
From harbhanu@huawei.com  Sun Mar  1 18:55:19 2009
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Date: Mon, 02 Mar 2009 08:25:29 +0530
From: harbhanu <harbhanu@huawei.com>
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To: sip-implementors@lists.cs.columbia.edu, =?iso-8859-1?Q?'I=F1aki_Baz_Castillo'?= <ibc@aliax.net>, sip@ietf.org
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Subject: Re: [Sip] [Sip-implementors] Tel URI query
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I actually wanted to use +3314529!****! as example.
But then as per your interpretation too it's not allowed.=20

Also, please refer to the specification below:
3GPP 23003-830: [Section 13.5	Public Service Identity (PSI)]
             A wildcarded PSI consists of a delimited regular expression
located either in the userinfo portion of the SIP URI or in the
telephone-subscriber portion of the Tel URI.=20

TISPAN 182 025: [Section 6.1.2 Identification]
             The wildcarded public user identity consists of a delimited
regular expression located in the telephone-subscriber portion of a tel =
URI
(e.g.         tel: +3314529!****!) or in the user portion of a SIP URI =
(e.g.
sip:*@example.com).

Our point of contention is that the TEL Uri is modified as per the text
given in above mentioned specifications.=20

So do we have any parallel draft in IETF to apply these modification to =
Tel
Uri ??
How are we supposed to handle scenarios like this ?



Regards,
Harbhanu

-----Original Message-----
From: I=F1aki Baz Castillo [mailto:ibc@aliax.net]=20
Sent: Saturday, February 28, 2009 6:19 PM
To: harbhanu@huawei.com
Cc: sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] Tel URI query

2009/2/27 harbhanu <harbhanu@huawei.com>:

> Example: The following PSI could be stored in the HSS -
> "sip:chatlist!.*!@example.com".
>
> "
>
> As per the above statements, the same example there would hold true =
for
tel
> uri too. Such as:
>
> tel:chatlist!*!@example.com
>
>
>
> But after referring to the ABNF defined by RFC-3966 I believe the same =
is
> not supported.


The above TEL uri is fully invalid. Basically a TEL uri can be:

- Local number:
tel: + 0123456789aBCDEF-,() + ;phone-context=3DXXXX

- Global number:
tel: + "+" 0123456789-,()

A tel URI number cannot contain other symbols. Also, note that in case
of a local number (so doesn't start by "+") then "phone-context"
parameter is mandatory.

Regards.


--=20
I=F1aki Baz Castillo
<ibc@aliax.net>


From HKaplan@acmepacket.com  Sun Mar  1 19:21:40 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Theo Zourzouvillys <theo@voip.co.uk>
Date: Sun, 1 Mar 2009 22:22:01 -0500
Thread-Topic: [Sip] SIP amplification vulnerability -01 (draft-zourzouvillys-sip-via-cookie-01)
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> -----Original Message-----
> From: theo@crazygreek.co.uk [mailto:theo@crazygreek.co.uk] On Behalf Of
> Theo Zourzouvillys
> Sent: Monday, February 23, 2009 5:49 AM
>
> On Sun, Feb 22, 2009 at 11:55 PM, Hadriel Kaplan <HKaplan@acmepacket.com>
> wrote:
>
> > Essentially what you're proposing seems to me similar in concept to
> syncookies, but at the next layer up.
>
> yeah, it is - in fact probably why i subconsciously called them via
> cookies :-)

So if the protection mechanism involved requires a change on the proxy and =
a change on the UA's, to do the via-cookie mechanism, then why do the chang=
e so high in the SIP stack?  I mean essentially what you'd want is a *reall=
y* efficient response/check mechanism, before even parsing the message (or =
parsing it enough to create a legal 4xx response).

One way to do that would be to create a fixed-length, binary header as the =
first N bytes of the UDP payload, to carry the support indication and the c=
hallenge and challenge-response.  You could even treat this as a shim-layer=
 between UDP and the SIP message layer (and below sigcomp if it's there).  =
That way the client side also doesn't have to process the cookie challenge =
at its SIP layer either, because the retransmission with the challenge resp=
onse is done by the shim layer.  And of course a fixed-length header send-a=
ble in the SIP port somewhat already exists: STUN.

Obviously in a way you're really just re-creating TCP or SCTP in that regar=
d, but that's what the via-cookie is doing anyway, just at a higher layer.

-hadriel

From shivalk_80@yahoo.co.in  Sun Mar  1 21:19:51 2009
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From: atul garg <shivalk_80@yahoo.co.in>
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Subject: [Sip] Frequency of Register Message ??
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Hi,

1) Is register message used as Heartbeat in any scenario???

2) Can register message be sent after every second???

3) Is it logical to send register message after every second???

4) what will be the drawbacks of sending register message after every secon=
d???

Regards
Atul=0A=0A=0A      Add more friends to your messenger and enjoy! Go to http=
://messenger.yahoo.com/invite/


From mahesh.awaradi@globaledgesoft.com  Sun Mar  1 22:12:23 2009
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Hi All,
what will happen if caller din't get ACK.whats the behaviour of caller.

Regards
Mahesh



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Subject: Re: [Sip] SIP amplification vulnerability -01 (draft-zourzouvillys-sip-via-cookie-01)
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On Mon, Mar 2, 2009 at 3:22 AM, Hadriel Kaplan <HKaplan@acmepacket.com> wrote:

Hi Hadriel,

> So if the protection mechanism involved requires a change on the proxy and a change on the UA's, to do the via-cookie mechanism, then why do the change so high in the SIP stack?

only for adoption reasons:

First, It seemed to big a change to push and in reality get adopted in
the "short" term by people as any other lower change requires more
than just a few lines of code for people that want to implement it
without caring about performance (for example, a SIP phone or
something else that is not worried about high performance).  DCCP
offers the same features at a level already implemented by most major
operating systems, as does DTLS - so we could use either of those and
the problem goes away, too with potentially less complexity for
implementers (at least in the DCCP case).

Second, there are already a scary number of ALGs on DSL/cable routers
and deployed out there - any non 3261 compliant method would likely
require many of those be updated as well, although with all the SIP
diagnostic tools like ethereal/wireshark and monitoring etc.

 ~ Theo

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Cc: "draft-ietf-sip-ua-privacy@tools.ietf.org" <draft-ietf-sip-ua-privacy@tools.ietf.org>, Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] WGLC for draft-ietf-sip-ua-privacy
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(As WG cochair)

A reminder of the current WGLC on this document which completes today.

Please submit your comments as soon as possible.

If you are reviewing the document, and you consider there may be a delay in=
 your comments appearing, then please send a mail to the chairs so they can=
 ascertain whether more time is needed.

regards

Keith

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of DRAGE, Keith (Keith)
> Sent: Tuesday, February 17, 2009 12:01 AM
> To: SIP List
> Cc: draft-ietf-sip-ua-privacy@tools.ietf.org; Dean Willis
> Subject: [Sip] WGLC for draft-ietf-sip-ua-privacy
>=20
> (As WG cochair)
>=20
> This mail is to initiate working group last call on=20
> draft-ietf-sip-ua-privacy which may be found here:
>=20
> http://tools.ietf.org/internet-drafts/draft-ietf-sip-ua-privacy-05.txt
>=20
> Please provide comments to both the authors and to the SIP WG=20
> list by Monday 2nd March 2009.
>=20
> The intended status of the document is informational.
>=20
> regards
>=20
> Keith
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
> =

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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: SIP IETF <sip@ietf.org>
Date: Mon, 2 Mar 2009 11:48:07 +0100
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(As WG cochair)

A reminder of the current WGLC on this document, which completes Wednesday.

Please submit your comments as soon as possible.

If you are reviewing the document, and you consider there may be a delay in=
 your comments appearing, then please send a mail to the chairs so they can=
 ascertain whether more time is needed.

regards

Keith=20

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]=20
> Sent: Monday, February 16, 2009 7:08 PM
> To: SIP IETF
> Cc: Eric Burger; DRAGE, Keith (Keith)
> Subject: WGLC on draft-ietf-sip-info-events
>=20
>=20
> I'd like to request the SIP Working Group perform final=20
> review on the INFO Events draft:
>=20
>=20
> http://tools.ietf.org/id/draft-ietf-sip-info-events-03.txt
>=20
>=20
>=20
> Please review this draft and bring any issues to the mailing=20
> list, copying the author.  Please conclude this review=20
> process by March 4, 2009.
>=20
> --
> Dean Willis
> cp-Chair, SIP WG
> =

From satyanarayana.t@motorola.com  Mon Mar  2 02:58:12 2009
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From: "T Satyanarayana-A12694" <satyanarayana.t@motorola.com>
To: =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>, <harbhanu@huawei.com>
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There MAY be no violation of specs here. These wildcard chars (! etc) =
are used only in the HSS or other nodes (only to optimize handling of =
multiple related URIs) and not in a SIP message?. Just a thought, not =
sure. Have a look at the following text from 23.003.

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
The public service identity shall take the form of either a SIP URI (see =
IETF RFC 3261 [26]) or a Tel URI (see IETF RFC 3966 [45]). A public =
service identity identifies a service, or a specific resource created =
for a service on an application server. The domain part is pre-defined =
by the IMS operators and the IMS system provides the flexibility to =
dynamically create the user part of the PSIs.

The PSIs are stored in the HSS either as a distinct PSI or as a =
wildcarded PSI. A distinct PSI contains the PSI that is used in routing =
, whilst a wildcarded PSI represents a collection of PSIs. Wildcarded =
PSIs enable optimisation of the operation and maintenance of the nodes. =
A wildcarded PSI consists of a delimited regular expression located =
either in the userinfo portion of the SIP URI or in the =
telephone-subscriber portion of the Tel URI. The regular expression in =
the wildcarded PSI shall take the form of Extended Regular Expressions =
(ERE) as defined in chapter 9 in IEEE 1003.1-2004 Part 1 [60]. The =
delimiter shall be the exclamation mark character ("!"). If more than =
two exclamation mark characters are present in the userinfo portion or =
telephone-subscriber portion of a wildcarded PSI then the outside pair =
of exclamation mark characters is regarded as the pair of delimiters =
(i.e. no exclamation mark characters are allowed to be present in the =
fixed parts of the userinfo portion or telephone-subscriber portion).

When stored in the HSS, the wildcarded PSI shall include the delimiter =
character to indicate the extent of the part of the PSI that is =
wildcarded.  It is used  to separate the regular expression from the =
fixed part of the wildcarded PSI.

Example: The following PSI could be stored in the HSS - =
"sip:chatlist!.*!@example.com".
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Regards
Satya T

-----Original Message-----
From: sip-implementors-bounces@lists.cs.columbia.edu =
[mailto:sip-implementors-bounces@lists.cs.columbia.edu] On Behalf Of =
I=F1aki Baz Castillo
Sent: Monday, March 02, 2009 2:28 PM
To: harbhanu@huawei.com
Cc: sip@ietf.org; sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] Tel URI query

2009/3/2 harbhanu <harbhanu@huawei.com>:
> I actually wanted to use +3314529!****! as example.
> But then as per your interpretation too it's not allowed.

Sure it's not allowed per RFC 3966 BNF syntax rules.


> TISPAN 182 025: [Section 6.1.2 Identification]
> =A0 =A0 =A0 =A0 =A0 =A0 The wildcarded public user identity consists =
of a=20
> delimited regular expression located in the telephone-subscriber=20
> portion of a tel URI (e.g. =A0 =A0 =A0 =A0 tel: +3314529!****!) or in =
the user portion of a SIP URI (e.g.
> sip:*@example.com).

This is annoying. It means that this specification violates the grammar =
specified in RFC 3966. Theorically RFC 3966 is just updated by RFC 5341. =
Sincerelly I can't understand how the above spec can requiere such =
aberration in Tel URI.


> So do we have any parallel draft in IETF to apply these modification=20
> to Tel Uri ??
> How are we supposed to handle scenarios like this ?

Sorry, no idea, this is really confusing. Do people writting =
specifications really want TEL URI to be usable??



--
I=F1aki Baz Castillo
<ibc@aliax.net>

_______________________________________________
Sip-implementors mailing list
Sip-implementors@lists.cs.columbia.edu
https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.


	Title           : Response Code for Indication of Terminated Dialog
	Author(s)       : C. Holmberg
	Filename        : draft-ietf-sip-199-06.txt
	Pages           : 14
	Date            : 2009-03-02

This specification defines a new SIP response code, 199 Early Dialog
Terminated, which a SIP forking proxy and a UAS can use to indicate
upstream towards the UAC that an early dialog has been terminated,
before a final response is sent towards the UAC.

A URL for this Internet-Draft is:
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Subject: Re: [Sip] Frequency of Register Message ??
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Do you mean these questions about whether they're used in an IETF RFC?  Thi=
s is the sip mailing list for IETF RFC/drafts discussion - you're probably =
looking for sip-implementors@lists.cs.columbia.edu - see https://lists.cs.c=
olumbia.edu/cucslists/listinfo/sip-implementors.

I assume you mean in practice, not theory.

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of atu=
l
> garg
> Sent: Monday, March 02, 2009 12:20 AM
>
> 1) Is register message used as Heartbeat in any scenario???

In an RFC no, but in practice yes.  In fact it's widely used as a Heartbeat=
 in practice.  But for a specific case: endpoints or small PBX's to their r=
egistrar/proxy.  Not for proxy-to-proxy or server-to-server; OPTIONS messag=
es are often used for the server-to-server type of Heartbeat in practice.


> 2) Can register message be sent after every second???

You mean can it be sent every second?  Well technically you could of course=
, but in practice that's not a good idea.  There are some devices which wil=
l blacklist/block your UA if it sends one that frequently, disabling servic=
e to your user.  It would be considered an attack or at least a broken impl=
ementation.  Your code would essentially be considered malware.


> 3) Is it logical to send register message after every second???

No, not really.  There is no real reason to do so.  NATs, for example, don'=
t expire their bindings nearly that fast.  The worst I've seen a NAT do is =
about 30 seconds, but some are longer (much longer).  So you only need to s=
end one every 15 or 20 seconds worst-case.  But really you should just use =
the Expires header/param in the 200 ok response to the first Register, for =
the timer/frequency value for Register messages. (in other words follow RFC=
 3261 section 10.2.1 - 10.2.4, and see 10.3)


> 4) what will be the drawbacks of sending register message after every
> second???

Well, for starters the processing load on your Registrar or proxy will be v=
ery high.  Your UA is one of potentially hundreds of thousands or millions =
of endpoints on a Registrar.  If each UA sent a Register every second, it w=
ouldn't scale.

And if you implement your stack in a full layered model, then your transact=
ion layer could also be sending retransmissions while you're trying to send=
 new ones - resulting in either out of order sending, or at least a possibl=
e pile up of retransmissions for multiple Registers if the network's out fo=
r a dozen seconds or so.

-hadriel

From christer.holmberg@ericsson.com  Mon Mar  2 04:58:20 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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Subject: [Sip] New version (-06) of the 199 draft
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Hi,

I received some editorial comments on the security impact chapter of the
draft, so I fixed those and submitted a new version of the draft.

It can also be found at:

http://users.piuha.net/cholmber/drafts/draft-ietf-sip-199-06.txt

Regards,

Christer



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<P><FONT SIZE=3D2 FACE=3D"Arial">Hi,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">I received some editorial comments on =
the security impact chapter of the draft, so I fixed those and submitted =
a new version of the draft.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">It can also be found at:</FONT>
</P>

<P><A =
HREF=3D"http://users.piuha.net/cholmber/drafts/draft-ietf-sip-199-06.txt"=
><U><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Arial">http://users.piuha.net/cholmber/drafts/draft-ietf-sip-199-=
06.txt</FONT></U></A>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Regards,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Christer</FONT>
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From brett@broadsoft.com  Mon Mar  2 05:19:25 2009
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From: Brett Tate <brett@broadsoft.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, SIP IETF <sip@ietf.org>
Date: Mon, 2 Mar 2009 05:18:37 -0800
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Subject: Re: [Sip] WGLC on draft-ietf-sip-info-events
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> A reminder of the current WGLC on this document, which completes
> Wednesday.
>=20
> Please submit your comments as soon as possible.
>=20
> If you are reviewing the document, and you consider there may be a delay
> in your comments appearing, then please send a mail to the chairs so they
> can ascertain whether more time is needed.

I haven't received a reply concerning my February 16 (archived on 17) email=
: "draft-ietf-sip-info-events-03: comments and questions".  Thus I'm not su=
re if I will have follow-up comments based upon the reply.

http://www.ietf.org/mail-archive/web/sip/current/msg26621.html

The following is a snapshot of my questions and comments in case the author=
s did not see my prior email.

Thanks,
Brett

----

Section 3.1 paragraph 1:  Concerning "MUST advertise" and "provisional 1xx"=
, please clarify that the "MUST advertise" is not applicable to the 100 res=
ponse.  Does the "MUST advertise" and "provisional 1xx" apply to all (first=
 and subsequent within dialog) INVITE non 101-199 (intentionally indicated =
199)?  Concerning ACK for initial INVITE's 2xx, does the "MUST advertise" a=
lways apply?

Section 3.1 paragraph 2 sentences 1 and 2: Sentence 1 indicates "session pa=
rameters" and sentence 2 indicates "dialog parameters" concerning Info Pack=
age negotiation, I'm not sure when the negotiation/renegotiation occurs.  D=
oes it follow SDP offer/answer rules or something else; if something else, =
what?  I have the same question concerning "dialog refresh" text discussed =
later within the same section since incorrect interpretations might trigger=
 deactivation.

Section 4.3: When sending a 489, it indicates "MUST terminate the INVITE di=
alog".  The "MUST terminate" seems severe even if all the negotiation race =
conditions are adequately addressed.  The mentioned "protocol failure" reas=
on seems contrary to the "strictly send and lenient receive" mantra.

Section 5.1 Table 1: The usage of "g" seems potentially outdated or incorre=
ctly reference since discussed within RFC 2543 instead of RFC 3261.  The sa=
me applies to usage Via row concerning "(2)".

Section 5.2.2: The "general-header" reference is outdated or incorrectly re=
ference since defined within RFC 2543 instead of RFC 3261.  RFC 3261 uses "=
Message-Header".

Section 7.7 paragraph 2: The following potentially could be reworded since =
it is okay to return failure responses (401, 503, 500, ...) even when overl=
apping occurs: "in the event a UAC issues overlapping INFO requests for an =
Info Package, the UAS MUST NOT return an error response".

Section 8: Why specifically defining and using "nil"?  I'm mainly just curi=
ous of the benefit over not having defined "nil" for similar use within the=
 Supported header.


From Mike.Szilagyi@inin.com  Mon Mar  2 06:20:42 2009
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From: "Szilagyi, Mike" <Mike.Szilagyi@inin.com>
To: "SIP@ietf.org" <SIP@ietf.org>
Date: Mon, 2 Mar 2009 09:20:59 -0500
Thread-Topic: Question regarding conflicting grammar for IPV6 SIP URI and RFC 3986
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Subject: Re: [Sip] Question regarding conflicting grammar for IPV6 SIP URI and	RFC 3986
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Could someone let me know where this discussion has been moved?

Regards,
Mike

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Szila=
gyi, Mike
Sent: Thursday, December 18, 2008 10:53 AM
To: SIP@ietf.org
Subject: [Sip] Question regarding conflicting grammar for IPV6 SIP URI and =
RFC 3986

RFC 3986 defines the following grammar for a URI:

   URI           =3D scheme ":" hier-part [ "?" query ] [ "#" fragment ]

   hier-part     =3D "//" authority path-abempty
                 / path-absolute
                 / path-rootless
                 / path-empty

A SIP URI falls into the 'path-rootless' rule:


   path-rootless =3D segment-nz *( "/" segment )


   segment-nz    =3D 1*pchar


   pchar         =3D unreserved / pct-encoded / sub-delims / ":" / "@"


   pct-encoded   =3D "%" HEXDIG HEXDIG



   unreserved    =3D ALPHA / DIGIT / "-" / "." / "_" / "~"


   sub-delims    =3D "!" / "$" / "&" / "'" / "(" / ")"

                 / "*" / "+" / "," / ";" / "=3D"

path-rootless does not allow "[" or "]" characters, so the SIP-URI grammar =
for IPV6 violates RFC 3986 grammar.

SIP-URI          =3D  "sip:" [ userinfo ] hostport
                    uri-parameters [ headers ]

hostport         =3D  host [ ":" port ]
host             =3D  hostname / IPv4address / IPv6reference
hostname         =3D  *( domainlabel "." ) toplabel [ "." ]
domainlabel      =3D  alphanum
                    / alphanum *( alphanum / "-" ) alphanum
toplabel         =3D  ALPHA / ALPHA *( alphanum / "-" ) alphanum
IPv4address    =3D  1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT
IPv6reference  =3D  "[" IPv6address "]"  <-- *** violates RFC 3986 ***
IPv6address    =3D  hexpart [ ":" IPv4address ]
hexpart        =3D  hexseq / hexseq "::" [ hexseq ] / "::" [ hexseq ]
hexseq         =3D  hex4 *( ":" hex4)
hex4           =3D  1*4HEXDIG
port           =3D  1*DIGIT


These, though valid SIP URIs are not valid URIs:

    <sip:[2001:db8::10]>
    <sip:caller@[2001:db8::1]>

Has this already been addressed?

Regards,
Mike

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<div class=3DSection1>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Could someone let me kno=
w where
this discussion has been moved?<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Regards,<o:p></o:p></spa=
n></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Mike<o:p></o:p></span></=
p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in'>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma=
","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <b>On Behalf Of </b>Szil=
agyi,
Mike<br>
<b>Sent:</b> Thursday, December 18, 2008 10:53 AM<br>
<b>To:</b> SIP@ietf.org<br>
<b>Subject:</b> [Sip] Question regarding conflicting grammar for IPV6 SIP U=
RI
and RFC 3986<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>RFC 3986 defines the following grammar for a URI:<o:p>=
</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal style=3D'background:white'><span style=3D'font-size:10=
.0pt;
font-family:"Courier New";color:black'>&nbsp;&nbsp;
URI&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D scheme
&quot;:&quot; hier-part [ &quot;?&quot; query ] [ &quot;#&quot; fragment ]<=
o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'background:white'><span style=3D'font-size:10=
.0pt;
font-family:"Courier New";color:black'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>&nbsp; &nbsp;hier-part&nbsp;&nbsp; &nbsp;&nbsp;=
=3D
&quot;//&quot; authority path-abempty<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
&nbsp;&nbsp;&nbsp;/ path-absolute<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
&nbsp;&nbsp;&nbsp;/ path-rootless<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
&nbsp;&nbsp;&nbsp;/ path-empty<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'>A SIP URI falls into the
&#8216;path-rootless&#8217; rule:<span style=3D'font-size:10.0pt;font-famil=
y:
"Courier New"'><o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif";
color:blue'><o:p>&nbsp;</o:p></span></p>

<pre style=3D'background:white'><span style=3D'color:black'>&nbsp;&nbsp; pa=
th-rootless =3D segment-nz *( &quot;/&quot; segment )<o:p></o:p></span></pr=
e>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif";
color:blue'><o:p>&nbsp;</o:p></span></p>

<pre style=3D'background:white'><span style=3D'color:black'>&nbsp;&nbsp; se=
gment-nz&nbsp;&nbsp;&nbsp; =3D 1*pchar<o:p></o:p></span></pre>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif";
color:blue'><o:p>&nbsp;</o:p></span></p>

<pre style=3D'background:white'><span style=3D'color:black'>&nbsp;&nbsp; pc=
har&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;=3D unreserved / pct-en=
coded / sub-delims / &quot;:&quot; / &quot;@&quot;<o:p></o:p></span></pre>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif";
color:blue'><o:p>&nbsp;</o:p></span></p>

<pre style=3D'background:white'><span style=3D'color:black'>&nbsp;&nbsp; pc=
t-encoded&nbsp;&nbsp; =3D &quot;%&quot; HEXDIG HEXDIG<o:p></o:p></span></pr=
e><pre
style=3D'background:white'><span style=3D'color:black'><o:p>&nbsp;</o:p></s=
pan></pre><pre
style=3D'background:white'><span style=3D'color:black'>&nbsp;&nbsp; unreser=
ved&nbsp;&nbsp;&nbsp; =3D ALPHA / DIGIT / &quot;-&quot; / &quot;.&quot; / &=
quot;_&quot; / &quot;~&quot;<o:p></o:p></span></pre>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif";
color:blue'><o:p>&nbsp;</o:p></span></p>

<pre style=3D'background:white'><span style=3D'color:black'>&nbsp;&nbsp; su=
b-delims&nbsp;&nbsp;&nbsp; =3D &quot;!&quot; / &quot;$&quot; / &quot;&amp;&=
quot; / &quot;'&quot; / &quot;(&quot; / &quot;)&quot;<o:p></o:p></span></pr=
e><pre
style=3D'background:white'><span style=3D'color:black'>&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
; / &quot;*&quot; / &quot;+&quot; / &quot;,&quot; / &quot;;&quot; / &quot;=
=3D&quot;<o:p></o:p></span></pre>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>path-rootless does not allow &#8220;[&#8220; or
&#8220;]&#8221; characters, so the SIP-URI grammar for IPV6 violates RFC 39=
86
grammar.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>SIP-URI&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;
=3D&nbsp; &quot;sip:&quot; [ userinfo ] hostport<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
uri-parameters [ headers ]<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>hostport&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;
=3D&nbsp; host [ &quot;:&quot; port ]<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>host&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;
=3D&nbsp; hostname / IPv4address / IPv6reference<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>hostname&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;
=3D&nbsp; *( domainlabel &quot;.&quot; ) toplabel [ &quot;.&quot; ]<o:p></o=
:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>domainlabel&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D&nb=
sp;
alphanum<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
/ alphanum *( alphanum / &quot;-&quot; ) alphanum<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>toplabel&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;
=3D&nbsp; ALPHA / ALPHA *( alphanum / &quot;-&quot; ) alphanum<o:p></o:p></=
span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>IPv4address&nbsp;&nbsp;&nbsp; =3D&nbsp; 1*3DIGIT
&quot;.&quot; 1*3DIGIT &quot;.&quot; 1*3DIGIT &quot;.&quot; 1*3DIGIT<o:p></=
o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New";color:red'>IPv6reference&nbsp; =3D&nbsp; &quot;[&=
quot;
IPv6address &quot;]&quot;&nbsp; &lt;-- *** violates RFC 3986 ***<o:p></o:p>=
</span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>IPv6address&nbsp;&nbsp;&nbsp; =3D&nbsp; hexpart =
[
&quot;:&quot; IPv4address ]<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>hexpart&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;
=3D&nbsp; hexseq / hexseq &quot;::&quot; [ hexseq ] / &quot;::&quot; [ hexs=
eq ]<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>hexseq&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;
=3D&nbsp; hex4 *( &quot;:&quot; hex4)<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>hex4&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;
=3D&nbsp; 1*4HEXDIG<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>port&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;
=3D&nbsp; 1*DIGIT<o:p></o:p></span></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>These, though valid SIP URIs are not valid URIs:<o:p><=
/o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New";color:red'>&nbsp;&nbsp;&nbsp;
&lt;sip:[2001:db8::10]&gt;<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New";color:red'>&nbsp;&nbsp;&nbsp;
&lt;sip:caller@[2001:db8::1]&gt;<o:p></o:p></span></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Has this already been addressed?<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Regards,<o:p></o:p></p>

<p class=3DMsoNormal>Mike<o:p></o:p></p>

</div>

</body>

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From: Brett Tate <brett@broadsoft.com>
To: "Szilagyi, Mike" <Mike.Szilagyi@inin.com>, "SIP@ietf.org" <SIP@ietf.org>
Date: Mon, 2 Mar 2009 06:37:23 -0800
Thread-Topic: Question regarding conflicting grammar for IPV6 SIP URI and RFC 3986
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Subject: Re: [Sip] Question regarding conflicting grammar for IPV6 SIP URI and	RFC 3986
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Cullen's reply killed or moved the discussion.

http://www.ietf.org/mail-archive/web/sip/current/msg26385.html



"WIth my AD hat on ...



If you want to update/discuss/change/abuse 3986, probably the best place to=
 to that would be on the APPS area list at apps-discuss@ietf.org<mailto:app=
s-discuss@ietf.org>



It would be up to the Apps ADs, but I suspect that change to the BNF sugges=
ted below is too significant a change to be done with an Errata.



Cullen"


From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Szila=
gyi, Mike
Sent: Monday, March 02, 2009 9:21 AM
To: SIP@ietf.org
Subject: Re: [Sip] Question regarding conflicting grammar for IPV6 SIP URI =
and RFC 3986

Could someone let me know where this discussion has been moved?

Regards,
Mike

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Szila=
gyi, Mike
Sent: Thursday, December 18, 2008 10:53 AM
To: SIP@ietf.org
Subject: [Sip] Question regarding conflicting grammar for IPV6 SIP URI and =
RFC 3986

RFC 3986 defines the following grammar for a URI:

   URI           =3D scheme ":" hier-part [ "?" query ] [ "#" fragment ]

   hier-part     =3D "//" authority path-abempty
                 / path-absolute
                 / path-rootless
                 / path-empty

A SIP URI falls into the 'path-rootless' rule:


   path-rootless =3D segment-nz *( "/" segment )


   segment-nz    =3D 1*pchar


   pchar         =3D unreserved / pct-encoded / sub-delims / ":" / "@"


   pct-encoded   =3D "%" HEXDIG HEXDIG



   unreserved    =3D ALPHA / DIGIT / "-" / "." / "_" / "~"


   sub-delims    =3D "!" / "$" / "&" / "'" / "(" / ")"

                 / "*" / "+" / "," / ";" / "=3D"

path-rootless does not allow "[" or "]" characters, so the SIP-URI grammar =
for IPV6 violates RFC 3986 grammar.

SIP-URI          =3D  "sip:" [ userinfo ] hostport
                    uri-parameters [ headers ]

hostport         =3D  host [ ":" port ]
host             =3D  hostname / IPv4address / IPv6reference
hostname         =3D  *( domainlabel "." ) toplabel [ "." ]
domainlabel      =3D  alphanum
                    / alphanum *( alphanum / "-" ) alphanum
toplabel         =3D  ALPHA / ALPHA *( alphanum / "-" ) alphanum
IPv4address    =3D  1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT
IPv6reference  =3D  "[" IPv6address "]"  <-- *** violates RFC 3986 ***
IPv6address    =3D  hexpart [ ":" IPv4address ]
hexpart        =3D  hexseq / hexseq "::" [ hexseq ] / "::" [ hexseq ]
hexseq         =3D  hex4 *( ":" hex4)
hex4           =3D  1*4HEXDIG
port           =3D  1*DIGIT


These, though valid SIP URIs are not valid URIs:

    <sip:[2001:db8::10]>
    <sip:caller@[2001:db8::1]>

Has this already been addressed?

Regards,
Mike

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<div class=3DSection1>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew";
color:#1F497D'>Cullen's reply killed or moved the discussion.<o:p></o:p></s=
pan></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew";
color:#1F497D'><a
href=3D"http://www.ietf.org/mail-archive/web/sip/current/msg26385.html">htt=
p://www.ietf.org/mail-archive/web/sip/current/msg26385.html</a><o:p></o:p><=
/span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoPlainText>&quot;WIth my AD hat on ...<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>If you want to update/discuss/change/abuse 3986, pr=
obably
the best place to to that would be on the APPS area list at <a
href=3D"mailto:apps-discuss@ietf.org">apps-discuss@ietf.org</a><o:p></o:p><=
/p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>It would be up to the Apps ADs, but I suspect that =
change
to the BNF suggested below is too significant a change to be done with an
Errata.<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>Cullen&quot;<o:p></o:p></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in'>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma=
","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <b>On Behalf Of </b>Szil=
agyi,
Mike<br>
<b>Sent:</b> Monday, March 02, 2009 9:21 AM<br>
<b>To:</b> SIP@ietf.org<br>
<b>Subject:</b> Re: [Sip] Question regarding conflicting grammar for IPV6 S=
IP
URI and RFC 3986<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Could someone let me kno=
w where
this discussion has been moved?<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Regards,<o:p></o:p></spa=
n></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Mike<o:p></o:p></span></=
p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in'>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma=
","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <b>On Behalf Of </b>Szil=
agyi,
Mike<br>
<b>Sent:</b> Thursday, December 18, 2008 10:53 AM<br>
<b>To:</b> SIP@ietf.org<br>
<b>Subject:</b> [Sip] Question regarding conflicting grammar for IPV6 SIP U=
RI
and RFC 3986<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>RFC 3986 defines the following grammar for a URI:<o:p>=
</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal style=3D'background:white'><span style=3D'font-size:10=
.0pt;
font-family:"Courier New";color:black'>&nbsp;&nbsp;
URI&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D scheme
&quot;:&quot; hier-part [ &quot;?&quot; query ] [ &quot;#&quot; fragment ]<=
o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'background:white'><span style=3D'font-size:10=
.0pt;
font-family:"Courier New";color:black'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>&nbsp; &nbsp;hier-part&nbsp;&nbsp; &nbsp;&nbsp;=
=3D
&quot;//&quot; authority path-abempty<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
&nbsp;&nbsp;&nbsp;/ path-absolute<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
&nbsp;&nbsp;&nbsp;/ path-rootless<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
&nbsp;&nbsp;&nbsp;/ path-empty<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'>A SIP URI falls into the
&#8216;path-rootless&#8217; rule:<span style=3D'font-size:10.0pt;font-famil=
y:
"Courier New"'><o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif";
color:blue'><o:p>&nbsp;</o:p></span></p>

<pre style=3D'background:white'><span style=3D'color:black'>&nbsp;&nbsp; pa=
th-rootless =3D segment-nz *( &quot;/&quot; segment )<o:p></o:p></span></pr=
e>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif";
color:blue'><o:p>&nbsp;</o:p></span></p>

<pre style=3D'background:white'><span style=3D'color:black'>&nbsp;&nbsp; se=
gment-nz&nbsp;&nbsp;&nbsp; =3D 1*pchar<o:p></o:p></span></pre>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif";
color:blue'><o:p>&nbsp;</o:p></span></p>

<pre style=3D'background:white'><span style=3D'color:black'>&nbsp;&nbsp; pc=
har&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;=3D unreserved / pct-en=
coded / sub-delims / &quot;:&quot; / &quot;@&quot;<o:p></o:p></span></pre>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif";
color:blue'><o:p>&nbsp;</o:p></span></p>

<pre style=3D'background:white'><span style=3D'color:black'>&nbsp;&nbsp; pc=
t-encoded&nbsp;&nbsp; =3D &quot;%&quot; HEXDIG HEXDIG<o:p></o:p></span></pr=
e><pre
style=3D'background:white'><span style=3D'color:black'><o:p>&nbsp;</o:p></s=
pan></pre><pre
style=3D'background:white'><span style=3D'color:black'>&nbsp;&nbsp; unreser=
ved&nbsp;&nbsp;&nbsp; =3D ALPHA / DIGIT / &quot;-&quot; / &quot;.&quot; / &=
quot;_&quot; / &quot;~&quot;<o:p></o:p></span></pre>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif";
color:blue'><o:p>&nbsp;</o:p></span></p>

<pre style=3D'background:white'><span style=3D'color:black'>&nbsp;&nbsp; su=
b-delims&nbsp;&nbsp;&nbsp; =3D &quot;!&quot; / &quot;$&quot; / &quot;&amp;&=
quot; / &quot;'&quot; / &quot;(&quot; / &quot;)&quot;<o:p></o:p></span></pr=
e><pre
style=3D'background:white'><span style=3D'color:black'>&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
; / &quot;*&quot; / &quot;+&quot; / &quot;,&quot; / &quot;;&quot; / &quot;=
=3D&quot;<o:p></o:p></span></pre>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>path-rootless does not allow &#8220;[&#8220; or
&#8220;]&#8221; characters, so the SIP-URI grammar for IPV6 violates RFC 39=
86
grammar.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>SIP-URI&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;
=3D&nbsp; &quot;sip:&quot; [ userinfo ] hostport<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
uri-parameters [ headers ]<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>hostport&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;
=3D&nbsp; host [ &quot;:&quot; port ]<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>host&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;
=3D&nbsp; hostname / IPv4address / IPv6reference<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>hostname&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;
=3D&nbsp; *( domainlabel &quot;.&quot; ) toplabel [ &quot;.&quot; ]<o:p></o=
:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>domainlabel&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D&nb=
sp;
alphanum<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
/ alphanum *( alphanum / &quot;-&quot; ) alphanum<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>toplabel&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;
=3D&nbsp; ALPHA / ALPHA *( alphanum / &quot;-&quot; ) alphanum<o:p></o:p></=
span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>IPv4address&nbsp;&nbsp;&nbsp; =3D&nbsp; 1*3DIGIT
&quot;.&quot; 1*3DIGIT &quot;.&quot; 1*3DIGIT &quot;.&quot; 1*3DIGIT<o:p></=
o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New";color:red'>IPv6reference&nbsp; =3D&nbsp; &quot;[&=
quot;
IPv6address &quot;]&quot;&nbsp; &lt;-- *** violates RFC 3986 ***<o:p></o:p>=
</span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>IPv6address&nbsp;&nbsp;&nbsp; =3D&nbsp; hexpart =
[
&quot;:&quot; IPv4address ]<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>hexpart&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;
=3D&nbsp; hexseq / hexseq &quot;::&quot; [ hexseq ] / &quot;::&quot; [ hexs=
eq ]<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>hexseq&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;
=3D&nbsp; hex4 *( &quot;:&quot; hex4)<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>hex4&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;
=3D&nbsp; 1*4HEXDIG<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New"'>port&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;
=3D&nbsp; 1*DIGIT<o:p></o:p></span></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>These, though valid SIP URIs are not valid URIs:<o:p><=
/o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New";color:red'>&nbsp;&nbsp;&nbsp;
&lt;sip:[2001:db8::10]&gt;<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span style=3D'font-size=
:10.0pt;
font-family:"Courier New";color:red'>&nbsp;&nbsp;&nbsp;
&lt;sip:caller@[2001:db8::1]&gt;<o:p></o:p></span></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Has this already been addressed?<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Regards,<o:p></o:p></p>

<p class=3DMsoNormal>Mike<o:p></o:p></p>

</div>

</div>

</body>

</html>

--_000_9B2A061A1137254BBE4F4B2CD843646A10B1FAA3F8mbx02citserve_--

From john.elwell@siemens.com  Mon Mar  2 09:24:17 2009
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Date: Mon, 02 Mar 2009 17:23:25 +0000
From: "Elwell, John" <john.elwell@siemens.com>
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To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, SIP List <SIP@ietf.org>
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Cc: draft-ietf-sip-ua-privacy@tools.ietf.org, Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] WGLC for draft-ietf-sip-ua-privacy
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1. In 4.1:
"If revealing the domain name in the
   Contact header field is a concern, usage of a third-party GRUU server
   to obtain a temp-gruu that is irrelevant to users' domain, which is
   outside the scope of this document SHOULD be considered."
"SHOULD be considered" is not a testable normative statement. It is not
the user's domain that is outside the scope, but usage of a third-party
GRUU server. The word "irrelevant" in inappropriate here. To take care
of these concerns, I think it should say:
"If revealing the domain name in the  Contact header field is a concern,
use of a third party GRUU server is a possible solution, but this is
outside the scope of this document."=20

2. In 5.1.1:
"For out-of-dialog
   request, anonymous URI MAY be set when anonymization is required."
What is the purpose of this statement? I assume out-of-dialog does not
include dialog-forming (which is covered earlier in the paragraph). For
a completely out-of-dialog request such as a MESSAGE request, why would
one include Contact? For REGISTER, one certainly cannot use an anonymous
URI.

John


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of DRAGE, Keith (Keith)
> Sent: 02 March 2009 10:47
> To: SIP List
> Cc: draft-ietf-sip-ua-privacy@tools.ietf.org; Dean Willis
> Subject: Re: [Sip] WGLC for draft-ietf-sip-ua-privacy
>=20
> (As WG cochair)
>=20
> A reminder of the current WGLC on this document which completes today.
>=20
> Please submit your comments as soon as possible.
>=20
> If you are reviewing the document, and you consider there may=20
> be a delay in your comments appearing, then please send a=20
> mail to the chairs so they can ascertain whether more time is needed.
>=20
> regards
>=20
> Keith
>=20
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> > Behalf Of DRAGE, Keith (Keith)
> > Sent: Tuesday, February 17, 2009 12:01 AM
> > To: SIP List
> > Cc: draft-ietf-sip-ua-privacy@tools.ietf.org; Dean Willis
> > Subject: [Sip] WGLC for draft-ietf-sip-ua-privacy
> >=20
> > (As WG cochair)
> >=20
> > This mail is to initiate working group last call on=20
> > draft-ietf-sip-ua-privacy which may be found here:
> >=20
> >=20
> http://tools.ietf.org/internet-drafts/draft-ietf-sip-ua-privacy-05.txt
> >=20
> > Please provide comments to both the authors and to the SIP WG=20
> > list by Monday 2nd March 2009.
> >=20
> > The intended status of the document is informational.
> >=20
> > regards
> >=20
> > Keith
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip=20
> > Use sipping@ietf.org for new developments on the application of sip
> >=20
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

From Bala_Neelakantan@Quintum.com  Mon Mar  2 10:09:45 2009
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From: Neelakantan Balasubramanian <Bala_Neelakantan@Quintum.com>
To: Mahesh <mahesh.awaradi@globaledgesoft.com>, "sip@ietf.org" <sip@ietf.org>
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Please read RFC 3261.  The caller generates INVITE.  The caller on receipt =
of final response, generates ACK.

Please direct questions like this on SIP implementers list Sip-implementors=
@lists.cs.columbia.edu



> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Mahesh
> Sent: Monday, March 02, 2009 12:11 AM
> To: sip@ietf.org
> Subject: [Sip] ACK response
>
> Hi All,
> what will happen if caller din't get ACK.whats the behaviour of caller.
>
> Regards
> Mahesh
>
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

From dean.willis@softarmor.com  Mon Mar  2 12:04:59 2009
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Subject: [Sip] Connection Reuse draft to focus on TLS connections
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We intend to slightly rescope the Connection Reuse draft:

http://tools.ietf.org/id/draft-ietf-sip-connect-reuse-12.txt

Based on input from our ADs, the WG chairs and the editor of the have  
agreed to eliminate the part of the draft that discusses the  
implementation of reusing non-TLS connections. The stage for this is  
set in the abstract of the current draft, from which I quote:

> From the security perspective, it is bad practice to reuse a single  
> connection for the TCP or SCTP transport between two peers, and this  
> document provides specific insights into why this is the case. As a  
> remedy, it suggests using two TCP connections (or two SCTP  
> associations), each opened pro-actively towards the recipient by the  
> sender.


We may  discuss at a later date the idea of developing a draft for  
connection reuse in the absence of TLS.

I'm really not expecting anybody to object strongly to this approach,  
but am bringing this to the list to avoid surprises down the road. If  
you have major issues with doing it this way, please bring them up now.

--
Dean Willis (as chair, ten years and counting . . .)



From christer.holmberg@ericsson.com  Mon Mar  2 12:12:33 2009
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Hi,

Just a question: does this rescope affect chapter 8 (Overlap with
connection reuse) of the keep draft
(http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-keep-03.txt)?


Regards,

Christer

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Dean Willis
Sent: Monday, March 02, 2009 10:04 PM
To: SIP WG
Cc: Cullen Jennings; Keith Drage
Subject: [Sip] Connection Reuse draft to focus on TLS connections


We intend to slightly rescope the Connection Reuse draft:

http://tools.ietf.org/id/draft-ietf-sip-connect-reuse-12.txt

Based on input from our ADs, the WG chairs and the editor of the have
agreed to eliminate the part of the draft that discusses the
implementation of reusing non-TLS connections. The stage for this is set
in the abstract of the current draft, from which I quote:

> From the security perspective, it is bad practice to reuse a single=20
> connection for the TCP or SCTP transport between two peers, and this=20
> document provides specific insights into why this is the case. As a=20
> remedy, it suggests using two TCP connections (or two SCTP=20
> associations), each opened pro-actively towards the recipient by the=20
> sender.


We may  discuss at a later date the idea of developing a draft for
connection reuse in the absence of TLS.

I'm really not expecting anybody to object strongly to this approach,
but am bringing this to the list to avoid surprises down the road. If
you have major issues with doing it this way, please bring them up now.

--
Dean Willis (as chair, ten years and counting . . .)


_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol Use
sip-implementors@cs.columbia.edu for questions on current sip Use
sipping@ietf.org for new developments on the application of sip

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Subject: [Sip] Fwd: I-D Action:draft-sparks-sip-3261-norm-ref-status-00.txt
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I split this out of the steps-to-draft document for separate tracking

Begin forwarded message:

> From: Internet-Drafts@ietf.org
> Date: March 2, 2009 8:30:02 AM CST
> To: i-d-announce@ietf.org
> Subject: I-D Action:draft-sparks-sip-3261-norm-ref-status-00.txt
> Reply-To: internet-drafts@ietf.org
>
> A New Internet-Draft is available from the on-line Internet-Drafts  
> directories.
>
> 	Title           : Status of Normative References in RFC3261
> 	Author(s)       : R. Sparks
> 	Filename        : draft-sparks-sip-3261-norm-ref-status-00.txt
> 	Pages           : 6
> 	Date            : 2009-03-02
>
> This document captures the current status of the normative references
> in RFC3261.  It is intended to inform continuing discussions on how
> to maintain the SIP protocol.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-sparks-sip-3261-norm-ref-status-00.txt
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; ">I split this out of the =
steps-to-draft document for separate tracking<br><div><br><div>Begin =
forwarded message:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><b>From: =
</b></font><font face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px =
Helvetica"><a =
href=3D"mailto:Internet-Drafts@ietf.org">Internet-Drafts@ietf.org</a></fon=
t></div><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: =
0px; margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" =
color=3D"#000000" style=3D"font: 12.0px Helvetica; color: =
#000000"><b>Date: </b></font><font face=3D"Helvetica" size=3D"3" =
style=3D"font: 12.0px Helvetica">March 2, 2009 8:30:02 AM =
CST</font></div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><font face=3D"Helvetica" =
size=3D"3" color=3D"#000000" style=3D"font: 12.0px Helvetica; color: =
#000000"><b>To: </b></font><font face=3D"Helvetica" size=3D"3" =
style=3D"font: 12.0px Helvetica"><a =
href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a></font></di=
v><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><b>Subject: =
</b></font><font face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px =
Helvetica"><b>I-D =
Action:draft-sparks-sip-3261-norm-ref-status-00.txt<span =
class=3D"Apple-converted-space">&nbsp;</span></b></font></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><b>Reply-To: =
</b></font><font face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px =
Helvetica"><a =
href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a></fon=
t></div><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: =
0px; margin-left: 0px; min-height: 14px; "><br></div> </div><div>A New =
Internet-Draft is available from the on-line Internet-Drafts =
directories.<br><br><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>Title =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: Status of =
Normative References in RFC3261<br><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>Author(s) =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: R. Sparks<br><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>Filename =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: =
draft-sparks-sip-3261-norm-ref-status-00.txt<br><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>Pages =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: 6<br><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>Date =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: =
2009-03-02<br><br>This document captures the current status of the =
normative references<br>in RFC3261. &nbsp;It is intended to inform =
continuing discussions on how<br>to maintain the SIP protocol.<br><br>A =
URL for this Internet-Draft is:<br><a =
href=3D"http://www.ietf.org/internet-drafts/draft-sparks-sip-3261-norm-ref=
-status-00.txt">http://www.ietf.org/internet-drafts/draft-sparks-sip-3261-=
norm-ref-status-00.txt</a><br><br>Internet-Drafts are also available by =
anonymous FTP at:<br>ftp://ftp.ietf.org/internet-drafts/<br><br>Below is =
the data which will enable a MIME compliant mail =
reader<br>implementation to automatically retrieve the ASCII version of =
the<br>Internet-Draft.<br></div></blockquote></div></body></html>=

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<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div><blockquote type="cite"><div>_______________________________________________<br>I-D-Announce mailing list<br>I-D-Announce@ietf.org<br>https://www.ietf.org/mailman/listinfo/i-d-announce<br>Internet-Draft directories: http://www.ietf.org/shadow.html<br>or ftp://ftp.ietf.org/ietf/1shadow-sites.txt<br></div></blockquote></div><br></body></html>
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From rjsparks@nostrum.com  Mon Mar  2 12:19:18 2009
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Subject: [Sip] Fwd: I-D Action:draft-sparks-sip-3261-interop-statement-00.txt
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Also split out of steps-to-draft

Begin forwarded message:

> From: Internet-Drafts@ietf.org
> Date: March 2, 2009 8:30:02 AM CST
> To: i-d-announce@ietf.org
> Subject: I-D Action:draft-sparks-sip-3261-interop-statement-00.txt
> Reply-To: internet-drafts@ietf.org
>
> A New Internet-Draft is available from the on-line Internet-Drafts  
> directories.
>
> 	Title           : RFC3261 Interop Statement
> 	Author(s)       : R. Sparks
> 	Filename        : draft-sparks-sip-3261-interop-statement-00.txt
> 	Pages           : 9
> 	Date            : 2009-03-02
>
> This document captures an outline of the interoperability statements
> that will be collected to construct an interoperability report for
> RFC 3261.  The outline is stil under review and should not be treated
> as complete, but will drive data collection at upcoming
> interoperability events.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-sparks-sip-3261-interop-statement-00.txt
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><div>Also split out of =
steps-to-draft</div><div><br><div>Begin forwarded message:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><b>From: =
</b></font><font face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px =
Helvetica"><a =
href=3D"mailto:Internet-Drafts@ietf.org">Internet-Drafts@ietf.org</a></fon=
t></div><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: =
0px; margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" =
color=3D"#000000" style=3D"font: 12.0px Helvetica; color: =
#000000"><b>Date: </b></font><font face=3D"Helvetica" size=3D"3" =
style=3D"font: 12.0px Helvetica">March 2, 2009 8:30:02 AM =
CST</font></div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><font face=3D"Helvetica" =
size=3D"3" color=3D"#000000" style=3D"font: 12.0px Helvetica; color: =
#000000"><b>To: </b></font><font face=3D"Helvetica" size=3D"3" =
style=3D"font: 12.0px Helvetica"><a =
href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a></font></di=
v><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><b>Subject: =
</b></font><font face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px =
Helvetica"><b>I-D =
Action:draft-sparks-sip-3261-interop-statement-00.txt<span =
class=3D"Apple-converted-space">&nbsp;</span></b></font></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><b>Reply-To: =
</b></font><font face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px =
Helvetica"><a =
href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a></fon=
t></div><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: =
0px; margin-left: 0px; min-height: 14px; "><br></div> </div><div>A New =
Internet-Draft is available from the on-line Internet-Drafts =
directories.<br><br><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>Title =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: RFC3261 =
Interop Statement<br><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>Author(s) =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: R. Sparks<br><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>Filename =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: =
draft-sparks-sip-3261-interop-statement-00.txt<br><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>Pages =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: 9<br><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>Date =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: =
2009-03-02<br><br>This document captures an outline of the =
interoperability statements<br>that will be collected to construct an =
interoperability report for<br>RFC 3261. &nbsp;The outline is stil under =
review and should not be treated<br>as complete, but will drive data =
collection at upcoming<br>interoperability events.<br><br>A URL for this =
Internet-Draft is:<br><a =
href=3D"http://www.ietf.org/internet-drafts/draft-sparks-sip-3261-interop-=
statement-00.txt">http://www.ietf.org/internet-drafts/draft-sparks-sip-326=
1-interop-statement-00.txt</a><br><br>Internet-Drafts are also available =
by anonymous FTP at:<br>ftp://ftp.ietf.org/internet-drafts/<br><br>Below =
is the data which will enable a MIME compliant mail =
reader<br>implementation to automatically retrieve the ASCII version of =
the<br>Internet-Draft.<br></div></blockquote></div></body></html>=

--Apple-Mail-60-19930079
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<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div><blockquote type="cite"><div>_______________________________________________<br>I-D-Announce mailing list<br>I-D-Announce@ietf.org<br>https://www.ietf.org/mailman/listinfo/i-d-announce<br>Internet-Draft directories: http://www.ietf.org/shadow.html<br>or ftp://ftp.ietf.org/ietf/1shadow-sites.txt<br></div></blockquote></div><br></body></html>
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From vkg@alcatel-lucent.com  Mon Mar  2 12:19:35 2009
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To: Christer Holmberg <christer.holmberg@ericsson.com>
References: <7CDFD441-A190-4BC9-8BAA-7821A12FF4FF@softarmor.com> <CA9998CD4A020D418654FCDEF4E707DF0B167F63@esealmw113.eemea.ericsson.se>
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Cc: SIP WG <sip@ietf.org>
Subject: Re: [Sip] Connection Reuse draft to focus on TLS connections
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Christer Holmberg wrote:
> Hi,
> 
> Just a question: does this rescope affect chapter 8 (Overlap with
> connection reuse) of the keep draft
> (http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-keep-03.txt)?

Christer: I believe the text you have in keep-03 stays as it
is.  When we start work on a new draft for connection oriented
non-TLS transports, we may have to update that section with
one more reference to the new draft.  That should be the
extent of impact on keep-03.

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From rjsparks@nostrum.com  Mon Mar  2 12:19:46 2009
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This got a fairly big touch due to better handling of transport error  
conditions. Please review this carefully.

RjS

Begin forwarded message:

> From: Internet-Drafts@ietf.org
> Date: March 2, 2009 8:45:01 AM CST
> To: i-d-announce@ietf.org
> Subject: I-D Action:draft-sparks-sip-invfix-03.txt
> Reply-To: internet-drafts@ietf.org
>
> A New Internet-Draft is available from the on-line Internet-Drafts  
> directories.
>
> 	Title           : Correct transaction handling for 200 responses to  
> Session Initiation Protocol INVITE requests
> 	Author(s)       : R. Sparks
> 	Filename        : draft-sparks-sip-invfix-03.txt
> 	Pages           : 20
> 	Date            : 2009-03-02
>
> This document normatively updates RFC 3261, the Session Initiation
> Protocol (SIP), to address an error in the specified handling of
> success (200 class) responses to INVITE requests.  Elements following
> RFC 3261 exactly will misidentify retransmissions of the request as a
> new, unassociated, request.  The correction involves modifying the
> INVITE transaction state machines.  The correction also changes the
> way responses that cannot be matched to an existing transaction are
> handled to address a security risk.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-sparks-sip-invfix-03.txt
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; ">This got a fairly big touch due =
to better handling of transport error conditions. Please review this =
carefully.<div><br></div><div>RjS</div><div><div><br><div>Begin =
forwarded message:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><b>From: =
</b></font><font face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px =
Helvetica"><a =
href=3D"mailto:Internet-Drafts@ietf.org">Internet-Drafts@ietf.org</a></fon=
t></div><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: =
0px; margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" =
color=3D"#000000" style=3D"font: 12.0px Helvetica; color: =
#000000"><b>Date: </b></font><font face=3D"Helvetica" size=3D"3" =
style=3D"font: 12.0px Helvetica">March 2, 2009 8:45:01 AM =
CST</font></div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><font face=3D"Helvetica" =
size=3D"3" color=3D"#000000" style=3D"font: 12.0px Helvetica; color: =
#000000"><b>To: </b></font><font face=3D"Helvetica" size=3D"3" =
style=3D"font: 12.0px Helvetica"><a =
href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a></font></di=
v><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><b>Subject: =
</b></font><font face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px =
Helvetica"><b>I-D Action:draft-sparks-sip-invfix-03.txt<span =
class=3D"Apple-converted-space">&nbsp;</span></b></font></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><b>Reply-To: =
</b></font><font face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px =
Helvetica"><a =
href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a></fon=
t></div><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: =
0px; margin-left: 0px; min-height: 14px; "><br></div> </div><div>A New =
Internet-Draft is available from the on-line Internet-Drafts =
directories.<br><br><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>Title =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: Correct =
transaction handling for 200 responses to Session Initiation Protocol =
INVITE requests<br><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>Author(s) =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: R. Sparks<br><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>Filename =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: =
draft-sparks-sip-invfix-03.txt<br><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>Pages =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: =
20<br><span class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span>Date =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: =
2009-03-02<br><br>This document normatively updates RFC 3261, the =
Session Initiation<br>Protocol (SIP), to address an error in the =
specified handling of<br>success (200 class) responses to INVITE =
requests. &nbsp;Elements following<br>RFC 3261 exactly will misidentify =
retransmissions of the request as a<br>new, unassociated, request. =
&nbsp;The correction involves modifying the<br>INVITE transaction state =
machines. &nbsp;The correction also changes the<br>way responses that =
cannot be matched to an existing transaction are<br>handled to address a =
security risk.<br><br>A URL for this Internet-Draft is:<br><a =
href=3D"http://www.ietf.org/internet-drafts/draft-sparks-sip-invfix-03.txt=
">http://www.ietf.org/internet-drafts/draft-sparks-sip-invfix-03.txt</a><b=
r><br>Internet-Drafts are also available by anonymous FTP =
at:<br>ftp://ftp.ietf.org/internet-drafts/<br><br>Below is the data =
which will enable a MIME compliant mail reader<br>implementation to =
automatically retrieve the ASCII version of =
the<br>Internet-Draft.<br></div></blockquote></div></div></body></html>=

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<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div><div><blockquote type="cite"><div>_______________________________________________<br>I-D-Announce mailing list<br>I-D-Announce@ietf.org<br>https://www.ietf.org/mailman/listinfo/i-d-announce<br>Internet-Draft directories: http://www.ietf.org/shadow.html<br>or ftp://ftp.ietf.org/ietf/1shadow-sites.txt<br></div></blockquote></div><br></div></body></html>
--Apple-Mail-63-19971108--

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From HKaplan@acmepacket.com  Mon Mar  2 12:58:42 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dean Willis <dean.willis@softarmor.com>, SIP WG <sip@ietf.org>
Date: Mon, 2 Mar 2009 15:58:45 -0500
Thread-Topic: [Sip] Connection Reuse draft to focus on TLS connections
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Cc: Cullen Jennings <fluffy@cisco.com>, Keith Drage <drage@alcatel-lucent.com>
Subject: Re: [Sip] Connection Reuse draft to focus on TLS connections
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One should note it's not applicable to just "TLS" - it's only applicable to=
 mutual-TLS.  I don't mean to sound negative, but really you might as well =
limit it's applicability to IPv6 as well at this rate.  :(

Can this draft at least remove the statement in section 8.1 that the client=
 SHOULD NOT add an alias parameter to TCP/SCTP transports?   There's actual=
ly nothing wrong with adding one - it's the server side of the connection t=
hat needs to not use/honor it based on its policy. (and if you think adding=
 one would be malicious, then trust me malicious devices will simply add on=
e and ignore this draft's SHOULD NOT statement)  All the parameter says is =
effectively: "I am willing to accept requests on this connection I created,=
 if you wish to send them to me over it".

-hadriel


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Dea=
n
> Willis
> Sent: Monday, March 02, 2009 3:04 PM
> To: SIP WG
> Cc: Cullen Jennings; Keith Drage
> Subject: [Sip] Connection Reuse draft to focus on TLS connections
>
>
> We intend to slightly rescope the Connection Reuse draft:
>
> http://tools.ietf.org/id/draft-ietf-sip-connect-reuse-12.txt
>
> Based on input from our ADs, the WG chairs and the editor of the have
> agreed to eliminate the part of the draft that discusses the
> implementation of reusing non-TLS connections. The stage for this is
> set in the abstract of the current draft, from which I quote:
>
> > From the security perspective, it is bad practice to reuse a single
> > connection for the TCP or SCTP transport between two peers, and this
> > document provides specific insights into why this is the case. As a
> > remedy, it suggests using two TCP connections (or two SCTP
> > associations), each opened pro-actively towards the recipient by the
> > sender.
>
>
> We may  discuss at a later date the idea of developing a draft for
> connection reuse in the absence of TLS.
>
> I'm really not expecting anybody to object strongly to this approach,
> but am bringing this to the list to avoid surprises down the road. If
> you have major issues with doing it this way, please bring them up now.
>
> --
> Dean Willis (as chair, ten years and counting . . .)
>
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

From HKaplan@acmepacket.com  Mon Mar  2 13:11:50 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Robert Sparks <rjsparks@nostrum.com>, SIP List <sip@ietf.org>
Date: Mon, 2 Mar 2009 16:12:09 -0500
Thread-Topic: [Sip] Fwd: I-D Action:draft-sparks-sip-3261-interop-statement-00.txt
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Nice list.  Thanks for putting this together!

I propose you delete " - Demonstrate successful exchange of S/MIME protecte=
d entities".
I further propose the answer to "Right now there is nothing covering sips:.=
 What do those statements need to look like?" is an empty line.
And I propose the answer to "How do we make a statement that captures we've=
 found two implementations that discover their registrar through configurat=
ion rather than using URI resolution." Is we don't need to - the real world=
 has done it for us.

And I like your bullet point in the middle of the Application-level feature=
s. :)

-hadriel

________________________________________
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Rober=
t Sparks
Sent: Monday, March 02, 2009 3:19 PM
To: SIP List
Subject: [Sip] Fwd: I-D Action:draft-sparks-sip-3261-interop-statement-00.t=
xt

Also split out of steps-to-draft

Begin forwarded message:


From: Internet-Drafts@ietf.org
Date: March 2, 2009 8:30:02 AM CST
To: i-d-announce@ietf.org
Subject: I-D Action:draft-sparks-sip-3261-interop-statement-00.txt
Reply-To: internet-drafts@ietf.org

A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.

        Title           : RFC3261 Interop Statement
        Author(s)       : R. Sparks
        Filename        : draft-sparks-sip-3261-interop-statement-00.txt
        Pages           : 9
        Date            : 2009-03-02

This document captures an outline of the interoperability statements
that will be collected to construct an interoperability report for
RFC 3261.  The outline is stil under review and should not be treated
as complete, but will drive data collection at upcoming
interoperability events.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-sparks-sip-3261-interop-statement=
-00.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

Below is the data which will enable a MIME compliant mail reader
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Internet-Draft.

From dworley@nortel.com  Mon Mar  2 13:16:44 2009
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On Mon, 2009-03-02 at 11:40 +0530, Mahesh wrote:
> what will happen if caller din't get ACK.whats the behaviour of caller.

See RFC 3261 section 17.2.1, especially figure 7.  If the UAS does not
receive an ACK, it re-sends the final response to the INVITE.  If Timer
H fires before an ACK arrives, the UAS deletes the dialog.

Dale



From vkg@alcatel-lucent.com  Mon Mar  2 13:21:30 2009
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Hadriel Kaplan wrote:
> One should note it's not applicable to just "TLS" - it's only
> applicable to mutual-TLS.  I don't mean to sound negative, but really
> you might as well limit it's applicability to IPv6 as well at this
> rate.  :(
> 
> Can this draft at least remove the statement in section 8.1 that the
> client SHOULD NOT add an alias parameter to TCP/SCTP transports?

Hadriel: I have had discussions with the AD on TCP/SCTP connection
reuse and their impact on the current TLS connection reuse draft.
The AD is of the opinion that it will be hard to progress
the current connect-reuse draft in IESG while allowing TCP/SCTP
connection reuse in the big-I Internet.

Obviously, the fact remains that people will use TCP connection
reuse, whether or not this draft prohibits it or not.  The
question boils down to how to best move *this* work ahead
(i.e., TLS connection reuse), and then deal with TCP connection
reuse.

The best way forward is to excise any text relating to TCP/SCTP
connection reuse, and focus this draft on TLS only (the original
motivation for the draft.)  This allows us to get the TLS
connect-reuse draft out (it has been pending now for four
arduous, laborious, suspenseful, pondering, brutal, intemperate,
and exacting years.)

In parallel, right after the -00 deadline for IETF 74 is over,
I will put out a new draft on TCP/SCTP connection reuse that
allows reuse for these transports when certain assumptions
are met.  I have been assured by the AD that a milestone can
be quickly added to SIP/DISPATCH/SIPCORE WG to allow the new
work to progress expeditiously.

Is that a workable solution for you?  Please let me know.

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From HKaplan@acmepacket.com  Mon Mar  2 13:25:50 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Vijay K. Gurbani" <vkg@alcatel-lucent.com>
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Thread-Topic: [Sip] Connection Reuse draft to focus on TLS connections
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Ahhh, so you'd like to end your misery.  That sounds fair.  Ignore my comme=
nt.

-hadriel
p.s. but you're a glutton for punishment if you submit another I-D for non-=
TLS connect-reuse. ;)


> -----Original Message-----
> From: Vijay K. Gurbani [mailto:vkg@alcatel-lucent.com]
> Sent: Monday, March 02, 2009 4:22 PM
> To: Hadriel Kaplan
> Cc: SIP WG
> Subject: Re: [Sip] Connection Reuse draft to focus on TLS connections
>
> Hadriel Kaplan wrote:
> > One should note it's not applicable to just "TLS" - it's only
> > applicable to mutual-TLS.  I don't mean to sound negative, but really
> > you might as well limit it's applicability to IPv6 as well at this
> > rate.  :(
> >
> > Can this draft at least remove the statement in section 8.1 that the
> > client SHOULD NOT add an alias parameter to TCP/SCTP transports?
>
> Hadriel: I have had discussions with the AD on TCP/SCTP connection
> reuse and their impact on the current TLS connection reuse draft.
> The AD is of the opinion that it will be hard to progress
> the current connect-reuse draft in IESG while allowing TCP/SCTP
> connection reuse in the big-I Internet.
>
> Obviously, the fact remains that people will use TCP connection
> reuse, whether or not this draft prohibits it or not.  The
> question boils down to how to best move *this* work ahead
> (i.e., TLS connection reuse), and then deal with TCP connection
> reuse.
>
> The best way forward is to excise any text relating to TCP/SCTP
> connection reuse, and focus this draft on TLS only (the original
> motivation for the draft.)  This allows us to get the TLS
> connect-reuse draft out (it has been pending now for four
> arduous, laborious, suspenseful, pondering, brutal, intemperate,
> and exacting years.)
>
> In parallel, right after the -00 deadline for IETF 74 is over,
> I will put out a new draft on TCP/SCTP connection reuse that
> allows reuse for these transports when certain assumptions
> are met.  I have been assured by the AD that a milestone can
> be quickly added to SIP/DISPATCH/SIPCORE WG to allow the new
> work to progress expeditiously.
>
> Is that a workable solution for you?  Please let me know.
>
> - vijay
> --
> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
> 1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
> Web:   http://ect.bell-labs.com/who/vkg/

From vkg@alcatel-lucent.com  Mon Mar  2 13:30:19 2009
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Hadriel Kaplan wrote:
> Ahhh, so you'd like to end your misery.  That sounds fair.  Ignore my
> comment.

Ending my connect-reuse induced misery is one part, yes.

> -hadriel p.s. but you're a glutton for punishment if you submit
> another I-D for non-TLS connect-reuse. ;)

Aha, but I can have you co-edit the TCP/SCTP connect-reuse draft
and make sure that the punishment is equitably shared between us ;-)

But seriously, as long as the need to do TCP/SCTP connection reuse
is there, I don't mind spending time doing that effort, with you
or anyone else (in fact, I have an editor in mind for the new
draft ... ;-) )

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From lists@ohlmeier.org  Mon Mar  2 14:29:29 2009
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Hi Robert,

would appreciate it if you could add to the "URI specific features" (or 
to the "Extensibility mechanism"?) from my interop experience at SIPit:

- Demonstrate the correct usage of URIs with known and unknown 
parameters in Route URIs and Request-URI

Cheers
   Nils

Am 02.03.2009 21:19 Uhr, schrieb Robert Sparks:
> Also split out of steps-to-draft
>
> Begin forwarded message:
>
>> *From: *Internet-Drafts@ietf.org <mailto:Internet-Drafts@ietf.org>
>> *Date: *March 2, 2009 8:30:02 AM CST
>> *To: *i-d-announce@ietf.org <mailto:i-d-announce@ietf.org>
>> *Subject: **I-D Action:draft-sparks-sip-3261-interop-statement-00.txt *
>> *Reply-To: *internet-drafts@ietf.org <mailto:internet-drafts@ietf.org>
>>
>> A New Internet-Draft is available from the on-line Internet-Drafts
>> directories.
>>
>> Title : RFC3261 Interop Statement
>> Author(s) : R. Sparks
>> Filename : draft-sparks-sip-3261-interop-statement-00.txt
>> Pages : 9
>> Date : 2009-03-02
>>
>> This document captures an outline of the interoperability statements
>> that will be collected to construct an interoperability report for
>> RFC 3261. The outline is stil under review and should not be treated
>> as complete, but will drive data collection at upcoming
>> interoperability events.
>>
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-sparks-sip-3261-interop-statement-00.txt
>>
>> Internet-Drafts are also available by anonymous FTP at:
>> ftp://ftp.ietf.org/internet-drafts/
>>
>> Below is the data which will enable a MIME compliant mail reader
>> implementation to automatically retrieve the ASCII version of the
>> Internet-Draft.
>
> ------------------------------------------------------------------------
>
>> _______________________________________________
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>> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>
>
> ------------------------------------------------------------------------
>
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> Use sip-implementors@cs.columbia.edu for questions on current sip
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From dworley@nortel.com  Mon Mar  2 15:44:55 2009
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From: "Dale Worley" <dworley@nortel.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
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Subject: Re: [Sip] FW: Initial Version I-D Submission Deadline Extended to	March 4, 2009
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Although please note that the Internet Draft Submission Tool does not
know that the deadline has been extended.

Dale



From christer.holmberg@ericsson.com  Mon Mar  2 22:23:30 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Robert Sparks" <rjsparks@nostrum.com>, "SIP List" <sip@ietf.org>
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Hi,
=20
A couple of questions:
=20
1. The draft says "Demonstrate successful offer/answer exchange".
Offer/answer is one of the biggest interop issues, so I think we would
need to be a little more specific than that. We should probably have a
complete separate interop-statement on offer/answer...
=20
2. For the things that shall be demonstrated, should there be references
to where the correct behavior is specified? It's difficult to
demonstrate correct behavior if you don't know what that correct
behavior is :)
=20
Regards,
=20
Christer

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Robert Sparks
Sent: Monday, March 02, 2009 10:19 PM
To: SIP List
Subject: [Sip] Fwd: I-D
Action:draft-sparks-sip-3261-interop-statement-00.txt


Also split out of steps-to-draft

Begin forwarded message:


	From: Internet-Drafts@ietf.org
	Date: March 2, 2009 8:30:02 AM CST
	To: i-d-announce@ietf.org
	Subject: I-D
Action:draft-sparks-sip-3261-interop-statement-00.txt=20
	Reply-To: internet-drafts@ietf.org

	A New Internet-Draft is available from the on-line
Internet-Drafts directories.
=09
	Title           : RFC3261 Interop Statement
	Author(s)       : R. Sparks
	Filename        : draft-sparks-sip-3261-interop-statement-00.txt
	Pages           : 9
	Date            : 2009-03-02
=09
	This document captures an outline of the interoperability
statements
	that will be collected to construct an interoperability report
for
	RFC 3261.  The outline is stil under review and should not be
treated
	as complete, but will drive data collection at upcoming
	interoperability events.
=09
	A URL for this Internet-Draft is:
=09
http://www.ietf.org/internet-drafts/draft-sparks-sip-3261-interop-statem
ent-00.txt
=09
	Internet-Drafts are also available by anonymous FTP at:
	ftp://ftp.ietf.org/internet-drafts/
=09
	Below is the data which will enable a MIME compliant mail reader
	implementation to automatically retrieve the ASCII version of
the
	Internet-Draft.
=09


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Content-Type: text/html;
	charset="us-ascii"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.6000.16809" name=3DGENERATOR></HEAD>
<BODY=20
style=3D"WORD-WRAP: break-word; webkit-nbsp-mode: space; =
webkit-line-break: after-white-space">
<DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Hi,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>A couple of questions:</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>1. The draft says "Demonstrate successful =
offer/answer=20
exchange". Offer/answer is one of the biggest interop issues, so I think =
we=20
would need to be a little more specific than that. We should probably =
have a=20
complete separate interop-statement on =
offer/answer...</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>2. For the things that shall be demonstrated, =
should there=20
be references to where the correct behavior is specified? It's difficult =
to=20
demonstrate correct behavior if you don't know what that correct =
behavior is=20
:)</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Christer</FONT></SPAN></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Robert =
Sparks<BR><B>Sent:</B>=20
Monday, March 02, 2009 10:19 PM<BR><B>To:</B> SIP =
List<BR><B>Subject:</B> [Sip]=20
Fwd: I-D=20
Action:draft-sparks-sip-3261-interop-statement-00.txt<BR></FONT><BR></DIV=
>
<DIV></DIV>
<DIV>Also split out of steps-to-draft</DIV>
<DIV><BR>
<DIV>Begin forwarded message:</DIV><BR =
class=3DApple-interchange-newline>
<BLOCKQUOTE type=3D"cite">
  <DIV>
  <DIV style=3D"MARGIN: 0px"><FONT style=3D"FONT: 12px Helvetica; COLOR: =
#000000"=20
  face=3DHelvetica color=3D#000000 size=3D3><B>From: </B></FONT><FONT=20
  style=3D"FONT: 12px Helvetica" face=3DHelvetica size=3D3><A=20
  =
href=3D"mailto:Internet-Drafts@ietf.org">Internet-Drafts@ietf.org</A></FO=
NT></DIV>
  <DIV style=3D"MARGIN: 0px"><FONT style=3D"FONT: 12px Helvetica; COLOR: =
#000000"=20
  face=3DHelvetica color=3D#000000 size=3D3><B>Date: </B></FONT><FONT=20
  style=3D"FONT: 12px Helvetica" face=3DHelvetica size=3D3>March 2, 2009 =
8:30:02 AM=20
  CST</FONT></DIV>
  <DIV style=3D"MARGIN: 0px"><FONT style=3D"FONT: 12px Helvetica; COLOR: =
#000000"=20
  face=3DHelvetica color=3D#000000 size=3D3><B>To: </B></FONT><FONT=20
  style=3D"FONT: 12px Helvetica" face=3DHelvetica size=3D3><A=20
  =
href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</A></FONT></D=
IV>
  <DIV style=3D"MARGIN: 0px"><FONT style=3D"FONT: 12px Helvetica; COLOR: =
#000000"=20
  face=3DHelvetica color=3D#000000 size=3D3><B>Subject: </B></FONT><FONT =

  style=3D"FONT: 12px Helvetica" face=3DHelvetica size=3D3><B>I-D=20
  Action:draft-sparks-sip-3261-interop-statement-00.txt<SPAN=20
  class=3DApple-converted-space>&nbsp;</SPAN></B></FONT></DIV>
  <DIV style=3D"MARGIN: 0px"><FONT style=3D"FONT: 12px Helvetica; COLOR: =
#000000"=20
  face=3DHelvetica color=3D#000000 size=3D3><B>Reply-To: =
</B></FONT><FONT=20
  style=3D"FONT: 12px Helvetica" face=3DHelvetica size=3D3><A=20
  =
href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</A></FO=
NT></DIV>
  <DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px"><BR></DIV></DIV>
  <DIV>A New Internet-Draft is available from the on-line =
Internet-Drafts=20
  directories.<BR><BR><SPAN class=3DApple-tab-span=20
  style=3D"WHITE-SPACE: pre"></SPAN>Title=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: RFC3261 =
Interop=20
  Statement<BR><SPAN class=3DApple-tab-span=20
  style=3D"WHITE-SPACE: pre"></SPAN>Author(s)=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: R. Sparks<BR><SPAN =
class=3DApple-tab-span=20
  style=3D"WHITE-SPACE: pre"></SPAN>Filename=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;:=20
  draft-sparks-sip-3261-interop-statement-00.txt<BR><SPAN =
class=3DApple-tab-span=20
  style=3D"WHITE-SPACE: pre"></SPAN>Pages=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: =
9<BR><SPAN=20
  class=3DApple-tab-span style=3D"WHITE-SPACE: pre"></SPAN>Date=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;:=20
  2009-03-02<BR><BR>This document captures an outline of the =
interoperability=20
  statements<BR>that will be collected to construct an interoperability =
report=20
  for<BR>RFC 3261. &nbsp;The outline is stil under review and should not =
be=20
  treated<BR>as complete, but will drive data collection at=20
  upcoming<BR>interoperability events.<BR><BR>A URL for this =
Internet-Draft=20
  is:<BR><A=20
  =
href=3D"http://www.ietf.org/internet-drafts/draft-sparks-sip-3261-interop=
-statement-00.txt">http://www.ietf.org/internet-drafts/draft-sparks-sip-3=
261-interop-statement-00.txt</A><BR><BR>Internet-Drafts=20
  are also available by anonymous FTP=20
  at:<BR>ftp://ftp.ietf.org/internet-drafts/<BR><BR>Below is the data =
which will=20
  enable a MIME compliant mail reader<BR>implementation to automatically =

  retrieve the ASCII version of=20
the<BR>Internet-Draft.<BR></DIV></BLOCKQUOTE></DIV></BODY></HTML>

------_=_NextPart_001_01C99BC8.90F8A128--

From dean.willis@softarmor.com  Mon Mar  2 23:59:06 2009
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Cc: Keith Drage <drage@alcatel-lucent.com>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] New version (-05) of the 199 draft - ready for WGLC?
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We didn't get any real answers due to a bit of confusion between this  
and keepalive. So, let's ask the question again.

Is 199 ready for WGLC?

--
dean


On Feb 19, 2009, at 5:12 AM, Christer Holmberg wrote:

>
> Hi,
>
> I've submitted a new version of draft-ietf-sip-199. I have done  
> changes and updates based on feedback and discussions (please let me  
> know if you think I've forgotten something), and I've also added  
> text to the security chapter.
>
> I propose that we start the WGLC process for the draft.
>
> The draft can also be found at:
>
> http://users.piuha.net/cholmber/drafts/draft-ietf-sip-199-05.txt
>
> Regards,
>
> Christer
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


From munakata.mayumi@lab.ntt.co.jp  Tue Mar  3 00:13:10 2009
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Date: Tue, 03 Mar 2009 17:12:34 +0900
From: Mayumi Munakata <munakata.mayumi@lab.ntt.co.jp>
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Cc: SIP List <SIP@ietf.org>, draft-ietf-sip-ua-privacy@tools.ietf.org, "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>, Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] WGLC for draft-ietf-sip-ua-privacy
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Hi John,

Thank you for the comments.

See inline.

 > 1. In 4.1:
 > "If revealing the domain name in the
 >    Contact header field is a concern, usage of a third-party GRUU server
 >    to obtain a temp-gruu that is irrelevant to users' domain, which is
 >    outside the scope of this document SHOULD be considered."
 > "SHOULD be considered" is not a testable normative statement. It is not
 > the user's domain that is outside the scope, but usage of a third-party
 > GRUU server. The word "irrelevant" in inappropriate here. To take care
 > of these concerns, I think it should say:
 > "If revealing the domain name in the  Contact header field is a concern,
 > use of a third party GRUU server is a possible solution, but this is
 > outside the scope of this document."

I agree and will replace the text with the one you proposed.

 >
 > 2. In 5.1.1:
 > "For out-of-dialog
 >    request, anonymous URI MAY be set when anonymization is required."
 > What is the purpose of this statement? I assume out-of-dialog does not
 > include dialog-forming (which is covered earlier in the paragraph). For
 > a completely out-of-dialog request such as a MESSAGE request, why would
 > one include Contact? For REGISTER, one certainly cannot use an anonymous
 > URI.

The out-of-dialog request indeed meant the MESSAGE request.
I checked the RFC and found that MESSAGE requests don't
include the Contact header as you pointed out.
I will delete the sentence.

Thank you!

Mayumi

 >
 > John

From john.elwell@siemens.com  Tue Mar  3 02:23:42 2009
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Date: Tue, 03 Mar 2009 10:23:22 +0000
From: "Elwell, John" <john.elwell@siemens.com>
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Subject: Re: [Sip] Fwd: I-DAction:draft-sparks-sip-3261-interop-statement-00.txt
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=20


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Christer Holmberg
	Sent: 03 March 2009 06:23
	To: Robert Sparks; SIP List
	Subject: Re: [Sip] Fwd:
I-DAction:draft-sparks-sip-3261-interop-statement-00.txt
=09
=09
	Hi,
	=20
	A couple of questions:
	=20
	1. The draft says "Demonstrate successful offer/answer
exchange". Offer/answer is one of the biggest interop issues, so I think
we would need to be a little more specific than that. We should probably
have a complete separate interop-statement on offer/answer...
	[JRE] Yes, I agree with this. I guess we need to draw a line
between SIP and SDP, but I think we should try to cover RFC 3264 aspects
to a reasonable degree.
	=20
	John
	=20
	=20
	2. For the things that shall be demonstrated, should there be
references to where the correct behavior is specified? It's difficult to
demonstrate correct behavior if you don't know what that correct
behavior is :)
	=20
	Regards,
	=20
	Christer

________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Robert Sparks
	Sent: Monday, March 02, 2009 10:19 PM
	To: SIP List
	Subject: [Sip] Fwd: I-D
Action:draft-sparks-sip-3261-interop-statement-00.txt
=09
=09
	Also split out of steps-to-draft

	Begin forwarded message:


		From: Internet-Drafts@ietf.org
		Date: March 2, 2009 8:30:02 AM CST
		To: i-d-announce@ietf.org
		Subject: I-D
Action:draft-sparks-sip-3261-interop-statement-00.txt=20
		Reply-To: internet-drafts@ietf.org

		A New Internet-Draft is available from the on-line
Internet-Drafts directories.
	=09
		Title           : RFC3261 Interop Statement
		Author(s)       : R. Sparks
		Filename        :
draft-sparks-sip-3261-interop-statement-00.txt
		Pages           : 9
		Date            : 2009-03-02
	=09
		This document captures an outline of the
interoperability statements
		that will be collected to construct an interoperability
report for
		RFC 3261.  The outline is stil under review and should
not be treated
		as complete, but will drive data collection at upcoming
		interoperability events.
	=09
		A URL for this Internet-Draft is:
=09
http://www.ietf.org/internet-drafts/draft-sparks-sip-3261-interop-statem
ent-00.txt
	=09
		Internet-Drafts are also available by anonymous FTP at:
		ftp://ftp.ietf.org/internet-drafts/
	=09
		Below is the data which will enable a MIME compliant
mail reader
		implementation to automatically retrieve the ASCII
version of the
		Internet-Draft.
	=09


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<BODY=20
style=3D"WORD-WRAP: break-word; webkit-nbsp-mode: space; =
webkit-line-break: after-white-space">
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial =
color=3D#000080></FONT>&nbsp;</DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000080 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Christer=20
  Holmberg<BR><B>Sent:</B> 03 March 2009 06:23<BR><B>To:</B> Robert =
Sparks; SIP=20
  List<BR><B>Subject:</B> Re: [Sip] Fwd:=20
  =
I-DAction:draft-sparks-sip-3261-interop-statement-00.txt<BR></FONT><BR></=
DIV>
  <DIV></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2>Hi,</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2>A couple of questions:</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
color=3D#0000ff=20
  size=3D2><FONT face=3DArial>1. The draft says "Demonstrate successful =
offer/answer=20
  exchange". Offer/answer is one of the biggest interop issues, so I =
think we=20
  would need to be a little more specific than that. We should probably =
have a=20
  complete separate interop-statement on offer/answer...<BR><SPAN=20
  class=3D272232210-03032009><FONT color=3D#000080 =
size=3D3>[JRE]&nbsp;Yes, I agree=20
  with this. I guess we need to draw a line between SIP and SDP, but I =
think we=20
  should try to cover RFC 3264 aspects to a reasonable=20
  degree.</FONT></SPAN></FONT></FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
color=3D#0000ff=20
  size=3D2><FONT face=3DArial><SPAN=20
  class=3D272232210-03032009></SPAN></FONT></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
color=3D#0000ff=20
  size=3D2><FONT face=3DArial><SPAN class=3D272232210-03032009><FONT =
color=3D#000080=20
  size=3D3>John</FONT></SPAN></FONT></FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
color=3D#0000ff=20
  size=3D2><FONT face=3DArial><SPAN=20
  class=3D272232210-03032009>&nbsp;</SPAN></FONT></FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2>2. For the things that shall be demonstrated, =
should=20
  there be references to where the correct behavior is specified? It's =
difficult=20
  to demonstrate correct behavior if you don't know what that correct =
behavior=20
  is :)</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2>Regards,</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D607171721-02032009><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2>Christer</FONT></SPAN></DIV><BR>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Robert=20
  Sparks<BR><B>Sent:</B> Monday, March 02, 2009 10:19 PM<BR><B>To:</B> =
SIP=20
  List<BR><B>Subject:</B> [Sip] Fwd: I-D=20
  =
Action:draft-sparks-sip-3261-interop-statement-00.txt<BR></FONT><BR></DIV=
>
  <DIV></DIV>
  <DIV>Also split out of steps-to-draft</DIV>
  <DIV><BR>
  <DIV>Begin forwarded message:</DIV><BR =
class=3DApple-interchange-newline>
  <BLOCKQUOTE type=3D"cite">
    <DIV>
    <DIV style=3D"MARGIN: 0px"><FONT style=3D"FONT: 12px Helvetica; =
COLOR: #000000"=20
    face=3DHelvetica color=3D#000000 size=3D3><B>From: </B></FONT><FONT=20
    style=3D"FONT: 12px Helvetica" face=3DHelvetica size=3D3><A=20
    =
href=3D"mailto:Internet-Drafts@ietf.org">Internet-Drafts@ietf.org</A></FO=
NT></DIV>
    <DIV style=3D"MARGIN: 0px"><FONT style=3D"FONT: 12px Helvetica; =
COLOR: #000000"=20
    face=3DHelvetica color=3D#000000 size=3D3><B>Date: </B></FONT><FONT=20
    style=3D"FONT: 12px Helvetica" face=3DHelvetica size=3D3>March 2, =
2009 8:30:02 AM=20
    CST</FONT></DIV>
    <DIV style=3D"MARGIN: 0px"><FONT style=3D"FONT: 12px Helvetica; =
COLOR: #000000"=20
    face=3DHelvetica color=3D#000000 size=3D3><B>To: </B></FONT><FONT=20
    style=3D"FONT: 12px Helvetica" face=3DHelvetica size=3D3><A=20
    =
href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</A></FONT></D=
IV>
    <DIV style=3D"MARGIN: 0px"><FONT style=3D"FONT: 12px Helvetica; =
COLOR: #000000"=20
    face=3DHelvetica color=3D#000000 size=3D3><B>Subject: =
</B></FONT><FONT=20
    style=3D"FONT: 12px Helvetica" face=3DHelvetica size=3D3><B>I-D=20
    Action:draft-sparks-sip-3261-interop-statement-00.txt<SPAN=20
    class=3DApple-converted-space>&nbsp;</SPAN></B></FONT></DIV>
    <DIV style=3D"MARGIN: 0px"><FONT style=3D"FONT: 12px Helvetica; =
COLOR: #000000"=20
    face=3DHelvetica color=3D#000000 size=3D3><B>Reply-To: =
</B></FONT><FONT=20
    style=3D"FONT: 12px Helvetica" face=3DHelvetica size=3D3><A=20
    =
href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</A></FO=
NT></DIV>
    <DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px"><BR></DIV></DIV>
    <DIV>A New Internet-Draft is available from the on-line =
Internet-Drafts=20
    directories.<BR><BR><SPAN class=3DApple-tab-span=20
    style=3D"WHITE-SPACE: pre"></SPAN>Title=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: =
RFC3261=20
    Interop Statement<BR><SPAN class=3DApple-tab-span=20
    style=3D"WHITE-SPACE: pre"></SPAN>Author(s)=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: R. Sparks<BR><SPAN=20
    class=3DApple-tab-span style=3D"WHITE-SPACE: pre"></SPAN>Filename=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;:=20
    draft-sparks-sip-3261-interop-statement-00.txt<BR><SPAN =
class=3DApple-tab-span=20
    style=3D"WHITE-SPACE: pre"></SPAN>Pages=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: =
9<BR><SPAN=20
    class=3DApple-tab-span style=3D"WHITE-SPACE: pre"></SPAN>Date=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;:=20
    2009-03-02<BR><BR>This document captures an outline of the =
interoperability=20
    statements<BR>that will be collected to construct an =
interoperability report=20
    for<BR>RFC 3261. &nbsp;The outline is stil under review and should =
not be=20
    treated<BR>as complete, but will drive data collection at=20
    upcoming<BR>interoperability events.<BR><BR>A URL for this =
Internet-Draft=20
    is:<BR><A=20
    =
href=3D"http://www.ietf.org/internet-drafts/draft-sparks-sip-3261-interop=
-statement-00.txt">http://www.ietf.org/internet-drafts/draft-sparks-sip-3=
261-interop-statement-00.txt</A><BR><BR>Internet-Drafts=20
    are also available by anonymous FTP=20
    at:<BR>ftp://ftp.ietf.org/internet-drafts/<BR><BR>Below is the data =
which=20
    will enable a MIME compliant mail reader<BR>implementation to =
automatically=20
    retrieve the ASCII version of=20
  =
the<BR>Internet-Draft.<BR></DIV></BLOCKQUOTE></DIV></BLOCKQUOTE></BODY></=
HTML>

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From christer.holmberg@ericsson.com  Tue Mar  3 04:58:10 2009
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Thread-Topic: [Sip] New version (-05) of the 199 draft - ready for WGLC?
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Dean Willis" <dean.willis@softarmor.com>, "SIP WG" <sip@ietf.org>
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Subject: Re: [Sip] New version (-05) of the 199 draft - ready for WGLC?
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Please note that the latest version is -06.

Regards,

Christer=20

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]=20
> Sent: 3. maaliskuuta 2009 9:59
> To: SIP WG
> Cc: Keith Drage; Christer Holmberg
> Subject: Re: [Sip] New version (-05) of the 199 draft - ready=20
> for WGLC?
>=20
>=20
> We didn't get any real answers due to a bit of confusion=20
> between this and keepalive. So, let's ask the question again.
>=20
> Is 199 ready for WGLC?
>=20
> --
> dean
>=20
>=20
> On Feb 19, 2009, at 5:12 AM, Christer Holmberg wrote:
>=20
> >
> > Hi,
> >
> > I've submitted a new version of draft-ietf-sip-199. I have done=20
> > changes and updates based on feedback and discussions=20
> (please let me=20
> > know if you think I've forgotten something), and I've also=20
> added text=20
> > to the security chapter.
> >
> > I propose that we start the WGLC process for the draft.
> >
> > The draft can also be found at:
> >
> > http://users.piuha.net/cholmber/drafts/draft-ietf-sip-199-05.txt
> >
> > Regards,
> >
> > Christer
> >
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
>=20
>=20

From brett@broadsoft.com  Tue Mar  3 06:10:36 2009
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To: Dean Willis <dean.willis@softarmor.com>, SIP WG <sip@ietf.org>
Date: Tue, 3 Mar 2009 06:10:56 -0800
Thread-Topic: [Sip] New version (-05) of the 199 draft - ready for WGLC?
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Subject: Re: [Sip] New version (-05) of the 199 draft - ready for WGLC?
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Yes; I think draft-ietf-sip-199-06 is ready for WGLC.

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Dea=
n
> Willis
> Sent: Tuesday, March 03, 2009 2:59 AM
> To: SIP WG
> Cc: Keith Drage; Christer Holmberg
> Subject: Re: [Sip] New version (-05) of the 199 draft - ready for WGLC?
>=20
>=20
> We didn't get any real answers due to a bit of confusion between this
> and keepalive. So, let's ask the question again.
>=20
> Is 199 ready for WGLC?
>=20
> --
> dean


From drage@alcatel-lucent.com  Tue Mar  3 06:59:27 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: Brett Tate <brett@broadsoft.com>, SIP IETF <sip@ietf.org>
Date: Tue, 3 Mar 2009 15:59:40 +0100
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Cc: "hkaplan@acmepacket.com" <hkaplan@acmepacket.com>, "christer.holmberg@ericsson.com" <christer.holmberg@ericsson.com>
Subject: Re: [Sip] WGLC on draft-ietf-sip-info-events
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Frequently the editor's wait until the end of the WGLC before processing th=
e comments, so I am sure that it has not been forgotten.

It does have the advantage of making sure that we do not swap and change wh=
en we get conflicting comments.

regards

Keith=20

> -----Original Message-----
> From: Brett Tate [mailto:brett@broadsoft.com]=20
> Sent: Monday, March 02, 2009 1:19 PM
> To: DRAGE, Keith (Keith); SIP IETF
> Cc: hkaplan@acmepacket.com; christer.holmberg@ericsson.com;=20
> eburger@standardstrack.com
> Subject: RE: WGLC on draft-ietf-sip-info-events
>=20
> > A reminder of the current WGLC on this document, which completes=20
> > Wednesday.
> >=20
> > Please submit your comments as soon as possible.
> >=20
> > If you are reviewing the document, and you consider there may be a=20
> > delay in your comments appearing, then please send a mail to the=20
> > chairs so they can ascertain whether more time is needed.
>=20
> I haven't received a reply concerning my February 16=20
> (archived on 17) email: "draft-ietf-sip-info-events-03:=20
> comments and questions".  Thus I'm not sure if I will have=20
> follow-up comments based upon the reply.
>=20
> http://www.ietf.org/mail-archive/web/sip/current/msg26621.html
>=20
> The following is a snapshot of my questions and comments in=20
> case the authors did not see my prior email.
>=20
> Thanks,
> Brett
>=20
> ----
>=20
> Section 3.1 paragraph 1:  Concerning "MUST advertise" and=20
> "provisional 1xx", please clarify that the "MUST advertise"=20
> is not applicable to the 100 response.  Does the "MUST=20
> advertise" and "provisional 1xx" apply to all (first and=20
> subsequent within dialog) INVITE non 101-199 (intentionally=20
> indicated 199)?  Concerning ACK for initial INVITE's 2xx,=20
> does the "MUST advertise" always apply?
>=20
> Section 3.1 paragraph 2 sentences 1 and 2: Sentence 1=20
> indicates "session parameters" and sentence 2 indicates=20
> "dialog parameters" concerning Info Package negotiation, I'm=20
> not sure when the negotiation/renegotiation occurs.  Does it=20
> follow SDP offer/answer rules or something else; if something=20
> else, what?  I have the same question concerning "dialog=20
> refresh" text discussed later within the same section since=20
> incorrect interpretations might trigger deactivation.
>=20
> Section 4.3: When sending a 489, it indicates "MUST terminate=20
> the INVITE dialog".  The "MUST terminate" seems severe even=20
> if all the negotiation race conditions are adequately=20
> addressed.  The mentioned "protocol failure" reason seems=20
> contrary to the "strictly send and lenient receive" mantra.
>=20
> Section 5.1 Table 1: The usage of "g" seems potentially=20
> outdated or incorrectly reference since discussed within RFC=20
> 2543 instead of RFC 3261.  The same applies to usage Via row=20
> concerning "(2)".
>=20
> Section 5.2.2: The "general-header" reference is outdated or=20
> incorrectly reference since defined within RFC 2543 instead=20
> of RFC 3261.  RFC 3261 uses "Message-Header".
>=20
> Section 7.7 paragraph 2: The following potentially could be=20
> reworded since it is okay to return failure responses (401,=20
> 503, 500, ...) even when overlapping occurs: "in the event a=20
> UAC issues overlapping INFO requests for an Info Package, the=20
> UAS MUST NOT return an error response".
>=20
> Section 8: Why specifically defining and using "nil"?  I'm=20
> mainly just curious of the benefit over not having defined=20
> "nil" for similar use within the Supported header.
>=20
> =

From vkg@alcatel-lucent.com  Tue Mar  3 07:02:03 2009
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Subject: Re: [Sip] New version (-05) of the 199 draft - ready for WGLC?
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Dean Willis wrote:
> We didn't get any real answers due to a bit of confusion between this 
> and keepalive. So, let's ask the question again.
> 
> Is 199 ready for WGLC?

Yes; it is well written (I had looked at a much earlier version
and just re-read -06.)

One question for Christer:

- Does 199 carry a message/sipfrag MIME type that conveys the
  nature of the final non-2xx response?  This may have been
  discussed in the past, and if so, please let me know.  I
  do note that rfc3420 is specified in the draft as a normative
  reference, but it is not referred to anywhere in the body
  of the draft.

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From john.elwell@siemens.com  Tue Mar  3 08:46:15 2009
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From: "Elwell, John" <john.elwell@siemens.com>
To: SIP List <sip@ietf.org>, Cullen Jennings <fluffy@cisco.com>, Jon Peterson <jon.peterson@neustar.biz>
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Draft 03:
http://tools.ietf.org/html/draft-elwell-sip-e2e-identity-important-03
includes for the first time an analysis of changes to SIP requests by
B2BUAs (section 8). Also the Conclusions (section 9) have been extended.

I and a number of people who have assisted in revising this draft would
like to seek consensus on a number of the points. We aim to ask these
questions in San Francisco, but obviously it would help to get opinions
from the list in advance.

1. Tolerance to changes of certain elements as a SIP request is
forwarded through intermediate domains.

Section 8 analyses a number of elements of SIP messages that are signed
by the present RFC 4474 solution to end-to-end (or near end-to-end)
authenticated identification. For some elements it claims that any
solution for end-to-end authenticated identification needs to be
tolerant of changes by intermediate domains, since there appear to be
valid reasons why B2BUAs (including but not limited to SBCs) change
these elements. Picking on some of these:

A) Is there agreement that a solution needs to be tolerant of changes to
IP addresses and ports in c- and m-lines in SDP? In other words, that
intermediate domains can legitimately modify these fields (e.g., for
media steering) and a solution is needed for end-to-end authenticated
identification in such circumstances?

B) Is there agreement that a solution needs to be tolerant of changes to
Contact (addr-spec), Call-Id (call-id) and CSeq (digit) header fields?
In other words, that intermediate domains can legitimately modify these
fields (e.g., for topology hiding) and a solution is needed for
end-to-end authenticated identification in such circumstances?

C) Is there agreement that a solution does not need to be tolerant of
changes to the addr-spec of From and To header fields? In other words,
that intermediate domains should not modify these fields and can expect
to break end-to-end authenticated identification if they do so?


2. Are people in broad agreement with the conclusions in section 9
concerning the importance of end-to-end identification and in particular
end-to-end authenticated identification?

John


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Registration for SIPit 24 is open!
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From christer.holmberg@ericsson.com  Tue Mar  3 09:07:25 2009
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Thread-Topic: [Sip] New version (-05) of the 199 draft - ready for WGLC?
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Vijay K. Gurbani" <vkg@alcatel-lucent.com>, "Dean Willis" <dean.willis@softarmor.com>
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Subject: Re: [Sip] New version (-05) of the 199 draft - ready for WGLC?
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Hi Vijay,=20

>One question for Christer:
>
>- Does 199 carry a message/sipfrag MIME type that conveys the
>  nature of the final non-2xx response?  This may have been
>  discussed in the past, and if so, please let me know.  I
>  do note that rfc3420 is specified in the draft as a normative
>  reference, but it is not referred to anywhere in the body
>  of the draft.

We discussed it quite much, and the outcome was a compromise: we don't
say anything :)

So, if someone wants to carry message/sipfrag in the 199 it is allowed,
but the base 199 draft does not require it.

Regards,

Christer


From BPenfield@acmepacket.com  Tue Mar  3 09:11:22 2009
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From: Bob Penfield <BPenfield@acmepacket.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, SIP IETF <sip@ietf.org>
Date: Tue, 3 Mar 2009 12:11:46 -0500
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Here are some comments on the INFO draft:

Section 2 says:

   Each UA enumerates which Info Packages it can receive.  If the far
   end indicates it can receive a package offered by the near end, the
   near end can send INFO methods containing the payload for that
   package.

"offered" needs to changed to "supported" since an INFO sender does
not indicate what packages it can send.

Section 3 is called "Info Package Negotiation", but since we removed
Send-Info, it is not a negotiation since there is no feedback from the
peer UA. It is simply a declaration of receive capability. At first
read, I was totally confused by section 3.1 because I was expecting
some actual negotiation given the above statement in Section 2. There
are a few other places in the document that mentions "negotiation"
that also might need to be changed.

I think the usage of UAC/UAS terminology in Section 3.1 is still
confusing. I would prefer to see something like:

   A UA supporting this document MUST advertise the set of Info
   Packages it is willing to receive in Recv-Info header(s) in all
   "INVITE dialog usage" requests and responses (101-199 and 2xx only)
   for dialog establishment or target refresh. This includes INVITE,
   UPDATE, PRACK and ACK and their non-failure responses.

   A UA MUST NOT send INFO requests for a given INFO package until the
   UA has received an "INVITE dialog usage" request or response (for
   dialog establishment or target refresh) with a Recv-Info header
   listing the given Info Package.

   A UA MUST cease sending INFO requests for a given INFO package when
   the UA receives an "INVITE dialog usage" request or response (for
   dialog establishment or target refresh) that does not contain a
   Recv-Info header listing the given Info Package.

Section 3.1 - "the target UA may not send form package P"

   s/form/from/

Section 3.1 - "... the other UA in the dial will assume the ..."

  s/dial/dialog/

Section 3.1 - "server MAY terminate the session with a CANCEL or BYE"

  When the "server" is the UAS of the request, CANCEL is not
  appropriate, and BYE would not be appropriate if it is an initial
  INVITE. In this case, the request would need to be rejected. When
  the "server" is the UAC, I don't think CANCEL is appropriate since
  it is the far-end UA of a specific dialog that the UAC has an issue
  with, not all possible UAs in the case of an initial INVITE.

Section 7 - Can Info Packages strengthen "MAY" to "SHOULD"?

Section 8 - Why is the separator between Info-package-name and
Info-package-param a DOT? Shouldn't it be a SEMI? I don't think that
Info-package-param is similar to event-template in 3265. Its more like
event-param.

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To: "Vijay K. Gurbani" <vkg@alcatel-lucent.com>, Dean Willis <dean.willis@softarmor.com>
Date: Tue, 3 Mar 2009 09:19:20 -0800
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Subject: Re: [Sip] New version (-05) of the 199 draft - ready for WGLC?
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Vijay K. Gurbani
> Sent: Tuesday, March 03, 2009 7:02 AM
> To: Dean Willis
> Cc: SIP WG; Christer Holmberg
> Subject: Re: [Sip] New version (-05) of the 199 draft - ready for WGLC?
>=20
> Dean Willis wrote:
> > We didn't get any real answers due to a bit of confusion between this
> > and keepalive. So, let's ask the question again.
> >
> > Is 199 ready for WGLC?
>=20
> Yes; it is well written (I had looked at a much earlier version
> and just re-read -06.)
>=20
> One question for Christer:
>=20
> - Does 199 carry a message/sipfrag MIME type that conveys the
>   nature of the final non-2xx response?  This may have been
>   discussed in the past, and if so, please let me know.  I
>   do note that rfc3420 is specified in the draft as a normative
>   reference, but it is not referred to anywhere in the body
>   of the draft.

There was discussion around 199 carrying information about why the fork/dia=
log was terminated and sipfrag was mentioned to be used to convey that. How=
ever, later on Reason header within 199 is agreed to be used for that purpo=
se. Hence, mention of 3420 might be obsolete by now.

Thanks,
Serhad=20


>=20
> Thanks,
>=20
> - vijay
> --
> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
> 1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
> Web:   http://ect.bell-labs.com/who/vkg/
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

From dworley@nortel.com  Tue Mar  3 10:26:27 2009
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From: "Dale Worley" <dworley@nortel.com>
To: SIP <sip@ietf.org>
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Date: Tue, 03 Mar 2009 13:26:47 -0500
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Subject: Re: [Sip] Question regarding conflicting grammar for IPV6 SIP URI andRFC	3986
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I would like to resume the discussion of the problem of IPv6 host
addresses in SIP URIs.

To abbreviate my previous message, SIP URIs appear to be intended to be
a subset of RFC 3986 "path-rootless" URIs (as are mailto: URIs).  Of the
characters that RFC 3261 permits in the user/host part of the URI, which
corresponds to the "hier-part" of RFC 3986, the following characters are
not allowed by the syntax of RFC 3986:

? (permitted in user-parts)
/ (permitted in user-parts)
[ (permitted as part of an IPv6 address)
] (permitted as part of an IPv6 address)

In order to straighten out this problem, I propose that we modify the
RFC 3261 definition to bring it into line with RFC 3986.  Specifically,
we should:

1. Remove "?" and "/" from the user-unreserved set.  (These characters
can still be used in their escaped form, which is equivalent to their
unescaped form under the current rules.)

2. Replace "[" and "]" in host-addresses by their escaped forms, "%5B"
and "%5D" (as is done in mailto: URIs).

This will inconvenience any implementation currently using the two
characters in user-names, or using IPv6 addresses.  But it avoids having
to change the master URI syntax of RFC 3986, which is a task of much
broader scope.

Comments?

Dale



From adam@nostrum.com  Tue Mar  3 14:56:00 2009
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Subject: [Sip] RFC3265bis
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After 6 years of testing at SIPits and other experience in the field, it 
feels like time to fix some of the known bugs in RFC 3265. I've put 
together a 3265bis that addresses most of the bugs that have been 
reported to date (although some are called out as known issues):

http://www.ietf.org/internet-drafts/draft-roach-sipcore-rfc3265bis-00.txt

If you're already familiar with RFC 3265, I would suggest using Appendix 
C as your roadmap to navigating the changes. The areas I know I need 
feedback on are the open issues listed in Appendix B. Of course, it's 
always possible that the new document will prompt people to share 
implementation experience that hasn't yet been captured -- I'm happy to 
hear about that too.

/a

From eburger@standardstrack.com  Tue Mar  3 15:10:13 2009
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Cc: SIP IETF <sip@ietf.org>
Subject: Re: [Sip] WGLC on draft-ietf-sip-info-events
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It absolutely has not been forgotten: I was hoping to get more  
comments, but I'll get back with comments to the comments by the end  
of the week.

Thanks for your read of the document!

On Mar 3, 2009, at 6:59 AM, DRAGE, Keith (Keith) wrote:

> Frequently the editor's wait until the end of the WGLC before  
> processing the comments, so I am sure that it has not been forgotten.
>
> It does have the advantage of making sure that we do not swap and  
> change when we get conflicting comments.
>
> regards
>
> Keith
>
>> -----Original Message-----
>> From: Brett Tate [mailto:brett@broadsoft.com]
>> Sent: Monday, March 02, 2009 1:19 PM
>> To: DRAGE, Keith (Keith); SIP IETF
>> Cc: hkaplan@acmepacket.com; christer.holmberg@ericsson.com;
>> eburger@standardstrack.com
>> Subject: RE: WGLC on draft-ietf-sip-info-events
>>
>>> A reminder of the current WGLC on this document, which completes
>>> Wednesday.
>>>
>>> Please submit your comments as soon as possible.
>>>
>>> If you are reviewing the document, and you consider there may be a
>>> delay in your comments appearing, then please send a mail to the
>>> chairs so they can ascertain whether more time is needed.
>>
>> I haven't received a reply concerning my February 16
>> (archived on 17) email: "draft-ietf-sip-info-events-03:
>> comments and questions".  Thus I'm not sure if I will have
>> follow-up comments based upon the reply.
>>
>> http://www.ietf.org/mail-archive/web/sip/current/msg26621.html
>>
>> The following is a snapshot of my questions and comments in
>> case the authors did not see my prior email.
>>
>> Thanks,
>> Brett
>>
>> ----
>>
>> Section 3.1 paragraph 1:  Concerning "MUST advertise" and
>> "provisional 1xx", please clarify that the "MUST advertise"
>> is not applicable to the 100 response.  Does the "MUST
>> advertise" and "provisional 1xx" apply to all (first and
>> subsequent within dialog) INVITE non 101-199 (intentionally
>> indicated 199)?  Concerning ACK for initial INVITE's 2xx,
>> does the "MUST advertise" always apply?
>>
>> Section 3.1 paragraph 2 sentences 1 and 2: Sentence 1
>> indicates "session parameters" and sentence 2 indicates
>> "dialog parameters" concerning Info Package negotiation, I'm
>> not sure when the negotiation/renegotiation occurs.  Does it
>> follow SDP offer/answer rules or something else; if something
>> else, what?  I have the same question concerning "dialog
>> refresh" text discussed later within the same section since
>> incorrect interpretations might trigger deactivation.
>>
>> Section 4.3: When sending a 489, it indicates "MUST terminate
>> the INVITE dialog".  The "MUST terminate" seems severe even
>> if all the negotiation race conditions are adequately
>> addressed.  The mentioned "protocol failure" reason seems
>> contrary to the "strictly send and lenient receive" mantra.
>>
>> Section 5.1 Table 1: The usage of "g" seems potentially
>> outdated or incorrectly reference since discussed within RFC
>> 2543 instead of RFC 3261.  The same applies to usage Via row
>> concerning "(2)".
>>
>> Section 5.2.2: The "general-header" reference is outdated or
>> incorrectly reference since defined within RFC 2543 instead
>> of RFC 3261.  RFC 3261 uses "Message-Header".
>>
>> Section 7.7 paragraph 2: The following potentially could be
>> reworded since it is okay to return failure responses (401,
>> 503, 500, ...) even when overlapping occurs: "in the event a
>> UAC issues overlapping INFO requests for an Info Package, the
>> UAS MUST NOT return an error response".
>>
>> Section 8: Why specifically defining and using "nil"?  I'm
>> mainly just curious of the benefit over not having defined
>> "nil" for similar use within the Supported header.
>>
>>


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From: Bob Penfield <BPenfield@acmepacket.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Date: Tue, 3 Mar 2009 19:57:53 -0500
Thread-Topic: [Sip] I-D Action:draft-ietf-sip-199-06.txt
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Subject: Re: [Sip] I-D Action:draft-ietf-sip-199-06.txt
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Hi Christer,

I have few comments on draft-ietf-sip-199-06.txt. I apologize if any of the=
se are duplicates.

Section 6. The statement requiring the proxy to keep track of early dialogs=
:

   A proxy which supports generating of 199 response codes MUST keep
   track of early dialogs, in order to determine whether to generate a
   199 response when the proxy receives a non-2xx final response.  In
   addition, the proxy MUST keep track on which early dialogs it has
   received and forwarded 199 responses, in order to not generate
   additional 199 responses for those early dialogs.

Should precede the statement:

   If the forking proxy generates a 199 response, and if the
   forking proxy is able to identify additional early dialogs which are
   terminated by the same non-2xx final response, it MUST also generate
   and send a 199 response upstream for each of those early dialogs,

In fact, if the proxy is keeping track of early dialogs, under what
circumstances would it not be "able to identify additional early
dialogs which are terminated by the same non-2xx final response"?

Section 8 (199 Early Dialog Terminated) seems out of place. Shouldn't
this be in the introduction? or in an "Overview Of Operation" between
sections 3 and 4?

The text in Section 9 (Usage with SDP offer/answer) should be in
section 5 (User Agent Server behavior).

The statement in section 10 recommending that the UAS send 199
unreliably should be in section 5.

The statement in section 10 forbidding proxies from sending 199
reliably should be in section 6 (Proxy behavior).

Cheers,
(-:bob

From john.elwell@siemens.com  Wed Mar  4 00:46:51 2009
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From: "Elwell, John" <john.elwell@siemens.com>
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Thread-Topic: [Sip] WGLC on draft-ietf-sip-info-events
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Subject: Re: [Sip] WGLC on draft-ietf-sip-info-events
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I agree with Bob that the use of UAC and UAS terminology is confusing. I
cite some particular examples below.

1. In 3 it defines the two terms, but early in 3.1 it states "The
initiating UA (UAC of the INVITE)", which uses "UAC" in a sense
different from what is defined.

2. "the UAC MUST NOT send
   INFO requests for a given Info Package until the UAS lists the given
   Info Package in a Recv-Info header."
According to the definition of UAC, a UAC is the sender of an INFO
request. So while waiting to receive Recv-Info it is not a UAC.
Similarly, the other end is not a UAS by virtue of sending "Recv-Info" -
it only becomes a UAS when it receives an INFO request. Don't forget
that a UA is a UAC or a UAS only in the context of a particular
transaction, and before that transaction takes place and after that
transaction has finished it is just a UA. Any attempt to change the
meaning of UAC or UAS would be a departure from RFC 3261.

3. "A UAS lists multiple packages by enumerating the package name(s),
   separated by commas, as values for the Recv-Info header in the
   session establishment exchange.  A UAS can also list multiple
   packages by including multiple Recv-Info headers...."
We are talking about sending Recv-Info - nothing to do with an INFO
transaction, so the term UAS is inappropriate.

4. "This initial dialog establishment
   phase is the only time the UAS need be prepared to receive multiple
   INFO requests, as we require post-session-establishment negotiation
   to fully complete before a UAC can send an INFO request."
followed shortly afterwards by:
"A UAC MAY send an INFO
   request on both an early and confirmed dialog."
Either these are conflicting statements or there is some mix-up of terms
UAC and UAS.

5. In "Conventions Used in this Document" it states: "Since this
document strictly follows RFC 3261, we refer to the UA
   that issues the INVITE as the "initiating UA" and the UA that
   responds to the INVITE as the "target UA" to remove any confusion."
I don't think this is good practice, and I would even say that by doing
this we no longer strictly follow RFC 3261 terminology. If it is clear
we are talking about an INVITE transaction we should be free to use UAC
and UAS.

So in general, where it is clear which transaction we are talking about
(e.g., because it has appeared earlier in the paragraph concerned), we
should just use UAC or UAS. Otherwise we should use expressions like
"the UAC for the XXX transaction" or "the UA that sends the XXX
request".

John

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Bob Penfield
> Sent: 03 March 2009 17:12
> To: DRAGE, Keith (Keith); SIP IETF
> Subject: Re: [Sip] WGLC on draft-ietf-sip-info-events
>=20
>=20
> Here are some comments on the INFO draft:
>=20
> Section 2 says:
>=20
>    Each UA enumerates which Info Packages it can receive.  If the far
>    end indicates it can receive a package offered by the near end, the
>    near end can send INFO methods containing the payload for that
>    package.
>=20
> "offered" needs to changed to "supported" since an INFO sender does
> not indicate what packages it can send.
>=20
> Section 3 is called "Info Package Negotiation", but since we removed
> Send-Info, it is not a negotiation since there is no feedback from the
> peer UA. It is simply a declaration of receive capability. At first
> read, I was totally confused by section 3.1 because I was expecting
> some actual negotiation given the above statement in Section 2. There
> are a few other places in the document that mentions "negotiation"
> that also might need to be changed.
>=20
> I think the usage of UAC/UAS terminology in Section 3.1 is still
> confusing. I would prefer to see something like:
>=20
>    A UA supporting this document MUST advertise the set of Info
>    Packages it is willing to receive in Recv-Info header(s) in all
>    "INVITE dialog usage" requests and responses (101-199 and 2xx only)
>    for dialog establishment or target refresh. This includes INVITE,
>    UPDATE, PRACK and ACK and their non-failure responses.
>=20
>    A UA MUST NOT send INFO requests for a given INFO package until the
>    UA has received an "INVITE dialog usage" request or response (for
>    dialog establishment or target refresh) with a Recv-Info header
>    listing the given Info Package.
>=20
>    A UA MUST cease sending INFO requests for a given INFO package when
>    the UA receives an "INVITE dialog usage" request or response (for
>    dialog establishment or target refresh) that does not contain a
>    Recv-Info header listing the given Info Package.
>=20
> Section 3.1 - "the target UA may not send form package P"
>=20
>    s/form/from/
>=20
> Section 3.1 - "... the other UA in the dial will assume the ..."
>=20
>   s/dial/dialog/
>=20
> Section 3.1 - "server MAY terminate the session with a CANCEL or BYE"
>=20
>   When the "server" is the UAS of the request, CANCEL is not
>   appropriate, and BYE would not be appropriate if it is an initial
>   INVITE. In this case, the request would need to be rejected. When
>   the "server" is the UAC, I don't think CANCEL is appropriate since
>   it is the far-end UA of a specific dialog that the UAC has an issue
>   with, not all possible UAs in the case of an initial INVITE.
>=20
> Section 7 - Can Info Packages strengthen "MAY" to "SHOULD"?
>=20
> Section 8 - Why is the separator between Info-package-name and
> Info-package-param a DOT? Shouldn't it be a SEMI? I don't think that
> Info-package-param is similar to event-template in 3265. Its more like
> event-param.
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

From rco@iptel.org  Wed Mar  4 02:36:12 2009
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Subject: [Sip] draft-state-sip-relay-attack-00
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Hello,

a new internet draft has been published concerning the relay attack on 
digest authentication and SIP. The attack itself has been first 
disclosed 2 years ago by the maydnes team from the french INRIA. Until 
now, no document has been pushlished that documents the attack and 
provides guidance to SIP operators or handset manufacturers.

http://tools.ietf.org/html/draft-state-sip-relay-attack-00

The appropriate mitigations of problem resolutions are still not 100% 
clear. We hope that this draft can help start a discussion on how to 
best resolve this problem.


Regards,

Raphael Coeffic.
(on behalf of all the authors of this draft)

---------------------------------------------------------------------------------------------------

Filename:	   draft-state-sip-relay-attack
Version:	   00
Staging URL:	   http://www3.ietf.org/proceedings/staging/draft-state-sip-relay-attack-00.txt
Title:		   SIP digest authentication relay attack
Creation_date:	   2009-03-02
WG ID:		   Indvidual Submission
Number_of_pages: 18
Abstract:
The Session Initiation Protocol (SIP [RFC3261]) provides a mechanism
for creating, modifying, and terminating sessions with one or more
participants.  This document describes a vulnerability of SIP
combined with HTTP Digest Access Authentication [RFC2617] through
which an attacker can leverage the victim's credentials to send
authenticated requests on his behalf.  This attack is different from
the man-in-the-middle (MITM) attack and does not require any
eavesdropping, DNS or IP spoofing.




From theo@crazygreek.co.uk  Wed Mar  4 04:34:34 2009
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From: Theo Zourzouvillys <theo@crazygreek.co.uk>
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On Tue, Mar 3, 2009 at 6:26 PM, Dale Worley <dworley@nortel.com> wrote:

> But it avoids having
> to change the master URI syntax of RFC 3986, which is a task of much
> broader scope.

Both are going to require change (although we don't need world AND dog
to agree with a 3261 update), but seems to me that updating 3986 seems
like a far more appropriate path to take given the number of other
affected URI schemes as Hadriel pointed out.

I have no objection to the removal of '/' and '?' from the user part,
but the replacing of '[' and ']' seems quiet urgh.

... i've just quickly scanned through apps-discuss and haven't seen
any discussion on it there, should we perhaps test the water to see
what sort of reaction there would be to suggesting such an update if
it's the right thing to do?

 ~ Theo

From gshaham@juniper.net  Wed Mar  4 06:35:56 2009
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From: Gilad Shaham <gshaham@juniper.net>
To: "Elwell, John" <john.elwell@siemens.com>, "SIP List" <sip@ietf.org>, "Cullen Jennings" <fluffy@cisco.com>, "Jon Peterson" <jon.peterson@neustar.biz>
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Subject: Re: [Sip] Seeking consensus on draft-elwell-sip-e2e-identity-important
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Hi John,

I personally think your conclusions are correct. One comment I have is
regarding From and To headers. It's possible that the comment is a bit
ahead of time as this draft is not yet aimed at the solutions, but I
believe that when we come to a solution, the endpoints themselves should
enforce some rules on the From and To headers if they want to support
end-to-end identity.
For example, the endpoints should not (or MUST not) place an IP address
in these fields if they request end-to-end identity to be enabled.
Making this a requirement on the endpoint behalf would help coping with
the dilemma of the intermediate device as to whether or not it should
break the identity.

Thanks,
Gilad

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Elwell, John
Sent: Tuesday, March 03, 2009 6:47 PM
To: SIP List; Cullen Jennings; Jon Peterson
Subject: [Sip] Seeking consensus on
draft-elwell-sip-e2e-identity-important

Draft 03:
http://tools.ietf.org/html/draft-elwell-sip-e2e-identity-important-03
includes for the first time an analysis of changes to SIP requests by
B2BUAs (section 8). Also the Conclusions (section 9) have been extended.

I and a number of people who have assisted in revising this draft would
like to seek consensus on a number of the points. We aim to ask these
questions in San Francisco, but obviously it would help to get opinions
from the list in advance.

1. Tolerance to changes of certain elements as a SIP request is
forwarded through intermediate domains.

Section 8 analyses a number of elements of SIP messages that are signed
by the present RFC 4474 solution to end-to-end (or near end-to-end)
authenticated identification. For some elements it claims that any
solution for end-to-end authenticated identification needs to be
tolerant of changes by intermediate domains, since there appear to be
valid reasons why B2BUAs (including but not limited to SBCs) change
these elements. Picking on some of these:

A) Is there agreement that a solution needs to be tolerant of changes to
IP addresses and ports in c- and m-lines in SDP? In other words, that
intermediate domains can legitimately modify these fields (e.g., for
media steering) and a solution is needed for end-to-end authenticated
identification in such circumstances?

B) Is there agreement that a solution needs to be tolerant of changes to
Contact (addr-spec), Call-Id (call-id) and CSeq (digit) header fields?
In other words, that intermediate domains can legitimately modify these
fields (e.g., for topology hiding) and a solution is needed for
end-to-end authenticated identification in such circumstances?

C) Is there agreement that a solution does not need to be tolerant of
changes to the addr-spec of From and To header fields? In other words,
that intermediate domains should not modify these fields and can expect
to break end-to-end authenticated identification if they do so?


2. Are people in broad agreement with the conclusions in section 9
concerning the importance of end-to-end identification and in particular
end-to-end authenticated identification?

John

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip

From dean.willis@softarmor.com  Wed Mar  4 10:04:20 2009
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From: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: [Sip] Question regarding conflicting grammar for IPV6 SIP URI andRFC 3986
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On Mar 4, 2009, at 6:34 AM, Theo Zourzouvillys wrote:

> On Tue, Mar 3, 2009 at 6:26 PM, Dale Worley <dworley@nortel.com>  
> wrote:
>
>> But it avoids having
>> to change the master URI syntax of RFC 3986, which is a task of much
>> broader scope.
>
> Both are going to require change (although we don't need world AND dog
> to agree with a 3261 update), but seems to me that updating 3986 seems
> like a far more appropriate path to take given the number of other
> affected URI schemes as Hadriel pointed out.
>
> I have no objection to the removal of '/' and '?' from the user part,
> but the replacing of '[' and ']' seems quiet urgh.
>
> ... i've just quickly scanned through apps-discuss and haven't seen
> any discussion on it there, should we perhaps test the water to see
> what sort of reaction there would be to suggesting such an update if
> it's the right thing to do?


I'm afraid I have to concur with Theo. 3986 is just broken.

--
Dean


From dworley@nortel.com  Wed Mar  4 10:28:08 2009
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Subject: [Sip] SIP Tracing Facility
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I wrote up the "170" trace proposal as an I-D:

A new version of I-D, draft-worley-trace-00.txt has been successfuly
submitted by Dale Worley and posted to the IETF repository.

Filename:        draft-worley-trace
Revision:        00
Title:           SIP Tracing Facility
Creation_date:   2009-03-03
WG ID:           Independent Submission
Number_of_pages: 18

Abstract:
This document defines a SIP option tag, "trace", to be used within
SIP messages to request that SIP elements (both proxies and UASs)
that receive the message reflect to the UAC the request they received
and the response they gave by encapsulating the request and response
in a provisional response.  A new provisional response code "170" is
defined to carry the request and response.  This option tag is
expected to be used solely for diagnostic purposes.



From drage@alcatel-lucent.com  Wed Mar  4 10:32:11 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: SIP IETF <sip@ietf.org>
Date: Wed, 4 Mar 2009 19:32:32 +0100
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Subject: [Sip] Agenda call for SIP
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(As WG co chair)

If you wish to have time allocated to a particular discussion in the SIP se=
ssions in San Francisco, and have not already submitted a request, then ple=
ase do so.

Please include information like:

-	name(s) of any draft(s) to be discussed.
-	proposed name of presenter
-	what is/are the key question(s) to put to the WG as a result of the discu=
ssion.
-	how much time you think you will need.

Note that in addressing the third bullet we would like a bit more detail th=
an "WG item?", for example give us some help in identifying the text for th=
e milestone you actually want in the SIP charter, which is the problem you =
want to solve, rather than the solution.

Please do this now as we will be putting the preliminary agenda together to=
morrow (Thursday). Agendas are always subject to change but would prefer no=
t to have any late surprises.

regards

Keith


From dean.willis@softarmor.com  Wed Mar  4 10:51:43 2009
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Note that I'm accumulating agenda requests at our (yes, Henry, we one)  
wiki:

http://www.softarmor.com/mediawiki/index.php/SIP_Agenda_Requests_IETF_74

If your request doesn't show up there fairly quickly (48 hours), I've  
probably lost it.

--
Dean




From john.elwell@siemens.com  Wed Mar  4 12:43:48 2009
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From: "Elwell, John" <john.elwell@siemens.com>
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=20

> -----Original Message-----
> From: Gilad Shaham [mailto:gshaham@juniper.net]=20
> Sent: 04 March 2009 14:32
> To: Elwell, John; SIP List; Cullen Jennings; Jon Peterson
> Subject: RE: [Sip] Seeking consensus on=20
> draft-elwell-sip-e2e-identity-important
>=20
> Hi John,
>=20
> I personally think your conclusions are correct. One comment I have is
> regarding From and To headers. It's possible that the comment is a bit
> ahead of time as this draft is not yet aimed at the solutions, but I
> believe that when we come to a solution, the endpoints=20
> themselves should
> enforce some rules on the From and To headers if they want to support
> end-to-end identity.
> For example, the endpoints should not (or MUST not) place an=20
> IP address
> in these fields if they request end-to-end identity to be enabled.
> Making this a requirement on the endpoint behalf would help=20
> coping with
> the dilemma of the intermediate device as to whether or not it should
> break the identity.
[JRE] That would be good, but on the other hand this sort of fix-up can
in practice be done in the originating domain prior to the
authentication service. In fact replacing an IP address with a domain
name must be done before the authentication service.

John

>=20
> Thanks,
> Gilad
>=20
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Elwell, John
> Sent: Tuesday, March 03, 2009 6:47 PM
> To: SIP List; Cullen Jennings; Jon Peterson
> Subject: [Sip] Seeking consensus on
> draft-elwell-sip-e2e-identity-important
>=20
> Draft 03:
> http://tools.ietf.org/html/draft-elwell-sip-e2e-identity-important-03
> includes for the first time an analysis of changes to SIP requests by
> B2BUAs (section 8). Also the Conclusions (section 9) have=20
> been extended.
>=20
> I and a number of people who have assisted in revising this=20
> draft would
> like to seek consensus on a number of the points. We aim to ask these
> questions in San Francisco, but obviously it would help to=20
> get opinions
> from the list in advance.
>=20
> 1. Tolerance to changes of certain elements as a SIP request is
> forwarded through intermediate domains.
>=20
> Section 8 analyses a number of elements of SIP messages that=20
> are signed
> by the present RFC 4474 solution to end-to-end (or near end-to-end)
> authenticated identification. For some elements it claims that any
> solution for end-to-end authenticated identification needs to be
> tolerant of changes by intermediate domains, since there appear to be
> valid reasons why B2BUAs (including but not limited to SBCs) change
> these elements. Picking on some of these:
>=20
> A) Is there agreement that a solution needs to be tolerant of=20
> changes to
> IP addresses and ports in c- and m-lines in SDP? In other words, that
> intermediate domains can legitimately modify these fields (e.g., for
> media steering) and a solution is needed for end-to-end authenticated
> identification in such circumstances?
>=20
> B) Is there agreement that a solution needs to be tolerant of=20
> changes to
> Contact (addr-spec), Call-Id (call-id) and CSeq (digit) header fields?
> In other words, that intermediate domains can legitimately=20
> modify these
> fields (e.g., for topology hiding) and a solution is needed for
> end-to-end authenticated identification in such circumstances?
>=20
> C) Is there agreement that a solution does not need to be tolerant of
> changes to the addr-spec of From and To header fields? In other words,
> that intermediate domains should not modify these fields and=20
> can expect
> to break end-to-end authenticated identification if they do so?
>=20
>=20
> 2. Are people in broad agreement with the conclusions in section 9
> concerning the importance of end-to-end identification and in=20
> particular
> end-to-end authenticated identification?
>=20
> John
>=20
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

From dworley@nortel.com  Wed Mar  4 13:08:03 2009
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Subject: Re: [Sip] Question regarding conflicting grammar for IPV6 SIP URI andRFC 3986
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On Wed, 2009-03-04 at 12:34 +0000, Theo Zourzouvillys wrote:
> Both are going to require change (although we don't need world AND dog
> to agree with a 3261 update), but seems to me that updating 3986 seems
> like a far more appropriate path to take given the number of other
> affected URI schemes as Hadriel pointed out.

As far as I can see, all the schemes that Hadriel pointed out have
exactly the same problem as sip:, they permit "mailbox"-style components
but do not specify how they are to be encoded if the characters in
question are not valid by RFC 3986.

> I have no objection to the removal of '/' and '?' from the user part,
> but the replacing of '[' and ']' seems [quite] urgh.

It's "urgh" only if you've implemented IPv6, I suppose.  Conveniently,
our project hasn't.

Dale



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From: "Dale Worley" <dworley@nortel.com>
To: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: [Sip] Question regarding conflicting grammar for IPV6 SIP URI	andRFC 3986
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On Wed, 2009-03-04 at 12:03 -0600, Dean Willis wrote:
> I'm afraid I have to concur with Theo. 3986 is just broken.

Perhaps so, given that the different URI formats have different sets of
acceptable characters -- why would that be?

But what is the plan for going forward?

Dale



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Subject: Re: [Sip] draft-state-sip-relay-attack-00
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Thanks for publishing that attack.  It has similarities with the attack
described by Hadriel in
http://tools.ietf.org/html/draft-kaplan-sip-baiting-attack.

-d


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Raphael Coeffic
> Sent: Wednesday, March 04, 2009 2:35 AM
> To: sip@ietf.org
> Subject: [Sip] draft-state-sip-relay-attack-00
> 
> Hello,
> 
> a new internet draft has been published concerning the relay 
> attack on 
> digest authentication and SIP. The attack itself has been first 
> disclosed 2 years ago by the maydnes team from the french 
> INRIA. Until 
> now, no document has been pushlished that documents the attack and 
> provides guidance to SIP operators or handset manufacturers.
> 
> http://tools.ietf.org/html/draft-state-sip-relay-attack-00
> 
> The appropriate mitigations of problem resolutions are still not 100% 
> clear. We hope that this draft can help start a discussion on how to 
> best resolve this problem.
> 
> 
> Regards,
> 
> Raphael Coeffic.
> (on behalf of all the authors of this draft)
> 
> --------------------------------------------------------------
> -------------------------------------
> 
> Filename:	   draft-state-sip-relay-attack
> Version:	   00
> Staging URL:	   
> http://www3.ietf.org/proceedings/staging/draft-state-sip-relay
> -attack-00.txt
> Title:		   SIP digest authentication relay attack
> Creation_date:	   2009-03-02
> WG ID:		   Indvidual Submission
> Number_of_pages: 18
> Abstract:
> The Session Initiation Protocol (SIP [RFC3261]) provides a mechanism
> for creating, modifying, and terminating sessions with one or more
> participants.  This document describes a vulnerability of SIP
> combined with HTTP Digest Access Authentication [RFC2617] through
> which an attacker can leverage the victim's credentials to send
> authenticated requests on his behalf.  This attack is different from
> the man-in-the-middle (MITM) attack and does not require any
> eavesdropping, DNS or IP spoofing.
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


From HKaplan@acmepacket.com  Wed Mar  4 14:54:25 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dale Worley <dworley@nortel.com>, Dean Willis <dean.willis@softarmor.com>
Date: Wed, 4 Mar 2009 17:54:51 -0500
Thread-Topic: [Sip] Question regarding conflicting grammar for IPV6 SIP	URI andRFC 3986
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How about: ignorance is bliss?  :)

Seriously, what does RFC 3986 do??  How does it affect anything for common =
URI formats not to comply with it? (or really, for *it* not to comply with =
common URI formats)


-hadriel

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Dal=
e
> Worley
> Sent: Wednesday, March 04, 2009 4:10 PM
> To: Dean Willis
> Cc: SIP
> Subject: Re: [Sip] Question regarding conflicting grammar for IPV6 SIP UR=
I
> andRFC 3986
>
> On Wed, 2009-03-04 at 12:03 -0600, Dean Willis wrote:
> > I'm afraid I have to concur with Theo. 3986 is just broken.
>
> Perhaps so, given that the different URI formats have different sets of
> acceptable characters -- why would that be?
>
> But what is the plan for going forward?
>
> Dale
>
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

From theo@crazygreek.co.uk  Wed Mar  4 15:22:34 2009
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On Wed, Mar 4, 2009 at 10:34 AM, Raphael Coeffic <rco@iptel.org> wrote:

> The appropriate mitigations of problem resolutions are still not 100% clear.
> We hope that this draft can help start a discussion on how to best resolve
> this problem.

Hi Raphael,

Thanks for documenting this!

page 10:

   It is worth noting that the protection
   provided on the request URI is purely theoretical, as [RFC3261]
   introduces an exception to the request URI checking required by
   [RFC2617] in section 22.4:

Another important consideration to keep in mind while thinking about
solutions is that the Contact header in the dialog creating request
can be pretty much anything bob likes if he also adds a Record-Route
header in, leading to the dialog target URI at alice's UA being
whatever bob wants, and thus the Authentication header can be
manipulated to contain whatever the attacker wants in the uri
parameter.  To explain - consider Figure 1 in the draft

F1 could be sent by bob@rouge.com as:

  INVITE sip:alice@proxy.com SIP/2.0
  Contact: <sip:+1-900-xxx@proxy.com>
  Record-Route: <sip:bob@rouge.com;lr>

thus, when alice creates the dialog, the remote URI will indeed be
+1-900-xxx@proxy.com, and your in-dialog F10 message will be:

   F10 INVITE Alice -> Bob

      INVITE sip:+1-900-xxx@proxy.com SIP/2.0
      Route: <sip:bob@rouge.com;lr>
      Proxy-Authorization: Digest username="alice", realm="proxy.com",
       uri="sip:+1-900-xxx@proxy.com", nonce="f84f1cec41e6cbe5aea9c8e88d359543",
       response="3bea678acef9875433487f23a567d876",
       opaque="", algorithm=MD5
      Content-Type: application/sdp
      Content-Length:...

presto - the authentication header now even contains the URI you want
to call.  note that a re-INVITE could also be done to change the
target URI a few times to get different Authentication headers with
different URIs in, all legitimate.

As to the potential solutions: both only accepting from registered
contact or any attempt at avoiding sending re-INVITE [1] are in my
opinion unfeasible and broken - i'll have some thought on others,
though :-)

 ~ Theo

1 - REFER is another in-dialog method that could be abused - an OOD
REFER may be accepted with authentication by - for example - a PSTN
gateway.  Social engineering of alice could result in alice's UA
sending a REFER that could then be used.

From hsinnrei@adobe.com  Wed Mar  4 15:27:53 2009
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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Dean Willis <dean.willis@softarmor.com>, SIP IETF <sip@ietf.org>
Date: Wed, 4 Mar 2009 15:28:06 -0800
Thread-Topic: [Sip] Agenda call for SIP
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--_000_C5D46AA6BDB1hsinnreiadobecom_
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I suppose this is the 1st wiki in RAI?
If so, it is a good start.
Congratulations!

Henry


On 3/4/09 12:51 PM, "Dean Willis" <dean.willis@softarmor.com> wrote:




Note that I'm accumulating agenda requests at our (yes, Henry, we one)
wiki:

http://www.softarmor.com/mediawiki/index.php/SIP_Agenda_Requests_IETF_74

If your request doesn't show up there fairly quickly (48 hours), I've
probably lost it.

--
Dean



_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


--_000_C5D46AA6BDB1hsinnreiadobecom_
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<HTML>
<HEAD>
<TITLE>Re: [Sip] Agenda call for SIP</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
14pt'>I suppose this is the 1st wiki in RAI?<BR>
If so, it is a good start.<BR>
Congratulations!<BR>
<BR>
Henry<BR>
<BR>
<BR>
On 3/4/09 12:51 PM, &quot;Dean Willis&quot; &lt;<a href=3D"dean.willis@soft=
armor.com">dean.willis@softarmor.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:14pt'><BR>
<BR>
<BR>
Note that I'm accumulating agenda requests at our (yes, Henry, we one) <BR>
wiki:<BR>
<BR>
<a href=3D"http://www.softarmor.com/mediawiki/index.php/SIP_Agenda_Requests=
_IETF_74">http://www.softarmor.com/mediawiki/index.php/SIP_Agenda_Requests_=
IETF_74</a><BR>
<BR>
If your request doesn't show up there fairly quickly (48 hours), I've <BR>
probably lost it.<BR>
<BR>
--<BR>
Dean<BR>
<BR>
<BR>
<BR>
_______________________________________________<BR>
Sip mailing list &nbsp;<a href=3D"https://www.ietf.org/mailman/listinfo/sip=
">https://www.ietf.org/mailman/listinfo/sip</a><BR>
This list is for NEW development of the core SIP Protocol<BR>
Use <a href=3D"sip-implementors@cs.columbia.edu">sip-implementors@cs.columb=
ia.edu</a> for questions on current sip<BR>
Use <a href=3D"sipping@ietf.org">sipping@ietf.org</a> for new developments =
on the application of sip<BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
</BODY>
</HTML>


--_000_C5D46AA6BDB1hsinnreiadobecom_--

From HKaplan@acmepacket.com  Wed Mar  4 16:28:16 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: SIP <sip@ietf.org>
Date: Wed, 4 Mar 2009 19:28:37 -0500
Thread-Topic: I-D Inaction: draft-kaplan-sip-secure-call-id-00.txt
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Howdy,
In an effort to avoid endless thrashing, I will be updating the Session-ID =
draft to only be about troubleshooting use.
For dialog matching issues, I have followed Adam's advice and submitted a s=
eparate draft to update RFC 3261, linked below.
This issue will only get worse, and the code changes to do this should be f=
airly small for most folks, I would think.
And it's only 6 pages, so we should be able to agree on this in about 3 yea=
rs.  :)

A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.

        Title           : A Secure Call-ID for the Session Initiation Proto=
col (SIP)
        Author(s)       : H. Kaplan
        Filename        : draft-kaplan-sip-secure-call-id-00.txt
        Pages           : 6
        Date            : 2009-03-04

Many SIP devices generate Call-ID values which contain their system IP Addr=
ess, due to examples and normative text in RFC 3261.  This has led to some =
middleboxes, such as SBC's, to change the Call-ID for security reasons.  Th=
is draft updates RFC 3261 to require SIP User Agents to generate benign Cal=
l-IDs, in such a manner that they can be detected as secure and not need to=
 be changed.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-kaplan-sip-secure-call-id-00.txt


-hadriel

From dean.willis@softarmor.com  Wed Mar  4 18:49:30 2009
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On Mar 4, 2009, at 5:28 PM, Henry Sinnreich wrote:

> I suppose this is the 1st wiki in RAI?
> If so, it is a good start.
> Congratulations!

Actually, we have been using it since IETF 69.  And the IETF tools  
team has also made a WIKI available to each working group, but so far  
we haven't put it into use.

--
Dean


From john.elwell@siemens.com  Wed Mar  4 23:31:38 2009
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Date: Thu, 05 Mar 2009 07:29:09 +0000
From: "Elwell, John" <john.elwell@siemens.com>
To: SIP List <sip@ietf.org>, Mayumi Munakata <munakata.mayumi@lab.ntt.co.jp>
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I previously commented on the last sentence of the following text in
5.1.1 (Contact header field) of draft-ietf-sip-ua-privacy-05:
"   When using this header field in a dialog-forming request or response
   or in a mid-dialog request or response, the user agent MUST anonymize
   the Contact header field using an anonymous URI ("temp-gruu")
   obtained through the GRUU mechanism, unless an equivalent functional
   anonymous URI is provided by some other means.  For out-of-dialog
   request, anonymous URI MAY be set when anonymization is required."

The reason for my comment was that I wasn't certain where the last
sentence would apply.

Of course, a REGISTER request or 200 response is "out of dialog", but
clearly an anonymized Contact URI is inappropriate.

In addition I have identified the following requests:
- OPTIONS - Contact MAY be included.
- PUBLISH - Contact, if present, doesn't have any meaning in the context
of event publication.
- MESSAGE - Contact not allowed.

Furthermore, non-dialog-forming responses to things like INVITE and
SUBSCRIBE do not require Contact, with the exception of 3xx where the
URI clearly cannot be anonymized.

So how to cover this concisely? I would propose replacing the last
sentence above with the following:

"For other requests and responses, with the exception of 3xx responses,
REGISTER requests and REGISTER 200 responses, the UA MUST either omit
the Contact header field or use an anonymous URI."

Would this be acceptable? Have I captured all exceptions?

John

From munakata.mayumi@lab.ntt.co.jp  Thu Mar  5 00:15:08 2009
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Subject: Re: [Sip] New text for ua-privacy
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John,

See inline.

 > I previously commented on the last sentence of the following text in
 > 5.1.1 (Contact header field) of draft-ietf-sip-ua-privacy-05:
 > "   When using this header field in a dialog-forming request or response
 >    or in a mid-dialog request or response, the user agent MUST anonymize
 >    the Contact header field using an anonymous URI ("temp-gruu")
 >    obtained through the GRUU mechanism, unless an equivalent functional
 >    anonymous URI is provided by some other means.  For out-of-dialog
 >    request, anonymous URI MAY be set when anonymization is required."
 >
 > The reason for my comment was that I wasn't certain where the last
 > sentence would apply.

The original intent of the sentence was to anonymize
the Contact except for the REGISTER requests and 3xx
responses, as you pointed out in November, but I agree
that it is hard to tell from the current sentence.

 >
 > Of course, a REGISTER request or 200 response is "out of dialog", but
 > clearly an anonymized Contact URI is inappropriate.
 >
 > In addition I have identified the following requests:
 > - OPTIONS - Contact MAY be included.
 > - PUBLISH - Contact, if present, doesn't have any meaning in the context
 > of event publication.
 > - MESSAGE - Contact not allowed.

The Contact header is not allowed in PUBLISH request,
I believe.  Therefore, the Contact in OPTIONS is the
only case we should care as out-of-dialog request.

 >
 > Furthermore, non-dialog-forming responses to things like INVITE and
 > SUBSCRIBE do not require Contact, with the exception of 3xx where the
 > URI clearly cannot be anonymized.
 >
 > So how to cover this concisely? I would propose replacing the last
 > sentence above with the following:
 >
 > "For other requests and responses, with the exception of 3xx responses,
 > REGISTER requests and REGISTER 200 responses, the UA MUST either omit
 > the Contact header field or use an anonymous URI."
 >
 > Would this be acceptable? Have I captured all exceptions?

This text is much clearer than the original text.
I think this captures all the exceptions.
If there is no more exceptions, I will replace
the last sentence with your text.

Thank you!

Mayumi


 >
 > John
 > _______________________________________________
 > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
 > This list is for NEW development of the core SIP Protocol
 > Use sip-implementors@cs.columbia.edu for questions on current sip
 > Use sipping@ietf.org for new developments on the application of sip
 >

From christer.holmberg@ericsson.com  Thu Mar  5 01:10:29 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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Hi Bob,
=20
>I have few comments on draft-ietf-sip-199-06.txt. I apologize=20
>if any of these are duplicates.
>=20
>Section 6. The statement requiring the proxy to keep track of=20
>early dialogs:
>=20
>    A proxy which supports generating of 199 response codes MUST keep
>    track of early dialogs, in order to determine whether to generate a
>    199 response when the proxy receives a non-2xx final response.  In
>    addition, the proxy MUST keep track on which early dialogs it has
>    received and forwarded 199 responses, in order to not generate
>    additional 199 responses for those early dialogs.
>=20
>Should precede the statement:
>=20
>    If the forking proxy generates a 199 response, and if the
>    forking proxy is able to identify additional early dialogs=20
>    which are terminated by the same non-2xx final response, it MUST=20
>    also generate and send a 199 response upstream for each of those
early dialogs,
>=20
>In fact, if the proxy is keeping track of early dialogs,=20
>under what circumstances would it not be "able to identify=20
>additional early dialogs which are terminated by the same=20
>non-2xx final response"?

Just because the proxy keeps track of early dialogs, it doesn't
necessarily know that an error response on early dialog X also
terminates early dialog Y. In order to know that, the forking proxy
needs to know that early dialogs X and Y come from the same forking
proxy further down the path.


>Section 8 (199 Early Dialog Terminated) seems out of place.=20
>Shouldn't this be in the introduction? or in an "Overview Of=20
>Operation" between sections 3 and 4?

I have tried to use other similar drafts as "template".


>The text in Section 9 (Usage with SDP offer/answer) should be=20
>in section 5 (User Agent Server behavior).

I think it is good to have a separate chapter. Of course, I could refer
to chapter 9 from chapter 5.

>The statement in section 10 recommending that the UAS send=20
>199 unreliably should be in section 5.
>=20
>The statement in section 10 forbidding proxies from sending=20
>199 reliably should be in section 6 (Proxy behavior).

Same as for offer/answer.

Since the 199 procedures related to offer/answer and reliable responses
are special, I think it is good to have specific chapters for it.

Regards,

Christer


From rco@iptel.org  Thu Mar  5 01:47:25 2009
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Dan Wing wrote:
> Thanks for publishing that attack.  It has similarities with the attack
> described by Hadriel in
> http://tools.ietf.org/html/draft-kaplan-sip-baiting-attack.
>
>   
Yes, it also modifies part of the request which are not signed by the 
authentication/signature algorithm. I should mention this draft in a 
later version.

However, the goal is not the same. Hadriel's draft is focused on 
impersonation, whereby the other draft is focused on breaking the 
authentication done at the proxy. The second difference is that it works 
right now with any publicly reachable SIP provider.

Raphael.


From michael@voip.co.uk  Thu Mar  5 01:55:43 2009
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Cc: sip@ietf.org, Dan Wing <dwing@cisco.com>
Subject: Re: [Sip] draft-state-sip-relay-attack-00
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Raphael Coeffic wrote:
> Dan Wing wrote:
>> Thanks for publishing that attack.  It has similarities with the attack
>> described by Hadriel in
>> http://tools.ietf.org/html/draft-kaplan-sip-baiting-attack.
>>
>>   
> Yes, it also modifies part of the request which are not signed by the
> authentication/signature algorithm. I should mention this draft in a
> later version.
>
> However, the goal is not the same. Hadriel's draft is focused on
> impersonation, whereby the other draft is focused on breaking the
> authentication done at the proxy. The second difference is that it
> works right now with any publicly reachable SIP provider.
>

I'm intrigued by the variation in fig2.  How often are you finding two
proxies in different administrative domains that use the same
credentials?  Or is the attack more focussed towards Alice using
multiple sets of credentials?

I think that the variation in fig3 can be addressed to some degree by
using GRUU, but I don't think that completely solves the problem.

Regards,

Michael

From rco@iptel.org  Thu Mar  5 01:59:17 2009
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Michael Procter wrote:
> Raphael Coeffic wrote:
>   
>> Dan Wing wrote:
>>     
>>> Thanks for publishing that attack.  It has similarities with the attack
>>> described by Hadriel in
>>> http://tools.ietf.org/html/draft-kaplan-sip-baiting-attack.
>>>
>>>   
>>>       
>> Yes, it also modifies part of the request which are not signed by the
>> authentication/signature algorithm. I should mention this draft in a
>> later version.
>>
>> However, the goal is not the same. Hadriel's draft is focused on
>> impersonation, whereby the other draft is focused on breaking the
>> authentication done at the proxy. The second difference is that it
>> works right now with any publicly reachable SIP provider.
>>
>>     
>
> I'm intrigued by the variation in fig2.  How often are you finding two
> proxies in different administrative domains that use the same
> credentials?  Or is the attack more focussed towards Alice using
> multiple sets of credentials?
>
>   
They are not using the same credentials, which is the reason why 
'proxy.com' won't remove Alice's credentials from the request, and thus 
enable the attack.


> I think that the variation in fig3 can be addressed to some degree by
> using GRUU, but I don't think that completely solves the problem.
>
>   
I am not quite sure that I understand how this helps: could you develop 
your thoughts a little bit more?


Regards,
Raphael.


From rco@iptel.org  Thu Mar  5 02:03:32 2009
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Cc: SIP IETF <sip@ietf.org>, Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] Agenda call for SIP
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Hello,

Radu and I would like to present the relay attack draft recently 
published (http://tools.ietf.org/html/draft-state-sip-relay-attack-00).
 From the small feedback we became until now, it seems that there is 
some interrest in finding solutions, whereby the problem is clear.

I think we need only 10 to 15 minutes. The question we'd like to ask is 
whether or not this is something we want to work on. We also had a 
discussion (between the authors, not on the list) whether the document 
should address both the issue and a possible solution, or whether this 
should be split into 2 separate documents. I tend to think that one 
document is enough.

Regards
Raphael Coeffic.


DRAGE, Keith (Keith) wrote:
> (As WG co chair)
>
> If you wish to have time allocated to a particular discussion in the SIP sessions in San Francisco, and have not already submitted a request, then please do so.
>
> Please include information like:
>
> -	name(s) of any draft(s) to be discussed.
> -	proposed name of presenter
> -	what is/are the key question(s) to put to the WG as a result of the discussion.
> -	how much time you think you will need.
>
> Note that in addressing the third bullet we would like a bit more detail than "WG item?", for example give us some help in identifying the text for the milestone you actually want in the SIP charter, which is the problem you want to solve, rather than the solution.
>
> Please do this now as we will be putting the preliminary agenda together tomorrow (Thursday). Agendas are always subject to change but would prefer not to have any late surprises.
>
> regards
>
> Keith
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>   


From jan@ryngle.com  Thu Mar  5 02:59:32 2009
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From: Jan Janak <jan@ryngle.com>
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Cc: sip@ietf.org, Dan Wing <dwing@cisco.com>
Subject: Re: [Sip] draft-state-sip-relay-attack-00
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On 05-03 09:56, Michael Procter wrote:
> Raphael Coeffic wrote:
> > Dan Wing wrote:
> >> Thanks for publishing that attack.  It has similarities with the attack
> >> described by Hadriel in
> >> http://tools.ietf.org/html/draft-kaplan-sip-baiting-attack.
> >>
> >>   
> > Yes, it also modifies part of the request which are not signed by the
> > authentication/signature algorithm. I should mention this draft in a
> > later version.
> >
> > However, the goal is not the same. Hadriel's draft is focused on
> > impersonation, whereby the other draft is focused on breaking the
> > authentication done at the proxy. The second difference is that it
> > works right now with any publicly reachable SIP provider.
> >
> 
> I'm intrigued by the variation in fig2.  How often are you finding two
> proxies in different administrative domains that use the same
> credentials?  Or is the attack more focussed towards Alice using
> multiple sets of credentials?

Yes. The scenario in fig2 shows that such attack is possible even if Alice can
receive SIP requests from its outbound proxy only (or one of its outbound
proxies), either because her phone is configured so or because it is behind
NAT.

Even if the outbound proxy removes digest credentials that belong to its
administrative domain before forwarding SIP requests downstream, you can still
obtain digest credentials for another administrative domain through this
outbound proxy from Alice if her phone is configured with multiple sets of
credentials.

If you have a phone which is configured to use two different ITSPs and has two
sets of digest credentials then it is most likely vulnerable to this kind of
attack.

  Jan.

From michael@voip.co.uk  Thu Mar  5 03:04:27 2009
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Cc: sip@ietf.org, Dan Wing <dwing@cisco.com>
Subject: Re: [Sip] draft-state-sip-relay-attack-00
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Raphael Coeffic wrote:
> Michael Procter wrote: 
>> I'm intrigued by the variation in fig2.  How often are you finding two
>> proxies in different administrative domains that use the same
>> credentials?  Or is the attack more focussed towards Alice using
>> multiple sets of credentials?
>>
>>   
> They are not using the same credentials, which is the reason why
> 'proxy.com' won't remove Alice's credentials from the request, and
> thus enable the attack.

I see - it wasn't entirely clear from your text which case you were
addressing.  So that scenario is essentially the normal MITM digest
attack.  Whenever you receive a challenge, you can't really authenticate
the server that is challenging you to determine whether you should issue
a valid response or not.

>> I think that the variation in fig3 can be addressed to some degree by
>> using GRUU, but I don't think that completely solves the problem.
>>   
> I am not quite sure that I understand how this helps: could you
> develop your thoughts a little bit more?

Sorry - I mistyped!  What I meant to say was that 'outbound' addresses
the problem to some degree.  To expand on that a little, the key in fig3
is for Bob to be able to send a response directly to Alice, without
going through the proxy.  In the hosted scenario (which I considered
first), Alice is typically behind a NAT/FW and uses outbound to maintain
flows to proxy.com.  As a consequence of this, Alice should only act
upon responses received over these flows.  This would defeat the attack
in fig3, since Bob's altered response would probably not even reach
Alice, let alone be acted upon.

In the non-hosted scenario, where Alice and proxy.com are on the same
network (i.e. without NAT/FW issues) such as might be the case in a
typical enterprise deployment, then outbound wouldn't be used.  But in
this case, I would expect the enterprise firewall to prevent
communications from Bob to Alice, except when mediated by an appropriate
device (such as proxy.com).  Again, the scenario in fig3 would be defeated.

That isn't to say that the attack in fig3 isn't a real attack, only that
I wouldn't expect most deployed networks to suffer from it.

Regards,

Michael

From rco@iptel.org  Thu Mar  5 03:12:16 2009
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Cc: sip@ietf.org, Dan Wing <dwing@cisco.com>
Subject: Re: [Sip] draft-state-sip-relay-attack-00
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Hello Michael,

Michael Procter wrote:
> Raphael Coeffic wrote:
>   
>> Michael Procter wrote: 
>>     
>>> I'm intrigued by the variation in fig2.  How often are you finding two
>>> proxies in different administrative domains that use the same
>>> credentials?  Or is the attack more focussed towards Alice using
>>> multiple sets of credentials?
>>>
>>>   
>>>       
>> They are not using the same credentials, which is the reason why
>> 'proxy.com' won't remove Alice's credentials from the request, and
>> thus enable the attack.
>>     
>
> I see - it wasn't entirely clear from your text which case you were
> addressing.  So that scenario is essentially the normal MITM digest
> attack.  Whenever you receive a challenge, you can't really authenticate
> the server that is challenging you to determine whether you should issue
> a valid response or not.
>
>   

That's true. The novelty is that such a MITM attack can be induced by 
SIP means, without to much interventions into the network topology (no 
DNS or ARP spoofing for example).

>>> I think that the variation in fig3 can be addressed to some degree by
>>> using GRUU, but I don't think that completely solves the problem.
>>>   
>>>       
>> I am not quite sure that I understand how this helps: could you
>> develop your thoughts a little bit more?
>>     
>
> Sorry - I mistyped!  What I meant to say was that 'outbound' addresses
> the problem to some degree.  To expand on that a little, the key in fig3
> is for Bob to be able to send a response directly to Alice, without
> going through the proxy.  In the hosted scenario (which I considered
> first), Alice is typically behind a NAT/FW and uses outbound to maintain
> flows to proxy.com.  As a consequence of this, Alice should only act
> upon responses received over these flows.  This would defeat the attack
> in fig3, since Bob's altered response would probably not even reach
> Alice, let alone be acted upon.
>
> In the non-hosted scenario, where Alice and proxy.com are on the same
> network (i.e. without NAT/FW issues) such as might be the case in a
> typical enterprise deployment, then outbound wouldn't be used.  But in
> this case, I would expect the enterprise firewall to prevent
> communications from Bob to Alice, except when mediated by an appropriate
> device (such as proxy.com).  Again, the scenario in fig3 would be defeated.
>
>   
Imho, not necessarily. It is very easy to forge an IP (that of the 
proxy) if you only need one-way communication, which is the case for the 
200 reply. This means that it should be quite easy to get this reply 
directly to Alice. The question is what Alice and the outbound proxy 
will do after that.

> That isn't to say that the attack in fig3 isn't a real attack, only that
> I wouldn't expect most deployed networks to suffer from it.
>
>   
You may be right.

Best regards,
Raphael.

From lists@ohlmeier.org  Thu Mar  5 05:40:29 2009
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Subject: Re: [Sip] draft-state-sip-relay-attack-00
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Hello,

first of all I agree that the draft describes a potential problem.

One thing which is not that obvious but is implictly a requirement for the
attack: the proxies has to challenge in-dialog requests. I do not see a
big benefit in challeging in-dialog requests as these are hopefully
rejected by the remote side if no matching dialog exists. If the UA would
know that his proxy does not challenge in-dialog requests it could simply
ignore the challenge :-)

Some more comments:

Figure 1:
- As the draft already states this direct attack becomes inpossible if
Alice UA accepts incoming requests only from one of its proxys (we have to
keep in mind that we rely here on pure IP layer security which is weak as
we all know).
- Additionally Alice UA should notice that the call got established
without a route set. And without any Route set I would not see why the UA
should get a challenge at all. But I admit that such a check most likely
does not exist in UA today.

Figure 2:
- If p2.com uses a different realm then proxy.com you should mention in
your draft that Bob needs to know that Alice has accounts configured for
both servers in her UA.
- I never unterstood why a proxy should pass through the authentication
request from a foreign domain. In which scenario do I have credentials for
two different domains, but the traffic including authentication is passed
through only one of them?
Sure I could have a call forwarding from p2.com to proxy.com. Again only
if p2.com challenges in-dialog requests the problem arrises with this call
forwarding.
- Another possible improvement for an UA would be to strictly bind the
credentials to one proxy. Which means it would only answer challenges for
proxy.com if the request came IP wise from proxy.com and it would not
answer challenges for other proxies if they arrive IP wise from proxy.com.
This improvement does not work if you use daisy chained proxy with
authentication for outbound calls (but I have never seen or heard of such
a setup), or if the proxy with active call forwarding challenges in-dialog
requests.
- If we admit that this authentication pass through is of no real use,
either Alice UA could ignore it, or proxy.com could simply strip the
authentication from the foreign domain p2.com.
- If we still need authentication pass through proxy.com could at least
check that it does not pass through challenges with the same realm as its
own one (which brakes daisy chains of challenging proxy with identical
realms - which is hopefully not too comon).

I admit that my comments are no real solution for the problem itself and
some of them would first need to be implemented. But hopefully they help
to improve the situation.

Regards
  Nils Ohlmeier
  VoIP Freelancer

> Hello,
>
> a new internet draft has been published concerning the relay attack on
> digest authentication and SIP. The attack itself has been first
> disclosed 2 years ago by the maydnes team from the french INRIA. Until
> now, no document has been pushlished that documents the attack and
> provides guidance to SIP operators or handset manufacturers.
>
> http://tools.ietf.org/html/draft-state-sip-relay-attack-00
>
> The appropriate mitigations of problem resolutions are still not 100%
> clear. We hope that this draft can help start a discussion on how to
> best resolve this problem.
>
>
> Regards,
>
> Raphael Coeffic.
> (on behalf of all the authors of this draft)
>
> ---------------------------------------------------------------------------------------------------
>
> Filename:	   draft-state-sip-relay-attack
> Version:	   00
> Staging URL:
> http://www3.ietf.org/proceedings/staging/draft-state-sip-relay-attack-00.txt
> Title:		   SIP digest authentication relay attack
> Creation_date:	   2009-03-02
> WG ID:		   Indvidual Submission
> Number_of_pages: 18
> Abstract:
> The Session Initiation Protocol (SIP [RFC3261]) provides a mechanism
> for creating, modifying, and terminating sessions with one or more
> participants.  This document describes a vulnerability of SIP
> combined with HTTP Digest Access Authentication [RFC2617] through
> which an attacker can leverage the victim's credentials to send
> authenticated requests on his behalf.  This attack is different from
> the man-in-the-middle (MITM) attack and does not require any
> eavesdropping, DNS or IP spoofing.
>
>
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
>



From theo@crazygreek.co.uk  Thu Mar  5 06:30:39 2009
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On Wed, Mar 4, 2009 at 10:34 AM, Raphael Coeffic <rco@iptel.org> wrote:
> The appropriate mitigations of problem resolutions are still not 100% clear.
> We hope that this draft can help start a discussion on how to best resolve
> this problem.

After more thinking about this, I don't think this document presents
anything other than there are poorly implemented SIP endpoints and
proxies out there - as there is nothing wrong with SIP itself [1] in
these attacks:  It all boils down to UA vendors currently giving
credentials to anyone who asks for them, even if they shouldn't be.

All of the scenarios you mention in this document apply directly to
proxy authentication in the case where a UA would use outbound
authentication with it's proxy.  A UA only ever sending proxy
credentials to it's proxy seems like a dead simple and sensible thing
to do (and although i'm not really surprised UAs don't do this, i'll
pretend to be), and makes all the attacks you list just go away.

The only issue i see is in fig 3, how to identify a next hop is
actually proxy.com and not bob, as a proxy will commonly RR with a URI
such as node-01.proxy.com instead of proxy.com, or even use an IP
address in it.  The first, and most and correct way is use TLS or DTLS
with mutual auth, and you can verify that the next hop is really your
proxy and thus it's ok to send credentials to it.  The second way is
to always send requests to proxy.com (i.e, using outboind), even in
dialog ones (and indeed many UAs have a "send in-dialog requests via
outbound proxy" option,which makes this attack go away) - although
this might breaks some broken SBCs out there that SPs deploy today.
The third option (and most ickly and hacky) it to build and keep a
whitelist of IP addresses that proxy.com is, and ensure the SP only
ever record-routes using an IP address that a forward lookup on
proxy.com would resolve to.  crappy but in reality mostly works today,
as long as devices are sensible and do things like refresh after TTLs
expire on RRsets.

So, what does need to be done is beat UA vendors and SIP operators
with a great big security clue stick (how that's done i'd love to
know.  My stick is almost broken, and even when applied directly,
mostly ineffective.) - perhaps a BCP to highlight the issues?

However, you don't cover the more interesting cases of multi-hop proxy
authentication or end-to-end WWW authentication: these are the harder
ones to deal with, and may result in some "real" issues in SIP itself
rather than shoddy implementations and insecure proxies.

 ~ Theo

1 - heh. sarcasm.

From rco@iptel.org  Thu Mar  5 06:41:25 2009
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Hi Theo,

I really think that mutual authentication is the way to go. 
Unfortunatly, there is nothing deployed in this direction right now. 
Most providers (at least in germany) try to tackle those issues by 
separating networks. The SIP phones are then jailed into things like 
dedicated virtual circuits (on the DSL level). On the provider-side, 
they use the full panel of firewalls to get the full control of 
everything. This is definitely not the way to go.

Having the BCP gathered into this document (or another one, I don't 
care) would be a good thing to do. If you let people think by 
themselves, you might end-up with the situation we have right now.

Theo Zourzouvillys wrote:
> On Wed, Mar 4, 2009 at 10:34 AM, Raphael Coeffic <rco@iptel.org> wrote:
>   
>> The appropriate mitigations of problem resolutions are still not 100% clear.
>> We hope that this draft can help start a discussion on how to best resolve
>> this problem.
>>     
>
> After more thinking about this, I don't think this document presents
> anything other than there are poorly implemented SIP endpoints and
> proxies out there - as there is nothing wrong with SIP itself [1] in
> these attacks:  It all boils down to UA vendors currently giving
> credentials to anyone who asks for them, even if they shouldn't be.
>
>   
> All of the scenarios you mention in this document apply directly to
> proxy authentication in the case where a UA would use outbound
> authentication with it's proxy.  A UA only ever sending proxy
> credentials to it's proxy seems like a dead simple and sensible thing
> to do (and although i'm not really surprised UAs don't do this, i'll
> pretend to be), and makes all the attacks you list just go away.
>
> The only issue i see is in fig 3, how to identify a next hop is
> actually proxy.com and not bob, as a proxy will commonly RR with a URI
> such as node-01.proxy.com instead of proxy.com, or even use an IP
> address in it.  The first, and most and correct way is use TLS or DTLS
> with mutual auth, and you can verify that the next hop is really your
> proxy and thus it's ok to send credentials to it.  The second way is
> to always send requests to proxy.com (i.e, using outboind), even in
> dialog ones (and indeed many UAs have a "send in-dialog requests via
> outbound proxy" option,which makes this attack go away) - although
> this might breaks some broken SBCs out there that SPs deploy today.
> The third option (and most ickly and hacky) it to build and keep a
> whitelist of IP addresses that proxy.com is, and ensure the SP only
> ever record-routes using an IP address that a forward lookup on
> proxy.com would resolve to.  crappy but in reality mostly works today,
> as long as devices are sensible and do things like refresh after TTLs
> expire on RRsets.
>
> So, what does need to be done is beat UA vendors and SIP operators
> with a great big security clue stick (how that's done i'd love to
> know.  My stick is almost broken, and even when applied directly,
> mostly ineffective.) - perhaps a BCP to highlight the issues?
>
> However, you don't cover the more interesting cases of multi-hop proxy
> authentication or end-to-end WWW authentication: these are the harder
> ones to deal with, and may result in some "real" issues in SIP itself
> rather than shoddy implementations and insecure proxies.
>
>   

If you provide me with the input, I'd be very happy to document those 
issues.

Regards
Raphael.


From dworley@nortel.com  Thu Mar  5 07:25:16 2009
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On Thu, 2009-03-05 at 08:16 +0000, Elwell, John wrote:
> What about non-INVITEs? Provisional responses are designed for INVITEs
> and I am not certain about their applicability to non-INVITEs.

Yes, provisional responses to non-INVITEs may be slightly out of
specification, but I don't expect any SIP element to malfunction if it
receives a provisional response to a non-INVITE request.  OTOH, I would
expect most uses of 'trace' to be attached to INVITEs, although that
makes me think that tracing an OPTIONS might be a useful test tool.

(Which leads to an unexpected observation:  RFC 3261 presupposes that
all non-INVITE responses are processed promptly, but in the face of
transport errors, this is not always so.  In sipX, we've adjusted
transaction handling so that if a re-send of a non-INVITE message is
seen the recipient sends a 100 to quench further re-sends.  It seems to
work exactly as expected, and noticeably reduces re-sends in situations
with transport errors.)

> In the case of INVITE, what about Contact? Is it permissible to send a
> 1xx (other than 100) without a Contact URI? RFC 3261 seems to indicate
> it is optional in a 1xx, but that might be solely to cater for omission
> in 100.

The Contact value of a 170 should never be used, so it doesn't really
matter if one is provided, or if it is usable.  OTOH, every SIP element
should be able to create a usable Contact URI.

Dale



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To: Christer Holmberg <christer.holmberg@ericsson.com>
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inline

>-----Original Message-----
>From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>Sent: Thursday, March 05, 2009 4:11 AM
>To: Bob Penfield
>Cc: sip@ietf.org
>Subject: RE: [Sip] I-D Action:draft-ietf-sip-199-06.txt
>
>
>Hi Bob,
>
>>I have few comments on draft-ietf-sip-199-06.txt. I apologize
>>if any of these are duplicates.
>>
>>Section 6. The statement requiring the proxy to keep track of
>>early dialogs:
>>
>>    A proxy which supports generating of 199 response codes MUST keep
>>    track of early dialogs, in order to determine whether to generate a
>>    199 response when the proxy receives a non-2xx final response.  In
>>    addition, the proxy MUST keep track on which early dialogs it has
>>    received and forwarded 199 responses, in order to not generate
>>    additional 199 responses for those early dialogs.
>>
>>Should precede the statement:
>>
>>    If the forking proxy generates a 199 response, and if the
>>    forking proxy is able to identify additional early dialogs
>>    which are terminated by the same non-2xx final response, it MUST
>>    also generate and send a 199 response upstream for each of those
>>    early dialogs,
>>
>>In fact, if the proxy is keeping track of early dialogs,
>>under what circumstances would it not be "able to identify
>>additional early dialogs which are terminated by the same
>>non-2xx final response"?
>
>Just because the proxy keeps track of early dialogs, it doesn't
>necessarily know that an error response on early dialog X also
>terminates early dialog Y. In order to know that, the forking proxy
>needs to know that early dialogs X and Y come from the same forking
>proxy further down the path.
>

Of course it can. If the proxy forwarded the request to Z, and Z forked
to X and Y, the provisional responses from X and Y that created the
early dialogs would have the same branch-id. The error response from Z
would have that same branch-id. The proxy would know that all early
dialogs created by a 1xx with that branch-id would have terminated
(i.e. both X and Y).

>
>>Section 8 (199 Early Dialog Terminated) seems out of place.
>>Shouldn't this be in the introduction? or in an "Overview Of
>>Operation" between sections 3 and 4?
>
>I have tried to use other similar drafts as "template".
>
>
>>The text in Section 9 (Usage with SDP offer/answer) should be
>>in section 5 (User Agent Server behavior).
>
>I think it is good to have a separate chapter. Of course, I could refer
>to chapter 9 from chapter 5.

My point is that all too often behavior requirements (MUST, SHOULD,
MAY, etc.) get buried is separate sections and folks can miss
them. If all the requirements are in the applicable "behavior"
section, they are easy to find and developers are less likely to miss
them. If you want it to be a separate section, make it 5.1. All UAS
requirements should be in section 5 or a subsection thereof.

>
>>The statement in section 10 recommending that the UAS send
>>199 unreliably should be in section 5.
>>
>>The statement in section 10 forbidding proxies from sending
>>199 reliably should be in section 6 (Proxy behavior).
>
>Same as for offer/answer.
>
>Since the 199 procedures related to offer/answer and reliable responses
>are special, I think it is good to have specific chapters for it.
>

The requirements/restrictions for reliable provisional responses are
very important. They could easily be missed by having them in a
separate chapter in the back of the document. Again, all UAS behavior
requirements should be in section 5 (or sub-section) and all proxy
behavior requirements should be in chapter 6 (or sub-section).

>Regards,
>
>Christer
>

cheers,
(-:bob

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> Dan Wing wrote:
> > Thanks for publishing that attack.  It has similarities 
> > with the attack described by Hadriel in
> > http://tools.ietf.org/html/draft-kaplan-sip-baiting-attack.
> 
> Yes, it also modifies part of the request which are not signed by the 
> authentication/signature algorithm. I should mention this draft in a 
> later version.
> 
> However, the goal is not the same. Hadriel's draft is focused on 
> impersonation, whereby the other draft is focused on breaking the 
> authentication done at the proxy.

By breaking the authentication the attacker can impersonate
another user.  For example, that proxy might create an RFC4474 
signature.

> The second difference is that it works right now with any 
> publicly reachable SIP provider.

Yes, I agree that makes it much more interesting.

-d



From rco@iptel.org  Thu Mar  5 07:45:26 2009
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Dan Wing wrote:
>> Dan Wing wrote:
>>     
>>> Thanks for publishing that attack.  It has similarities 
>>> with the attack described by Hadriel in
>>> http://tools.ietf.org/html/draft-kaplan-sip-baiting-attack.
>>>       
>> Yes, it also modifies part of the request which are not signed by the 
>> authentication/signature algorithm. I should mention this draft in a 
>> later version.
>>
>> However, the goal is not the same. Hadriel's draft is focused on 
>> impersonation, whereby the other draft is focused on breaking the 
>> authentication done at the proxy.
>>     
>
> By breaking the authentication the attacker can impersonate
> another user.  For example, that proxy might create an RFC4474 
> signature.
>
>   
Another interresting application indeed! As it looks like, a lot of 
people on the list have very good ideas that we will have to document in 
version 01 (if it is the will of the WG, of course!).

Regards
Raphael.
>> The second difference is that it works right now with any 
>> publicly reachable SIP provider.
>>     
>
> Yes, I agree that makes it much more interesting.
>
> -d
>
>   


From theo@crazygreek.co.uk  Thu Mar  5 08:01:20 2009
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From: Theo Zourzouvillys <theo@crazygreek.co.uk>
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On Thu, Mar 5, 2009 at 2:41 PM, Raphael Coeffic <rco@iptel.org> wrote:

> I really think that mutual authentication is the way to go.

MTLS is indeed a good way to go for so many reasons, but it's not the
only way this can be fixed.

For the fig 3 attack, receiving the response down the same stream
oriented connection indicates it came form the proxy, not the
attacker.  In the case of datagram, an adjustment to
draft-zourzouvillys-via-cookies to get the proxy to re-add the cookie
value in the response would solve the direct-to-source response
scenario without having the evil DNS lookup hacks in place.

> Having the BCP gathered into this document (or another one, I don't care)
> would be a good thing to do. If you let people think by themselves, you
> might end-up with the situation we have right now.

We'll end up with insecure implementations whatever we do - although
education is the key to minimising them.

> If you provide me with the input, I'd be very happy to document those
> issues.

i'll see if i can find time :-)

 ~ Theo

From HKaplan@acmepacket.com  Thu Mar  5 08:32:36 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Raphael Coeffic <rco@iptel.org>, Dan Wing <dwing@cisco.com>, "sip@ietf.org" <sip@ietf.org>
Date: Thu, 5 Mar 2009 11:33:00 -0500
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Raphael Coeffic
> Sent: Thursday, March 05, 2009 4:48 AM
>
> The second difference is that it works
> right now with any publicly reachable SIP provider.

I think I know what you mean by this statement, but maybe I don't.  I think=
 you mean it "works" right now only on a publicly reachable SIP provider th=
at accepts INVITE requests from non-Registered Contacts.  No?  Or do you me=
an you've done testing of all publicly reachable SIP providers and found th=
is to be an issue for all of them, right now?

Since the REGISTERs are almost always challenged, this attack is not an iss=
ue for providers which reject INVITEs from non-Registered Contacts.  Since =
SBC's have offered that restriction policy for at least the past 5 years, a=
nd presumably outbound-proxies do too (because it's a very popular policy),=
 then you're really only talking about a theoretical subset of publicly rea=
chable SIP Providers right now.

And this is not to say the attack isn't interesting, or that publicly reach=
able SIP providers actually *use* such security policies.  I've been surpri=
sed before about how some of them don't actually employ the security mechan=
isms they have at their disposal.  But not using available mechanisms is ve=
ry different from not having a way to do them.

-hadriel


From HKaplan@acmepacket.com  Thu Mar  5 08:43:44 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Theo Zourzouvillys <theo@crazygreek.co.uk>, Raphael Coeffic <rco@iptel.org>
Date: Thu, 5 Mar 2009 11:44:11 -0500
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of The=
o
> Zourzouvillys
> Sent: Thursday, March 05, 2009 9:31 AM
>
> The only issue i see is in fig 3, how to identify a next hop is
> actually proxy.com and not bob, as a proxy will commonly RR with a URI
> such as node-01.proxy.com instead of proxy.com, or even use an IP
> address in it.  ...  The second way is
> to always send requests to proxy.com (i.e, using outboind), even in
> dialog ones (and indeed many UAs have a "send in-dialog requests via
> outbound proxy" option,which makes this attack go away) - although
> this might breaks some broken SBCs out there that SPs deploy today.

Ummm... in what way?  The SBC *is* the outbound proxy.  And figure-3 isn't =
possible since clearly SBC's remove Alice's Via and Contact URI's, such tha=
t Bob won't know how to bypass the SBC to reach Alice.  And if the SBC does=
n't do that, then you're right it's broken, and technically not an SBC but =
more like a 3GPP P-CSCF.  (And I heave heard some "SBC's" are really inline=
 as ALG's, but the ALG model is known to be broken in real-world SIP use)

-hadriel

From theo@crazygreek.co.uk  Thu Mar  5 08:49:00 2009
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Subject: Re: [Sip] draft-state-sip-relay-attack-00
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On Thu, Mar 5, 2009 at 4:44 PM, Hadriel Kaplan <HKaplan@acmepacket.com> wro=
te:

> And if the SBC doesn't do that,
> then you're right it's broken, and technically not an SBC but more like a=
 3GPP P-CSCF. =A0(And I heave heard some "SBC's" are really inline as ALG's=
, but the ALG model is known to be broken in real-world SIP use)

Indeed this is what i was referring too - although i'm sure there are
other types that are just as broken, not *all* vendors make SBCs that
work well.

if only we had a name for our rhinos, giraffes and gazelle, then we'd
be able to call them by their names, rather than just "aniamls".

:-)

 ~ Theo

From john.elwell@siemens.com  Thu Mar  5 09:14:05 2009
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Date: Thu, 05 Mar 2009 17:14:30 +0000
From: "Elwell, John" <john.elwell@siemens.com>
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Subject: Re: [Sip] I-D Inaction: draft-kaplan-sip-secure-call-id-00.txt
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Hadriel,

Should we specify something that would make life easier for
proxies/B2BUAs to detect compliance with this, e.g., a magic string at
the start of the call-ID, rather than simply the absence of "@"?

John=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Hadriel Kaplan
> Sent: 05 March 2009 00:29
> To: SIP
> Subject: [Sip] I-D Inaction: draft-kaplan-sip-secure-call-id-00.txt
>=20
> Howdy,
> In an effort to avoid endless thrashing, I will be updating=20
> the Session-ID draft to only be about troubleshooting use.
> For dialog matching issues, I have followed Adam's advice and=20
> submitted a separate draft to update RFC 3261, linked below.
> This issue will only get worse, and the code changes to do=20
> this should be fairly small for most folks, I would think.
> And it's only 6 pages, so we should be able to agree on this=20
> in about 3 years.  :)
>=20
> A New Internet-Draft is available from the on-line=20
> Internet-Drafts directories.
>=20
>         Title           : A Secure Call-ID for the Session=20
> Initiation Protocol (SIP)
>         Author(s)       : H. Kaplan
>         Filename        : draft-kaplan-sip-secure-call-id-00.txt
>         Pages           : 6
>         Date            : 2009-03-04
>=20
> Many SIP devices generate Call-ID values which contain their=20
> system IP Address, due to examples and normative text in RFC=20
> 3261.  This has led to some middleboxes, such as SBC's, to=20
> change the Call-ID for security reasons.  This draft updates=20
> RFC 3261 to require SIP User Agents to generate benign=20
> Call-IDs, in such a manner that they can be detected as=20
> secure and not need to be changed.
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-kaplan-sip-secure-ca
> ll-id-00.txt
>=20
>=20
> -hadriel
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

From HKaplan@acmepacket.com  Thu Mar  5 15:43:50 2009
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To: "Elwell, John" <john.elwell@siemens.com>, SIP <sip@ietf.org>
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> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com]
> Sent: Thursday, March 05, 2009 12:15 PM
>
> Should we specify something that would make life easier for
> proxies/B2BUAs to detect compliance with this, e.g., a magic string at
> the start of the call-ID, rather than simply the absence of "@"?

At one point I was thinking of just making it either be an "invalid" host:
Call-ID: randomstuffheregibberish@invalid

Or a fixed-length hex-ascii type thing (like Session-ID).  But then someone=
 pointed out to me that some devices already don't put their IP/host in it,=
 so there's no reason to go make them change their implementation, nor repl=
ace their call-id until they do.

The thing is I think that a proxy/b2bua which cares about this for security=
 properties, would need/want to scan the whole call-id for something which =
looks like a host/IP anyway, to make a dynamic decision.  So having a magic=
 string won't help them.  And B2BUA's which change it for non-security reas=
ons will change it no matter what, because it's not the privacy property th=
ey care about.

-hadriel

From theo@crazygreek.co.uk  Thu Mar  5 16:35:59 2009
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References: <1236104807.3740.48.camel@victoria-pingtel-com.us.nortel.com> <167dfb9b0903040434x4b76e3fex83f94358e64a085@mail.gmail.com> <6EF7E0CD-58D2-4218-9408-1FC584DE7C04@softarmor.com> <1236200991.3771.45.camel@victoria-pingtel-com.us.nortel.com> <E6C2E8958BA59A4FB960963D475F7AC314C232BD20@mail>
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From: Theo Zourzouvillys <theo@crazygreek.co.uk>
To: Hadriel Kaplan <HKaplan@acmepacket.com>
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Cc: SIP <sip@ietf.org>, Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] Question regarding conflicting grammar for IPV6 SIP URI andRFC 3986
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On Wed, Mar 4, 2009 at 10:54 PM, Hadriel Kaplan <HKaplan@acmepacket.com> wr=
ote:

> How about: ignorance is bliss? =A0:)
>
> Seriously, what does RFC 3986 do??

   The URI syntax defines a grammar that is a superset of all
   valid URIs, allowing an implementation to parse the common components
   of a URI reference without knowing the scheme-specific requirements
   of every possible identifier.

> How does it affect anything for common URI formats not to comply with it?

We'd be saying that a SIP URI is no longer guaranteed to be compatible
with the generic URI syntax, and there in lies dragons.

While it would unlikely have any direct effect on SIP elements
themselves, there are a whole bunch of assumptions made (and we should
be able to continue to make) around a SIP URI being compatible with
the syntax of a URI, notably in 3rd party tools.  some things that
spring to mind that would be affected are:

 - XML Schema syntax for a URI
 - in <a href=3D"xxx"> - i.e, any hyperlink on a HTML page.
 - PHP (or any other language) parse_uri()
 - draft-roach-sip-http-subscribe Link header values
 - subjectAltName URI in a SSL certificate
 - a RDBMS "URI" data type

... being able to put a SIP URI anywhere that there is a "URI" field
is a fairly useful thing for developers when it comes to re-usability

even worse, anything that presumes they are a URI will work most the
time, until an IPv6 address is used.

> (or really, for *it* not to comply with common URI formats)

i'm not sure how many other common URI formats have notably been
affected by the missing IPv6 yet: the lack of universal IPv6
deployment along with SIP being fairly likely to want to include
literal IPv6 addresses in the URI makes it more obvious for us to
notice other other schemes, so perhaps we're just ahead of the game?

 ~ Theo

From munakata.mayumi@lab.ntt.co.jp  Thu Mar  5 16:56:32 2009
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From: Mayumi Munakata <munakata.mayumi@lab.ntt.co.jp>
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Subject: [Sip] Update of draft-ietf-sip-ua-privacy
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All;

I have just updated the ua-privacy draft
to incorporate all the WGLC comments.
http://www.ietf.org/internet-drafts/draft-ietf-sip-ua-privacy-06.txt

Changes from the previous version are as follows:

1) In Section 4.1,
changed the text in the last paragraph
to make it clear that usage of a third-party GRUU server
is a possible solution but out of scope.

2) In Section 5.1.1,
changed the text in the second paragraph
in order not to anonymize the Contact in 3xx responses,
REGISTER requests and REGISTER 200 responses.

3) In Section 6,
added a paragraph to the Security Considerations section
that explains the concern of 'reg' event package [RFC3680].

The editors believe that all the WGLC comments are solved
and incorporated into the document.

Thanks,
Mayumi

From rfc-editor@rfc-editor.org  Thu Mar  5 16:57:20 2009
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Subject: [Sip] RFC 5478 on IANA Registration of New Session Initiation Protocol (SIP) Resource-Priority Namespaces
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 5478

        Title:      IANA Registration of New Session 
                    Initiation Protocol (SIP) Resource-Priority Namespaces 
        Author:     J. Polk
        Status:     Standards Track
        Date:       March 2009
        Mailbox:    jmpolk@cisco.com
        Pages:      6
        Characters: 12810
        Updates/Obsoletes/SeeAlso:   None

        I-D Tag:    draft-ietf-sip-rph-new-namespaces-04.txt

        URL:        http://www.rfc-editor.org/rfc/rfc5478.txt

This document creates additional Session Initiation Protocol (SIP)
Resource-Priority namespaces to meet the requirements of the US
Defense Information Systems Agency, and places these namespaces in
the IANA registry.  [STANDARDS TRACK]

This document is a product of the Session Initiation Protocol Working Group of the IETF.

This is now a Proposed Standard Protocol.

STANDARDS TRACK: This document specifies an Internet standards track
protocol for the Internet community,and requests discussion and suggestions
for improvements.  Please refer to the current edition of the Internet
Official Protocol Standards (STD 1) for the standardization state and
status of this protocol.  Distribution of this memo is unlimited.

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.


	Title           : UA-Driven Privacy Mechanism for SIP
	Author(s)       : M. Munakata, et al.
	Filename        : draft-ietf-sip-ua-privacy-06.txt
	Pages           : 11
	Date            : 2009-03-05

This document defines a guideline for a user agent to generate an
anonymous SIP message by utilizing mechanisms such as GRUU (Globally
Routable User Agent URIs) and TURN (Traversal Using Relays around
NAT) without the need for a privacy service defined in RFC 3323.

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Theo Zourzouvillys <theo@crazygreek.co.uk>
Date: Thu, 5 Mar 2009 21:28:13 -0500
Thread-Topic: [Sip] Question regarding conflicting grammar for IPV6 SIP URI andRFC 3986
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Subject: Re: [Sip] Question regarding conflicting grammar for IPV6 SIP URI andRFC 3986
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OK, sounds useful enough - in a previous email from Cullen on this topic (w=
ith his AD hat), he suggested emailing the APPS Area list (apps-disucss@iet=
f.org) since rfc3986 is in their camp.

-hadriel


> -----Original Message-----
> From: Theo Zourzouvillys [mailto:theo@crazygreek.co.uk]
> Sent: Thursday, March 05, 2009 7:36 PM
>
> > How does it affect anything for common URI formats not to comply with
> it?
>
> We'd be saying that a SIP URI is no longer guaranteed to be compatible
> with the generic URI syntax, and there in lies dragons.
>
> While it would unlikely have any direct effect on SIP elements
> themselves, there are a whole bunch of assumptions made (and we should
> be able to continue to make) around a SIP URI being compatible with
> the syntax of a URI, notably in 3rd party tools.  some things that
> spring to mind that would be affected are:
>
>  - XML Schema syntax for a URI
>  - in <a href=3D"xxx"> - i.e, any hyperlink on a HTML page.
>  - PHP (or any other language) parse_uri()
>  - draft-roach-sip-http-subscribe Link header values
>  - subjectAltName URI in a SSL certificate
>  - a RDBMS "URI" data type
>
> ... being able to put a SIP URI anywhere that there is a "URI" field
> is a fairly useful thing for developers when it comes to re-usability
>
> even worse, anything that presumes they are a URI will work most the
> time, until an IPv6 address is used.
>
> > (or really, for *it* not to comply with common URI formats)
>
> i'm not sure how many other common URI formats have notably been
> affected by the missing IPv6 yet: the lack of universal IPv6
> deployment along with SIP being fairly likely to want to include
> literal IPv6 addresses in the URI makes it more obvious for us to
> notice other other schemes, so perhaps we're just ahead of the game?
>
>  ~ Theo

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Thread-Topic: [Sip] Question regarding conflicting grammar for IPV6 SIP URI andRFC 3986
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From: "Horvath, Ernst" <ernst.horvath@siemens.com>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>, "Theo Zourzouvillys" <theo@crazygreek.co.uk>
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Cc: SIP <sip@ietf.org>, Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] Question regarding conflicting grammar for IPV6 SIP URI andRFC 3986
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I would also think the problem should be solved within RFC3986, for a
pure economic reason.=20
On a quick scan through registered URI schemes I identified the
following schemes having the same problem as sip/sips:=20

	mailto, im, pres, h323, iax, xmpp.

and I may have missed some. Makes more than have a dozen RFCs to be
updated against one (RFC3986).

Regards,
Ernst


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Hadriel Kaplan
> Sent: Friday, March 06, 2009 3:28 AM
> To: Theo Zourzouvillys
> Cc: SIP; Dean Willis
> Subject: Re: [Sip] Question regarding conflicting grammar for=20
> IPV6 SIP URI andRFC 3986
>=20
>=20
> OK, sounds useful enough - in a previous email from Cullen on=20
> this topic (with his AD hat), he suggested emailing the APPS=20
> Area list (apps-disucss@ietf.org) since rfc3986 is in their camp.
>=20
> -hadriel
>=20
>=20
> > -----Original Message-----
> > From: Theo Zourzouvillys [mailto:theo@crazygreek.co.uk]
> > Sent: Thursday, March 05, 2009 7:36 PM
> >
> > > How does it affect anything for common URI formats not to=20
> comply with
> > it?
> >
> > We'd be saying that a SIP URI is no longer guaranteed to be=20
> compatible
> > with the generic URI syntax, and there in lies dragons.
> >
> > While it would unlikely have any direct effect on SIP elements
> > themselves, there are a whole bunch of assumptions made=20
> (and we should
> > be able to continue to make) around a SIP URI being compatible with
> > the syntax of a URI, notably in 3rd party tools.  some things that
> > spring to mind that would be affected are:
> >
> >  - XML Schema syntax for a URI
> >  - in <a href=3D"xxx"> - i.e, any hyperlink on a HTML page.
> >  - PHP (or any other language) parse_uri()
> >  - draft-roach-sip-http-subscribe Link header values
> >  - subjectAltName URI in a SSL certificate
> >  - a RDBMS "URI" data type
> >
> > ... being able to put a SIP URI anywhere that there is a "URI" field
> > is a fairly useful thing for developers when it comes to=20
> re-usability
> >
> > even worse, anything that presumes they are a URI will work most the
> > time, until an IPv6 address is used.
> >
> > > (or really, for *it* not to comply with common URI formats)
> >
> > i'm not sure how many other common URI formats have notably been
> > affected by the missing IPv6 yet: the lack of universal IPv6
> > deployment along with SIP being fairly likely to want to include
> > literal IPv6 addresses in the URI makes it more obvious for us to
> > notice other other schemes, so perhaps we're just ahead of the game?
> >
> >  ~ Theo
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.


	Title           : Addressing Record-Route issues in the Session Initiation Protocol (SIP)
	Author(s)       : T. Froment, et al.
	Filename        : draft-ietf-sip-record-route-fix-06.txt
	Pages           : 24
	Date            : 2009-03-06

A typical function of a Session Initiation Protocol (SIP) Proxy is to
insert a Record-Route header into initial, dialog creating requests
in order to make subsequent, in-dialog requests pass through it.
This header contains a SIP Uniform Resource Identifier (URI)
indicating where and how the subsequent requests should be sent to
reach the proxy.  Like any SIP URI, it can contain sip or sips
schemes, IPv4 or IPv6 addresses, and URI parameters that could
influence the routing such as the transport parameter (for example
transport=tcp), or a compression indication like "comp=sigcomp".
When a proxy has to change some of those parameters between its
incoming and outgoing interfaces (multi-homed proxies, transport
protocol switching or IPv4 to IPv6 scenarios...), the question arises
on what should be put in Record-Route header(s).  It is just not
possible to make one header having the characteristics of both sides
at the same time.  This document aims to clarify these scenarios and
fix bugs already identified on this topic; it formally recommends the
use of the double Record-Route technique as an alternative to the
current RFC3261 text, which describes only a Record-Route rewriting
solution.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-record-route-fix-06.txt

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dale Worley <dworley@nortel.com>, "Elwell, John" <john.elwell@siemens.com>
Date: Fri, 6 Mar 2009 12:05:19 -0500
Thread-Topic: [Sip] SIP Tracing Facility
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Subject: Re: [Sip] SIP Tracing Facility
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Dal=
e
> Worley
> Sent: Thursday, March 05, 2009 10:26 AM
>
> On Thu, 2009-03-05 at 08:16 +0000, Elwell, John wrote:
> > What about non-INVITEs? Provisional responses are designed for INVITEs
> > and I am not certain about their applicability to non-INVITEs.
>
> Yes, provisional responses to non-INVITEs may be slightly out of
> specification, but I don't expect any SIP element to malfunction if it
> receives a provisional response to a non-INVITE request.  OTOH, I would
> expect most uses of 'trace' to be attached to INVITEs, although that
> makes me think that tracing an OPTIONS might be a useful test tool.

OPTIONS doesn't always get routed through the same path as INVITE, fwiw.


> (Which leads to an unexpected observation:  RFC 3261 presupposes that
> all non-INVITE responses are processed promptly, but in the face of
> transport errors, this is not always so.  In sipX, we've adjusted
> transaction handling so that if a re-send of a non-INVITE message is
> seen the recipient sends a 100 to quench further re-sends.  It seems to
> work exactly as expected, and noticeably reduces re-sends in situations
> with transport errors.)

I agree for most methods, but we've seen problems with 100 for the OPTIONS.=
  Some devices use the OPTIONS as a reachability ping/test, so sometimes th=
e 100 makes them think the far-end is reachable even if they get a 408 late=
r; in other cases some devices look for specific response codes to detect a=
dministrative out-of-service, for example a 500/503 to an OPTIONS sometimes=
 is used for that, and sometimes they didn't code for the 100 Trying as bei=
ng a possibility, so they see it and assume it's not out-of-service, and th=
e 503 sent later doesn't get processed the same.  I consider these bugs/bro=
ken implementations, but just noting sending the 100 is not bug-free.

 -hadriel

From HKaplan@acmepacket.com  Fri Mar  6 09:51:47 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dale Worley <dworley@nortel.com>, SIP <sip@ietf.org>
Date: Fri, 6 Mar 2009 12:52:14 -0500
Thread-Topic: [Sip] SIP Tracing Facility
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Hi Dale,

I appreciate that doing troubleshooting of SIP in SIP is very appealing, bu=
t I'm worried about it.  There are some real security issues with this, eve=
n if we don't want to admit it.  The reality is that if this trace mechanis=
m is actually adopted by any sizeable measure, then middleboxes will be for=
ced to munge the messages more than they already do, to prevent it from bei=
ng used/work.  In other words it's a self-defeating mechanism.

Today you can do incrementing Max-Forwards traceroute, similar to the ICMP =
traceroute TTL trick, and even that's gotten some folks worried enough to m=
ake us munge the Max-Forwards value (despite the obvious dangers in doing s=
o).  But anyway why isn't that type of trace enough for your needs?


Also, a couple specific questions/comments regarding this draft:

1) You use the term "final response" a couple times as being sent in the 17=
0's body.  How would a final response be sent in a 170 response, since the =
170 would be before a final response?

2) You state that a UAC should not create an early-dialog for the 170, but =
for any B2BUA in the path which did not know about this new trace mechanism=
, wouldn't it create early dialogs on the b2bua since it would treat it as =
a 183?

-hadriel


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Dal=
e
> Worley
> Sent: Wednesday, March 04, 2009 1:29 PM
> To: SIP
> Subject: [Sip] SIP Tracing Facility
>
> I wrote up the "170" trace proposal as an I-D:
>
> A new version of I-D, draft-worley-trace-00.txt has been successfuly
> submitted by Dale Worley and posted to the IETF repository.
>
> Filename:        draft-worley-trace
> Revision:        00
> Title:           SIP Tracing Facility
> Creation_date:   2009-03-03
> WG ID:           Independent Submission
> Number_of_pages: 18
>
> Abstract:
> This document defines a SIP option tag, "trace", to be used within
> SIP messages to request that SIP elements (both proxies and UASs)
> that receive the message reflect to the UAC the request they received
> and the response they gave by encapsulating the request and response
> in a provisional response.  A new provisional response code "170" is
> defined to carry the request and response.  This option tag is
> expected to be used solely for diagnostic purposes.
>
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

From eburger@standardstrack.com  Fri Mar  6 09:59:12 2009
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Cc: SIP IETF <sip@ietf.org>
Subject: Re: [Sip] SIP Tracing Facility
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Wouldn't SIP-CLF be a better way to go?

On Mar 6, 2009, at 12:52 PM, Hadriel Kaplan wrote:

> Hi Dale,
>
> I appreciate that doing troubleshooting of SIP in SIP is very  
> appealing, but I'm worried about it.  There are some real security  
> issues with this, even if we don't want to admit it.  The reality is  
> that if this trace mechanism is actually adopted by any sizeable  
> measure, then middleboxes will be forced to munge the messages more  
> than they already do, to prevent it from being used/work.  In other  
> words it's a self-defeating mechanism.
>
> Today you can do incrementing Max-Forwards traceroute, similar to  
> the ICMP traceroute TTL trick, and even that's gotten some folks  
> worried enough to make us munge the Max-Forwards value (despite the  
> obvious dangers in doing so).  But anyway why isn't that type of  
> trace enough for your needs?
>
>
> Also, a couple specific questions/comments regarding this draft:
>
> 1) You use the term "final response" a couple times as being sent in  
> the 170's body.  How would a final response be sent in a 170  
> response, since the 170 would be before a final response?
>
> 2) You state that a UAC should not create an early-dialog for the  
> 170, but for any B2BUA in the path which did not know about this new  
> trace mechanism, wouldn't it create early dialogs on the b2bua since  
> it would treat it as a 183?
>
> -hadriel
>
>
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf  
>> Of Dale
>> Worley
>> Sent: Wednesday, March 04, 2009 1:29 PM
>> To: SIP
>> Subject: [Sip] SIP Tracing Facility
>>
>> I wrote up the "170" trace proposal as an I-D:
>>
>> A new version of I-D, draft-worley-trace-00.txt has been successfuly
>> submitted by Dale Worley and posted to the IETF repository.
>>
>> Filename:        draft-worley-trace
>> Revision:        00
>> Title:           SIP Tracing Facility
>> Creation_date:   2009-03-03
>> WG ID:           Independent Submission
>> Number_of_pages: 18
>>
>> Abstract:
>> This document defines a SIP option tag, "trace", to be used within
>> SIP messages to request that SIP elements (both proxies and UASs)
>> that receive the message reflect to the UAC the request they received
>> and the response they gave by encapsulating the request and response
>> in a provisional response.  A new provisional response code "170" is
>> defined to carry the request and response.  This option tag is
>> expected to be used solely for diagnostic purposes.
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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From HKaplan@acmepacket.com  Fri Mar  6 10:08:55 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Eric Burger <eburger@standardstrack.com>, Dale Worley <dworley@nortel.com>
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Yes that's what I was thinking too - for basic connectivity check type thin=
gs, do SIP ping/traceroute.  To troubleshoot actual proxy and UA behavior i=
ssues, retargeting issues, perform any type of root-cause, etc., you need l=
ogs and there's just no getting away from that.  And only the authoritative=
 administrator of each SIP device can (and *should*) be able to get such in=
fo.

The real question is whether we can (or could) provide standardized ways of=
 helping admin's get the info they need.
And I think that is actually a really good topic for a BOF, or at least a d=
iscussion on RAI-DISPATCH.

-hadriel

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Eri=
c
> Burger
> Sent: Friday, March 06, 2009 1:00 PM
>
> Wouldn't SIP-CLF be a better way to go?

From vkg@alcatel-lucent.com  Fri Mar  6 10:27:13 2009
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Eric Burger wrote:
> Wouldn't SIP-CLF be a better way to go?

Hadriel Kaplan wrote:
> Yes that's what I was thinking too - for basic connectivity check
> type things, do SIP ping/traceroute.  To troubleshoot actual proxy
> and UA behavior issues, retargeting issues, perform any type of
> root-cause, etc., you need logs and there's just no getting away from
> that.  And only the authoritative administrator of each SIP device
> can (and *should*) be able to get such info.
> 
> The real question is whether we can (or could) provide standardized
> ways of helping admin's get the info they need. And I think that is
> actually a really good topic for a BOF, or at least a discussion on
> RAI-DISPATCH.

Funny you should mention a BoF.  Eric has also suggested that we
hold a Bar BoF to discuss sipping-clf.

I am modifying sipping-clf based on Dale's extensive comments
from the last round and from other comments received.  It
will be submitted before the Monday deadline and we can think
of holding at least a Bar BoF on this.

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From dworley@nortel.com  Fri Mar  6 11:41:34 2009
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On Thu, 2009-03-05 at 14:40 +0100, Nils Ohlmeier wrote:
> One thing which is not that obvious but is implictly a requirement for the
> attack: the proxies has to challenge in-dialog requests. I do not see a
> big benefit in challeging in-dialog requests as these are hopefully
> rejected by the remote side if no matching dialog exists. If the UA would
> know that his proxy does not challenge in-dialog requests it could simply
> ignore the challenge :-)

Except that there are legitimate uses for challenging in-dialog
requests:  sipX uses it to allow a phone to transfer a caller to any
destination that the executing phone has permission to call.  The first
step of this process is that when the executing phone sends a REFER, the
proxy challenges the REFER so that the executing phone attaches its
credentials to the REFER.  The proxy then analyzes these credentials to
determine the user that is responsible for the transfer operation, etc.
Without the in-dialog challenge, there is no way for the proxy to
determine the user that is responsible for transfer operation.

Dale



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Am 06.03.2009 20:41 Uhr, schrieb Dale Worley:
> On Thu, 2009-03-05 at 14:40 +0100, Nils Ohlmeier wrote:
>> One thing which is not that obvious but is implictly a requirement for the
>> attack: the proxies has to challenge in-dialog requests. I do not see a
>> big benefit in challeging in-dialog requests as these are hopefully
>> rejected by the remote side if no matching dialog exists. If the UA would
>> know that his proxy does not challenge in-dialog requests it could simply
>> ignore the challenge :-)
>
> Except that there are legitimate uses for challenging in-dialog
> requests:  sipX uses it to allow a phone to transfer a caller to any
> destination that the executing phone has permission to call.  The first
> step of this process is that when the executing phone sends a REFER, the
> proxy challenges the REFER so that the executing phone attaches its
> credentials to the REFER.  The proxy then analyzes these credentials to
> determine the user that is responsible for the transfer operation, etc.
> Without the in-dialog challenge, there is no way for the proxy to
> determine the user that is responsible for transfer operation.

Good point. Except that a REFER is not relevant for this replay attack. 
So would re-phrase it: I do not see a big benefit in challeging 
in-dialog INVITE requests. :-)

Greetings
   Nils

From theo@crazygreek.co.uk  Fri Mar  6 14:29:03 2009
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On Fri, Mar 6, 2009 at 10:21 PM, Nils Ohlmeier <lists@ohlmeier.org> wrote:

> Except that a REFER is not relevant for this replay attack

an authenticated OOD REFER may be used to initiate a call with some SIP devices.

 ~ Theo

From lists@ohlmeier.org  Fri Mar  6 14:51:44 2009
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Subject: Re: [Sip] draft-state-sip-relay-attack-00
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Am 06.03.2009 23:29 Uhr, schrieb Theo Zourzouvillys:
> On Fri, Mar 6, 2009 at 10:21 PM, Nils Ohlmeier<lists@ohlmeier.org>  wrote:
>
>> Except that a REFER is not relevant for this replay attack
>
> an authenticated OOD REFER may be used to initiate a call with some SIP devices.

Sure, but that is totally different story (of another bad UA 
implementation) which is not related to this auth relay attack.

   Nils

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Subject: Re: [Sip] draft-ietf-sip-info-events-03: comments and questions
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On Feb 16, 2009, at 7:18 PM, Brett Tate wrote:

> Howdy,
>
> The following are a few comments and questions concerning draft-ietf- 
> sip-info-events-03.
>
> Thanks,
> Brett
>
> -------------
>
> Section 3.1 paragraph 1:  Concerning "MUST advertise" and  
> "provisional 1xx", please clarify that the "MUST advertise" is not  
> applicable to the 100 response.  Does the "MUST advertise" and  
> "provisional 1xx" apply to all (first and subsequent within dialog)  
> INVITE non 101-199 (intentionally indicated 199)?  Concerning ACK  
> for initial INVITE's 2xx, does the "MUST advertise" always apply?

I would offer "MUST advertise" is appropriate. What is the rationale  
to not require it? We are expecting "early INFO," so no reason to  
screw up UACs by having random appearances and disappearances of Recv- 
Info headers.


> Section 3.1 paragraph 2 sentences 1 and 2: Sentence 1 indicates  
> "session parameters" and sentence 2 indicates "dialog parameters"  
> concerning Info Package negotiation, I'm not sure when the  
> negotiation/renegotiation occurs.  Does it follow SDP offer/answer  
> rules or something else; if something else, what?  I have the same  
> question concerning "dialog refresh" text discussed later within the  
> same section since incorrect interpretations might trigger  
> deactivation.

Should be dialog parameters everywhere.  Would that clear up the  
confusion?

> Section 4.3: When sending a 489, it indicates "MUST terminate the  
> INVITE dialog".  The "MUST terminate" seems severe even if all the  
> negotiation race conditions are adequately addressed.  The mentioned  
> "protocol failure" reason seems contrary to the "strictly send and  
> lenient receive" mantra.

Maybe living in New Hampshire has made me more conservative. A lot of  
folks will today say that while "Be conservative in what you send and  
liberal in what you receive" may have been the greatest mistake we  
ever made in protocol design. Let us look at this from a practical  
perspective. If you are a bad implementor and send a 489 you didn't  
mean to, who knows what else you will get wrong. Conversely, if you  
are a bad implementor and forget to tear down a 489-ed INFO, bad  
things probably will not happen.

I will defer to the WG on this one. Any other opinions?

> Section 5.1 Table 1: The usage of "g" seems potentially outdated or  
> incorrectly reference since discussed within RFC 2543 instead of RFC  
> 3261.  The same applies to usage Via row concerning "(2)".

Poor cutting & pasting. Will fix.

> Section 5.2.2: The "general-header" reference is outdated or  
> incorrectly reference since defined within RFC 2543 instead of RFC  
> 3261.  RFC 3261 uses "Message-Header".

Ditto.

> Section 7.7 paragraph 2: The following potentially could be reworded  
> since it is okay to return failure responses (401, 503, 500, ...)  
> even when overlapping occurs: "in the event a UAC issues overlapping  
> INFO requests for an Info Package, the UAS MUST NOT return an error  
> response".

Good idea.  Will do.


> Section 8: Why specifically defining and using "nil"?  I'm mainly  
> just curious of the benefit over not having defined "nil" for  
> similar use within the Supported header.

That is a good one. since we go to the trouble of registering "nil",  
it is just an Info-package-type.  Will fix.


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Inline.

On Mar 3, 2009, at 12:11 PM, Bob Penfield wrote:

>
> Here are some comments on the INFO draft:
>
> Section 2 says:
>
>   Each UA enumerates which Info Packages it can receive.  If the far
>   end indicates it can receive a package offered by the near end, the
>   near end can send INFO methods containing the payload for that
>   package.
>
> "offered" needs to changed to "supported" since an INFO sender does
> not indicate what packages it can send.

Correct.  Editing shrapnel. It is really in Section 1 (this is a note  
for me).

> Section 3 is called "Info Package Negotiation", but since we removed
> Send-Info, it is not a negotiation since there is no feedback from the
> peer UA. It is simply a declaration of receive capability. At first
> read, I was totally confused by section 3.1 because I was expecting
> some actual negotiation given the above statement in Section 2. There
> are a few other places in the document that mentions "negotiation"
> that also might need to be changed.

How about "Info Package Behavior"?

> I think the usage of UAC/UAS terminology in Section 3.1 is still
> confusing. I would prefer to see something like:
>
>   A UA supporting this document MUST advertise the set of Info
>   Packages it is willing to receive in Recv-Info header(s) in all
>   "INVITE dialog usage" requests and responses (101-199 and 2xx only)
>   for dialog establishment or target refresh. This includes INVITE,
>   UPDATE, PRACK and ACK and their non-failure responses.
>
>   A UA MUST NOT send INFO requests for a given INFO package until the
>   UA has received an "INVITE dialog usage" request or response (for
>   dialog establishment or target refresh) with a Recv-Info header
>   listing the given Info Package.
>
>   A UA MUST cease sending INFO requests for a given INFO package when
>   the UA receives an "INVITE dialog usage" request or response (for
>   dialog establishment or target refresh) that does not contain a
>   Recv-Info header listing the given Info Package.

Sounds good to me.

> Section 3.1 - "the target UA may not send form package P"
>
>   s/form/from/
>
> Section 3.1 - "... the other UA in the dial will assume the ..."
>
>  s/dial/dialog/
Got it.

> Section 3.1 - "server MAY terminate the session with a CANCEL or BYE"
>
>  When the "server" is the UAS of the request, CANCEL is not
>  appropriate, and BYE would not be appropriate if it is an initial
>  INVITE. In this case, the request would need to be rejected. When
>  the "server" is the UAC, I don't think CANCEL is appropriate since
>  it is the far-end UA of a specific dialog that the UAC has an issue
>  with, not all possible UAs in the case of an initial INVITE.

Whoops - clearly we are talking about the UAC in this case.


> Section 7 - Can Info Packages strengthen "MAY" to "SHOULD"?

In the real world, MAY = SHOULD.  This would not add any value.  The  
value of the wording here is to encourage more MUSTS :-)


> Section 8 - Why is the separator between Info-package-name and
> Info-package-param a DOT? Shouldn't it be a SEMI? I don't think that
> Info-package-param is similar to event-template in 3265. Its more like
> event-param.

Hmmm. That is one for my co-authors.  I would offer semicolon is the  
way to go.
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Cc: SIP IETF <sip@ietf.org>
Subject: Re: [Sip] WGLC on draft-ietf-sip-info-events
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Sounds sensible.

On Mar 4, 2009, at 3:47 AM, Elwell, John wrote:

> I agree with Bob that the use of UAC and UAS terminology is  
> confusing. I
> cite some particular examples below.
>
> 1. In 3 it defines the two terms, but early in 3.1 it states "The
> initiating UA (UAC of the INVITE)", which uses "UAC" in a sense
> different from what is defined.
>
> 2. "the UAC MUST NOT send
>   INFO requests for a given Info Package until the UAS lists the given
>   Info Package in a Recv-Info header."
> According to the definition of UAC, a UAC is the sender of an INFO
> request. So while waiting to receive Recv-Info it is not a UAC.
> Similarly, the other end is not a UAS by virtue of sending "Recv- 
> Info" -
> it only becomes a UAS when it receives an INFO request. Don't forget
> that a UA is a UAC or a UAS only in the context of a particular
> transaction, and before that transaction takes place and after that
> transaction has finished it is just a UA. Any attempt to change the
> meaning of UAC or UAS would be a departure from RFC 3261.
>
> 3. "A UAS lists multiple packages by enumerating the package name(s),
>   separated by commas, as values for the Recv-Info header in the
>   session establishment exchange.  A UAS can also list multiple
>   packages by including multiple Recv-Info headers...."
> We are talking about sending Recv-Info - nothing to do with an INFO
> transaction, so the term UAS is inappropriate.
>
> 4. "This initial dialog establishment
>   phase is the only time the UAS need be prepared to receive multiple
>   INFO requests, as we require post-session-establishment negotiation
>   to fully complete before a UAC can send an INFO request."
> followed shortly afterwards by:
> "A UAC MAY send an INFO
>   request on both an early and confirmed dialog."
> Either these are conflicting statements or there is some mix-up of  
> terms
> UAC and UAS.
>
> 5. In "Conventions Used in this Document" it states: "Since this
> document strictly follows RFC 3261, we refer to the UA
>   that issues the INVITE as the "initiating UA" and the UA that
>   responds to the INVITE as the "target UA" to remove any confusion."
> I don't think this is good practice, and I would even say that by  
> doing
> this we no longer strictly follow RFC 3261 terminology. If it is clear
> we are talking about an INVITE transaction we should be free to use  
> UAC
> and UAS.
>
> So in general, where it is clear which transaction we are talking  
> about
> (e.g., because it has appeared earlier in the paragraph concerned), we
> should just use UAC or UAS. Otherwise we should use expressions like
> "the UAC for the XXX transaction" or "the UA that sends the XXX
> request".
>
> John
>
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
>> Behalf Of Bob Penfield
>> Sent: 03 March 2009 17:12
>> To: DRAGE, Keith (Keith); SIP IETF
>> Subject: Re: [Sip] WGLC on draft-ietf-sip-info-events
>>
>>
>> Here are some comments on the INFO draft:
>>
>> Section 2 says:
>>
>>   Each UA enumerates which Info Packages it can receive.  If the far
>>   end indicates it can receive a package offered by the near end, the
>>   near end can send INFO methods containing the payload for that
>>   package.
>>
>> "offered" needs to changed to "supported" since an INFO sender does
>> not indicate what packages it can send.
>>
>> Section 3 is called "Info Package Negotiation", but since we removed
>> Send-Info, it is not a negotiation since there is no feedback from  
>> the
>> peer UA. It is simply a declaration of receive capability. At first
>> read, I was totally confused by section 3.1 because I was expecting
>> some actual negotiation given the above statement in Section 2. There
>> are a few other places in the document that mentions "negotiation"
>> that also might need to be changed.
>>
>> I think the usage of UAC/UAS terminology in Section 3.1 is still
>> confusing. I would prefer to see something like:
>>
>>   A UA supporting this document MUST advertise the set of Info
>>   Packages it is willing to receive in Recv-Info header(s) in all
>>   "INVITE dialog usage" requests and responses (101-199 and 2xx only)
>>   for dialog establishment or target refresh. This includes INVITE,
>>   UPDATE, PRACK and ACK and their non-failure responses.
>>
>>   A UA MUST NOT send INFO requests for a given INFO package until the
>>   UA has received an "INVITE dialog usage" request or response (for
>>   dialog establishment or target refresh) with a Recv-Info header
>>   listing the given Info Package.
>>
>>   A UA MUST cease sending INFO requests for a given INFO package when
>>   the UA receives an "INVITE dialog usage" request or response (for
>>   dialog establishment or target refresh) that does not contain a
>>   Recv-Info header listing the given Info Package.
>>
>> Section 3.1 - "the target UA may not send form package P"
>>
>>   s/form/from/
>>
>> Section 3.1 - "... the other UA in the dial will assume the ..."
>>
>>  s/dial/dialog/
>>
>> Section 3.1 - "server MAY terminate the session with a CANCEL or BYE"
>>
>>  When the "server" is the UAS of the request, CANCEL is not
>>  appropriate, and BYE would not be appropriate if it is an initial
>>  INVITE. In this case, the request would need to be rejected. When
>>  the "server" is the UAC, I don't think CANCEL is appropriate since
>>  it is the far-end UA of a specific dialog that the UAC has an issue
>>  with, not all possible UAs in the case of an initial INVITE.
>>
>> Section 7 - Can Info Packages strengthen "MAY" to "SHOULD"?
>>
>> Section 8 - Why is the separator between Info-package-name and
>> Info-package-param a DOT? Shouldn't it be a SEMI? I don't think that
>> Info-package-param is similar to event-template in 3265. Its more  
>> like
>> event-param.
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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From: Bob Penfield <BPenfield@acmepacket.com>
To: Eric Burger <eburger@standardstrack.com>, Brett Tate <brett@broadsoft.com>
Date: Sat, 7 Mar 2009 10:09:46 -0500
Thread-Topic: [Sip] draft-ietf-sip-info-events-03: comments and questions
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inline


>-----Original Message-----
>From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Eric=
 Burger
>Sent: Friday, March 06, 2009 4:56 PM
<snip>
>
>On Feb 16, 2009, at 7:18 PM, Brett Tate wrote:
>
<snip>
>> Section 4.3: When sending a 489, it indicates "MUST terminate the
>> INVITE dialog".  The "MUST terminate" seems severe even if all the
>> negotiation race conditions are adequately addressed.  The mentioned
>> "protocol failure" reason seems contrary to the "strictly send and
>> lenient receive" mantra.
>
>Maybe living in New Hampshire has made me more conservative. A lot of
>folks will today say that while "Be conservative in what you send and
>liberal in what you receive" may have been the greatest mistake we
>ever made in protocol design. Let us look at this from a practical
>perspective. If you are a bad implementor and send a 489 you didn't
>mean to, who knows what else you will get wrong. Conversely, if you
>are a bad implementor and forget to tear down a 489-ed INFO, bad
>things probably will not happen.
>
>I will defer to the WG on this one. Any other opinions?

RFC 5057 indicates that a 489 should destroy the dialog usage.
However, it also says:

   (12) 489 Bad Event:  In our example scenario, [5] declares that the
      subscription usage in which the NOTIFY is sent is terminated.
      This response is only valid in the context of SUBSCRIBE and
      NOTIFY.  UAC behavior for receiving this response to other methods
      is not specified, but treating it as an unknown 4xx is a
      reasonable practice.

I'm inclined to say that since the INFO is not suppose to change
dialog state, a 489 response to an INFO should not destroy the INVITE
dialog usage. With SUBSCRIBE/NOTIFY its a failure of the dialog
establishing/target-refresh request which is essential to the dialog
usage. But INFO is not essential to the INVITE dialog usage, so
applying the last sentence above makes sense to me.

cheers,
(-:bob

From jan@ryngle.com  Sat Mar  7 10:32:57 2009
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From: Jan Janak <jan@ryngle.com>
To: Nils Ohlmeier <lists@ohlmeier.org>
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Nils,

On 05-03 14:40, Nils Ohlmeier wrote:
> Hello,
> 
> first of all I agree that the draft describes a potential problem.
> 
> One thing which is not that obvious but is implictly a requirement for the
> attack: the proxies has to challenge in-dialog requests. 

It is not the proxy who challenges the in-dialog request, it is the
attacker (Bob). 

Even if the proxy is configured not to challenge in-dialog requests, or even
if digest authentication is not used at all on the proxy, as long as it
forwards the 407 from the attacker, the attack would still be possible. The
proxy (in figure 2) is only used to make sure that the 407 generated by Bob
makes it to Alice.

See figure 2 in the I-D, the 407 is generated by Bob based on the 407 he
received from p2.com. And the 407 from p2.com is a response to an
out-of-dialog INVITE.

So a requirement to make the attack possible is that the user agent responds
to challenges generated for in-dialog requests.

> I do not see a
> big benefit in challeging in-dialog requests as these are hopefully
> rejected by the remote side if no matching dialog exists. 

Yes, hopefully :-). But what if not? What if you deploy a poorly implemented
PSTN gateway which would happily process an in-dialog request with no matching
dialog?  Of course you can test the gateway before you deploy it if it is
under your control.

There are other cases where you know nothing about the target UA
implementation, such as when the users of your proxy call other users and you
have no control over the UA implementation they use. 

> - I never unterstood why a proxy should pass through the authentication
> request from a foreign domain. 

Because this is how it is specified in section 22.3 of RFC3261.

> In which scenario do I have credentials for
> two different domains, but the traffic including authentication is passed
> through only one of them?

One such scenario is that you have a UA configured with credentials for two
different ITSPs and the UA uses another proxy server as the outbound proxy for
both of them. This outbound proxy could be your corporate proxy sitting on a
firewall, NAT conditioning proxy, SBC, etc..

   Jan.

From jan@ryngle.com  Sat Mar  7 11:08:54 2009
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From: Jan Janak <jan@ryngle.com>
To: Theo Zourzouvillys <theo@crazygreek.co.uk>
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On 05-03 14:31, Theo Zourzouvillys wrote:
> However, you don't cover the more interesting cases of multi-hop proxy
> authentication or end-to-end WWW authentication: these are the harder
> ones to deal with, and may result in some "real" issues in SIP itself
> rather than shoddy implementations and insecure proxies.

Yes, after reading the discussion here I agree, perhaps the next revision of
the ID (if there is going to be any) should describe more difficult cases,
such as multi-hop authentication and challenging proxies reachable through one
common proxy. These are much harder cases to deal with.

   Jan.

From HKaplan@acmepacket.com  Sat Mar  7 11:18:32 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Jan Janak <jan@ryngle.com>, Nils Ohlmeier <lists@ohlmeier.org>
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Jan
> Janak
> Sent: Saturday, March 07, 2009 1:33 PM
>
> So a requirement to make the attack possible is that the user agent
> responds
> to challenges generated for in-dialog requests.

Right, and that the attacked domain accepts INVITEs from its AoR's with non=
-registered Contacts; or accepts INVITEs from its static AoR's to come in f=
rom unknown locations.  That's pretty rare in my world, but ymmv.


> > I do not see a
> > big benefit in challeging in-dialog requests as these are hopefully
> > rejected by the remote side if no matching dialog exists.
>
> Yes, hopefully :-). But what if not? What if you deploy a poorly
> implemented
> PSTN gateway which would happily process an in-dialog request with no
> matching
> dialog?  Of course you can test the gateway before you deploy it if it is
> under your control.

At that point you're describing something that is fundamentally broken.  Al=
l bets are off, including any preventative security measures you could take=
, etc.  I mean at that point you could just say "what if the user/password =
is published on a web site?"


> There are other cases where you know nothing about the target UA
> implementation, such as when the users of your proxy call other users and
> you
> have no control over the UA implementation they use.

Then get yourself a box that enforces the protocol rules in the middle. :)


> > - I never unterstood why a proxy should pass through the authentication
> > request from a foreign domain.
>
> Because this is how it is specified in section 22.3 of RFC3261.

And it would have to continue to do so.  There are actual use-cases for thi=
s.  I think there's even a reasonable use-case for challenging in-dialog re=
quests: connected-identity, for example.

But you don't even need to challenge in-dialog requests for this form of at=
tack: if the victim calls you, then you can challenge the initial INVITE.

-hadriel

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From: Jan Janak <jan@ryngle.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>
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Cc: "sip@ietf.org" <sip@ietf.org>, Dan Wing <dwing@cisco.com>
Subject: Re: [Sip] draft-state-sip-relay-attack-00
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On 05-03 11:33, Hadriel Kaplan wrote:
> 
> 
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> > Raphael Coeffic
> > Sent: Thursday, March 05, 2009 4:48 AM
> >
> > The second difference is that it works
> > right now with any publicly reachable SIP provider.
> 
> I think I know what you mean by this statement, but maybe I don't.  I think you mean it "works" right now only on a publicly reachable SIP provider that accepts INVITE requests from non-Registered Contacts.  No?  Or do you mean you've done testing of all publicly reachable SIP providers and found this to be an issue for all of them, right now?

I am not sure I understand how accepting/not-accepting INVITEs from
non-registered contacts makes it different, could you elaborate?

Roughly, as long as the attacker can establish a call with the victim and make
him/her respond to 407 within the session (for example by forcing the UA to
generate a re-INVITE), the attacker can get access to the victims
credentials. 

In some cases (if the victim UA is only reachable through a proxy
and the proxy removes its own credentials), the attacker would not be able to
get the credentials for that proxy, but might be able to get credentials for
another domain if the victim's UA is configured with multiple sets of
credentials.

I don't want to put words in Raphael's mouth, but he probably mentioned
publicly reachable SIP providers because they typically place no restrictions
on incoming calls for their users and thus this kind of attack can be easily
tried with them.

> And this is not to say the attack isn't interesting, or that publicly
> reachable SIP providers actually *use* such security policies.  I've been
> surprised before about how some of them don't actually employ the security
> mechanisms they have at their disposal.  But not using available mechanisms
> is very different from not having a way to do them.

Yes, I agree, in this case it would be interesting to know how many of them
actually remove used digest credentials before they forward SIP requests.

  Jan.

From HKaplan@acmepacket.com  Sat Mar  7 11:51:28 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Jan Janak <jan@ryngle.com>, Theo Zourzouvillys <theo@crazygreek.co.uk>
Date: Sat, 7 Mar 2009 14:51:54 -0500
Thread-Topic: [Sip] draft-state-sip-relay-attack-00
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Subject: Re: [Sip] draft-state-sip-relay-attack-00
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Another question is whether an attacker can create special nonce's for one =
or more challenges it sends, that will help it figure out the password.  Fo=
r example can they help a rainbow-table type of password cracking, and how =
much the cnonce can prevent that.

-hadriel

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Jan
> Janak
> Sent: Saturday, March 07, 2009 2:09 PM
>
> On 05-03 14:31, Theo Zourzouvillys wrote:
> > However, you don't cover the more interesting cases of multi-hop proxy
> > authentication or end-to-end WWW authentication: these are the harder
> > ones to deal with, and may result in some "real" issues in SIP itself
> > rather than shoddy implementations and insecure proxies.
>
> Yes, after reading the discussion here I agree, perhaps the next revision
> of
> the ID (if there is going to be any) should describe more difficult cases=
,
> such as multi-hop authentication and challenging proxies reachable throug=
h
> one
> common proxy. These are much harder cases to deal with.
>
>    Jan.
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

From victor.pascual.avila@gmail.com  Sat Mar  7 12:44:03 2009
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From: =?UTF-8?Q?Victor_Pascual_=C3=81vila?= <victor.pascual.avila@gmail.com>
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Subject: Re: [Sip] I-D Inaction: draft-kaplan-sip-secure-call-id-00.txt
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On Fri, Mar 6, 2009 at 12:44 AM, Hadriel Kaplan <HKaplan@acmepacket.com> wr=
ote:
>
>
>> -----Original Message-----
>> From: Elwell, John [mailto:john.elwell@siemens.com]
>> Sent: Thursday, March 05, 2009 12:15 PM
>>
>> Should we specify something that would make life easier for
>> proxies/B2BUAs to detect compliance with this, e.g., a magic string at
>> the start of the call-ID, rather than simply the absence of "@"?
>
> At one point I was thinking of just making it either be an "invalid" host=
:
> Call-ID: randomstuffheregibberish@invalid
>
> Or a fixed-length hex-ascii type thing (like Session-ID). =C2=A0But then =
someone pointed out to me that some devices already don't put their IP/host=
 in it, so there's no reason to go make them change their implementation, n=
or replace their call-id until they do.
>
> The thing is I think that a proxy/b2bua which cares about this for securi=
ty properties, would need/want to scan the whole call-id for something whic=
h looks like a host/IP anyway, to make a dynamic decision. =C2=A0So having =
a magic string won't help them. =C2=A0And B2BUA's which change it for non-s=
ecurity reasons will change it no matter what, because it's not the privacy=
 property they care about.

IMO the UAS shall be able to know whether or not the call-id is
end-to-end (e.g. INVITE-Initiated Dialog Event Package: a B2BUA can't
fix the call-id if the return-routed subscribe request gets routed
along a different path).
--=20
Victor Pascual =C3=81vila

From HKaplan@acmepacket.com  Sat Mar  7 12:45:31 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Jan Janak <jan@ryngle.com>
Date: Sat, 7 Mar 2009 15:45:57 -0500
Thread-Topic: [Sip] draft-state-sip-relay-attack-00
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> -----Original Message-----
> From: Jan Janak [mailto:jan@ryngle.com]
> Sent: Saturday, March 07, 2009 2:51 PM
>
> I am not sure I understand how accepting/not-accepting INVITEs from
> non-registered contacts makes it different, could you elaborate?

Assume Bob is bad, Alice is the victim.
The setup for the attack is such that Bob sends an INVITE to/through Alice'=
s domain, pretending to be Alice.  Alice's domain challenges the INVITE, wh=
ich Bob passes on to Alice, and using her challenge-response Bob challenge-=
responds to Alice's domain.  Right?

I am arguing that in common practice (in my particular market space, anyway=
), Alice's domain wouldn't accept Bob's spoofed INVITE to begin with.  Beca=
use it requires Bob's UA to actually be Registered as Alice in order to sen=
d in an INVITE pretending to be Alice.  Since REGISTER requests are challen=
ged, and the digest-challenge mechanism includes the Method name in the cal=
culation, Bob can't try to REGISTER into Alice's domain and use a challenge=
-response from her INVITE to do so.  Ergo, no problem.

Even for cases where they accept INVITE's from non-registering UA's (e.g, b=
ig IP-PBX SIP Trunks), my particular customer types often restrict what AoR=
 can be claimed or allowed to make calls in from that trunk.

On the peering side, it's a different story - but the model they currently =
employ is a chain-of-trust, so they trust the peer to authenticate and cont=
rol things in a similar fashion.  Whether that's a good idea or not is a di=
fferent story, but that's the current model.  There are protection mechanis=
ms available to prevent or at least mitigate this particular issue for peer=
ing, but I have less confidence that people use them in practice.


> I don't want to put words in Raphael's mouth, but he probably mentioned
> publicly reachable SIP providers because they typically place no
> restrictions
> on incoming calls for their users and thus this kind of attack can be
> easily
> tried with them.

Not in my experience.  They place *many* restrictions on incoming calls.  B=
ut then those are my customers, which are just a subset, and obviously my v=
iew is undoubtedly myopic because those are the types of providers that wou=
ld be talking to me to begin with.  And again, whether they actually use th=
e tools at hand I don't know.

-hadriel

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To: =?iso-8859-1?Q?Victor_Pascual_=C1vila?= <victor.pascual.avila@gmail.com>
Date: Sat, 7 Mar 2009 16:08:35 -0500
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> -----Original Message-----
> From: Victor Pascual =C1vila [mailto:victor.pascual.avila@gmail.com]
> Sent: Saturday, March 07, 2009 3:45 PM
>
> IMO the UAS shall be able to know whether or not the call-id is
> end-to-end (e.g. INVITE-Initiated Dialog Event Package: a B2BUA can't
> fix the call-id if the return-routed subscribe request gets routed
> along a different path).

Are you suggesting that be added to this draft's goals, or just that we sho=
uld tackle that problem (separately)?

I personally don't think it should be a goal of this draft.  There's a can =
of worms in that goal, imho.  This draft is short and to the point, address=
ing one specific issue and solution.  My hope is to get it fast-tracked if =
people agree with it, or I'll change it to a BCP recommendation as an indiv=
idual draft if people don't care.

-hadriel

From victor.pascual.avila@gmail.com  Sat Mar  7 13:14:57 2009
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Date: Sat, 7 Mar 2009 22:15:27 +0100
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From: =?UTF-8?Q?Victor_Pascual_=C3=81vila?= <victor.pascual.avila@gmail.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>
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Cc: "sip@ietf.org" <sip@ietf.org>
Subject: Re: [Sip] draft-state-sip-relay-attack-00
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On Sat, Mar 7, 2009 at 9:45 PM, Hadriel Kaplan <HKaplan@acmepacket.com> wro=
te:
>
>
>> -----Original Message-----
>> From: Jan Janak [mailto:jan@ryngle.com]
>> Sent: Saturday, March 07, 2009 2:51 PM
>>
>> I am not sure I understand how accepting/not-accepting INVITEs from
>> non-registered contacts makes it different, could you elaborate?
>
> Assume Bob is bad, Alice is the victim.
> The setup for the attack is such that Bob sends an INVITE to/through Alic=
e's domain, pretending to be Alice. =C2=A0Alice's domain challenges the INV=
ITE, which Bob passes on to Alice, and using her challenge-response Bob cha=
llenge-responds to Alice's domain. =C2=A0Right?
>
> I am arguing that in common practice (in my particular market space, anyw=
ay), Alice's domain wouldn't accept Bob's spoofed INVITE to begin with. =C2=
=A0Because it requires Bob's UA to actually be Registered as Alice in order=
 to send in an INVITE pretending to be Alice.

As an example: what would proxy.com check upon receipt of Bob's
spoofed INVITE over UDP (forging packet source ip and port)?

Thanks!
--=20
Victor Pascual =C3=81vila

From theo@crazygreek.co.uk  Sat Mar  7 13:25:39 2009
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From: Theo Zourzouvillys <theo@crazygreek.co.uk>
To: Hadriel Kaplan <HKaplan@acmepacket.com>
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Subject: Re: [Sip] draft-state-sip-relay-attack-00
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On Sat, Mar 7, 2009 at 8:45 PM, Hadriel Kaplan <HKaplan@acmepacket.com> wro=
te:

> Not in my experience. =A0They place *many* restrictions on incoming calls=
.

>From what i see (at least in the UK), it seems to depend on if the
provider is selling "PSTN termination" or "SIP services".  The former
tend to place many restrictions (make calls only if registered and
from the IP that registered, only allow a single binding in AOR, etc),
and the later tend not to (although may provide customers with the
ability to).  In the UK, I'm seeing a slow shift from the former to
the later, although no idea about the rest of the world.

hmm, the above paragraph could almost have been talking about
AOL/compuserv vs a "real" ISP 15 years ago :-)

 ~ Theo

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Subject: Re: [Sip] I-D Inaction: draft-kaplan-sip-secure-call-id-00.txt
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On Sat, Mar 7, 2009 at 10:08 PM, Hadriel Kaplan <HKaplan@acmepacket.com> wr=
ote:
>
>
>> -----Original Message-----
>> From: Victor Pascual =C3=81vila [mailto:victor.pascual.avila@gmail.com]
>> Sent: Saturday, March 07, 2009 3:45 PM
>>
>> IMO the UAS shall be able to know whether or not the call-id is
>> end-to-end (e.g. INVITE-Initiated Dialog Event Package: a B2BUA can't
>> fix the call-id if the return-routed subscribe request gets routed
>> along a different path).
>
> Are you suggesting that be added to this draft's goals, or just that we s=
hould tackle that problem (separately)?

I was just wondering if a magic cookie could help the UAS to identify
end-to-end identifiers.
--=20
Victor Pascual =C3=81vila

From HKaplan@acmepacket.com  Sat Mar  7 14:02:06 2009
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To: =?iso-8859-1?Q?Victor_Pascual_=C1vila?= <victor.pascual.avila@gmail.com>
Date: Sat, 7 Mar 2009 17:02:32 -0500
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> -----Original Message-----
> From: Victor Pascual =C1vila [mailto:victor.pascual.avila@gmail.com]
> Sent: Saturday, March 07, 2009 4:15 PM
>
> As an example: what would proxy.com check upon receipt of Bob's
> spoofed INVITE over UDP (forging packet source ip and port)?

Well that is a trade secret, and specific to the proxy/SBC vendor being use=
d, and probably its configuration. :)

But anyway if Bob could guess Alice's source UDP IP and port (which by the =
way is not always trivial to do), then it changes the form and scope of the=
 attack.  The interesting characteristics of a relay-attack, baiting-attack=
, etc., are that they do not require IP-layer spoofing.

Typically, though, spoofing enough to get through would be less useful for =
Bob to do unless he's actually a true MitM, because the SIP responses for t=
he call will go to Alice, so Bob won't be able to pass along the challenge =
and won't be able to establish a call.  And even if Alice's UA is broken en=
ough to answer a challenge for a request it didn't send, Bob would not get =
the SIP responses to setup the call fully, nor send or receive media for it=
.  Bob could try to mount a resource-exhaustion attack on the provider, but=
 that too is throttled (and besides Bob won't be able to get Alice to chall=
enge-response fast enough for each INVITE).  Bob could try to perform an an=
noyance attack by making phones ring, if he figures out how to avoid gettin=
g Alice blocked automatically, but there are even countermeasures for that =
available.  Bob could try an attack for the purpose of getting Alice blocke=
d from getting service for a period of time, but if Bob knows Alice's UDP s=
ource info he can do that directly on Alice anyway, and the best the provid=
er can do is stop it from affecting others.

-hadriel

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To: =?iso-8859-1?Q?Victor_Pascual_=C1vila?= <victor.pascual.avila@gmail.com>
Date: Sat, 7 Mar 2009 17:06:27 -0500
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Subject: Re: [Sip] I-D Inaction: draft-kaplan-sip-secure-call-id-00.txt
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> -----Original Message-----
> From: Victor Pascual =C1vila [mailto:victor.pascual.avila@gmail.com]
> Sent: Saturday, March 07, 2009 4:27 PM
>
> I was just wondering if a magic cookie could help the UAS to identify
> end-to-end identifiers.

Ahh, I see.  Nope, it won't, I don't think.  If a B2BUA changes the Call-ID=
 for reasons other than the security aspects, it would/should still create =
a secure-call-id format - simply so that other B2BUA's along the path don't=
 feel compelled to as well and exacerbate the problem.  I should make that =
clear in the draft.

-hadriel

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Theo Zourzouvillys <theo@crazygreek.co.uk>
Date: Sat, 7 Mar 2009 17:39:59 -0500
Thread-Topic: [Sip] draft-state-sip-relay-attack-00
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Cc: "sip@ietf.org" <sip@ietf.org>, Dan Wing <dwing@cisco.com>
Subject: Re: [Sip] draft-state-sip-relay-attack-00
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> -----Original Message-----
> From: Theo Zourzouvillys [mailto:theo@crazygreek.co.uk]
> Sent: Saturday, March 07, 2009 4:26 PM
>
> hmm, the above paragraph could almost have been talking about
> AOL/compuserv vs a "real" ISP 15 years ago :-)

Indeed. :)
And email has been getting less spam and phishing and viruses ever since.  =
And email has an even stronger architecture for security than SIP in some w=
ays, ironically.  And less impact on the user-experience when it fails.  In=
 many ways SIP's user-experience model is closer to IM than email, but with=
 some worse security properties.  (and it's debatable if one would call IM =
"open", or end-to-end)

But even in that context of closed vs. open, you find web hosts having to e=
mploy strong and stronger means of user authentication, such as captcha's, =
and connection-layer security, such as TLS.  If the open SIP providers don'=
t employ some counter-measures for spoofing, user authentication, and servi=
ce control, then they will when the issues crop up. (if they have enough us=
ers to make attacking them interesting, which is the big "if")

But anyway, that's why I want a SIP Identity mechanism that actually works,=
 fwiw.

-hadriel

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Subject: Re: [Sip] draft-state-sip-relay-attack-00
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Am 07.03.2009 20:18 Uhr, schrieb Hadriel Kaplan:
>> So a requirement to make the attack possible is that the user agent
>> responds
>> to challenges generated for in-dialog requests.
>
> Right, and that the attacked domain accepts INVITEs from its AoR's with non-registered Contacts; or accepts INVITEs from its static AoR's to come in from unknown locations.  That's pretty rare in my world, but ymmv.

Luckily it seems we are not living in the same world :-)
I call it a feature that I can make authenticated calls without being 
registered.

[...]

>>> - I never unterstood why a proxy should pass through the authentication
>>> request from a foreign domain.
>> Because this is how it is specified in section 22.3 of RFC3261.
>
> And it would have to continue to do so.  There are actual use-cases for this.

Could you please share one of these use-cases with me.

> I think there's even a reasonable use-case for challenging in-dialog requests: connected-identity, for example.
>
> But you don't even need to challenge in-dialog requests for this form of attack: if the victim calls you, then you can challenge the initial INVITE.

Sorry, but how is this going to work in world without a SBC which knows 
my credentials?
Remember my proxy can not answer the challenge (CSeq mis-match). And the 
caller hopefully does not know my credentials, otherwise the whole 
attack would be pointless.

Cheers
   Nils

From theo@crazygreek.co.uk  Sat Mar  7 15:22:54 2009
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From: Theo Zourzouvillys <theo@crazygreek.co.uk>
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Subject: Re: [Sip] draft-state-sip-relay-attack-00
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On Sat, Mar 7, 2009 at 10:39 PM, Hadriel Kaplan <HKaplan@acmepacket.com> wr=
ote:

> And email has been getting less spam and phishing and viruses ever since.

SIP also has one major advantage than email does: we've not *yet* got
the same penetration that SMTP did when it started really suffering.
Although I do see exactly the same apathy to the problem in voice
service providers and network operators now that I did in ISPs 10
years ago: "not a problem yet, can't afford to spend time on that".

>=A0If the open SIP providers don't employ some counter-measures for spoofi=
ng, user authentication, and service control, then they will when the issue=
s crop up.

It's difficult to do *that* much: a lot of the counter-measures need
support from the endpoint vendors for various features.  A look at the
quality (or even existence) of mutual TLS and related things in almost
all UAs speaks loads on their interest in such matters!

> But anyway, that's why I want a SIP Identity mechanism that actually work=
s, fwiw.

i'll give 10 pounds (of the Great British variety) and a bottle of
sheppy's finest Somerset cider to whoever comes up with such a thing.

 ~ Theo

From dean.willis@softarmor.com  Sat Mar  7 15:30:01 2009
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Subject: Re: [Sip] draft-ietf-sip-info-events-03: comments and questions
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Bob said:
I'm inclined to say that since the INFO is not suppose to change
> dialog state, a 489 response to an INFO should not destroy the INVITE
> dialog usage. With SUBSCRIBE/NOTIFY its a failure of the dialog
> establishing/target-refresh request which is essential to the dialog
> usage. But INFO is not essential to the INVITE dialog usage, so
> applying the last sentence above makes sense to me.

Doesn't a 489 in any in-dialog Non-Invite-Transaction destroy the  
dialog usage? It seems like we break the state machine if we make INFO  
a special case.

--
Dean


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From: "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>
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Subject: Re: [Sip] draft-ietf-sip-info-events-03: comments and questions
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From HKaplan@acmepacket.com  Sat Mar  7 16:30:10 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Nils Ohlmeier <lists@ohlmeier.org>
Date: Sat, 7 Mar 2009 19:30:30 -0500
Thread-Topic: [Sip] draft-state-sip-relay-attack-00
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Subject: Re: [Sip] draft-state-sip-relay-attack-00
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> -----Original Message-----
> From: Nils Ohlmeier [mailto:lists@ohlmeier.org]
> Sent: Saturday, March 07, 2009 5:41 PM
>
> Am 07.03.2009 20:18 Uhr, schrieb Hadriel Kaplan:
> >> So a requirement to make the attack possible is that the user agent
> >> responds
> >> to challenges generated for in-dialog requests.
> >
> > Right, and that the attacked domain accepts INVITEs from its AoR's with
> non-registered Contacts; or accepts INVITEs from its static AoR's to come
> in from unknown locations.  That's pretty rare in my world, but ymmv.
>
> Luckily it seems we are not living in the same world :-)
> I call it a feature that I can make authenticated calls without being
> registered.

Why, is sending a REGISTER hard??  It looks like a simple enough message to=
 generate.  How about OPTIONS?
(has anyone ever noticed that OPTIONS is an anagram for POTIONS, or "O, NOT=
 SIP" or "ON TO SIP" or "TON-O-SIP" or "TIS NO-OP" or "I NO POTS"? weird)

But anyway, sure - I'm not saying the attack isn't interesting or worth fig=
uring out a solution/prevention for.  I think it *is*.  I was only challeng=
ing a claim that Service Providers on the Internet are currently exposed to=
 it.  Not all of them are.


> >>> - I never unterstood why a proxy should pass through the
> authentication
> >>> request from a foreign domain.
> >> Because this is how it is specified in section 22.3 of RFC3261.
> >
> > And it would have to continue to do so.  There are actual use-cases for
> this.
>
> Could you please share one of these use-cases with me.

Well, the 3GPP roaming case, for one.
A pictorial version: http://www.tech-invite.com/Ti-ims-regflow-1.html


> > I think there's even a reasonable use-case for challenging in-dialog
> requests: connected-identity, for example.
> >
> > But you don't even need to challenge in-dialog requests for this form o=
f
> attack: if the victim calls you, then you can challenge the initial
> INVITE.
>
> Sorry, but how is this going to work in world without a SBC which knows
> my credentials?

SBC's don't usually know your credentials, even now.  They don't need to.

But anyway, I'm not sure what you mean by the question.  How is what going =
to work?  Stopping INVITE-based authentication relay-attacks?  You don't ne=
ed an SBC-type box to stop that.  Just disconnect the cable.  :)
Or, use the counter-measures in the draft.  Or change the protocol, or at l=
east the authentication mechanism.

-hadriel

From HKaplan@acmepacket.com  Sat Mar  7 20:24:57 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, Nils Ohlmeier <lists@ohlmeier.org>
Date: Sat, 7 Mar 2009 23:25:20 -0500
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Hadriel Kaplan
> Sent: Saturday, March 07, 2009 7:31 PM
>
> But anyway, I'm not sure what you mean by the question.  How is what goin=
g
> to work?  Stopping INVITE-based authentication relay-attacks?  You don't
> need an SBC-type box to stop that.  Just disconnect the cable.  :)
> Or, use the counter-measures in the draft.  Or change the protocol, or at
> least the authentication mechanism.

BTW, you don't need to be an SBC to have such policies.  There is nothing i=
n RFC 3261 which prohibits a pure Proxy from restricting who it accepts INV=
ITEs from, and when.  There are even some clever tricks one can do to make =
it a stateless mechanism.

-hadriel

From HKaplan@acmepacket.com  Sun Mar  8 12:43:50 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "sip@ietf.org" <sip@ietf.org>
Date: Sun, 8 Mar 2009 15:44:02 -0400
Thread-Topic: I-D Action:draft-kaplan-sip-session-id-02.txt 
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Howdy,
A new version of the session-id draft has been uploaded, removing the match=
ing properties from the requirements and solution, in the hopes of progress=
ing it faster.
The purpose of it is now solely for troubleshooting across SIP nodes.

-hadriel

> -----Original Message-----
> From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org=
]
> On Behalf Of Internet-Drafts@ietf.org
> Sent: Sunday, March 08, 2009 3:30 PM
> To: i-d-announce@ietf.org
> Subject: I-D Action:draft-kaplan-sip-session-id-02.txt
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>
>         Title           : A Session Identifier for the Session Initiation
> Protocol (SIP)
>         Author(s)       : H. Kaplan
>         Filename        : draft-kaplan-sip-session-id-02.txt
>         Pages           : 11
>         Date            : 2009-03-08
>
> There are several reasons for having a globally unique session
> identifier for the same SIP session, which can be maintained across
> B2BUA's and other SIP middle-boxes.  This draft proposes a new SIP
> header to carry such a value: Session-ID.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-kaplan-sip-session-id-02.txt
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "sip@ietf.org" <sip@ietf.org>
Date: Sun, 8 Mar 2009 15:52:23 -0400
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Howdy,
An updated version of SIP Asserter Identity has been uploaded, though with =
virtually no changes other than to avoid expiration.
But since the topic of SIP Identity is coming up (again) at the upcoming me=
eting, and this draft is related, I thought I'd post this again.

There was not much feedback to the previous rev, so I'm guessing folks eith=
er don't like it, or don't care, or just haven't had the time to read it.

As always, comments are appreciated - including just saying "don't care".

-hadriel

> -----Original Message-----
> From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org=
]
> On Behalf Of Internet-Drafts@ietf.org
> Sent: Sunday, March 08, 2009 2:30 PM
> To: i-d-announce@ietf.org
> Subject: I-D Action:draft-kaplan-sip-asserter-identity-01.txt
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>
>         Title           : Private Extensions to the Session Initiation
> Protocol (SIP) for Asserter Identification within Trusted Networks
>         Author(s)       : H. Kaplan
>         Filename        : draft-kaplan-sip-asserter-identity-01.txt
>         Pages           : 17
>         Date            : 2009-03-08
>
> This document describes private extensions to the Session
> Initiation Protocol (SIP) that enable a network of trusted SIP
> servers to identify the asserter of private user identity defined
> in RFC 3325.  The use of these extensions is only applicable
> inside a set of administrative domains with previously agreed-upon
> policies for generation, transport and usage of such information.
> This document does NOT offer a general identity model suitable for
> use between different trust domains, or use in the Internet at
> large.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-kaplan-sip-asserter-identity-
> 01.txt
>
> Internet-Drafts are also available by anonymous FTP at:
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>
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From brett@broadsoft.com  Sun Mar  8 14:31:03 2009
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From: Brett Tate <brett@broadsoft.com>
To: Eric Burger <eburger@standardstrack.com>, SIP IETF <sip@ietf.org>
Date: Sun, 8 Mar 2009 14:31:32 -0700
Thread-Topic: [Sip] draft-ietf-sip-info-events-03: comments and questions
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Thanks for the response; reply inline.

> > Section 3.1 paragraph 1:  Concerning "MUST advertise" and
> > "provisional 1xx", please clarify that the "MUST advertise"=20
> > is not applicable to the 100 response.  Does the=20
> > "MUST advertise" and "provisional 1xx" apply to all (first=20
> > and subsequent within dialog) INVITE non 101-199=20
> > (intentionally indicated 199)?  Concerning ACK for initial=20
> > INVITE's 2xx, does the "MUST advertise" always apply?
>=20
> I would offer "MUST advertise" is appropriate. What is the=20
> rationale to not require it? We are expecting "early INFO,"=20
> so no reason to screw up UACs by having random appearances=20
> and disappearances of Recv-Info headers.

A 100 response does not create a dialog; thus including it during setup has=
 no meaning or purpose.  A proxy can send 100 responses within dialog; howe=
ver the lack of Recv-Info appears to disable draft support per the followin=
g draft snippet.  Thus I prefer the text cover 101-299 Reponses instead of =
100-299 responses.

Section 3.1: "if the UA neglects to include the Recv-Info header, the other=
 UA in the dial will assume the first UA no longer supports INFO as specifi=
ed by this document."

Nit: "dial" should be "dialog".


> > Section 3.1 paragraph 2 sentences 1 and 2: Sentence 1=20
> > indicates "session parameters" and sentence 2 indicates=20
> > "dialog parameters" concerning Info Package negotiation,=20
> > I'm not sure when the negotiation/renegotiation occurs. =20
> > Does it follow SDP offer/answer rules or something else;=20
> > if something else, what?  I have the same question=20
> > concerning "dialog refresh" text discussed later within=20
> > the same section since incorrect interpretations might=20
> > trigger deactivation.
>=20
> Should be dialog parameters everywhere.  Would that clear=20
> up the confusion?

Partially; when are dialog parameters renegotiated?  My main concern is wit=
h impacts of "neglects to include the Recv-Info" as reflected within above =
section 3.1 snippet; thus the reason I'm trying to understand.

To avoid disabling, Recv-Info must be within re-INVITE, UPDATE, ACK-for-2xx=
, and 101-299 responses for these requests.  Is Recv-Info mandatory within =
PRACK and PRACK 2xx to avoid disabling?  Is it mandatory within REFER and R=
EFER 2xx to avoid disabling (REFER isn't listed within section 5.2)?


> > Section 4.3: When sending a 489, it indicates "MUST=20
> > terminate the INVITE dialog".  The "MUST terminate"=20
> > seems severe even if all the negotiation race conditions=20
> > are adequately addressed.  The mentioned "protocol failure"=20
> > reason seems contrary to the "strictly send and lenient=20
> > receive" mantra.
<snip>
> Let us look at this from a practical perspective. If you are=20
> a bad implementor and send a 489 you didn't mean to, who=20
> knows what else you will get wrong. Conversely, if you
> are a bad implementor and forget to tear down a 489-ed INFO,=20
> bad things probably will not happen.

Consider ACK from caller glaring with INFO from called.  If the ACK adjuste=
d Recv-Info to no longer support foo and INFO contained foo, why MUST the c=
aller terminate the dialog upon sending 489 instead of realizing that it wa=
s just a glare?  I actually think a good implementer would ignore the MUST =
and choose to be non compliant to avoid customer complaints; however I'd ra=
ther the draft be changed instead.

Because updated Recv-Info can be within so many requests and responses with=
in a dialog, there can be glare and ordering issues.  Adding to the issue (=
change and change back), is that the Recv-Info is required to be within so =
many requests and responses to avoid disabling.


From gao.yang2@zte.com.cn  Sun Mar  8 19:49:30 2009
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Subject: [Sip] "UPDATE during Re-INVITE" discussion
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There are two thought about "UPDATE during Re-INVITE". Which one is better 
or suits for current RFCs

There are two thought about "UPDATE during Re-INVITE".

1. All UPDATEs during Re-INVITE are Re-INVITE's sub-transaction
RFC3312(Precondition) is a case. As there is such case, we can making all 
UPDATEs during Re-INVITE as Re-INVITE's sub-transaction.

2. Not all UPDATE(during Re-INVITE) can be considered as sub-transaction 
of Re-INVITE
There is really need nested-modification, such as precondition and more in 
the future.  But the cascade of nested transaction should be defined in 
application-level, not signal-level. So, it is better to make Re-INVITE 
and UPDATE separatly expect for definition such as precondition.  And this 
obeys current definition of RFC3311.

In RFC3311, when the UPDATE is accepted by the other side(UPDATE/200OK), 
the change of states is committed and effort at once. And making all 
UPDATEs during Re-INVITE as Re-INVITE's sub-transaction can be violation 
of RFC3311. So, we shold not treat all UPDATEs during Re-INVITE as 
Re-INVITE's sub-transaction.

While UPDATE/200OK just refresh "precondition state table" in 
RFC3312("precondition state table" is modified at once, so this obeys 
RFC3312), having no impat on the commitment of modification triggered by 
Re-INVITE, these UPDATE/200OK can be treated as sub-transaction of 
Re-INVITE.
Other UPDATE/200OK can not be treated as sub-transaction of Re-INVITE.

Comments/feeling are welcome!


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<br><font size=2 face="sans-serif">There are two thought about &quot;UPDATE
during Re-INVITE&quot;. Which one is better or suits for current RFCs</font>
<br>
<br><font size=2 face="sans-serif"><b>There are two thought about &quot;UPDATE
during Re-INVITE&quot;.</b></font>
<br>
<br><font size=2 face="sans-serif"><b>1. All UPDATEs during Re-INVITE are
Re-INVITE's sub-transaction</b></font>
<br><font size=2 face="sans-serif">RFC3312(Precondition) is a case. As
there is such case, we can making all UPDATEs during Re-INVITE as Re-INVITE's
sub-transaction.</font>
<br>
<br><font size=2 face="sans-serif"><b>2. Not all UPDATE(during Re-INVITE)
can be considered as sub-transaction of Re-INVITE</b></font>
<br><font size=2 face="sans-serif">There is really need nested-modification,
such as precondition and more in the future. &nbsp;But the cascade of nested
transaction should be defined in application-level, not signal-level. <b>So,
it is better to make Re-INVITE and UPDATE separatly expect for definition
such as precondition.</b> &nbsp;<b>And this obeys current definition of
RFC3311.</b></font>
<br>
<br><font size=2 face="sans-serif">In RFC3311, when the UPDATE is accepted
by the other side(UPDATE/200OK), the change of states is committed and
effort at once. And making all UPDATEs during Re-INVITE as Re-INVITE's
sub-transaction can be violation of RFC3311. So, we shold not treat all
UPDATEs during Re-INVITE as Re-INVITE's sub-transaction.</font>
<br>
<br><font size=2 face="sans-serif">While UPDATE/200OK just refresh &quot;precondition
state table&quot; in RFC3312(&quot;precondition state table&quot; is modified
at once, so this obeys RFC3312), having no impat on the commitment of modification
triggered by Re-INVITE, these UPDATE/200OK can be treated as sub-transaction
of Re-INVITE.</font>
<br><font size=2 face="sans-serif">Other UPDATE/200OK can not be treated
as sub-transaction of Re-INVITE.</font>
<br>
<br><font size=2 face="sans-serif"><b>Comments/feeling are welcome!</b></font>
<br><br><pre>
--------------------------------------------------------
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From rco@iptel.org  Mon Mar  9 04:41:03 2009
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From: Raphael Coeffic <rco@iptel.org>
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Subject: Re: [Sip] draft-state-sip-relay-attack-00
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Hadriel Kaplan wrote:
>   
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
>> Hadriel Kaplan
>> Sent: Saturday, March 07, 2009 7:31 PM
>>
>> But anyway, I'm not sure what you mean by the question.  How is what going
>> to work?  Stopping INVITE-based authentication relay-attacks?  You don't
>> need an SBC-type box to stop that.  Just disconnect the cable.  :)
>> Or, use the counter-measures in the draft.  Or change the protocol, or at
>> least the authentication mechanism.
>>     
>
> BTW, you don't need to be an SBC to have such policies.  There is nothing in RFC 3261 which prohibits a pure Proxy from restricting who it accepts INVITEs from, and when.  There are even some clever tricks one can do to make it a stateless mechanism.
>
>   
That's a good point. Requiring the user to be registered and only 
accepting requests from the registered contacts provide a fair-enough 
level of security concerning the attack debated. But I am still hoping 
that we could find a solution not requiring this kind of measures, which 
I would call "user-restricting".

-Raphael.

From eric.wangxr@gmail.com  Mon Mar  9 05:05:31 2009
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From: "Eric.wang" <eric.wangxr@gmail.com>
To: <gao.yang2@zte.com.cn>, "'SIP'" <sip@ietf.org>, "'SIPPING'" <sipping@ietf.org>
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Hi,

=20

I support view 2,not all update can be considered as sub-transaction of
re-INVITE.

According to current RFCs, it=A1=AFs NOT proper to consider all UPDATEs =
as
re-INVITE=A1=AFs sub-transaction=20

Eg:

UPDATE nested in re-INVITE  for target refresh can=A1=AFt be considered =
as
sub-transaction.

=20

=B7=A2=BC=FE=C8=CB: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] =
=B4=FA=B1=ED
gao-.yang2@zte.com.cn
=B7=A2=CB=CD=CA=B1=BC=E4: 2009=C4=EA3=D4=C29=C8=D5 10:49
=CA=D5=BC=FE=C8=CB: SIP; SIPPING
=D6=F7=CC=E2: [Sip] "UPDATE during Re-INVITE" discussion

=20


There are two thought about "UPDATE during Re-INVITE". Which one is =
better
or suits for current RFCs=20

There are two thought about "UPDATE during Re-INVITE".=20

1. All UPDATEs during Re-INVITE are Re-INVITE's sub-transaction=20
RFC3312(Precondition) is a case. As there is such case, we can making =
all
UPDATEs during Re-INVITE as Re-INVITE's sub-transaction.=20

2. Not all UPDATE(during Re-INVITE) can be considered as sub-transaction =
of
Re-INVITE=20
There is really need nested-modification, such as precondition and more =
in
the future.  But the cascade of nested transaction should be defined in
application-level, not signal-level. So, it is better to make Re-INVITE =
and
UPDATE separatly expect for definition such as precondition.  And this =
obeys
current definition of RFC3311.=20

In RFC3311, when the UPDATE is accepted by the other side(UPDATE/200OK), =
the
change of states is committed and effort at once. And making all UPDATEs
during Re-INVITE as Re-INVITE's sub-transaction can be violation of =
RFC3311.
So, we shold not treat all UPDATEs during Re-INVITE as Re-INVITE's
sub-transaction.=20

While UPDATE/200OK just refresh "precondition state table" in
RFC3312("precondition state table" is modified at once, so this obeys
RFC3312), having no impat on the commitment of modification triggered by
Re-INVITE, these UPDATE/200OK can be treated as sub-transaction of
Re-INVITE.=20
Other UPDATE/200OK can not be treated as sub-transaction of Re-INVITE.=20

Comments/feeling are welcome!=20

=20
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This email and any files transmitted with it are confidential and =
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If you have received this email in error please notify the originator of =
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This message has been scanned for viruses and Spam by ZTE Anti-Spam =
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=20

__________ Information from ESET NOD32 Antivirus, version of virus =
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=20

The message was checked by ESET NOD32 Antivirus.

=20

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<div class=3DSection1>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi,<o:p></o:p></span></p>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I support view 2,not all update can be considered as =
sub-transaction
of re-INVITE.<o:p></o:p></span></p>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'>According to current RFCs, it=A1=AFs NOT proper to =
consider all
UPDATEs as re-INVITE=A1=AFs sub-transaction <o:p></o:p></span></p>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Eg:<o:p></o:p></span></p>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'>UPDATE nested in re-INVITE &nbsp;for target refresh =
can=A1=AFt be
considered as sub-transaction.<o:p></o:p></span></p>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0cm 0cm 0cm'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:=CB=CE=CC=E5'>=B7=A2=BC=FE=C8=CB<sp=
an
lang=3DEN-US>:</span></span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;
font-family:=CB=CE=CC=E5'> sip-bounces@ietf.org =
[mailto:sip-bounces@ietf.org] </span><b><span
style=3D'font-size:10.0pt;font-family:=CB=CE=CC=E5'>=B4=FA=B1=ED =
</span></b><span lang=3DEN-US
style=3D'font-size:10.0pt;font-family:=CB=CE=CC=E5'>gao-.yang2@zte.com.cn=
<br>
</span><b><span =
style=3D'font-size:10.0pt;font-family:=CB=CE=CC=E5'>=B7=A2=CB=CD=CA=B1=BC=
=E4<span lang=3DEN-US>:</span></span></b><span
lang=3DEN-US style=3D'font-size:10.0pt;font-family:=CB=CE=CC=E5'> =
2009</span><span
style=3D'font-size:10.0pt;font-family:=CB=CE=CC=E5'>=C4=EA<span =
lang=3DEN-US>3</span>=D4=C2<span
lang=3DEN-US>9</span>=C8=D5<span lang=3DEN-US> 10:49<br>
</span><b>=CA=D5=BC=FE=C8=CB<span lang=3DEN-US>:</span></b><span =
lang=3DEN-US> SIP; SIPPING<br>
</span><b>=D6=F7=CC=E2<span lang=3DEN-US>:</span></b><span lang=3DEN-US> =
[Sip] &quot;UPDATE
during Re-INVITE&quot; discussion<o:p></o:p></span></span></p>

</div>

<p class=3DMsoNormal><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
lang=3DEN-US><br>
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>There
are two thought about &quot;UPDATE during Re-INVITE&quot;. Which one is =
better
or suits for current RFCs</span><span lang=3DEN-US> <br>
<br>
</span><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>There
are two thought about &quot;UPDATE during =
Re-INVITE&quot;.</span></b><span
lang=3DEN-US> <br>
<br>
</span><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>1.
All UPDATEs during Re-INVITE are Re-INVITE's =
sub-transaction</span></b><span
lang=3DEN-US> <br>
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>RFC3312(Preco=
ndition)
is a case. As there is such case, we can making all UPDATEs during =
Re-INVITE as
Re-INVITE's sub-transaction.</span><span lang=3DEN-US> <br>
<br>
</span><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>2.
Not all UPDATE(during Re-INVITE) can be considered as sub-transaction of =
Re-INVITE</span></b><span
lang=3DEN-US> <br>
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>There
is really need nested-modification, such as precondition and more in the
future. &nbsp;But the cascade of nested transaction should be defined in
application-level, not signal-level. <b>So, it is better to make =
Re-INVITE and
UPDATE separatly expect for definition such as precondition.</b> =
&nbsp;<b>And
this obeys current definition of RFC3311.</b></span><span lang=3DEN-US> =
<br>
<br>
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>In
RFC3311, when the UPDATE is accepted by the other side(UPDATE/200OK), =
the
change of states is committed and effort at once. And making all UPDATEs =
during
Re-INVITE as Re-INVITE's sub-transaction can be violation of RFC3311. =
So, we
shold not treat all UPDATEs during Re-INVITE as Re-INVITE's =
sub-transaction.</span><span
lang=3DEN-US> <br>
<br>
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>While
UPDATE/200OK just refresh &quot;precondition state table&quot; in
RFC3312(&quot;precondition state table&quot; is modified at once, so =
this obeys
RFC3312), having no impat on the commitment of modification triggered by
Re-INVITE, these UPDATE/200OK can be treated as sub-transaction of =
Re-INVITE.</span><span
lang=3DEN-US> <br>
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Other
UPDATE/200OK can not be treated as sub-transaction of =
Re-INVITE.</span><span
lang=3DEN-US> <br>
<br>
</span><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Comments/feel=
ing
are welcome!</span></b><span lang=3DEN-US> <o:p></o:p></span></p>

<pre><span lang=3DEN-US><o:p>&nbsp;</o:p></span></pre><pre><span =
lang=3DEN-US>--------------------------------------------------------<o:p=
></o:p></span></pre><pre><span
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<p><span lang=3DEN-US>__________ Information from ESET NOD32 Antivirus, =
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of virus signature database 3918 (20090309) =
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Subject: Re: [Sip] draft-ietf-sip-info-events-03: comments and questions
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> SIP has a state machine??? You could fool me....
>
> Where can I find it? Do not say 3261

Search in your favorite SIP stack. Some have it, others not... :-)
Some even have a state per transaction and dialog. But don't get fooled by
the marketing/sales department ;-)

SCNR
  Nils


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	Title           : Connection Reuse in the Session Initiation Protocol (SIP)
	Author(s)       : V. Gurbani, et al.
	Filename        : draft-ietf-sip-connect-reuse-13.txt
	Pages           : 18
	Date            : 2009-03-09

This document enables a pair of communicating proxies to reuse a
congestion-controlled connection between themselves for sending
requests in the forward and backwards direction.  Because the
connection is essentially aliased for requests going in the backwards
direction, reuse is predicated upon both the communicating endpoints
authenticating themselves using X.509 certificates through TLS.  For
this reason, we only consider connection reuse for TLS over TCP and
TLS over SCTP.  This document also provides guidelines on connection
reuse and virtual SIP servers and the interaction of connection reuse
and DNS SRV lookups in SIP.

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From vkg@alcatel-lucent.com  Mon Mar  9 07:53:25 2009
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Date: Mon, 09 Mar 2009 08:54:33 -0500
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Folks: connect-reuse-13 has been released to address AD
comments and move the work forward in IESG.

By far the most pervasive change was to restrict connection
reuse to TLS connections, as we discussed on the list last
week.  We will explore a separate draft for TCP connection reuse
after IETF 74.

The draft is now available in the archives at:

http://www.ietf.org/internet-drafts/draft-ietf-sip-connect-reuse-13.txt

On behalf of the author team, I am hoping that version 13 will prove
to be a charm ;-)

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
WWW:   http://ect.bell-labs.com/who/vkg

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>> >>> - I never unterstood why a proxy should pass through the
>> authentication
>> >>> request from a foreign domain.
>> >> Because this is how it is specified in section 22.3 of RFC3261.
>> >
>> > And it would have to continue to do so.  There are actual use-cases
>> for
>> this.
>>
>> Could you please share one of these use-cases with me.
>
> Well, the 3GPP roaming case, for one.
> A pictorial version: http://www.tech-invite.com/Ti-ims-regflow-1.html

Well you are right, I was just thinking about the "typical"
proxy/registrar combination. But there are outbound or travesering
proxyies which don't do authentication at all.

To stay at your 3GPP romain case: I don't think this attack would be
possible if the S-CSCF does not pass through "unknown" auth requests.
Right?

The P-CSCF has to forward REGISTERs to non-local S-CSCF which then will
authenticate the registration with "unknown"/non-local realms. But as soon
as the UA is registered the P-CSCF should only accept terminating INVITE
requests for that UA from the S-CSCF where the user is registered to. And
again the S-CSCF of the user/UA has in my opinion no need to forward
"foreign" authentication requests, or?!

BTW as you pointed out already earlier debating this attack for IMS is
only hypothetical, because IMS allows calls only from registered UAs.

And from a more general perspective:
I believe that such a transitive proxy which does not do authentication by
himself should only accept auth requests from the serving/registrar proxy
(or however you call the element which does the authentication part).
Furthermore this transitive proxy should not allow dialogs over "just
transitive proxys" (which have now clue about authentication) only
(because that would allow the attacker again to send a challenge).

>> > I think there's even a reasonable use-case for challenging in-dialog
>> requests: connected-identity, for example.
>> >
>> > But you don't even need to challenge in-dialog requests for this form
>> of
>> attack: if the victim calls you, then you can challenge the initial
>> INVITE.
>>
>> Sorry, but how is this going to work in world without a SBC which knows
>> my credentials?
>
> SBC's don't usually know your credentials, even now.  They don't need to.
>
> But anyway, I'm not sure what you mean by the question.  How is what going
> to work?  Stopping INVITE-based authentication relay-attacks?  You don't
> need an SBC-type box to stop that.  Just disconnect the cable.  :)
> Or, use the counter-measures in the draft.  Or change the protocol, or at
> least the authentication mechanism.

My fault. I mis-read your original point.

Cheers
  Nils


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	Title           : Framework for Establishing an SRTP Security Context using DTLS
	Author(s)       : J. Fischl, et al.
	Filename        : draft-ietf-sip-dtls-srtp-framework-07.txt
	Pages           : 38
	Date            : 2009-03-09

This document specifies how to use the Session Initiation Protocol
(SIP) to establish an Secure Real-time Transport Protocol (SRTP)
security context using the Datagram Transport Layer Security (DTLS)
protocol.  It describes a mechanism of transporting a fingerprint
attribute in the Session Description Protocol (SDP) that identifies
the key that will be presented during the DTLS handshake.  The key
exchange travels along the media path as opposed to the signaling
path.  The SIP Identity mechanism can be used to protect the
integrity of the fingerprint attribute from modification by
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	Title		: Location Conveyance for the Session Initiation Protocol
	Author(s)	: J. Polk, B. Rosen
	Filename	: draft-ietf-sip-location-conveyance-13.txt
	Pages		: 46
	Date		: 2009-3-9
	
This document defines an extension to the Session Initiation 
   Protocol (SIP) to convey geographic location information from one 
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   conveyance as well as location-based routing, where SIP servers 
   make routing decisions based on the location of the user agent 
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	Title           : SIP SAML Profile and Binding
	Author(s)       : H. Tschofenig, et al.
	Filename        : draft-ietf-sip-saml-06.txt
	Pages           : 52
	Date            : 2009-03-09

This document specifies a Session Initiation Protocol (SIP) profile
of Security Assertion Markup Language (SAML) as well as a SAML SIP
binding.  The defined SIP SAML Profile composes with the mechanisms
defined in the SIP Identity specification and satisfy requirements
presented in "Trait-based Authorization Requirements for the Session
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Subject: Re: [Sip] draft-state-sip-relay-attack-00
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> -----Original Message-----
> From: Raphael Coeffic [mailto:rco@iptel.org]
> Sent: Monday, March 09, 2009 7:42 AM
>
> That's a good point. Requiring the user to be registered and only
> accepting requests from the registered contacts provide a fair-enough
> level of security concerning the attack debated. But I am still hoping
> that we could find a solution not requiring this kind of measures, which
> I would call "user-restricting".

How is it "user-restricting" to require a UA to generate a REGISTER request=
?  Most humans don't generate the REGISTER request by hand - typically soft=
ware does it for them.  :)
I mean at that point you can also just say "digest authentication is user-r=
estricting".

-hadriel

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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: Re: [Sip] Comments on draft-ietf-sip-body-handling-05
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Hi Eric,

thanks for your comments. We had already agreed to request the 
publication of 05. We were just waiting for the chairs to do or get 
someone to do the PROTO write up for the draft (the PROTO write up has 
already been done at this point).

In any case, I will revise the draft so that the revision we use for the 
publication request (06) will include your comments. My answers inline:

> Good work and almost done!
> 
> General
> =======
> Many of the rules for multipart/mumble say things like "all parts must 
> be foo." This construction is not strictly correct. For example, in 
> multipart/alternative, I could have a multipart/mixed. Saying something 
> like "all parts in a multipart/alternative must be marked foo" means 
> that ALL parts, including walking down the multipart/mixed contained 
> therein, must be marked foo.  What we really want to say is "all 
> top-level parts in a multipart/alternative must be marked foo." That 
> means, for example, if one of the parts is a multipart/mixed, the 
> multipart/mixed needs to have the foo marking, but things within the 
> multipart/mixed, I presume, will follow the appropriate rules for 
> multipart/mixed and the body's context.

Good point. I have fixed it.

> Section 3.2
> ===========
> Please make clear that binary *SIP* message body parts MUST be binary, 
> as the text currently says. However, it is quite possible, and should be 
> legal, to encode application parts transported in SIP, such as INFO or 
> NOTIFY bodies.  For example, one might forward a base64 e-mail or a 
> uuencoded file.  Since that would be an application-level message, there 
> is no reason to burden the application with decoding the body part to 
> get it to binary, just for the other side to re-encode it.

I have added the following:

"The only
case where a UA MAY use a different encoding is when transferring
application data between applications that only handle a different
encoding (e.g., base64)."


> Section 6.2
> ===========
> What confused me is section 6.2 starts with what looks like a normative 
> statement, but it really is a forward reference. To me, there is no 
> reference to which 'same' we are referring to.  The text then seems to 
> jump to what looks like a special case for 'session' and 
> 'early-session'.  I would suggest the following:
> 
> 
> 6.2.  UA Behavior to Generate 'multipart/alternative' Message Bodies
> 
>   Section 8.2 mandates all top-level body parts within a 
> 'multipart/alternative' have the same disposition type as the enclosing 
> 'multipart/alternative'.
> 
>    The 'session' and 'early-session' [RFC3959] disposition types require 
> that all body parts of a 'multipart/alternative' body have different 
> content types.   Consequently, for the 'session' and 'early-session' 
> disposition  types, each top-level body type within the 
> 'multipart/alternative' body MUST be different.  Otherwise, the rules 
> for the specific disposition type apply.
>       NOTE: One could envision situations where a UA presents multiple 
> alternatives with the same body type, where the receiving UA picks the 
> best or richest alternative it can use.

I have split the paragraphs as you suggest. I agree it makes the text 
clearer. However, I have stick to using the active voice for normative 
text, which is the trend nowadays (e.g., the UA MUST set the parameters 
to different values instead of the values of the parameters MUST be 
different).


> Section 6.3
> ===========
> Ah, but there ARE handling rules for multipart/alternative.  I would 
> replace this section with:
> 
>    When a UA receives a message with a 'multipart/alternative' body 
> part, if the body part has a 'required' handling, then the UA MUST pick 
> the best alternative body part to use.  If the UA is unable to use any 
> of the body parts, the UA MUST reject the request with a 415 error 
> response.  This could occur if the UA did not understand any of the 
> top-level body parts within the 'multipart/alternative'.
> 
>    The handling marking for top-level body parts in a 
> 'multipart/alternative' has no meaning.  Therefore, the UA MUST ignore 
> any 'handling' tags in the top-level body parts of the 
> 'multipart/alternative' body part.  Once the UA picks one of the 
> top-level body parts to use, then the UA MUST use the appropriate 
> processing rules for the 'handling' tags within the selected top-level 
> body part.

Rules about the handling parameter are given in Section 8.3. Below, you 
suggest to remove Section 8.3 and keep this one instead. The thing is 
that we had discussions on the structure of the document and this is the 
structure we agreed to have, including having sections like 6,3 for 
completeness. At this point, I do not want to reopen that agreement.


> Section 8.2
> ===========
> For multipart/alternative, a top-level body type with 
> 'handling=required' makes no sense.  By definition, using 
> multipart/alternative means we expect the UA to not be able to handle 
> one or more of the body parts - we just do not know which one.  I would 
> remove the note and incorporate the gist into the text:
> 
>    If the sending UA requires the handling of a 'multipart/alternative' 
> body as a whole, the UA MUST set the 'handling' parameter of the 
> 'multipart/alternative' body to 'required'.  Otherwise, the UA MUST set 
> the 'handling' parameter to 'optional'.  The sending UA MUST set all the 
> top-level body parts within the 'multipart/alternative' to 
> 'handling=optional'.
> 

the draft already said this. I have added the clarification about 
"top-level" body parts.

> Section 8.2
> ===========
> For multipart/related, I disagree that if any part is required, the 
> /related is required.

the text talks about the "root body part", not about the 
multipart/related body. The text in the draft already says what you 
propose below.

>  The last sentence is a non-sequitur.  Either I 
> mandate doing something with the 'multipart/related' body or I do not.  
> I could see, for example, saying that processing the multipart/related 
> is optional, but if you do process it, one or more of the sub-parts is 
> required. In this case, the multipart/related is optional, yet one of 
> the sub-parts is required.
> 
> How about:
> 
>    If the sending UA requires the handling of a 'multipart/related' body 
> as a whole, the UA MUST set the 'handling' parameter of the 
> 'multipart/related' body to 'required'.  Otherwise, the UA MUST set the 
> handling to 'optional'.  The UA sets the 'handling' parameters of the 
> body parts within the 'multipart/related' body independently from the 
> 'handling' parameter o the 'multipart/related' body.  If the sending UA 
> requires the handling of a particular body, the UA MUST set the 
> 'handling' parameter of that body part 'required'.  Otherwise, the UA 
> MUST set the parameter to 'optional'.
> 
> 
> 
> Section 8.2
> ===========
> General: I really do not like the construction "If the UA requires the 
> handling of foo the UA MUST mark it with 'handling=required'."  The 
> reason I do not like this is this is precisely what RFC 3459 says to 
> do.  All this draft does is repeat what is in that draft.  Saying "You 
> MUST do what you MUST do" seems redundant and error prone.

the feedback we got is that people found this section useful to clarify 
what to do in each case.

> 
> Section 8.3
> ===========
> This section is redundant with section 6.3.  Remove section 8.3, as 
> section 6.3 talks about processing.

See my comments above about this issue.


> Section 8.4
> ===========
> OK, I'll bite: what if I do not feel like returning a 415?  Either 
> enumerate the situation where I MUST return a 415 or enumerate the 
> situations where I would return something other than 415.

When specifying response codes, we typically use SHOULD because the 
request may have some other problem the UAS needs to report using a 
different error response. Specifying all those cases would be something 
like "go and read RFC 3261" :o)... I think using SHOULD in this context 
is clear enough.


> Section 9.2
> ===========
> Nit in last line of Note: s/constrain/constraint/
> 

Fixed.

> 
> Section 9.4
> ===========
> Since RFC 3459 updated 3204, would not the normative current reference 
> be to 3459?

both references are listed as normative because the draft also 
references the parts of RFC 3204 that deal with signalling transport.


> Section 11
> ==========
> I would offer we point out the potential for DoS attacks from complex 
> MIME bodies.  UA's need to be aware of potential stack depth overflow, 
> line length overflow, pointers to missing content, incorrect 
> content-lengths, correct content-lengths that span what a parser that 
> does not look at content-lengths might consider to be content boundaries 
> with other body parts snuck in.

If you provide me with a reference to a discussion of this attack, I 
will be glad to add it to this section. In any case, I will add it to 
the next revision of the draft, because I intend to submit 06 in a while.

Thanks for your comments,

Gonzalo


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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.


	Title           : Message Body Handling in the Session Initiation Protocol (SIP)
	Author(s)       : G. Camarillo
	Filename        : draft-ietf-sip-body-handling-06.txt
	Pages           : 19
	Date            : 2009-03-09

This document specifies how message bodies are handled in SIP.
Additionally, this document specifies SIP user agent support for MIME
(Multipurpose Internet Mail Extensions) in message bodies.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-body-handling-06.txt

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From christer.holmberg@ericsson.com  Tue Mar 10 00:21:42 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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I support view 1.
=20
Regards,
=20
Christer


________________________________

	From: sipping-bounces@ietf.org [mailto:sipping-bounces@ietf.org] On =
Behalf Of gao.yang2@zte.com.cn
	Sent: 10. maaliskuuta 2009 8:54
	To: Eric.wang
	Cc: 'SIP'; 'SIPPING'
	Subject: [Sipping] =B4=F0=B8=B4: [Sip] "UPDATE during Re-INVITE" =
discussion
=09
=09


	Yes. If we accept view 1, there would be some backward compatible =
problem. And this is a change to RFC3311.=20
=09
	I am waiting for more comments from people want to keep RFC3311 from =
re-write.=20
=09
	Gao=20
=09
=09
	"Eric.wang" <eric.wangxr@gmail.com> =D0=B4=D3=DA 2009-03-09 20:05:51:
=09
	> Hi,=20
	>  =20
	> I support view 2,not all update can be considered as sub-transaction
	> of re-INVITE.=20
	> According to current RFCs, it=A1=AFs NOT proper to consider all =
UPDATEs=20
	> as re-INVITE=A1=AFs sub-transaction=20
	> Eg:=20
	> UPDATE nested in re-INVITE  for target refresh can=A1=AFt be =
considered=20
	> as sub-transaction.=20
=09
=09
	=20
	> There are two thought about "UPDATE during Re-INVITE". Which one is=20
	> better or suits for current RFCs=20
	>=20
	> There are two thought about "UPDATE during Re-INVITE".=20
	>=20
	> 1. All UPDATEs during Re-INVITE are Re-INVITE's sub-transaction=20
	> RFC3312(Precondition) is a case. As there is such case, we can=20
	> making all UPDATEs during Re-INVITE as Re-INVITE's sub-transaction.=20
	>=20
	> 2. Not all UPDATE(during Re-INVITE) can be considered as sub-
	> transaction of Re-INVITE=20
	> There is really need nested-modification, such as precondition and=20
	> more in the future.  But the cascade of nested transaction should be
	> defined in application-level, not signal-level. So, it is better to=20
	> make Re-INVITE and UPDATE separatly expect for definition such as=20
	> precondition.  And this obeys current definition of RFC3311.=20
	>=20
	> In RFC3311, when the UPDATE is accepted by the other=20
	> side(UPDATE/200OK), the change of states is committed and effort at=20
	> once. And making all UPDATEs during Re-INVITE as Re-INVITE's sub-
	> transaction can be violation of RFC3311. So, we shold not treat all=20
	> UPDATEs during Re-INVITE as Re-INVITE's sub-transaction.=20
	>=20
	> While UPDATE/200OK just refresh "precondition state table" in=20
	> RFC3312("precondition state table" is modified at once, so this=20
	> obeys RFC3312), having no impat on the commitment of modification=20
	> triggered by Re-INVITE, these UPDATE/200OK can be treated as sub-
	> transaction of Re-INVITE.=20
	> Other UPDATE/200OK can not be treated as sub-transaction of =
Re-INVITE.=20
	>=20
	> Comments/feeling are welcome!=20
=09
=09
	--------------------------------------------------------
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------_=_NextPart_001_01C9A150.EE5CD0F8
Content-Type: text/html;
	charset="GB2312"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dgb2312">
<META content=3D"MSHTML 6.00.6000.16809" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D508212007-10032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>I support view 1.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D508212007-10032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D508212007-10032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D508212007-10032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D508212007-10032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Christer</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sipping-bounces@ietf.org=20
  [mailto:sipping-bounces@ietf.org] <B>On Behalf Of=20
  </B>gao.yang2@zte.com.cn<BR><B>Sent:</B> 10. maaliskuuta 2009=20
  8:54<BR><B>To:</B> Eric.wang<BR><B>Cc:</B> 'SIP'; =
'SIPPING'<BR><B>Subject:</B>=20
  [Sipping] =B4=F0=B8=B4: [Sip] "UPDATE during Re-INVITE" =
discussion<BR></FONT><BR></DIV>
  <DIV></DIV><BR><BR><FONT face=3Dsans-serif size=3D2>Yes. If we accept =
view 1,=20
  there would be some backward compatible problem. And this is a change =
to=20
  RFC3311.</FONT> <BR><BR><FONT face=3Dsans-serif size=3D2>I am waiting =
for more=20
  comments from people want to keep RFC3311 from re-write.</FONT> =
<BR><BR><FONT=20
  face=3Dsans-serif size=3D2>Gao</FONT> <BR><BR><BR><FONT =
size=3D2><TT>"Eric.wang"=20
  &lt;eric.wangxr@gmail.com&gt; =D0=B4=D3=DA 2009-03-09 =
20:05:51:<BR><BR>&gt;=20
  Hi,</TT></FONT> <BR><FONT size=3D2><TT>&gt; &nbsp;</TT></FONT> =
<BR><FONT=20
  size=3D2><TT>&gt; I support view 2,not all update can be considered as =

  sub-transaction<BR>&gt; of re-INVITE.</TT></FONT> <BR><FONT =
size=3D2><TT>&gt;=20
  According to current RFCs, it=A1=AFs NOT proper to consider all =
UPDATEs <BR>&gt; as=20
  re-INVITE=A1=AFs sub-transaction </TT></FONT><BR><FONT =
size=3D2><TT>&gt;=20
  Eg:</TT></FONT> <BR><FONT size=3D2><TT>&gt; UPDATE nested in re-INVITE =
&nbsp;for=20
  target refresh can=A1=AFt be considered <BR>&gt; as =
sub-transaction.</TT></FONT>=20
  <BR><BR><BR><FONT size=3D2><TT>&nbsp;<BR>&gt; There are two thought =
about=20
  "UPDATE during Re-INVITE". Which one is <BR>&gt; better or suits for =
current=20
  RFCs <BR>&gt; <BR>&gt; There are two thought about "UPDATE during =
Re-INVITE".=20
  <BR>&gt; <BR>&gt; 1. All UPDATEs during Re-INVITE are Re-INVITE's=20
  sub-transaction <BR>&gt; RFC3312(Precondition) is a case. As there is =
such=20
  case, we can <BR>&gt; making all UPDATEs during Re-INVITE as =
Re-INVITE's=20
  sub-transaction. <BR>&gt; <BR>&gt; 2. Not all UPDATE(during Re-INVITE) =
can be=20
  considered as sub-<BR>&gt; transaction of Re-INVITE <BR>&gt; There is =
really=20
  need nested-modification, such as precondition and <BR>&gt; more in =
the=20
  future. &nbsp;But the cascade of nested transaction should be<BR>&gt; =
defined=20
  in application-level, not signal-level. So, it is better to <BR>&gt; =
make=20
  Re-INVITE and UPDATE separatly expect for definition such as <BR>&gt;=20
  precondition. &nbsp;And this obeys current definition of RFC3311. =
<BR>&gt;=20
  <BR>&gt; In RFC3311, when the UPDATE is accepted by the other <BR>&gt; =

  side(UPDATE/200OK), the change of states is committed and effort at =
<BR>&gt;=20
  once. And making all UPDATEs during Re-INVITE as Re-INVITE's =
sub-<BR>&gt;=20
  transaction can be violation of RFC3311. So, we shold not treat all =
<BR>&gt;=20
  UPDATEs during Re-INVITE as Re-INVITE's sub-transaction. <BR>&gt; =
<BR>&gt;=20
  While UPDATE/200OK just refresh "precondition state table" in <BR>&gt; =

  RFC3312("precondition state table" is modified at once, so this =
<BR>&gt; obeys=20
  RFC3312), having no impat on the commitment of modification <BR>&gt; =
triggered=20
  by Re-INVITE, these UPDATE/200OK can be treated as sub-<BR>&gt; =
transaction of=20
  Re-INVITE. <BR>&gt; Other UPDATE/200OK can not be treated as =
sub-transaction=20
  of Re-INVITE. <BR>&gt; <BR>&gt; Comments/feeling are welcome!=20
  =
</TT></FONT><BR><BR><PRE>------------------------------------------------=
--------
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: <gao.yang2@zte.com.cn>
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Subject: Re: [Sip] "UPDATE during Re-INVITE" discussion
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The reason why we have this whole discussion is because the current =
specifications are unclear. If everything was clear, and everybody had =
the same understanding, we wouldn't need to clarify anything.
=20
So, whatever solution we choose, I am sure that someone will have to =
"rewrite their software".
=20
....IF that software exists in the first place, that is.=20
=20
I am not aware of any deployments which would support re-INVITEs with =
nested UPDATEs/PRACKs. Maybe there are such deployments, but I doubt =
they would all behave in the way as you describe.
=20
Regarding your second bullet, I don't even think that one should send =
"nested" UPDATEs, if they don't have anything to do with the re-INVITE. =
I think that is bad application design. Non-related changes should be =
done outside the re-INVITE transaction.
=20
Regards,
=20
Christer


________________________________

	From: gao.yang2@zte.com.cn [mailto:gao.yang2@zte.com.cn]=20
	Sent: 10. maaliskuuta 2009 9:36
	To: Christer Holmberg
	Cc: Eric.wang; SIP; SIPPING
	Subject: [Sip] "UPDATE during Re-INVITE" discussion
=09
=09

	OK. Thanks for your standpoint.=20
=09
	But accepting this means:=20
	1. Rewrite of current software to be as the behavior defined by =
"draft-camarillo-sipping-reinvite-00".=20
	2. Making all UPDATE/200OK "during" Re-INVITE as Re-INVITE's part.=20
=09
	Gao=20
=09
=09
=09
=09
"Christer Holmberg" <christer.holmberg@ericsson.com>=20

2009-03-10 15:22=20

=CA=D5=BC=FE=C8=CB
<gao.yang2@zte.com.cn>, "Eric.wang" <eric.wangxr@gmail.com>=20
=B3=AD=CB=CD
"SIP" <sip@ietf.org>, "SIPPING" <sipping@ietf.org>=20
=D6=F7=CC=E2
RE: [Sipping] =B4=F0=B8=B4: [Sip] "UPDATE during Re-INVITE" discussion

=09




	I support view 1.=20
	 =20
	Regards,=20
	 =20
	Christer=20
=09
=09
________________________________

	From: sipping-bounces@ietf.org [mailto:sipping-bounces@ietf.org] On =
Behalf Of gao.yang2@zte.com.cn
	Sent: 10. maaliskuuta 2009 8:54
	To: Eric.wang
	Cc: 'SIP'; 'SIPPING'
	Subject: [Sipping] =B4=F0=B8=B4: [Sip] "UPDATE during Re-INVITE" =
discussion
=09
=09
=09
	Yes. If we accept view 1, there would be some backward compatible =
problem. And this is a change to RFC3311.=20
=09
	I am waiting for more comments from people want to keep RFC3311 from =
re-write.=20
=09
	Gao=20
=09
=09
	"Eric.wang" <eric.wangxr@gmail.com> =D0=B4=D3=DA 2009-03-09 20:05:51:
=09
	> Hi,=20
	>  =20
	> I support view 2,not all update can be considered as sub-transaction
	> of re-INVITE.=20
	> According to current RFCs, it=A1=AFs NOT proper to consider all =
UPDATEs=20
	> as re-INVITE=A1=AFs sub-transaction=20
	> Eg:=20
	> UPDATE nested in re-INVITE  for target refresh can=A1=AFt be =
considered=20
	> as sub-transaction.=20
=09
=09
=09
	> There are two thought about "UPDATE during Re-INVITE". Which one is=20
	> better or suits for current RFCs=20
	>=20
	> There are two thought about "UPDATE during Re-INVITE".=20
	>=20
	> 1. All UPDATEs during Re-INVITE are Re-INVITE's sub-transaction=20
	> RFC3312(Precondition) is a case. As there is such case, we can=20
	> making all UPDATEs during Re-INVITE as Re-INVITE's sub-transaction.=20
	>=20
	> 2. Not all UPDATE(during Re-INVITE) can be considered as sub-
	> transaction of Re-INVITE=20
	> There is really need nested-modification, such as precondition and=20
	> more in the future.  But the cascade of nested transaction should be
	> defined in application-level, not signal-level. So, it is better to=20
	> make Re-INVITE and UPDATE separatly expect for definition such as=20
	> precondition.  And this obeys current definition of RFC3311.=20
	>=20
	> In RFC3311, when the UPDATE is accepted by the other=20
	> side(UPDATE/200OK), the change of states is committed and effort at=20
	> once. And making all UPDATEs during Re-INVITE as Re-INVITE's sub-
	> transaction can be violation of RFC3311. So, we shold not treat all=20
	> UPDATEs during Re-INVITE as Re-INVITE's sub-transaction.=20
	>=20
	> While UPDATE/200OK just refresh "precondition state table" in=20
	> RFC3312("precondition state table" is modified at once, so this=20
	> obeys RFC3312), having no impat on the commitment of modification=20
	> triggered by Re-INVITE, these UPDATE/200OK can be treated as sub-
	> transaction of Re-INVITE.=20
	> Other UPDATE/200OK can not be treated as sub-transaction of =
Re-INVITE.=20
	>=20
	> Comments/feeling are welcome!=20
=09
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dgb2312">
<META content=3D"MSHTML 6.00.6000.16809" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D404235507-10032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>The reason why we have this =
whole&nbsp;discussion is=20
because the current specifications are unclear. If everything was clear, =
and=20
everybody had the same understanding,&nbsp;we wouldn't need to clarify=20
anything.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D404235507-10032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D404235507-10032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>So, whatever solution we choose, I am sure that =
someone=20
will have to "rewrite their software".</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D404235507-10032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D404235507-10032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>....IF that software exists in the first place, =
that is.=20
</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D404235507-10032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D404235507-10032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>I am not aware of any deployments which would =
support=20
re-INVITEs with nested UPDATEs/PRACKs. Maybe there are such deployments, =
but I=20
doubt they would all behave in the way as you =
describe.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D404235507-10032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D404235507-10032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Regarding your second bullet, I don't even =
think that one=20
should send "nested"&nbsp;UPDATEs, if they don't have anything to do =
with the=20
re-INVITE. I think that is bad application design. Non-related changes =
should be=20
done outside the re-INVITE transaction.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D404235507-10032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D404235507-10032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D404235507-10032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D404235507-10032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Christer</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> gao.yang2@zte.com.cn=20
  [mailto:gao.yang2@zte.com.cn] <BR><B>Sent:</B> 10. maaliskuuta 2009=20
  9:36<BR><B>To:</B> Christer Holmberg<BR><B>Cc:</B> Eric.wang; SIP;=20
  SIPPING<BR><B>Subject:</B> [Sip] "UPDATE during Re-INVITE"=20
  discussion<BR></FONT><BR></DIV>
  <DIV></DIV><BR><FONT face=3Dsans-serif size=3D2>OK. Thanks for your=20
  standpoint.</FONT> <BR><BR><FONT face=3Dsans-serif size=3D2>But =
accepting this=20
  means:</FONT> <BR><FONT face=3Dsans-serif size=3D2>1. Rewrite of =
current software=20
  to be as the behavior defined by =
"draft-camarillo-sipping-reinvite-00".</FONT>=20
  <BR><FONT face=3Dsans-serif size=3D2>2. Making all UPDATE/200OK =
"during" Re-INVITE=20
  as Re-INVITE's part.</FONT> <BR><BR><FONT face=3Dsans-serif =
size=3D2>Gao</FONT>=20
  <BR><BR><BR><BR>
  <TABLE width=3D"100%">
    <TBODY>
    <TR vAlign=3Dtop>
      <TD width=3D"26%"><FONT face=3Dsans-serif size=3D1><B>"Christer =
Holmberg"=20
        &lt;christer.holmberg@ericsson.com&gt;</B> </FONT>
        <P><FONT face=3Dsans-serif size=3D1>2009-03-10 15:22</FONT> </P>
      <TD width=3D"73%">
        <TABLE width=3D"100%">
          <TBODY>
          <TR vAlign=3Dtop>
            <TD>
              <DIV align=3Dright><FONT face=3Dsans-serif =
size=3D1>=CA=D5=BC=FE=C8=CB</FONT></DIV>
            <TD><FONT face=3Dsans-serif =
size=3D1>&lt;gao.yang2@zte.com.cn&gt;,=20
              "Eric.wang" &lt;eric.wangxr@gmail.com&gt;</FONT>=20
          <TR vAlign=3Dtop>
            <TD>
              <DIV align=3Dright><FONT face=3Dsans-serif =
size=3D1>=B3=AD=CB=CD</FONT></DIV>
            <TD><FONT face=3Dsans-serif size=3D1>"SIP" =
&lt;sip@ietf.org&gt;,=20
              "SIPPING" &lt;sipping@ietf.org&gt;</FONT>=20
          <TR vAlign=3Dtop>
            <TD>
              <DIV align=3Dright><FONT face=3Dsans-serif =
size=3D1>=D6=F7=CC=E2</FONT></DIV>
            <TD><FONT face=3Dsans-serif size=3D1>RE: [Sipping] =
=B4=F0=B8=B4: [Sip] "UPDATE=20
              during Re-INVITE" =
discussion</FONT></TR></TBODY></TABLE><BR>
        <TABLE>
          <TBODY>
          <TR vAlign=3Dtop>
            <TD>
            =
<TD></TR></TBODY></TABLE><BR></TR></TBODY></TABLE><BR><BR><BR><FONT =
face=3DArial=20
  color=3Dblue size=3D2>I support view 1.</FONT> <BR><FONT =
face=3Dsans-serif=20
  size=3D3>&nbsp;</FONT> <BR><FONT face=3DArial color=3Dblue =
size=3D2>Regards,</FONT>=20
  <BR><FONT face=3Dsans-serif size=3D3>&nbsp;</FONT> <BR><FONT =
face=3DArial color=3Dblue=20
  size=3D2>Christer</FONT> <BR><BR>
  <HR>
  <FONT face=3DTahoma size=3D2><B>From:</B> sipping-bounces@ietf.org=20
  [mailto:sipping-bounces@ietf.org] <B>On Behalf Of=20
  </B>gao.yang2@zte.com.cn<B><BR>Sent:</B> 10. maaliskuuta 2009=20
  8:54<B><BR>To:</B> Eric.wang<B><BR>Cc:</B> 'SIP'; =
'SIPPING'<B><BR>Subject:</B>=20
  [Sipping] =B4=F0=B8=B4: [Sip] "UPDATE during Re-INVITE" =
discussion</FONT><FONT=20
  face=3Dsans-serif size=3D3><BR></FONT><BR><FONT face=3Dsans-serif=20
  size=3D3><BR></FONT><FONT face=3Dsans-serif size=3D2><BR>Yes. If we =
accept view 1,=20
  there would be some backward compatible problem. And this is a change =
to=20
  RFC3311.</FONT><FONT face=3Dsans-serif size=3D3> <BR></FONT><FONT =
face=3Dsans-serif=20
  size=3D2><BR>I am waiting for more comments from people want to keep =
RFC3311=20
  from re-write.</FONT><FONT face=3Dsans-serif size=3D3> =
<BR></FONT><FONT=20
  face=3Dsans-serif size=3D2><BR>Gao</FONT><FONT face=3Dsans-serif =
size=3D3>=20
  <BR><BR></FONT><FONT size=3D2><TT><BR>"Eric.wang" =
&lt;eric.wangxr@gmail.com&gt;=20
  =D0=B4=D3=DA 2009-03-09 20:05:51:<BR><BR>&gt; Hi,</TT></FONT><FONT =
face=3Dsans-serif=20
  size=3D3> </FONT><FONT size=3D2><TT><BR>&gt; &nbsp;</TT></FONT><FONT=20
  face=3Dsans-serif size=3D3> </FONT><FONT size=3D2><TT><BR>&gt; I =
support view 2,not=20
  all update can be considered as sub-transaction<BR>&gt; of=20
  re-INVITE.</TT></FONT><FONT face=3Dsans-serif size=3D3> </FONT><FONT=20
  size=3D2><TT><BR>&gt; According to current RFCs, it=A1=AFs NOT proper =
to consider all=20
  UPDATEs <BR>&gt; as re-INVITE=A1=AFs sub-transaction <BR>&gt; =
Eg:</TT></FONT><FONT=20
  face=3Dsans-serif size=3D3> </FONT><FONT size=3D2><TT><BR>&gt; UPDATE =
nested in=20
  re-INVITE &nbsp;for target refresh can=A1=AFt be considered <BR>&gt; =
as=20
  sub-transaction.</TT></FONT><FONT face=3Dsans-serif size=3D3> =
<BR><BR></FONT><FONT=20
  size=3D2><TT><BR><BR>&gt; There are two thought about "UPDATE during =
Re-INVITE".=20
  Which one is <BR>&gt; better or suits for current RFCs <BR>&gt; =
<BR>&gt; There=20
  are two thought about "UPDATE during Re-INVITE". <BR>&gt; <BR>&gt; 1. =
All=20
  UPDATEs during Re-INVITE are Re-INVITE's sub-transaction <BR>&gt;=20
  RFC3312(Precondition) is a case. As there is such case, we can =
<BR>&gt; making=20
  all UPDATEs during Re-INVITE as Re-INVITE's sub-transaction. <BR>&gt; =
<BR>&gt;=20
  2. Not all UPDATE(during Re-INVITE) can be considered as sub-<BR>&gt;=20
  transaction of Re-INVITE <BR>&gt; There is really need =
nested-modification,=20
  such as precondition and <BR>&gt; more in the future. &nbsp;But the =
cascade of=20
  nested transaction should be<BR>&gt; defined in application-level, not =

  signal-level. So, it is better to <BR>&gt; make Re-INVITE and UPDATE =
separatly=20
  expect for definition such as <BR>&gt; precondition. &nbsp;And this =
obeys=20
  current definition of RFC3311. <BR>&gt; <BR>&gt; In RFC3311, when the =
UPDATE=20
  is accepted by the other <BR>&gt; side(UPDATE/200OK), the change of =
states is=20
  committed and effort at <BR>&gt; once. And making all UPDATEs during =
Re-INVITE=20
  as Re-INVITE's sub-<BR>&gt; transaction can be violation of RFC3311. =
So, we=20
  shold not treat all <BR>&gt; UPDATEs during Re-INVITE as Re-INVITE's=20
  sub-transaction. <BR>&gt; <BR>&gt; While UPDATE/200OK just refresh=20
  "precondition state table" in <BR>&gt; RFC3312("precondition state =
table" is=20
  modified at once, so this <BR>&gt; obeys RFC3312), having no impat on =
the=20
  commitment of modification <BR>&gt; triggered by Re-INVITE, these =
UPDATE/200OK=20
  can be treated as sub-<BR>&gt; transaction of Re-INVITE. <BR>&gt; =
Other=20
  UPDATE/200OK can not be treated as sub-transaction of Re-INVITE. =
<BR>&gt;=20
  <BR>&gt; Comments/feeling are welcome! </TT></FONT><FONT =
face=3Dsans-serif=20
  size=3D3><BR></FONT><BR><FONT=20
  =
size=3D3><TT>--------------------------------------------------------<BR>=
ZTE=20
  Information Security Notice: The information contained in this mail is =
solely=20
  property of the sender's organization. This mail communication is=20
  confidential. Recipients named above are obligated to maintain secrecy =
and are=20
  not permitted to disclose the contents of this communication to=20
  others.<BR>This email and any files transmitted with it are =
confidential and=20
  intended solely for the use of the individual or entity to whom they =
are=20
  addressed. If you have received this email in error please notify the=20
  originator of the message. Any views expressed in this message are =
those of=20
  the individual sender.<BR>This message has been scanned for viruses =
and Spam=20
  by ZTE Anti-Spam =
system.<BR></TT></FONT><BR><BR><PRE>-------------------------------------=
-------------------
ZTE&nbsp;Information&nbsp;Security&nbsp;Notice:&nbsp;The&nbsp;information=
&nbsp;contained&nbsp;in&nbsp;this&nbsp;mail&nbsp;is&nbsp;solely&nbsp;prop=
erty&nbsp;of&nbsp;the&nbsp;sender's&nbsp;organization.&nbsp;This&nbsp;mai=
l&nbsp;communication&nbsp;is&nbsp;confidential.&nbsp;Recipients&nbsp;name=
d&nbsp;above&nbsp;are&nbsp;obligated&nbsp;to&nbsp;maintain&nbsp;secrecy&n=
bsp;and&nbsp;are&nbsp;not&nbsp;permitted&nbsp;to&nbsp;disclose&nbsp;the&n=
bsp;contents&nbsp;of&nbsp;this&nbsp;communication&nbsp;to&nbsp;others.
This&nbsp;email&nbsp;and&nbsp;any&nbsp;files&nbsp;transmitted&nbsp;with&n=
bsp;it&nbsp;are&nbsp;confidential&nbsp;and&nbsp;intended&nbsp;solely&nbsp=
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ty&nbsp;to&nbsp;whom&nbsp;they&nbsp;are&nbsp;addressed.&nbsp;If&nbsp;you&=
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sp;Any&nbsp;views&nbsp;expressed&nbsp;in&nbsp;this&nbsp;message&nbsp;are&=
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This&nbsp;message&nbsp;has&nbsp;been&nbsp;scanned&nbsp;for&nbsp;viruses&n=
bsp;and&nbsp;Spam&nbsp;by&nbsp;ZTE&nbsp;Anti-Spam&nbsp;system.
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Thread-Topic: RE: [Sip] "UPDATE during Re-INVITE" discussion
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: <gao.yang2@zte.com.cn>
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Subject: Re: [Sip] "UPDATE during Re-INVITE" discussion
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With bullet I didn't mean that you were trying to shoot me, or =
something. "Bullet" in this context refers to the second statement you =
gave :)


________________________________

	From: gao.yang2@zte.com.cn [mailto:gao.yang2@zte.com.cn]=20
	Sent: 10. maaliskuuta 2009 10:28
	To: Christer Holmberg
	Cc: Eric.wang; SIP; SIPPING
	Subject: =B4=F0=B8=B4: RE: [Sip] "UPDATE during Re-INVITE" discussion
=09
=09

	Hi,=20
=09
=09
=09
=09
"Christer Holmberg" <christer.holmberg@ericsson.com>=20

2009-03-10 16:01=20

=CA=D5=BC=FE=C8=CB
<gao.yang2@zte.com.cn>=20
=B3=AD=CB=CD
"Eric.wang" <eric.wangxr@gmail.com>, "SIP" <sip@ietf.org>, "SIPPING" =
<sipping@ietf.org>=20
=D6=F7=CC=E2
RE: [Sip] "UPDATE during Re-INVITE" discussion

=09




	The reason why we have this whole discussion is because the current =
specifications are unclear. If everything was clear, and everybody had =
the same understanding, we wouldn't need to clarify anything.=20
	[Gao] I think everything is clear. But people have different view =
towards UPDATE. So we need clarification than re-definition here.=20
	 =20
	So, whatever solution we choose, I am sure that someone will have to =
"rewrite their software".=20
	[Gao] Yes. But we should guard the right understanding. And compel =
people with misunderstanding to "rewrite their software". I think make =
every UPDATE/200OK during Re-INVITE as part of Re-INVITE is a "big" =
change for some software.=20
	I support to take UPDATE/200OK just refreshing "precondition state =
table" as part of Re-INVITE. And I think most of current software is =
doing so.=20
	 =20
	....IF that software exists in the first place, that is.=20
	 =20
	I am not aware of any deployments which would support re-INVITEs with =
nested UPDATEs/PRACKs. Maybe there are such deployments, but I doubt =
they would all behave in the way as you describe.=20
	[Gao] I just talked about if we want to obey RFC3311, it is not right =
to take all UPDATE/200OK(during Re-INVITE) as part of Re-INVITE.=20
	 =20
	Regarding your second bullet, I don't even think that one should send =
"nested" UPDATEs, if they don't have anything to do with the re-INVITE. =
I think that is bad application design. Non-related changes should be =
done outside the re-INVITE transaction.=20
	[Gao] It is not bullet, just friendly discussion.=20
	And I think the relationship of Re-INVITE and UPDATEs should be defined =
by application level definition, such as precondition. We should not =
just making relationship  by "during".=20
	 =20
=09
	Gao=20
=09
=09
________________________________

	From: gao.yang2@zte.com.cn [mailto:gao.yang2@zte.com.cn]=20
	Sent: 10. maaliskuuta 2009 9:36
	To: Christer Holmberg
	Cc: Eric.wang; SIP; SIPPING
	Subject: [Sip] "UPDATE during Re-INVITE" discussion
=09
=09
	OK. Thanks for your standpoint.=20
=09
	But accepting this means:=20
	1. Rewrite of current software to be as the behavior defined by =
"draft-camarillo-sipping-reinvite-00".=20
	2. Making all UPDATE/200OK "during" Re-INVITE as Re-INVITE's part.=20
=09
	Gao=20
=09
=09
=09
"Christer Holmberg" <christer.holmberg@ericsson.com>=20

2009-03-10 15:22=20



=CA=D5=BC=FE=C8=CB
<gao.yang2@zte.com.cn>, "Eric.wang" <eric.wangxr@gmail.com>=20
=B3=AD=CB=CD
"SIP" <sip@ietf.org>, "SIPPING" <sipping@ietf.org>=20
=D6=F7=CC=E2
RE: [Sipping] =B4=F0=B8=B4: [Sip] "UPDATE during Re-INVITE" discussion


=09


=09
=09
=09
	I support view 1.=20
	=20
	Regards,=20
	=20
	Christer=20
=09
=09
________________________________

	From: sipping-bounces@ietf.org [mailto:sipping-bounces@ietf.org] On =
Behalf Of gao.yang2@zte.com.cn
	Sent: 10. maaliskuuta 2009 8:54
	To: Eric.wang
	Cc: 'SIP'; 'SIPPING'
	Subject: [Sipping] =B4=F0=B8=B4: [Sip] "UPDATE during Re-INVITE" =
discussion
=09
=09
=09
	Yes. If we accept view 1, there would be some backward compatible =
problem. And this is a change to RFC3311.=20
=09
	I am waiting for more comments from people want to keep RFC3311 from =
re-write.=20
=09
	Gao=20
=09
=09
	"Eric.wang" <eric.wangxr@gmail.com> =D0=B4=D3=DA 2009-03-09 20:05:51:
=09
	> Hi,=20
	>  =20
	> I support view 2,not all update can be considered as sub-transaction
	> of re-INVITE.=20
	> According to current RFCs, it=A1=AFs NOT proper to consider all =
UPDATEs=20
	> as re-INVITE=A1=AFs sub-transaction=20
	> Eg:=20
	> UPDATE nested in re-INVITE  for target refresh can=A1=AFt be =
considered=20
	> as sub-transaction.=20
=09
=09
=09
	> There are two thought about "UPDATE during Re-INVITE". Which one is=20
	> better or suits for current RFCs=20
	>=20
	> There are two thought about "UPDATE during Re-INVITE".=20
	>=20
	> 1. All UPDATEs during Re-INVITE are Re-INVITE's sub-transaction=20
	> RFC3312(Precondition) is a case. As there is such case, we can=20
	> making all UPDATEs during Re-INVITE as Re-INVITE's sub-transaction.=20
	>=20
	> 2. Not all UPDATE(during Re-INVITE) can be considered as sub-
	> transaction of Re-INVITE=20
	> There is really need nested-modification, such as precondition and=20
	> more in the future.  But the cascade of nested transaction should be
	> defined in application-level, not signal-level. So, it is better to=20
	> make Re-INVITE and UPDATE separatly expect for definition such as=20
	> precondition.  And this obeys current definition of RFC3311.=20
	>=20
	> In RFC3311, when the UPDATE is accepted by the other=20
	> side(UPDATE/200OK), the change of states is committed and effort at=20
	> once. And making all UPDATEs during Re-INVITE as Re-INVITE's sub-
	> transaction can be violation of RFC3311. So, we shold not treat all=20
	> UPDATEs during Re-INVITE as Re-INVITE's sub-transaction.=20
	>=20
	> While UPDATE/200OK just refresh "precondition state table" in=20
	> RFC3312("precondition state table" is modified at once, so this=20
	> obeys RFC3312), having no impat on the commitment of modification=20
	> triggered by Re-INVITE, these UPDATE/200OK can be treated as sub-
	> transaction of Re-INVITE.=20
	> Other UPDATE/200OK can not be treated as sub-transaction of =
Re-INVITE.=20
	>=20
	> Comments/feeling are welcome!=20
=09
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<DIV dir=3Dltr align=3Dleft><SPAN class=3D658014408-10032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>With bullet I didn't mean that you&nbsp;were =
trying to=20
shoot me, or something. "Bullet" in this context refers to the second =
statement=20
you gave :)</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> gao.yang2@zte.com.cn=20
  [mailto:gao.yang2@zte.com.cn] <BR><B>Sent:</B> 10. maaliskuuta 2009=20
  10:28<BR><B>To:</B> Christer Holmberg<BR><B>Cc:</B> Eric.wang; SIP;=20
  SIPPING<BR><B>Subject:</B> =B4=F0=B8=B4: RE: [Sip] "UPDATE during =
Re-INVITE"=20
  discussion<BR></FONT><BR></DIV>
  <DIV></DIV><BR><FONT face=3Dsans-serif size=3D2>Hi,</FONT> =
<BR><BR><BR><BR>
  <TABLE width=3D"100%">
    <TBODY>
    <TR vAlign=3Dtop>
      <TD width=3D"26%"><FONT face=3Dsans-serif size=3D1><B>"Christer =
Holmberg"=20
        &lt;christer.holmberg@ericsson.com&gt;</B> </FONT>
        <P><FONT face=3Dsans-serif size=3D1>2009-03-10 16:01</FONT> </P>
      <TD width=3D"73%">
        <TABLE width=3D"100%">
          <TBODY>
          <TR vAlign=3Dtop>
            <TD>
              <DIV align=3Dright><FONT face=3Dsans-serif =
size=3D1>=CA=D5=BC=FE=C8=CB</FONT></DIV>
            <TD><FONT face=3Dsans-serif=20
              size=3D1>&lt;gao.yang2@zte.com.cn&gt;</FONT>=20
          <TR vAlign=3Dtop>
            <TD>
              <DIV align=3Dright><FONT face=3Dsans-serif =
size=3D1>=B3=AD=CB=CD</FONT></DIV>
            <TD><FONT face=3Dsans-serif size=3D1>"Eric.wang"=20
              &lt;eric.wangxr@gmail.com&gt;, "SIP" &lt;sip@ietf.org&gt;, =

              "SIPPING" &lt;sipping@ietf.org&gt;</FONT>=20
          <TR vAlign=3Dtop>
            <TD>
              <DIV align=3Dright><FONT face=3Dsans-serif =
size=3D1>=D6=F7=CC=E2</FONT></DIV>
            <TD><FONT face=3Dsans-serif size=3D1>RE: [Sip] "UPDATE =
during=20
              Re-INVITE" discussion</FONT></TR></TBODY></TABLE><BR>
        <TABLE>
          <TBODY>
          <TR vAlign=3Dtop>
            <TD>
            =
<TD></TR></TBODY></TABLE><BR></TR></TBODY></TABLE><BR><BR><BR><FONT =
face=3DArial=20
  size=3D2>The reason why we have this whole discussion is because the =
current=20
  specifications are unclear. If everything was clear, and everybody had =
the=20
  same understanding, we wouldn't need to clarify anything.</FONT> =
<BR><FONT=20
  face=3DArial color=3Dblue size=3D2>[Gao] I think everything is clear. =
But people=20
  have different view towards UPDATE. So we need clarification than=20
  re-definition here.</FONT> <BR><FONT face=3Dsans-serif =
size=3D3>&nbsp;</FONT>=20
  <BR><FONT face=3DArial size=3D2>So, whatever solution we choose, I am =
sure that=20
  someone will have to "rewrite their software".</FONT> <BR><FONT =
face=3DArial=20
  color=3Dblue size=3D2>[Gao] Yes. But we should guard the right =
understanding. And=20
  compel people with misunderstanding to "rewrite their software". I =
think make=20
  every UPDATE/200OK during Re-INVITE as part of Re-INVITE is a "big" =
change for=20
  some software.</FONT> <BR><FONT face=3DArial color=3Dblue size=3D2>I =
support to take=20
  UPDATE/200OK just refreshing "precondition state table" as part of =
Re-INVITE.=20
  And I think most of current software is doing so.</FONT> <BR><FONT=20
  face=3Dsans-serif size=3D3>&nbsp;</FONT> <BR><FONT face=3DArial =
size=3D2>....IF that=20
  software exists in the first place, that is. </FONT><BR><FONT =
face=3Dsans-serif=20
  size=3D3>&nbsp;</FONT> <BR><FONT face=3DArial size=3D2>I am not aware =
of any=20
  deployments which would support re-INVITEs with nested UPDATEs/PRACKs. =
Maybe=20
  there are such deployments, but I doubt they would all behave in the =
way as=20
  you describe.</FONT> <BR><FONT face=3DArial color=3Dblue =
size=3D2>[Gao] I just=20
  talked about if we want to obey RFC3311, it is not right to take all=20
  UPDATE/200OK(during Re-INVITE) as part of Re-INVITE.</FONT> <BR><FONT=20
  face=3Dsans-serif size=3D3>&nbsp;</FONT> <BR><FONT face=3DArial =
size=3D2>Regarding=20
  your second bullet, I don't even think that one should send "nested" =
UPDATEs,=20
  if they don't have anything to do with the re-INVITE. I think that is =
bad=20
  application design. Non-related changes should be done outside the =
re-INVITE=20
  transaction.</FONT> <BR><FONT face=3DArial color=3Dblue size=3D2>[Gao] =
It is not=20
  bullet, just friendly discussion.</FONT> <BR><FONT face=3DArial =
color=3Dblue=20
  size=3D2>And I think the relationship of Re-INVITE and UPDATEs should =
be defined=20
  by application level definition, such as precondition. We should not =
just=20
  making relationship &nbsp;by "during".</FONT> <BR><FONT =
face=3Dsans-serif=20
  size=3D3>&nbsp;</FONT> <BR><BR><FONT face=3Dsans-serif =
size=3D2>Gao</FONT> <BR><BR>
  <HR>
  <FONT face=3DTahoma size=3D2><B>From:</B> gao.yang2@zte.com.cn=20
  [mailto:gao.yang2@zte.com.cn] <B><BR>Sent:</B> 10. maaliskuuta 2009=20
  9:36<B><BR>To:</B> Christer Holmberg<B><BR>Cc:</B> Eric.wang; SIP;=20
  SIPPING<B><BR>Subject:</B> [Sip] "UPDATE during Re-INVITE"=20
  discussion</FONT><FONT face=3Dsans-serif size=3D3><BR></FONT><BR><FONT =

  face=3Dsans-serif size=3D2><BR>OK. Thanks for your =
standpoint.</FONT><FONT=20
  face=3Dsans-serif size=3D3> <BR></FONT><FONT face=3Dsans-serif =
size=3D2><BR>But=20
  accepting this means:</FONT><FONT face=3Dsans-serif size=3D3> =
</FONT><FONT=20
  face=3Dsans-serif size=3D2><BR>1. Rewrite of current software to be as =
the=20
  behavior defined by "draft-camarillo-sipping-reinvite-00".</FONT><FONT =

  face=3Dsans-serif size=3D3> </FONT><FONT face=3Dsans-serif =
size=3D2><BR>2. Making all=20
  UPDATE/200OK "during" Re-INVITE as Re-INVITE's part.</FONT><FONT=20
  face=3Dsans-serif size=3D3> <BR></FONT><FONT face=3Dsans-serif=20
  size=3D2><BR>Gao</FONT><FONT face=3Dsans-serif size=3D3> =
<BR><BR><BR></FONT>
  <TABLE width=3D"100%">
    <TBODY>
    <TR vAlign=3Dtop>
      <TD width=3D"47%"><FONT face=3Dsans-serif size=3D1><B>"Christer =
Holmberg"=20
        &lt;christer.holmberg@ericsson.com&gt;</B> </FONT>
        <P><FONT face=3Dsans-serif size=3D1>2009-03-10 15:22</FONT><FONT =

        face=3Dsans-serif size=3D3> </FONT></P>
      <TD width=3D"52%"><BR>
        <TABLE width=3D"100%">
          <TBODY>
          <TR vAlign=3Dtop>
            <TD width=3D"8%">
              <DIV align=3Dright><FONT face=3Dsans-serif =
size=3D1>=CA=D5=BC=FE=C8=CB</FONT></DIV>
            <TD width=3D"91%"><FONT face=3Dsans-serif=20
              size=3D1>&lt;gao.yang2@zte.com.cn&gt;, "Eric.wang"=20
              &lt;eric.wangxr@gmail.com&gt;</FONT><FONT =
face=3Dsans-serif size=3D3>=20
              </FONT>
          <TR vAlign=3Dtop>
            <TD>
              <DIV align=3Dright><FONT face=3Dsans-serif =
size=3D1>=B3=AD=CB=CD</FONT></DIV>
            <TD><FONT face=3Dsans-serif size=3D1>"SIP" =
&lt;sip@ietf.org&gt;,=20
              "SIPPING" &lt;sipping@ietf.org&gt;</FONT><FONT =
face=3Dsans-serif=20
              size=3D3> </FONT>
          <TR vAlign=3Dtop>
            <TD>
              <DIV align=3Dright><FONT face=3Dsans-serif =
size=3D1>=D6=F7=CC=E2</FONT></DIV>
            <TD><FONT face=3Dsans-serif size=3D1>RE: [Sipping] =
=B4=F0=B8=B4: [Sip] "UPDATE=20
              during Re-INVITE" =
discussion</FONT></TR></TBODY></TABLE><BR><BR>
        <TABLE width=3D"100%">
          <TBODY>
          <TR vAlign=3Dtop>
            <TD width=3D"50%">
            <TD =
width=3D"50%"></TR></TBODY></TABLE><BR></TR></TBODY></TABLE><BR><FONT=20
  face=3Dsans-serif size=3D3><BR><BR></FONT><FONT face=3DArial =
color=3Dblue size=3D2><BR>I=20
  support view 1.</FONT><FONT face=3Dsans-serif size=3D3> =
<BR>&nbsp;</FONT><FONT=20
  face=3DArial color=3Dblue size=3D2><BR>Regards,</FONT><FONT =
face=3Dsans-serif size=3D3>=20
  <BR>&nbsp;</FONT><FONT face=3DArial color=3Dblue =
size=3D2><BR>Christer</FONT><FONT=20
  face=3Dsans-serif size=3D3> <BR><BR></FONT>
  <HR>
  <FONT face=3DTahoma size=3D2><B>From:</B> sipping-bounces@ietf.org=20
  [mailto:sipping-bounces@ietf.org] <B>On Behalf Of=20
  </B>gao.yang2@zte.com.cn<B><BR>Sent:</B> 10. maaliskuuta 2009=20
  8:54<B><BR>To:</B> Eric.wang<B><BR>Cc:</B> 'SIP'; =
'SIPPING'<B><BR>Subject:</B>=20
  [Sipping] =B4=F0=B8=B4: [Sip] "UPDATE during Re-INVITE" =
discussion</FONT><FONT=20
  face=3Dsans-serif size=3D3><BR><BR></FONT><FONT face=3Dsans-serif=20
  size=3D2><BR><BR>Yes. If we accept view 1, there would be some =
backward=20
  compatible problem. And this is a change to RFC3311.</FONT><FONT=20
  face=3Dsans-serif size=3D3> </FONT><FONT face=3Dsans-serif =
size=3D2><BR><BR>I am=20
  waiting for more comments from people want to keep RFC3311 from=20
  re-write.</FONT><FONT face=3Dsans-serif size=3D3> </FONT><FONT =
face=3Dsans-serif=20
  size=3D2><BR><BR>Gao</FONT><FONT face=3Dsans-serif size=3D3> =
<BR></FONT><FONT=20
  size=3D2><TT><BR><BR>"Eric.wang" &lt;eric.wangxr@gmail.com&gt; =
=D0=B4=D3=DA 2009-03-09=20
  20:05:51:<BR><BR>&gt; Hi,</TT></FONT><FONT face=3Dsans-serif size=3D3> =

  </FONT><FONT size=3D2><TT><BR>&gt; &nbsp;</TT></FONT><FONT =
face=3Dsans-serif=20
  size=3D3> </FONT><FONT size=3D2><TT><BR>&gt; I support view 2,not all =
update can=20
  be considered as sub-transaction<BR>&gt; of =
re-INVITE.</TT></FONT><FONT=20
  face=3Dsans-serif size=3D3> </FONT><FONT size=3D2><TT><BR>&gt; =
According to current=20
  RFCs, it=A1=AFs NOT proper to consider all UPDATEs <BR>&gt; as =
re-INVITE=A1=AFs=20
  sub-transaction <BR>&gt; Eg:</TT></FONT><FONT face=3Dsans-serif =
size=3D3>=20
  </FONT><FONT size=3D2><TT><BR>&gt; UPDATE nested in re-INVITE =
&nbsp;for target=20
  refresh can=A1=AFt be considered <BR>&gt; as =
sub-transaction.</TT></FONT><FONT=20
  face=3Dsans-serif size=3D3> <BR></FONT><FONT =
size=3D2><TT><BR><BR><BR>&gt; There are=20
  two thought about "UPDATE during Re-INVITE". Which one is <BR>&gt; =
better or=20
  suits for current RFCs <BR>&gt; <BR>&gt; There are two thought about =
"UPDATE=20
  during Re-INVITE". <BR>&gt; <BR>&gt; 1. All UPDATEs during Re-INVITE =
are=20
  Re-INVITE's sub-transaction <BR>&gt; RFC3312(Precondition) is a case. =
As there=20
  is such case, we can <BR>&gt; making all UPDATEs during Re-INVITE as=20
  Re-INVITE's sub-transaction. <BR>&gt; <BR>&gt; 2. Not all =
UPDATE(during=20
  Re-INVITE) can be considered as sub-<BR>&gt; transaction of Re-INVITE =
<BR>&gt;=20
  There is really need nested-modification, such as precondition and =
<BR>&gt;=20
  more in the future. &nbsp;But the cascade of nested transaction should =

  be<BR>&gt; defined in application-level, not signal-level. So, it is =
better to=20
  <BR>&gt; make Re-INVITE and UPDATE separatly expect for definition =
such as=20
  <BR>&gt; precondition. &nbsp;And this obeys current definition of =
RFC3311.=20
  <BR>&gt; <BR>&gt; In RFC3311, when the UPDATE is accepted by the other =

  <BR>&gt; side(UPDATE/200OK), the change of states is committed and =
effort at=20
  <BR>&gt; once. And making all UPDATEs during Re-INVITE as Re-INVITE's=20
  sub-<BR>&gt; transaction can be violation of RFC3311. So, we shold not =
treat=20
  all <BR>&gt; UPDATEs during Re-INVITE as Re-INVITE's sub-transaction. =
<BR>&gt;=20
  <BR>&gt; While UPDATE/200OK just refresh "precondition state table" in =

  <BR>&gt; RFC3312("precondition state table" is modified at once, so =
this=20
  <BR>&gt; obeys RFC3312), having no impat on the commitment of =
modification=20
  <BR>&gt; triggered by Re-INVITE, these UPDATE/200OK can be treated as=20
  sub-<BR>&gt; transaction of Re-INVITE. <BR>&gt; Other UPDATE/200OK can =
not be=20
  treated as sub-transaction of Re-INVITE. <BR>&gt; <BR>&gt; =
Comments/feeling=20
  are welcome! </TT></FONT><FONT face=3Dsans-serif =
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------_=_NextPart_001_01C9A15C.80A59A4E--

From john.elwell@siemens.com  Tue Mar 10 02:28:12 2009
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Date: Tue, 10 Mar 2009 09:28:35 +0000
From: "Elwell, John" <john.elwell@siemens.com>
In-reply-to: <E6C2E8958BA59A4FB960963D475F7AC314C4DE6378@mail>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, sip@ietf.org
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Subject: Re: [Sip] FW: I-D Action:draft-kaplan-sip-asserter-identity-01.txt
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Hadriel,

An interesting draft, perhaps useful in two ways: first in the way you
intend (as asserter ID in closed domains or federations of closed
domains), but secondly some aspects might be relevant for inclusion in
an eventual solution to the SIP-Identity problem, notably the means of
selecting particular body parts and SDP attributes to sign.

On that particular issue, I am not sure whether the asserter is in a
position to decide on a request-by-request basis which body parts and
SDP attributes to sign, since it cannot know what downstream entities
are likely to need to do to those fields, so we may need some guidance
(e.g., a default list of things to sign).

John

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Hadriel Kaplan
> Sent: 08 March 2009 19:52
> To: sip@ietf.org
> Subject: [Sip] FW: I-D=20
> Action:draft-kaplan-sip-asserter-identity-01.txt
>=20
>=20
> Howdy,
> An updated version of SIP Asserter Identity has been=20
> uploaded, though with virtually no changes other than to=20
> avoid expiration.
> But since the topic of SIP Identity is coming up (again) at=20
> the upcoming meeting, and this draft is related, I thought=20
> I'd post this again.
>=20
> There was not much feedback to the previous rev, so I'm=20
> guessing folks either don't like it, or don't care, or just=20
> haven't had the time to read it.
>=20
> As always, comments are appreciated - including just saying=20
> "don't care".
>=20
> -hadriel
>=20
> > -----Original Message-----
> > From: i-d-announce-bounces@ietf.org=20
> [mailto:i-d-announce-bounces@ietf.org]
> > On Behalf Of Internet-Drafts@ietf.org
> > Sent: Sunday, March 08, 2009 2:30 PM
> > To: i-d-announce@ietf.org
> > Subject: I-D Action:draft-kaplan-sip-asserter-identity-01.txt
> >
> > A New Internet-Draft is available from the on-line Internet-Drafts
> > directories.
> >
> >         Title           : Private Extensions to the Session=20
> Initiation
> > Protocol (SIP) for Asserter Identification within Trusted Networks
> >         Author(s)       : H. Kaplan
> >         Filename        : draft-kaplan-sip-asserter-identity-01.txt
> >         Pages           : 17
> >         Date            : 2009-03-08
> >
> > This document describes private extensions to the Session
> > Initiation Protocol (SIP) that enable a network of trusted SIP
> > servers to identify the asserter of private user identity defined
> > in RFC 3325.  The use of these extensions is only applicable
> > inside a set of administrative domains with previously agreed-upon
> > policies for generation, transport and usage of such information.
> > This document does NOT offer a general identity model suitable for
> > use between different trust domains, or use in the Internet at
> > large.
> >
> > A URL for this Internet-Draft is:
> >=20
> http://www.ietf.org/internet-drafts/draft-kaplan-sip-asserter-
> identity-
> > 01.txt
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > Below is the data which will enable a MIME compliant mail reader
> > implementation to automatically retrieve the ASCII version of the
> > Internet-Draft.
>=20

From rco@iptel.org  Tue Mar 10 03:05:53 2009
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Subject: Re: [Sip] draft-state-sip-relay-attack-00
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Hadriel Kaplan wrote:
>   
>> -----Original Message-----
>> From: Raphael Coeffic [mailto:rco@iptel.org]
>> Sent: Monday, March 09, 2009 7:42 AM
>>
>> That's a good point. Requiring the user to be registered and only
>> accepting requests from the registered contacts provide a fair-enough
>> level of security concerning the attack debated. But I am still hoping
>> that we could find a solution not requiring this kind of measures, which
>> I would call "user-restricting".
>>     
>
> How is it "user-restricting" to require a UA to generate a REGISTER request?  Most humans don't generate the REGISTER request by hand - typically software does it for them.  :)
>   

Forcing registrations is the path that IMS went for, I believe. But if 
you want to take advantage of this, you may have to deploy a little more 
IMS than you'd like to. This reminds me of some email providers that 
require you to connect through POP3 prior to send any message through 
SMTP, instead of deploying any secure authentication mechnism.

Maybe just an example: let's say you have a home SIP server, doing the 
usual least cost routing. Your least cost router might have something 
like 50 different routes. Do you want this box, or maybe your phones to 
have 50 running registrations, just for the purpose of having cheap 
calls? Well, personaly, I would prefer to just install my certificate on 
this box, and use TLS. But as very very few of those PSTN providers do 
support TLS, I cannot. By the way, there are already commercial products 
supporting this scenario.

Regards,
Raphael.


From wwwrun@core3.amsl.com  Tue Mar 10 07:47:48 2009
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Subject: [Sip] Protocol Action: 'Framework for Establishing an SRTP Security Context using DTLS' to Proposed Standard
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The IESG has approved the following document:

- 'Framework for Establishing an SRTP Security Context using DTLS '
   <draft-ietf-sip-dtls-srtp-framework-07.txt> as a Proposed Standard

This document is the product of the Session Initiation Protocol Working 
Group. 

The IESG contact persons are Cullen Jennings and Jon Peterson.

A URL of this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-dtls-srtp-framework-07.txt

Technical Summary

This document is part of a suite of documents that specify a
mechanism for keying the Secure Real-time Transport Protocol (SRTP)
using Datagram Transport Layer Security (DTLS). Together, these
protocols provide for in-band establishment of secure multimedia
sessions without dependencies on external authentication
infrastructures. This document specifies the SIP framework for
DTLS-SRTP. A companion document, draft-ietf-avt-dtls-srtp,
specifies the SRTP portion.


Working Group Summary

This document is a product of the Session Initiation Protocol
(SIP) Working Group. It was developed in response to coordinated
meetings held with several working groups, and in consideration of
several competing specifications.


Document Quality

There are at least two known implementations of earlier versions of
this draft, one Open Source and one commercial. Several
implementors have expressed interest in implementation and
deployment of the final version.

Personnel

The Proto Shepherd for this document is Dean Willis, and the
responsible Area Director is Cullen Jennings.


From brett@broadsoft.com  Tue Mar 10 08:04:24 2009
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I think number 2 more accurately represents RFC 3261, RFC 3262, and RFC 331=
1.


From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of gao.y=
ang2@zte.com.cn
Sent: Sunday, March 08, 2009 10:49 PM
To: SIP; SIPPING
Subject: [Sip] "UPDATE during Re-INVITE" discussion


There are two thought about "UPDATE during Re-INVITE". Which one is better =
or suits for current RFCs

There are two thought about "UPDATE during Re-INVITE".

1. All UPDATEs during Re-INVITE are Re-INVITE's sub-transaction
RFC3312(Precondition) is a case. As there is such case, we can making all U=
PDATEs during Re-INVITE as Re-INVITE's sub-transaction.

2. Not all UPDATE(during Re-INVITE) can be considered as sub-transaction of=
 Re-INVITE
There is really need nested-modification, such as precondition and more in =
the future.  But the cascade of nested transaction should be defined in app=
lication-level, not signal-level. So, it is better to make Re-INVITE and UP=
DATE separatly expect for definition such as precondition.  And this obeys =
current definition of RFC3311.

In RFC3311, when the UPDATE is accepted by the other side(UPDATE/200OK), th=
e change of states is committed and effort at once. And making all UPDATEs =
during Re-INVITE as Re-INVITE's sub-transaction can be violation of RFC3311=
. So, we shold not treat all UPDATEs during Re-INVITE as Re-INVITE's sub-tr=
ansaction.

While UPDATE/200OK just refresh "precondition state table" in RFC3312("prec=
ondition state table" is modified at once, so this obeys RFC3312), having n=
o impat on the commitment of modification triggered by Re-INVITE, these UPD=
ATE/200OK can be treated as sub-transaction of Re-INVITE.
Other UPDATE/200OK can not be treated as sub-transaction of Re-INVITE.

Comments/feeling are welcome!



--------------------------------------------------------

ZTE Information Security Notice: The information contained in this mail is =
solely property of the sender's organization. This mail communication is co=
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<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew";
color:#1F497D'>I think number 2 more accurately represents RFC 3261, RFC 32=
62,
and RFC 3311.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'>

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<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in'>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma=
","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> sip-bounces@i=
etf.org
[mailto:sip-bounces@ietf.org] <b>On Behalf Of </b>gao.yang2@zte.com.cn<br>
<b>Sent:</b> Sunday, March 08, 2009 10:49 PM<br>
<b>To:</b> SIP; SIPPING<br>
<b>Subject:</b> [Sip] &quot;UPDATE during Re-INVITE&quot; discussion<o:p></=
o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>There are=
 two
thought about &quot;UPDATE during Re-INVITE&quot;. Which one is better or s=
uits
for current RFCs</span> <br>
<br>
<b><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>There =
are
two thought about &quot;UPDATE during Re-INVITE&quot;.</span></b> <br>
<br>
<b><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>1. All
UPDATEs during Re-INVITE are Re-INVITE's sub-transaction</span></b> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>RFC3312(P=
recondition)
is a case. As there is such case, we can making all UPDATEs during Re-INVIT=
E as
Re-INVITE's sub-transaction.</span> <br>
<br>
<b><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>2. Not=
 all
UPDATE(during Re-INVITE) can be considered as sub-transaction of Re-INVITE<=
/span></b>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>There is =
really
need nested-modification, such as precondition and more in the future.
&nbsp;But the cascade of nested transaction should be defined in
application-level, not signal-level. <b>So, it is better to make Re-INVITE =
and
UPDATE separatly expect for definition such as precondition.</b> &nbsp;<b>A=
nd
this obeys current definition of RFC3311.</b></span> <br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>In RFC331=
1,
when the UPDATE is accepted by the other side(UPDATE/200OK), the change of
states is committed and effort at once. And making all UPDATEs during Re-IN=
VITE
as Re-INVITE's sub-transaction can be violation of RFC3311. So, we shold no=
t
treat all UPDATEs during Re-INVITE as Re-INVITE's sub-transaction.</span> <=
br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>While
UPDATE/200OK just refresh &quot;precondition state table&quot; in
RFC3312(&quot;precondition state table&quot; is modified at once, so this o=
beys
RFC3312), having no impat on the commitment of modification triggered by
Re-INVITE, these UPDATE/200OK can be treated as sub-transaction of Re-INVIT=
E.</span>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Other
UPDATE/200OK can not be treated as sub-transaction of Re-INVITE.</span> <br=
>
<br>
<b><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Commen=
ts/feeling
are welcome!</span></b> <o:p></o:p></p>

<pre><o:p>&nbsp;</o:p></pre><pre>------------------------------------------=
--------------<o:p></o:p></pre><pre>ZTE&nbsp;Information&nbsp;Security&nbsp=
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From brett@broadsoft.com  Tue Mar 10 08:04:25 2009
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From: Brett Tate <brett@broadsoft.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, SIPPING <sipping@ietf.org>, SIP <sip@ietf.org>
Date: Tue, 10 Mar 2009 08:04:53 -0700
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> The reason why we have this whole=A0discussion is because the=20
> current specifications are unclear.

As mentioned before, I think the following RFC 3261 quote is clear although=
 some disagree.

RFC 3261 section 14.1: "If a UA receives a non-2xx final response to a re-I=
NVITE, the session parameters MUST remain unchanged, as if no re-INVITE had=
 been issued."


> Regarding your second bullet, I don't even think that one=20
> should send "nested"=A0UPDATEs, if they don't have anything to=20
> do with the re-INVITE. I think that is bad application design.=20
> Non-related changes should be done outside the re-INVITE transaction.
=A0
Sending UPDATE per RFC 4028 to refresh or audit the dialog is one of the ti=
mes UPDATE may be tried while a re-INVITE is occurring.


From christer.holmberg@ericsson.com  Tue Mar 10 08:15:28 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Brett Tate" <brett@broadsoft.com>, "SIPPING" <sipping@ietf.org>, "SIP" <sip@ietf.org>
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Hi,=20

>>Regarding your second bullet, I don't even think that one should send=20
>>"nested"=A0UPDATEs, if they don't have anything to do with the=20
>>re-INVITE. I think that is bad application design.
>>Non-related changes should be done outside the re-INVITE transaction.
> =A0
>Sending UPDATE per RFC 4028 to refresh or audit the dialog is=20
>one of the times UPDATE may be tried while a re-INVITE is occurring.

True, but normally those UPDATEs are not going to modify session =
parameters.

So, yes, it may happen, but I think we should recommend against doing =
so.

Regards,

Christer


From scott.lawrence@nortel.com  Tue Mar 10 11:02:09 2009
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From: "Scott Lawrence" <scott.lawrence@nortel.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>
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On Sat, 2009-03-07 at 14:51 -0500, Hadriel Kaplan wrote:

> Another question is whether an attacker can create special nonce's for
> one or more challenges it sends, that will help it figure out the
> password.  For example can they help a rainbow-table type of password
> cracking, and how much the cnonce can prevent that.

If the cnonce is well chosen and the nc value incremented, I think that
gets pretty hard, but I'm not by any means an expert on cracking one-way
hash functions.



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Thread-Topic: [Sip] "UPDATE during Re-INVITE" discussion
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Subject: Re: [Sip] "UPDATE during Re-INVITE" discussion
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>>> Regarding your second bullet, I don't even think that one=20
>>> should send "nested"=A0UPDATEs, if they don't have anything=20
>>> to do with the re-INVITE. I think that is bad application=20
>>> design.  Non-related changes should be done outside the=20
>>> re-INVITE transaction.
>>
>> Sending UPDATE per RFC 4028 to refresh or audit the dialog
>> is one of the times UPDATE may be tried while a re-INVITE is
>> occurring.
>=20
> True, but normally those UPDATEs are not going to modify session
> parameters.
>
> So, yes, it may happen, but I think we should recommend=20
> against doing so.

There are two key phrases within rfc3261 which might be adding to the confu=
sion: "session parameters" and "state changes".  RFC 3261 section 12.2.2 an=
d 14.1 basically indicate to rollback as though the re-INVITE was not sent.=
  There have been numerous proposal concerning what should and shouldn't ro=
llback; it is occasional hard to distinguish if people mean 1) SDP 2) non S=
DP state data (like Contact, Session-Expires, ...), or 3) both. =20

I agree that UPDATE likely wouldn't include SDP when only attempting to ref=
resh or audit the session; however it can.

I think 12.2.2 is sufficiently clear concerning what should occur upon re-I=
NVITE failure; however I think it was unfortunate that section 14.1 indicat=
ed "session parameters" instead of "state changes".

RFC 3261 section 12.2.2: "Requests sent within a dialog, as any other reque=
sts, are atomic.  If a particular request is accepted by the UAS, all the s=
tate changes associated with it are performed.  If the request is rejected,=
 none of the state changes are performed."

Concerning UPDATE in relation to section 12.2.2, I consider it an atomic tr=
ansaction.  And for it is worth, I consider PRACK to not be an atomic trans=
action.


From HKaplan@acmepacket.com  Tue Mar 10 14:35:52 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Raphael Coeffic <rco@iptel.org>
Date: Tue, 10 Mar 2009 17:36:24 -0400
Thread-Topic: [Sip] draft-state-sip-relay-attack-00
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Cc: Nils Ohlmeier <lists@ohlmeier.org>, "sip@ietf.org" <sip@ietf.org>
Subject: Re: [Sip] draft-state-sip-relay-attack-00
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> -----Original Message-----
> From: Raphael Coeffic [mailto:rco@iptel.org]
> Sent: Tuesday, March 10, 2009 6:06 AM
>=20
> Hadriel Kaplan wrote:
>=20
> Forcing registrations is the path that IMS went for, I believe. But if
> you want to take advantage of this, you may have to deploy a little more
> IMS than you'd like to.=20

The concept and practice of "forcing" UA's, which have AoR's belonging to a=
 domain, to REGISTER their contact binding for those AoR's in order to make=
 calls, has been around a lot longer than IMS has.  Invoking the "that soun=
ds like IMS" defense is a cheap guilt-by-association argument, and not rele=
vant.  One might as well argue that using INVITE's or RTP "is the path that=
 IMS went for".

You keep making it sound like sending a SIP request which does not create a=
 dialog is somehow a difficult mechanism to implement, or "forces" people t=
o do something they don't want to.  What is it forcing?

You're looking for a means of authenticating a UA without being subject to =
a relay attack.  Using a different SIP Method name, which can only be initi=
ated by the client with the credentials, and one that UA's cannot be baited=
 to provide by random third parties, is one solution to the problem.  If yo=
u don't like the string "REGISTER", then how about "OPTIONS"?


> This reminds me of some email providers that
> require you to connect through POP3 prior to send any message through
> SMTP, instead of deploying any secure authentication mechnism.

No, this is more like a Website which offers email services requiring you t=
o login using your credentials before letting you send.  It's not analogous=
 to making you use POP3 because a REGISTER does not use a different protoco=
l, and can be tied to the same connection-layer "flow" as the INVITE et al.

You may not like it, but any form of authentication by definition is more r=
estrictive than no form of authentication.

=20
> Maybe just an example: let's say you have a home SIP server, doing the
> usual least cost routing. Your least cost router might have something
> like 50 different routes. Do you want this box, or maybe your phones to
> have 50 running registrations, just for the purpose of having cheap
> calls? Well, personaly, I would prefer to just install my certificate on
> this box, and use TLS. But as very very few of those PSTN providers do
> support TLS, I cannot. By the way, there are already commercial products
> supporting this scenario.

So you have a mechanism that people don't seem to want to implement - and m=
eanwhile you have another mechanism, which is widely implemented.  And the =
problem is...?

-hadriel

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From dean.willis@softarmor.com  Tue Mar 10 21:18:29 2009
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A SIP agenda for IETF 74 has been posted at:

http://www.ietf.org/proceedings/09mar/agenda/sip.html

The WIKI version remains active at:
http://www.softarmor.com/mediawiki/index.php/SIP_Agenda_IETF_74

Note that we're trying out the SIPCORE/DISPATCH split.

The Tuesday session will run as SIPCORE and start with an intro, but  
no discussion of, the new model.

The Wednesday session will run as DISPATCH.  This may be something of  
a gong show.

Ongoing SIP and SIPPING work that is not clearly in the draft charters  
of SIPCORE or DISPATCH will be handled in the Friday morning SIPPING  
session. We will also discuss the new work model and RFC3274bis in the  
Friday session.

Here's an mhtml version of the agenda for you folks (like cullen) who  
really want it in email. Note that it includes on the reading list a  
bunch of drafts that we haven't allocated time to; the inclusion is so  
you folks can see where those drafts might go in the new model.



Draft Agenda for SIP at IETF 74 -- SIPCORE Session
Victim	Topic	Drafts	Time
Jon Peterson	SIP Change Process Intro
(Discussion on Friday)	draft-peterson-rai-rfc3427bis-01	20
Christer Holmberg	Early Dialog termination	draft-ietf-sip-199-06	20
Robert Sparks	3261 Interop Statement	draft-sparks-sip-3261-interop- 
statement-00	15
Robert Sparks	3261 Normative References	draft-sparks-sip-3261-norm-ref- 
status-00	15
Robert Sparks	Fix INVITE transaction	draft-sparks-sip-invfix-03	15
Mary Barnes	RFC 4244bis	draft-barnes-sip-rfc4244bis-00	20
Placeholder	No Agenda Time	draft-ietf-sip-ipv6-abnf-fix-03
draft-rosenberg-sip-target-uri-delivery-00
0

Draft Agenda for SIP at IETF 74 -- DISPATCH
Victim	Topic	Drafts	Time
Theo Zourzouvillys	Via Cookies	draft-zourzouvillys-sip-via-cookie-02	15
Christer Holmberg	Keepalive Without Outbound	draft-holmberg-sip- 
keep-03	15
Hadriel Kaplan	Secure Call ID	draft-kaplan-sip-secure-call-id-00	15
Hadriel Kaplan	Session ID	draft-kaplan-sip-session-id-01	15
Alan Johnston	Batch NOTIFYs	draft-johnston-sipping-batch-notify-00	10
Alan Johnston	User to User for ISDN	draft-johnston-sipping-cc-uui-07	10
Salvatore Loreto	Context-ID Requirements	draft-loreto-sipping-context- 
id-requirements	10
Salvatore Loreto	Changes to Referred-By	draft-loreto- 
sipping-3892bis-01	10
Hadriel Kaplan	Updates to the Updates to Asserted Identity in SIP	 
draft-kaplan-sipping-pai-responses-00	10
Raphael Coeffic
Radu State	Digest Relay Attack	draft-state-sip-relay-attack-00	10
Placeholder	Agenda Time Not Requested	draft-avasarala-sipping-reason- 
header-dynamic-icb-00
draft-boucadair-sipping-ipv6-atypes-00
draft-gurbani-sipping-clf-00
draft-niccolini-sipping-siphandover-05
draft-rosenberg-sip-app-media-tag-02
draft-kuthan-sip-derive-00
draft-polk-geopriv-sip-lo-retrans-rec-concerns-00
draft-roach-sip-http-subscribe-01
draft-rosenberg-sip-tote-02
draft-york-sip-visual-identifier-trusted-identity-01
0

Draft Agenda for SIP at IETF 74 -- SIP/SIPPING Continuing Work
Victim	Topic	Drafts	Time
Gonzalo Camarillo	Re-INVITE Handling in SIP & Media State under  
Preconditions in SIP	draft-camarillo-sipping-reinvite-00
draft-camarillo-sipping-precons-00
15
Christer Holmberg	Re-Invite Fallback (based on ML discussion)	draft- 
ietf-sipping-sip-offeranswer-10	15
Eric Burger	Info Events	draft-ietf-sip-info-events-03	15
Hadriel Kaplan	Identity	draft-elwell-sip-e2e-identity-important-03	30
Dale Worley	Profile Datasets	draft-ietf-sipping-profile-datasets-02	15
Volker Hilt	SIP Overload Control	draft-ietf-sipping-overload-design-02	 
15
Salvatore Loreto	Event Throttling	draft-niemi-sipping-event- 
throttle-08	15
Jon Peterson	SIP Change Process Intro (Discussion on Friday)	draft- 
peterson-rai-rfc3427bis-01	30
Placeholder	No Agenda Time	draft-ietf-sip-outbound-16
draft-ietf-sip-connect-reuse-12
draft-ietf-sipping-sip-offeranswer-10
draft-ietf-sip-certs-07
draft-ietf-sip-dtls-srtp-framework-06
draft-ietf-sip-session-policy-framework-05
draft-ietf-sip-xcapevent-04
draft-ietf-sip-location-conveyance-12
draft-ietf-sip-body-handling-05
draft-ietf-sip-domain-certs-02
draft-ietf-sip-eku-03
draft-ietf-sip-record-route-fix-05
draft-ietf-sip-saml-05
draft-ietf-sip-ua-privacy-05
draft-rosenberg-sip-info-litmus-02
draft-ietf-sip-ua-privacy-06
0



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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><div><br></div><div>A SIP =
agenda for IETF 74 has been posted at:</div><div><br></div><div><a =
href=3D"http://www.ietf.org/proceedings/09mar/agenda/sip.html">http://www.=
ietf.org/proceedings/09mar/agenda/sip.html</a></div><div><br></div><div>Th=
e WIKI version remains active at:</div><div><a =
href=3D"http://www.softarmor.com/mediawiki/index.php/SIP_Agenda_IETF_74">h=
ttp://www.softarmor.com/mediawiki/index.php/SIP_Agenda_IETF_74</a></div><d=
iv><br></div><div>Note that we're trying out the SIPCORE/DISPATCH =
split.</div><div><br></div><div>The Tuesday session will run as SIPCORE =
and start with an intro, but no discussion of, the new =
model.&nbsp;</div><div><br></div><div>The Wednesday session will run as =
DISPATCH. &nbsp;This may be something of a gong =
show.</div><div><br></div><div>Ongoing SIP and SIPPING work that is not =
clearly in the draft charters of SIPCORE or DISPATCH will be handled in =
the Friday morning SIPPING session. We will also discuss the new work =
model and RFC3274bis in the Friday =
session.</div><div><br></div><div>Here's an mhtml version of the agenda =
for you folks (like cullen) who really want it in email. Note that it =
includes on the reading list a bunch of drafts that we haven't allocated =
time to; the inclusion is so you folks can see where those drafts might =
go in the new =
model.</div><div><br></div><div><br></div><div><br></div><div><span =
class=3D"Apple-style-span" style=3D"font-family: -webkit-sans-serif; =
font-size: 13px; line-height: 19px; "><table border=3D"1" =
style=3D"background-image: initial; background-repeat: initial; =
background-attachment: initial; -webkit-background-clip: initial; =
-webkit-background-origin: initial; background-color: white; font-size: =
100%; color: black; background-position: initial initial; =
"><caption>Draft Agenda for SIP at IETF 74 -- SIPCORE =
Session</caption><tbody><tr><th>Victim</th><th>Topic</th><th>Drafts</th><t=
h>Time</th></tr><tr><td>Jon Peterson</td><td>SIP Change Process =
Intro&nbsp;<br>(Discussion on Friday)</td><td><a =
href=3D"http://tools.ietf.org/id/draft-peterson-rai-rfc3427bis-01.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-peterson-rai-rfc3427bis-01.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-peterson-rai-rfc3427bis-01</a></td><td>20</td></tr><tr><td>Christe=
r Holmberg</td><td>Early Dialog termination</td><td><a =
href=3D"http://tools.ietf.org/id/draft-ietf-sip-199-06.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-ietf-sip-199-06.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-ietf-sip-199-06</a></td><td>20</td></tr><tr><td>Robert =
Sparks</td><td>3261 Interop Statement</td><td><a =
href=3D"http://tools.ietf.org/id/draft-sparks-sip-3261-interop-statement-0=
0.txt" class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-sparks-sip-3261-interop-statement-=
00.txt" rel=3D"nofollow" style=3D"text-decoration: none; =
background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-sparks-sip-3261-interop-statement-00</a></td><td>15</td></tr><tr><=
td>Robert Sparks</td><td>3261 Normative References</td><td><a =
href=3D"http://tools.ietf.org/id/draft-sparks-sip-3261-norm-ref-status-00.=
txt" class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-sparks-sip-3261-norm-ref-status-00=
.txt" rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-sparks-sip-3261-norm-ref-status-00</a></td><td>15</td></tr><tr><td=
>Robert Sparks</td><td>Fix INVITE transaction</td><td><a =
href=3D"http://tools.ietf.org/html/draft-sparks-sip-invfix-03" =
class=3D"external text" =
title=3D"http://tools.ietf.org/html/draft-sparks-sip-invfix-03" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-sparks-sip-invfix-03</a></td><td>15</td></tr><tr><td>Mary =
Barnes</td><td><a href=3D"http://www.ietf.org/rfc/rfc4244.txt" =
class=3D"external" title=3D"http://www.ietf.org/rfc/rfc4244.txt" =
style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; ">RFC 4244</a>bis</td><td><a =
href=3D"http://tools.ietf.org/id/draft-barnes-sip-rfc4244bis-00.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-barnes-sip-rfc4244bis-00.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-barnes-sip-rfc4244bis-00</a></td><td>20</td></tr><tr><td>Placehold=
er</td><td>No Agenda Time</td><td><a =
href=3D"http://tools.ietf.org/id/draft-ietf-sip-ipv6-abnf-fix-03.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-ietf-sip-ipv6-abnf-fix-03.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-ietf-sip-ipv6-abnf-fix-03</a><br><div style=3D"margin-top: =
0.4em; margin-right: 0px; margin-bottom: 0.5em; margin-left: 0px; =
line-height: 1.5em; "><a =
href=3D"http://tools.ietf.org/id/draft-rosenberg-sip-target-uri-delivery-0=
0.txt" class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-rosenberg-sip-target-uri-delivery-=
00.txt" rel=3D"nofollow" style=3D"text-decoration: none; =
background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-rosenberg-sip-target-uri-delivery-00</a><br></div></td><td>0</td><=
/tr></tbody></table><div style=3D"margin-top: 0.4em; margin-right: 0px; =
margin-bottom: 0.5em; margin-left: 0px; line-height: 1.5em; =
"><br></div><table border=3D"1" style=3D"background-image: initial; =
background-repeat: initial; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: white; font-size: 100%; color: black; =
background-position: initial initial; "><caption>Draft Agenda for SIP at =
IETF 74 -- =
DISPATCH</caption><tbody><tr><th>Victim</th><th>Topic</th><th>Drafts</th><=
th>Time</th></tr><tr><td>Theo Zourzouvillys</td><td>Via =
Cookies</td><td><a =
href=3D"http://tools.ietf.org/id/draft-zourzouvillys-sip-via-cookie-02.txt=
" class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-zourzouvillys-sip-via-cookie-02.tx=
t" rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-zourzouvillys-sip-via-cookie-02</a></td><td>15</td></tr><tr><td>Ch=
rister Holmberg</td><td>Keepalive Without Outbound</td><td><a =
href=3D"http://tools.ietf.org/html/draft-holmberg-sip-keep-03" =
class=3D"external text" =
title=3D"http://tools.ietf.org/html/draft-holmberg-sip-keep-03" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-holmberg-sip-keep-03</a></td><td>15</td></tr><tr><td>Hadriel =
Kaplan</td><td>Secure Call ID</td><td><a =
href=3D"http://www.ietf.org/internet-drafts/draft-kaplan-sip-secure-call-i=
d-00.txt" class=3D"external text" =
title=3D"http://www.ietf.org/internet-drafts/draft-kaplan-sip-secure-call-=
id-00.txt" rel=3D"nofollow" style=3D"text-decoration: none; =
background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-kaplan-sip-secure-call-id-00</a></td><td>15</td></tr><tr><td>Hadri=
el Kaplan</td><td>Session ID</td><td><a =
href=3D"http://tools.ietf.org/id/draft-kaplan-sip-session-id-01.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-kaplan-sip-session-id-01.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-kaplan-sip-session-id-01</a></td><td>15</td></tr><tr><td>Alan =
Johnston</td><td>Batch NOTIFYs</td><td><a =
href=3D"http://tools.ietf.org/id/draft-johnston-sipping-batch-notify-00.tx=
t" class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-johnston-sipping-batch-notify-00.t=
xt" rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-johnston-sipping-batch-notify-00</a></td><td>10</td></tr><tr><td>A=
lan Johnston</td><td>User to User for ISDN</td><td><a =
href=3D"http://tools.ietf.org/id/draft-johnston-sipping-cc-uui-07.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-johnston-sipping-cc-uui-07.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-johnston-sipping-cc-uui-07</a></td><td>10</td></tr><tr><td>Salvato=
re Loreto</td><td>Context-ID Requirements</td><td><a =
href=3D"http://tools.ietf.org/id/draft-loreto-sipping-context-id-requireme=
nts-01.txt" class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-loreto-sipping-context-id-requirem=
ents-01.txt" rel=3D"nofollow" style=3D"text-decoration: none; =
background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-loreto-sipping-context-id-requirements</a></td><td>10</td></tr><tr=
><td>Salvatore Loreto</td><td>Changes to Referred-By</td><td><a =
href=3D"http://tools.ietf.org/id/draft-loreto-sipping-3892bis-01.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-loreto-sipping-3892bis-01.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-loreto-sipping-3892bis-01</a></td><td>10</td></tr><tr><td>Hadriel =
Kaplan</td><td>Updates to the Updates to Asserted Identity in =
SIP</td><td><a =
href=3D"http://tools.ietf.org/id/draft-kaplan-sipping-pai-responses-00.txt=
" class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-kaplan-sipping-pai-responses-00.tx=
t" rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-kaplan-sipping-pai-responses-00</a></td><td>10</td></tr><tr><td>Ra=
phael Coeffic<br>Radu State</td><td>Digest Relay Attack</td><td><a =
href=3D"http://tools.ietf.org/id/draft-state-sip-relay-attack-00.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-state-sip-relay-attack-00.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-state-sip-relay-attack-00</a></td><td>10</td></tr><tr><td>Placehol=
der</td><td>Agenda Time Not Requested</td><td><a =
href=3D"http://tools.ietf.org/id/draft-avasarala-sipping-reason-header-dyn=
amic-icb-00.txt" class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-avasarala-sipping-reason-header-dy=
namic-icb-00.txt" rel=3D"nofollow" style=3D"text-decoration: none; =
background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-avasarala-sipping-reason-header-dynamic-icb-00</a><br><div =
style=3D"margin-top: 0.4em; margin-right: 0px; margin-bottom: 0.5em; =
margin-left: 0px; line-height: 1.5em; "><a =
href=3D"http://tools.ietf.org/id/draft-boucadair-sipping-ipv6-atypes-00.tx=
t" class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-boucadair-sipping-ipv6-atypes-00.t=
xt" rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-boucadair-sipping-ipv6-atypes-00</a><br><a =
href=3D"http://tools.ietf.org/id/draft-gurbani-sipping-clf-00.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-gurbani-sipping-clf-00.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-gurbani-sipping-clf-00</a><br><a =
href=3D"http://tools.ietf.org/id/draft-niccolini-sipping-siphandover-05.tx=
t" class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-niccolini-sipping-siphandover-05.t=
xt" rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-niccolini-sipping-siphandover-05</a><br><a =
href=3D"http://tools.ietf.org/html/draft-rosenberg-sip-app-media-tag-02" =
class=3D"external text" =
title=3D"http://tools.ietf.org/html/draft-rosenberg-sip-app-media-tag-02" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-rosenberg-sip-app-media-tag-02</a><br><a =
href=3D"http://tools.ietf.org/id/draft-kuthan-sip-derive-00.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-kuthan-sip-derive-00.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-kuthan-sip-derive-00</a><br><a =
href=3D"http://tools.ietf.org/html/draft-polk-geopriv-sip-lo-retrans-rec-c=
oncerns-00" class=3D"external text" =
title=3D"http://tools.ietf.org/html/draft-polk-geopriv-sip-lo-retrans-rec-=
concerns-00" rel=3D"nofollow" style=3D"text-decoration: none; =
background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-polk-geopriv-sip-lo-retrans-rec-concerns-00</a><br><a =
href=3D"http://tools.ietf.org/id/draft-roach-sip-http-subscribe-01.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-roach-sip-http-subscribe-01.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-roach-sip-http-subscribe-01</a><br><a =
href=3D"http://tools.ietf.org/id/draft-rosenberg-sip-tote-02.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-rosenberg-sip-tote-02.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-rosenberg-sip-tote-02</a><br><a =
href=3D"http://tools.ietf.org/id/draft-york-sip-visual-identifier-trusted-=
identity-01.txt" class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-york-sip-visual-identifier-trusted=
-identity-01.txt" rel=3D"nofollow" style=3D"text-decoration: none; =
background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-york-sip-visual-identifier-trusted-identity-01</a></div></td><td>0=
</td></tr></tbody></table><div style=3D"margin-top: 0.4em; margin-right: =
0px; margin-bottom: 0.5em; margin-left: 0px; line-height: 1.5em; =
"><br></div><table border=3D"1" style=3D"background-image: initial; =
background-repeat: initial; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: white; font-size: 100%; color: black; position: =
static; z-index: auto; "><caption>Draft Agenda for SIP at IETF 74 -- =
SIP/SIPPING Continuing =
Work</caption><tbody><tr><th>Victim</th><th>Topic</th><th>Drafts</th><th>T=
ime</th></tr><tr><td>Gonzalo Camarillo</td><td>Re-INVITE Handling in SIP =
&amp; Media State under Preconditions in SIP</td><td><a =
href=3D"http://tools.ietf.org/id/draft-camarillo-sipping-reinvite-00.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-camarillo-sipping-reinvite-00.txt"=
 rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-camarillo-sipping-reinvite-00</a><br><div style=3D"margin-top: =
0.4em; margin-right: 0px; margin-bottom: 0.5em; margin-left: 0px; =
line-height: 1.5em; "><a =
href=3D"http://tools.ietf.org/id/draft-camarillo-sipping-precons-00.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-camarillo-sipping-precons-00.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-camarillo-sipping-precons-00</a></div></td><td>15</td></tr><tr><td=
>Christer Holmberg</td><td>Re-Invite Fallback (based on ML =
discussion)</td><td><a =
href=3D"http://tools.ietf.org/id/draft-ietf-sipping-sip-offeranswer-10.txt=
" class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-ietf-sipping-sip-offeranswer-10.tx=
t" rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-ietf-sipping-sip-offeranswer-10</a></td><td>15</td></tr><tr><td>Er=
ic Burger</td><td>Info Events</td><td><a =
href=3D"http://tools.ietf.org/html/draft-ietf-sip-info-events-03" =
class=3D"external text" =
title=3D"http://tools.ietf.org/html/draft-ietf-sip-info-events-03" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-ietf-sip-info-events-03</a></td><td>15</td></tr><tr><td>Hadriel =
Kaplan</td><td>Identity</td><td><a =
href=3D"http://tools.ietf.org/id/draft-elwell-sip-e2e-identity-important-0=
3.txt" class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-elwell-sip-e2e-identity-important-=
03.txt" rel=3D"nofollow" style=3D"text-decoration: none; =
background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-elwell-sip-e2e-identity-important-03</a></td><td>30</td></tr><tr><=
td>Dale Worley</td><td>Profile Datasets</td><td><a =
href=3D"http://tools.ietf.org/id/draft-ietf-sipping-profile-datasets-02.tx=
t" class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-ietf-sipping-profile-datasets-02.t=
xt" rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-ietf-sipping-profile-datasets-02</a></td><td>15</td></tr><tr><td>V=
olker Hilt</td><td>SIP Overload Control</td><td><a =
href=3D"http://tools.ietf.org/id/draft-ietf-sipping-overload-design-02.txt=
" class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-ietf-sipping-overload-design-02.tx=
t" rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-ietf-sipping-overload-design-02</a></td><td>15</td></tr><tr><td>Sa=
lvatore Loreto</td><td>Event Throttling</td><td><a =
href=3D"http://tools.ietf.org/id/draft-niemi-sipping-event-throttle-08.txt=
" class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-niemi-sipping-event-throttle-08.tx=
t" rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-niemi-sipping-event-throttle-08</a></td><td>15</td></tr><tr><td>Jo=
n Peterson</td><td>SIP Change Process Intro (Discussion on =
Friday)</td><td><a =
href=3D"http://tools.ietf.org/id/draft-peterson-rai-rfc3427bis-01.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-peterson-rai-rfc3427bis-01.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-peterson-rai-rfc3427bis-01</a></td><td>30</td></tr><tr><td>Placeho=
lder</td><td>No Agenda Time</td><td><a =
href=3D"http://tools.ietf.org/id/draft-ietf-sip-outbound-16.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-ietf-sip-outbound-16.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-ietf-sip-outbound-16</a><br><div style=3D"margin-top: 0.4em; =
margin-right: 0px; margin-bottom: 0.5em; margin-left: 0px; line-height: =
1.5em; "><a =
href=3D"http://tools.ietf.org/id/draft-ietf-sip-connect-reuse-12.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-ietf-sip-connect-reuse-12.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-ietf-sip-connect-reuse-12</a><br><a =
href=3D"http://tools.ietf.org/id/draft-ietf-sipping-sip-offeranswer-10.txt=
" class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-ietf-sipping-sip-offeranswer-10.tx=
t" rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-ietf-sipping-sip-offeranswer-10</a><br><a =
href=3D"http://tools.ietf.org/id/draft-ietf-sip-certs-07.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-ietf-sip-certs-07.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; ">draft-ietf-sip-certs-07</a><br><a =
href=3D"http://tools.ietf.org/id/draft-ietf-sip-dtls-srtp-framework-06.txt=
" class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-ietf-sip-dtls-srtp-framework-06.tx=
t" rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-ietf-sip-dtls-srtp-framework-06</a><br><a =
href=3D"http://tools.ietf.org/html/draft-ietf-sip-session-policy-framework=
-05" class=3D"external text" =
title=3D"http://tools.ietf.org/html/draft-ietf-sip-session-policy-framewor=
k-05" rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-ietf-sip-session-policy-framework-05</a><br><a =
href=3D"http://tools.ietf.org/id/draft-ietf-sip-xcapevent-04.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-ietf-sip-xcapevent-04.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-ietf-sip-xcapevent-04</a><br><a =
href=3D"http://tools.ietf.org/id/draft-ietf-sip-location-conveyance-12.txt=
" class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-ietf-sip-location-conveyance-12.tx=
t" rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-ietf-sip-location-conveyance-12</a><br><a =
href=3D"http://tools.ietf.org/id/draft-ietf-sip-body-handling-05.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-ietf-sip-body-handling-05.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-ietf-sip-body-handling-05</a><br><a =
href=3D"http://tools.ietf.org/id/draft-ietf-sip-domain-certs-02.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-ietf-sip-domain-certs-02.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-ietf-sip-domain-certs-02</a><br><a =
href=3D"http://tools.ietf.org/id/draft-ietf-sip-eku-03.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-ietf-sip-eku-03.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; ">draft-ietf-sip-eku-03</a><br><a =
href=3D"http://tools.ietf.org/id/draft-ietf-sip-record-route-fix-05.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-ietf-sip-record-route-fix-05.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-ietf-sip-record-route-fix-05</a><br><a =
href=3D"http://tools.ietf.org/id/draft-ietf-sip-saml-05.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-ietf-sip-saml-05.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; ">draft-ietf-sip-saml-05</a><br><a =
href=3D"http://tools.ietf.org/id/draft-ietf-sip-ua-privacy-05.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-ietf-sip-ua-privacy-05.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-ietf-sip-ua-privacy-05</a><br><a =
href=3D"http://tools.ietf.org/id/draft-rosenberg-sip-info-litmus-02.txt" =
class=3D"external text" =
title=3D"http://tools.ietf.org/id/draft-rosenberg-sip-info-litmus-02.txt" =
rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-rosenberg-sip-info-litmus-02</a><br><a =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-ua-privacy-06.t=
xt" class=3D"external text" =
title=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-ua-privacy-06.=
txt" rel=3D"nofollow" style=3D"text-decoration: none; background-image: =
url(http://www.softarmor.com/mediawiki/skins/monobook/external.png); =
background-repeat: no-repeat; background-attachment: initial; =
-webkit-background-clip: initial; -webkit-background-origin: initial; =
background-color: initial; padding-right: 13px; color: rgb(51, 102, =
187); background-position: 100% 50%; =
">draft-ietf-sip-ua-privacy-06</a></div></td><td>0</td></tr></tbody></tabl=
e></span></div><div><br></div><div><br></div></body></html>=

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Subject: [Sip] Pub request for draft-ietf-sip-body-handling-06
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We have requested publication of draft-ietf-sip-body-handling-06.

I'd like to thank Theo Zourzouvillys for volunteering to serve as the  
draft shepherd. This represents a departure from prior practice, in  
that we've traditionally had the chairs act as shepherds. I recently  
realized that sharing the load is a good way to increase our pool of  
administratively-trained folks, and get the job done faster.

The pub-request writeup, prepared by Theo, follows.

--
Dean


-----------

The SIP working group would like to request publication of  draft-ietf- 
sip-body-handling-06. This is a product of the SIP working group and  
is intended for standards track.

Theo Zourzouvillys <theo@voip.co.uk> will serve as the draft shepherd,  
supported by SIP working group chair Dean Willis.

A write-up, prepared by Theo, is attached. Note that there is an  
Instruction to the RFC Editor attached to correct two minor typos in  
XML and example SIP messages that prevent validation.

[This template version is dated September 17, 2008.]

(for draft-ietf-sip-body-handling-06)

    (1.a) Who is the Document Shepherd for this document? Has the
          Document Shepherd personally reviewed this version of the
          document and, in particular, does he or she believe this
          version is ready for forwarding to the IESG for publication?

The Document Shepherd for this document is Theo Zourzouvillys.  I have  
read
this version of the draft and believe it is ready for forwarding to  
the IESG
for publication.

    (1.b) Has the document had adequate review both from key WG members
          and from key non-WG members? Does the Document Shepherd have
          any concerns about the depth or breadth of the reviews that
          have been performed?

This document has had solid, in-depth reviews from working group  
members.

    (1.c) Does the Document Shepherd have concerns that the document
          needs more review from a particular or broader perspective,
          e.g., security, operational complexity, someone familiar with
          AAA, internationalization or XML?

There are no concerns.

    (1.d) Does the Document Shepherd have any specific concerns or
          issues with this document that the Responsible Area Director
          and/or the IESG should be aware of? For example, perhaps he
          or she is uncomfortable with certain parts of the document, or
          has concerns whether there really is a need for it. In any
          event, if the WG has discussed those issues and has indicated
          that it still wishes to advance the document, detail those
          concerns here. Has an IPR disclosure related to this document
          been filed? If so, please include a reference to the
          disclosure and summarize the WG discussion and conclusion on
          this issue.

I have no concerns with this document, and believe there is a need for  
it.

There have been no IPR disclosures for this document.

    (1.e) How solid is the WG consensus behind this document? Does it
          represent the strong concurrence of a few individuals, with
          others being silent, or does the WG as a whole understand and
          agree with it?

WG consensus appears to be solid.

    (1.f) Has anyone threatened an appeal or otherwise indicated extreme
          discontent? If so, please summarise the areas of conflict in
          separate email messages to the Responsible Area Director. (It
          should be in a separate email because this questionnaire is
          entered into the ID Tracker.)

No one has expressed any discontent with this document.

    (1.g) Has the Document Shepherd personally verified that the
          document satisfies all ID nits? (See
          http://www.ietf.org/ID-Checklist.html and
          http://tools.ietf.org/tools/idnits/). Boilerplate checks are
          not enough; this check needs to be thorough. Has the document
          met all formal review criteria it needs to, such as the MIB
          Doctor, media type and URI type reviews?

The document shepherd has personally verified that this version of the
document satisfies all of the ID nits, bar a warning about legal
boilerplate about pre-RFC5378 work.

    (1.h) Has the document split its references into normative and
          informative? Are there normative references to documents that
          are not ready for advancement or are otherwise in an unclear
          state? If such normative references exist, what is the
          strategy for their completion? Are there normative references
          that are downward references, as described in [RFC3967]? If
          so, list these downward references to support the Area
          Director in the Last Call procedure for them [RFC3967].

The document has been split into normative and informative references.

All normative references are stable.

    (1.i) Has the Document Shepherd verified that the document IANA
          consideration section exists and is consistent with the body
          of the document? If the document specifies protocol
          extensions, are reservations requested in appropriate IANA
          registries? Are the IANA registries clearly identified? If
          the document creates a new registry, does it define the
          proposed initial contents of the registry and an allocation
          procedure for future registrations? Does it suggest a
          reasonable name for the new registry? See [RFC5226]. If the
          document describes an Expert Review process has Shepherd
          conferred with the Responsible Area Director so that the IESG
          can appoint the needed Expert during the IESG Evaluation?

This document contains two considerations for IANA. in the IANA  
considerations
section (section 12):

  * Registration of the 'by-reference' Disposition Type in the  
existing IANA
    registry for Content-Disposition 'Handling Parameter Values',  
created
    by RFC3204 at http://www.iana.org/assignments/mail-cont-disp

  * request an update to the 'handling' parameter in the existing 'SIP  
Header
    Field Parameters and Parameter Values' registry created by  
RFC3968, at
    http://www.iana.org/assignments/sip-parameters

Neither of these registries require Expert Review for additions or
modifications.


    (1.j) Has the Document Shepherd verified that sections of the
          document that are written in a formal language, such as XML
          code, BNF rules, MIB definitions, etc., validate correctly in
          an automated checker?

The XML in figure 2 does not validate due to a missing closing XML '>'  
on
the 'resource-lists' element (line 250), and the Content-Length on  
line 230
is also incorrect.   Please see [Instructions for RFC Editor] for  
instructions
to correct this issue.

SIP messages and SDP bodies in examples have been manually checked and  
appear
valid.

    (1.k) The IESG approval announcement includes a Document
          Announcement Write-Up. Please provide such a Document
          Announcement Write-Up? Recent examples can be found in the
          "Action" announcements for approved documents. The approval
          announcement contains the following sections:

          Technical Summary
             Relevant content can frequently be found in the abstract
             and/or introduction of the document. If not, this may be
             an indication that there are deficiencies in the abstract
             or introduction.


   This document specifies how message bodies should be handled in SIP.
   Additionally, this document specifies SIP user agent support for MIME
   (Multipurpose Internet Mail Extensions) in message bodies.


          Working Group Summary
             Was there anything in WG process that is worth noting? For
             example, was there controversy about particular points or
             were there decisions where the consensus was particularly
             rough?


There is consensus in the working group to publish this document, and is
targeted for Standards Track.  Work on this document began in May 2007,
and was adopted as a working group item in August 2007. WGLC was issued
in June 2008.


          Document Quality
             Are there existing implementations of the protocol? Have a
             significant number of vendors indicated their plan to
             implement the specification? Are there any reviewers that
             merit special mention as having done a thorough review,
             e.g., one that resulted in important changes or a
             conclusion that the document had no substantive issues? If
             there was a MIB Doctor, Media Type or other expert review,
             what was its course (briefly)? In the case of a Media Type
             review, on what date was the request posted?

This document specifically addresses an area of SIP that has been an
interoperability problem in the past.  The SIPit interoperability  
events have
seen many problems in the area of interoperability of MIME handling.

This document has been reviews by many participants over the lifetime  
of the
document, by the following members of the WG:

  - Paul Kyzivat
  - John Elwell
  - Francois Audet
  - Dan Wing
  - Eric Burger
  - Dale Worley
  - Jonathan Rosenberg
  - Cullen Jennings
  - Adam Roach

Additionally, an extensive APPS area review of the document was been  
performed
by Dave Crocker in an eary version of the this document.






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We have submitted the updated target-uri draft based on the comments
submitted to the list and comments received at IETF73.

I have taken over as editor as Jonathan didn't have the cycles to
update the draft, with Francois, Christer and Hans Erick as additional
co-authors and great deal of help from Mary.

The following summarizes the changes made to the target-uri document
1. Added use-case for toll-free number back
2. Added definition of "retarget" operation.
3. Removed a reference to URN
4. Added a text discussing the difference to P-Called-Party-Id
5. Changed parameter name from "target" to "istarget"

Note, that the target-uri document still contains the normative text  
for the
History-Info header.

In addition, Mary (with Francois as co-author) has submitted a
rfc4244bis, with the following changes:
1. Incorporated the normative aspects of the target-uri document
into the existing normative text in RFC 4244 - the functionality is
virtually identical (as is some of the text) as the HI based solution
described in the target-uri document.  It's important that the solution
be integrated into RFC 4244 as it MUST work and be based on the  
normative
aspects of RFC 4244.
2.  Added the use cases from target-uri the the summary
in the overview of rfc4244bis.
3. Added an additional requirement to capture the "target-uri"  
information.
4. Fixed an error in the RFC 4244 ABNF and added "retarget" parameter.
5. Added a more simplified example.


We had some very long offline exchanges as to the best way forward and
remaining work for both documents.

Some of the issues identified are:

::Issues::
    1. Should we remove the normative text from target-uri and progress
        4244bis along with the target-uri document to meet the chartered
       SIP WG milestone?


    2. Name of the parameter.
       At the last meeting, parameter "target" was said inappropriate
       because voicemail-uri spec already defines a parameter called
       "target" which also can be found in hi-entry, thus potentially
       causing confusion.

       Currently the target-uri draft uses "istarget" and 4244bis uses
       "retarget"  but we could never come to
       a consensus on what name is appropriate.  Other suggestions have
       included the following:
       "target-identity" (someone didn't like that "identity" is also  
a SIP header)
       "reg-uri"  (can be paired with "mapped-uri" for item 3 below)
       "aor"
       "jibberish"
       etc.

       One reason this is so difficult relates to the problem  
statement in target-uri in that
       RFC 3261 doesn't differentiate the mechanism by which the new
       (target) Request-URI is selected.  Another issue is that some  
of the terminology in
       RFC 3261 is overloaded - e.g., "forwarding" refers both to a  
Proxy
       which does not have responsibility for the domain of the  
request-URI
       in the incoming request, thus the proxy just "forwards" the  
request to
       the next hop AND "forwarding" is used to describe the process  
whereby
       the outgoing request is built and "forwarded" to the next hop  
at which
       point the proxy does not know how the new request-uri was  
selected.
       RFC 4244 has attempted to clarify the terms and attempts to use  
"forward"
       in the context of the former situation and "retarget" for the  
case whereby
       a proxy is responsible for the domain and thus can use a number  
of
       mechanism to select the new target for the request - e.g., a  
REGISTRAR,
       configured data, etc.

  3.  Related to the last point in item 2 above, it has been proposed  
that
       we differentiate the hi-entries even more by defining separate  
parameters
      for registered and configured/mapped contacts.
      Currently when the R-URI is translated to a URI which is either  
derived
      from location service lookup(registered by UA) or from mapping
      table, there is no differentiation as to how the URI was derived  
once it is
      added to the list of potential targets.

      The general consensus of the authors of the two documents was  
that it may
      be useful for some services to have the hi-entries tagged with the
      more specific information.

      And, of course, this gets us into another naming contest. In the  
end, the naming
      is not so important as long as the term isn't too overloaded and  
it is defined
      precisely in the document(s).

We would appreciate WG feedback on these issues and any other comments  
on
the two documents prior to IETF-74.

Regards,
Shida and Mary.



Begin forwarded message:
> From: Internet-Drafts@ietf.org
> Date: March 10, 2009 2:30:01 AM JST
> To: i-d-announce@ietf.org
> Subject: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
> Reply-To: internet-drafts@ietf.org
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>
>
> 	Title		: Delivery of Request-URI Targets to User Agents
> 	Author(s)	: J. Rosenberg, H. van Elburg, C. Holmberg, F. Audet, S.  
> Schubert
> 	Filename	: draft-rosenberg-sip-target-uri-delivery-01.txt
> 	Pages		: 16
> 	Date		: 2009-3-9
> 	
> When a Session Initiation Protocol (SIP) proxy receives a request
>   targeted at a URI identifying a user or resource it is responsible
>   for, the proxy translates the URI to a registered or configured
>   contact URI of an agent representing that user or resource.  In the
>   process, the original URI is removed from the request.  Numerous use
>   cases have arisen which require this information to be delivered to
>   the user agent.  This document describes these use cases and defines
>   an extension to the History-Info header field which allows it to be
>   used to support those cases.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri-delivery-01.txt
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
Content-Type: text/plain<BR>Content-ID: &lt;2009-3-9102756.I- 
D@ietf.org&gt;<BR><BR>
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt

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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><div><br></div><div>We have =
submitted the updated target-uri draft based on the =
comments&nbsp;</div><div>submitted to the list and comments received at =
IETF73.&nbsp;</div><div><br></div><div>I have taken over as editor as =
Jonathan didn't have the cycles to&nbsp;</div><div>update the draft, =
with Francois, Christer and Hans Erick as =
additional&nbsp;</div><div>co-authors and great deal of help from Mary. =
&nbsp;</div><div>&nbsp;</div><div>The following summarizes the changes =
made to the target-uri document</div><div>1. Added use-case for =
toll-free number back</div><div>2. Added definition of "retarget" =
operation.&nbsp;</div><div>3. Removed a reference to URN</div><div>4. =
Added a text discussing the difference to P-Called-Party-Id</div><div>5. =
Changed parameter name from "target" to =
"istarget"</div><div>&nbsp;</div><div>Note, that the target-uri document =
still contains the normative text for the</div><div>History-Info =
header.</div><div>&nbsp;</div><div>In addition, Mary (with Francois as =
co-author) has submitted a</div><div>rfc4244bis, with the following =
changes:</div><div>1. Incorporated the normative aspects of the =
target-uri document</div><div>into the existing normative text in RFC =
4244 - the functionality is&nbsp;</div><div>virtually identical (as is =
some of the text) as the HI based solution&nbsp;</div><div>described in =
the target-uri document. &nbsp;It's important that the =
solution</div><div>be integrated into RFC 4244 as it MUST work and be =
based on the normative</div><div>aspects of RFC 4244.</div><div>2. =
&nbsp;Added the use cases from target-uri the the summary</div><div>in =
the overview of rfc4244bis.</div><div>3. Added an additional requirement =
to capture the "target-uri" information.</div><div>4. Fixed an error in =
the RFC 4244 ABNF and added "retarget" parameter.</div><div>5. Added a =
more simplified example.</div><div>&nbsp;</div><div>&nbsp;</div><div>We =
had some very long offline exchanges as to the best way forward =
and</div><div>remaining work for both =
documents.</div><div>&nbsp;</div><div>Some of the issues identified =
are:</div><div><br></div><div>::Issues::</div><div>&nbsp;&nbsp; 1. =
Should we remove the normative text from target-uri and =
progress</div><div>&nbsp;&nbsp; &nbsp; &nbsp; 4244bis along with the =
target-uri document to meet the chartered</div><div>&nbsp;&nbsp; &nbsp; =
&nbsp;SIP WG =
milestone?&nbsp;</div><div>&nbsp;</div><div>&nbsp;</div><div>&nbsp;&nbsp; =
2. Name of the parameter.&nbsp;</div><div>&nbsp;&nbsp; &nbsp; &nbsp;At =
the last meeting, parameter "target" was said =
inappropriate&nbsp;</div><div>&nbsp;&nbsp; &nbsp; &nbsp;because =
voicemail-uri spec already defines a parameter =
called&nbsp;</div><div>&nbsp;&nbsp; &nbsp; &nbsp;"target" which also can =
be found in hi-entry, thus potentially&nbsp;</div><div>&nbsp;&nbsp; =
&nbsp; &nbsp;causing confusion.</div><div>&nbsp;&nbsp; =
&nbsp;&nbsp;</div><div>&nbsp;&nbsp; &nbsp; &nbsp;Currently the =
target-uri draft uses "istarget" and 4244bis uses</div><div>&nbsp;&nbsp; =
&nbsp; &nbsp;"retarget" &nbsp;but we could never come =
to&nbsp;</div><div>&nbsp;&nbsp; &nbsp; &nbsp;a consensus on what name is =
appropriate. &nbsp;Other suggestions have</div><div>&nbsp;&nbsp; &nbsp; =
&nbsp;included the following:</div><div>&nbsp;&nbsp; &nbsp; =
&nbsp;"target-identity" (someone didn't like that "identity" is also a =
SIP header)</div><div>&nbsp;&nbsp; &nbsp; &nbsp;"reg-uri" &nbsp;(can be =
paired with "mapped-uri" for item 3 below)</div><div>&nbsp;&nbsp; &nbsp; =
&nbsp;"aor"</div><div>&nbsp;&nbsp; &nbsp; =
&nbsp;"jibberish"&nbsp;</div><div>&nbsp;&nbsp; &nbsp; =
&nbsp;etc.</div><div>&nbsp;</div><div>&nbsp;&nbsp; &nbsp; &nbsp;One =
reason this is so difficult relates to the problem statement in =
target-uri in that</div><div>&nbsp;&nbsp; &nbsp; &nbsp;RFC 3261 doesn't =
differentiate the mechanism by which the new</div><div>&nbsp;&nbsp; =
&nbsp; &nbsp;(target) Request-URI is selected. &nbsp;Another issue is =
that some of the terminology in</div><div>&nbsp;&nbsp; &nbsp; &nbsp;RFC =
3261 is overloaded - e.g., "forwarding" refers both to a =
Proxy</div><div>&nbsp;&nbsp; &nbsp; &nbsp;which does not have =
responsibility for the domain of the request-URI</div><div>&nbsp;&nbsp; =
&nbsp; &nbsp;in the incoming request, thus the proxy just "forwards" the =
request to</div><div>&nbsp;&nbsp; &nbsp; &nbsp;the next hop AND =
"forwarding" is used to describe the process =
whereby</div><div>&nbsp;&nbsp; &nbsp; &nbsp;the outgoing request is =
built and "forwarded" to the next hop at which</div><div>&nbsp;&nbsp; =
&nbsp; &nbsp;point the proxy does not know how the new request-uri was =
selected.</div><div>&nbsp;&nbsp; &nbsp; &nbsp;RFC 4244 has attempted to =
clarify the terms and attempts to use =
"forward"&nbsp;</div><div>&nbsp;&nbsp; &nbsp; &nbsp;in the context of =
the former situation and "retarget" for the case =
whereby</div><div>&nbsp;&nbsp; &nbsp; &nbsp;a proxy is responsible for =
the domain and thus can use a number of&nbsp;</div><div>&nbsp;&nbsp; =
&nbsp; &nbsp;mechanism to select the new target for the request - e.g., =
a REGISTRAR,&nbsp;</div><div>&nbsp;&nbsp; &nbsp; &nbsp;configured data, =
etc. &nbsp;</div><div><br></div><div>&nbsp;3. &nbsp;Related to the last =
point in item 2 above, it has been proposed that</div><div>&nbsp;&nbsp; =
&nbsp; &nbsp;we differentiate the hi-entries even more by defining =
separate parameters</div><div>&nbsp;&nbsp; &nbsp; for registered and =
configured/mapped contacts.&nbsp;</div><div>&nbsp;&nbsp; &nbsp; =
Currently when the R-URI is translated to a URI which is either =
derived&nbsp;</div><div>&nbsp;&nbsp; &nbsp; from location service =
lookup(registered by UA) or from mapping&nbsp;</div><div>&nbsp;&nbsp; =
&nbsp; table, there is no differentiation as to how the URI was derived =
once it is</div><div>&nbsp;&nbsp; &nbsp; added to the list of potential =
targets.</div><div>&nbsp;&nbsp; &nbsp;&nbsp;</div><div>&nbsp;&nbsp; =
&nbsp; The general consensus of the authors of the two documents was =
that it may&nbsp;</div><div>&nbsp;&nbsp; &nbsp; be useful for some =
services to have the hi-entries tagged with the</div><div>&nbsp;&nbsp; =
&nbsp; more specific =
information.&nbsp;</div><div>&nbsp;</div><div>&nbsp;&nbsp; &nbsp; And, =
of course, this gets us into another naming contest. In the end, the =
naming</div><div>&nbsp;&nbsp; &nbsp; is not so important as long as the =
term isn't too overloaded and it is defined</div><div>&nbsp;&nbsp; =
&nbsp; precisely in the document(s).</div><div>&nbsp;</div><div>We would =
appreciate WG feedback on these issues and any other comments =
on</div><div>the two documents prior to =
IETF-74.</div><div>&nbsp;</div><div>Regards,</div><div>Shida and =
Mary.</div><div><br></div><div><br></div><div><div><div><div><br =
class=3D"Apple-interchange-newline">Begin forwarded =
message:</div><blockquote type=3D"cite"><div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><font =
face=3D"Helvetica" size=3D"3" color=3D"#000000" style=3D"font: normal =
normal normal 12px/normal Helvetica; color: rgb(0, 0, 0); =
"><b>From:&nbsp;</b></font><font face=3D"Helvetica" size=3D"3" =
style=3D"font: normal normal normal 12px/normal Helvetica; "><a =
href=3D"mailto:Internet-Drafts@ietf.org">Internet-Drafts@ietf.org</a></fon=
t></div><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: =
0px; margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" =
color=3D"#000000" style=3D"font: normal normal normal 12px/normal =
Helvetica; color: rgb(0, 0, 0); "><b>Date:&nbsp;</b></font><font =
face=3D"Helvetica" size=3D"3" style=3D"font: normal normal normal =
12px/normal Helvetica; ">March 10, 2009 2:30:01 AM JST</font></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: normal normal normal 12px/normal Helvetica; color: rgb(0, =
0, 0); "><b>To:&nbsp;</b></font><font face=3D"Helvetica" size=3D"3" =
style=3D"font: normal normal normal 12px/normal Helvetica; "><a =
href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a></font></di=
v><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: normal normal normal 12px/normal Helvetica; color: rgb(0, =
0, 0); "><b>Subject:&nbsp;</b></font><font face=3D"Helvetica" size=3D"3" =
style=3D"font: normal normal normal 12px/normal Helvetica; "><b>I-D =
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt&nbsp;</b></font></di=
v><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: normal normal normal 12px/normal Helvetica; color: rgb(0, =
0, 0); "><b>Reply-To:&nbsp;</b></font><font face=3D"Helvetica" size=3D"3" =
style=3D"font: normal normal normal 12px/normal Helvetica; "><a =
href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a></fon=
t></div><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: =
0px; margin-left: 0px; min-height: 14px; "><br></div></div><div>A New =
Internet-Draft is available from the on-line =
Internet-Drafts&nbsp;<br>directories.<br><br><br><span =
class=3D"Apple-tab-span" style=3D"white-space: pre; ">	=
</span>Title<span class=3D"Apple-tab-span" style=3D"white-space: pre; ">	=
</span><span class=3D"Apple-tab-span" style=3D"white-space: pre; ">	=
</span>: Delivery of Request-URI Targets to User Agents<br><span =
class=3D"Apple-tab-span" style=3D"white-space: pre; ">	=
</span>Author(s)<span class=3D"Apple-tab-span" style=3D"white-space: =
pre; ">	</span>: J. Rosenberg, H. van Elburg, C. Holmberg, F. Audet, S. =
Schubert<br><span class=3D"Apple-tab-span" style=3D"white-space: pre; ">	=
</span>Filename<span class=3D"Apple-tab-span" style=3D"white-space: pre; =
">	</span>: draft-rosenberg-sip-target-uri-delivery-01.txt<br><span =
class=3D"Apple-tab-span" style=3D"white-space: pre; ">	=
</span>Pages<span class=3D"Apple-tab-span" style=3D"white-space: pre; ">	=
</span><span class=3D"Apple-tab-span" style=3D"white-space: pre; ">	=
</span>: 16<br><span class=3D"Apple-tab-span" style=3D"white-space: pre; =
">	</span>Date<span class=3D"Apple-tab-span" style=3D"white-space: =
pre; ">	</span><span class=3D"Apple-tab-span" style=3D"white-space: pre; =
">	</span>: 2009-3-9<br><span class=3D"Apple-tab-span" =
style=3D"white-space: pre; ">	</span><br>When a Session Initiation =
Protocol (SIP) proxy receives a request<br>&nbsp;&nbsp;targeted at a URI =
identifying a user or resource it is responsible<br>&nbsp;&nbsp;for, the =
proxy translates the URI to a registered or =
configured<br>&nbsp;&nbsp;contact URI of an agent representing that user =
or resource. &nbsp;In the<br>&nbsp;&nbsp;process, the original URI is =
removed from the request. &nbsp;Numerous use<br>&nbsp;&nbsp;cases have =
arisen which require this information to be delivered =
to<br>&nbsp;&nbsp;the user agent. &nbsp;This document describes these =
use cases and defines<br>&nbsp;&nbsp;an extension to the History-Info =
header field which allows it to be<br>&nbsp;&nbsp;used to support those =
cases.<br><br>A URL for this Internet-Draft is:<br><a =
href=3D"http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri=
-delivery-01.txt">http://www.ietf.org/internet-drafts/draft-rosenberg-sip-=
target-uri-delivery-01.txt</a><br><br>Internet-Drafts are also available =
by anonymous FTP at:<br><a =
href=3D"ftp://ftp.ietf.org/internet-drafts/">ftp://ftp.ietf.org/internet-d=
rafts/</a><br><br>Below is the data which will enable a MIME compliant =
mail reader<br>implementation to automatically retrieve the ASCII =
version of =
the<br>Internet-Draft.<br></div></blockquote></div></div>Content-Type: =
text/plain&lt;BR>Content-ID: &amp;lt;<a =
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>&amp;gt;&lt;BR>&lt;BR><div><div><blockquote =
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From dean.willis@softarmor.com  Tue Mar 10 23:37:03 2009
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On Mar 11, 2009, at 1:13 AM, Shida Schubert wrote:

>
>       One reason this is so difficult relates to the problem  
> statement in target-uri in that
>       RFC 3261 doesn't differentiate the mechanism by which the new
>       (target) Request-URI is selected.  Another issue is that some  
> of the terminology in
>       RFC 3261 is overloaded - e.g., "forwarding" refers both to a  
> Proxy
>       which does not have responsibility for the domain of the  
> request-URI
>       in the incoming request, thus the proxy just "forwards" the  
> request to
>       the next hop AND "forwarding" is used to describe the process  
> whereby
>       the outgoing request is built and "forwarded" to the next hop  
> at which
>       point the proxy does not know how the new request-uri was  
> selected.
>       RFC 4244 has attempted to clarify the terms and attempts to  
> use "forward"
>       in the context of the former situation and "retarget" for the  
> case whereby
>       a proxy is responsible for the domain and thus can use a  
> number of
>       mechanism to select the new target for the request - e.g., a  
> REGISTRAR,
>       configured data, etc.
>

Thanks, Shida.

I personally would not be averse to an Essential Correction that  
corrects the terminology in RFC 3261, but I suspect that will make  
many heads spin around elsewhere in the WG.

--
Dean

From dwillis@peerance.com  Tue Mar 10 23:49:04 2009
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--000e0cd2176c8ab2e70464d245f5
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I've been having difficulties impersonating myself with my backup gmail
account. Maybe this will work, maybe not.

--
Dean

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<br>I&#39;ve been having difficulties impersonating myself with my backup gmail account. Maybe this will work, maybe not.<br><br>--<br>Dean<br>

--000e0cd2176c8ab2e70464d245f5--

From HKaplan@acmepacket.com  Tue Mar 10 23:54:55 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Shida Schubert <shida@ntt-at.com>, "sip@ietf.org List" <sip@ietf.org>
Date: Wed, 11 Mar 2009 02:55:28 -0400
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At the risk of opening up a topic we have already agreed on, but in the spi=
rit of open communication... [how's that for an opener!?]

Are we absolutely sure we want to put this target-URI in the History-Info h=
eader?

I am not asking this because I think it's hard.  I am asking because I am n=
ot sure it will be used.  Several of us from this same WG are involved in t=
he SIP-Forum's SIP-Connect profile, and when faced with either using this n=
ew History-Info mechanism for a target-uri delivery issue, vs. not, we chos=
e not to.  And we were mostly the same people who liked the idea of putting=
 it in History-Info in the IETF! (myself included)  But I am concerned that=
 when given a reasonable opportunity to use a mechanism we ourselves promot=
e when wearing a different hat, we chose not to use it in practice.  It doe=
sn't bode well. :(

-hadriel
p.s. sorry Mary for raising this, but there isn't much email traffic anyway=
. :)

________________________________________
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Shida=
 Schubert
Sent: Wednesday, March 11, 2009 2:13 AM
To: sip@ietf.org List

We have submitted the updated target-uri draft based on the comments=A0
submitted to the list and comments received at IETF73.=A0

I have taken over as editor as Jonathan didn't have the cycles to=A0
update the draft, with Francois, Christer and Hans Erick as additional=A0
co-authors and great deal of help from Mary. =A0
=A0
The following summarizes the changes made to the target-uri document
1. Added use-case for toll-free number back
2. Added definition of "retarget" operation.=A0
3. Removed a reference to URN
4. Added a text discussing the difference to P-Called-Party-Id
5. Changed parameter name from "target" to "istarget"
=A0
Note, that the target-uri document still contains the normative text for th=
e
History-Info header.
=A0
In addition, Mary (with Francois as co-author) has submitted a
rfc4244bis, with the following changes:
1. Incorporated the normative aspects of the target-uri document
into the existing normative text in RFC 4244 - the functionality is=A0
virtually identical (as is some of the text) as the HI based solution=A0
described in the target-uri document. =A0It's important that the solution
be integrated into RFC 4244 as it MUST work and be based on the normative
aspects of RFC 4244.
2. =A0Added the use cases from target-uri the the summary
in the overview of rfc4244bis.
3. Added an additional requirement to capture the "target-uri" information.
4. Fixed an error in the RFC 4244 ABNF and added "retarget" parameter.
5. Added a more simplified example.
=A0
=A0
We had some very long offline exchanges as to the best way forward and
remaining work for both documents.
=A0
Some of the issues identified are:

::Issues::
=A0=A0 1. Should we remove the normative text from target-uri and progress
=A0=A0 =A0 =A0 4244bis along with the target-uri document to meet the chart=
ered
=A0=A0 =A0 =A0SIP WG milestone?=A0
=A0
=A0
=A0=A0 2. Name of the parameter.=A0
=A0=A0 =A0 =A0At the last meeting, parameter "target" was said inappropriat=
e=A0
=A0=A0 =A0 =A0because voicemail-uri spec already defines a parameter called=
=A0
=A0=A0 =A0 =A0"target" which also can be found in hi-entry, thus potentiall=
y=A0
=A0=A0 =A0 =A0causing confusion.
=A0=A0 =A0=A0
=A0=A0 =A0 =A0Currently the target-uri draft uses "istarget" and 4244bis us=
es
=A0=A0 =A0 =A0"retarget" =A0but we could never come to=A0
=A0=A0 =A0 =A0a consensus on what name is appropriate. =A0Other suggestions=
 have
=A0=A0 =A0 =A0included the following:
=A0=A0 =A0 =A0"target-identity" (someone didn't like that "identity" is als=
o a SIP header)
=A0=A0 =A0 =A0"reg-uri" =A0(can be paired with "mapped-uri" for item 3 belo=
w)
=A0=A0 =A0 =A0"aor"
=A0=A0 =A0 =A0"jibberish"=A0
=A0=A0 =A0 =A0etc.
=A0
=A0=A0 =A0 =A0One reason this is so difficult relates to the problem statem=
ent in target-uri in that
=A0=A0 =A0 =A0RFC 3261 doesn't differentiate the mechanism by which the new
=A0=A0 =A0 =A0(target) Request-URI is selected. =A0Another issue is that so=
me of the terminology in
=A0=A0 =A0 =A0RFC 3261 is overloaded - e.g., "forwarding" refers both to a =
Proxy
=A0=A0 =A0 =A0which does not have responsibility for the domain of the requ=
est-URI
=A0=A0 =A0 =A0in the incoming request, thus the proxy just "forwards" the r=
equest to
=A0=A0 =A0 =A0the next hop AND "forwarding" is used to describe the process=
 whereby
=A0=A0 =A0 =A0the outgoing request is built and "forwarded" to the next hop=
 at which
=A0=A0 =A0 =A0point the proxy does not know how the new request-uri was sel=
ected.
=A0=A0 =A0 =A0RFC 4244 has attempted to clarify the terms and attempts to u=
se "forward"=A0
=A0=A0 =A0 =A0in the context of the former situation and "retarget" for the=
 case whereby
=A0=A0 =A0 =A0a proxy is responsible for the domain and thus can use a numb=
er of=A0
=A0=A0 =A0 =A0mechanism to select the new target for the request - e.g., a =
REGISTRAR,=A0
=A0=A0 =A0 =A0configured data, etc. =A0

=A03. =A0Related to the last point in item 2 above, it has been proposed th=
at
=A0=A0 =A0 =A0we differentiate the hi-entries even more by defining separat=
e parameters
=A0=A0 =A0 for registered and configured/mapped contacts.=A0
=A0=A0 =A0 Currently when the R-URI is translated to a URI which is either =
derived=A0
=A0=A0 =A0 from location service lookup(registered by UA) or from mapping=
=A0
=A0=A0 =A0 table, there is no differentiation as to how the URI was derived=
 once it is
=A0=A0 =A0 added to the list of potential targets.
=A0=A0 =A0=A0
=A0=A0 =A0 The general consensus of the authors of the two documents was th=
at it may=A0
=A0=A0 =A0 be useful for some services to have the hi-entries tagged with t=
he
=A0=A0 =A0 more specific information.=A0
=A0
=A0=A0 =A0 And, of course, this gets us into another naming contest. In the=
 end, the naming
=A0=A0 =A0 is not so important as long as the term isn't too overloaded and=
 it is defined
=A0=A0 =A0 precisely in the document(s).
=A0
We would appreciate WG feedback on these issues and any other comments on
the two documents prior to IETF-74.
=A0
Regards,
Shida and Mary.



Begin forwarded message:
From:=A0Internet-Drafts@ietf.org
Date:=A0March 10, 2009 2:30:01 AM JST
To:=A0i-d-announce@ietf.org
Subject:=A0I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt=A0
Reply-To:=A0internet-drafts@ietf.org

A New Internet-Draft is available from the on-line Internet-Drafts=A0
directories.


	Title		: Delivery of Request-URI Targets to User Agents
	Author(s)	: J. Rosenberg, H. van Elburg, C. Holmberg, F. Audet, S. Schuber=
t
	Filename	: draft-rosenberg-sip-target-uri-delivery-01.txt
	Pages		: 16
	Date		: 2009-3-9
=09
When a Session Initiation Protocol (SIP) proxy receives a request
=A0=A0targeted at a URI identifying a user or resource it is responsible
=A0=A0for, the proxy translates the URI to a registered or configured
=A0=A0contact URI of an agent representing that user or resource. =A0In the
=A0=A0process, the original URI is removed from the request. =A0Numerous us=
e
=A0=A0cases have arisen which require this information to be delivered to
=A0=A0the user agent. =A0This document describes these use cases and define=
s
=A0=A0an extension to the History-Info header field which allows it to be
=A0=A0used to support those cases.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri-delivery=
-01.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.
Content-Type: text/plain<BR>Content-ID: &lt;2009-3-9102756.I-D@ietf.org&gt;=
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_______________________________________________
I-D-Announce mailing list
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https://www.ietf.org/mailman/listinfo/i-d-announce
Internet-Draft directories:=A0http://www.ietf.org/shadow.html
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=A03. =A0Related to the last point in item 2 above, it has been proposed th=
at
=A0=A0 =A0 =A0we differentiate the hi-entries even more by defining separat=
e parameters
=A0=A0 =A0 for registered and configured/mapped contacts.=A0


I vote no on 3, fwiw.  Keep it Simple.

-hadriel

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Date: Wed, 11 Mar 2009 09:36:26 +0100
Message-ID: <9ae56b1e0903110136t2d127012p68a113d650451463@mail.gmail.com>
From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>
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Cc: "sip@ietf.org List" <sip@ietf.org>, Mary Barnes <mary.barnes@nortel.com>
Subject: Re: [Sip] Fwd: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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--0015174c16486e1b7b0464d3c3f1
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I think that is a good question, because it is real world use that we would
like to serve. But at the same time when we want to progress this
requirement, it would be good to know what is an acceptable means to
transport the target-URI towards the UAS?

Would you reconsider using a distinct header for this?

/Hans Erik van Elburg


On Wed, Mar 11, 2009 at 7:55 AM, Hadriel Kaplan <HKaplan@acmepacket.com>wrote:

>
> At the risk of opening up a topic we have already agreed on, but in the
> spirit of open communication... [how's that for an opener!?]
>
> Are we absolutely sure we want to put this target-URI in the History-Info
> header?
>
> I am not asking this because I think it's hard.  I am asking because I am
> not sure it will be used.  Several of us from this same WG are involved in
> the SIP-Forum's SIP-Connect profile, and when faced with either using this
> new History-Info mechanism for a target-uri delivery issue, vs. not, we
> chose not to.  And we were mostly the same people who liked the idea of
> putting it in History-Info in the IETF! (myself included)  But I am
> concerned that when given a reasonable opportunity to use a mechanism we
> ourselves promote when wearing a different hat, we chose not to use it in
> practice.  It doesn't bode well. :(
>
> -hadriel
> p.s. sorry Mary for raising this, but there isn't much email traffic
> anyway. :)
>
> ________________________________________
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Shida Schubert
> Sent: Wednesday, March 11, 2009 2:13 AM
> To: sip@ietf.org List
>
> We have submitted the updated target-uri draft based on the comments
> submitted to the list and comments received at IETF73.
>
> I have taken over as editor as Jonathan didn't have the cycles to
> update the draft, with Francois, Christer and Hans Erick as additional
> co-authors and great deal of help from Mary.
>
> The following summarizes the changes made to the target-uri document
> 1. Added use-case for toll-free number back
> 2. Added definition of "retarget" operation.
> 3. Removed a reference to URN
> 4. Added a text discussing the difference to P-Called-Party-Id
> 5. Changed parameter name from "target" to "istarget"
>
> Note, that the target-uri document still contains the normative text for
> the
> History-Info header.
>
> In addition, Mary (with Francois as co-author) has submitted a
> rfc4244bis, with the following changes:
> 1. Incorporated the normative aspects of the target-uri document
> into the existing normative text in RFC 4244 - the functionality is
> virtually identical (as is some of the text) as the HI based solution
> described in the target-uri document.  It's important that the solution
> be integrated into RFC 4244 as it MUST work and be based on the normative
> aspects of RFC 4244.
> 2.  Added the use cases from target-uri the the summary
> in the overview of rfc4244bis.
> 3. Added an additional requirement to capture the "target-uri" information.
> 4. Fixed an error in the RFC 4244 ABNF and added "retarget" parameter.
> 5. Added a more simplified example.
>
>
> We had some very long offline exchanges as to the best way forward and
> remaining work for both documents.
>
> Some of the issues identified are:
>
> ::Issues::
>    1. Should we remove the normative text from target-uri and progress
>        4244bis along with the target-uri document to meet the chartered
>       SIP WG milestone?
>
>
>    2. Name of the parameter.
>       At the last meeting, parameter "target" was said inappropriate
>       because voicemail-uri spec already defines a parameter called
>       "target" which also can be found in hi-entry, thus potentially
>       causing confusion.
>
>       Currently the target-uri draft uses "istarget" and 4244bis uses
>       "retarget"  but we could never come to
>       a consensus on what name is appropriate.  Other suggestions have
>       included the following:
>       "target-identity" (someone didn't like that "identity" is also a SIP
> header)
>       "reg-uri"  (can be paired with "mapped-uri" for item 3 below)
>       "aor"
>       "jibberish"
>       etc.
>
>       One reason this is so difficult relates to the problem statement in
> target-uri in that
>       RFC 3261 doesn't differentiate the mechanism by which the new
>       (target) Request-URI is selected.  Another issue is that some of the
> terminology in
>       RFC 3261 is overloaded - e.g., "forwarding" refers both to a Proxy
>       which does not have responsibility for the domain of the request-URI
>       in the incoming request, thus the proxy just "forwards" the request
> to
>       the next hop AND "forwarding" is used to describe the process whereby
>       the outgoing request is built and "forwarded" to the next hop at
> which
>       point the proxy does not know how the new request-uri was selected.
>       RFC 4244 has attempted to clarify the terms and attempts to use
> "forward"
>       in the context of the former situation and "retarget" for the case
> whereby
>       a proxy is responsible for the domain and thus can use a number of
>       mechanism to select the new target for the request - e.g., a
> REGISTRAR,
>       configured data, etc.
>
>  3.  Related to the last point in item 2 above, it has been proposed that
>       we differentiate the hi-entries even more by defining separate
> parameters
>      for registered and configured/mapped contacts.
>      Currently when the R-URI is translated to a URI which is either
> derived
>      from location service lookup(registered by UA) or from mapping
>      table, there is no differentiation as to how the URI was derived once
> it is
>      added to the list of potential targets.
>
>      The general consensus of the authors of the two documents was that it
> may
>      be useful for some services to have the hi-entries tagged with the
>      more specific information.
>
>      And, of course, this gets us into another naming contest. In the end,
> the naming
>      is not so important as long as the term isn't too overloaded and it is
> defined
>      precisely in the document(s).
>
> We would appreciate WG feedback on these issues and any other comments on
> the two documents prior to IETF-74.
>
> Regards,
> Shida and Mary.
>
>
>
> Begin forwarded message:
> From: Internet-Drafts@ietf.org
> Date: March 10, 2009 2:30:01 AM JST
> To: i-d-announce@ietf.org
> Subject: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
> Reply-To: internet-drafts@ietf.org
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>
>
>        Title           : Delivery of Request-URI Targets to User Agents
>        Author(s)       : J. Rosenberg, H. van Elburg, C. Holmberg, F.
> Audet, S. Schubert
>        Filename        : draft-rosenberg-sip-target-uri-delivery-01.txt
>        Pages           : 16
>        Date            : 2009-3-9
>
> When a Session Initiation Protocol (SIP) proxy receives a request
>   targeted at a URI identifying a user or resource it is responsible
>   for, the proxy translates the URI to a registered or configured
>   contact URI of an agent representing that user or resource.  In the
>   process, the original URI is removed from the request.  Numerous use
>   cases have arisen which require this information to be delivered to
>   the user agent.  This document describes these use cases and defines
>   an extension to the History-Info header field which allows it to be
>   used to support those cases.
>
> A URL for this Internet-Draft is:
>
> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri-delivery-01.txt
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
> Content-Type: text/plain<BR>Content-ID: &lt;2009-3-9102756.I-D@ietf.org<lt%3B2009-3-9102756.I-D@ietf.org>
> &gt;<BR><BR>
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft<https://www.ietf.org/mailman/listinfo/i-d-announce%0AInternet-Draft>directories:
> http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

--0015174c16486e1b7b0464d3c3f1
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

I think that is a good question, because it is real world use that we would=
 like to serve. But at the same time when we want to progress this requirem=
ent, it would be good to know what is an acceptable means to transport the =
target-URI towards the UAS?<br>
<br>Would you reconsider using a distinct header for this?<br><br clear=3D"=
all">/Hans Erik van Elburg<br>
<br><br><div class=3D"gmail_quote">On Wed, Mar 11, 2009 at 7:55 AM, Hadriel=
 Kaplan <span dir=3D"ltr">&lt;<a href=3D"mailto:HKaplan@acmepacket.com">HKa=
plan@acmepacket.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quo=
te" style=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt=
 0.8ex; padding-left: 1ex;">
<br>
At the risk of opening up a topic we have already agreed on, but in the spi=
rit of open communication... [how&#39;s that for an opener!?]<br>
<br>
Are we absolutely sure we want to put this target-URI in the History-Info h=
eader?<br>
<br>
I am not asking this because I think it&#39;s hard. =A0I am asking because =
I am not sure it will be used. =A0Several of us from this same WG are invol=
ved in the SIP-Forum&#39;s SIP-Connect profile, and when faced with either =
using this new History-Info mechanism for a target-uri delivery issue, vs. =
not, we chose not to. =A0And we were mostly the same people who liked the i=
dea of putting it in History-Info in the IETF! (myself included) =A0But I a=
m concerned that when given a reasonable opportunity to use a mechanism we =
ourselves promote when wearing a different hat, we chose not to use it in p=
ractice. =A0It doesn&#39;t bode well. :(<br>

<br>
-hadriel<br>
p.s. sorry Mary for raising this, but there isn&#39;t much email traffic an=
yway. :)<br>
<br>
________________________________________<br>
From: <a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a> [mai=
lto:<a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a>] On Be=
half Of Shida Schubert<br>
Sent: Wednesday, March 11, 2009 2:13 AM<br>
To: <a href=3D"mailto:sip@ietf.org">sip@ietf.org</a> List<br>
<div><div></div><div class=3D"h5"><br>
We have submitted the updated target-uri draft based on the comments=A0<br>
submitted to the list and comments received at IETF73.=A0<br>
<br>
I have taken over as editor as Jonathan didn&#39;t have the cycles to=A0<br=
>
update the draft, with Francois, Christer and Hans Erick as additional=A0<b=
r>
co-authors and great deal of help from Mary. =A0<br>
=A0<br>
The following summarizes the changes made to the target-uri document<br>
1. Added use-case for toll-free number back<br>
2. Added definition of &quot;retarget&quot; operation.=A0<br>
3. Removed a reference to URN<br>
4. Added a text discussing the difference to P-Called-Party-Id<br>
5. Changed parameter name from &quot;target&quot; to &quot;istarget&quot;<b=
r>
=A0<br>
Note, that the target-uri document still contains the normative text for th=
e<br>
History-Info header.<br>
=A0<br>
In addition, Mary (with Francois as co-author) has submitted a<br>
rfc4244bis, with the following changes:<br>
1. Incorporated the normative aspects of the target-uri document<br>
into the existing normative text in RFC 4244 - the functionality is=A0<br>
virtually identical (as is some of the text) as the HI based solution=A0<br=
>
described in the target-uri document. =A0It&#39;s important that the soluti=
on<br>
be integrated into RFC 4244 as it MUST work and be based on the normative<b=
r>
aspects of RFC 4244.<br>
2. =A0Added the use cases from target-uri the the summary<br>
in the overview of rfc4244bis.<br>
3. Added an additional requirement to capture the &quot;target-uri&quot; in=
formation.<br>
4. Fixed an error in the RFC 4244 ABNF and added &quot;retarget&quot; param=
eter.<br>
5. Added a more simplified example.<br>
=A0<br>
=A0<br>
We had some very long offline exchanges as to the best way forward and<br>
remaining work for both documents.<br>
=A0<br>
Some of the issues identified are:<br>
<br>
::Issues::<br>
=A0=A0 1. Should we remove the normative text from target-uri and progress<=
br>
=A0=A0 =A0 =A0 4244bis along with the target-uri document to meet the chart=
ered<br>
=A0=A0 =A0 =A0SIP WG milestone?=A0<br>
=A0<br>
=A0<br>
=A0=A0 2. Name of the parameter.=A0<br>
=A0=A0 =A0 =A0At the last meeting, parameter &quot;target&quot; was said in=
appropriate=A0<br>
=A0=A0 =A0 =A0because voicemail-uri spec already defines a parameter called=
=A0<br>
=A0=A0 =A0 =A0&quot;target&quot; which also can be found in hi-entry, thus =
potentially=A0<br>
=A0=A0 =A0 =A0causing confusion.<br>
=A0=A0 =A0=A0<br>
=A0=A0 =A0 =A0Currently the target-uri draft uses &quot;istarget&quot; and =
4244bis uses<br>
=A0=A0 =A0 =A0&quot;retarget&quot; =A0but we could never come to=A0<br>
=A0=A0 =A0 =A0a consensus on what name is appropriate. =A0Other suggestions=
 have<br>
=A0=A0 =A0 =A0included the following:<br>
=A0=A0 =A0 =A0&quot;target-identity&quot; (someone didn&#39;t like that &qu=
ot;identity&quot; is also a SIP header)<br>
=A0=A0 =A0 =A0&quot;reg-uri&quot; =A0(can be paired with &quot;mapped-uri&q=
uot; for item 3 below)<br>
=A0=A0 =A0 =A0&quot;aor&quot;<br>
=A0=A0 =A0 =A0&quot;jibberish&quot;=A0<br>
=A0=A0 =A0 =A0etc.<br>
=A0<br>
=A0=A0 =A0 =A0One reason this is so difficult relates to the problem statem=
ent in target-uri in that<br>
=A0=A0 =A0 =A0RFC 3261 doesn&#39;t differentiate the mechanism by which the=
 new<br>
=A0=A0 =A0 =A0(target) Request-URI is selected. =A0Another issue is that so=
me of the terminology in<br>
=A0=A0 =A0 =A0RFC 3261 is overloaded - e.g., &quot;forwarding&quot; refers =
both to a Proxy<br>
=A0=A0 =A0 =A0which does not have responsibility for the domain of the requ=
est-URI<br>
=A0=A0 =A0 =A0in the incoming request, thus the proxy just &quot;forwards&q=
uot; the request to<br>
=A0=A0 =A0 =A0the next hop AND &quot;forwarding&quot; is used to describe t=
he process whereby<br>
=A0=A0 =A0 =A0the outgoing request is built and &quot;forwarded&quot; to th=
e next hop at which<br>
=A0=A0 =A0 =A0point the proxy does not know how the new request-uri was sel=
ected.<br>
=A0=A0 =A0 =A0RFC 4244 has attempted to clarify the terms and attempts to u=
se &quot;forward&quot;=A0<br>
=A0=A0 =A0 =A0in the context of the former situation and &quot;retarget&quo=
t; for the case whereby<br>
=A0=A0 =A0 =A0a proxy is responsible for the domain and thus can use a numb=
er of=A0<br>
=A0=A0 =A0 =A0mechanism to select the new target for the request - e.g., a =
REGISTRAR,=A0<br>
=A0=A0 =A0 =A0configured data, etc. =A0<br>
<br>
=A03. =A0Related to the last point in item 2 above, it has been proposed th=
at<br>
=A0=A0 =A0 =A0we differentiate the hi-entries even more by defining separat=
e parameters<br>
=A0=A0 =A0 for registered and configured/mapped contacts.=A0<br>
=A0=A0 =A0 Currently when the R-URI is translated to a URI which is either =
derived=A0<br>
=A0=A0 =A0 from location service lookup(registered by UA) or from mapping=
=A0<br>
=A0=A0 =A0 table, there is no differentiation as to how the URI was derived=
 once it is<br>
=A0=A0 =A0 added to the list of potential targets.<br>
=A0=A0 =A0=A0<br>
=A0=A0 =A0 The general consensus of the authors of the two documents was th=
at it may=A0<br>
=A0=A0 =A0 be useful for some services to have the hi-entries tagged with t=
he<br>
=A0=A0 =A0 more specific information.=A0<br>
=A0<br>
=A0=A0 =A0 And, of course, this gets us into another naming contest. In the=
 end, the naming<br>
=A0=A0 =A0 is not so important as long as the term isn&#39;t too overloaded=
 and it is defined<br>
=A0=A0 =A0 precisely in the document(s).<br>
=A0<br>
We would appreciate WG feedback on these issues and any other comments on<b=
r>
the two documents prior to IETF-74.<br>
=A0<br>
Regards,<br>
Shida and Mary.<br>
<br>
<br>
<br>
Begin forwarded message:<br>
From:=A0<a href=3D"mailto:Internet-Drafts@ietf.org">Internet-Drafts@ietf.or=
g</a><br>
Date:=A0March 10, 2009 2:30:01 AM JST<br>
To:=A0<a href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a><br=
>
Subject:=A0I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt=A0<br>
Reply-To:=A0<a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@iet=
f.org</a><br>
<br>
A New Internet-Draft is available from the on-line Internet-Drafts=A0<br>
directories.<br>
<br>
<br>
 =A0 =A0 =A0 =A0Title =A0 =A0 =A0 =A0 =A0 : Delivery of Request-URI Targets=
 to User Agents<br>
 =A0 =A0 =A0 =A0Author(s) =A0 =A0 =A0 : J. Rosenberg, H. van Elburg, C. Hol=
mberg, F. Audet, S. Schubert<br>
 =A0 =A0 =A0 =A0Filename =A0 =A0 =A0 =A0: draft-rosenberg-sip-target-uri-de=
livery-01.txt<br>
 =A0 =A0 =A0 =A0Pages =A0 =A0 =A0 =A0 =A0 : 16<br>
 =A0 =A0 =A0 =A0Date =A0 =A0 =A0 =A0 =A0 =A0: 2009-3-9<br>
<br>
When a Session Initiation Protocol (SIP) proxy receives a request<br>
=A0=A0targeted at a URI identifying a user or resource it is responsible<br=
>
=A0=A0for, the proxy translates the URI to a registered or configured<br>
=A0=A0contact URI of an agent representing that user or resource. =A0In the=
<br>
=A0=A0process, the original URI is removed from the request. =A0Numerous us=
e<br>
=A0=A0cases have arisen which require this information to be delivered to<b=
r>
=A0=A0the user agent. =A0This document describes these use cases and define=
s<br>
=A0=A0an extension to the History-Info header field which allows it to be<b=
r>
=A0=A0used to support those cases.<br>
<br>
A URL for this Internet-Draft is:<br>
<a href=3D"http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-u=
ri-delivery-01.txt" target=3D"_blank">http://www.ietf.org/internet-drafts/d=
raft-rosenberg-sip-target-uri-delivery-01.txt</a><br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
Below is the data which will enable a MIME compliant mail reader<br>
implementation to automatically retrieve the ASCII version of the<br>
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</div></div><div><div></div><div class=3D"h5">_____________________________=
__________________<br>
Sip mailing list =A0<a href=3D"https://www.ietf.org/mailman/listinfo/sip" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
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.columbia.edu</a> for questions on current sip<br>
Use <a href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a> for new develo=
pments on the application of sip<br>
</div></div></blockquote></div><br>

--0015174c16486e1b7b0464d3c3f1--

From rco@iptel.org  Wed Mar 11 01:52:27 2009
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From: Raphael Coeffic <rco@iptel.org>
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To: Hadriel Kaplan <HKaplan@acmepacket.com>
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Cc: Nils Ohlmeier <lists@ohlmeier.org>, "sip@ietf.org" <sip@ietf.org>
Subject: Re: [Sip] draft-state-sip-relay-attack-00
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Hadriel Kaplan wrote:
>   
>> -----Original Message-----
>> From: Raphael Coeffic [mailto:rco@iptel.org]
>> Sent: Tuesday, March 10, 2009 6:06 AM
>>
>> Hadriel Kaplan wrote:
>>
>> Forcing registrations is the path that IMS went for, I believe. But if
>> you want to take advantage of this, you may have to deploy a little more
>> IMS than you'd like to. 
>>     
>
> The concept and practice of "forcing" UA's, which have AoR's belonging to a domain, to REGISTER their contact binding for those AoR's in order to make calls, has been around a lot longer than IMS has.  Invoking the "that sounds like IMS" defense is a cheap guilt-by-association argument, and not relevant.  One might as well argue that using INVITE's or RTP "is the path that IMS went for".
>
> You keep making it sound like sending a SIP request which does not create a dialog is somehow a difficult mechanism to implement, or "forces" people to do something they don't want to.  What is it forcing?
>
>   
Well, it is not required as per RFC 3261 ;-)

> You're looking for a means of authenticating a UA without being subject to a relay attack.  Using a different SIP Method name, which can only be initiated by the client with the credentials, and one that UA's cannot be baited to provide by random third parties, is one solution to the problem.  If you don't like the string "REGISTER", then how about "OPTIONS"?
>
>   
No, I think you missunderstood me. I still think that requiring a 
registration and checking the source IP and port against the registered 
contact is a fair measure to take. However, it introduces more traffic 
in some scenarios, where it is not necessarily needed. REGISTERs could 
become the "background load" in the servers. This also puts a lot of 
state into some proxies that does not need it, etc...
My concern is just that it seems like there is no better solution.
>> Maybe just an example: let's say you have a home SIP server, doing the
>> usual least cost routing. Your least cost router might have something
>> like 50 different routes. Do you want this box, or maybe your phones to
>> have 50 running registrations, just for the purpose of having cheap
>> calls? Well, personaly, I would prefer to just install my certificate on
>> this box, and use TLS. But as very very few of those PSTN providers do
>> support TLS, I cannot. By the way, there are already commercial products
>> supporting this scenario.
>>     
>
> So you have a mechanism that people don't seem to want to implement - and meanwhile you have another mechanism, which is widely implemented.  And the problem is...?
>
>   
See previous answer.

Cheers
Raphael.

From ietf.hanserik@gmail.com  Wed Mar 11 01:58:15 2009
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Date: Wed, 11 Mar 2009 09:58:48 +0100
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From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: Shida Schubert <shida@ntt-at.com>
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Subject: Re: [Sip] Fwd: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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--0015174bdeaa6c08510464d4130f
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One of the problems that made the discussion quite tedious is the complete
inversed meaning of the terms "retarget" and "reroute" between 4244bis and
target-uri factions have.

Conclusion was that the terminology need to be properly defined and some
attempt was made to start such activity.

On 3. I have strong concerns when we start tagging the URI as to how they
come about "retarget"/"mapped conficuration" etc. It is better to embellish
them with a tag that just represents there meaning for example "istarget" or
"hop". For the following reasons:
1. It is much more intuitive for the user of this information, its meaning
can basically be guessed.
2. Such meaning also probably survives application in other problem spaces
that we had not foreseen when introducing the concept.

The solution that is  described now in the target-URI delivery draft is not
yet complete, it does not solve the freephone use case for example and in
its current form it will deliver exactly the same target-URI as that which
would be delivered by the  P-Called-Party-ID header. Contrary to what the
draft says. So some work is needed still.

/Hans Erik van Elburg


On Wed, Mar 11, 2009 at 7:13 AM, Shida Schubert <shida@ntt-at.com> wrote:

>
> We have submitted the updated target-uri draft based on the comments
> submitted to the list and comments received at IETF73.
>
> I have taken over as editor as Jonathan didn't have the cycles to
> update the draft, with Francois, Christer and Hans Erick as additional
> co-authors and great deal of help from Mary.
>
> The following summarizes the changes made to the target-uri document
> 1. Added use-case for toll-free number back
> 2. Added definition of "retarget" operation.
> 3. Removed a reference to URN
> 4. Added a text discussing the difference to P-Called-Party-Id
> 5. Changed parameter name from "target" to "istarget"
>
> Note, that the target-uri document still contains the normative text for
> the
> History-Info header.
>
> In addition, Mary (with Francois as co-author) has submitted a
> rfc4244bis, with the following changes:
> 1. Incorporated the normative aspects of the target-uri document
> into the existing normative text in RFC 4244 - the functionality is
> virtually identical (as is some of the text) as the HI based solution
> described in the target-uri document.  It's important that the solution
> be integrated into RFC 4244 as it MUST work and be based on the normative
> aspects of RFC 4244.
> 2.  Added the use cases from target-uri the the summary
> in the overview of rfc4244bis.
> 3. Added an additional requirement to capture the "target-uri" information.
> 4. Fixed an error in the RFC 4244 ABNF and added "retarget" parameter.
> 5. Added a more simplified example.
>
>
> We had some very long offline exchanges as to the best way forward and
> remaining work for both documents.
>
> Some of the issues identified are:
>
> ::Issues::
>    1. Should we remove the normative text from target-uri and progress
>        4244bis along with the target-uri document to meet the chartered
>       SIP WG milestone?
>
>
>    2. Name of the parameter.
>       At the last meeting, parameter "target" was said inappropriate
>       because voicemail-uri spec already defines a parameter called
>       "target" which also can be found in hi-entry, thus potentially
>       causing confusion.
>
>       Currently the target-uri draft uses "istarget" and 4244bis uses
>       "retarget"  but we could never come to
>       a consensus on what name is appropriate.  Other suggestions have
>       included the following:
>       "target-identity" (someone didn't like that "identity" is also a SIP
> header)
>       "reg-uri"  (can be paired with "mapped-uri" for item 3 below)
>       "aor"
>       "jibberish"
>       etc.
>
>       One reason this is so difficult relates to the problem statement in
> target-uri in that
>       RFC 3261 doesn't differentiate the mechanism by which the new
>       (target) Request-URI is selected.  Another issue is that some of the
> terminology in
>       RFC 3261 is overloaded - e.g., "forwarding" refers both to a Proxy
>       which does not have responsibility for the domain of the request-URI
>       in the incoming request, thus the proxy just "forwards" the request
> to
>       the next hop AND "forwarding" is used to describe the process whereby
>       the outgoing request is built and "forwarded" to the next hop at
> which
>       point the proxy does not know how the new request-uri was selected.
>       RFC 4244 has attempted to clarify the terms and attempts to use
> "forward"
>       in the context of the former situation and "retarget" for the case
> whereby
>       a proxy is responsible for the domain and thus can use a number of
>       mechanism to select the new target for the request - e.g., a
> REGISTRAR,
>       configured data, etc.
>
>  3.  Related to the last point in item 2 above, it has been proposed that
>       we differentiate the hi-entries even more by defining separate
> parameters
>      for registered and configured/mapped contacts.
>      Currently when the R-URI is translated to a URI which is either
> derived
>      from location service lookup(registered by UA) or from mapping
>      table, there is no differentiation as to how the URI was derived once
> it is
>      added to the list of potential targets.
>
>      The general consensus of the authors of the two documents was that it
> may
>      be useful for some services to have the hi-entries tagged with the
>      more specific information.
>
>      And, of course, this gets us into another naming contest. In the end,
> the naming
>      is not so important as long as the term isn't too overloaded and it is
> defined
>      precisely in the document(s).
>
> We would appreciate WG feedback on these issues and any other comments on
> the two documents prior to IETF-74.
>
> Regards,
> Shida and Mary.
>
>
>
> Begin forwarded message:
>
> *From: *Internet-Drafts@ietf.org
> *Date: *March 10, 2009 2:30:01 AM JST
> *To: *i-d-announce@ietf.org
> *Subject: **I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt *
> *Reply-To: *internet-drafts@ietf.org
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>
>
> Title : Delivery of Request-URI Targets to User Agents
> Author(s) : J. Rosenberg, H. van Elburg, C. Holmberg, F. Audet, S.
> Schubert
> Filename : draft-rosenberg-sip-target-uri-delivery-01.txt
> Pages : 16
> Date : 2009-3-9
>
> When a Session Initiation Protocol (SIP) proxy receives a request
>   targeted at a URI identifying a user or resource it is responsible
>   for, the proxy translates the URI to a registered or configured
>   contact URI of an agent representing that user or resource.  In the
>   process, the original URI is removed from the request.  Numerous use
>   cases have arisen which require this information to be delivered to
>   the user agent.  This document describes these use cases and defines
>   an extension to the History-Info header field which allows it to be
>   used to support those cases.
>
> A URL for this Internet-Draft is:
>
> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri-delivery-01.txt
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
>
> Content-Type: text/plain<BR>Content-ID: &lt;2009-3-9102756.I-D@ietf.org
> &gt;<BR><BR>
>
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

--0015174bdeaa6c08510464d4130f
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

One of the problems that made the discussion quite tedious is the complete =
inversed meaning of the terms &quot;retarget&quot; and &quot;reroute&quot; =
between 4244bis and target-uri factions have.<br><br>Conclusion was that th=
e terminology need to be properly defined and some attempt was made to star=
t such activity.<br>
<br>On 3. I have strong concerns when we start tagging the URI as to how th=
ey come about &quot;retarget&quot;/&quot;mapped conficuration&quot; etc. It=
 is better to embellish them with a tag that just represents there meaning =
for example &quot;istarget&quot; or &quot;hop&quot;. For the following reas=
ons:<br>
1. It is much more intuitive for the user of this information, its meaning =
can basically be guessed.<br>2. Such meaning also probably survives applica=
tion in other problem spaces that we had not foreseen when introducing the =
concept.<br>
<br>The solution that is=A0 described now in the target-URI delivery draft =
is not yet complete, it does not solve the freephone use case for example a=
nd in its current form it will deliver exactly the same target-URI as that =
which would be delivered by the=A0 P-Called-Party-ID header. Contrary to wh=
at the draft says. So some work is needed still.<br>
<br>/Hans Erik van Elburg<br>
<br><br><div class=3D"gmail_quote">On Wed, Mar 11, 2009 at 7:13 AM, Shida S=
chubert <span dir=3D"ltr">&lt;<a href=3D"mailto:shida@ntt-at.com">shida@ntt=
-at.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D=
"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padd=
ing-left: 1ex;">
<div style=3D""><div><br></div><div>We have submitted the updated target-ur=
i draft based on the comments=A0</div><div>submitted to the list and commen=
ts received at IETF73.=A0</div><div><br></div><div>I have taken over as edi=
tor as Jonathan didn&#39;t have the cycles to=A0</div>
<div>update the draft, with Francois, Christer and Hans Erick as additional=
=A0</div><div>co-authors and great deal of help from Mary. =A0</div><div>=
=A0</div><div>The following summarizes the changes made to the target-uri d=
ocument</div>
<div>1. Added use-case for toll-free number back</div><div>2. Added definit=
ion of &quot;retarget&quot; operation.=A0</div><div>3. Removed a reference =
to URN</div><div>4. Added a text discussing the difference to P-Called-Part=
y-Id</div>
<div>5. Changed parameter name from &quot;target&quot; to &quot;istarget&qu=
ot;</div><div>=A0</div><div>Note, that the target-uri document still contai=
ns the normative text for the</div><div>History-Info header.</div><div>=A0<=
/div>
<div>In addition, Mary (with Francois as co-author) has submitted a</div><d=
iv>rfc4244bis, with the following changes:</div><div>1. Incorporated the no=
rmative aspects of the target-uri document</div><div>into the existing norm=
ative text in RFC 4244 - the functionality is=A0</div>
<div>virtually identical (as is some of the text) as the HI based solution=
=A0</div><div>described in the target-uri document. =A0It&#39;s important t=
hat the solution</div><div>be integrated into RFC 4244 as it MUST work and =
be based on the normative</div>
<div>aspects of RFC 4244.</div><div>2. =A0Added the use cases from target-u=
ri the the summary</div><div>in the overview of rfc4244bis.</div><div>3. Ad=
ded an additional requirement to capture the &quot;target-uri&quot; informa=
tion.</div>
<div>4. Fixed an error in the RFC 4244 ABNF and added &quot;retarget&quot; =
parameter.</div><div>5. Added a more simplified example.</div><div>=A0</div=
><div>=A0</div><div>We had some very long offline exchanges as to the best =
way forward and</div>
<div>remaining work for both documents.</div><div>=A0</div><div>Some of the=
 issues identified are:</div><div><br></div><div>::Issues::</div><div>=A0=
=A0 1. Should we remove the normative text from target-uri and progress</di=
v><div>
=A0=A0 =A0 =A0 4244bis along with the target-uri document to meet the chart=
ered</div><div>=A0=A0 =A0 =A0SIP WG milestone?=A0</div><div>=A0</div><div>=
=A0</div><div>=A0=A0 2. Name of the parameter.=A0</div><div>=A0=A0 =A0 =A0A=
t the last meeting, parameter &quot;target&quot; was said inappropriate=A0<=
/div>
<div>=A0=A0 =A0 =A0because voicemail-uri spec already defines a parameter c=
alled=A0</div><div>=A0=A0 =A0 =A0&quot;target&quot; which also can be found=
 in hi-entry, thus potentially=A0</div><div>=A0=A0 =A0 =A0causing confusion=
.</div><div>=A0=A0 =A0=A0</div><div>
=A0=A0 =A0 =A0Currently the target-uri draft uses &quot;istarget&quot; and =
4244bis uses</div><div>=A0=A0 =A0 =A0&quot;retarget&quot; =A0but we could n=
ever come to=A0</div><div>=A0=A0 =A0 =A0a consensus on what name is appropr=
iate. =A0Other suggestions have</div>
<div>=A0=A0 =A0 =A0included the following:</div><div>=A0=A0 =A0 =A0&quot;ta=
rget-identity&quot; (someone didn&#39;t like that &quot;identity&quot; is a=
lso a SIP header)</div><div>=A0=A0 =A0 =A0&quot;reg-uri&quot; =A0(can be pa=
ired with &quot;mapped-uri&quot; for item 3 below)</div>
<div>=A0=A0 =A0 =A0&quot;aor&quot;</div><div>=A0=A0 =A0 =A0&quot;jibberish&=
quot;=A0</div><div>=A0=A0 =A0 =A0etc.</div><div>=A0</div><div>=A0=A0 =A0 =
=A0One reason this is so difficult relates to the problem statement in targ=
et-uri in that</div><div>=A0=A0 =A0 =A0RFC 3261 doesn&#39;t differentiate t=
he mechanism by which the new</div>
<div>=A0=A0 =A0 =A0(target) Request-URI is selected. =A0Another issue is th=
at some of the terminology in</div><div>=A0=A0 =A0 =A0RFC 3261 is overloade=
d - e.g., &quot;forwarding&quot; refers both to a Proxy</div><div>=A0=A0 =
=A0 =A0which does not have responsibility for the domain of the request-URI=
</div>
<div>=A0=A0 =A0 =A0in the incoming request, thus the proxy just &quot;forwa=
rds&quot; the request to</div><div>=A0=A0 =A0 =A0the next hop AND &quot;for=
warding&quot; is used to describe the process whereby</div><div>=A0=A0 =A0 =
=A0the outgoing request is built and &quot;forwarded&quot; to the next hop =
at which</div>
<div>=A0=A0 =A0 =A0point the proxy does not know how the new request-uri wa=
s selected.</div><div>=A0=A0 =A0 =A0RFC 4244 has attempted to clarify the t=
erms and attempts to use &quot;forward&quot;=A0</div><div>=A0=A0 =A0 =A0in =
the context of the former situation and &quot;retarget&quot; for the case w=
hereby</div>
<div>=A0=A0 =A0 =A0a proxy is responsible for the domain and thus can use a=
 number of=A0</div><div>=A0=A0 =A0 =A0mechanism to select the new target fo=
r the request - e.g., a REGISTRAR,=A0</div><div>=A0=A0 =A0 =A0configured da=
ta, etc. =A0</div><div><br>
</div><div>=A03. =A0Related to the last point in item 2 above, it has been =
proposed that</div><div>=A0=A0 =A0 =A0we differentiate the hi-entries even =
more by defining separate parameters</div><div>=A0=A0 =A0 for registered an=
d configured/mapped contacts.=A0</div>
<div>=A0=A0 =A0 Currently when the R-URI is translated to a URI which is ei=
ther derived=A0</div><div>=A0=A0 =A0 from location service lookup(registere=
d by UA) or from mapping=A0</div><div>=A0=A0 =A0 table, there is no differe=
ntiation as to how the URI was derived once it is</div>
<div>=A0=A0 =A0 added to the list of potential targets.</div><div>=A0=A0 =
=A0=A0</div><div>=A0=A0 =A0 The general consensus of the authors of the two=
 documents was that it may=A0</div><div>=A0=A0 =A0 be useful for some servi=
ces to have the hi-entries tagged with the</div>
<div>=A0=A0 =A0 more specific information.=A0</div><div>=A0</div><div>=A0=
=A0 =A0 And, of course, this gets us into another naming contest. In the en=
d, the naming</div><div>=A0=A0 =A0 is not so important as long as the term =
isn&#39;t too overloaded and it is defined</div>
<div>=A0=A0 =A0 precisely in the document(s).</div><div>=A0</div><div>We wo=
uld appreciate WG feedback on these issues and any other comments on</div><=
div>the two documents prior to IETF-74.</div><div>=A0</div><div>Regards,</d=
iv><div>
Shida and Mary.</div><div><br></div><div><br></div><div><div><div><div><br>=
Begin forwarded message:</div><blockquote type=3D"cite"><div><div style=3D"=
margin: 0px;"><font style=3D"color: rgb(0, 0, 0);" size=3D"3" color=3D"#000=
000" face=3D"Helvetica"><b>From:=A0</b></font><font size=3D"3" face=3D"Helv=
etica"><a href=3D"mailto:Internet-Drafts@ietf.org" target=3D"_blank">Intern=
et-Drafts@ietf.org</a></font></div>
<div style=3D"margin: 0px;"><font style=3D"color: rgb(0, 0, 0);" size=3D"3"=
 color=3D"#000000" face=3D"Helvetica"><b>Date:=A0</b></font><font size=3D"3=
" face=3D"Helvetica">March 10, 2009 2:30:01 AM JST</font></div><div style=
=3D"margin: 0px;">
<font style=3D"color: rgb(0, 0, 0);" size=3D"3" color=3D"#000000" face=3D"H=
elvetica"><b>To:=A0</b></font><font size=3D"3" face=3D"Helvetica"><a href=
=3D"mailto:i-d-announce@ietf.org" target=3D"_blank">i-d-announce@ietf.org</=
a></font></div><div style=3D"margin: 0px;">
<font style=3D"color: rgb(0, 0, 0);" size=3D"3" color=3D"#000000" face=3D"H=
elvetica"><b>Subject:=A0</b></font><font size=3D"3" face=3D"Helvetica"><b>I=
-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt=A0</b></font></div=
><div style=3D"margin: 0px;">
<font style=3D"color: rgb(0, 0, 0);" size=3D"3" color=3D"#000000" face=3D"H=
elvetica"><b>Reply-To:=A0</b></font><font size=3D"3" face=3D"Helvetica"><a =
href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">internet-drafts@=
ietf.org</a></font></div>
<div style=3D"margin: 0px; min-height: 14px;"><br></div></div><div>A New In=
ternet-Draft is available from the on-line Internet-Drafts=A0<br>directorie=
s.<br><br><br><span style=3D"white-space: pre;">	</span>Title<span style=3D=
"white-space: pre;">	</span><span style=3D"white-space: pre;">	</span>: Del=
ivery of Request-URI Targets to User Agents<br>
<span style=3D"white-space: pre;">	</span>Author(s)<span style=3D"white-spa=
ce: pre;">	</span>: J. Rosenberg, H. van Elburg, C. Holmberg, F. Audet, S. =
Schubert<br><span style=3D"white-space: pre;">	</span>Filename<span style=
=3D"white-space: pre;">	</span>: draft-rosenberg-sip-target-uri-delivery-01=
.txt<br>
<span style=3D"white-space: pre;">	</span>Pages<span style=3D"white-space: =
pre;">	</span><span style=3D"white-space: pre;">	</span>: 16<br><span style=
=3D"white-space: pre;">	</span>Date<span style=3D"white-space: pre;">	</spa=
n><span style=3D"white-space: pre;">	</span>: 2009-3-9<br>
<span style=3D"white-space: pre;">	</span><br>When a Session Initiation Pro=
tocol (SIP) proxy receives a request<br>=A0=A0targeted at a URI identifying=
 a user or resource it is responsible<br>=A0=A0for, the proxy translates th=
e URI to a registered or configured<br>
=A0=A0contact URI of an agent representing that user or resource. =A0In the=
<br>=A0=A0process, the original URI is removed from the request. =A0Numerou=
s use<br>=A0=A0cases have arisen which require this information to be deliv=
ered to<br>=A0=A0the user agent. =A0This document describes these use cases=
 and defines<br>
=A0=A0an extension to the History-Info header field which allows it to be<b=
r>=A0=A0used to support those cases.<br><br>A URL for this Internet-Draft i=
s:<br><a href=3D"http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ta=
rget-uri-delivery-01.txt" target=3D"_blank">http://www.ietf.org/internet-dr=
afts/draft-rosenberg-sip-target-uri-delivery-01.txt</a><br>
<br>Internet-Drafts are also available by anonymous FTP at:<br><a href=3D"f=
tp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp.ietf.org/in=
ternet-drafts/</a><br><br>Below is the data which will enable a MIME compli=
ant mail reader<br>
implementation to automatically retrieve the ASCII version of the<br>Intern=
et-Draft.<br></div></blockquote></div></div>Content-Type: text/plain&lt;BR&=
gt;Content-ID: &amp;lt;<a href=3D"mailto:2009-3-9102756.I-D@ietf.org" targe=
t=3D"_blank">2009-3-9102756.I-D@ietf.org</a>&amp;gt;&lt;BR&gt;&lt;BR&gt;<di=
v>
<div><blockquote type=3D"cite"><div>_______________________________________=
________<br>I-D-Announce mailing list<br><a href=3D"mailto:I-D-Announce@iet=
f.org" target=3D"_blank">I-D-Announce@ietf.org</a><br><a href=3D"https://ww=
w.ietf.org/mailman/listinfo/i-d-announce" target=3D"_blank">https://www.iet=
f.org/mailman/listinfo/i-d-announce</a><br>
Internet-Draft directories:=A0<a href=3D"http://www.ietf.org/shadow.html" t=
arget=3D"_blank">http://www.ietf.org/shadow.html</a><br>or=A0<a href=3D"ftp=
://ftp.ietf.org/ietf/1shadow-sites.txt" target=3D"_blank">ftp://ftp.ietf.or=
g/ietf/1shadow-sites.txt</a></div>
</blockquote></div></div></div></div><br>__________________________________=
_____________<br>
Sip mailing list =A0<a href=3D"https://www.ietf.org/mailman/listinfo/sip" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
This list is for NEW development of the core SIP Protocol<br>
Use <a href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs=
.columbia.edu</a> for questions on current sip<br>
Use <a href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a> for new develo=
pments on the application of sip<br></blockquote></div><br>

--0015174bdeaa6c08510464d4130f--

From ian.elz@ericsson.com  Wed Mar 11 02:52:42 2009
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From: "Ian Elz" <ian.elz@ericsson.com>
To: "Mary Barnes" <mary.barnes@nortel.com>, "Francois Audet" <audet@nortel.com>
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Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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Mary, Francois,

Section 4.1 of the draft allows for the addition of the Privacy header
parameter with a value of "history" to be included with the
hi-targeted-to-uri.

Can this be extended to also allow the value "none".

A node adding a H-I entry is allowed to specify the privacy value
"history" either in the Privacy header or as a header parameter
associated with the hi-targeted-to-uri. If the former action is taken by
one node this results in privacy for all history info header entries
even if this is not required. There is no way for privacy of an
individual H-I entry to be set to none if no Privacy is required for the
uri.

This creates a specific problem when interworking with the PSTN where
the privacy is associated with each E.164 number included in the
protocol. If an INVITE is being mapped to ISUP and the H-I entries are
being used to map to redirection numbers then a single Privacy header
with the value "history" will result in all numbers being restricted.

If the Privacy header parameter could include the value "none" then this
would explicitly indicate that the associated uri was allowed to be
presented.

The ability to explicitly allow presentation of specific H-I entries may
also be useful in pure sip implementations.

I don't believe that this change would create any backward compatibility
issues. The existing implementations will continue as deployed but new
implementations could explicitly set no privacy for a specific uri.

There are no security issues other than those already defined in the
draft that an intervening node could modify an existing H-I entry. The
Privacy header value of "none" should only be included by the node in
the same manner as the existing inclusion of the "history" value.

Ian Elz


-----Original Message-----
From: i-d-announce-bounces@ietf.org
[mailto:i-d-announce-bounces@ietf.org] On Behalf Of
Internet-Drafts@ietf.org
Sent: 04 March 2009 15:15
To: i-d-announce@ietf.org
Subject: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

A New Internet-Draft is available from the on-line Internet-Drafts
directories.

	Title           : An Extension to the Session Initiation
Protocol (SIP) for Request History Information
	Author(s)       : M. Barnes, F. Audet
	Filename        : draft-barnes-sip-rfc4244bis-00.txt
	Pages           : 49
	Date            : 2009-03-04

This document defines a standard mechanism for capturing the history
information associated with a Session Initiation Protocol (SIP)
request.  This capability enables many enhanced services by providing
the information as to how and why a call arrives at a specific
application or user.  This document defines a new optional SIP
header, History-Info, for capturing the history information in
requests.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-barnes-sip-rfc4244bis-00.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

From john.elwell@siemens.com  Wed Mar 11 04:33:40 2009
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From: "Elwell, John" <john.elwell@siemens.com>
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Just to expand on what Hadriel says, the SIP Forum is addressing the =
case where a request is delivered by a SIP Service Provider to a =
SIP-PBX, the SIP-PBX having registered with the SIP-SP. The SIP-PBX =
needs to be in possession of the target, in order to route onwards =
correctly within the enterprise. If only the SIP-PBX's contact URI is =
available in the Request-URI, the SIP-PBX would have to look elsewhere =
for the target. In accordance with recent SIP WG agreements, this would =
be the History-Info header field, but that requires the present =
extension.

Advantages of putting the target in the Request-URI for this particular =
situation:
- This is where the SIP-PBX would normally expect to find the target =
when a request is received from elsewhere (e.g., another SIP-PBX within =
the enterprise), so it is reasonable also to expect it there when =
received from a SIP SP.
- SIPConnect 1.1 will not require History-Info for any other purpose, so =
it would make it a mandatory specification for SIPConnect 1.1 just in =
order to solve this particular problem, thereby raising the bar for SPs =
and SIP-PBXs wanting to comply with the spec.
- We would need to define procedures for what to do if History-Info is =
absent from a received request or if there is no entry marked as =
'istarget'.
- SIPConnect 1.1 is due for completion in the next few weeks, and cannot =
wait the usual IETF cycle time for History-Info update to hit the RFC =
Editor's queue.
- If History-Info is used anyway (for its original purpose), the history =
revealed when a request comes from a SIP-SP via a SIP-PBX to another =
proxy and finally to a UAS would be unnecessarily complicated and =
therefore confusing, because the target would be replaced by a contact =
and would then appear again at the next hop.

John
=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Hadriel Kaplan
> Sent: 11 March 2009 06:55
> To: Shida Schubert; sip@ietf.org List
> Cc: Mary Barnes
> Subject: Re: [Sip] Fwd: I-D=20
> ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>=20
>=20
> At the risk of opening up a topic we have already agreed on,=20
> but in the spirit of open communication... [how's that for an=20
> opener!?]
>=20
> Are we absolutely sure we want to put this target-URI in the=20
> History-Info header?
>=20
> I am not asking this because I think it's hard.  I am asking=20
> because I am not sure it will be used.  Several of us from=20
> this same WG are involved in the SIP-Forum's SIP-Connect=20
> profile, and when faced with either using this new=20
> History-Info mechanism for a target-uri delivery issue, vs.=20
> not, we chose not to.  And we were mostly the same people who=20
> liked the idea of putting it in History-Info in the IETF!=20
> (myself included)  But I am concerned that when given a=20
> reasonable opportunity to use a mechanism we ourselves=20
> promote when wearing a different hat, we chose not to use it=20
> in practice.  It doesn't bode well. :(
>=20
> -hadriel
> p.s. sorry Mary for raising this, but there isn't much email=20
> traffic anyway. :)
>=20
> ________________________________________
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Shida Schubert
> Sent: Wednesday, March 11, 2009 2:13 AM
> To: sip@ietf.org List
>=20
> We have submitted the updated target-uri draft based on the =
comments=A0
> submitted to the list and comments received at IETF73.=A0
>=20
> I have taken over as editor as Jonathan didn't have the cycles to=A0
> update the draft, with Francois, Christer and Hans Erick as=20
> additional=A0
> co-authors and great deal of help from Mary. =A0
> =A0
> The following summarizes the changes made to the target-uri document
> 1. Added use-case for toll-free number back
> 2. Added definition of "retarget" operation.=A0
> 3. Removed a reference to URN
> 4. Added a text discussing the difference to P-Called-Party-Id
> 5. Changed parameter name from "target" to "istarget"
> =A0
> Note, that the target-uri document still contains the=20
> normative text for the
> History-Info header.
> =A0
> In addition, Mary (with Francois as co-author) has submitted a
> rfc4244bis, with the following changes:
> 1. Incorporated the normative aspects of the target-uri document
> into the existing normative text in RFC 4244 - the functionality is=A0
> virtually identical (as is some of the text) as the HI based =
solution=A0
> described in the target-uri document. =A0It's important that=20
> the solution
> be integrated into RFC 4244 as it MUST work and be based on=20
> the normative
> aspects of RFC 4244.
> 2. =A0Added the use cases from target-uri the the summary
> in the overview of rfc4244bis.
> 3. Added an additional requirement to capture the=20
> "target-uri" information.
> 4. Fixed an error in the RFC 4244 ABNF and added "retarget" parameter.
> 5. Added a more simplified example.
> =A0
> =A0
> We had some very long offline exchanges as to the best way forward and
> remaining work for both documents.
> =A0
> Some of the issues identified are:
>=20
> ::Issues::
> =A0=A0 1. Should we remove the normative text from target-uri and =
progress
> =A0=A0 =A0 =A0 4244bis along with the target-uri document to meet the=20
> chartered
> =A0=A0 =A0 =A0SIP WG milestone?=A0
> =A0
> =A0
> =A0=A0 2. Name of the parameter.=A0
> =A0=A0 =A0 =A0At the last meeting, parameter "target" was said =
inappropriate=A0
> =A0=A0 =A0 =A0because voicemail-uri spec already defines a parameter =
called=A0
> =A0=A0 =A0 =A0"target" which also can be found in hi-entry, thus =
potentially=A0
> =A0=A0 =A0 =A0causing confusion.
> =A0=A0 =A0=A0
> =A0=A0 =A0 =A0Currently the target-uri draft uses "istarget" and =
4244bis uses
> =A0=A0 =A0 =A0"retarget" =A0but we could never come to=A0
> =A0=A0 =A0 =A0a consensus on what name is appropriate. =A0Other =
suggestions have
> =A0=A0 =A0 =A0included the following:
> =A0=A0 =A0 =A0"target-identity" (someone didn't like that "identity"=20
> is also a SIP header)
> =A0=A0 =A0 =A0"reg-uri" =A0(can be paired with "mapped-uri" for item 3 =
below)
> =A0=A0 =A0 =A0"aor"
> =A0=A0 =A0 =A0"jibberish"=A0
> =A0=A0 =A0 =A0etc.
> =A0
> =A0=A0 =A0 =A0One reason this is so difficult relates to the problem=20
> statement in target-uri in that
> =A0=A0 =A0 =A0RFC 3261 doesn't differentiate the mechanism by which =
the new
> =A0=A0 =A0 =A0(target) Request-URI is selected. =A0Another issue is=20
> that some of the terminology in
> =A0=A0 =A0 =A0RFC 3261 is overloaded - e.g., "forwarding" refers both=20
> to a Proxy
> =A0=A0 =A0 =A0which does not have responsibility for the domain of=20
> the request-URI
> =A0=A0 =A0 =A0in the incoming request, thus the proxy just "forwards"=20
> the request to
> =A0=A0 =A0 =A0the next hop AND "forwarding" is used to describe the=20
> process whereby
> =A0=A0 =A0 =A0the outgoing request is built and "forwarded" to the=20
> next hop at which
> =A0=A0 =A0 =A0point the proxy does not know how the new request-uri=20
> was selected.
> =A0=A0 =A0 =A0RFC 4244 has attempted to clarify the terms and=20
> attempts to use "forward"=A0
> =A0=A0 =A0 =A0in the context of the former situation and "retarget"=20
> for the case whereby
> =A0=A0 =A0 =A0a proxy is responsible for the domain and thus can use=20
> a number of=A0
> =A0=A0 =A0 =A0mechanism to select the new target for the request -=20
> e.g., a REGISTRAR,=A0
> =A0=A0 =A0 =A0configured data, etc. =A0
>=20
> =A03. =A0Related to the last point in item 2 above, it has been=20
> proposed that
> =A0=A0 =A0 =A0we differentiate the hi-entries even more by defining=20
> separate parameters
> =A0=A0 =A0 for registered and configured/mapped contacts.=A0
> =A0=A0 =A0 Currently when the R-URI is translated to a URI which is=20
> either derived=A0
> =A0=A0 =A0 from location service lookup(registered by UA) or from =
mapping=A0
> =A0=A0 =A0 table, there is no differentiation as to how the URI was=20
> derived once it is
> =A0=A0 =A0 added to the list of potential targets.
> =A0=A0 =A0=A0
> =A0=A0 =A0 The general consensus of the authors of the two=20
> documents was that it may=A0
> =A0=A0 =A0 be useful for some services to have the hi-entries=20
> tagged with the
> =A0=A0 =A0 more specific information.=A0
> =A0
> =A0=A0 =A0 And, of course, this gets us into another naming=20
> contest. In the end, the naming
> =A0=A0 =A0 is not so important as long as the term isn't too=20
> overloaded and it is defined
> =A0=A0 =A0 precisely in the document(s).
> =A0
> We would appreciate WG feedback on these issues and any other=20
> comments on
> the two documents prior to IETF-74.
> =A0
> Regards,
> Shida and Mary.
>=20
>=20
>=20
> Begin forwarded message:
> From:=A0Internet-Drafts@ietf.org
> Date:=A0March 10, 2009 2:30:01 AM JST
> To:=A0i-d-announce@ietf.org
> Subject:=A0I-D =
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt=A0
> Reply-To:=A0internet-drafts@ietf.org
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts=A0
> directories.
>=20
>=20
> 	Title		: Delivery of Request-URI Targets to User Agents
> 	Author(s)	: J. Rosenberg, H. van Elburg, C.=20
> Holmberg, F. Audet, S. Schubert
> 	Filename	: draft-rosenberg-sip-target-uri-delivery-01.txt
> 	Pages		: 16
> 	Date		: 2009-3-9
> =09
> When a Session Initiation Protocol (SIP) proxy receives a request
> =A0=A0targeted at a URI identifying a user or resource it is =
responsible
> =A0=A0for, the proxy translates the URI to a registered or configured
> =A0=A0contact URI of an agent representing that user or resource. =
=A0In the
> =A0=A0process, the original URI is removed from the request. =
=A0Numerous use
> =A0=A0cases have arisen which require this information to be delivered =
to
> =A0=A0the user agent. =A0This document describes these use cases and =
defines
> =A0=A0an extension to the History-Info header field which allows it to =
be
> =A0=A0used to support those cases.
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target
-uri-delivery-01.txt
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
> Content-Type: text/plain<BR>Content-ID:=20
> &lt;2009-3-9102756.I-D@ietf.org&gt;<BR><BR>
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories:=A0http://www.ietf.org/shadow.html
> or=A0ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

From john.elwell@siemens.com  Wed Mar 11 04:34:44 2009
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Date: Wed, 11 Mar 2009 11:34:14 +0000
From: "Elwell, John" <john.elwell@siemens.com>
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Subject: Re: [Sip] Fwd: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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=20


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Shida Schubert
	Sent: 11 March 2009 06:13
	To: sip@ietf.org List
	Cc: Mary Barnes
	Subject: [Sip] Fwd: I-D
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
=09
=09

	We have submitted the updated target-uri draft based on the
comments=20
	submitted to the list and comments received at IETF73.=20

	I have taken over as editor as Jonathan didn't have the cycles
to=20
	update the draft, with Francois, Christer and Hans Erick as
additional=20
	co-authors and great deal of help from Mary. =20
	=20
	The following summarizes the changes made to the target-uri
document
	1. Added use-case for toll-free number back
	2. Added definition of "retarget" operation.=20
	3. Removed a reference to URN
	4. Added a text discussing the difference to P-Called-Party-Id
	5. Changed parameter name from "target" to "istarget"
	=20
	Note, that the target-uri document still contains the normative
text for the
	History-Info header.
[JRE] I don't like this. I would prefer to have all the normative stuff
in one place, if RFC 4244 is being revised anyway.

John
	=20
	In addition, Mary (with Francois as co-author) has submitted a
	rfc4244bis, with the following changes:
	1. Incorporated the normative aspects of the target-uri document
	into the existing normative text in RFC 4244 - the functionality
is=20
	virtually identical (as is some of the text) as the HI based
solution=20
	described in the target-uri document.  It's important that the
solution
	be integrated into RFC 4244 as it MUST work and be based on the
normative
	aspects of RFC 4244.
	2.  Added the use cases from target-uri the the summary
	in the overview of rfc4244bis.
	3. Added an additional requirement to capture the "target-uri"
information.
	4. Fixed an error in the RFC 4244 ABNF and added "retarget"
parameter.
	5. Added a more simplified example.
	=20
	=20
	We had some very long offline exchanges as to the best way
forward and
	remaining work for both documents.
	=20
	Some of the issues identified are:

	::Issues::
	   1. Should we remove the normative text from target-uri and
progress
	       4244bis along with the target-uri document to meet the
chartered
	      SIP WG milestone?=20
	=20
	=20
	   2. Name of the parameter.=20
	      At the last meeting, parameter "target" was said
inappropriate=20
	      because voicemail-uri spec already defines a parameter
called=20
	      "target" which also can be found in hi-entry, thus
potentially=20
	      causing confusion.
	    =20
	      Currently the target-uri draft uses "istarget" and 4244bis
uses
	      "retarget"  but we could never come to=20
	      a consensus on what name is appropriate.  Other
suggestions have
	      included the following:
	      "target-identity" (someone didn't like that "identity" is
also a SIP header)
	      "reg-uri"  (can be paired with "mapped-uri" for item 3
below)
	      "aor"
	      "jibberish"=20
	      etc.
	=20
	      One reason this is so difficult relates to the problem
statement in target-uri in that
	      RFC 3261 doesn't differentiate the mechanism by which the
new
	      (target) Request-URI is selected.  Another issue is that
some of the terminology in
	      RFC 3261 is overloaded - e.g., "forwarding" refers both to
a Proxy
	      which does not have responsibility for the domain of the
request-URI
	      in the incoming request, thus the proxy just "forwards"
the request to
	      the next hop AND "forwarding" is used to describe the
process whereby
	      the outgoing request is built and "forwarded" to the next
hop at which
	      point the proxy does not know how the new request-uri was
selected.
	      RFC 4244 has attempted to clarify the terms and attempts
to use "forward"=20
	      in the context of the former situation and "retarget" for
the case whereby
	      a proxy is responsible for the domain and thus can use a
number of=20
	      mechanism to select the new target for the request - e.g.,
a REGISTRAR,=20
	      configured data, etc. =20

	 3.  Related to the last point in item 2 above, it has been
proposed that
	      we differentiate the hi-entries even more by defining
separate parameters
	     for registered and configured/mapped contacts.=20
	     Currently when the R-URI is translated to a URI which is
either derived=20
	     from location service lookup(registered by UA) or from
mapping=20
	     table, there is no differentiation as to how the URI was
derived once it is
	     added to the list of potential targets.
	    =20
	     The general consensus of the authors of the two documents
was that it may=20
	     be useful for some services to have the hi-entries tagged
with the
	     more specific information.=20
	=20
	     And, of course, this gets us into another naming contest.
In the end, the naming
	     is not so important as long as the term isn't too
overloaded and it is defined
	     precisely in the document(s).
	=20
	We would appreciate WG feedback on these issues and any other
comments on
	the two documents prior to IETF-74.
	=20
	Regards,
	Shida and Mary.



	Begin forwarded message:

		From: Internet-Drafts@ietf.org
		Date: March 10, 2009 2:30:01 AM JST
		To: i-d-announce@ietf.org
		Subject: I-D
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt=20
		Reply-To: internet-drafts@ietf.org

		A New Internet-Draft is available from the on-line
Internet-Drafts=20
		directories.
	=09
	=09
		Title : Delivery of Request-URI Targets to User Agents
		Author(s) : J. Rosenberg, H. van Elburg, C. Holmberg, F.
Audet, S. Schubert
		Filename :
draft-rosenberg-sip-target-uri-delivery-01.txt
		Pages : 16
		Date : 2009-3-9
	=09
		When a Session Initiation Protocol (SIP) proxy receives
a request
		  targeted at a URI identifying a user or resource it is
responsible
		  for, the proxy translates the URI to a registered or
configured
		  contact URI of an agent representing that user or
resource.  In the
		  process, the original URI is removed from the request.
Numerous use
		  cases have arisen which require this information to be
delivered to
		  the user agent.  This document describes these use
cases and defines
		  an extension to the History-Info header field which
allows it to be
		  used to support those cases.
	=09
		A URL for this Internet-Draft is:
=09
http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri-deliv
ery-01.txt
	=09
		Internet-Drafts are also available by anonymous FTP at:
		ftp://ftp.ietf.org/internet-drafts/
	=09
		Below is the data which will enable a MIME compliant
mail reader
		implementation to automatically retrieve the ASCII
version of the
		Internet-Draft.
	=09

	Content-Type: text/plain<BR>Content-ID:
&lt;2009-3-9102756.I-D@ietf.org&gt;<BR><BR>=20

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		or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


From christer.holmberg@ericsson.com  Wed Mar 11 05:26:23 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens.com>, "Hadriel Kaplan" <HKaplan@acmepacket.com>, "Shida Schubert" <shida@ntt-at.com>, <sip@ietf.org>
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Subject: Re: [Sip] Fwd: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Hi,

We had a looooong discussion about that (ua-loose-route vs Target), and =
I really hope we should not go back there now.

I think we should focus on whether we want to have this in H-I, as =
currently proposed, or as a separate header.

Regards,

Christer
=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Elwell, John
> Sent: 11. maaliskuuta 2009 13:32
> To: Hadriel Kaplan; Shida Schubert; sip@ietf.org
> Cc: Mary Barnes
> Subject: Re: [Sip] Fwd: I-D=20
> ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>=20
> Just to expand on what Hadriel says, the SIP Forum is=20
> addressing the case where a request is delivered by a SIP=20
> Service Provider to a SIP-PBX, the SIP-PBX having registered=20
> with the SIP-SP. The SIP-PBX needs to be in possession of the=20
> target, in order to route onwards correctly within the=20
> enterprise. If only the SIP-PBX's contact URI is available in=20
> the Request-URI, the SIP-PBX would have to look elsewhere for=20
> the target. In accordance with recent SIP WG agreements, this=20
> would be the History-Info header field, but that requires the=20
> present extension.
>=20
> Advantages of putting the target in the Request-URI for this=20
> particular situation:
> - This is where the SIP-PBX would normally expect to find the=20
> target when a request is received from elsewhere (e.g.,=20
> another SIP-PBX within the enterprise), so it is reasonable=20
> also to expect it there when received from a SIP SP.
> - SIPConnect 1.1 will not require History-Info for any other=20
> purpose, so it would make it a mandatory specification for=20
> SIPConnect 1.1 just in order to solve this particular=20
> problem, thereby raising the bar for SPs and SIP-PBXs wanting=20
> to comply with the spec.
> - We would need to define procedures for what to do if=20
> History-Info is absent from a received request or if there is=20
> no entry marked as 'istarget'.
> - SIPConnect 1.1 is due for completion in the next few weeks,=20
> and cannot wait the usual IETF cycle time for History-Info=20
> update to hit the RFC Editor's queue.
> - If History-Info is used anyway (for its original purpose),=20
> the history revealed when a request comes from a SIP-SP via a=20
> SIP-PBX to another proxy and finally to a UAS would be=20
> unnecessarily complicated and therefore confusing, because=20
> the target would be replaced by a contact and would then=20
> appear again at the next hop.
>=20
> John
> =20
>=20
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of=20
> > Hadriel Kaplan
> > Sent: 11 March 2009 06:55
> > To: Shida Schubert; sip@ietf.org List
> > Cc: Mary Barnes
> > Subject: Re: [Sip] Fwd: I-D
> > ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
> >=20
> >=20
> > At the risk of opening up a topic we have already agreed on, but in=20
> > the spirit of open communication... [how's that for an opener!?]
> >=20
> > Are we absolutely sure we want to put this target-URI in the=20
> > History-Info header?
> >=20
> > I am not asking this because I think it's hard.  I am=20
> asking because I=20
> > am not sure it will be used.  Several of us from this same WG are=20
> > involved in the SIP-Forum's SIP-Connect profile, and when=20
> faced with=20
> > either using this new History-Info mechanism for a=20
> target-uri delivery=20
> > issue, vs.
> > not, we chose not to.  And we were mostly the same people who liked=20
> > the idea of putting it in History-Info in the IETF!
> > (myself included)  But I am concerned that when given a reasonable=20
> > opportunity to use a mechanism we ourselves promote when wearing a=20
> > different hat, we chose not to use it in practice.  It doesn't bode=20
> > well. :(
> >=20
> > -hadriel
> > p.s. sorry Mary for raising this, but there isn't much=20
> email traffic=20
> > anyway. :)
> >=20
> > ________________________________________
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of=20
> > Shida Schubert
> > Sent: Wednesday, March 11, 2009 2:13 AM
> > To: sip@ietf.org List
> >=20
> > We have submitted the updated target-uri draft based on the=20
> comments=20
> > submitted to the list and comments received at IETF73.
> >=20
> > I have taken over as editor as Jonathan didn't have the cycles to=20
> > update the draft, with Francois, Christer and Hans Erick as=20
> additional=20
> > co-authors and great deal of help from Mary.
> > =A0
> > The following summarizes the changes made to the target-uri=20
> document=20
> > 1. Added use-case for toll-free number back 2. Added definition of=20
> > "retarget" operation.
> > 3. Removed a reference to URN
> > 4. Added a text discussing the difference to P-Called-Party-Id 5.=20
> > Changed parameter name from "target" to "istarget"
> > =A0
> > Note, that the target-uri document still contains the=20
> normative text=20
> > for the History-Info header.
> > =A0
> > In addition, Mary (with Francois as co-author) has submitted a=20
> > rfc4244bis, with the following changes:
> > 1. Incorporated the normative aspects of the target-uri=20
> document into=20
> > the existing normative text in RFC 4244 - the functionality is=20
> > virtually identical (as is some of the text) as the HI=20
> based solution=20
> > described in the target-uri document. =A0It's important that the=20
> > solution be integrated into RFC 4244 as it MUST work and be=20
> based on=20
> > the normative aspects of RFC 4244.
> > 2. =A0Added the use cases from target-uri the the summary in the=20
> > overview of rfc4244bis.
> > 3. Added an additional requirement to capture the "target-uri"=20
> > information.
> > 4. Fixed an error in the RFC 4244 ABNF and added "retarget"=20
> parameter.
> > 5. Added a more simplified example.
> > =A0
> > =A0
> > We had some very long offline exchanges as to the best way=20
> forward and=20
> > remaining work for both documents.
> > =A0
> > Some of the issues identified are:
> >=20
> > ::Issues::
> > =A0=A0 1. Should we remove the normative text from target-uri=20
> and progress
> > =A0=A0 =A0 =A0 4244bis along with the target-uri document to meet =
the=20
> > chartered
> > =A0=A0 =A0 =A0SIP WG milestone?
> > =A0
> > =A0
> > =A0=A0 2. Name of the parameter.
> > =A0=A0 =A0 =A0At the last meeting, parameter "target" was said =
inappropriate
> > =A0=A0 =A0 =A0because voicemail-uri spec already defines a parameter =
called
> > =A0=A0 =A0 =A0"target" which also can be found in hi-entry, thus =
potentially
> > =A0=A0 =A0 =A0causing confusion.
> > =A0=A0 =A0=A0
> > =A0=A0 =A0 =A0Currently the target-uri draft uses "istarget" and=20
> 4244bis uses
> > =A0=A0 =A0 =A0"retarget" =A0but we could never come to
> > =A0=A0 =A0 =A0a consensus on what name is appropriate. =A0Other=20
> suggestions have
> > =A0=A0 =A0 =A0included the following:
> > =A0=A0 =A0 =A0"target-identity" (someone didn't like that "identity" =

> > is also a SIP header)
> > =A0=A0 =A0 =A0"reg-uri" =A0(can be paired with "mapped-uri" for item =
3 below)
> > =A0=A0 =A0 =A0"aor"
> > =A0=A0 =A0 =A0"jibberish"
> > =A0=A0 =A0 =A0etc.
> > =A0
> > =A0=A0 =A0 =A0One reason this is so difficult relates to the=20
> problem statement=20
> > in target-uri in that
> > =A0=A0 =A0 =A0RFC 3261 doesn't differentiate the mechanism by which =
the new
> > =A0=A0 =A0 =A0(target) Request-URI is selected. =A0Another issue is=20
> that some of=20
> > the terminology in
> > =A0=A0 =A0 =A0RFC 3261 is overloaded - e.g., "forwarding" refers =
both to a=20
> > Proxy
> > =A0=A0 =A0 =A0which does not have responsibility for the domain of =
the=20
> > request-URI
> > =A0=A0 =A0 =A0in the incoming request, thus the proxy just =
"forwards"=20
> > the request to
> > =A0=A0 =A0 =A0the next hop AND "forwarding" is used to describe the =
process=20
> > whereby
> > =A0=A0 =A0 =A0the outgoing request is built and "forwarded" to the=20
> next hop at=20
> > which
> > =A0=A0 =A0 =A0point the proxy does not know how the new request-uri =
was=20
> > selected.
> > =A0=A0 =A0 =A0RFC 4244 has attempted to clarify the terms and=20
> attempts to use=20
> > "forward"
> > =A0=A0 =A0 =A0in the context of the former situation and "retarget"=20
> > for the case whereby
> > =A0=A0 =A0 =A0a proxy is responsible for the domain and thus can=20
> use a number=20
> > of
> > =A0=A0 =A0 =A0mechanism to select the new target for the request - =
e.g., a=20
> > REGISTRAR,
> > =A0=A0 =A0 =A0configured data, etc.
> >=20
> > =A03. =A0Related to the last point in item 2 above, it has been=20
> proposed=20
> > that
> > =A0=A0 =A0 =A0we differentiate the hi-entries even more by defining=20
> separate=20
> > parameters
> > =A0=A0 =A0 for registered and configured/mapped contacts.
> > =A0=A0 =A0 Currently when the R-URI is translated to a URI which=20
> is either=20
> > derived
> > =A0=A0 =A0 from location service lookup(registered by UA) or from =
mapping
> > =A0=A0 =A0 table, there is no differentiation as to how the URI=20
> was derived=20
> > once it is
> > =A0=A0 =A0 added to the list of potential targets.
> > =A0=A0 =A0=A0
> > =A0=A0 =A0 The general consensus of the authors of the two documents =
was=20
> > that it may
> > =A0=A0 =A0 be useful for some services to have the hi-entries tagged =
with=20
> > the
> > =A0=A0 =A0 more specific information.
> > =A0
> > =A0=A0 =A0 And, of course, this gets us into another naming=20
> contest. In the=20
> > end, the naming
> > =A0=A0 =A0 is not so important as long as the term isn't too=20
> overloaded and=20
> > it is defined
> > =A0=A0 =A0 precisely in the document(s).
> > =A0
> > We would appreciate WG feedback on these issues and any=20
> other comments=20
> > on the two documents prior to IETF-74.
> > =A0
> > Regards,
> > Shida and Mary.
> >=20
> >=20
> >=20
> > Begin forwarded message:
> > From:=A0Internet-Drafts@ietf.org
> > Date:=A0March 10, 2009 2:30:01 AM JST
> > To:=A0i-d-announce@ietf.org
> > Subject:=A0I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
> > Reply-To:=A0internet-drafts@ietf.org
> >=20
> > A New Internet-Draft is available from the on-line Internet-Drafts=20
> > directories.
> >=20
> >=20
> > 	Title		: Delivery of Request-URI Targets to User Agents
> > 	Author(s)	: J. Rosenberg, H. van Elburg, C.=20
> > Holmberg, F. Audet, S. Schubert
> > 	Filename	: draft-rosenberg-sip-target-uri-delivery-01.txt
> > 	Pages		: 16
> > 	Date		: 2009-3-9
> > =09
> > When a Session Initiation Protocol (SIP) proxy receives a request
> > =A0=A0targeted at a URI identifying a user or resource it is =
responsible
> > =A0=A0for, the proxy translates the URI to a registered or =
configured
> > =A0=A0contact URI of an agent representing that user or=20
> resource. =A0In the
> > =A0=A0process, the original URI is removed from the request. =A0
> Numerous use
> > =A0=A0cases have arisen which require this information to be=20
> delivered to
> > =A0=A0the user agent. =A0This document describes these use cases=20
> and defines
> > =A0=A0an extension to the History-Info header field which=20
> allows it to be
> > =A0=A0used to support those cases.
> >=20
> > A URL for this Internet-Draft is:
> > http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target
> -uri-delivery-01.txt
> >=20
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >=20
> > Below is the data which will enable a MIME compliant mail reader=20
> > implementation to automatically retrieve the ASCII version of the=20
> > Internet-Draft.
> > Content-Type: text/plain<BR>Content-ID:=20
> > &lt;2009-3-9102756.I-D@ietf.org&gt;<BR><BR>
> > _______________________________________________
> > I-D-Announce mailing list
> > I-D-Announce@ietf.org
> > https://www.ietf.org/mailman/listinfo/i-d-announce
> > Internet-Draft directories:=A0http://www.ietf.org/shadow.html or=A0
> > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

From christer.holmberg@ericsson.com  Wed Mar 11 05:55:09 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Mary Barnes" <mary.barnes@nortel.com>, "Francois Audet" <audet@nortel.com>
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Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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Hi,

My first comment/question on the draft: eventhough there are lots of
examples, do we really need 46 pages? Are we really making History-Info
more clear if we still need that many pages?

I know my comment is not that useful, since I don't propose what to
change/remove, but just as a general comment.

Regards,

Christer




> -----Original Message-----
> From: i-d-announce-bounces@ietf.org
> [mailto:i-d-announce-bounces@ietf.org] On Behalf Of=20
> Internet-Drafts@ietf.org
> Sent: 04 March 2009 15:15
> To: i-d-announce@ietf.org
> Subject: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20
>=20
> A New Internet-Draft is available from the on-line=20
> Internet-Drafts directories.
>=20
> 	Title           : An Extension to the Session Initiation
> Protocol (SIP) for Request History Information
> 	Author(s)       : M. Barnes, F. Audet
> 	Filename        : draft-barnes-sip-rfc4244bis-00.txt
> 	Pages           : 49
> 	Date            : 2009-03-04
>=20
> This document defines a standard mechanism for capturing the=20
> history information associated with a Session Initiation=20
> Protocol (SIP) request.  This capability enables many=20
> enhanced services by providing the information as to how and=20
> why a call arrives at a specific application or user.  This=20
> document defines a new optional SIP header, History-Info, for=20
> capturing the history information in requests.
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-barnes-sip-rfc4244bis-00.txt
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> Below is the data which will enable a MIME compliant mail=20
> reader implementation to automatically retrieve the ASCII=20
> version of the Internet-Draft.
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

From christer.holmberg@ericsson.com  Wed Mar 11 06:19:34 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Bob Penfield" <BPenfield@acmepacket.com>
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Subject: Re: [Sip] I-D Action:draft-ietf-sip-199-06.txt
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Hi Bob,=20

>>>In fact, if the proxy is keeping track of early dialogs, under what=20
>>>circumstances would it not be "able to identify additional early=20
>>>dialogs which are terminated by the same non-2xx final response"?
>>
>>Just because the proxy keeps track of early dialogs, it doesn't=20
>>necessarily know that an error response on early dialog X also=20
>>terminates early dialog Y. In order to know that, the forking proxy=20
>>needs to know that early dialogs X and Y come from the same forking=20
>>proxy further down the path.
>>
>=20
>Of course it can. If the proxy forwarded the request to Z,=20
>and Z forked to X and Y, the provisional responses from X and=20
>Y that created the early dialogs would have the same=20
>branch-id. The error response from Z would have that same=20
>branch-id. The proxy would know that all early dialogs=20
>created by a 1xx with that branch-id would have terminated=20
>(i.e. both X and Y).

But, the branch-id is part of the Via, which is removed by each entity
when forwarding the response. So, the forking proxy is never going to
get the branch-id inserted by Z, is it?


>>>Section 8 (199 Early Dialog Terminated) seems out of place.
>>>Shouldn't this be in the introduction? or in an "Overview Of=20
>>>Operation" between sections 3 and 4?
>>
>>I have tried to use other similar drafts as "template".
>>
>>
>>>The text in Section 9 (Usage with SDP offer/answer) should be in=20
>>>section 5 (User Agent Server behavior).
>>
>>I think it is good to have a separate chapter. Of course, I could
refer=20
>>to chapter 9 from chapter 5.
>=20
>My point is that all too often behavior requirements (MUST,=20
>SHOULD, MAY, etc.) get buried is separate sections and folks=20
>can miss them. If all the requirements are in the applicable=20
>"behavior" section, they are easy to find and developers are less
likely=20
>to miss them. If you want it to be a separate section, make=20
>it 5.1. All UAS requirements should be in section 5 or a=20
>subsection thereof.
>
>>>The statement in section 10 recommending that the UAS send
>>>199 unreliably should be in section 5.
>>>
>>>The statement in section 10 forbidding proxies from sending
>>>199 reliably should be in section 6 (Proxy behavior).
>>
>>Same as for offer/answer.
>>
>>Since the 199 procedures related to offer/answer and reliable
responses=20
>>are special, I think it is good to have specific chapters for it.
>>=20
>The requirements/restrictions for reliable provisional=20
>responses are very important. They could easily be missed by=20
>having them in a separate chapter in the back of the=20
>document. Again, all UAS behavior requirements should be in=20
>section 5 (or sub-section) and all proxy behavior=20
>requirements should be in chapter 6 (or sub-section).

I'll think of something. But, as you ok with the text as such? It's only
the location of the text which you have an issue with?


Regards,

Christer


From mary.barnes@nortel.com  Wed Mar 11 06:21:10 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>, "Francois Audet" <audet@nortel.com>
Cc: sip@ietf.org
Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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Hi Christer,=20

Yes, we can remove a lot of the extra examples in the appendix - that
was what we had in mind for the next revision, since they all need to be
updated (all of them are dated in that they use strict routing). Also,
since there is now a document like target-uri that has better use cases,
we really don't need so many in the protocol spec. So, perhaps, you can
consider adding some detailed flows to target-uri? =20

And, you have to keep in mind that the normative text in 4244 is
comprised of about 12 pages (7-18), so yes the size of the document is
misleading in terms of what really has to be implemented

Mary.=20

-----Original Message-----
From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
Sent: Wednesday, March 11, 2009 7:55 AM
To: Barnes, Mary (RICH2:AR00); Audet, Francois (SC100:3055)
Cc: sip@ietf.org
Subject: RE: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt


Hi,

My first comment/question on the draft: eventhough there are lots of
examples, do we really need 46 pages? Are we really making History-Info
more clear if we still need that many pages?

I know my comment is not that useful, since I don't propose what to
change/remove, but just as a general comment.

Regards,

Christer




> -----Original Message-----
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> [mailto:i-d-announce-bounces@ietf.org] On Behalf Of=20
> Internet-Drafts@ietf.org
> Sent: 04 March 2009 15:15
> To: i-d-announce@ietf.org
> Subject: I-D Action:draft-barnes-sip-rfc4244bis-00.txt
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts=20
> directories.
>=20
> 	Title           : An Extension to the Session Initiation
> Protocol (SIP) for Request History Information
> 	Author(s)       : M. Barnes, F. Audet
> 	Filename        : draft-barnes-sip-rfc4244bis-00.txt
> 	Pages           : 49
> 	Date            : 2009-03-04
>=20
> This document defines a standard mechanism for capturing the history=20
> information associated with a Session Initiation Protocol (SIP)=20
> request.  This capability enables many enhanced services by providing=20
> the information as to how and why a call arrives at a specific=20
> application or user.  This document defines a new optional SIP header,

> History-Info, for capturing the history information in requests.
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-barnes-sip-rfc4244bis-00.txt
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> Below is the data which will enable a MIME compliant mail reader=20
> implementation to automatically retrieve the ASCII version of the=20
> Internet-Draft.
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> sipping@ietf.org for new developments on the application of sip
>=20

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Date: Wed, 11 Mar 2009 14:49:50 +0100 (CET)
From: "Nils Ohlmeier" <lists@ohlmeier.org>
To: "Raphael Coeffic" <rco@iptel.org>
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Cc: Nils Ohlmeier <lists@ohlmeier.org>, "sip@ietf.org" <sip@ietf.org>, Hadriel Kaplan <hkaplan@acmepacket.com>
Subject: Re: [Sip] draft-state-sip-relay-attack-00
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> Hadriel Kaplan wrote:
>>
>>> -----Original Message-----
>>> From: Raphael Coeffic [mailto:rco@iptel.org]
>>> Sent: Tuesday, March 10, 2009 6:06 AM
>>>
>>> Hadriel Kaplan wrote:
>>>
>>> Forcing registrations is the path that IMS went for, I believe. But if
>>> you want to take advantage of this, you may have to deploy a little
>>> more
>>> IMS than you'd like to.
>>>
>>
>> The concept and practice of "forcing" UA's, which have AoR's belonging
>> to a domain, to REGISTER their contact binding for those AoR's in order
>> to make calls, has been around a lot longer than IMS has.  Invoking the
>> "that sounds like IMS" defense is a cheap guilt-by-association argument,
>> and not relevant.  One might as well argue that using INVITE's or RTP
>> "is the path that IMS went for".
>>
>> You keep making it sound like sending a SIP request which does not
>> create a dialog is somehow a difficult mechanism to implement, or
>> "forces" people to do something they don't want to.  What is it forcing?
>>
>>
> Well, it is not required as per RFC 3261 ;-)
>
>> You're looking for a means of authenticating a UA without being subject
>> to a relay attack.  Using a different SIP Method name, which can only be
>> initiated by the client with the credentials, and one that UA's cannot
>> be baited to provide by random third parties, is one solution to the
>> problem.  If you don't like the string "REGISTER", then how about
>> "OPTIONS"?
>>
>>
> No, I think you missunderstood me. I still think that requiring a
> registration and checking the source IP and port against the registered
> contact is a fair measure to take. However, it introduces more traffic
> in some scenarios, where it is not necessarily needed. REGISTERs could
> become the "background load" in the servers. This also puts a lot of
> state into some proxies that does not need it, etc...
> My concern is just that it seems like there is no better solution.

Sure to create a Contact binding via an authenticated REGISTER is one
possible solution.
But I think this only works because you rely on the fact that REGISTERs
are not being retargeted like INVITEs. Which then in the end means you
simply can forget about authenticating all other requests, as long as the
Contact or IP/port was successfully authenticated via the "one hop"
REGISTER method (refer-to: IMS).
The advantage of this solution is clearly that it can be easily deployed
by hopefully most of the service providers today.

But I believe the proper technical solution would be that the server
authenticates itself in the challenge, plus adding protected informations
to the challenge which allows the receiver of the challenge to verify that
this challenge was/is targeted to himself.
The dis-advantage is clearly that this would be only possible with
extensions of the existing protocols. But we would/might gain other
benefits by such a solution as well.

Regards
  Nils Ohlmeier


From mary.barnes@nortel.com  Wed Mar 11 09:23:53 2009
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I don't think it should cause so much concern as I think it just
requires clarifying text and we just need to agree to more precise
terms: e.g., qualifying the use of "forward" with either "next hop
forwarding" or "request forwarding" for this particular case. As long as
we can be precise in 4244bis, then that should suffice for now, but
whenever 3261 is updated or bised, then I think the clarifications
should be included.

Mary.=20

-----Original Message-----
From: Dean Willis [mailto:dean.willis@softarmor.com]=20
Sent: Wednesday, March 11, 2009 1:38 AM
To: Shida Schubert
Cc: sip@ietf.org List; Barnes, Mary (RICH2:AR00)
Subject: Re: [Sip] Fwd: I-D
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


On Mar 11, 2009, at 1:13 AM, Shida Schubert wrote:

>
>       One reason this is so difficult relates to the problem statement

> in target-uri in that
>       RFC 3261 doesn't differentiate the mechanism by which the new
>       (target) Request-URI is selected.  Another issue is that some of

> the terminology in
>       RFC 3261 is overloaded - e.g., "forwarding" refers both to a=20
> Proxy
>       which does not have responsibility for the domain of the=20
> request-URI
>       in the incoming request, thus the proxy just "forwards" the=20
> request to
>       the next hop AND "forwarding" is used to describe the process=20
> whereby
>       the outgoing request is built and "forwarded" to the next hop at

> which
>       point the proxy does not know how the new request-uri was=20
> selected.
>       RFC 4244 has attempted to clarify the terms and attempts to use=20
> "forward"
>       in the context of the former situation and "retarget" for the=20
> case whereby
>       a proxy is responsible for the domain and thus can use a number=20
> of
>       mechanism to select the new target for the request - e.g., a=20
> REGISTRAR,
>       configured data, etc.
>

Thanks, Shida.

I personally would not be averse to an Essential Correction that
corrects the terminology in RFC 3261, but I suspect that will make many
heads spin around elsewhere in the WG.

--
Dean

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Thread-Topic: [Sip] Fwd: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>, "Shida Schubert" <shida@ntt-at.com>, <sip@ietf.org>
Subject: Re: [Sip] Fwd: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Yeah, we can define a new header and just ignore the use of a basic SIP =
building block that the WG spent 2.5 years  (note this doesn't include =
the 2+ years spent on the doc as an individual and as a SIPPING WG =
document ;)  And, while we're at it, let's just go ahead and obsolete =
4244 altogether and standardize diversion (something the WG rejected =
almost a decade ago). That will make one vendor and a handful of SPs =
very happy and those of us that have implemented the specs can ripout =
some code after while when we have some spare change to pay folks to do =
so (and of course, this is after the folks that have not implemented =
diversion actually implement it - there are actually some of those out =
there).=20

As an aside, this is another fine example as to why we don't seem to =
make great progress or produce specs that are deemed useful - we agree =
on a way forward and then change our minds later (sometimes for good =
reasons and sometimes because a shortcut seems easier or it's just =
arbitrary based on the phase of the moon it seems). =20

Mary.=20

-----Original Message-----
From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]=20
Sent: Wednesday, March 11, 2009 1:55 AM
To: Shida Schubert; sip@ietf.org List
Cc: Barnes, Mary (RICH2:AR00)
Subject: RE: [Sip] Fwd: I-D =
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


At the risk of opening up a topic we have already agreed on, but in the =
spirit of open communication... [how's that for an opener!?]

Are we absolutely sure we want to put this target-URI in the =
History-Info header?

I am not asking this because I think it's hard.  I am asking because I =
am not sure it will be used.  Several of us from this same WG are =
involved in the SIP-Forum's SIP-Connect profile, and when faced with =
either using this new History-Info mechanism for a target-uri delivery =
issue, vs. not, we chose not to.  And we were mostly the same people who =
liked the idea of putting it in History-Info in the IETF! (myself =
included)  But I am concerned that when given a reasonable opportunity =
to use a mechanism we ourselves promote when wearing a different hat, we =
chose not to use it in practice.  It doesn't bode well. :(

-hadriel
p.s. sorry Mary for raising this, but there isn't much email traffic =
anyway. :)

________________________________________
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
Shida Schubert
Sent: Wednesday, March 11, 2009 2:13 AM
To: sip@ietf.org List

We have submitted the updated target-uri draft based on the comments =
submitted to the list and comments received at IETF73.=A0

I have taken over as editor as Jonathan didn't have the cycles to update =
the draft, with Francois, Christer and Hans Erick as additional =
co-authors and great deal of help from Mary. =A0
=A0
The following summarizes the changes made to the target-uri document 1. =
Added use-case for toll-free number back 2. Added definition of =
"retarget" operation. 3. Removed a reference to URN 4. Added a text =
discussing the difference to P-Called-Party-Id 5. Changed parameter name =
from "target" to "istarget"
=A0
Note, that the target-uri document still contains the normative text for =
the History-Info header.
=A0
In addition, Mary (with Francois as co-author) has submitted a =
rfc4244bis, with the following changes:
1. Incorporated the normative aspects of the target-uri document into =
the existing normative text in RFC 4244 - the functionality is virtually =
identical (as is some of the text) as the HI based solution described in =
the target-uri document. =A0It's important that the solution be =
integrated into RFC 4244 as it MUST work and be based on the normative =
aspects of RFC 4244.
2. =A0Added the use cases from target-uri the the summary in the =
overview of rfc4244bis.
3. Added an additional requirement to capture the "target-uri" =
information.
4. Fixed an error in the RFC 4244 ABNF and added "retarget" parameter.
5. Added a more simplified example.
=A0
=A0
We had some very long offline exchanges as to the best way forward and
remaining work for both documents.
=A0
Some of the issues identified are:

::Issues::
=A0=A0 1. Should we remove the normative text from target-uri and =
progress
=A0=A0 =A0 =A0 4244bis along with the target-uri document to meet the =
chartered
=A0=A0 =A0 =A0SIP WG milestone?=A0
=A0
=A0
=A0=A0 2. Name of the parameter.=A0
=A0=A0 =A0 =A0At the last meeting, parameter "target" was said =
inappropriate=A0
=A0=A0 =A0 =A0because voicemail-uri spec already defines a parameter =
called=A0
=A0=A0 =A0 =A0"target" which also can be found in hi-entry, thus =
potentially=A0
=A0=A0 =A0 =A0causing confusion.
=A0=A0 =A0=A0
=A0=A0 =A0 =A0Currently the target-uri draft uses "istarget" and 4244bis =
uses
=A0=A0 =A0 =A0"retarget" =A0but we could never come to=A0
=A0=A0 =A0 =A0a consensus on what name is appropriate. =A0Other =
suggestions have
=A0=A0 =A0 =A0included the following:
=A0=A0 =A0 =A0"target-identity" (someone didn't like that "identity" is =
also a SIP header)
=A0=A0 =A0 =A0"reg-uri" =A0(can be paired with "mapped-uri" for item 3 =
below)
=A0=A0 =A0 =A0"aor"
=A0=A0 =A0 =A0"jibberish"=A0
=A0=A0 =A0 =A0etc.
=A0
=A0=A0 =A0 =A0One reason this is so difficult relates to the problem =
statement in target-uri in that
=A0=A0 =A0 =A0RFC 3261 doesn't differentiate the mechanism by which the =
new
=A0=A0 =A0 =A0(target) Request-URI is selected. =A0Another issue is that =
some of the terminology in
=A0=A0 =A0 =A0RFC 3261 is overloaded - e.g., "forwarding" refers both to =
a Proxy
=A0=A0 =A0 =A0which does not have responsibility for the domain of the =
request-URI
=A0=A0 =A0 =A0in the incoming request, thus the proxy just "forwards" =
the request to
=A0=A0 =A0 =A0the next hop AND "forwarding" is used to describe the =
process whereby
=A0=A0 =A0 =A0the outgoing request is built and "forwarded" to the next =
hop at which
=A0=A0 =A0 =A0point the proxy does not know how the new request-uri was =
selected.
=A0=A0 =A0 =A0RFC 4244 has attempted to clarify the terms and attempts =
to use "forward"=A0
=A0=A0 =A0 =A0in the context of the former situation and "retarget" for =
the case whereby
=A0=A0 =A0 =A0a proxy is responsible for the domain and thus can use a =
number of=A0
=A0=A0 =A0 =A0mechanism to select the new target for the request - e.g., =
a REGISTRAR,=A0
=A0=A0 =A0 =A0configured data, etc. =A0

=A03. =A0Related to the last point in item 2 above, it has been proposed =
that
=A0=A0 =A0 =A0we differentiate the hi-entries even more by defining =
separate parameters
=A0=A0 =A0 for registered and configured/mapped contacts.=A0
=A0=A0 =A0 Currently when the R-URI is translated to a URI which is =
either derived=A0
=A0=A0 =A0 from location service lookup(registered by UA) or from =
mapping=A0
=A0=A0 =A0 table, there is no differentiation as to how the URI was =
derived once it is
=A0=A0 =A0 added to the list of potential targets.
=A0=A0 =A0=A0
=A0=A0 =A0 The general consensus of the authors of the two documents was =
that it may=A0
=A0=A0 =A0 be useful for some services to have the hi-entries tagged =
with the
=A0=A0 =A0 more specific information.=A0
=A0
=A0=A0 =A0 And, of course, this gets us into another naming contest. In =
the end, the naming
=A0=A0 =A0 is not so important as long as the term isn't too overloaded =
and it is defined
=A0=A0 =A0 precisely in the document(s).
=A0
We would appreciate WG feedback on these issues and any other comments =
on
the two documents prior to IETF-74.
=A0
Regards,
Shida and Mary.



Begin forwarded message:
From:=A0Internet-Drafts@ietf.org
Date:=A0March 10, 2009 2:30:01 AM JST
To:=A0i-d-announce@ietf.org
Subject:=A0I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt=A0
Reply-To:=A0internet-drafts@ietf.org

A New Internet-Draft is available from the on-line Internet-Drafts=A0
directories.


	Title		: Delivery of Request-URI Targets to User Agents
	Author(s)	: J. Rosenberg, H. van Elburg, C. Holmberg, F. Audet, S. =
Schubert
	Filename	: draft-rosenberg-sip-target-uri-delivery-01.txt
	Pages		: 16
	Date		: 2009-3-9
=09
When a Session Initiation Protocol (SIP) proxy receives a request
=A0=A0targeted at a URI identifying a user or resource it is responsible
=A0=A0for, the proxy translates the URI to a registered or configured
=A0=A0contact URI of an agent representing that user or resource. =A0In =
the
=A0=A0process, the original URI is removed from the request. =A0Numerous =
use
=A0=A0cases have arisen which require this information to be delivered =
to
=A0=A0the user agent. =A0This document describes these use cases and =
defines
=A0=A0an extension to the History-Info header field which allows it to =
be
=A0=A0used to support those cases.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri-delive=
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "Elwell, John" <john.elwell@siemens.com>, Shida Schubert <shida@ntt-at.com>, "sip@ietf.org" <sip@ietf.org>
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Subject: Re: [Sip] Fwd: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Wednesday, March 11, 2009 8:27 AM
> To: Elwell, John; Hadriel Kaplan; Shida Schubert; sip@ietf.org
>=20
> We had a looooong discussion about that (ua-loose-route vs Target), and I
> really hope we should not go back there now.

I wouldn't have, were it not for the Sip-Forum experience.  If a bunch of p=
eople who spent time on the topic don't even want to use the mechanism, wha=
t hopes do we have that it will succeed in the marketplace? =20

And don't get me wrong - it could have been in a "Target" header too and we=
 wouldn't have used that either, I think.  Although part of the reason is b=
ecause we know IP-PBX's actually support Jonathan's original ua-loose-route=
 concept (just without the Route header part), whereas I'm not so sure what=
 legacy endpoints will do with an AoR-req-uri; nor whether the req-uri woul=
d ever survive end-to-end in Peering cases. =20

So I was in favor of doing a separate Header because it was "safer", in ter=
ms of not breaking interoperability - but it's less chance for actually get=
ting used, which in a way does break something too (customer expectations).=
  Ugh.  Screwed either way.

-hadriel

From mary.barnes@nortel.com  Wed Mar 11 09:49:57 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
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Ian,

This is an interesting question. I need to think about it some more, but
I don't like the "none" idea as it really must be up to the entity that
added the Privacy header to the request, as to whether it wants the HI
entries that it adds to go outside a domain. My initial thought is that
we can't overrule the overall Privacy header. The thing is that the
Privacy header doesn't preclude gathering the information and using
within a domain AND if it were to not include an hi-entry when the
request leaves the domain for which the proxy is responsible, the
recipient can add the hi-entry for the current request-uri before it
adds the new hi-entry. =20

Your PSTN example doesn't strictly map obviously to a SIP model and I
would think you might consider the PSTN hop to be within the same domain
for which the proxy (or gw, I guess) is responsible OR consider this a
walled garden whereby the Privacy header doesn't apply. Which brings me
to a more general question as to how you all deal with other headers
when you do your mapping to the PSTN when there is a Privacy header?  It
would seem this problem isn't unique to History-Info, although I know
little about the details of PSTN I/W.

Regards,
Mary.=20

-----Original Message-----
From: Ian Elz [mailto:ian.elz@ericsson.com]=20
Sent: Wednesday, March 11, 2009 4:54 AM
To: Barnes, Mary (RICH2:AR00); Audet, Francois (SC100:3055)
Cc: sip@ietf.org
Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

Mary, Francois,

Section 4.1 of the draft allows for the addition of the Privacy header
parameter with a value of "history" to be included with the
hi-targeted-to-uri.

Can this be extended to also allow the value "none".

A node adding a H-I entry is allowed to specify the privacy value
"history" either in the Privacy header or as a header parameter
associated with the hi-targeted-to-uri. If the former action is taken by
one node this results in privacy for all history info header entries
even if this is not required. There is no way for privacy of an
individual H-I entry to be set to none if no Privacy is required for the
uri.

This creates a specific problem when interworking with the PSTN where
the privacy is associated with each E.164 number included in the
protocol. If an INVITE is being mapped to ISUP and the H-I entries are
being used to map to redirection numbers then a single Privacy header
with the value "history" will result in all numbers being restricted.

If the Privacy header parameter could include the value "none" then this
would explicitly indicate that the associated uri was allowed to be
presented.

The ability to explicitly allow presentation of specific H-I entries may
also be useful in pure sip implementations.

I don't believe that this change would create any backward compatibility
issues. The existing implementations will continue as deployed but new
implementations could explicitly set no privacy for a specific uri.

There are no security issues other than those already defined in the
draft that an intervening node could modify an existing H-I entry. The
Privacy header value of "none" should only be included by the node in
the same manner as the existing inclusion of the "history" value.

Ian Elz


-----Original Message-----
From: i-d-announce-bounces@ietf.org
[mailto:i-d-announce-bounces@ietf.org] On Behalf Of
Internet-Drafts@ietf.org
Sent: 04 March 2009 15:15
To: i-d-announce@ietf.org
Subject: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

A New Internet-Draft is available from the on-line Internet-Drafts
directories.

	Title           : An Extension to the Session Initiation
Protocol (SIP) for Request History Information
	Author(s)       : M. Barnes, F. Audet
	Filename        : draft-barnes-sip-rfc4244bis-00.txt
	Pages           : 49
	Date            : 2009-03-04

This document defines a standard mechanism for capturing the history
information associated with a Session Initiation Protocol (SIP) request.
This capability enables many enhanced services by providing the
information as to how and why a call arrives at a specific application
or user.  This document defines a new optional SIP header, History-Info,
for capturing the history information in requests.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-barnes-sip-rfc4244bis-00.txt

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From mary.barnes@nortel.com  Wed Mar 11 10:00:09 2009
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Thread-Topic: Terminology (was RE: [Sip] Fwd: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Hans Erik van Elburg" <ietf.hanserik@gmail.com>, "Shida Schubert" <shida@ntt-at.com>
Cc: sip@ietf.org
Subject: [Sip] Terminology (was RE: Fwd: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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And, I think the big issue with the terminology is that 4244bis is
application agnostic, so it is tagging what happens to the URIs (why/how
the specific URI was overwritten).  Whereas, specific applications can
derive information, such as what "is" the real "target" for the request,
from the History-Info header.   So, IMHO, it's a matter of the
target-uri document specifying that the last entry tagged by whatever
name we come up with "is" the real "target" for the request.  I believe
it's up to the applications to describe how they make use of the
History-Info and not for the History-Info to describe how applications
can use it.  History-Info purely reflects what happened from a SIP
Protocol perspective and should not imply any application semantics.
Indeed the intent of section 5 of 4244 was to inform the applications
how they should decribe their usage of the header.=20
=20
Mary.=20
Note: I changed the subject to hopefully make this topic easier to keep
track of.=20

________________________________

From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
Sent: Wednesday, March 11, 2009 3:59 AM
To: Shida Schubert
Cc: sip@ietf.org List; Barnes, Mary (RICH2:AR00)
Subject: Re: [Sip] Fwd: I-D
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


One of the problems that made the discussion quite tedious is the
complete inversed meaning of the terms "retarget" and "reroute" between
4244bis and target-uri factions have.

Conclusion was that the terminology need to be properly defined and some
attempt was made to start such activity.

On 3. I have strong concerns when we start tagging the URI as to how
they come about "retarget"/"mapped conficuration" etc. It is better to
embellish them with a tag that just represents there meaning for example
"istarget" or "hop". For the following reasons:
1. It is much more intuitive for the user of this information, its
meaning can basically be guessed.
2. Such meaning also probably survives application in other problem
spaces that we had not foreseen when introducing the concept.

The solution that is  described now in the target-URI delivery draft is
not yet complete, it does not solve the freephone use case for example
and in its current form it will deliver exactly the same target-URI as
that which would be delivered by the  P-Called-Party-ID header. Contrary
to what the draft says. So some work is needed still.

/Hans Erik van Elburg



On Wed, Mar 11, 2009 at 7:13 AM, Shida Schubert <shida@ntt-at.com>
wrote:



	We have submitted the updated target-uri draft based on the
comments=20
	submitted to the list and comments received at IETF73.=20

	I have taken over as editor as Jonathan didn't have the cycles
to=20
	update the draft, with Francois, Christer and Hans Erick as
additional=20
	co-authors and great deal of help from Mary. =20
	=20
	The following summarizes the changes made to the target-uri
document
	1. Added use-case for toll-free number back
	2. Added definition of "retarget" operation.=20
	3. Removed a reference to URN
	4. Added a text discussing the difference to P-Called-Party-Id
	5. Changed parameter name from "target" to "istarget"
	=20
	Note, that the target-uri document still contains the normative
text for the
	History-Info header.
	=20
	In addition, Mary (with Francois as co-author) has submitted a
	rfc4244bis, with the following changes:
	1. Incorporated the normative aspects of the target-uri document
	into the existing normative text in RFC 4244 - the functionality
is=20
	virtually identical (as is some of the text) as the HI based
solution=20
	described in the target-uri document.  It's important that the
solution
	be integrated into RFC 4244 as it MUST work and be based on the
normative
	aspects of RFC 4244.
	2.  Added the use cases from target-uri the the summary
	in the overview of rfc4244bis.
	3. Added an additional requirement to capture the "target-uri"
information.
	4. Fixed an error in the RFC 4244 ABNF and added "retarget"
parameter.
	5. Added a more simplified example.
	=20
	=20
	We had some very long offline exchanges as to the best way
forward and
	remaining work for both documents.
	=20
	Some of the issues identified are:

	::Issues::
	   1. Should we remove the normative text from target-uri and
progress
	       4244bis along with the target-uri document to meet the
chartered
	      SIP WG milestone?=20
	=20
	=20
	   2. Name of the parameter.=20
	      At the last meeting, parameter "target" was said
inappropriate=20
	      because voicemail-uri spec already defines a parameter
called=20
	      "target" which also can be found in hi-entry, thus
potentially=20
	      causing confusion.
	    =20
	      Currently the target-uri draft uses "istarget" and 4244bis
uses
	      "retarget"  but we could never come to=20
	      a consensus on what name is appropriate.  Other
suggestions have
	      included the following:
	      "target-identity" (someone didn't like that "identity" is
also a SIP header)
	      "reg-uri"  (can be paired with "mapped-uri" for item 3
below)
	      "aor"
	      "jibberish"=20
	      etc.
	=20
	      One reason this is so difficult relates to the problem
statement in target-uri in that
	      RFC 3261 doesn't differentiate the mechanism by which the
new
	      (target) Request-URI is selected.  Another issue is that
some of the terminology in
	      RFC 3261 is overloaded - e.g., "forwarding" refers both to
a Proxy
	      which does not have responsibility for the domain of the
request-URI
	      in the incoming request, thus the proxy just "forwards"
the request to
	      the next hop AND "forwarding" is used to describe the
process whereby
	      the outgoing request is built and "forwarded" to the next
hop at which
	      point the proxy does not know how the new request-uri was
selected.
	      RFC 4244 has attempted to clarify the terms and attempts
to use "forward"=20
	      in the context of the former situation and "retarget" for
the case whereby
	      a proxy is responsible for the domain and thus can use a
number of=20
	      mechanism to select the new target for the request - e.g.,
a REGISTRAR,=20
	      configured data, etc. =20

	 3.  Related to the last point in item 2 above, it has been
proposed that
	      we differentiate the hi-entries even more by defining
separate parameters
	     for registered and configured/mapped contacts.=20
	     Currently when the R-URI is translated to a URI which is
either derived=20
	     from location service lookup(registered by UA) or from
mapping=20
	     table, there is no differentiation as to how the URI was
derived once it is
	     added to the list of potential targets.
	    =20
	     The general consensus of the authors of the two documents
was that it may=20
	     be useful for some services to have the hi-entries tagged
with the
	     more specific information.=20
	=20
	     And, of course, this gets us into another naming contest.
In the end, the naming
	     is not so important as long as the term isn't too
overloaded and it is defined
	     precisely in the document(s).
	=20
	We would appreciate WG feedback on these issues and any other
comments on
	the two documents prior to IETF-74.
	=20
	Regards,
	Shida and Mary.



	Begin forwarded message:

		From: Internet-Drafts@ietf.org
		Date: March 10, 2009 2:30:01 AM JST
		To: i-d-announce@ietf.org
		Subject: I-D
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt=20
		Reply-To: internet-drafts@ietf.org

		A New Internet-Draft is available from the on-line
Internet-Drafts=20
		directories.
	=09
	=09
		Title : Delivery of Request-URI Targets to User Agents
		Author(s) : J. Rosenberg, H. van Elburg, C. Holmberg, F.
Audet, S. Schubert
		Filename :
draft-rosenberg-sip-target-uri-delivery-01.txt
		Pages : 16
		Date : 2009-3-9
	=09
		When a Session Initiation Protocol (SIP) proxy receives
a request
		  targeted at a URI identifying a user or resource it is
responsible
		  for, the proxy translates the URI to a registered or
configured
		  contact URI of an agent representing that user or
resource.  In the
		  process, the original URI is removed from the request.
Numerous use
		  cases have arisen which require this information to be
delivered to
		  the user agent.  This document describes these use
cases and defines
		  an extension to the History-Info header field which
allows it to be
		  used to support those cases.
	=09
		A URL for this Internet-Draft is:
=09
http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri-deliv
ery-01.txt
	=09
		Internet-Drafts are also available by anonymous FTP at:
		ftp://ftp.ietf.org/internet-drafts/
	=09
		Below is the data which will enable a MIME compliant
mail reader
		implementation to automatically retrieve the ASCII
version of the
		Internet-Draft.
	=09

	Content-Type: text/plain<BR>Content-ID:
&lt;2009-3-9102756.I-D@ietf.org&gt;<BR><BR>=20

		_______________________________________________
		I-D-Announce mailing list
		I-D-Announce@ietf.org
		https://www.ietf.org/mailman/listinfo/i-d-announce
		Internet-Draft directories:
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		or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


	_______________________________________________
	Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
	This list is for NEW development of the core SIP Protocol
	Use sip-implementors@cs.columbia.edu for questions on current
sip
	Use sipping@ietf.org for new developments on the application of
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=09



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<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D171575516-11032009>And, I=20
think the big issue with the terminology is that 4244bis is application=20
agnostic, so it is tagging what happens to the URIs (why/how the =
specific URI=20
was overwritten).&nbsp; Whereas, specific applications can derive =
information,=20
such as what "is" the real "target" for the request, from =
the&nbsp;History-Info=20
header. &nbsp; So, IMHO, it's a matter of the target-uri document =
specifying=20
that the last entry tagged by whatever name we come up with "is" the =
real=20
"target" for the request.&nbsp; I believe it's up to the applications to =

describe how they make use of the History-Info and not for the =
History-Info to=20
describe how applications can use it. &nbsp;History-Info purely reflects =
what=20
happened from a SIP Protocol perspective and should&nbsp;not imply=20
any&nbsp;application semantics. Indeed the intent of section 5 of 4244 =
was to=20
inform the applications how they should decribe their usage of the =
header.=20
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D171575516-11032009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D171575516-11032009>Mary.=20
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D171575516-11032009>Note:=20
I changed the subject to hopefully make this topic easier to keep track =
of.=20
</SPAN></FONT></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Hans Erik van Elburg=20
[mailto:ietf.hanserik@gmail.com] <BR><B>Sent:</B> Wednesday, March 11, =
2009 3:59=20
AM<BR><B>To:</B> Shida Schubert<BR><B>Cc:</B> sip@ietf.org List; Barnes, =
Mary=20
(RICH2:AR00)<BR><B>Subject:</B> Re: [Sip] Fwd: I-D=20
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR></FONT><BR></DIV=
>
<DIV></DIV>One of the problems that made the discussion quite tedious is =
the=20
complete inversed meaning of the terms "retarget" and "reroute" between =
4244bis=20
and target-uri factions have.<BR><BR>Conclusion was that the terminology =
need to=20
be properly defined and some attempt was made to start such =
activity.<BR><BR>On=20
3. I have strong concerns when we start tagging the URI as to how they =
come=20
about "retarget"/"mapped conficuration" etc. It is better to embellish =
them with=20
a tag that just represents there meaning for example "istarget" or =
"hop". For=20
the following reasons:<BR>1. It is much more intuitive for the user of =
this=20
information, its meaning can basically be guessed.<BR>2. Such meaning =
also=20
probably survives application in other problem spaces that we had not =
foreseen=20
when introducing the concept.<BR><BR>The solution that is&nbsp; =
described now in=20
the target-URI delivery draft is not yet complete, it does not solve the =

freephone use case for example and in its current form it will deliver =
exactly=20
the same target-URI as that which would be delivered by the&nbsp;=20
P-Called-Party-ID header. Contrary to what the draft says. So some work =
is=20
needed still.<BR><BR>/Hans Erik van Elburg<BR><BR><BR>
<DIV class=3Dgmail_quote>On Wed, Mar 11, 2009 at 7:13 AM, Shida Schubert =
<SPAN=20
dir=3Dltr>&lt;<A =
href=3D"mailto:shida@ntt-at.com">shida@ntt-at.com</A>&gt;</SPAN>=20
wrote:<BR>
<BLOCKQUOTE class=3Dgmail_quote=20
style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">
  <DIV>
  <DIV><BR></DIV>
  <DIV>We have submitted the updated target-uri draft based on the=20
  comments&nbsp;</DIV>
  <DIV>submitted to the list and comments received at =
IETF73.&nbsp;</DIV>
  <DIV><BR></DIV>
  <DIV>I have taken over as editor as Jonathan didn't have the cycles=20
  to&nbsp;</DIV>
  <DIV>update the draft, with Francois, Christer and Hans Erick as=20
  additional&nbsp;</DIV>
  <DIV>co-authors and great deal of help from Mary. &nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>The following summarizes the changes made to the target-uri=20
document</DIV>
  <DIV>1. Added use-case for toll-free number back</DIV>
  <DIV>2. Added definition of "retarget" operation.&nbsp;</DIV>
  <DIV>3. Removed a reference to URN</DIV>
  <DIV>4. Added a text discussing the difference to =
P-Called-Party-Id</DIV>
  <DIV>5. Changed parameter name from "target" to "istarget"</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>Note, that the target-uri document still contains the normative =
text for=20
  the</DIV>
  <DIV>History-Info header.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>In addition, Mary (with Francois as co-author) has submitted =
a</DIV>
  <DIV>rfc4244bis, with the following changes:</DIV>
  <DIV>1. Incorporated the normative aspects of the target-uri =
document</DIV>
  <DIV>into the existing normative text in RFC 4244 - the functionality=20
  is&nbsp;</DIV>
  <DIV>virtually identical (as is some of the text) as the HI based=20
  solution&nbsp;</DIV>
  <DIV>described in the target-uri document. &nbsp;It's important that =
the=20
  solution</DIV>
  <DIV>be integrated into RFC 4244 as it MUST work and be based on the=20
  normative</DIV>
  <DIV>aspects of RFC 4244.</DIV>
  <DIV>2. &nbsp;Added the use cases from target-uri the the =
summary</DIV>
  <DIV>in the overview of rfc4244bis.</DIV>
  <DIV>3. Added an additional requirement to capture the "target-uri"=20
  information.</DIV>
  <DIV>4. Fixed an error in the RFC 4244 ABNF and added "retarget"=20
  parameter.</DIV>
  <DIV>5. Added a more simplified example.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>We had some very long offline exchanges as to the best way =
forward=20
  and</DIV>
  <DIV>remaining work for both documents.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>Some of the issues identified are:</DIV>
  <DIV><BR></DIV>
  <DIV>::Issues::</DIV>
  <DIV>&nbsp;&nbsp; 1. Should we remove the normative text from =
target-uri and=20
  progress</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp; 4244bis along with the target-uri =
document to=20
  meet the chartered</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;SIP WG milestone?&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; 2. Name of the parameter.&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;At the last meeting, parameter "target" =
was=20
  said inappropriate&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;because voicemail-uri spec already =
defines a=20
  parameter called&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;"target" which also can be found in =
hi-entry,=20
  thus potentially&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;causing confusion.</DIV>
  <DIV>&nbsp;&nbsp; &nbsp;&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;Currently the target-uri draft uses =
"istarget"=20
  and 4244bis uses</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;"retarget" &nbsp;but we could never =
come=20
  to&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;a consensus on what name is =
appropriate.=20
  &nbsp;Other suggestions have</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;included the following:</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;"target-identity" (someone didn't like =
that=20
  "identity" is also a SIP header)</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;"reg-uri" &nbsp;(can be paired with=20
  "mapped-uri" for item 3 below)</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;"aor"</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;"jibberish"&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;etc.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;One reason this is so difficult relates =
to the=20
  problem statement in target-uri in that</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;RFC 3261 doesn't differentiate the =
mechanism by=20
  which the new</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;(target) Request-URI is selected. =
&nbsp;Another=20
  issue is that some of the terminology in</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;RFC 3261 is overloaded - e.g., =
"forwarding"=20
  refers both to a Proxy</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;which does not have responsibility for =
the=20
  domain of the request-URI</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;in the incoming request, thus the proxy =
just=20
  "forwards" the request to</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;the next hop AND "forwarding" is used =
to=20
  describe the process whereby</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;the outgoing request is built and =
"forwarded"=20
  to the next hop at which</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;point the proxy does not know how the =
new=20
  request-uri was selected.</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;RFC 4244 has attempted to clarify the =
terms and=20
  attempts to use "forward"&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;in the context of the former situation =
and=20
  "retarget" for the case whereby</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;a proxy is responsible for the domain =
and thus=20
  can use a number of&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;mechanism to select the new target for =
the=20
  request - e.g., a REGISTRAR,&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;configured data, etc. &nbsp;</DIV>
  <DIV><BR></DIV>
  <DIV>&nbsp;3. &nbsp;Related to the last point in item 2 above, it has =
been=20
  proposed that</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;we differentiate the hi-entries even =
more by=20
  defining separate parameters</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; for registered and configured/mapped=20
  contacts.&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; Currently when the R-URI is translated to a =
URI which=20
  is either derived&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; from location service lookup(registered by =
UA) or=20
  from mapping&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; table, there is no differentiation as to how =
the URI=20
  was derived once it is</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; added to the list of potential targets.</DIV>
  <DIV>&nbsp;&nbsp; &nbsp;&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; The general consensus of the authors of the =
two=20
  documents was that it may&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; be useful for some services to have the =
hi-entries=20
  tagged with the</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; more specific information.&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; And, of course, this gets us into another =
naming=20
  contest. In the end, the naming</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; is not so important as long as the term isn't =
too=20
  overloaded and it is defined</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; precisely in the document(s).</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>We would appreciate WG feedback on these issues and any other =
comments=20
  on</DIV>
  <DIV>the two documents prior to IETF-74.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>Regards,</DIV>
  <DIV>Shida and Mary.</DIV>
  <DIV><BR></DIV>
  <DIV><BR></DIV>
  <DIV>
  <DIV>
  <DIV>
  <DIV><BR>Begin forwarded message:</DIV>
  <BLOCKQUOTE type=3D"cite">
    <DIV>
    <DIV style=3D"MARGIN: 0px"><FONT style=3D"COLOR: rgb(0,0,0)" =
face=3DHelvetica=20
    color=3D#000000 size=3D3><B>From:&nbsp;</B></FONT><FONT =
face=3DHelvetica size=3D3><A=20
    href=3D"mailto:Internet-Drafts@ietf.org"=20
    target=3D_blank>Internet-Drafts@ietf.org</A></FONT></DIV>
    <DIV style=3D"MARGIN: 0px"><FONT style=3D"COLOR: rgb(0,0,0)" =
face=3DHelvetica=20
    color=3D#000000 size=3D3><B>Date:&nbsp;</B></FONT><FONT =
face=3DHelvetica=20
    size=3D3>March 10, 2009 2:30:01 AM JST</FONT></DIV>
    <DIV style=3D"MARGIN: 0px"><FONT style=3D"COLOR: rgb(0,0,0)" =
face=3DHelvetica=20
    color=3D#000000 size=3D3><B>To:&nbsp;</B></FONT><FONT =
face=3DHelvetica size=3D3><A=20
    href=3D"mailto:i-d-announce@ietf.org"=20
    target=3D_blank>i-d-announce@ietf.org</A></FONT></DIV>
    <DIV style=3D"MARGIN: 0px"><FONT style=3D"COLOR: rgb(0,0,0)" =
face=3DHelvetica=20
    color=3D#000000 size=3D3><B>Subject:&nbsp;</B></FONT><FONT =
face=3DHelvetica=20
    size=3D3><B>I-D=20
    =
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt&nbsp;</B></FONT></D=
IV>
    <DIV style=3D"MARGIN: 0px"><FONT style=3D"COLOR: rgb(0,0,0)" =
face=3DHelvetica=20
    color=3D#000000 size=3D3><B>Reply-To:&nbsp;</B></FONT><FONT =
face=3DHelvetica=20
    size=3D3><A href=3D"mailto:internet-drafts@ietf.org"=20
    target=3D_blank>internet-drafts@ietf.org</A></FONT></DIV>
    <DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px"><BR></DIV></DIV>
    <DIV>A New Internet-Draft is available from the on-line=20
    Internet-Drafts&nbsp;<BR>directories.<BR><BR><BR><SPAN=20
    style=3D"WHITE-SPACE: pre"></SPAN>Title<SPAN style=3D"WHITE-SPACE: =
pre">=20
    </SPAN><SPAN style=3D"WHITE-SPACE: pre"></SPAN>: Delivery of =
Request-URI=20
    Targets to User Agents<BR><SPAN=20
    style=3D"WHITE-SPACE: pre"></SPAN>Author(s)<SPAN =
style=3D"WHITE-SPACE: pre">=20
    </SPAN>: J. Rosenberg, H. van Elburg, C. Holmberg, F. Audet, S.=20
    Schubert<BR><SPAN style=3D"WHITE-SPACE: pre"></SPAN>Filename<SPAN=20
    style=3D"WHITE-SPACE: pre"> </SPAN>:=20
    draft-rosenberg-sip-target-uri-delivery-01.txt<BR><SPAN=20
    style=3D"WHITE-SPACE: pre"></SPAN>Pages<SPAN style=3D"WHITE-SPACE: =
pre">=20
    </SPAN><SPAN style=3D"WHITE-SPACE: pre"></SPAN>: 16<BR><SPAN=20
    style=3D"WHITE-SPACE: pre"></SPAN>Date<SPAN style=3D"WHITE-SPACE: =
pre">=20
    </SPAN><SPAN style=3D"WHITE-SPACE: pre"></SPAN>: 2009-3-9<BR><SPAN=20
    style=3D"WHITE-SPACE: pre"></SPAN><BR>When a Session Initiation =
Protocol (SIP)=20
    proxy receives a request<BR>&nbsp;&nbsp;targeted at a URI =
identifying a user=20
    or resource it is responsible<BR>&nbsp;&nbsp;for, the proxy =
translates the=20
    URI to a registered or configured<BR>&nbsp;&nbsp;contact URI of an =
agent=20
    representing that user or resource. &nbsp;In =
the<BR>&nbsp;&nbsp;process, the=20
    original URI is removed from the request. &nbsp;Numerous=20
    use<BR>&nbsp;&nbsp;cases have arisen which require this information =
to be=20
    delivered to<BR>&nbsp;&nbsp;the user agent. &nbsp;This document =
describes=20
    these use cases and defines<BR>&nbsp;&nbsp;an extension to the =
History-Info=20
    header field which allows it to be<BR>&nbsp;&nbsp;used to support =
those=20
    cases.<BR><BR>A URL for this Internet-Draft is:<BR><A=20
    =
href=3D"http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-ur=
i-delivery-01.txt"=20
    =
target=3D_blank>http://www.ietf.org/internet-drafts/draft-rosenberg-sip-t=
arget-uri-delivery-01.txt</A><BR><BR>Internet-Drafts=20
    are also available by anonymous FTP at:<BR><A=20
    href=3D"ftp://ftp.ietf.org/internet-drafts/"=20
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KQUOTE></DIV></DIV></DIV></DIV><BR>______________________________________=
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  mailing list &nbsp;<A =
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  target=3D_blank>https://www.ietf.org/mailman/listinfo/sip</A><BR>This =
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  for NEW development of the core SIP Protocol<BR>Use <A=20
  =
href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.colu=
mbia.edu</A>=20
  for questions on current sip<BR>Use <A=20
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From mary.barnes@nortel.com  Wed Mar 11 10:03:50 2009
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Thread-Topic: Terminology (was RE: [Sip] Fwd: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Mary Barnes" <mary.barnes@nortel.com>, "Hans Erik van Elburg" <ietf.hanserik@gmail.com>, "Shida Schubert" <shida@ntt-at.com>
Cc: sip@ietf.org
Subject: Re: [Sip] Terminology (was RE: Fwd: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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BTW - in the end, I really don't care what we call the tag - the term
just needs to be precisely defined and I'd rather it not be based on a
term in RFC 3261 that is already overloaded and doesn't precisely match
the semantics for the tag in 4244bis.

________________________________

From: Barnes, Mary (RICH2:AR00)=20
Sent: Wednesday, March 11, 2009 12:03 PM
To: 'Hans Erik van Elburg'; Shida Schubert
Cc: sip@ietf.org List
Subject: Terminology (was RE: [Sip] Fwd: I-D
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


And, I think the big issue with the terminology is that 4244bis is
application agnostic, so it is tagging what happens to the URIs (why/how
the specific URI was overwritten).  Whereas, specific applications can
derive information, such as what "is" the real "target" for the request,
from the History-Info header.   So, IMHO, it's a matter of the
target-uri document specifying that the last entry tagged by whatever
name we come up with "is" the real "target" for the request.  I believe
it's up to the applications to describe how they make use of the
History-Info and not for the History-Info to describe how applications
can use it.  History-Info purely reflects what happened from a SIP
Protocol perspective and should not imply any application semantics.
Indeed the intent of section 5 of 4244 was to inform the applications
how they should decribe their usage of the header.=20
=20
Mary.=20
Note: I changed the subject to hopefully make this topic easier to keep
track of.=20

________________________________

From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
Sent: Wednesday, March 11, 2009 3:59 AM
To: Shida Schubert
Cc: sip@ietf.org List; Barnes, Mary (RICH2:AR00)
Subject: Re: [Sip] Fwd: I-D
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


One of the problems that made the discussion quite tedious is the
complete inversed meaning of the terms "retarget" and "reroute" between
4244bis and target-uri factions have.

Conclusion was that the terminology need to be properly defined and some
attempt was made to start such activity.

On 3. I have strong concerns when we start tagging the URI as to how
they come about "retarget"/"mapped conficuration" etc. It is better to
embellish them with a tag that just represents there meaning for example
"istarget" or "hop". For the following reasons:
1. It is much more intuitive for the user of this information, its
meaning can basically be guessed.
2. Such meaning also probably survives application in other problem
spaces that we had not foreseen when introducing the concept.

The solution that is  described now in the target-URI delivery draft is
not yet complete, it does not solve the freephone use case for example
and in its current form it will deliver exactly the same target-URI as
that which would be delivered by the  P-Called-Party-ID header. Contrary
to what the draft says. So some work is needed still.

/Hans Erik van Elburg



On Wed, Mar 11, 2009 at 7:13 AM, Shida Schubert <shida@ntt-at.com>
wrote:



	We have submitted the updated target-uri draft based on the
comments=20
	submitted to the list and comments received at IETF73.=20

	I have taken over as editor as Jonathan didn't have the cycles
to=20
	update the draft, with Francois, Christer and Hans Erick as
additional=20
	co-authors and great deal of help from Mary. =20
	=20
	The following summarizes the changes made to the target-uri
document
	1. Added use-case for toll-free number back
	2. Added definition of "retarget" operation.=20
	3. Removed a reference to URN
	4. Added a text discussing the difference to P-Called-Party-Id
	5. Changed parameter name from "target" to "istarget"
	=20
	Note, that the target-uri document still contains the normative
text for the
	History-Info header.
	=20
	In addition, Mary (with Francois as co-author) has submitted a
	rfc4244bis, with the following changes:
	1. Incorporated the normative aspects of the target-uri document
	into the existing normative text in RFC 4244 - the functionality
is=20
	virtually identical (as is some of the text) as the HI based
solution=20
	described in the target-uri document.  It's important that the
solution
	be integrated into RFC 4244 as it MUST work and be based on the
normative
	aspects of RFC 4244.
	2.  Added the use cases from target-uri the the summary
	in the overview of rfc4244bis.
	3. Added an additional requirement to capture the "target-uri"
information.
	4. Fixed an error in the RFC 4244 ABNF and added "retarget"
parameter.
	5. Added a more simplified example.
	=20
	=20
	We had some very long offline exchanges as to the best way
forward and
	remaining work for both documents.
	=20
	Some of the issues identified are:

	::Issues::
	   1. Should we remove the normative text from target-uri and
progress
	       4244bis along with the target-uri document to meet the
chartered
	      SIP WG milestone?=20
	=20
	=20
	   2. Name of the parameter.=20
	      At the last meeting, parameter "target" was said
inappropriate=20
	      because voicemail-uri spec already defines a parameter
called=20
	      "target" which also can be found in hi-entry, thus
potentially=20
	      causing confusion.
	    =20
	      Currently the target-uri draft uses "istarget" and 4244bis
uses
	      "retarget"  but we could never come to=20
	      a consensus on what name is appropriate.  Other
suggestions have
	      included the following:
	      "target-identity" (someone didn't like that "identity" is
also a SIP header)
	      "reg-uri"  (can be paired with "mapped-uri" for item 3
below)
	      "aor"
	      "jibberish"=20
	      etc.
	=20
	      One reason this is so difficult relates to the problem
statement in target-uri in that
	      RFC 3261 doesn't differentiate the mechanism by which the
new
	      (target) Request-URI is selected.  Another issue is that
some of the terminology in
	      RFC 3261 is overloaded - e.g., "forwarding" refers both to
a Proxy
	      which does not have responsibility for the domain of the
request-URI
	      in the incoming request, thus the proxy just "forwards"
the request to
	      the next hop AND "forwarding" is used to describe the
process whereby
	      the outgoing request is built and "forwarded" to the next
hop at which
	      point the proxy does not know how the new request-uri was
selected.
	      RFC 4244 has attempted to clarify the terms and attempts
to use "forward"=20
	      in the context of the former situation and "retarget" for
the case whereby
	      a proxy is responsible for the domain and thus can use a
number of=20
	      mechanism to select the new target for the request - e.g.,
a REGISTRAR,=20
	      configured data, etc. =20

	 3.  Related to the last point in item 2 above, it has been
proposed that
	      we differentiate the hi-entries even more by defining
separate parameters
	     for registered and configured/mapped contacts.=20
	     Currently when the R-URI is translated to a URI which is
either derived=20
	     from location service lookup(registered by UA) or from
mapping=20
	     table, there is no differentiation as to how the URI was
derived once it is
	     added to the list of potential targets.
	    =20
	     The general consensus of the authors of the two documents
was that it may=20
	     be useful for some services to have the hi-entries tagged
with the
	     more specific information.=20
	=20
	     And, of course, this gets us into another naming contest.
In the end, the naming
	     is not so important as long as the term isn't too
overloaded and it is defined
	     precisely in the document(s).
	=20
	We would appreciate WG feedback on these issues and any other
comments on
	the two documents prior to IETF-74.
	=20
	Regards,
	Shida and Mary.



	Begin forwarded message:

		From: Internet-Drafts@ietf.org
		Date: March 10, 2009 2:30:01 AM JST
		To: i-d-announce@ietf.org
		Subject: I-D
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt=20
		Reply-To: internet-drafts@ietf.org

		A New Internet-Draft is available from the on-line
Internet-Drafts=20
		directories.
	=09
	=09
		Title : Delivery of Request-URI Targets to User Agents
		Author(s) : J. Rosenberg, H. van Elburg, C. Holmberg, F.
Audet, S. Schubert
		Filename :
draft-rosenberg-sip-target-uri-delivery-01.txt
		Pages : 16
		Date : 2009-3-9
	=09
		When a Session Initiation Protocol (SIP) proxy receives
a request
		  targeted at a URI identifying a user or resource it is
responsible
		  for, the proxy translates the URI to a registered or
configured
		  contact URI of an agent representing that user or
resource.  In the
		  process, the original URI is removed from the request.
Numerous use
		  cases have arisen which require this information to be
delivered to
		  the user agent.  This document describes these use
cases and defines
		  an extension to the History-Info header field which
allows it to be
		  used to support those cases.
	=09
		A URL for this Internet-Draft is:
=09
http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri-deliv
ery-01.txt
	=09
		Internet-Drafts are also available by anonymous FTP at:
		ftp://ftp.ietf.org/internet-drafts/
	=09
		Below is the data which will enable a MIME compliant
mail reader
		implementation to automatically retrieve the ASCII
version of the
		Internet-Draft.
	=09

	Content-Type: text/plain<BR>Content-ID:
&lt;2009-3-9102756.I-D@ietf.org&gt;<BR><BR>=20

		_______________________________________________
		I-D-Announce mailing list
		I-D-Announce@ietf.org
		https://www.ietf.org/mailman/listinfo/i-d-announce
		Internet-Draft directories:
http://www.ietf.org/shadow.html
		or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


	_______________________________________________
	Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
	This list is for NEW development of the core SIP Protocol
	Use sip-implementors@cs.columbia.edu for questions on current
sip
	Use sipping@ietf.org for new developments on the application of
sip
=09



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<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D937390417-11032009>BTW&nbsp;- in the end, I really don't care =
what we call=20
the tag - the term just needs to be precisely defined and I'd rather it =
not be=20
based on a term in RFC 3261 that is already overloaded and doesn't =
precisely=20
match the semantics for the tag in 4244bis.</SPAN></FONT></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Barnes, Mary (RICH2:AR00) =
<BR><B>Sent:</B>=20
Wednesday, March 11, 2009 12:03 PM<BR><B>To:</B> 'Hans Erik van Elburg'; =
Shida=20
Schubert<BR><B>Cc:</B> sip@ietf.org List<BR><B>Subject:</B> Terminology =
(was RE:=20
[Sip] Fwd: I-D=20
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR></FONT><BR></DIV=
>
<DIV></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D171575516-11032009>And, I=20
think the big issue with the terminology is that 4244bis is application=20
agnostic, so it is tagging what happens to the URIs (why/how the =
specific URI=20
was overwritten).&nbsp; Whereas, specific applications can derive =
information,=20
such as what "is" the real "target" for the request, from =
the&nbsp;History-Info=20
header. &nbsp; So, IMHO, it's a matter of the target-uri document =
specifying=20
that the last entry tagged by whatever name we come up with "is" the =
real=20
"target" for the request.&nbsp; I believe it's up to the applications to =

describe how they make use of the History-Info and not for the =
History-Info to=20
describe how applications can use it. &nbsp;History-Info purely reflects =
what=20
happened from a SIP Protocol perspective and should&nbsp;not imply=20
any&nbsp;application semantics. Indeed the intent of section 5 of 4244 =
was to=20
inform the applications how they should decribe their usage of the =
header.=20
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D171575516-11032009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D171575516-11032009>Mary.=20
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D171575516-11032009>Note:=20
I changed the subject to hopefully make this topic easier to keep track =
of.=20
</SPAN></FONT></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Hans Erik van Elburg=20
[mailto:ietf.hanserik@gmail.com] <BR><B>Sent:</B> Wednesday, March 11, =
2009 3:59=20
AM<BR><B>To:</B> Shida Schubert<BR><B>Cc:</B> sip@ietf.org List; Barnes, =
Mary=20
(RICH2:AR00)<BR><B>Subject:</B> Re: [Sip] Fwd: I-D=20
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR></FONT><BR></DIV=
>
<DIV></DIV>One of the problems that made the discussion quite tedious is =
the=20
complete inversed meaning of the terms "retarget" and "reroute" between =
4244bis=20
and target-uri factions have.<BR><BR>Conclusion was that the terminology =
need to=20
be properly defined and some attempt was made to start such =
activity.<BR><BR>On=20
3. I have strong concerns when we start tagging the URI as to how they =
come=20
about "retarget"/"mapped conficuration" etc. It is better to embellish =
them with=20
a tag that just represents there meaning for example "istarget" or =
"hop". For=20
the following reasons:<BR>1. It is much more intuitive for the user of =
this=20
information, its meaning can basically be guessed.<BR>2. Such meaning =
also=20
probably survives application in other problem spaces that we had not =
foreseen=20
when introducing the concept.<BR><BR>The solution that is&nbsp; =
described now in=20
the target-URI delivery draft is not yet complete, it does not solve the =

freephone use case for example and in its current form it will deliver =
exactly=20
the same target-URI as that which would be delivered by the&nbsp;=20
P-Called-Party-ID header. Contrary to what the draft says. So some work =
is=20
needed still.<BR><BR>/Hans Erik van Elburg<BR><BR><BR>
<DIV class=3Dgmail_quote>On Wed, Mar 11, 2009 at 7:13 AM, Shida Schubert =
<SPAN=20
dir=3Dltr>&lt;<A =
href=3D"mailto:shida@ntt-at.com">shida@ntt-at.com</A>&gt;</SPAN>=20
wrote:<BR>
<BLOCKQUOTE class=3Dgmail_quote=20
style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">
  <DIV>
  <DIV><BR></DIV>
  <DIV>We have submitted the updated target-uri draft based on the=20
  comments&nbsp;</DIV>
  <DIV>submitted to the list and comments received at =
IETF73.&nbsp;</DIV>
  <DIV><BR></DIV>
  <DIV>I have taken over as editor as Jonathan didn't have the cycles=20
  to&nbsp;</DIV>
  <DIV>update the draft, with Francois, Christer and Hans Erick as=20
  additional&nbsp;</DIV>
  <DIV>co-authors and great deal of help from Mary. &nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>The following summarizes the changes made to the target-uri=20
document</DIV>
  <DIV>1. Added use-case for toll-free number back</DIV>
  <DIV>2. Added definition of "retarget" operation.&nbsp;</DIV>
  <DIV>3. Removed a reference to URN</DIV>
  <DIV>4. Added a text discussing the difference to =
P-Called-Party-Id</DIV>
  <DIV>5. Changed parameter name from "target" to "istarget"</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>Note, that the target-uri document still contains the normative =
text for=20
  the</DIV>
  <DIV>History-Info header.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>In addition, Mary (with Francois as co-author) has submitted =
a</DIV>
  <DIV>rfc4244bis, with the following changes:</DIV>
  <DIV>1. Incorporated the normative aspects of the target-uri =
document</DIV>
  <DIV>into the existing normative text in RFC 4244 - the functionality=20
  is&nbsp;</DIV>
  <DIV>virtually identical (as is some of the text) as the HI based=20
  solution&nbsp;</DIV>
  <DIV>described in the target-uri document. &nbsp;It's important that =
the=20
  solution</DIV>
  <DIV>be integrated into RFC 4244 as it MUST work and be based on the=20
  normative</DIV>
  <DIV>aspects of RFC 4244.</DIV>
  <DIV>2. &nbsp;Added the use cases from target-uri the the =
summary</DIV>
  <DIV>in the overview of rfc4244bis.</DIV>
  <DIV>3. Added an additional requirement to capture the "target-uri"=20
  information.</DIV>
  <DIV>4. Fixed an error in the RFC 4244 ABNF and added "retarget"=20
  parameter.</DIV>
  <DIV>5. Added a more simplified example.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>We had some very long offline exchanges as to the best way =
forward=20
  and</DIV>
  <DIV>remaining work for both documents.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>Some of the issues identified are:</DIV>
  <DIV><BR></DIV>
  <DIV>::Issues::</DIV>
  <DIV>&nbsp;&nbsp; 1. Should we remove the normative text from =
target-uri and=20
  progress</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp; 4244bis along with the target-uri =
document to=20
  meet the chartered</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;SIP WG milestone?&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; 2. Name of the parameter.&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;At the last meeting, parameter "target" =
was=20
  said inappropriate&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;because voicemail-uri spec already =
defines a=20
  parameter called&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;"target" which also can be found in =
hi-entry,=20
  thus potentially&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;causing confusion.</DIV>
  <DIV>&nbsp;&nbsp; &nbsp;&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;Currently the target-uri draft uses =
"istarget"=20
  and 4244bis uses</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;"retarget" &nbsp;but we could never =
come=20
  to&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;a consensus on what name is =
appropriate.=20
  &nbsp;Other suggestions have</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;included the following:</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;"target-identity" (someone didn't like =
that=20
  "identity" is also a SIP header)</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;"reg-uri" &nbsp;(can be paired with=20
  "mapped-uri" for item 3 below)</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;"aor"</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;"jibberish"&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;etc.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;One reason this is so difficult relates =
to the=20
  problem statement in target-uri in that</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;RFC 3261 doesn't differentiate the =
mechanism by=20
  which the new</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;(target) Request-URI is selected. =
&nbsp;Another=20
  issue is that some of the terminology in</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;RFC 3261 is overloaded - e.g., =
"forwarding"=20
  refers both to a Proxy</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;which does not have responsibility for =
the=20
  domain of the request-URI</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;in the incoming request, thus the proxy =
just=20
  "forwards" the request to</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;the next hop AND "forwarding" is used =
to=20
  describe the process whereby</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;the outgoing request is built and =
"forwarded"=20
  to the next hop at which</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;point the proxy does not know how the =
new=20
  request-uri was selected.</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;RFC 4244 has attempted to clarify the =
terms and=20
  attempts to use "forward"&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;in the context of the former situation =
and=20
  "retarget" for the case whereby</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;a proxy is responsible for the domain =
and thus=20
  can use a number of&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;mechanism to select the new target for =
the=20
  request - e.g., a REGISTRAR,&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;configured data, etc. &nbsp;</DIV>
  <DIV><BR></DIV>
  <DIV>&nbsp;3. &nbsp;Related to the last point in item 2 above, it has =
been=20
  proposed that</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; &nbsp;we differentiate the hi-entries even =
more by=20
  defining separate parameters</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; for registered and configured/mapped=20
  contacts.&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; Currently when the R-URI is translated to a =
URI which=20
  is either derived&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; from location service lookup(registered by =
UA) or=20
  from mapping&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; table, there is no differentiation as to how =
the URI=20
  was derived once it is</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; added to the list of potential targets.</DIV>
  <DIV>&nbsp;&nbsp; &nbsp;&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; The general consensus of the authors of the =
two=20
  documents was that it may&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; be useful for some services to have the =
hi-entries=20
  tagged with the</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; more specific information.&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; And, of course, this gets us into another =
naming=20
  contest. In the end, the naming</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; is not so important as long as the term isn't =
too=20
  overloaded and it is defined</DIV>
  <DIV>&nbsp;&nbsp; &nbsp; precisely in the document(s).</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>We would appreciate WG feedback on these issues and any other =
comments=20
  on</DIV>
  <DIV>the two documents prior to IETF-74.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>Regards,</DIV>
  <DIV>Shida and Mary.</DIV>
  <DIV><BR></DIV>
  <DIV><BR></DIV>
  <DIV>
  <DIV>
  <DIV>
  <DIV><BR>Begin forwarded message:</DIV>
  <BLOCKQUOTE type=3D"cite">
    <DIV>
    <DIV style=3D"MARGIN: 0px"><FONT style=3D"COLOR: rgb(0,0,0)" =
face=3DHelvetica=20
    color=3D#000000 size=3D3><B>From:&nbsp;</B></FONT><FONT =
face=3DHelvetica size=3D3><A=20
    href=3D"mailto:Internet-Drafts@ietf.org"=20
    target=3D_blank>Internet-Drafts@ietf.org</A></FONT></DIV>
    <DIV style=3D"MARGIN: 0px"><FONT style=3D"COLOR: rgb(0,0,0)" =
face=3DHelvetica=20
    color=3D#000000 size=3D3><B>Date:&nbsp;</B></FONT><FONT =
face=3DHelvetica=20
    size=3D3>March 10, 2009 2:30:01 AM JST</FONT></DIV>
    <DIV style=3D"MARGIN: 0px"><FONT style=3D"COLOR: rgb(0,0,0)" =
face=3DHelvetica=20
    color=3D#000000 size=3D3><B>To:&nbsp;</B></FONT><FONT =
face=3DHelvetica size=3D3><A=20
    href=3D"mailto:i-d-announce@ietf.org"=20
    target=3D_blank>i-d-announce@ietf.org</A></FONT></DIV>
    <DIV style=3D"MARGIN: 0px"><FONT style=3D"COLOR: rgb(0,0,0)" =
face=3DHelvetica=20
    color=3D#000000 size=3D3><B>Subject:&nbsp;</B></FONT><FONT =
face=3DHelvetica=20
    size=3D3><B>I-D=20
    =
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt&nbsp;</B></FONT></D=
IV>
    <DIV style=3D"MARGIN: 0px"><FONT style=3D"COLOR: rgb(0,0,0)" =
face=3DHelvetica=20
    color=3D#000000 size=3D3><B>Reply-To:&nbsp;</B></FONT><FONT =
face=3DHelvetica=20
    size=3D3><A href=3D"mailto:internet-drafts@ietf.org"=20
    target=3D_blank>internet-drafts@ietf.org</A></FONT></DIV>
    <DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px"><BR></DIV></DIV>
    <DIV>A New Internet-Draft is available from the on-line=20
    Internet-Drafts&nbsp;<BR>directories.<BR><BR><BR><SPAN=20
    style=3D"WHITE-SPACE: pre"></SPAN>Title<SPAN style=3D"WHITE-SPACE: =
pre">=20
    </SPAN><SPAN style=3D"WHITE-SPACE: pre"></SPAN>: Delivery of =
Request-URI=20
    Targets to User Agents<BR><SPAN=20
    style=3D"WHITE-SPACE: pre"></SPAN>Author(s)<SPAN =
style=3D"WHITE-SPACE: pre">=20
    </SPAN>: J. Rosenberg, H. van Elburg, C. Holmberg, F. Audet, S.=20
    Schubert<BR><SPAN style=3D"WHITE-SPACE: pre"></SPAN>Filename<SPAN=20
    style=3D"WHITE-SPACE: pre"> </SPAN>:=20
    draft-rosenberg-sip-target-uri-delivery-01.txt<BR><SPAN=20
    style=3D"WHITE-SPACE: pre"></SPAN>Pages<SPAN style=3D"WHITE-SPACE: =
pre">=20
    </SPAN><SPAN style=3D"WHITE-SPACE: pre"></SPAN>: 16<BR><SPAN=20
    style=3D"WHITE-SPACE: pre"></SPAN>Date<SPAN style=3D"WHITE-SPACE: =
pre">=20
    </SPAN><SPAN style=3D"WHITE-SPACE: pre"></SPAN>: 2009-3-9<BR><SPAN=20
    style=3D"WHITE-SPACE: pre"></SPAN><BR>When a Session Initiation =
Protocol (SIP)=20
    proxy receives a request<BR>&nbsp;&nbsp;targeted at a URI =
identifying a user=20
    or resource it is responsible<BR>&nbsp;&nbsp;for, the proxy =
translates the=20
    URI to a registered or configured<BR>&nbsp;&nbsp;contact URI of an =
agent=20
    representing that user or resource. &nbsp;In =
the<BR>&nbsp;&nbsp;process, the=20
    original URI is removed from the request. &nbsp;Numerous=20
    use<BR>&nbsp;&nbsp;cases have arisen which require this information =
to be=20
    delivered to<BR>&nbsp;&nbsp;the user agent. &nbsp;This document =
describes=20
    these use cases and defines<BR>&nbsp;&nbsp;an extension to the =
History-Info=20
    header field which allows it to be<BR>&nbsp;&nbsp;used to support =
those=20
    cases.<BR><BR>A URL for this Internet-Draft is:<BR><A=20
    =
href=3D"http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-ur=
i-delivery-01.txt"=20
    =
target=3D_blank>http://www.ietf.org/internet-drafts/draft-rosenberg-sip-t=
arget-uri-delivery-01.txt</A><BR><BR>Internet-Drafts=20
    are also available by anonymous FTP at:<BR><A=20
    href=3D"ftp://ftp.ietf.org/internet-drafts/"=20
    target=3D_blank>ftp://ftp.ietf.org/internet-drafts/</A><BR><BR>Below =
is the=20
    data which will enable a MIME compliant mail =
reader<BR>implementation to=20
    automatically retrieve the ASCII version of=20
  the<BR>Internet-Draft.<BR></DIV></BLOCKQUOTE></DIV></DIV>Content-Type: =

  text/plain&lt;BR&gt;Content-ID: &amp;lt;<A=20
  href=3D"mailto:2009-3-9102756.I-D@ietf.org"=20
  =
target=3D_blank>2009-3-9102756.I-D@ietf.org</A>&amp;gt;&lt;BR&gt;&lt;BR&g=
t;=20
  <DIV>
  <DIV>
  <BLOCKQUOTE type=3D"cite">
    <DIV>_______________________________________________<BR>I-D-Announce =
mailing=20
    list<BR><A href=3D"mailto:I-D-Announce@ietf.org"=20
    target=3D_blank>I-D-Announce@ietf.org</A><BR><A=20
    href=3D"https://www.ietf.org/mailman/listinfo/i-d-announce"=20
    =
target=3D_blank>https://www.ietf.org/mailman/listinfo/i-d-announce</A><BR=
>Internet-Draft=20
    directories:&nbsp;<A href=3D"http://www.ietf.org/shadow.html"=20
    target=3D_blank>http://www.ietf.org/shadow.html</A><BR>or&nbsp;<A=20
    href=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt"=20
    =
target=3D_blank>ftp://ftp.ietf.org/ietf/1shadow-sites.txt</A></DIV></BLOC=
KQUOTE></DIV></DIV></DIV></DIV><BR>______________________________________=
_________<BR>Sip=20
  mailing list &nbsp;<A =
href=3D"https://www.ietf.org/mailman/listinfo/sip"=20
  target=3D_blank>https://www.ietf.org/mailman/listinfo/sip</A><BR>This =
list is=20
  for NEW development of the core SIP Protocol<BR>Use <A=20
  =
href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.colu=
mbia.edu</A>=20
  for questions on current sip<BR>Use <A=20
  href=3D"mailto:sipping@ietf.org">sipping@ietf.org</A> for new =
developments on=20
  the application of sip<BR></BLOCKQUOTE></DIV><BR></BODY></HTML>

------_=_NextPart_001_01C9A26B.63D0589F--

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To: Mary Barnes <mary.barnes@nortel.com>, Shida Schubert <shida@ntt-at.com>, "sip@ietf.org" <sip@ietf.org>
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I don't actually have much stake in Diversion - you know it wasn't us who i=
mplemented it first, and it's a softswitch/PBX thing not SBC thing anyway -=
 I just hate having two ways of doing the same thing, or having to convert =
between the (multiple) vendors that do one or the other.  And before you th=
row stones at me for it, you can look into your own product line and find t=
hat there are products that only do Diversion and not Hist-Info.  In fact s=
everal vendors have a mixture of both across product lines, afaict.

But anyway I'm not debating the merits of History-Info vs. Diversion.  That=
 time has passed, and we will just have to live with both for a while, and =
hopefully someday Hist-Info will win.  I will be happy to convert between i=
t and Diversion for now, just so it can be un-converted on the other side o=
f the link, so that a piece of wire can claim to be standards-compliant. ;)=
=20

I'm not even arguing for a new header. =20
I'm just saying something ain't right when we don't even want to use it in =
practice.  If we can't eat our own dog food, there's something wrong with t=
he food.

-hadriel

> -----Original Message-----
> From: Mary Barnes [mailto:mary.barnes@nortel.com]
> Sent: Wednesday, March 11, 2009 12:39 PM
> To: Hadriel Kaplan; Shida Schubert; sip@ietf.org
>=20
> Yeah, we can define a new header and just ignore the use of a basic SIP
> building block that the WG spent 2.5 years  (note this doesn't include th=
e
> 2+ years spent on the doc as an individual and as a SIPPING WG document ;=
)
> And, while we're at it, let's just go ahead and obsolete 4244 altogether
> and standardize diversion (something the WG rejected almost a decade ago)=
.
> That will make one vendor and a handful of SPs very happy and those of us
> that have implemented the specs can ripout some code after while when we
> have some spare change to pay folks to do so (and of course, this is afte=
r
> the folks that have not implemented diversion actually implement it -
> there are actually some of those out there).
>=20
> As an aside, this is another fine example as to why we don't seem to make
> great progress or produce specs that are deemed useful - we agree on a wa=
y
> forward and then change our minds later (sometimes for good reasons and
> sometimes because a shortcut seems easier or it's just arbitrary based on
> the phase of the moon it seems).
>=20
> Mary.

From mary.barnes@nortel.com  Wed Mar 11 10:17:24 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>, "Shida Schubert" <shida@ntt-at.com>, <sip@ietf.org>
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Well, we do advise the products to implement History-Info and the one
that I was responsible for in the past does implement all three
(including the remote-party-id solution). =20

On your last point, it's not clear to me the reason why folks haven't
implemented it, other than that it's just in the pile of stuff that they
haven't implemented. And, it's not all the members of SIP Forum that
haven't implemented it (perhaps the subset that are currently
contributing to the SIP Connect spec).  There is a handful of vendors
that interop with one of our products (that I know has implemented HI)
that do implement it, as well. So, I don't think it's fair to say across
the industry it has not been implemented. =20

Now, I can see folks not wanting to implement it from the perspective
that there are already a number of core interop issues per your interop
document, thus there's really no point in adding something like HI when
the basics are broken. But, that's a broader problem and kinda takes us
to Dean's tricycle analogy on the RAI list.=20

Mary.=20

-----Original Message-----
From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]=20
Sent: Wednesday, March 11, 2009 12:05 PM
To: Barnes, Mary (RICH2:AR00); Shida Schubert; sip@ietf.org
Subject: RE: [Sip] Fwd: I-D
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


I don't actually have much stake in Diversion - you know it wasn't us
who implemented it first, and it's a softswitch/PBX thing not SBC thing
anyway - I just hate having two ways of doing the same thing, or having
to convert between the (multiple) vendors that do one or the other.  And
before you throw stones at me for it, you can look into your own product
line and find that there are products that only do Diversion and not
Hist-Info.  In fact several vendors have a mixture of both across
product lines, afaict.

But anyway I'm not debating the merits of History-Info vs. Diversion.
That time has passed, and we will just have to live with both for a
while, and hopefully someday Hist-Info will win.  I will be happy to
convert between it and Diversion for now, just so it can be un-converted
on the other side of the link, so that a piece of wire can claim to be
standards-compliant. ;)=20

I'm not even arguing for a new header. =20
I'm just saying something ain't right when we don't even want to use it
in practice.  If we can't eat our own dog food, there's something wrong
with the food.

-hadriel

> -----Original Message-----
> From: Mary Barnes [mailto:mary.barnes@nortel.com]
> Sent: Wednesday, March 11, 2009 12:39 PM
> To: Hadriel Kaplan; Shida Schubert; sip@ietf.org
>=20
> Yeah, we can define a new header and just ignore the use of a basic=20
> SIP building block that the WG spent 2.5 years  (note this doesn't=20
> include the
> 2+ years spent on the doc as an individual and as a SIPPING WG=20
> 2+ document ;)
> And, while we're at it, let's just go ahead and obsolete 4244=20
> altogether and standardize diversion (something the WG rejected almost
a decade ago).
> That will make one vendor and a handful of SPs very happy and those of

> us that have implemented the specs can ripout some code after while=20
> when we have some spare change to pay folks to do so (and of course,=20
> this is after the folks that have not implemented diversion actually=20
> implement it - there are actually some of those out there).
>=20
> As an aside, this is another fine example as to why we don't seem to=20
> make great progress or produce specs that are deemed useful - we agree

> on a way forward and then change our minds later (sometimes for good=20
> reasons and sometimes because a shortcut seems easier or it's just=20
> arbitrary based on the phase of the moon it seems).
>=20
> Mary.

From HKaplan@acmepacket.com  Wed Mar 11 10:56:08 2009
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To: Mary Barnes <mary.barnes@nortel.com>, Shida Schubert <shida@ntt-at.com>, "sip@ietf.org" <sip@ietf.org>
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I never said across the industry it has not been implemented - I know it ha=
s by some because we have to convert it for some.  I think the reason it ha=
sn't gained dominance is because Diversion was done first by many, and if i=
t ain't broke they have no reason to fix it compared with other priorities;=
 or it just takes them years to do such things and get it deployed. =20

-hadriel

> -----Original Message-----
> From: Mary Barnes [mailto:mary.barnes@nortel.com]
> Sent: Wednesday, March 11, 2009 1:20 PM
>=20
> Well, we do advise the products to implement History-Info and the one
> that I was responsible for in the past does implement all three
> (including the remote-party-id solution).
>=20
> On your last point, it's not clear to me the reason why folks haven't
> implemented it, other than that it's just in the pile of stuff that they
> haven't implemented. And, it's not all the members of SIP Forum that
> haven't implemented it (perhaps the subset that are currently
> contributing to the SIP Connect spec).  There is a handful of vendors
> that interop with one of our products (that I know has implemented HI)
> that do implement it, as well. So, I don't think it's fair to say across
> the industry it has not been implemented.
>=20
> Now, I can see folks not wanting to implement it from the perspective
> that there are already a number of core interop issues per your interop
> document, thus there's really no point in adding something like HI when
> the basics are broken. But, that's a broader problem and kinda takes us
> to Dean's tricycle analogy on the RAI list.
>=20
> Mary.

From theo@crazygreek.co.uk  Wed Mar 11 11:19:28 2009
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From: Theo Zourzouvillys <theo@crazygreek.co.uk>
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On Wed, Mar 11, 2009 at 5:04 PM, Hadriel Kaplan <HKaplan@acmepacket.com> wrote:

> If we can't eat our own dog food, there's something wrong with the food.

or you've not yet worked out how to open the tin.

i am strongly in support of this mechanism.

 ~ Theo

From theo@crazygreek.co.uk  Wed Mar 11 13:07:21 2009
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On Wed, Mar 11, 2009 at 6:20 PM, Theo Zourzouvillys
<theo@crazygreek.co.uk> wrote:

> or you've not yet worked out how to open the tin.
>
> i am strongly in support of this mechanism.

sorry, too may lists, threads, and offlist discussions - and not enough sleep.

i'm not trying to suggest that we should be working on mechanisms for
opening tins (although will happily entertain any new mechanisms -
there is currently only one practical way i am aware of opening tins,
which means it doesn't meet SIPs minimum quota for "ways to do
thing"); so let me reword my previous message so it makes sense to
anyone other then me:

  i am strongly in favour of putting the target URI in HI.

:)

 ~ Theo

From HKaplan@acmepacket.com  Wed Mar 11 13:50:06 2009
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> -----Original Message-----
> From: Theo Zourzouvillys [mailto:theo@crazygreek.co.uk]
> Sent: Wednesday, March 11, 2009 2:20 PM
>=20
> On Wed, Mar 11, 2009 at 5:04 PM, Hadriel Kaplan <HKaplan@acmepacket.com>
> wrote:
>=20
> > If we can't eat our own dog food, there's something wrong with the food=
.
>=20
> or you've not yet worked out how to open the tin.

I think most everyone in the SIP Forum knows how the mechanism could work u=
sing HI.  The question was if we wanted to solve the issue using HI for the=
 SIP-Connect profile.  I think the rationale for "no" was the feeling we ha=
d on whether PBX's would actually support HI anytime soon, or whether in pr=
actice we would actually be doing something different in deployment to actu=
ally make calls work.  (though I can't really speak for others - I could be=
 totally wrong on that, this was just the sense I got)

Personally, I look at it from a motivation perspective: who has the motivat=
ion to change to make something work?  In the SIP-Connect case, the Service=
 Provider is the one selling the SIP-Trunk service, so to win the deal and =
satisfy customers they've got more of the motivation/burden to make it work=
.  The SP's also have a large motivation to support as many IP-PBX vendors =
as possible.  Any changes we mandate of IP-PBX vendors will reduce the pool=
 of them that are "SIP-Connect compliant", which is not good.  Especially w=
hen what we're proposing basically works *now* in SIP-Trunk deployments - i=
.e., there's running code.  It doesn't quite work that way today within the=
 SP's domain on the wire, but as I said they have the motivation (and the a=
bility) to change their side.

-hadriel=20


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Date: Wed, 11 Mar 2009 22:40:04 +0100
From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
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Subject: Re: [Sip] Terminology (was RE: Fwd: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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The target-uri concept when defined properly is entirely application 
agnostic.

When one uses History-Info or another header as a vehicle for this 
information then it still can be considered application agnostic as any 
application can use this information as it likes.

 From SIP perspective all the Request-URI rewrites look the same, that 
is what brought us discussing this in the first place.

/Hans Erik

Mary Barnes wrote:
> And, I think the big issue with the terminology is that 4244bis is 
> application agnostic, so it is tagging what happens to the URIs 
> (why/how the specific URI was overwritten).  Whereas, specific 
> applications can derive information, such as what "is" the real 
> "target" for the request, from the History-Info header.   So, IMHO, 
> it's a matter of the target-uri document specifying that the last 
> entry tagged by whatever name we come up with "is" the real "target" 
> for the request.  I believe it's up to the applications to describe 
> how they make use of the History-Info and not for the History-Info to 
> describe how applications can use it.  History-Info purely reflects 
> what happened from a SIP Protocol perspective and should not imply 
> any application semantics. Indeed the intent of section 5 of 4244 was 
> to inform the applications how they should decribe their usage of the 
> header.
>  
> Mary.
> Note: I changed the subject to hopefully make this topic easier to 
> keep track of.
>
> ------------------------------------------------------------------------
> *From:* Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
> *Sent:* Wednesday, March 11, 2009 3:59 AM
> *To:* Shida Schubert
> *Cc:* sip@ietf.org List; Barnes, Mary (RICH2:AR00)
> *Subject:* Re: [Sip] Fwd: I-D 
> ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>
> One of the problems that made the discussion quite tedious is the 
> complete inversed meaning of the terms "retarget" and "reroute" 
> between 4244bis and target-uri factions have.
>
> Conclusion was that the terminology need to be properly defined and 
> some attempt was made to start such activity.
>
> On 3. I have strong concerns when we start tagging the URI as to how 
> they come about "retarget"/"mapped conficuration" etc. It is better to 
> embellish them with a tag that just represents there meaning for 
> example "istarget" or "hop". For the following reasons:
> 1. It is much more intuitive for the user of this information, its 
> meaning can basically be guessed.
> 2. Such meaning also probably survives application in other problem 
> spaces that we had not foreseen when introducing the concept.
>
> The solution that is  described now in the target-URI delivery draft 
> is not yet complete, it does not solve the freephone use case for 
> example and in its current form it will deliver exactly the same 
> target-URI as that which would be delivered by the  P-Called-Party-ID 
> header. Contrary to what the draft says. So some work is needed still.
>
> /Hans Erik van Elburg
>
>
> On Wed, Mar 11, 2009 at 7:13 AM, Shida Schubert <shida@ntt-at.com 
> <mailto:shida@ntt-at.com>> wrote:
>
>
>     We have submitted the updated target-uri draft based on the comments 
>     submitted to the list and comments received at IETF73. 
>
>     I have taken over as editor as Jonathan didn't have the cycles to 
>     update the draft, with Francois, Christer and Hans Erick as
>     additional 
>     co-authors and great deal of help from Mary.  
>      
>     The following summarizes the changes made to the target-uri document
>     1. Added use-case for toll-free number back
>     2. Added definition of "retarget" operation. 
>     3. Removed a reference to URN
>     4. Added a text discussing the difference to P-Called-Party-Id
>     5. Changed parameter name from "target" to "istarget"
>      
>     Note, that the target-uri document still contains the normative
>     text for the
>     History-Info header.
>      
>     In addition, Mary (with Francois as co-author) has submitted a
>     rfc4244bis, with the following changes:
>     1. Incorporated the normative aspects of the target-uri document
>     into the existing normative text in RFC 4244 - the functionality is 
>     virtually identical (as is some of the text) as the HI based solution 
>     described in the target-uri document.  It's important that the
>     solution
>     be integrated into RFC 4244 as it MUST work and be based on the
>     normative
>     aspects of RFC 4244.
>     2.  Added the use cases from target-uri the the summary
>     in the overview of rfc4244bis.
>     3. Added an additional requirement to capture the "target-uri"
>     information.
>     4. Fixed an error in the RFC 4244 ABNF and added "retarget" parameter.
>     5. Added a more simplified example.
>      
>      
>     We had some very long offline exchanges as to the best way forward and
>     remaining work for both documents.
>      
>     Some of the issues identified are:
>
>     ::Issues::
>        1. Should we remove the normative text from target-uri and progress
>            4244bis along with the target-uri document to meet the
>     chartered
>           SIP WG milestone? 
>      
>      
>        2. Name of the parameter. 
>           At the last meeting, parameter "target" was said inappropriate 
>           because voicemail-uri spec already defines a parameter called 
>           "target" which also can be found in hi-entry, thus potentially 
>           causing confusion.
>          
>           Currently the target-uri draft uses "istarget" and 4244bis uses
>           "retarget"  but we could never come to 
>           a consensus on what name is appropriate.  Other suggestions have
>           included the following:
>           "target-identity" (someone didn't like that "identity" is
>     also a SIP header)
>           "reg-uri"  (can be paired with "mapped-uri" for item 3 below)
>           "aor"
>           "jibberish" 
>           etc.
>      
>           One reason this is so difficult relates to the problem
>     statement in target-uri in that
>           RFC 3261 doesn't differentiate the mechanism by which the new
>           (target) Request-URI is selected.  Another issue is that
>     some of the terminology in
>           RFC 3261 is overloaded - e.g., "forwarding" refers both to a
>     Proxy
>           which does not have responsibility for the domain of the
>     request-URI
>           in the incoming request, thus the proxy just "forwards" the
>     request to
>           the next hop AND "forwarding" is used to describe the
>     process whereby
>           the outgoing request is built and "forwarded" to the next
>     hop at which
>           point the proxy does not know how the new request-uri was
>     selected.
>           RFC 4244 has attempted to clarify the terms and attempts to
>     use "forward" 
>           in the context of the former situation and "retarget" for
>     the case whereby
>           a proxy is responsible for the domain and thus can use a
>     number of 
>           mechanism to select the new target for the request - e.g., a
>     REGISTRAR, 
>           configured data, etc.  
>
>      3.  Related to the last point in item 2 above, it has been
>     proposed that
>           we differentiate the hi-entries even more by defining
>     separate parameters
>          for registered and configured/mapped contacts. 
>          Currently when the R-URI is translated to a URI which is
>     either derived 
>          from location service lookup(registered by UA) or from mapping 
>          table, there is no differentiation as to how the URI was
>     derived once it is
>          added to the list of potential targets.
>          
>          The general consensus of the authors of the two documents was
>     that it may 
>          be useful for some services to have the hi-entries tagged
>     with the
>          more specific information. 
>      
>          And, of course, this gets us into another naming contest. In
>     the end, the naming
>          is not so important as long as the term isn't too overloaded
>     and it is defined
>          precisely in the document(s).
>      
>     We would appreciate WG feedback on these issues and any other
>     comments on
>     the two documents prior to IETF-74.
>      
>     Regards,
>     Shida and Mary.
>
>
>
>     Begin forwarded message:
>>     *From: *Internet-Drafts@ietf.org <mailto:Internet-Drafts@ietf.org>
>>     *Date: *March 10, 2009 2:30:01 AM JST
>>     *To: *i-d-announce@ietf.org <mailto:i-d-announce@ietf.org>
>>     *Subject: **I-D
>>     ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt *
>>     *Reply-To: *internet-drafts@ietf.org
>>     <mailto:internet-drafts@ietf.org>
>>
>>     A New Internet-Draft is available from the on-line Internet-Drafts 
>>     directories.
>>
>>
>>     Title
>>     : Delivery of Request-URI Targets to User Agents
>>     Author(s)
>>     : J. Rosenberg, H. van Elburg, C. Holmberg, F. Audet, S. Schubert
>>     Filename : draft-rosenberg-sip-target-uri-delivery-01.txt
>>     Pages
>>     : 16
>>     Date
>>     : 2009-3-9
>>
>>     When a Session Initiation Protocol (SIP) proxy receives a request
>>       targeted at a URI identifying a user or resource it is responsible
>>       for, the proxy translates the URI to a registered or configured
>>       contact URI of an agent representing that user or resource.  In the
>>       process, the original URI is removed from the request.
>>      Numerous use
>>       cases have arisen which require this information to be delivered to
>>       the user agent.  This document describes these use cases and
>>     defines
>>       an extension to the History-Info header field which allows it to be
>>       used to support those cases.
>>
>>     A URL for this Internet-Draft is:
>>     http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri-delivery-01.txt
>>
>>     Internet-Drafts are also available by anonymous FTP at:
>>     ftp://ftp.ietf.org/internet-drafts/
>>
>>     Below is the data which will enable a MIME compliant mail reader
>>     implementation to automatically retrieve the ASCII version of the
>>     Internet-Draft.
>     Content-Type: text/plain<BR>Content-ID:
>     &lt;2009-3-9102756.I-D@ietf.org
>     <mailto:2009-3-9102756.I-D@ietf.org>&gt;<BR><BR>
>>     _______________________________________________
>>     I-D-Announce mailing list
>>     I-D-Announce@ietf.org <mailto:I-D-Announce@ietf.org>
>>     https://www.ietf.org/mailman/listinfo/i-d-announce
>>     Internet-Draft directories: http://www.ietf.org/shadow.html
>>     or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>
>     _______________________________________________
>     Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>     This list is for NEW development of the core SIP Protocol
>     Use sip-implementors@cs.columbia.edu
>     <mailto:sip-implementors@cs.columbia.edu> for questions on current sip
>     Use sipping@ietf.org <mailto:sipping@ietf.org> for new
>     developments on the application of sip
>
>

From spencer@wonderhamster.org  Wed Mar 11 14:49:58 2009
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Subject: Re: [Sip] Fwd: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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> I think most everyone in the SIP Forum knows how the mechanism could work 
> using HI.  The question was if we wanted to solve the issue using HI for 
> the SIP-Connect profile.  I think the rationale for "no" was the feeling 
> we had on whether PBX's would actually support HI anytime soon, or whether 
> in practice we would actually be doing something different in deployment 
> to actually make calls work.  (though I can't really speak for others - I 
> could be totally wrong on that, this was just the sense I got)
>

Speaking only for myself, I thought that the obstacle for SIPconnect was 
that there was already a history-info-bis effort underway - we just weren't 
excited about requiring support for a mechanism that was being revised. It 
wasn't clear to me that we would be equally unexcited if 4244bis was 
approved for publication.

But as Hadriel says, I could also be totally wrong about that. And other 
people don't have to be shy about explaining things to me.

Thanks,

Spencer

> Personally, I look at it from a motivation perspective: who has the 
> motivation to change to make something work?  In the SIP-Connect case, the 
> Service Provider is the one selling the SIP-Trunk service, so to win the 
> deal and satisfy customers they've got more of the motivation/burden to 
> make it work.  The SP's also have a large motivation to support as many 
> IP-PBX vendors as possible.  Any changes we mandate of IP-PBX vendors will 
> reduce the pool of them that are "SIP-Connect compliant", which is not 
> good.  Especially when what we're proposing basically works *now* in 
> SIP-Trunk deployments - i.e., there's running code.  It doesn't quite work 
> that way today within the SP's domain on the wire, but as I said they have 
> the motivation (and the ability) to change their side.
>
> -hadriel
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 



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Date: Wed, 11 Mar 2009 23:04:46 +0100
From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
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Cc: sip@ietf.org, Ian Elz <ian.elz@ericsson.com>, Francois Audet <audet@nortel.com>
Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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I believe in this case the History-Info application-agnosticness 
principle applies.

Why should History-Info enforce something, that goes against the whishes 
of a domain that is adding a hi-entry and should be considered best 
equiped in judging what privacy rules apply for this specific entry.

/Hans Erik

Mary Barnes wrote:
> Ian,
>
> This is an interesting question. I need to think about it some more, but
> I don't like the "none" idea as it really must be up to the entity that
> added the Privacy header to the request, as to whether it wants the HI
> entries that it adds to go outside a domain. My initial thought is that
> we can't overrule the overall Privacy header. The thing is that the
> Privacy header doesn't preclude gathering the information and using
> within a domain AND if it were to not include an hi-entry when the
> request leaves the domain for which the proxy is responsible, the
> recipient can add the hi-entry for the current request-uri before it
> adds the new hi-entry.  
>
> Your PSTN example doesn't strictly map obviously to a SIP model and I
> would think you might consider the PSTN hop to be within the same domain
> for which the proxy (or gw, I guess) is responsible OR consider this a
> walled garden whereby the Privacy header doesn't apply. Which brings me
> to a more general question as to how you all deal with other headers
> when you do your mapping to the PSTN when there is a Privacy header?  It
> would seem this problem isn't unique to History-Info, although I know
> little about the details of PSTN I/W.
>
> Regards,
> Mary. 
>
> -----Original Message-----
> From: Ian Elz [mailto:ian.elz@ericsson.com] 
> Sent: Wednesday, March 11, 2009 4:54 AM
> To: Barnes, Mary (RICH2:AR00); Audet, Francois (SC100:3055)
> Cc: sip@ietf.org
> Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt 
>
> Mary, Francois,
>
> Section 4.1 of the draft allows for the addition of the Privacy header
> parameter with a value of "history" to be included with the
> hi-targeted-to-uri.
>
> Can this be extended to also allow the value "none".
>
> A node adding a H-I entry is allowed to specify the privacy value
> "history" either in the Privacy header or as a header parameter
> associated with the hi-targeted-to-uri. If the former action is taken by
> one node this results in privacy for all history info header entries
> even if this is not required. There is no way for privacy of an
> individual H-I entry to be set to none if no Privacy is required for the
> uri.
>
> This creates a specific problem when interworking with the PSTN where
> the privacy is associated with each E.164 number included in the
> protocol. If an INVITE is being mapped to ISUP and the H-I entries are
> being used to map to redirection numbers then a single Privacy header
> with the value "history" will result in all numbers being restricted.
>
> If the Privacy header parameter could include the value "none" then this
> would explicitly indicate that the associated uri was allowed to be
> presented.
>
> The ability to explicitly allow presentation of specific H-I entries may
> also be useful in pure sip implementations.
>
> I don't believe that this change would create any backward compatibility
> issues. The existing implementations will continue as deployed but new
> implementations could explicitly set no privacy for a specific uri.
>
> There are no security issues other than those already defined in the
> draft that an intervening node could modify an existing H-I entry. The
> Privacy header value of "none" should only be included by the node in
> the same manner as the existing inclusion of the "history" value.
>
> Ian Elz
>
>
> -----Original Message-----
> From: i-d-announce-bounces@ietf.org
> [mailto:i-d-announce-bounces@ietf.org] On Behalf Of
> Internet-Drafts@ietf.org
> Sent: 04 March 2009 15:15
> To: i-d-announce@ietf.org
> Subject: I-D Action:draft-barnes-sip-rfc4244bis-00.txt 
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>
> 	Title           : An Extension to the Session Initiation
> Protocol (SIP) for Request History Information
> 	Author(s)       : M. Barnes, F. Audet
> 	Filename        : draft-barnes-sip-rfc4244bis-00.txt
> 	Pages           : 49
> 	Date            : 2009-03-04
>
> This document defines a standard mechanism for capturing the history
> information associated with a Session Initiation Protocol (SIP) request.
> This capability enables many enhanced services by providing the
> information as to how and why a call arrives at a specific application
> or user.  This document defines a new optional SIP header, History-Info,
> for capturing the history information in requests.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-barnes-sip-rfc4244bis-00.txt
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> Below is the data which will enable a MIME compliant mail reader
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> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>   

From mary.barnes@nortel.com  Wed Mar 11 15:27:08 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Hans Erik van Elburg" <ietf.hanserik@gmail.com>
Cc: sip@ietf.org
Subject: Re: [Sip] Terminology (was RE: Fwd: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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So, by "target-uri concept" are you referring to the solution in the
current target-uri draft or the application usage as such in the
target-uri document (which is what I was referring to)?  =20

Yes, the problem is that SIP doesn't differentiate the various
mechanisms by which a request-URI may be changed, which is what some of
the updated text in 4244bis is trying to do and why we need precise
terms. I think we all agree that.=20

I honestly don't care what we call the tag, as long as we're clear about
the functionality.=20

Mary.=20

-----Original Message-----
From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
Sent: Wednesday, March 11, 2009 4:40 PM
To: Barnes, Mary (RICH2:AR00)
Cc: Shida Schubert; sip@ietf.org
Subject: Re: Terminology (was RE: [Sip] Fwd: I-D
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt

The target-uri concept when defined properly is entirely application
agnostic.

When one uses History-Info or another header as a vehicle for this
information then it still can be considered application agnostic as any
application can use this information as it likes.

 From SIP perspective all the Request-URI rewrites look the same, that
is what brought us discussing this in the first place.

/Hans Erik

Mary Barnes wrote:
> And, I think the big issue with the terminology is that 4244bis is=20
> application agnostic, so it is tagging what happens to the URIs=20
> (why/how the specific URI was overwritten).  Whereas, specific=20
> applications can derive information, such as what "is" the real
> "target" for the request, from the History-Info header.   So, IMHO,=20
> it's a matter of the target-uri document specifying that the last=20
> entry tagged by whatever name we come up with "is" the real "target"
> for the request.  I believe it's up to the applications to describe=20
> how they make use of the History-Info and not for the History-Info to=20
> describe how applications can use it.  History-Info purely reflects=20
> what happened from a SIP Protocol perspective and should not imply any

> application semantics. Indeed the intent of section 5 of 4244 was to=20
> inform the applications how they should decribe their usage of the=20
> header.
> =20
> Mary.
> Note: I changed the subject to hopefully make this topic easier to=20
> keep track of.
>
> ----------------------------------------------------------------------
> --
> *From:* Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
> *Sent:* Wednesday, March 11, 2009 3:59 AM
> *To:* Shida Schubert
> *Cc:* sip@ietf.org List; Barnes, Mary (RICH2:AR00)
> *Subject:* Re: [Sip] Fwd: I-D
> ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>
> One of the problems that made the discussion quite tedious is the=20
> complete inversed meaning of the terms "retarget" and "reroute"
> between 4244bis and target-uri factions have.
>
> Conclusion was that the terminology need to be properly defined and=20
> some attempt was made to start such activity.
>
> On 3. I have strong concerns when we start tagging the URI as to how=20
> they come about "retarget"/"mapped conficuration" etc. It is better to

> embellish them with a tag that just represents there meaning for=20
> example "istarget" or "hop". For the following reasons:
> 1. It is much more intuitive for the user of this information, its=20
> meaning can basically be guessed.
> 2. Such meaning also probably survives application in other problem=20
> spaces that we had not foreseen when introducing the concept.
>
> The solution that is  described now in the target-URI delivery draft=20
> is not yet complete, it does not solve the freephone use case for=20
> example and in its current form it will deliver exactly the same=20
> target-URI as that which would be delivered by the  P-Called-Party-ID=20
> header. Contrary to what the draft says. So some work is needed still.
>
> /Hans Erik van Elburg
>
>
> On Wed, Mar 11, 2009 at 7:13 AM, Shida Schubert <shida@ntt-at.com=20
> <mailto:shida@ntt-at.com>> wrote:
>
>
>     We have submitted the updated target-uri draft based on the
comments=20
>     submitted to the list and comments received at IETF73.=20
>
>     I have taken over as editor as Jonathan didn't have the cycles to=20
>     update the draft, with Francois, Christer and Hans Erick as
>     additional=20
>     co-authors and great deal of help from Mary. =20
>     =20
>     The following summarizes the changes made to the target-uri
document
>     1. Added use-case for toll-free number back
>     2. Added definition of "retarget" operation.=20
>     3. Removed a reference to URN
>     4. Added a text discussing the difference to P-Called-Party-Id
>     5. Changed parameter name from "target" to "istarget"
>     =20
>     Note, that the target-uri document still contains the normative
>     text for the
>     History-Info header.
>     =20
>     In addition, Mary (with Francois as co-author) has submitted a
>     rfc4244bis, with the following changes:
>     1. Incorporated the normative aspects of the target-uri document
>     into the existing normative text in RFC 4244 - the functionality
is=20
>     virtually identical (as is some of the text) as the HI based
solution=20
>     described in the target-uri document.  It's important that the
>     solution
>     be integrated into RFC 4244 as it MUST work and be based on the
>     normative
>     aspects of RFC 4244.
>     2.  Added the use cases from target-uri the the summary
>     in the overview of rfc4244bis.
>     3. Added an additional requirement to capture the "target-uri"
>     information.
>     4. Fixed an error in the RFC 4244 ABNF and added "retarget"
parameter.
>     5. Added a more simplified example.
>     =20
>     =20
>     We had some very long offline exchanges as to the best way forward
and
>     remaining work for both documents.
>     =20
>     Some of the issues identified are:
>
>     ::Issues::
>        1. Should we remove the normative text from target-uri and
progress
>            4244bis along with the target-uri document to meet the
>     chartered
>           SIP WG milestone?=20
>     =20
>     =20
>        2. Name of the parameter.=20
>           At the last meeting, parameter "target" was said
inappropriate=20
>           because voicemail-uri spec already defines a parameter
called=20
>           "target" which also can be found in hi-entry, thus
potentially=20
>           causing confusion.
>         =20
>           Currently the target-uri draft uses "istarget" and 4244bis
uses
>           "retarget"  but we could never come to=20
>           a consensus on what name is appropriate.  Other suggestions
have
>           included the following:
>           "target-identity" (someone didn't like that "identity" is
>     also a SIP header)
>           "reg-uri"  (can be paired with "mapped-uri" for item 3
below)
>           "aor"
>           "jibberish"=20
>           etc.
>     =20
>           One reason this is so difficult relates to the problem
>     statement in target-uri in that
>           RFC 3261 doesn't differentiate the mechanism by which the
new
>           (target) Request-URI is selected.  Another issue is that
>     some of the terminology in
>           RFC 3261 is overloaded - e.g., "forwarding" refers both to a
>     Proxy
>           which does not have responsibility for the domain of the
>     request-URI
>           in the incoming request, thus the proxy just "forwards" the
>     request to
>           the next hop AND "forwarding" is used to describe the
>     process whereby
>           the outgoing request is built and "forwarded" to the next
>     hop at which
>           point the proxy does not know how the new request-uri was
>     selected.
>           RFC 4244 has attempted to clarify the terms and attempts to
>     use "forward"=20
>           in the context of the former situation and "retarget" for
>     the case whereby
>           a proxy is responsible for the domain and thus can use a
>     number of=20
>           mechanism to select the new target for the request - e.g., a
>     REGISTRAR,=20
>           configured data, etc. =20
>
>      3.  Related to the last point in item 2 above, it has been
>     proposed that
>           we differentiate the hi-entries even more by defining
>     separate parameters
>          for registered and configured/mapped contacts.=20
>          Currently when the R-URI is translated to a URI which is
>     either derived=20
>          from location service lookup(registered by UA) or from
mapping=20
>          table, there is no differentiation as to how the URI was
>     derived once it is
>          added to the list of potential targets.
>         =20
>          The general consensus of the authors of the two documents was
>     that it may=20
>          be useful for some services to have the hi-entries tagged
>     with the
>          more specific information.=20
>     =20
>          And, of course, this gets us into another naming contest. In
>     the end, the naming
>          is not so important as long as the term isn't too overloaded
>     and it is defined
>          precisely in the document(s).
>     =20
>     We would appreciate WG feedback on these issues and any other
>     comments on
>     the two documents prior to IETF-74.
>     =20
>     Regards,
>     Shida and Mary.
>
>
>
>     Begin forwarded message:
>>     *From: *Internet-Drafts@ietf.org
<mailto:Internet-Drafts@ietf.org>
>>     *Date: *March 10, 2009 2:30:01 AM JST
>>     *To: *i-d-announce@ietf.org <mailto:i-d-announce@ietf.org>
>>     *Subject: **I-D
>>     ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt *
>>     *Reply-To: *internet-drafts@ietf.org
>>     <mailto:internet-drafts@ietf.org>
>>
>>     A New Internet-Draft is available from the on-line
Internet-Drafts=20
>>     directories.
>>
>>
>>     Title
>>     : Delivery of Request-URI Targets to User Agents
>>     Author(s)
>>     : J. Rosenberg, H. van Elburg, C. Holmberg, F. Audet, S. Schubert
>>     Filename : draft-rosenberg-sip-target-uri-delivery-01.txt
>>     Pages
>>     : 16
>>     Date
>>     : 2009-3-9
>>
>>     When a Session Initiation Protocol (SIP) proxy receives a request
>>       targeted at a URI identifying a user or resource it is
responsible
>>       for, the proxy translates the URI to a registered or configured
>>       contact URI of an agent representing that user or resource.  In
the
>>       process, the original URI is removed from the request.
>>      Numerous use
>>       cases have arisen which require this information to be
delivered to
>>       the user agent.  This document describes these use cases and
>>     defines
>>       an extension to the History-Info header field which allows it
to be
>>       used to support those cases.
>>
>>     A URL for this Internet-Draft is:
>>    =20
>> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri-de
>> livery-01.txt
>>
>>     Internet-Drafts are also available by anonymous FTP at:
>>     ftp://ftp.ietf.org/internet-drafts/
>>
>>     Below is the data which will enable a MIME compliant mail reader
>>     implementation to automatically retrieve the ASCII version of the
>>     Internet-Draft.
>     Content-Type: text/plain<BR>Content-ID:
>     &lt;2009-3-9102756.I-D@ietf.org
>     <mailto:2009-3-9102756.I-D@ietf.org>&gt;<BR><BR>
>>     _______________________________________________
>>     I-D-Announce mailing list
>>     I-D-Announce@ietf.org <mailto:I-D-Announce@ietf.org>
>>     https://www.ietf.org/mailman/listinfo/i-d-announce
>>     Internet-Draft directories: http://www.ietf.org/shadow.html
>>     or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>
>     _______________________________________________
>     Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>     This list is for NEW development of the core SIP Protocol
>     Use sip-implementors@cs.columbia.edu
>     <mailto:sip-implementors@cs.columbia.edu> for questions on current
sip
>     Use sipping@ietf.org <mailto:sipping@ietf.org> for new
>     developments on the application of sip
>
>

From mary.barnes@nortel.com  Wed Mar 11 15:40:43 2009
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Date: Wed, 11 Mar 2009 17:43:46 -0500
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Thread-Topic: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
Thread-Index: AcmilXHktBatMHeqQMOnx1ZHfxdInQAA4AEg
References: <20090304151501.4951728C331@core3.amsl.com>	<C0E80510684FE94DBDE3A4AF6B968D2D053B65B5@esealmw118.eemea.ericsson.se> <1ECE0EB50388174790F9694F77522CCF1CCA67CB@zrc2hxm0.corp.nortel.com> <49B8357E.2040809@gmail.com>
From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Hans Erik van Elburg" <ietf.hanserik@gmail.com>
Cc: sip@ietf.org, Ian Elz <ian.elz@ericsson.com>, Francois Audet <audet@nortel.com>
Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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I'm not sure if I'm clear on your concern here - what is the "something"
in the "Why should History-Info enforce..."?=20

If by the "something" you mean removing the history-info header (if it's
session or header level privacy), we have to go back to the fundamental
History-Info solution requirements and consider the functionality
provided by the Privacy header in RFC 3324. This isn't about
application-agnostic or not.  The information in the hi-entries can be
considered of the same ilk as the other information that is intended to
be kept private by the use of the Privacy header, thus if the request
indicates session or header levels of privacy, the proxy SHOULD NOT
forward the hi-entries. Note, it's a SHOULD NOT. If your network
configuration is such that there is no privacy issue with sharing that
information, then you can document as such and explain why it's
perfectly okay to forward the hi-entries.=20

However, per my note below, even if the prior hop strips out information
that is appropriate to the next domain, the last hi-entry can be added
by that next hop proxy to preserve the information before that proxy
might change the request-uri.=20

Mary.=20

-----Original Message-----
From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
Sent: Wednesday, March 11, 2009 5:05 PM
To: Barnes, Mary (RICH2:AR00)
Cc: Ian Elz; Audet, Francois (SC100:3055); sip@ietf.org
Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt

I believe in this case the History-Info application-agnosticness
principle applies.

Why should History-Info enforce something, that goes against the whishes
of a domain that is adding a hi-entry and should be considered best
equiped in judging what privacy rules apply for this specific entry.

/Hans Erik

Mary Barnes wrote:
> Ian,
>
> This is an interesting question. I need to think about it some more,=20
> but I don't like the "none" idea as it really must be up to the entity

> that added the Privacy header to the request, as to whether it wants=20
> the HI entries that it adds to go outside a domain. My initial thought

> is that we can't overrule the overall Privacy header. The thing is=20
> that the Privacy header doesn't preclude gathering the information and

> using within a domain AND if it were to not include an hi-entry when=20
> the request leaves the domain for which the proxy is responsible, the=20
> recipient can add the hi-entry for the current request-uri before it=20
> adds the new hi-entry.
>
> Your PSTN example doesn't strictly map obviously to a SIP model and I=20
> would think you might consider the PSTN hop to be within the same=20
> domain for which the proxy (or gw, I guess) is responsible OR consider

> this a walled garden whereby the Privacy header doesn't apply. Which=20
> brings me to a more general question as to how you all deal with other

> headers when you do your mapping to the PSTN when there is a Privacy=20
> header?  It would seem this problem isn't unique to History-Info,=20
> although I know little about the details of PSTN I/W.
>
> Regards,
> Mary.=20
>
> -----Original Message-----
> From: Ian Elz [mailto:ian.elz@ericsson.com]
> Sent: Wednesday, March 11, 2009 4:54 AM
> To: Barnes, Mary (RICH2:AR00); Audet, Francois (SC100:3055)
> Cc: sip@ietf.org
> Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt
>
> Mary, Francois,
>
> Section 4.1 of the draft allows for the addition of the Privacy header

> parameter with a value of "history" to be included with the=20
> hi-targeted-to-uri.
>
> Can this be extended to also allow the value "none".
>
> A node adding a H-I entry is allowed to specify the privacy value=20
> "history" either in the Privacy header or as a header parameter=20
> associated with the hi-targeted-to-uri. If the former action is taken=20
> by one node this results in privacy for all history info header=20
> entries even if this is not required. There is no way for privacy of=20
> an individual H-I entry to be set to none if no Privacy is required=20
> for the uri.
>
> This creates a specific problem when interworking with the PSTN where=20
> the privacy is associated with each E.164 number included in the=20
> protocol. If an INVITE is being mapped to ISUP and the H-I entries are

> being used to map to redirection numbers then a single Privacy header=20
> with the value "history" will result in all numbers being restricted.
>
> If the Privacy header parameter could include the value "none" then=20
> this would explicitly indicate that the associated uri was allowed to=20
> be presented.
>
> The ability to explicitly allow presentation of specific H-I entries=20
> may also be useful in pure sip implementations.
>
> I don't believe that this change would create any backward=20
> compatibility issues. The existing implementations will continue as=20
> deployed but new implementations could explicitly set no privacy for a
specific uri.
>
> There are no security issues other than those already defined in the=20
> draft that an intervening node could modify an existing H-I entry. The

> Privacy header value of "none" should only be included by the node in=20
> the same manner as the existing inclusion of the "history" value.
>
> Ian Elz
>
>
> -----Original Message-----
> From: i-d-announce-bounces@ietf.org
> [mailto:i-d-announce-bounces@ietf.org] On Behalf Of=20
> Internet-Drafts@ietf.org
> Sent: 04 March 2009 15:15
> To: i-d-announce@ietf.org
> Subject: I-D Action:draft-barnes-sip-rfc4244bis-00.txt
>
> A New Internet-Draft is available from the on-line Internet-Drafts=20
> directories.
>
> 	Title           : An Extension to the Session Initiation
> Protocol (SIP) for Request History Information
> 	Author(s)       : M. Barnes, F. Audet
> 	Filename        : draft-barnes-sip-rfc4244bis-00.txt
> 	Pages           : 49
> 	Date            : 2009-03-04
>
> This document defines a standard mechanism for capturing the history=20
> information associated with a Session Initiation Protocol (SIP)
request.
> This capability enables many enhanced services by providing the=20
> information as to how and why a call arrives at a specific application

> or user.  This document defines a new optional SIP header,=20
> History-Info, for capturing the history information in requests.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-barnes-sip-rfc4244bis-00.txt
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> Below is the data which will enable a MIME compliant mail reader=20
> implementation to automatically retrieve the ASCII version of the=20
> Internet-Draft.
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> sipping@ietf.org for new developments on the application of sip
>  =20

From ietf.hanserik@gmail.com  Wed Mar 11 16:01:06 2009
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Date: Thu, 12 Mar 2009 00:01:38 +0100
From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
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Cc: sip@ietf.org, Ian Elz <ian.elz@ericsson.com>, Francois Audet <audet@nortel.com>
Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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When a proxy1 adds a Privacy header with the value history, then why 
should a proxy 2 not be able to say: fine but I override the general 
privacy rule for the specific hi-entry that I add or am responsible for?

I was reacting to  "I don't like the "none" idea as it really must be up 
to the entity that  added the Privacy header to the request, as to 
whether it wants the HI entries that it adds to go outside a domain."

/Hans Erik

Mary Barnes wrote:
> I'm not sure if I'm clear on your concern here - what is the "something"
> in the "Why should History-Info enforce..."? 
>
> If by the "something" you mean removing the history-info header (if it's
> session or header level privacy), we have to go back to the fundamental
> History-Info solution requirements and consider the functionality
> provided by the Privacy header in RFC 3324. This isn't about
> application-agnostic or not.  The information in the hi-entries can be
> considered of the same ilk as the other information that is intended to
> be kept private by the use of the Privacy header, thus if the request
> indicates session or header levels of privacy, the proxy SHOULD NOT
> forward the hi-entries. Note, it's a SHOULD NOT. If your network
> configuration is such that there is no privacy issue with sharing that
> information, then you can document as such and explain why it's
> perfectly okay to forward the hi-entries. 
>
> However, per my note below, even if the prior hop strips out information
> that is appropriate to the next domain, the last hi-entry can be added
> by that next hop proxy to preserve the information before that proxy
> might change the request-uri. 
>
> Mary. 
>
> -----Original Message-----
> From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com] 
> Sent: Wednesday, March 11, 2009 5:05 PM
> To: Barnes, Mary (RICH2:AR00)
> Cc: Ian Elz; Audet, Francois (SC100:3055); sip@ietf.org
> Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
>
> I believe in this case the History-Info application-agnosticness
> principle applies.
>
> Why should History-Info enforce something, that goes against the whishes
> of a domain that is adding a hi-entry and should be considered best
> equiped in judging what privacy rules apply for this specific entry.
>
> /Hans Erik
>
> Mary Barnes wrote:
>   
>> Ian,
>>
>> This is an interesting question. I need to think about it some more, 
>> but I don't like the "none" idea as it really must be up to the entity
>>     
>
>   
>> that added the Privacy header to the request, as to whether it wants 
>> the HI entries that it adds to go outside a domain. My initial thought
>>     
>
>   
>> is that we can't overrule the overall Privacy header. The thing is 
>> that the Privacy header doesn't preclude gathering the information and
>>     
>
>   
>> using within a domain AND if it were to not include an hi-entry when 
>> the request leaves the domain for which the proxy is responsible, the 
>> recipient can add the hi-entry for the current request-uri before it 
>> adds the new hi-entry.
>>
>> Your PSTN example doesn't strictly map obviously to a SIP model and I 
>> would think you might consider the PSTN hop to be within the same 
>> domain for which the proxy (or gw, I guess) is responsible OR consider
>>     
>
>   
>> this a walled garden whereby the Privacy header doesn't apply. Which 
>> brings me to a more general question as to how you all deal with other
>>     
>
>   
>> headers when you do your mapping to the PSTN when there is a Privacy 
>> header?  It would seem this problem isn't unique to History-Info, 
>> although I know little about the details of PSTN I/W.
>>
>> Regards,
>> Mary. 
>>
>> -----Original Message-----
>> From: Ian Elz [mailto:ian.elz@ericsson.com]
>> Sent: Wednesday, March 11, 2009 4:54 AM
>> To: Barnes, Mary (RICH2:AR00); Audet, Francois (SC100:3055)
>> Cc: sip@ietf.org
>> Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt
>>
>> Mary, Francois,
>>
>> Section 4.1 of the draft allows for the addition of the Privacy header
>>     
>
>   
>> parameter with a value of "history" to be included with the 
>> hi-targeted-to-uri.
>>
>> Can this be extended to also allow the value "none".
>>
>> A node adding a H-I entry is allowed to specify the privacy value 
>> "history" either in the Privacy header or as a header parameter 
>> associated with the hi-targeted-to-uri. If the former action is taken 
>> by one node this results in privacy for all history info header 
>> entries even if this is not required. There is no way for privacy of 
>> an individual H-I entry to be set to none if no Privacy is required 
>> for the uri.
>>
>> This creates a specific problem when interworking with the PSTN where 
>> the privacy is associated with each E.164 number included in the 
>> protocol. If an INVITE is being mapped to ISUP and the H-I entries are
>>     
>
>   
>> being used to map to redirection numbers then a single Privacy header 
>> with the value "history" will result in all numbers being restricted.
>>
>> If the Privacy header parameter could include the value "none" then 
>> this would explicitly indicate that the associated uri was allowed to 
>> be presented.
>>
>> The ability to explicitly allow presentation of specific H-I entries 
>> may also be useful in pure sip implementations.
>>
>> I don't believe that this change would create any backward 
>> compatibility issues. The existing implementations will continue as 
>> deployed but new implementations could explicitly set no privacy for a
>>     
> specific uri.
>   
>> There are no security issues other than those already defined in the 
>> draft that an intervening node could modify an existing H-I entry. The
>>     
>
>   
>> Privacy header value of "none" should only be included by the node in 
>> the same manner as the existing inclusion of the "history" value.
>>
>> Ian Elz
>>
>>
>> -----Original Message-----
>> From: i-d-announce-bounces@ietf.org
>> [mailto:i-d-announce-bounces@ietf.org] On Behalf Of 
>> Internet-Drafts@ietf.org
>> Sent: 04 March 2009 15:15
>> To: i-d-announce@ietf.org
>> Subject: I-D Action:draft-barnes-sip-rfc4244bis-00.txt
>>
>> A New Internet-Draft is available from the on-line Internet-Drafts 
>> directories.
>>
>> 	Title           : An Extension to the Session Initiation
>> Protocol (SIP) for Request History Information
>> 	Author(s)       : M. Barnes, F. Audet
>> 	Filename        : draft-barnes-sip-rfc4244bis-00.txt
>> 	Pages           : 49
>> 	Date            : 2009-03-04
>>
>> This document defines a standard mechanism for capturing the history 
>> information associated with a Session Initiation Protocol (SIP)
>>     
> request.
>   
>> This capability enables many enhanced services by providing the 
>> information as to how and why a call arrives at a specific application
>>     
>
>   
>> or user.  This document defines a new optional SIP header, 
>> History-Info, for capturing the history information in requests.
>>
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-barnes-sip-rfc4244bis-00.txt
>>
>> Internet-Drafts are also available by anonymous FTP at:
>> ftp://ftp.ietf.org/internet-drafts/
>>
>> Below is the data which will enable a MIME compliant mail reader 
>> implementation to automatically retrieve the ASCII version of the 
>> Internet-Draft.
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use 
>> sip-implementors@cs.columbia.edu for questions on current sip Use 
>> sipping@ietf.org for new developments on the application of sip
>>   
>>     

From fluffy@cisco.com  Wed Mar 11 16:06:26 2009
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Cc: SIP List <sip@ietf.org>, Jon Peterson <jon.peterson@neustar.biz>
Subject: Re: [Sip] Seeking consensus on draft-elwell-sip-e2e-identity-important
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I continue to believe that we need to to come to some common  
understanding of the security problem we are trying to solve and the  
constraints on that solution. On one of the close to last phones calls  
involving this draft I asked everyone one on the call what they  
thought the problem was. I got no common answer.

There is a lot of good information in this draft related to the  
problem and solutions, but I don't believe this draft gets to what  
problem we are trying to solve thus making it very hard to answer a  
question like if the solution should involve changing m line or not.  
If we all agreed the problem was, just for example, stop MITM from  
listening to phone calls while still allowing intermediaries to do  
media steering, then I think we could get there. I know the problem is  
much more complicated than I just said - I'm not suggesting that my  
above statement is the right problem statement but just that it is  
closer to level we should be looking at the problem at.

Cullen


On Mar 3, 2009, at 9:46 AM, Elwell, John wrote:

> Draft 03:
> http://tools.ietf.org/html/draft-elwell-sip-e2e-identity-important-03
> includes for the first time an analysis of changes to SIP requests by
> B2BUAs (section 8). Also the Conclusions (section 9) have been  
> extended.
>
> I and a number of people who have assisted in revising this draft  
> would
> like to seek consensus on a number of the points. We aim to ask these
> questions in San Francisco, but obviously it would help to get  
> opinions
> from the list in advance.
>
> 1. Tolerance to changes of certain elements as a SIP request is
> forwarded through intermediate domains.
>
> Section 8 analyses a number of elements of SIP messages that are  
> signed
> by the present RFC 4474 solution to end-to-end (or near end-to-end)
> authenticated identification. For some elements it claims that any
> solution for end-to-end authenticated identification needs to be
> tolerant of changes by intermediate domains, since there appear to be
> valid reasons why B2BUAs (including but not limited to SBCs) change
> these elements. Picking on some of these:
>
> A) Is there agreement that a solution needs to be tolerant of  
> changes to
> IP addresses and ports in c- and m-lines in SDP? In other words, that
> intermediate domains can legitimately modify these fields (e.g., for
> media steering) and a solution is needed for end-to-end authenticated
> identification in such circumstances?
>
> B) Is there agreement that a solution needs to be tolerant of  
> changes to
> Contact (addr-spec), Call-Id (call-id) and CSeq (digit) header fields?
> In other words, that intermediate domains can legitimately modify  
> these
> fields (e.g., for topology hiding) and a solution is needed for
> end-to-end authenticated identification in such circumstances?
>
> C) Is there agreement that a solution does not need to be tolerant of
> changes to the addr-spec of From and To header fields? In other words,
> that intermediate domains should not modify these fields and can  
> expect
> to break end-to-end authenticated identification if they do so?
>
>
> 2. Are people in broad agreement with the conclusions in section 9
> concerning the importance of end-to-end identification and in  
> particular
> end-to-end authenticated identification?
>
> John
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


From mary.barnes@nortel.com  Wed Mar 11 16:28:43 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Hans Erik van Elburg" <ietf.hanserik@gmail.com>
Cc: sip@ietf.org, Ian Elz <ian.elz@ericsson.com>, Francois Audet <audet@nortel.com>
Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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And, if you have a network whereby you believe that none of the
information that Proxy 2 will put in hi-entries reveals any information
that is considered private, then you can set your flag, or whatever and
not drop the hi-entries when they leave that domain - that's why it's a
SHOULD NOT - i.e., don't do it unless you've evaluated why it's okay to
do it in that case.  It is clearly documented in 4244 that there is
indeed a loss of information when you use the Privacy header versus
tagging individual hi-entries with the privacy header. Now, we could add
a Recommendation that the Privacy header be used only to tag individual
entries.=20

Yes, I agree the "none" is not a good idea (and conflicts with the
Privacy header) as it kindof overrides it, so to speak.=20

Mary.=20


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Hans Erik van Elburg
Sent: Wednesday, March 11, 2009 6:02 PM
To: Barnes, Mary (RICH2:AR00)
Cc: sip@ietf.org; Ian Elz; Audet, Francois (SC100:3055)
Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt

When a proxy1 adds a Privacy header with the value history, then why
should a proxy 2 not be able to say: fine but I override the general
privacy rule for the specific hi-entry that I add or am responsible for?

I was reacting to  "I don't like the "none" idea as it really must be up
to the entity that  added the Privacy header to the request, as to
whether it wants the HI entries that it adds to go outside a domain."

/Hans Erik

Mary Barnes wrote:
> I'm not sure if I'm clear on your concern here - what is the
"something"
> in the "Why should History-Info enforce..."?=20
>
> If by the "something" you mean removing the history-info header (if=20
> it's session or header level privacy), we have to go back to the=20
> fundamental History-Info solution requirements and consider the=20
> functionality provided by the Privacy header in RFC 3324. This isn't=20
> about application-agnostic or not.  The information in the hi-entries=20
> can be considered of the same ilk as the other information that is=20
> intended to be kept private by the use of the Privacy header, thus if=20
> the request indicates session or header levels of privacy, the proxy=20
> SHOULD NOT forward the hi-entries. Note, it's a SHOULD NOT. If your=20
> network configuration is such that there is no privacy issue with=20
> sharing that information, then you can document as such and explain=20
> why it's perfectly okay to forward the hi-entries.
>
> However, per my note below, even if the prior hop strips out=20
> information that is appropriate to the next domain, the last hi-entry=20
> can be added by that next hop proxy to preserve the information before

> that proxy might change the request-uri.
>
> Mary.=20
>
> -----Original Message-----
> From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
> Sent: Wednesday, March 11, 2009 5:05 PM
> To: Barnes, Mary (RICH2:AR00)
> Cc: Ian Elz; Audet, Francois (SC100:3055); sip@ietf.org
> Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
>
> I believe in this case the History-Info application-agnosticness=20
> principle applies.
>
> Why should History-Info enforce something, that goes against the=20
> whishes of a domain that is adding a hi-entry and should be considered

> best equiped in judging what privacy rules apply for this specific
entry.
>
> /Hans Erik
>
> Mary Barnes wrote:
>  =20
>> Ian,
>>
>> This is an interesting question. I need to think about it some more,=20
>> but I don't like the "none" idea as it really must be up to the=20
>> entity
>>    =20
>
>  =20
>> that added the Privacy header to the request, as to whether it wants=20
>> the HI entries that it adds to go outside a domain. My initial=20
>> thought
>>    =20
>
>  =20
>> is that we can't overrule the overall Privacy header. The thing is=20
>> that the Privacy header doesn't preclude gathering the information=20
>> and
>>    =20
>
>  =20
>> using within a domain AND if it were to not include an hi-entry when=20
>> the request leaves the domain for which the proxy is responsible, the

>> recipient can add the hi-entry for the current request-uri before it=20
>> adds the new hi-entry.
>>
>> Your PSTN example doesn't strictly map obviously to a SIP model and I

>> would think you might consider the PSTN hop to be within the same=20
>> domain for which the proxy (or gw, I guess) is responsible OR=20
>> consider
>>    =20
>
>  =20
>> this a walled garden whereby the Privacy header doesn't apply. Which=20
>> brings me to a more general question as to how you all deal with=20
>> other
>>    =20
>
>  =20
>> headers when you do your mapping to the PSTN when there is a Privacy=20
>> header?  It would seem this problem isn't unique to History-Info,=20
>> although I know little about the details of PSTN I/W.
>>
>> Regards,
>> Mary.=20
>>
>> -----Original Message-----
>> From: Ian Elz [mailto:ian.elz@ericsson.com]
>> Sent: Wednesday, March 11, 2009 4:54 AM
>> To: Barnes, Mary (RICH2:AR00); Audet, Francois (SC100:3055)
>> Cc: sip@ietf.org
>> Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt
>>
>> Mary, Francois,
>>
>> Section 4.1 of the draft allows for the addition of the Privacy=20
>> header
>>    =20
>
>  =20
>> parameter with a value of "history" to be included with the=20
>> hi-targeted-to-uri.
>>
>> Can this be extended to also allow the value "none".
>>
>> A node adding a H-I entry is allowed to specify the privacy value=20
>> "history" either in the Privacy header or as a header parameter=20
>> associated with the hi-targeted-to-uri. If the former action is taken

>> by one node this results in privacy for all history info header=20
>> entries even if this is not required. There is no way for privacy of=20
>> an individual H-I entry to be set to none if no Privacy is required=20
>> for the uri.
>>
>> This creates a specific problem when interworking with the PSTN where

>> the privacy is associated with each E.164 number included in the=20
>> protocol. If an INVITE is being mapped to ISUP and the H-I entries=20
>> are
>>    =20
>
>  =20
>> being used to map to redirection numbers then a single Privacy header

>> with the value "history" will result in all numbers being restricted.
>>
>> If the Privacy header parameter could include the value "none" then=20
>> this would explicitly indicate that the associated uri was allowed to

>> be presented.
>>
>> The ability to explicitly allow presentation of specific H-I entries=20
>> may also be useful in pure sip implementations.
>>
>> I don't believe that this change would create any backward=20
>> compatibility issues. The existing implementations will continue as=20
>> deployed but new implementations could explicitly set no privacy for=20
>> a
>>    =20
> specific uri.
>  =20
>> There are no security issues other than those already defined in the=20
>> draft that an intervening node could modify an existing H-I entry.=20
>> The
>>    =20
>
>  =20
>> Privacy header value of "none" should only be included by the node in

>> the same manner as the existing inclusion of the "history" value.
>>
>> Ian Elz
>>
>>
>> -----Original Message-----
>> From: i-d-announce-bounces@ietf.org
>> [mailto:i-d-announce-bounces@ietf.org] On Behalf Of=20
>> Internet-Drafts@ietf.org
>> Sent: 04 March 2009 15:15
>> To: i-d-announce@ietf.org
>> Subject: I-D Action:draft-barnes-sip-rfc4244bis-00.txt
>>
>> A New Internet-Draft is available from the on-line Internet-Drafts=20
>> directories.
>>
>> 	Title           : An Extension to the Session Initiation
>> Protocol (SIP) for Request History Information
>> 	Author(s)       : M. Barnes, F. Audet
>> 	Filename        : draft-barnes-sip-rfc4244bis-00.txt
>> 	Pages           : 49
>> 	Date            : 2009-03-04
>>
>> This document defines a standard mechanism for capturing the history=20
>> information associated with a Session Initiation Protocol (SIP)
>>    =20
> request.
>  =20
>> This capability enables many enhanced services by providing the=20
>> information as to how and why a call arrives at a specific=20
>> application
>>    =20
>
>  =20
>> or user.  This document defines a new optional SIP header,=20
>> History-Info, for capturing the history information in requests.
>>
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-barnes-sip-rfc4244bis-00.tx
>> t
>>
>> Internet-Drafts are also available by anonymous FTP at:
>> ftp://ftp.ietf.org/internet-drafts/
>>
>> Below is the data which will enable a MIME compliant mail reader=20
>> implementation to automatically retrieve the ASCII version of the=20
>> Internet-Draft.
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use=20
>> sip-implementors@cs.columbia.edu for questions on current sip Use=20
>> sipping@ietf.org for new developments on the application of sip
>>  =20
>>    =20
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol Use
sip-implementors@cs.columbia.edu for questions on current sip Use
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Cc: sip@ietf.org
Subject: Re: [Sip] Terminology (was RE: Fwd: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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I was talking about the concept.

The use cases only describe  cases where the target-uri (lets call that 
current target from now) has been lost when an initial request for a 
dialog or standalone request arrives at the UAS.

What has been lost is the current target of the request:


      Current target

   The current target of an initial request for a dialog or standalone
   request is the name or address to which the request is targeted, i.e.
   either the initial target inserted in the Request-URI by the UAC that
   originates the request, or when a retarget occurred, the target
   provided in that retarget operation.  Reroute and translation
   operations never change the current target.


This defintion only makes sense when the following definitions are used:

Name:

   A name is a moniker for an entity which refers to it in a way which

   reveals nothing about where it is in a network.  In SIP, tel URI

   which doesn't represent the location of the entity is a name.   


Address:

   An address is an identifier for an entity which describes it by its

   location on the network.  In SIP, the SIP URI itself is a form of

   address because the host part of the URI, the only mandatory part of

   the URI besides the scheme itself, indicates the location of a SIP

   server that can be used to handle the request.

Route:

   Finally, a route is a   sequence of SIP entities (including the UA 
itself!) which are

 traversed in order to forward a request to an address or name.

Retarget (other term might be needed, as this is highly confusing):

    A Request-URI rewrite operation that changes the target identity of 
the request.

Reroute (other term might be needed):

    A Request-URI rewrite operation that does not change the current 
target of the request, but     determines the route/next hop taken to 
reach the target-identity.

/Hans Erik

Mary Barnes wrote:
> So, by "target-uri concept" are you referring to the solution in the
> current target-uri draft or the application usage as such in the
> target-uri document (which is what I was referring to)?   
>
> Yes, the problem is that SIP doesn't differentiate the various
> mechanisms by which a request-URI may be changed, which is what some of
> the updated text in 4244bis is trying to do and why we need precise
> terms. I think we all agree that. 
>
> I honestly don't care what we call the tag, as long as we're clear about
> the functionality. 
>
> Mary. 
>
> -----Original Message-----
> From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com] 
> Sent: Wednesday, March 11, 2009 4:40 PM
> To: Barnes, Mary (RICH2:AR00)
> Cc: Shida Schubert; sip@ietf.org
> Subject: Re: Terminology (was RE: [Sip] Fwd: I-D
> ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>
> The target-uri concept when defined properly is entirely application
> agnostic.
>
> When one uses History-Info or another header as a vehicle for this
> information then it still can be considered application agnostic as any
> application can use this information as it likes.
>
>  From SIP perspective all the Request-URI rewrites look the same, that
> is what brought us discussing this in the first place.
>
> /Hans Erik
>
> Mary Barnes wrote:
>   
>> And, I think the big issue with the terminology is that 4244bis is 
>> application agnostic, so it is tagging what happens to the URIs 
>> (why/how the specific URI was overwritten).  Whereas, specific 
>> applications can derive information, such as what "is" the real
>> "target" for the request, from the History-Info header.   So, IMHO, 
>> it's a matter of the target-uri document specifying that the last 
>> entry tagged by whatever name we come up with "is" the real "target"
>> for the request.  I believe it's up to the applications to describe 
>> how they make use of the History-Info and not for the History-Info to 
>> describe how applications can use it.  History-Info purely reflects 
>> what happened from a SIP Protocol perspective and should not imply any
>>     
>
>   
>> application semantics. Indeed the intent of section 5 of 4244 was to 
>> inform the applications how they should decribe their usage of the 
>> header.
>>  
>> Mary.
>> Note: I changed the subject to hopefully make this topic easier to 
>> keep track of.
>>
>> ----------------------------------------------------------------------
>> --
>> *From:* Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
>> *Sent:* Wednesday, March 11, 2009 3:59 AM
>> *To:* Shida Schubert
>> *Cc:* sip@ietf.org List; Barnes, Mary (RICH2:AR00)
>> *Subject:* Re: [Sip] Fwd: I-D
>> ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>>
>> One of the problems that made the discussion quite tedious is the 
>> complete inversed meaning of the terms "retarget" and "reroute"
>> between 4244bis and target-uri factions have.
>>
>> Conclusion was that the terminology need to be properly defined and 
>> some attempt was made to start such activity.
>>
>> On 3. I have strong concerns when we start tagging the URI as to how 
>> they come about "retarget"/"mapped conficuration" etc. It is better to
>>     
>
>   
>> embellish them with a tag that just represents there meaning for 
>> example "istarget" or "hop". For the following reasons:
>> 1. It is much more intuitive for the user of this information, its 
>> meaning can basically be guessed.
>> 2. Such meaning also probably survives application in other problem 
>> spaces that we had not foreseen when introducing the concept.
>>
>> The solution that is  described now in the target-URI delivery draft 
>> is not yet complete, it does not solve the freephone use case for 
>> example and in its current form it will deliver exactly the same 
>> target-URI as that which would be delivered by the  P-Called-Party-ID 
>> header. Contrary to what the draft says. So some work is needed still.
>>
>> /Hans Erik van Elburg
>>
>>
>> On Wed, Mar 11, 2009 at 7:13 AM, Shida Schubert <shida@ntt-at.com 
>> <mailto:shida@ntt-at.com>> wrote:
>>
>>
>>     We have submitted the updated target-uri draft based on the
>>     
> comments 
>   
>>     submitted to the list and comments received at IETF73. 
>>
>>     I have taken over as editor as Jonathan didn't have the cycles to 
>>     update the draft, with Francois, Christer and Hans Erick as
>>     additional 
>>     co-authors and great deal of help from Mary.  
>>      
>>     The following summarizes the changes made to the target-uri
>>     
> document
>   
>>     1. Added use-case for toll-free number back
>>     2. Added definition of "retarget" operation. 
>>     3. Removed a reference to URN
>>     4. Added a text discussing the difference to P-Called-Party-Id
>>     5. Changed parameter name from "target" to "istarget"
>>      
>>     Note, that the target-uri document still contains the normative
>>     text for the
>>     History-Info header.
>>      
>>     In addition, Mary (with Francois as co-author) has submitted a
>>     rfc4244bis, with the following changes:
>>     1. Incorporated the normative aspects of the target-uri document
>>     into the existing normative text in RFC 4244 - the functionality
>>     
> is 
>   
>>     virtually identical (as is some of the text) as the HI based
>>     
> solution 
>   
>>     described in the target-uri document.  It's important that the
>>     solution
>>     be integrated into RFC 4244 as it MUST work and be based on the
>>     normative
>>     aspects of RFC 4244.
>>     2.  Added the use cases from target-uri the the summary
>>     in the overview of rfc4244bis.
>>     3. Added an additional requirement to capture the "target-uri"
>>     information.
>>     4. Fixed an error in the RFC 4244 ABNF and added "retarget"
>>     
> parameter.
>   
>>     5. Added a more simplified example.
>>      
>>      
>>     We had some very long offline exchanges as to the best way forward
>>     
> and
>   
>>     remaining work for both documents.
>>      
>>     Some of the issues identified are:
>>
>>     ::Issues::
>>        1. Should we remove the normative text from target-uri and
>>     
> progress
>   
>>            4244bis along with the target-uri document to meet the
>>     chartered
>>           SIP WG milestone? 
>>      
>>      
>>        2. Name of the parameter. 
>>           At the last meeting, parameter "target" was said
>>     
> inappropriate 
>   
>>           because voicemail-uri spec already defines a parameter
>>     
> called 
>   
>>           "target" which also can be found in hi-entry, thus
>>     
> potentially 
>   
>>           causing confusion.
>>          
>>           Currently the target-uri draft uses "istarget" and 4244bis
>>     
> uses
>   
>>           "retarget"  but we could never come to 
>>           a consensus on what name is appropriate.  Other suggestions
>>     
> have
>   
>>           included the following:
>>           "target-identity" (someone didn't like that "identity" is
>>     also a SIP header)
>>           "reg-uri"  (can be paired with "mapped-uri" for item 3
>>     
> below)
>   
>>           "aor"
>>           "jibberish" 
>>           etc.
>>      
>>           One reason this is so difficult relates to the problem
>>     statement in target-uri in that
>>           RFC 3261 doesn't differentiate the mechanism by which the
>>     
> new
>   
>>           (target) Request-URI is selected.  Another issue is that
>>     some of the terminology in
>>           RFC 3261 is overloaded - e.g., "forwarding" refers both to a
>>     Proxy
>>           which does not have responsibility for the domain of the
>>     request-URI
>>           in the incoming request, thus the proxy just "forwards" the
>>     request to
>>           the next hop AND "forwarding" is used to describe the
>>     process whereby
>>           the outgoing request is built and "forwarded" to the next
>>     hop at which
>>           point the proxy does not know how the new request-uri was
>>     selected.
>>           RFC 4244 has attempted to clarify the terms and attempts to
>>     use "forward" 
>>           in the context of the former situation and "retarget" for
>>     the case whereby
>>           a proxy is responsible for the domain and thus can use a
>>     number of 
>>           mechanism to select the new target for the request - e.g., a
>>     REGISTRAR, 
>>           configured data, etc.  
>>
>>      3.  Related to the last point in item 2 above, it has been
>>     proposed that
>>           we differentiate the hi-entries even more by defining
>>     separate parameters
>>          for registered and configured/mapped contacts. 
>>          Currently when the R-URI is translated to a URI which is
>>     either derived 
>>          from location service lookup(registered by UA) or from
>>     
> mapping 
>   
>>          table, there is no differentiation as to how the URI was
>>     derived once it is
>>          added to the list of potential targets.
>>          
>>          The general consensus of the authors of the two documents was
>>     that it may 
>>          be useful for some services to have the hi-entries tagged
>>     with the
>>          more specific information. 
>>      
>>          And, of course, this gets us into another naming contest. In
>>     the end, the naming
>>          is not so important as long as the term isn't too overloaded
>>     and it is defined
>>          precisely in the document(s).
>>      
>>     We would appreciate WG feedback on these issues and any other
>>     comments on
>>     the two documents prior to IETF-74.
>>      
>>     Regards,
>>     Shida and Mary.
>>
>>
>>
>>     Begin forwarded message:
>>     
>>>     *From: *Internet-Drafts@ietf.org
>>>       
> <mailto:Internet-Drafts@ietf.org>
>   
>>>     *Date: *March 10, 2009 2:30:01 AM JST
>>>     *To: *i-d-announce@ietf.org <mailto:i-d-announce@ietf.org>
>>>     *Subject: **I-D
>>>     ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt *
>>>     *Reply-To: *internet-drafts@ietf.org
>>>     <mailto:internet-drafts@ietf.org>
>>>
>>>     A New Internet-Draft is available from the on-line
>>>       
> Internet-Drafts 
>   
>>>     directories.
>>>
>>>
>>>     Title
>>>     : Delivery of Request-URI Targets to User Agents
>>>     Author(s)
>>>     : J. Rosenberg, H. van Elburg, C. Holmberg, F. Audet, S. Schubert
>>>     Filename : draft-rosenberg-sip-target-uri-delivery-01.txt
>>>     Pages
>>>     : 16
>>>     Date
>>>     : 2009-3-9
>>>
>>>     When a Session Initiation Protocol (SIP) proxy receives a request
>>>       targeted at a URI identifying a user or resource it is
>>>       
> responsible
>   
>>>       for, the proxy translates the URI to a registered or configured
>>>       contact URI of an agent representing that user or resource.  In
>>>       
> the
>   
>>>       process, the original URI is removed from the request.
>>>      Numerous use
>>>       cases have arisen which require this information to be
>>>       
> delivered to
>   
>>>       the user agent.  This document describes these use cases and
>>>     defines
>>>       an extension to the History-Info header field which allows it
>>>       
> to be
>   
>>>       used to support those cases.
>>>
>>>     A URL for this Internet-Draft is:
>>>     
>>> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri-de
>>> livery-01.txt
>>>
>>>     Internet-Drafts are also available by anonymous FTP at:
>>>     ftp://ftp.ietf.org/internet-drafts/
>>>
>>>     Below is the data which will enable a MIME compliant mail reader
>>>     implementation to automatically retrieve the ASCII version of the
>>>     Internet-Draft.
>>>       
>>     Content-Type: text/plain<BR>Content-ID:
>>     &lt;2009-3-9102756.I-D@ietf.org
>>     <mailto:2009-3-9102756.I-D@ietf.org>&gt;<BR><BR>
>>     
>>>     _______________________________________________
>>>     I-D-Announce mailing list
>>>     I-D-Announce@ietf.org <mailto:I-D-Announce@ietf.org>
>>>     https://www.ietf.org/mailman/listinfo/i-d-announce
>>>     Internet-Draft directories: http://www.ietf.org/shadow.html
>>>     or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>>>       
>>     _______________________________________________
>>     Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>     This list is for NEW development of the core SIP Protocol
>>     Use sip-implementors@cs.columbia.edu
>>     <mailto:sip-implementors@cs.columbia.edu> for questions on current
>>     
> sip
>   
>>     Use sipping@ietf.org <mailto:sipping@ietf.org> for new
>>     developments on the application of sip
>>
>>
>>     

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Cc: sip@ietf.org, Ian Elz <ian.elz@ericsson.com>, Francois Audet <audet@nortel.com>
Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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But you do not have control over a previously visited proxy using the 
header. Why should that one dictate the privacy needs of the next hop 
proxy in the same domain?

I dont think there is a conflict it is just adding some higher level of 
granularity in controlling the privacy of specific entries.

/Hans Erik

Mary Barnes wrote:
> And, if you have a network whereby you believe that none of the
> information that Proxy 2 will put in hi-entries reveals any information
> that is considered private, then you can set your flag, or whatever and
> not drop the hi-entries when they leave that domain - that's why it's a
> SHOULD NOT - i.e., don't do it unless you've evaluated why it's okay to
> do it in that case.  It is clearly documented in 4244 that there is
> indeed a loss of information when you use the Privacy header versus
> tagging individual hi-entries with the privacy header. Now, we could add
> a Recommendation that the Privacy header be used only to tag individual
> entries. 
>
> Yes, I agree the "none" is not a good idea (and conflicts with the
> Privacy header) as it kindof overrides it, so to speak. 
>
> Mary. 
>
>
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Hans Erik van Elburg
> Sent: Wednesday, March 11, 2009 6:02 PM
> To: Barnes, Mary (RICH2:AR00)
> Cc: sip@ietf.org; Ian Elz; Audet, Francois (SC100:3055)
> Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
>
> When a proxy1 adds a Privacy header with the value history, then why
> should a proxy 2 not be able to say: fine but I override the general
> privacy rule for the specific hi-entry that I add or am responsible for?
>
> I was reacting to  "I don't like the "none" idea as it really must be up
> to the entity that  added the Privacy header to the request, as to
> whether it wants the HI entries that it adds to go outside a domain."
>
> /Hans Erik
>
> Mary Barnes wrote:
>   
>> I'm not sure if I'm clear on your concern here - what is the
>>     
> "something"
>   
>> in the "Why should History-Info enforce..."? 
>>
>> If by the "something" you mean removing the history-info header (if 
>> it's session or header level privacy), we have to go back to the 
>> fundamental History-Info solution requirements and consider the 
>> functionality provided by the Privacy header in RFC 3324. This isn't 
>> about application-agnostic or not.  The information in the hi-entries 
>> can be considered of the same ilk as the other information that is 
>> intended to be kept private by the use of the Privacy header, thus if 
>> the request indicates session or header levels of privacy, the proxy 
>> SHOULD NOT forward the hi-entries. Note, it's a SHOULD NOT. If your 
>> network configuration is such that there is no privacy issue with 
>> sharing that information, then you can document as such and explain 
>> why it's perfectly okay to forward the hi-entries.
>>
>> However, per my note below, even if the prior hop strips out 
>> information that is appropriate to the next domain, the last hi-entry 
>> can be added by that next hop proxy to preserve the information before
>>     
>
>   
>> that proxy might change the request-uri.
>>
>> Mary. 
>>
>> -----Original Message-----
>> From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
>> Sent: Wednesday, March 11, 2009 5:05 PM
>> To: Barnes, Mary (RICH2:AR00)
>> Cc: Ian Elz; Audet, Francois (SC100:3055); sip@ietf.org
>> Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
>>
>> I believe in this case the History-Info application-agnosticness 
>> principle applies.
>>
>> Why should History-Info enforce something, that goes against the 
>> whishes of a domain that is adding a hi-entry and should be considered
>>     
>
>   
>> best equiped in judging what privacy rules apply for this specific
>>     
> entry.
>   
>> /Hans Erik
>>
>> Mary Barnes wrote:
>>   
>>     
>>> Ian,
>>>
>>> This is an interesting question. I need to think about it some more, 
>>> but I don't like the "none" idea as it really must be up to the 
>>> entity
>>>     
>>>       
>>   
>>     
>>> that added the Privacy header to the request, as to whether it wants 
>>> the HI entries that it adds to go outside a domain. My initial 
>>> thought
>>>     
>>>       
>>   
>>     
>>> is that we can't overrule the overall Privacy header. The thing is 
>>> that the Privacy header doesn't preclude gathering the information 
>>> and
>>>     
>>>       
>>   
>>     
>>> using within a domain AND if it were to not include an hi-entry when 
>>> the request leaves the domain for which the proxy is responsible, the
>>>       
>
>   
>>> recipient can add the hi-entry for the current request-uri before it 
>>> adds the new hi-entry.
>>>
>>> Your PSTN example doesn't strictly map obviously to a SIP model and I
>>>       
>
>   
>>> would think you might consider the PSTN hop to be within the same 
>>> domain for which the proxy (or gw, I guess) is responsible OR 
>>> consider
>>>     
>>>       
>>   
>>     
>>> this a walled garden whereby the Privacy header doesn't apply. Which 
>>> brings me to a more general question as to how you all deal with 
>>> other
>>>     
>>>       
>>   
>>     
>>> headers when you do your mapping to the PSTN when there is a Privacy 
>>> header?  It would seem this problem isn't unique to History-Info, 
>>> although I know little about the details of PSTN I/W.
>>>
>>> Regards,
>>> Mary. 
>>>
>>> -----Original Message-----
>>> From: Ian Elz [mailto:ian.elz@ericsson.com]
>>> Sent: Wednesday, March 11, 2009 4:54 AM
>>> To: Barnes, Mary (RICH2:AR00); Audet, Francois (SC100:3055)
>>> Cc: sip@ietf.org
>>> Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt
>>>
>>> Mary, Francois,
>>>
>>> Section 4.1 of the draft allows for the addition of the Privacy 
>>> header
>>>     
>>>       
>>   
>>     
>>> parameter with a value of "history" to be included with the 
>>> hi-targeted-to-uri.
>>>
>>> Can this be extended to also allow the value "none".
>>>
>>> A node adding a H-I entry is allowed to specify the privacy value 
>>> "history" either in the Privacy header or as a header parameter 
>>> associated with the hi-targeted-to-uri. If the former action is taken
>>>       
>
>   
>>> by one node this results in privacy for all history info header 
>>> entries even if this is not required. There is no way for privacy of 
>>> an individual H-I entry to be set to none if no Privacy is required 
>>> for the uri.
>>>
>>> This creates a specific problem when interworking with the PSTN where
>>>       
>
>   
>>> the privacy is associated with each E.164 number included in the 
>>> protocol. If an INVITE is being mapped to ISUP and the H-I entries 
>>> are
>>>     
>>>       
>>   
>>     
>>> being used to map to redirection numbers then a single Privacy header
>>>       
>
>   
>>> with the value "history" will result in all numbers being restricted.
>>>
>>> If the Privacy header parameter could include the value "none" then 
>>> this would explicitly indicate that the associated uri was allowed to
>>>       
>
>   
>>> be presented.
>>>
>>> The ability to explicitly allow presentation of specific H-I entries 
>>> may also be useful in pure sip implementations.
>>>
>>> I don't believe that this change would create any backward 
>>> compatibility issues. The existing implementations will continue as 
>>> deployed but new implementations could explicitly set no privacy for 
>>> a
>>>     
>>>       
>> specific uri.
>>   
>>     
>>> There are no security issues other than those already defined in the 
>>> draft that an intervening node could modify an existing H-I entry. 
>>> The
>>>     
>>>       
>>   
>>     
>>> Privacy header value of "none" should only be included by the node in
>>>       
>
>   
>>> the same manner as the existing inclusion of the "history" value.
>>>
>>> Ian Elz
>>>
>>>
>>> -----Original Message-----
>>> From: i-d-announce-bounces@ietf.org
>>> [mailto:i-d-announce-bounces@ietf.org] On Behalf Of 
>>> Internet-Drafts@ietf.org
>>> Sent: 04 March 2009 15:15
>>> To: i-d-announce@ietf.org
>>> Subject: I-D Action:draft-barnes-sip-rfc4244bis-00.txt
>>>
>>> A New Internet-Draft is available from the on-line Internet-Drafts 
>>> directories.
>>>
>>> 	Title           : An Extension to the Session Initiation
>>> Protocol (SIP) for Request History Information
>>> 	Author(s)       : M. Barnes, F. Audet
>>> 	Filename        : draft-barnes-sip-rfc4244bis-00.txt
>>> 	Pages           : 49
>>> 	Date            : 2009-03-04
>>>
>>> This document defines a standard mechanism for capturing the history 
>>> information associated with a Session Initiation Protocol (SIP)
>>>     
>>>       
>> request.
>>   
>>     
>>> This capability enables many enhanced services by providing the 
>>> information as to how and why a call arrives at a specific 
>>> application
>>>     
>>>       
>>   
>>     
>>> or user.  This document defines a new optional SIP header, 
>>> History-Info, for capturing the history information in requests.
>>>
>>> A URL for this Internet-Draft is:
>>> http://www.ietf.org/internet-drafts/draft-barnes-sip-rfc4244bis-00.tx
>>> t
>>>
>>> Internet-Drafts are also available by anonymous FTP at:
>>> ftp://ftp.ietf.org/internet-drafts/
>>>
>>> Below is the data which will enable a MIME compliant mail reader 
>>> implementation to automatically retrieve the ASCII version of the 
>>> Internet-Draft.
>>> _______________________________________________
>>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol Use 
>>> sip-implementors@cs.columbia.edu for questions on current sip Use 
>>> sipping@ietf.org for new developments on the application of sip
>>>   
>>>     
>>>       
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use
> sip-implementors@cs.columbia.edu for questions on current sip Use
> sipping@ietf.org for new developments on the application of sip
>   

From mary.barnes@nortel.com  Wed Mar 11 17:41:00 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Hans Erik van Elburg" <ietf.hanserik@gmail.com>
Cc: sip@ietf.org, Ian Elz <ian.elz@ericsson.com>, Francois Audet <audet@nortel.com>
Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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Responses below [MB].

-----Original Message-----
From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
Sent: Wednesday, March 11, 2009 6:56 PM
To: Barnes, Mary (RICH2:AR00)
Cc: sip@ietf.org; Ian Elz; Audet, Francois (SC100:3055)
Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt

But you do not have control over a previously visited proxy using the
header. Why should that one dictate the privacy needs of the next hop
proxy in the same domain?
[MB] If the Privacy header is for the request as a whole (i.e., it's not
tagged against a specific hi-entry), then by the definition of the
header in RFC 3323 (with the addition of the priv-value=3D"history"), it
does apply to how the handled is request at every node.  And, again, it
is a SHOULD NOT, so it is possible that there might be conditions under
which you don't want to "dictate the privacy needs..." - you just need
to specify the conditions.  Also, within the same domain, the hi-entries
are NOT removed even if the Privacy header is used. If Proxy 1 and Proxy
2 are responsible for the same domain, then you don't remove the
hi-entries until you leave that domain.  [/MB]

I dont think there is a conflict it is just adding some higher level of
granularity in controlling the privacy of specific entries.
[MB] I had interpreted the "none" to be used with the Privacy header
(for the whole request, not just specific hi-entries), so that the
specific entry can be forwarded outside a domain.  [/MB]

/Hans Erik

Mary Barnes wrote:
> And, if you have a network whereby you believe that none of the=20
> information that Proxy 2 will put in hi-entries reveals any=20
> information that is considered private, then you can set your flag, or

> whatever and not drop the hi-entries when they leave that domain -=20
> that's why it's a SHOULD NOT - i.e., don't do it unless you've=20
> evaluated why it's okay to do it in that case.  It is clearly=20
> documented in 4244 that there is indeed a loss of information when you

> use the Privacy header versus tagging individual hi-entries with the=20
> privacy header. Now, we could add a Recommendation that the Privacy=20
> header be used only to tag individual entries.
>
> Yes, I agree the "none" is not a good idea (and conflicts with the=20
> Privacy header) as it kindof overrides it, so to speak.
>
> Mary.=20
>
>
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of=20
> Hans Erik van Elburg
> Sent: Wednesday, March 11, 2009 6:02 PM
> To: Barnes, Mary (RICH2:AR00)
> Cc: sip@ietf.org; Ian Elz; Audet, Francois (SC100:3055)
> Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
>
> When a proxy1 adds a Privacy header with the value history, then why=20
> should a proxy 2 not be able to say: fine but I override the general=20
> privacy rule for the specific hi-entry that I add or am responsible
for?
>
> I was reacting to  "I don't like the "none" idea as it really must be=20
> up to the entity that  added the Privacy header to the request, as to=20
> whether it wants the HI entries that it adds to go outside a domain."
>
> /Hans Erik
>
> Mary Barnes wrote:
>  =20
>> I'm not sure if I'm clear on your concern here - what is the
>>    =20
> "something"
>  =20
>> in the "Why should History-Info enforce..."?=20
>>
>> If by the "something" you mean removing the history-info header (if=20
>> it's session or header level privacy), we have to go back to the=20
>> fundamental History-Info solution requirements and consider the=20
>> functionality provided by the Privacy header in RFC 3324. This isn't=20
>> about application-agnostic or not.  The information in the hi-entries

>> can be considered of the same ilk as the other information that is=20
>> intended to be kept private by the use of the Privacy header, thus if

>> the request indicates session or header levels of privacy, the proxy=20
>> SHOULD NOT forward the hi-entries. Note, it's a SHOULD NOT. If your=20
>> network configuration is such that there is no privacy issue with=20
>> sharing that information, then you can document as such and explain=20
>> why it's perfectly okay to forward the hi-entries.
>>
>> However, per my note below, even if the prior hop strips out=20
>> information that is appropriate to the next domain, the last hi-entry

>> can be added by that next hop proxy to preserve the information=20
>> before
>>    =20
>
>  =20
>> that proxy might change the request-uri.
>>
>> Mary.=20
>>
>> -----Original Message-----
>> From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
>> Sent: Wednesday, March 11, 2009 5:05 PM
>> To: Barnes, Mary (RICH2:AR00)
>> Cc: Ian Elz; Audet, Francois (SC100:3055); sip@ietf.org
>> Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
>>
>> I believe in this case the History-Info application-agnosticness=20
>> principle applies.
>>
>> Why should History-Info enforce something, that goes against the=20
>> whishes of a domain that is adding a hi-entry and should be=20
>> considered
>>    =20
>
>  =20
>> best equiped in judging what privacy rules apply for this specific
>>    =20
> entry.
>  =20
>> /Hans Erik
>>
>> Mary Barnes wrote:
>>  =20
>>    =20
>>> Ian,
>>>
>>> This is an interesting question. I need to think about it some more,

>>> but I don't like the "none" idea as it really must be up to the=20
>>> entity
>>>    =20
>>>      =20
>>  =20
>>    =20
>>> that added the Privacy header to the request, as to whether it wants

>>> the HI entries that it adds to go outside a domain. My initial=20
>>> thought
>>>    =20
>>>      =20
>>  =20
>>    =20
>>> is that we can't overrule the overall Privacy header. The thing is=20
>>> that the Privacy header doesn't preclude gathering the information=20
>>> and
>>>    =20
>>>      =20
>>  =20
>>    =20
>>> using within a domain AND if it were to not include an hi-entry when

>>> the request leaves the domain for which the proxy is responsible,=20
>>> the
>>>      =20
>
>  =20
>>> recipient can add the hi-entry for the current request-uri before it

>>> adds the new hi-entry.
>>>
>>> Your PSTN example doesn't strictly map obviously to a SIP model and=20
>>> I
>>>      =20
>
>  =20
>>> would think you might consider the PSTN hop to be within the same=20
>>> domain for which the proxy (or gw, I guess) is responsible OR=20
>>> consider
>>>    =20
>>>      =20
>>  =20
>>    =20
>>> this a walled garden whereby the Privacy header doesn't apply. Which

>>> brings me to a more general question as to how you all deal with=20
>>> other
>>>    =20
>>>      =20
>>  =20
>>    =20
>>> headers when you do your mapping to the PSTN when there is a Privacy

>>> header?  It would seem this problem isn't unique to History-Info,=20
>>> although I know little about the details of PSTN I/W.
>>>
>>> Regards,
>>> Mary.=20
>>>
>>> -----Original Message-----
>>> From: Ian Elz [mailto:ian.elz@ericsson.com]
>>> Sent: Wednesday, March 11, 2009 4:54 AM
>>> To: Barnes, Mary (RICH2:AR00); Audet, Francois (SC100:3055)
>>> Cc: sip@ietf.org
>>> Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt
>>>
>>> Mary, Francois,
>>>
>>> Section 4.1 of the draft allows for the addition of the Privacy=20
>>> header
>>>    =20
>>>      =20
>>  =20
>>    =20
>>> parameter with a value of "history" to be included with the=20
>>> hi-targeted-to-uri.
>>>
>>> Can this be extended to also allow the value "none".
>>>
>>> A node adding a H-I entry is allowed to specify the privacy value=20
>>> "history" either in the Privacy header or as a header parameter=20
>>> associated with the hi-targeted-to-uri. If the former action is=20
>>> taken
>>>      =20
>
>  =20
>>> by one node this results in privacy for all history info header=20
>>> entries even if this is not required. There is no way for privacy of

>>> an individual H-I entry to be set to none if no Privacy is required=20
>>> for the uri.
>>>
>>> This creates a specific problem when interworking with the PSTN=20
>>> where
>>>      =20
>
>  =20
>>> the privacy is associated with each E.164 number included in the=20
>>> protocol. If an INVITE is being mapped to ISUP and the H-I entries=20
>>> are
>>>    =20
>>>      =20
>>  =20
>>    =20
>>> being used to map to redirection numbers then a single Privacy=20
>>> header
>>>      =20
>
>  =20
>>> with the value "history" will result in all numbers being
restricted.
>>>
>>> If the Privacy header parameter could include the value "none" then=20
>>> this would explicitly indicate that the associated uri was allowed=20
>>> to
>>>      =20
>
>  =20
>>> be presented.
>>>
>>> The ability to explicitly allow presentation of specific H-I entries

>>> may also be useful in pure sip implementations.
>>>
>>> I don't believe that this change would create any backward=20
>>> compatibility issues. The existing implementations will continue as=20
>>> deployed but new implementations could explicitly set no privacy for

>>> a
>>>    =20
>>>      =20
>> specific uri.
>>  =20
>>    =20
>>> There are no security issues other than those already defined in the

>>> draft that an intervening node could modify an existing H-I entry.
>>> The
>>>    =20
>>>      =20
>>  =20
>>    =20
>>> Privacy header value of "none" should only be included by the node=20
>>> in
>>>      =20
>
>  =20
>>> the same manner as the existing inclusion of the "history" value.
>>>
>>> Ian Elz
>>>
>>>
>>> -----Original Message-----
>>> From: i-d-announce-bounces@ietf.org
>>> [mailto:i-d-announce-bounces@ietf.org] On Behalf Of=20
>>> Internet-Drafts@ietf.org
>>> Sent: 04 March 2009 15:15
>>> To: i-d-announce@ietf.org
>>> Subject: I-D Action:draft-barnes-sip-rfc4244bis-00.txt
>>>
>>> A New Internet-Draft is available from the on-line Internet-Drafts=20
>>> directories.
>>>
>>> 	Title           : An Extension to the Session Initiation
>>> Protocol (SIP) for Request History Information
>>> 	Author(s)       : M. Barnes, F. Audet
>>> 	Filename        : draft-barnes-sip-rfc4244bis-00.txt
>>> 	Pages           : 49
>>> 	Date            : 2009-03-04
>>>
>>> This document defines a standard mechanism for capturing the history

>>> information associated with a Session Initiation Protocol (SIP)
>>>    =20
>>>      =20
>> request.
>>  =20
>>    =20
>>> This capability enables many enhanced services by providing the=20
>>> information as to how and why a call arrives at a specific=20
>>> application
>>>    =20
>>>      =20
>>  =20
>>    =20
>>> or user.  This document defines a new optional SIP header,=20
>>> History-Info, for capturing the history information in requests.
>>>
>>> A URL for this Internet-Draft is:
>>> http://www.ietf.org/internet-drafts/draft-barnes-sip-rfc4244bis-00.t
>>> x
>>> t
>>>
>>> Internet-Drafts are also available by anonymous FTP at:
>>> ftp://ftp.ietf.org/internet-drafts/
>>>
>>> Below is the data which will enable a MIME compliant mail reader=20
>>> implementation to automatically retrieve the ASCII version of the=20
>>> Internet-Draft.
>>> _______________________________________________
>>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol Use=20
>>> sip-implementors@cs.columbia.edu for questions on current sip Use=20
>>> sipping@ietf.org for new developments on the application of sip
>>>  =20
>>>    =20
>>>      =20
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> sipping@ietf.org for new developments on the application of sip
>  =20

From mary.barnes@nortel.com  Wed Mar 11 18:31:06 2009
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Subject: Re: [Sip] Terminology (was RE: Fwd: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Responses below [MB].

-----Original Message-----
From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
Sent: Wednesday, March 11, 2009 6:42 PM
To: Barnes, Mary (RICH2:AR00)
Cc: Shida Schubert; sip@ietf.org
Subject: Re: Terminology (was RE: [Sip] Fwd: I-D
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt

I was talking about the concept.

The use cases only describe  cases where the target-uri (lets call that
current target from now) has been lost when an initial request for a
dialog or standalone request arrives at the UAS.
[MB] And I think this is where the terminology confusion starts - I
don't think of the "lost" target-uri as being the "current" target. In
my mind, the "current" target is reflected by the last hi-entry and the
request-uri in the incoming request at the UAS. If the entity that sent
the request was the entity that added the last hi-entry, then the uri in
that hi-entry is the same as the request-URI in the SIP request that
arrives at the UAS. I refer to the "lost" uri as the one that was
"retargeted" - that's the one the UAS wants to pull from the hi-entries
in the incoming request. That hi-entry was not the one that was just
added by the entity that built the request just received by the UAS.
That hi-entry is tagged with whatever name we are going to tag it with
BEFORE the request is forwarded (using the term forward per section 16.6
of RFC 3261).  That tag is added once the target set of potential
candidates for the new request uri are determined in section 16.5 of
3261 (with "target set" being a 3261 term), just before the request is
forwarded in section 16.6 to one of those targets.  A new hi-entry
(which will be the last hi-entry in the request received by the UAS) is
added in section 16.6 of 3261 as the request is forwarded.  At this
point in time, the lost information is in the previous hi-entry when the
outgoing request is sent. [/MB]=20

What has been lost is the current target of the request:
[MB] Right - at which point in my mind, it's no longer the current
target ;) Maybe we call it "lost". [/MB]


      Current target

   The current target of an initial request for a dialog or standalone
   request is the name or address to which the request is targeted, i.e.
   either the initial target inserted in the Request-URI by the UAC that
   originates the request, or when a retarget occurred, the target
   provided in that retarget operation.  Reroute and translation
   operations never change the current target.

[MB] I don't think this definition fits what you want - i.e., if there
is no retargeting, then none of the hi-entries are tagged - i.e., you
won't have your concept of current. The way it works is that if no
hi-entries are tagged, then you know that there was no retargeting, thus
you know the request-uri has not been lost. [/MB]

This defintion only makes sense when the following definitions are used:

Name:

   A name is a moniker for an entity which refers to it in a way which

   reveals nothing about where it is in a network.  In SIP, tel URI

   which doesn't represent the location of the entity is a name.  =20


Address:

   An address is an identifier for an entity which describes it by its

   location on the network.  In SIP, the SIP URI itself is a form of

   address because the host part of the URI, the only mandatory part of

   the URI besides the scheme itself, indicates the location of a SIP

   server that can be used to handle the request.

Route:

   Finally, a route is a   sequence of SIP entities (including the UA=20
itself!) which are

 traversed in order to forward a request to an address or name.

Retarget (other term might be needed, as this is highly confusing):

    A Request-URI rewrite operation that changes the target identity of
the request.

Reroute (other term might be needed):

    A Request-URI rewrite operation that does not change the current=20
target of the request, but     determines the route/next hop taken to=20
reach the target-identity.

/Hans Erik

Mary Barnes wrote:
> So, by "target-uri concept" are you referring to the solution in the=20
> current target-uri draft or the application usage as such in the
> target-uri document (which is what I was referring to)?  =20
>
> Yes, the problem is that SIP doesn't differentiate the various=20
> mechanisms by which a request-URI may be changed, which is what some=20
> of the updated text in 4244bis is trying to do and why we need precise

> terms. I think we all agree that.
>
> I honestly don't care what we call the tag, as long as we're clear=20
> about the functionality.
>
> Mary.=20
>
> -----Original Message-----
> From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
> Sent: Wednesday, March 11, 2009 4:40 PM
> To: Barnes, Mary (RICH2:AR00)
> Cc: Shida Schubert; sip@ietf.org
> Subject: Re: Terminology (was RE: [Sip] Fwd: I-D=20
> ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>
> The target-uri concept when defined properly is entirely application=20
> agnostic.
>
> When one uses History-Info or another header as a vehicle for this=20
> information then it still can be considered application agnostic as=20
> any application can use this information as it likes.
>
>  From SIP perspective all the Request-URI rewrites look the same, that

> is what brought us discussing this in the first place.
>
> /Hans Erik
>
> Mary Barnes wrote:
>  =20
>> And, I think the big issue with the terminology is that 4244bis is=20
>> application agnostic, so it is tagging what happens to the URIs=20
>> (why/how the specific URI was overwritten).  Whereas, specific=20
>> applications can derive information, such as what "is" the real
>> "target" for the request, from the History-Info header.   So, IMHO,=20
>> it's a matter of the target-uri document specifying that the last=20
>> entry tagged by whatever name we come up with "is" the real "target"
>> for the request.  I believe it's up to the applications to describe=20
>> how they make use of the History-Info and not for the History-Info to

>> describe how applications can use it.  History-Info purely reflects=20
>> what happened from a SIP Protocol perspective and should not imply=20
>> any
>>    =20
>
>  =20
>> application semantics. Indeed the intent of section 5 of 4244 was to=20
>> inform the applications how they should decribe their usage of the=20
>> header.
>> =20
>> Mary.
>> Note: I changed the subject to hopefully make this topic easier to=20
>> keep track of.
>>
>> ---------------------------------------------------------------------
>> -
>> --
>> *From:* Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
>> *Sent:* Wednesday, March 11, 2009 3:59 AM
>> *To:* Shida Schubert
>> *Cc:* sip@ietf.org List; Barnes, Mary (RICH2:AR00)
>> *Subject:* Re: [Sip] Fwd: I-D
>> ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>>
>> One of the problems that made the discussion quite tedious is the=20
>> complete inversed meaning of the terms "retarget" and "reroute"
>> between 4244bis and target-uri factions have.
>>
>> Conclusion was that the terminology need to be properly defined and=20
>> some attempt was made to start such activity.
>>
>> On 3. I have strong concerns when we start tagging the URI as to how=20
>> they come about "retarget"/"mapped conficuration" etc. It is better=20
>> to
>>    =20
>
>  =20
>> embellish them with a tag that just represents there meaning for=20
>> example "istarget" or "hop". For the following reasons:
>> 1. It is much more intuitive for the user of this information, its=20
>> meaning can basically be guessed.
>> 2. Such meaning also probably survives application in other problem=20
>> spaces that we had not foreseen when introducing the concept.
>>
>> The solution that is  described now in the target-URI delivery draft=20
>> is not yet complete, it does not solve the freephone use case for=20
>> example and in its current form it will deliver exactly the same=20
>> target-URI as that which would be delivered by the  P-Called-Party-ID

>> header. Contrary to what the draft says. So some work is needed
still.
>>
>> /Hans Erik van Elburg
>>
>>
>> On Wed, Mar 11, 2009 at 7:13 AM, Shida Schubert <shida@ntt-at.com=20
>> <mailto:shida@ntt-at.com>> wrote:
>>
>>
>>     We have submitted the updated target-uri draft based on the
>>    =20
> comments
>  =20
>>     submitted to the list and comments received at IETF73.=20
>>
>>     I have taken over as editor as Jonathan didn't have the cycles to

>>     update the draft, with Francois, Christer and Hans Erick as
>>     additional=20
>>     co-authors and great deal of help from Mary. =20
>>     =20
>>     The following summarizes the changes made to the target-uri
>>    =20
> document
>  =20
>>     1. Added use-case for toll-free number back
>>     2. Added definition of "retarget" operation.=20
>>     3. Removed a reference to URN
>>     4. Added a text discussing the difference to P-Called-Party-Id
>>     5. Changed parameter name from "target" to "istarget"
>>     =20
>>     Note, that the target-uri document still contains the normative
>>     text for the
>>     History-Info header.
>>     =20
>>     In addition, Mary (with Francois as co-author) has submitted a
>>     rfc4244bis, with the following changes:
>>     1. Incorporated the normative aspects of the target-uri document
>>     into the existing normative text in RFC 4244 - the functionality
>>    =20
> is
>  =20
>>     virtually identical (as is some of the text) as the HI based
>>    =20
> solution
>  =20
>>     described in the target-uri document.  It's important that the
>>     solution
>>     be integrated into RFC 4244 as it MUST work and be based on the
>>     normative
>>     aspects of RFC 4244.
>>     2.  Added the use cases from target-uri the the summary
>>     in the overview of rfc4244bis.
>>     3. Added an additional requirement to capture the "target-uri"
>>     information.
>>     4. Fixed an error in the RFC 4244 ABNF and added "retarget"
>>    =20
> parameter.
>  =20
>>     5. Added a more simplified example.
>>     =20
>>     =20
>>     We had some very long offline exchanges as to the best way=20
>> forward
>>    =20
> and
>  =20
>>     remaining work for both documents.
>>     =20
>>     Some of the issues identified are:
>>
>>     ::Issues::
>>        1. Should we remove the normative text from target-uri and
>>    =20
> progress
>  =20
>>            4244bis along with the target-uri document to meet the
>>     chartered
>>           SIP WG milestone?=20
>>     =20
>>     =20
>>        2. Name of the parameter.=20
>>           At the last meeting, parameter "target" was said
>>    =20
> inappropriate
>  =20
>>           because voicemail-uri spec already defines a parameter
>>    =20
> called
>  =20
>>           "target" which also can be found in hi-entry, thus
>>    =20
> potentially
>  =20
>>           causing confusion.
>>         =20
>>           Currently the target-uri draft uses "istarget" and 4244bis
>>    =20
> uses
>  =20
>>           "retarget"  but we could never come to=20
>>           a consensus on what name is appropriate.  Other suggestions
>>    =20
> have
>  =20
>>           included the following:
>>           "target-identity" (someone didn't like that "identity" is
>>     also a SIP header)
>>           "reg-uri"  (can be paired with "mapped-uri" for item 3
>>    =20
> below)
>  =20
>>           "aor"
>>           "jibberish"=20
>>           etc.
>>     =20
>>           One reason this is so difficult relates to the problem
>>     statement in target-uri in that
>>           RFC 3261 doesn't differentiate the mechanism by which the
>>    =20
> new
>  =20
>>           (target) Request-URI is selected.  Another issue is that
>>     some of the terminology in
>>           RFC 3261 is overloaded - e.g., "forwarding" refers both to
a
>>     Proxy
>>           which does not have responsibility for the domain of the
>>     request-URI
>>           in the incoming request, thus the proxy just "forwards" the
>>     request to
>>           the next hop AND "forwarding" is used to describe the
>>     process whereby
>>           the outgoing request is built and "forwarded" to the next
>>     hop at which
>>           point the proxy does not know how the new request-uri was
>>     selected.
>>           RFC 4244 has attempted to clarify the terms and attempts to
>>     use "forward"=20
>>           in the context of the former situation and "retarget" for
>>     the case whereby
>>           a proxy is responsible for the domain and thus can use a
>>     number of=20
>>           mechanism to select the new target for the request - e.g.,
a
>>     REGISTRAR,=20
>>           configured data, etc. =20
>>
>>      3.  Related to the last point in item 2 above, it has been
>>     proposed that
>>           we differentiate the hi-entries even more by defining
>>     separate parameters
>>          for registered and configured/mapped contacts.=20
>>          Currently when the R-URI is translated to a URI which is
>>     either derived=20
>>          from location service lookup(registered by UA) or from
>>    =20
> mapping
>  =20
>>          table, there is no differentiation as to how the URI was
>>     derived once it is
>>          added to the list of potential targets.
>>         =20
>>          The general consensus of the authors of the two documents
was
>>     that it may=20
>>          be useful for some services to have the hi-entries tagged
>>     with the
>>          more specific information.=20
>>     =20
>>          And, of course, this gets us into another naming contest. In
>>     the end, the naming
>>          is not so important as long as the term isn't too overloaded
>>     and it is defined
>>          precisely in the document(s).
>>     =20
>>     We would appreciate WG feedback on these issues and any other
>>     comments on
>>     the two documents prior to IETF-74.
>>     =20
>>     Regards,
>>     Shida and Mary.
>>
>>
>>
>>     Begin forwarded message:
>>    =20
>>>     *From: *Internet-Drafts@ietf.org
>>>      =20
> <mailto:Internet-Drafts@ietf.org>
>  =20
>>>     *Date: *March 10, 2009 2:30:01 AM JST
>>>     *To: *i-d-announce@ietf.org <mailto:i-d-announce@ietf.org>
>>>     *Subject: **I-D
>>>     ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt *
>>>     *Reply-To: *internet-drafts@ietf.org
>>>     <mailto:internet-drafts@ietf.org>
>>>
>>>     A New Internet-Draft is available from the on-line
>>>      =20
> Internet-Drafts
>  =20
>>>     directories.
>>>
>>>
>>>     Title
>>>     : Delivery of Request-URI Targets to User Agents
>>>     Author(s)
>>>     : J. Rosenberg, H. van Elburg, C. Holmberg, F. Audet, S.
Schubert
>>>     Filename : draft-rosenberg-sip-target-uri-delivery-01.txt
>>>     Pages
>>>     : 16
>>>     Date
>>>     : 2009-3-9
>>>
>>>     When a Session Initiation Protocol (SIP) proxy receives a
request
>>>       targeted at a URI identifying a user or resource it is
>>>      =20
> responsible
>  =20
>>>       for, the proxy translates the URI to a registered or
configured
>>>       contact URI of an agent representing that user or resource. =20
>>> In
>>>      =20
> the
>  =20
>>>       process, the original URI is removed from the request.
>>>      Numerous use
>>>       cases have arisen which require this information to be
>>>      =20
> delivered to
>  =20
>>>       the user agent.  This document describes these use cases and
>>>     defines
>>>       an extension to the History-Info header field which allows it
>>>      =20
> to be
>  =20
>>>       used to support those cases.
>>>
>>>     A URL for this Internet-Draft is:
>>>    =20
>>> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri-d
>>> e
>>> livery-01.txt
>>>
>>>     Internet-Drafts are also available by anonymous FTP at:
>>>     ftp://ftp.ietf.org/internet-drafts/
>>>
>>>     Below is the data which will enable a MIME compliant mail reader
>>>     implementation to automatically retrieve the ASCII version of
the
>>>     Internet-Draft.
>>>      =20
>>     Content-Type: text/plain<BR>Content-ID:
>>     &lt;2009-3-9102756.I-D@ietf.org
>>     <mailto:2009-3-9102756.I-D@ietf.org>&gt;<BR><BR>
>>    =20
>>>     _______________________________________________
>>>     I-D-Announce mailing list
>>>     I-D-Announce@ietf.org <mailto:I-D-Announce@ietf.org>
>>>     https://www.ietf.org/mailman/listinfo/i-d-announce
>>>     Internet-Draft directories: http://www.ietf.org/shadow.html
>>>     or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>>>      =20
>>     _______________________________________________
>>     Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>     This list is for NEW development of the core SIP Protocol
>>     Use sip-implementors@cs.columbia.edu
>>     <mailto:sip-implementors@cs.columbia.edu> for questions on=20
>> current
>>    =20
> sip
>  =20
>>     Use sipping@ietf.org <mailto:sipping@ietf.org> for new
>>     developments on the application of sip
>>
>>
>>    =20

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From christer.holmberg@ericsson.com  Thu Mar 12 00:30:55 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: <gao.yang2@zte.com.cn>, "SIP" <sip@ietf.org>, "SIPPING" <sipping@ietf.org>
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Subject: Re: [Sip] One chart  //"UPDATE during Re-INVITE" discussion
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In my opinion 487 should trigger whatever actions/rollback that any other non-200 response sent by the UAS would.
 
Regards,
 
Christer


________________________________

	From: gao.yang2@zte.com.cn [mailto:gao.yang2@zte.com.cn] 
	Sent: 12. maaliskuuta 2009 5:47
	To: SIP; SIPPING
	Cc: Christer Holmberg; Gonzalo Camarillo
	Subject: One chart //"UPDATE during Re-INVITE" discussion
	
	

	      UAC                   UAS 
	       | session established | 
	       |<===================>| 
	       |                     | 
	       | F1  re-INVITE (SDP) | 
	       |-------------------->| 
	       | F2 1xx-rel (SDP)    | 
	       |<--------------------| 
	       | F3   UPDATE         | 
	       |<--------------------| Target refreshing 
	       | F4 2xx UPT          | 
	       |-------------------->| 
	       |                     | 
	       | F5/6 Cancel/200OK   | 
	       |-------------------->| 
	       | F7    487 INV       | 
	       |<--------------------| 
	       | F7     ACK          | 
	       |-------------------->| 
	       | F8  re-INVITE (SDP) | 
	       |-------------------->| Using which dialog state 
	
	F3/F4 is about target refreshing. 
	
	And if UAC sends Cancel$B!$(J UAS sends 487. The first Re-INVITE is discarded. 
	
	But if UAC sends another Re-INVITE, which dialog state should it use? 
	
	I think is the one refreshed by F3. That's the merit of regarding UPDATE/200OK's atomicity. 
	
	
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<DIV dir=ltr align=left><SPAN class=305232907-12032009><FONT face=Arial 
color=#0000ff size=2>In my opinion 487&nbsp;should trigger whatever 
actions/rollback that&nbsp;any other non-200 response sent by the UAS 
would.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=305232907-12032009><FONT face=Arial 
color=#0000ff size=2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=ltr align=left><SPAN class=305232907-12032009><FONT face=Arial 
color=#0000ff size=2>Regards,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=305232907-12032009><FONT face=Arial 
color=#0000ff size=2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=ltr align=left><SPAN class=305232907-12032009><FONT face=Arial 
color=#0000ff size=2>Christer</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=ltr 
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
  <DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
  <HR tabIndex=-1>
  <FONT face=Tahoma size=2><B>From:</B> gao.yang2@zte.com.cn 
  [mailto:gao.yang2@zte.com.cn] <BR><B>Sent:</B> 12. maaliskuuta 2009 
  5:47<BR><B>To:</B> SIP; SIPPING<BR><B>Cc:</B> Christer Holmberg; Gonzalo 
  Camarillo<BR><B>Subject:</B> One chart //"UPDATE during Re-INVITE" 
  discussion<BR></FONT><BR></DIV>
  <DIV></DIV><BR><FONT face=sans-serif size=2>&nbsp; &nbsp; &nbsp; UAC &nbsp; 
  &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; UAS</FONT> <BR><FONT 
  face=sans-serif size=2>&nbsp; &nbsp; &nbsp; &nbsp;| session established 
  |</FONT> <BR><FONT face=sans-serif size=2>&nbsp; &nbsp; &nbsp; 
  &nbsp;|&lt;===================&gt;|</FONT> <BR><FONT face=sans-serif 
  size=2>&nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
  &nbsp; &nbsp; &nbsp; &nbsp; |</FONT> <BR><FONT face=sans-serif size=2>&nbsp; 
  &nbsp; &nbsp; &nbsp;| F1 &nbsp;re-INVITE (SDP) |</FONT> <BR><FONT 
  face=sans-serif size=2>&nbsp; &nbsp; &nbsp; 
  &nbsp;|--------------------&gt;|</FONT> <BR><FONT face=sans-serif 
  size=2>&nbsp; &nbsp; &nbsp; &nbsp;| F2 1xx-rel (SDP) &nbsp; &nbsp;|</FONT> 
  <BR><FONT face=sans-serif size=2>&nbsp; &nbsp; &nbsp; 
  &nbsp;|&lt;--------------------|</FONT> <BR><FONT face=sans-serif 
  size=2>&nbsp; &nbsp; &nbsp; &nbsp;| F3 &nbsp; UPDATE &nbsp; &nbsp; &nbsp; 
  &nbsp; |</FONT> <BR><FONT face=sans-serif size=2>&nbsp; &nbsp; &nbsp; 
  &nbsp;|&lt;--------------------| Target refreshing</FONT> <BR><FONT 
  face=sans-serif size=2>&nbsp; &nbsp; &nbsp; &nbsp;| F4 2xx UPT &nbsp; &nbsp; 
  &nbsp; &nbsp; &nbsp;|</FONT> <BR><FONT face=sans-serif size=2>&nbsp; &nbsp; 
  &nbsp; &nbsp;|--------------------&gt;|</FONT> <BR><FONT face=sans-serif 
  size=2>&nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
  &nbsp; &nbsp; &nbsp; &nbsp; |</FONT> <BR><FONT face=sans-serif size=2>&nbsp; 
  &nbsp; &nbsp; &nbsp;| F5/6 Cancel/200OK &nbsp; |</FONT> <BR><FONT 
  face=sans-serif size=2>&nbsp; &nbsp; &nbsp; 
  &nbsp;|--------------------&gt;|</FONT> <BR><FONT face=sans-serif 
  size=2>&nbsp; &nbsp; &nbsp; &nbsp;| F7 &nbsp; &nbsp;487 INV &nbsp; &nbsp; 
  &nbsp; |</FONT> <BR><FONT face=sans-serif size=2>&nbsp; &nbsp; &nbsp; 
  &nbsp;|&lt;--------------------|</FONT> <BR><FONT face=sans-serif 
  size=2>&nbsp; &nbsp; &nbsp; &nbsp;| F7 &nbsp; &nbsp; ACK &nbsp; &nbsp; &nbsp; 
  &nbsp; &nbsp;|</FONT> <BR><FONT face=sans-serif size=2>&nbsp; &nbsp; &nbsp; 
  &nbsp;|--------------------&gt;|</FONT> <BR><FONT face=sans-serif 
  size=2>&nbsp; &nbsp; &nbsp; &nbsp;| F8 &nbsp;re-INVITE (SDP) |</FONT> 
  <BR><FONT face=sans-serif size=2>&nbsp; &nbsp; &nbsp; 
  &nbsp;|--------------------&gt;| Using which dialog state</FONT> <BR><BR><FONT 
  face=sans-serif size=2>F3/F4 is about target refreshing.</FONT> <BR><BR><FONT 
  face=sans-serif size=2>And if UAC sends Cancel$B!$(B UAS sends 487. The first 
  Re-INVITE is discarded.</FONT> <BR><BR><FONT face=sans-serif size=2>But if UAC 
  sends another Re-INVITE, which dialog state should it use?</FONT> 
  <BR><BR><FONT face=sans-serif size=2>I think is the one refreshed by F3. 
  That's the merit of regarding UPDATE/200OK's atomicity.</FONT> <BR><BR><PRE>--------------------------------------------------------
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Date: Thu, 12 Mar 2009 09:34:34 +0100
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From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: Mary Barnes <mary.barnes@nortel.com>
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Subject: Re: [Sip] Terminology (was RE: Fwd: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Yes, because you are using the 3261 use of target and the 4244bis introduced
definition of retarget. I thought it was clear that we need other words as
those definitions don't match the target-uri drafts use of the terms. Also
they do not suffice to provide a solution for the use cases in the
target-uri draft.

The 3261 text you refer to is exactly about the case where the home proxy
overwrites the Request-URI with a new target. This target is teh registered
contact address. And hence this would be what target-uri calls a hop or a
route. This case and this is where it gets confusing is not a "retarget" in
the target-uri draft use of the term.

The target-uri draft states:

   "To avoid confusion, we
   refer to a SIP URI that is an address for a user or resource as a
   "target" and a SIP URI that is a hop for reaching that user as a
   "hop".


Apparently that does not suffice to avoid confusion.

As for the tagging, speaking about the solution before agreeing on the
terminology and the problem it should solve is meaningless.

/Hans Erik van Elburg


On Thu, Mar 12, 2009 at 2:33 AM, Mary Barnes <mary.barnes@nortel.com> wrote:

> Responses below [MB].
>
> -----Original Message-----
> From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
> Sent: Wednesday, March 11, 2009 6:42 PM
> To: Barnes, Mary (RICH2:AR00)
> Cc: Shida Schubert; sip@ietf.org
> Subject: Re: Terminology (was RE: [Sip] Fwd: I-D
> ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>
> I was talking about the concept.
>
> The use cases only describe  cases where the target-uri (lets call that
> current target from now) has been lost when an initial request for a
> dialog or standalone request arrives at the UAS.
> [MB] And I think this is where the terminology confusion starts - I
> don't think of the "lost" target-uri as being the "current" target. In
> my mind, the "current" target is reflected by the last hi-entry and the
> request-uri in the incoming request at the UAS. If the entity that sent
> the request was the entity that added the last hi-entry, then the uri in
> that hi-entry is the same as the request-URI in the SIP request that
> arrives at the UAS. I refer to the "lost" uri as the one that was
> "retargeted" - that's the one the UAS wants to pull from the hi-entries
> in the incoming request. That hi-entry was not the one that was just
> added by the entity that built the request just received by the UAS.
> That hi-entry is tagged with whatever name we are going to tag it with
> BEFORE the request is forwarded (using the term forward per section 16.6
> of RFC 3261).  That tag is added once the target set of potential
> candidates for the new request uri are determined in section 16.5 of
> 3261 (with "target set" being a 3261 term), just before the request is
> forwarded in section 16.6 to one of those targets.  A new hi-entry
> (which will be the last hi-entry in the request received by the UAS) is
> added in section 16.6 of 3261 as the request is forwarded.  At this
> point in time, the lost information is in the previous hi-entry when the
> outgoing request is sent. [/MB]
>
> What has been lost is the current target of the request:
> [MB] Right - at which point in my mind, it's no longer the current
> target ;) Maybe we call it "lost". [/MB]
>
>
>      Current target
>
>   The current target of an initial request for a dialog or standalone
>   request is the name or address to which the request is targeted, i.e.
>   either the initial target inserted in the Request-URI by the UAC that
>   originates the request, or when a retarget occurred, the target
>   provided in that retarget operation.  Reroute and translation
>   operations never change the current target.
>
> [MB] I don't think this definition fits what you want - i.e., if there
> is no retargeting, then none of the hi-entries are tagged - i.e., you
> won't have your concept of current. The way it works is that if no
> hi-entries are tagged, then you know that there was no retargeting, thus
> you know the request-uri has not been lost. [/MB]
>
> This defintion only makes sense when the following definitions are used:
>
> Name:
>
>   A name is a moniker for an entity which refers to it in a way which
>
>   reveals nothing about where it is in a network.  In SIP, tel URI
>
>   which doesn't represent the location of the entity is a name.
>
>
> Address:
>
>   An address is an identifier for an entity which describes it by its
>
>   location on the network.  In SIP, the SIP URI itself is a form of
>
>   address because the host part of the URI, the only mandatory part of
>
>   the URI besides the scheme itself, indicates the location of a SIP
>
>   server that can be used to handle the request.
>
> Route:
>
>   Finally, a route is a   sequence of SIP entities (including the UA
> itself!) which are
>
>  traversed in order to forward a request to an address or name.
>
> Retarget (other term might be needed, as this is highly confusing):
>
>    A Request-URI rewrite operation that changes the target identity of
> the request.
>
> Reroute (other term might be needed):
>
>    A Request-URI rewrite operation that does not change the current
> target of the request, but     determines the route/next hop taken to
> reach the target-identity.
>
> /Hans Erik
>
> Mary Barnes wrote:
> > So, by "target-uri concept" are you referring to the solution in the
> > current target-uri draft or the application usage as such in the
> > target-uri document (which is what I was referring to)?
> >
> > Yes, the problem is that SIP doesn't differentiate the various
> > mechanisms by which a request-URI may be changed, which is what some
> > of the updated text in 4244bis is trying to do and why we need precise
>
> > terms. I think we all agree that.
> >
> > I honestly don't care what we call the tag, as long as we're clear
> > about the functionality.
> >
> > Mary.
> >
> > -----Original Message-----
> > From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
> > Sent: Wednesday, March 11, 2009 4:40 PM
> > To: Barnes, Mary (RICH2:AR00)
> > Cc: Shida Schubert; sip@ietf.org
> > Subject: Re: Terminology (was RE: [Sip] Fwd: I-D
> > ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
> >
> > The target-uri concept when defined properly is entirely application
> > agnostic.
> >
> > When one uses History-Info or another header as a vehicle for this
> > information then it still can be considered application agnostic as
> > any application can use this information as it likes.
> >
> >  From SIP perspective all the Request-URI rewrites look the same, that
>
> > is what brought us discussing this in the first place.
> >
> > /Hans Erik
> >
> > Mary Barnes wrote:
> >
> >> And, I think the big issue with the terminology is that 4244bis is
> >> application agnostic, so it is tagging what happens to the URIs
> >> (why/how the specific URI was overwritten).  Whereas, specific
> >> applications can derive information, such as what "is" the real
> >> "target" for the request, from the History-Info header.   So, IMHO,
> >> it's a matter of the target-uri document specifying that the last
> >> entry tagged by whatever name we come up with "is" the real "target"
> >> for the request.  I believe it's up to the applications to describe
> >> how they make use of the History-Info and not for the History-Info to
>
> >> describe how applications can use it.  History-Info purely reflects
> >> what happened from a SIP Protocol perspective and should not imply
> >> any
> >>
> >
> >
> >> application semantics. Indeed the intent of section 5 of 4244 was to
> >> inform the applications how they should decribe their usage of the
> >> header.
> >>
> >> Mary.
> >> Note: I changed the subject to hopefully make this topic easier to
> >> keep track of.
> >>
> >> ---------------------------------------------------------------------
> >> -
> >> --
> >> *From:* Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
> >> *Sent:* Wednesday, March 11, 2009 3:59 AM
> >> *To:* Shida Schubert
> >> *Cc:* sip@ietf.org List; Barnes, Mary (RICH2:AR00)
> >> *Subject:* Re: [Sip] Fwd: I-D
> >> ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
> >>
> >> One of the problems that made the discussion quite tedious is the
> >> complete inversed meaning of the terms "retarget" and "reroute"
> >> between 4244bis and target-uri factions have.
> >>
> >> Conclusion was that the terminology need to be properly defined and
> >> some attempt was made to start such activity.
> >>
> >> On 3. I have strong concerns when we start tagging the URI as to how
> >> they come about "retarget"/"mapped conficuration" etc. It is better
> >> to
> >>
> >
> >
> >> embellish them with a tag that just represents there meaning for
> >> example "istarget" or "hop". For the following reasons:
> >> 1. It is much more intuitive for the user of this information, its
> >> meaning can basically be guessed.
> >> 2. Such meaning also probably survives application in other problem
> >> spaces that we had not foreseen when introducing the concept.
> >>
> >> The solution that is  described now in the target-URI delivery draft
> >> is not yet complete, it does not solve the freephone use case for
> >> example and in its current form it will deliver exactly the same
> >> target-URI as that which would be delivered by the  P-Called-Party-ID
>
> >> header. Contrary to what the draft says. So some work is needed
> still.
> >>
> >> /Hans Erik van Elburg
> >>
> >>
> >> On Wed, Mar 11, 2009 at 7:13 AM, Shida Schubert <shida@ntt-at.com
> >> <mailto:shida@ntt-at.com>> wrote:
> >>
> >>
> >>     We have submitted the updated target-uri draft based on the
> >>
> > comments
> >
> >>     submitted to the list and comments received at IETF73.
> >>
> >>     I have taken over as editor as Jonathan didn't have the cycles to
>
> >>     update the draft, with Francois, Christer and Hans Erick as
> >>     additional
> >>     co-authors and great deal of help from Mary.
> >>
> >>     The following summarizes the changes made to the target-uri
> >>
> > document
> >
> >>     1. Added use-case for toll-free number back
> >>     2. Added definition of "retarget" operation.
> >>     3. Removed a reference to URN
> >>     4. Added a text discussing the difference to P-Called-Party-Id
> >>     5. Changed parameter name from "target" to "istarget"
> >>
> >>     Note, that the target-uri document still contains the normative
> >>     text for the
> >>     History-Info header.
> >>
> >>     In addition, Mary (with Francois as co-author) has submitted a
> >>     rfc4244bis, with the following changes:
> >>     1. Incorporated the normative aspects of the target-uri document
> >>     into the existing normative text in RFC 4244 - the functionality
> >>
> > is
> >
> >>     virtually identical (as is some of the text) as the HI based
> >>
> > solution
> >
> >>     described in the target-uri document.  It's important that the
> >>     solution
> >>     be integrated into RFC 4244 as it MUST work and be based on the
> >>     normative
> >>     aspects of RFC 4244.
> >>     2.  Added the use cases from target-uri the the summary
> >>     in the overview of rfc4244bis.
> >>     3. Added an additional requirement to capture the "target-uri"
> >>     information.
> >>     4. Fixed an error in the RFC 4244 ABNF and added "retarget"
> >>
> > parameter.
> >
> >>     5. Added a more simplified example.
> >>
> >>
> >>     We had some very long offline exchanges as to the best way
> >> forward
> >>
> > and
> >
> >>     remaining work for both documents.
> >>
> >>     Some of the issues identified are:
> >>
> >>     ::Issues::
> >>        1. Should we remove the normative text from target-uri and
> >>
> > progress
> >
> >>            4244bis along with the target-uri document to meet the
> >>     chartered
> >>           SIP WG milestone?
> >>
> >>
> >>        2. Name of the parameter.
> >>           At the last meeting, parameter "target" was said
> >>
> > inappropriate
> >
> >>           because voicemail-uri spec already defines a parameter
> >>
> > called
> >
> >>           "target" which also can be found in hi-entry, thus
> >>
> > potentially
> >
> >>           causing confusion.
> >>
> >>           Currently the target-uri draft uses "istarget" and 4244bis
> >>
> > uses
> >
> >>           "retarget"  but we could never come to
> >>           a consensus on what name is appropriate.  Other suggestions
> >>
> > have
> >
> >>           included the following:
> >>           "target-identity" (someone didn't like that "identity" is
> >>     also a SIP header)
> >>           "reg-uri"  (can be paired with "mapped-uri" for item 3
> >>
> > below)
> >
> >>           "aor"
> >>           "jibberish"
> >>           etc.
> >>
> >>           One reason this is so difficult relates to the problem
> >>     statement in target-uri in that
> >>           RFC 3261 doesn't differentiate the mechanism by which the
> >>
> > new
> >
> >>           (target) Request-URI is selected.  Another issue is that
> >>     some of the terminology in
> >>           RFC 3261 is overloaded - e.g., "forwarding" refers both to
> a
> >>     Proxy
> >>           which does not have responsibility for the domain of the
> >>     request-URI
> >>           in the incoming request, thus the proxy just "forwards" the
> >>     request to
> >>           the next hop AND "forwarding" is used to describe the
> >>     process whereby
> >>           the outgoing request is built and "forwarded" to the next
> >>     hop at which
> >>           point the proxy does not know how the new request-uri was
> >>     selected.
> >>           RFC 4244 has attempted to clarify the terms and attempts to
> >>     use "forward"
> >>           in the context of the former situation and "retarget" for
> >>     the case whereby
> >>           a proxy is responsible for the domain and thus can use a
> >>     number of
> >>           mechanism to select the new target for the request - e.g.,
> a
> >>     REGISTRAR,
> >>           configured data, etc.
> >>
> >>      3.  Related to the last point in item 2 above, it has been
> >>     proposed that
> >>           we differentiate the hi-entries even more by defining
> >>     separate parameters
> >>          for registered and configured/mapped contacts.
> >>          Currently when the R-URI is translated to a URI which is
> >>     either derived
> >>          from location service lookup(registered by UA) or from
> >>
> > mapping
> >
> >>          table, there is no differentiation as to how the URI was
> >>     derived once it is
> >>          added to the list of potential targets.
> >>
> >>          The general consensus of the authors of the two documents
> was
> >>     that it may
> >>          be useful for some services to have the hi-entries tagged
> >>     with the
> >>          more specific information.
> >>
> >>          And, of course, this gets us into another naming contest. In
> >>     the end, the naming
> >>          is not so important as long as the term isn't too overloaded
> >>     and it is defined
> >>          precisely in the document(s).
> >>
> >>     We would appreciate WG feedback on these issues and any other
> >>     comments on
> >>     the two documents prior to IETF-74.
> >>
> >>     Regards,
> >>     Shida and Mary.
> >>
> >>
> >>
> >>     Begin forwarded message:
> >>
> >>>     *From: *Internet-Drafts@ietf.org
> >>>
> > <mailto:Internet-Drafts@ietf.org>
> >
> >>>     *Date: *March 10, 2009 2:30:01 AM JST
> >>>     *To: *i-d-announce@ietf.org <mailto:i-d-announce@ietf.org>
> >>>     *Subject: **I-D
> >>>     ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt *
> >>>     *Reply-To: *internet-drafts@ietf.org
> >>>     <mailto:internet-drafts@ietf.org>
> >>>
> >>>     A New Internet-Draft is available from the on-line
> >>>
> > Internet-Drafts
> >
> >>>     directories.
> >>>
> >>>
> >>>     Title
> >>>     : Delivery of Request-URI Targets to User Agents
> >>>     Author(s)
> >>>     : J. Rosenberg, H. van Elburg, C. Holmberg, F. Audet, S.
> Schubert
> >>>     Filename : draft-rosenberg-sip-target-uri-delivery-01.txt
> >>>     Pages
> >>>     : 16
> >>>     Date
> >>>     : 2009-3-9
> >>>
> >>>     When a Session Initiation Protocol (SIP) proxy receives a
> request
> >>>       targeted at a URI identifying a user or resource it is
> >>>
> > responsible
> >
> >>>       for, the proxy translates the URI to a registered or
> configured
> >>>       contact URI of an agent representing that user or resource.
> >>> In
> >>>
> > the
> >
> >>>       process, the original URI is removed from the request.
> >>>      Numerous use
> >>>       cases have arisen which require this information to be
> >>>
> > delivered to
> >
> >>>       the user agent.  This document describes these use cases and
> >>>     defines
> >>>       an extension to the History-Info header field which allows it
> >>>
> > to be
> >
> >>>       used to support those cases.
> >>>
> >>>     A URL for this Internet-Draft is:
> >>>
> >>> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri-d
> >>> e
> >>> livery-01.txt
> >>>
> >>>     Internet-Drafts are also available by anonymous FTP at:
> >>>     ftp://ftp.ietf.org/internet-drafts/
> >>>
> >>>     Below is the data which will enable a MIME compliant mail reader
> >>>     implementation to automatically retrieve the ASCII version of
> the
> >>>     Internet-Draft.
> >>>
> >>     Content-Type: text/plain<BR>Content-ID:
> >>     &lt;2009-3-9102756.I-D@ietf.org <lt%3B2009-3-9102756.I-D@ietf.org>
> >>     <mailto:2009-3-9102756.I-D@ietf.org>&gt;<BR><BR>
> >>
> >>>     _______________________________________________
> >>>     I-D-Announce mailing list
> >>>     I-D-Announce@ietf.org <mailto:I-D-Announce@ietf.org>
> >>>     https://www.ietf.org/mailman/listinfo/i-d-announce
> >>>     Internet-Draft directories: http://www.ietf.org/shadow.html
> >>>     or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >>>
> >>     _______________________________________________
> >>     Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> >>     This list is for NEW development of the core SIP Protocol
> >>     Use sip-implementors@cs.columbia.edu
> >>     <mailto:sip-implementors@cs.columbia.edu> for questions on
> >> current
> >>
> > sip
> >
> >>     Use sipping@ietf.org <mailto:sipping@ietf.org> for new
> >>     developments on the application of sip
> >>
> >>
> >>
>

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Content-Type: text/html; charset=ISO-8859-1
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Yes, because you are using the 3261 use of target and the 4244bis introduce=
d definition of retarget. I thought it was clear that we need other words a=
s those definitions don&#39;t match the target-uri drafts use of the terms.=
 Also they do not suffice to provide a solution for the use cases in the ta=
rget-uri draft.<br>
<br>The 3261 text you refer to is exactly about the case where the home pro=
xy overwrites the Request-URI with a new target. This target is teh registe=
red contact address. And hence this would be what target-uri calls a hop or=
 a route. This case and this is where it gets confusing is not a &quot;reta=
rget&quot; in the target-uri draft use of the term.<br>
<br>The target-uri draft states:<br><pre class=3D"newpage">   &quot;To avoi=
d confusion, we<br>   refer to a SIP URI that is an address for a user or r=
esource as a<br>   &quot;target&quot; and a SIP URI that is a hop for reach=
ing that user as a<br>
   &quot;hop&quot;.<br></pre><br>Apparently that does not suffice to avoid =
confusion.<br><br>As for the tagging, speaking about the solution before ag=
reeing on the terminology and the problem it should solve is meaningless.<b=
r>
<br clear=3D"all">/Hans Erik van Elburg<br>
<br><br><div class=3D"gmail_quote">On Thu, Mar 12, 2009 at 2:33 AM, Mary Ba=
rnes <span dir=3D"ltr">&lt;<a href=3D"mailto:mary.barnes@nortel.com">mary.b=
arnes@nortel.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote"=
 style=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.=
8ex; padding-left: 1ex;">
Responses below [MB].<br>
<div class=3D"im"><br>
-----Original Message-----<br>
From: Hans Erik van Elburg [mailto:<a href=3D"mailto:ietf.hanserik@gmail.co=
m">ietf.hanserik@gmail.com</a>]<br>
</div><div class=3D"im">Sent: Wednesday, March 11, 2009 6:42 PM<br>
To: Barnes, Mary (RICH2:AR00)<br>
Cc: Shida Schubert; <a href=3D"mailto:sip@ietf.org">sip@ietf.org</a><br>
Subject: Re: Terminology (was RE: [Sip] Fwd: I-D<br>
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br>
<br>
</div><div class=3D"im">I was talking about the concept.<br>
<br>
The use cases only describe =A0cases where the target-uri (lets call that<b=
r>
current target from now) has been lost when an initial request for a<br>
dialog or standalone request arrives at the UAS.<br>
</div>[MB] And I think this is where the terminology confusion starts - I<b=
r>
don&#39;t think of the &quot;lost&quot; target-uri as being the &quot;curre=
nt&quot; target. In<br>
my mind, the &quot;current&quot; target is reflected by the last hi-entry a=
nd the<br>
request-uri in the incoming request at the UAS. If the entity that sent<br>
the request was the entity that added the last hi-entry, then the uri in<br=
>
that hi-entry is the same as the request-URI in the SIP request that<br>
arrives at the UAS. I refer to the &quot;lost&quot; uri as the one that was=
<br>
&quot;retargeted&quot; - that&#39;s the one the UAS wants to pull from the =
hi-entries<br>
in the incoming request. That hi-entry was not the one that was just<br>
added by the entity that built the request just received by the UAS.<br>
That hi-entry is tagged with whatever name we are going to tag it with<br>
BEFORE the request is forwarded (using the term forward per section 16.6<br=
>
of RFC 3261). =A0That tag is added once the target set of potential<br>
candidates for the new request uri are determined in section 16.5 of<br>
3261 (with &quot;target set&quot; being a 3261 term), just before the reque=
st is<br>
forwarded in section 16.6 to one of those targets. =A0A new hi-entry<br>
(which will be the last hi-entry in the request received by the UAS) is<br>
added in section 16.6 of 3261 as the request is forwarded. =A0At this<br>
point in time, the lost information is in the previous hi-entry when the<br=
>
outgoing request is sent. [/MB]<br>
<div class=3D"im"><br>
What has been lost is the current target of the request:<br>
</div>[MB] Right - at which point in my mind, it&#39;s no longer the curren=
t<br>
target ;) Maybe we call it &quot;lost&quot;. [/MB]<br>
<div class=3D"im"><br>
<br>
 =A0 =A0 =A0Current target<br>
<br>
 =A0 The current target of an initial request for a dialog or standalone<br=
>
 =A0 request is the name or address to which the request is targeted, i.e.<=
br>
 =A0 either the initial target inserted in the Request-URI by the UAC that<=
br>
 =A0 originates the request, or when a retarget occurred, the target<br>
 =A0 provided in that retarget operation. =A0Reroute and translation<br>
 =A0 operations never change the current target.<br>
<br>
</div>[MB] I don&#39;t think this definition fits what you want - i.e., if =
there<br>
is no retargeting, then none of the hi-entries are tagged - i.e., you<br>
won&#39;t have your concept of current. The way it works is that if no<br>
hi-entries are tagged, then you know that there was no retargeting, thus<br=
>
you know the request-uri has not been lost. [/MB]<br>
<div><div></div><div class=3D"h5"><br>
This defintion only makes sense when the following definitions are used:<br=
>
<br>
Name:<br>
<br>
 =A0 A name is a moniker for an entity which refers to it in a way which<br=
>
<br>
 =A0 reveals nothing about where it is in a network. =A0In SIP, tel URI<br>
<br>
 =A0 which doesn&#39;t represent the location of the entity is a name.<br>
<br>
<br>
Address:<br>
<br>
 =A0 An address is an identifier for an entity which describes it by its<br=
>
<br>
 =A0 location on the network. =A0In SIP, the SIP URI itself is a form of<br=
>
<br>
 =A0 address because the host part of the URI, the only mandatory part of<b=
r>
<br>
 =A0 the URI besides the scheme itself, indicates the location of a SIP<br>
<br>
 =A0 server that can be used to handle the request.<br>
<br>
Route:<br>
<br>
 =A0 Finally, a route is a =A0 sequence of SIP entities (including the UA<b=
r>
itself!) which are<br>
<br>
=A0traversed in order to forward a request to an address or name.<br>
<br>
Retarget (other term might be needed, as this is highly confusing):<br>
<br>
 =A0 =A0A Request-URI rewrite operation that changes the target identity of=
<br>
the request.<br>
<br>
Reroute (other term might be needed):<br>
<br>
 =A0 =A0A Request-URI rewrite operation that does not change the current<br=
>
target of the request, but =A0 =A0 determines the route/next hop taken to<b=
r>
reach the target-identity.<br>
<br>
/Hans Erik<br>
<br>
Mary Barnes wrote:<br>
&gt; So, by &quot;target-uri concept&quot; are you referring to the solutio=
n in the<br>
&gt; current target-uri draft or the application usage as such in the<br>
&gt; target-uri document (which is what I was referring to)?<br>
&gt;<br>
&gt; Yes, the problem is that SIP doesn&#39;t differentiate the various<br>
&gt; mechanisms by which a request-URI may be changed, which is what some<b=
r>
&gt; of the updated text in 4244bis is trying to do and why we need precise=
<br>
<br>
&gt; terms. I think we all agree that.<br>
&gt;<br>
&gt; I honestly don&#39;t care what we call the tag, as long as we&#39;re c=
lear<br>
&gt; about the functionality.<br>
&gt;<br>
&gt; Mary.<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: Hans Erik van Elburg [mailto:<a href=3D"mailto:ietf.hanserik@gma=
il.com">ietf.hanserik@gmail.com</a>]<br>
&gt; Sent: Wednesday, March 11, 2009 4:40 PM<br>
&gt; To: Barnes, Mary (RICH2:AR00)<br>
&gt; Cc: Shida Schubert; <a href=3D"mailto:sip@ietf.org">sip@ietf.org</a><b=
r>
&gt; Subject: Re: Terminology (was RE: [Sip] Fwd: I-D<br>
&gt; ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br>
&gt;<br>
&gt; The target-uri concept when defined properly is entirely application<b=
r>
&gt; agnostic.<br>
&gt;<br>
&gt; When one uses History-Info or another header as a vehicle for this<br>
&gt; information then it still can be considered application agnostic as<br=
>
&gt; any application can use this information as it likes.<br>
&gt;<br>
&gt; =A0From SIP perspective all the Request-URI rewrites look the same, th=
at<br>
<br>
&gt; is what brought us discussing this in the first place.<br>
&gt;<br>
&gt; /Hans Erik<br>
&gt;<br>
&gt; Mary Barnes wrote:<br>
&gt;<br>
&gt;&gt; And, I think the big issue with the terminology is that 4244bis is=
<br>
&gt;&gt; application agnostic, so it is tagging what happens to the URIs<br=
>
&gt;&gt; (why/how the specific URI was overwritten). =A0Whereas, specific<b=
r>
&gt;&gt; applications can derive information, such as what &quot;is&quot; t=
he real<br>
&gt;&gt; &quot;target&quot; for the request, from the History-Info header. =
=A0 So, IMHO,<br>
&gt;&gt; it&#39;s a matter of the target-uri document specifying that the l=
ast<br>
&gt;&gt; entry tagged by whatever name we come up with &quot;is&quot; the r=
eal &quot;target&quot;<br>
&gt;&gt; for the request. =A0I believe it&#39;s up to the applications to d=
escribe<br>
&gt;&gt; how they make use of the History-Info and not for the History-Info=
 to<br>
<br>
&gt;&gt; describe how applications can use it. =A0History-Info purely refle=
cts<br>
&gt;&gt; what happened from a SIP Protocol perspective and should not imply=
<br>
&gt;&gt; any<br>
&gt;&gt;<br>
&gt;<br>
&gt;<br>
&gt;&gt; application semantics. Indeed the intent of section 5 of 4244 was =
to<br>
&gt;&gt; inform the applications how they should decribe their usage of the=
<br>
&gt;&gt; header.<br>
&gt;&gt;<br>
&gt;&gt; Mary.<br>
&gt;&gt; Note: I changed the subject to hopefully make this topic easier to=
<br>
&gt;&gt; keep track of.<br>
&gt;&gt;<br>
&gt;&gt; ------------------------------------------------------------------=
---<br>
&gt;&gt; -<br>
&gt;&gt; --<br>
&gt;&gt; *From:* Hans Erik van Elburg [mailto:<a href=3D"mailto:ietf.hanser=
ik@gmail.com">ietf.hanserik@gmail.com</a>]<br>
&gt;&gt; *Sent:* Wednesday, March 11, 2009 3:59 AM<br>
&gt;&gt; *To:* Shida Schubert<br>
&gt;&gt; *Cc:* <a href=3D"mailto:sip@ietf.org">sip@ietf.org</a> List; Barne=
s, Mary (RICH2:AR00)<br>
&gt;&gt; *Subject:* Re: [Sip] Fwd: I-D<br>
&gt;&gt; ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br>
&gt;&gt;<br>
&gt;&gt; One of the problems that made the discussion quite tedious is the<=
br>
&gt;&gt; complete inversed meaning of the terms &quot;retarget&quot; and &q=
uot;reroute&quot;<br>
&gt;&gt; between 4244bis and target-uri factions have.<br>
&gt;&gt;<br>
&gt;&gt; Conclusion was that the terminology need to be properly defined an=
d<br>
&gt;&gt; some attempt was made to start such activity.<br>
&gt;&gt;<br>
&gt;&gt; On 3. I have strong concerns when we start tagging the URI as to h=
ow<br>
&gt;&gt; they come about &quot;retarget&quot;/&quot;mapped conficuration&qu=
ot; etc. It is better<br>
&gt;&gt; to<br>
&gt;&gt;<br>
&gt;<br>
&gt;<br>
&gt;&gt; embellish them with a tag that just represents there meaning for<b=
r>
&gt;&gt; example &quot;istarget&quot; or &quot;hop&quot;. For the following=
 reasons:<br>
&gt;&gt; 1. It is much more intuitive for the user of this information, its=
<br>
&gt;&gt; meaning can basically be guessed.<br>
&gt;&gt; 2. Such meaning also probably survives application in other proble=
m<br>
&gt;&gt; spaces that we had not foreseen when introducing the concept.<br>
&gt;&gt;<br>
&gt;&gt; The solution that is =A0described now in the target-URI delivery d=
raft<br>
&gt;&gt; is not yet complete, it does not solve the freephone use case for<=
br>
&gt;&gt; example and in its current form it will deliver exactly the same<b=
r>
&gt;&gt; target-URI as that which would be delivered by the =A0P-Called-Par=
ty-ID<br>
<br>
&gt;&gt; header. Contrary to what the draft says. So some work is needed<br=
>
still.<br>
&gt;&gt;<br>
&gt;&gt; /Hans Erik van Elburg<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; On Wed, Mar 11, 2009 at 7:13 AM, Shida Schubert &lt;<a href=3D"mai=
lto:shida@ntt-at.com">shida@ntt-at.com</a><br>
&gt;&gt; &lt;mailto:<a href=3D"mailto:shida@ntt-at.com">shida@ntt-at.com</a=
>&gt;&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 We have submitted the updated target-uri draft based on th=
e<br>
&gt;&gt;<br>
&gt; comments<br>
&gt;<br>
&gt;&gt; =A0 =A0 submitted to the list and comments received at IETF73.<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 I have taken over as editor as Jonathan didn&#39;t have th=
e cycles to<br>
<br>
&gt;&gt; =A0 =A0 update the draft, with Francois, Christer and Hans Erick a=
s<br>
&gt;&gt; =A0 =A0 additional<br>
&gt;&gt; =A0 =A0 co-authors and great deal of help from Mary.<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 The following summarizes the changes made to the target-ur=
i<br>
&gt;&gt;<br>
&gt; document<br>
&gt;<br>
&gt;&gt; =A0 =A0 1. Added use-case for toll-free number back<br>
&gt;&gt; =A0 =A0 2. Added definition of &quot;retarget&quot; operation.<br>
&gt;&gt; =A0 =A0 3. Removed a reference to URN<br>
&gt;&gt; =A0 =A0 4. Added a text discussing the difference to P-Called-Part=
y-Id<br>
&gt;&gt; =A0 =A0 5. Changed parameter name from &quot;target&quot; to &quot=
;istarget&quot;<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 Note, that the target-uri document still contains the norm=
ative<br>
&gt;&gt; =A0 =A0 text for the<br>
&gt;&gt; =A0 =A0 History-Info header.<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 In addition, Mary (with Francois as co-author) has submitt=
ed a<br>
&gt;&gt; =A0 =A0 rfc4244bis, with the following changes:<br>
&gt;&gt; =A0 =A0 1. Incorporated the normative aspects of the target-uri do=
cument<br>
&gt;&gt; =A0 =A0 into the existing normative text in RFC 4244 - the functio=
nality<br>
&gt;&gt;<br>
&gt; is<br>
&gt;<br>
&gt;&gt; =A0 =A0 virtually identical (as is some of the text) as the HI bas=
ed<br>
&gt;&gt;<br>
&gt; solution<br>
&gt;<br>
&gt;&gt; =A0 =A0 described in the target-uri document. =A0It&#39;s importan=
t that the<br>
&gt;&gt; =A0 =A0 solution<br>
&gt;&gt; =A0 =A0 be integrated into RFC 4244 as it MUST work and be based o=
n the<br>
&gt;&gt; =A0 =A0 normative<br>
&gt;&gt; =A0 =A0 aspects of RFC 4244.<br>
&gt;&gt; =A0 =A0 2. =A0Added the use cases from target-uri the the summary<=
br>
&gt;&gt; =A0 =A0 in the overview of rfc4244bis.<br>
&gt;&gt; =A0 =A0 3. Added an additional requirement to capture the &quot;ta=
rget-uri&quot;<br>
&gt;&gt; =A0 =A0 information.<br>
&gt;&gt; =A0 =A0 4. Fixed an error in the RFC 4244 ABNF and added &quot;ret=
arget&quot;<br>
&gt;&gt;<br>
&gt; parameter.<br>
&gt;<br>
&gt;&gt; =A0 =A0 5. Added a more simplified example.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 We had some very long offline exchanges as to the best way=
<br>
&gt;&gt; forward<br>
&gt;&gt;<br>
&gt; and<br>
&gt;<br>
&gt;&gt; =A0 =A0 remaining work for both documents.<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 Some of the issues identified are:<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 ::Issues::<br>
&gt;&gt; =A0 =A0 =A0 =A01. Should we remove the normative text from target-=
uri and<br>
&gt;&gt;<br>
&gt; progress<br>
&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 =A04244bis along with the target-uri document =
to meet the<br>
&gt;&gt; =A0 =A0 chartered<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 SIP WG milestone?<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A02. Name of the parameter.<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 At the last meeting, parameter &quot;target&qu=
ot; was said<br>
&gt;&gt;<br>
&gt; inappropriate<br>
&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 because voicemail-uri spec already defines a p=
arameter<br>
&gt;&gt;<br>
&gt; called<br>
&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 &quot;target&quot; which also can be found in =
hi-entry, thus<br>
&gt;&gt;<br>
&gt; potentially<br>
&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 causing confusion.<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 Currently the target-uri draft uses &quot;ista=
rget&quot; and 4244bis<br>
&gt;&gt;<br>
&gt; uses<br>
&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 &quot;retarget&quot; =A0but we could never com=
e to<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 a consensus on what name is appropriate. =A0Ot=
her suggestions<br>
&gt;&gt;<br>
&gt; have<br>
&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 included the following:<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 &quot;target-identity&quot; (someone didn&#39;=
t like that &quot;identity&quot; is<br>
&gt;&gt; =A0 =A0 also a SIP header)<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 &quot;reg-uri&quot; =A0(can be paired with &qu=
ot;mapped-uri&quot; for item 3<br>
&gt;&gt;<br>
&gt; below)<br>
&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 &quot;aor&quot;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 &quot;jibberish&quot;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 etc.<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 One reason this is so difficult relates to the=
 problem<br>
&gt;&gt; =A0 =A0 statement in target-uri in that<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 RFC 3261 doesn&#39;t differentiate the mechani=
sm by which the<br>
&gt;&gt;<br>
&gt; new<br>
&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 (target) Request-URI is selected. =A0Another i=
ssue is that<br>
&gt;&gt; =A0 =A0 some of the terminology in<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 RFC 3261 is overloaded - e.g., &quot;forwardin=
g&quot; refers both to<br>
a<br>
&gt;&gt; =A0 =A0 Proxy<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 which does not have responsibility for the dom=
ain of the<br>
&gt;&gt; =A0 =A0 request-URI<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 in the incoming request, thus the proxy just &=
quot;forwards&quot; the<br>
&gt;&gt; =A0 =A0 request to<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 the next hop AND &quot;forwarding&quot; is use=
d to describe the<br>
&gt;&gt; =A0 =A0 process whereby<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 the outgoing request is built and &quot;forwar=
ded&quot; to the next<br>
&gt;&gt; =A0 =A0 hop at which<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 point the proxy does not know how the new requ=
est-uri was<br>
&gt;&gt; =A0 =A0 selected.<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 RFC 4244 has attempted to clarify the terms an=
d attempts to<br>
&gt;&gt; =A0 =A0 use &quot;forward&quot;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 in the context of the former situation and &qu=
ot;retarget&quot; for<br>
&gt;&gt; =A0 =A0 the case whereby<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 a proxy is responsible for the domain and thus=
 can use a<br>
&gt;&gt; =A0 =A0 number of<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 mechanism to select the new target for the req=
uest - e.g.,<br>
a<br>
&gt;&gt; =A0 =A0 REGISTRAR,<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 configured data, etc.<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A03. =A0Related to the last point in item 2 above, it has=
 been<br>
&gt;&gt; =A0 =A0 proposed that<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 we differentiate the hi-entries even more by d=
efining<br>
&gt;&gt; =A0 =A0 separate parameters<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0for registered and configured/mapped contacts.<=
br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0Currently when the R-URI is translated to a URI=
 which is<br>
&gt;&gt; =A0 =A0 either derived<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0from location service lookup(registered by UA) =
or from<br>
&gt;&gt;<br>
&gt; mapping<br>
&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0table, there is no differentiation as to how th=
e URI was<br>
&gt;&gt; =A0 =A0 derived once it is<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0added to the list of potential targets.<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0The general consensus of the authors of the two=
 documents<br>
was<br>
&gt;&gt; =A0 =A0 that it may<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0be useful for some services to have the hi-entr=
ies tagged<br>
&gt;&gt; =A0 =A0 with the<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0more specific information.<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0And, of course, this gets us into another namin=
g contest. In<br>
&gt;&gt; =A0 =A0 the end, the naming<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0is not so important as long as the term isn&#39=
;t too overloaded<br>
&gt;&gt; =A0 =A0 and it is defined<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0precisely in the document(s).<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 We would appreciate WG feedback on these issues and any ot=
her<br>
&gt;&gt; =A0 =A0 comments on<br>
&gt;&gt; =A0 =A0 the two documents prior to IETF-74.<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 Regards,<br>
&gt;&gt; =A0 =A0 Shida and Mary.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 Begin forwarded message:<br>
&gt;&gt;<br>
&gt;&gt;&gt; =A0 =A0 *From: *<a href=3D"mailto:Internet-Drafts@ietf.org">In=
ternet-Drafts@ietf.org</a><br>
&gt;&gt;&gt;<br>
&gt; &lt;mailto:<a href=3D"mailto:Internet-Drafts@ietf.org">Internet-Drafts=
@ietf.org</a>&gt;<br>
&gt;<br>
&gt;&gt;&gt; =A0 =A0 *Date: *March 10, 2009 2:30:01 AM JST<br>
&gt;&gt;&gt; =A0 =A0 *To: *<a href=3D"mailto:i-d-announce@ietf.org">i-d-ann=
ounce@ietf.org</a> &lt;mailto:<a href=3D"mailto:i-d-announce@ietf.org">i-d-=
announce@ietf.org</a>&gt;<br>
&gt;&gt;&gt; =A0 =A0 *Subject: **I-D<br>
&gt;&gt;&gt; =A0 =A0 ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt =
*<br>
&gt;&gt;&gt; =A0 =A0 *Reply-To: *<a href=3D"mailto:internet-drafts@ietf.org=
">internet-drafts@ietf.org</a><br>
&gt;&gt;&gt; =A0 =A0 &lt;mailto:<a href=3D"mailto:internet-drafts@ietf.org"=
>internet-drafts@ietf.org</a>&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; =A0 =A0 A New Internet-Draft is available from the on-line<br>
&gt;&gt;&gt;<br>
&gt; Internet-Drafts<br>
&gt;<br>
&gt;&gt;&gt; =A0 =A0 directories.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; =A0 =A0 Title<br>
&gt;&gt;&gt; =A0 =A0 : Delivery of Request-URI Targets to User Agents<br>
&gt;&gt;&gt; =A0 =A0 Author(s)<br>
&gt;&gt;&gt; =A0 =A0 : J. Rosenberg, H. van Elburg, C. Holmberg, F. Audet, =
S.<br>
Schubert<br>
&gt;&gt;&gt; =A0 =A0 Filename : draft-rosenberg-sip-target-uri-delivery-01.=
txt<br>
&gt;&gt;&gt; =A0 =A0 Pages<br>
&gt;&gt;&gt; =A0 =A0 : 16<br>
&gt;&gt;&gt; =A0 =A0 Date<br>
&gt;&gt;&gt; =A0 =A0 : 2009-3-9<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; =A0 =A0 When a Session Initiation Protocol (SIP) proxy receive=
s a<br>
request<br>
&gt;&gt;&gt; =A0 =A0 =A0 targeted at a URI identifying a user or resource i=
t is<br>
&gt;&gt;&gt;<br>
&gt; responsible<br>
&gt;<br>
&gt;&gt;&gt; =A0 =A0 =A0 for, the proxy translates the URI to a registered =
or<br>
configured<br>
&gt;&gt;&gt; =A0 =A0 =A0 contact URI of an agent representing that user or =
resource.<br>
&gt;&gt;&gt; In<br>
&gt;&gt;&gt;<br>
&gt; the<br>
&gt;<br>
&gt;&gt;&gt; =A0 =A0 =A0 process, the original URI is removed from the requ=
est.<br>
&gt;&gt;&gt; =A0 =A0 =A0Numerous use<br>
&gt;&gt;&gt; =A0 =A0 =A0 cases have arisen which require this information t=
o be<br>
&gt;&gt;&gt;<br>
&gt; delivered to<br>
&gt;<br>
&gt;&gt;&gt; =A0 =A0 =A0 the user agent. =A0This document describes these u=
se cases and<br>
&gt;&gt;&gt; =A0 =A0 defines<br>
&gt;&gt;&gt; =A0 =A0 =A0 an extension to the History-Info header field whic=
h allows it<br>
&gt;&gt;&gt;<br>
&gt; to be<br>
&gt;<br>
&gt;&gt;&gt; =A0 =A0 =A0 used to support those cases.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; =A0 =A0 A URL for this Internet-Draft is:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; <a href=3D"http://www.ietf.org/internet-drafts/draft-rosenberg=
-sip-target-uri-d" target=3D"_blank">http://www.ietf.org/internet-drafts/dr=
aft-rosenberg-sip-target-uri-d</a><br>
&gt;&gt;&gt; e<br>
&gt;&gt;&gt; livery-01.txt<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; =A0 =A0 Internet-Drafts are also available by anonymous FTP at=
:<br>
&gt;&gt;&gt; =A0 =A0 <a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=
=3D"_blank">ftp://ftp.ietf.org/internet-drafts/</a><br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; =A0 =A0 Below is the data which will enable a MIME compliant m=
ail reader<br>
&gt;&gt;&gt; =A0 =A0 implementation to automatically retrieve the ASCII ver=
sion of<br>
the<br>
&gt;&gt;&gt; =A0 =A0 Internet-Draft.<br>
&gt;&gt;&gt;<br>
&gt;&gt; =A0 =A0 Content-Type: text/plain&lt;BR&gt;Content-ID:<br>
&gt;&gt; =A0 =A0 &amp;<a href=3D"mailto:lt%3B2009-3-9102756.I-D@ietf.org">l=
t;2009-3-9102756.I-D@ietf.org</a><br>
&gt;&gt; =A0 =A0 &lt;mailto:<a href=3D"mailto:2009-3-9102756.I-D@ietf.org">=
2009-3-9102756.I-D@ietf.org</a>&gt;&amp;gt;&lt;BR&gt;&lt;BR&gt;<br>
&gt;&gt;<br>
&gt;&gt;&gt; =A0 =A0 _______________________________________________<br>
&gt;&gt;&gt; =A0 =A0 I-D-Announce mailing list<br>
&gt;&gt;&gt; =A0 =A0 <a href=3D"mailto:I-D-Announce@ietf.org">I-D-Announce@=
ietf.org</a> &lt;mailto:<a href=3D"mailto:I-D-Announce@ietf.org">I-D-Announ=
ce@ietf.org</a>&gt;<br>
&gt;&gt;&gt; =A0 =A0 <a href=3D"https://www.ietf.org/mailman/listinfo/i-d-a=
nnounce" target=3D"_blank">https://www.ietf.org/mailman/listinfo/i-d-announ=
ce</a><br>
&gt;&gt;&gt; =A0 =A0 Internet-Draft directories: <a href=3D"http://www.ietf=
.org/shadow.html" target=3D"_blank">http://www.ietf.org/shadow.html</a><br>
&gt;&gt;&gt; =A0 =A0 or <a href=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.tx=
t" target=3D"_blank">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</a><br>
&gt;&gt;&gt;<br>
&gt;&gt; =A0 =A0 _______________________________________________<br>
&gt;&gt; =A0 =A0 Sip mailing list =A0<a href=3D"https://www.ietf.org/mailma=
n/listinfo/sip" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sip=
</a><br>
&gt;&gt; =A0 =A0 This list is for NEW development of the core SIP Protocol<=
br>
&gt;&gt; =A0 =A0 Use <a href=3D"mailto:sip-implementors@cs.columbia.edu">si=
p-implementors@cs.columbia.edu</a><br>
&gt;&gt; =A0 =A0 &lt;mailto:<a href=3D"mailto:sip-implementors@cs.columbia.=
edu">sip-implementors@cs.columbia.edu</a>&gt; for questions on<br>
&gt;&gt; current<br>
&gt;&gt;<br>
&gt; sip<br>
&gt;<br>
&gt;&gt; =A0 =A0 Use <a href=3D"mailto:sipping@ietf.org">sipping@ietf.org</=
a> &lt;mailto:<a href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a>&gt; =
for new<br>
&gt;&gt; =A0 =A0 developments on the application of sip<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
</div></div></blockquote></div><br>

--0015174c10a098e9240464e7daf9--

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Cc: sip@ietf.org
Subject: Re: [Sip] Terminology (was RE: Fwd: I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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If 'istarget' is used only to identify the value that was overwritten in
the Request-URI by a contact URI, an alternative approach would be to
flag the contact URI H-I entry as 'iscontact'. Then the UAS would just
need to look for the most recent H-I entry that is not marked
'iscontact'.

John


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Hans Erik van Elburg
	Sent: 12 March 2009 08:35
	To: Mary Barnes
	Cc: sip@ietf.org
	Subject: Re: [Sip] Terminology (was RE: Fwd:
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
=09
=09
	Yes, because you are using the 3261 use of target and the
4244bis introduced definition of retarget. I thought it was clear that
we need other words as those definitions don't match the target-uri
drafts use of the terms. Also they do not suffice to provide a solution
for the use cases in the target-uri draft.
=09
	The 3261 text you refer to is exactly about the case where the
home proxy overwrites the Request-URI with a new target. This target is
teh registered contact address. And hence this would be what target-uri
calls a hop or a route. This case and this is where it gets confusing is
not a "retarget" in the target-uri draft use of the term.
=09
	The target-uri draft states:
=09
	   "To avoid confusion, we
	   refer to a SIP URI that is an address for a user or resource
as a
	   "target" and a SIP URI that is a hop for reaching that user
as a
=09
	   "hop".
=09

	Apparently that does not suffice to avoid confusion.
=09
	As for the tagging, speaking about the solution before agreeing
on the terminology and the problem it should solve is meaningless.
=09
	/Hans Erik van Elburg
=09
=09
=09
	On Thu, Mar 12, 2009 at 2:33 AM, Mary Barnes
<mary.barnes@nortel.com> wrote:
=09

		Responses below [MB].
	=09

		-----Original Message-----
		From: Hans Erik van Elburg
[mailto:ietf.hanserik@gmail.com]
	=09
		Sent: Wednesday, March 11, 2009 6:42 PM
		To: Barnes, Mary (RICH2:AR00)
		Cc: Shida Schubert; sip@ietf.org
		Subject: Re: Terminology (was RE: [Sip] Fwd: I-D
		ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
	=09
	=09
		I was talking about the concept.
	=09
		The use cases only describe  cases where the target-uri
(lets call that
		current target from now) has been lost when an initial
request for a
		dialog or standalone request arrives at the UAS.
	=09
		[MB] And I think this is where the terminology confusion
starts - I
		don't think of the "lost" target-uri as being the
"current" target. In
		my mind, the "current" target is reflected by the last
hi-entry and the
		request-uri in the incoming request at the UAS. If the
entity that sent
		the request was the entity that added the last hi-entry,
then the uri in
		that hi-entry is the same as the request-URI in the SIP
request that
		arrives at the UAS. I refer to the "lost" uri as the one
that was
		"retargeted" - that's the one the UAS wants to pull from
the hi-entries
		in the incoming request. That hi-entry was not the one
that was just
		added by the entity that built the request just received
by the UAS.
		That hi-entry is tagged with whatever name we are going
to tag it with
		BEFORE the request is forwarded (using the term forward
per section 16.6
		of RFC 3261).  That tag is added once the target set of
potential
		candidates for the new request uri are determined in
section 16.5 of
		3261 (with "target set" being a 3261 term), just before
the request is
		forwarded in section 16.6 to one of those targets.  A
new hi-entry
		(which will be the last hi-entry in the request received
by the UAS) is
		added in section 16.6 of 3261 as the request is
forwarded.  At this
		point in time, the lost information is in the previous
hi-entry when the
		outgoing request is sent. [/MB]
	=09

		What has been lost is the current target of the request:
	=09
		[MB] Right - at which point in my mind, it's no longer
the current
		target ;) Maybe we call it "lost". [/MB]
	=09


		     Current target
	=09
		  The current target of an initial request for a dialog
or standalone
		  request is the name or address to which the request is
targeted, i.e.
		  either the initial target inserted in the Request-URI
by the UAC that
		  originates the request, or when a retarget occurred,
the target
		  provided in that retarget operation.  Reroute and
translation
		  operations never change the current target.
	=09
	=09
		[MB] I don't think this definition fits what you want -
i.e., if there
		is no retargeting, then none of the hi-entries are
tagged - i.e., you
		won't have your concept of current. The way it works is
that if no
		hi-entries are tagged, then you know that there was no
retargeting, thus
		you know the request-uri has not been lost. [/MB]
	=09

		This defintion only makes sense when the following
definitions are used:
	=09
		Name:
	=09
		  A name is a moniker for an entity which refers to it
in a way which
	=09
		  reveals nothing about where it is in a network.  In
SIP, tel URI
	=09
		  which doesn't represent the location of the entity is
a name.
	=09
	=09
		Address:
	=09
		  An address is an identifier for an entity which
describes it by its
	=09
		  location on the network.  In SIP, the SIP URI itself
is a form of
	=09
		  address because the host part of the URI, the only
mandatory part of
	=09
		  the URI besides the scheme itself, indicates the
location of a SIP
	=09
		  server that can be used to handle the request.
	=09
		Route:
	=09
		  Finally, a route is a   sequence of SIP entities
(including the UA
		itself!) which are
	=09
		 traversed in order to forward a request to an address
or name.
	=09
		Retarget (other term might be needed, as this is highly
confusing):
	=09
		   A Request-URI rewrite operation that changes the
target identity of
		the request.
	=09
		Reroute (other term might be needed):
	=09
		   A Request-URI rewrite operation that does not change
the current
		target of the request, but     determines the route/next
hop taken to
		reach the target-identity.
	=09
		/Hans Erik
	=09
		Mary Barnes wrote:
		> So, by "target-uri concept" are you referring to the
solution in the
		> current target-uri draft or the application usage as
such in the
		> target-uri document (which is what I was referring
to)?
		>
		> Yes, the problem is that SIP doesn't differentiate the
various
		> mechanisms by which a request-URI may be changed,
which is what some
		> of the updated text in 4244bis is trying to do and why
we need precise
	=09
		> terms. I think we all agree that.
		>
		> I honestly don't care what we call the tag, as long as
we're clear
		> about the functionality.
		>
		> Mary.
		>
		> -----Original Message-----
		> From: Hans Erik van Elburg
[mailto:ietf.hanserik@gmail.com]
		> Sent: Wednesday, March 11, 2009 4:40 PM
		> To: Barnes, Mary (RICH2:AR00)
		> Cc: Shida Schubert; sip@ietf.org
		> Subject: Re: Terminology (was RE: [Sip] Fwd: I-D
		> ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
		>
		> The target-uri concept when defined properly is
entirely application
		> agnostic.
		>
		> When one uses History-Info or another header as a
vehicle for this
		> information then it still can be considered
application agnostic as
		> any application can use this information as it likes.
		>
		>  From SIP perspective all the Request-URI rewrites
look the same, that
	=09
		> is what brought us discussing this in the first place.
		>
		> /Hans Erik
		>
		> Mary Barnes wrote:
		>
		>> And, I think the big issue with the terminology is
that 4244bis is
		>> application agnostic, so it is tagging what happens
to the URIs
		>> (why/how the specific URI was overwritten).  Whereas,
specific
		>> applications can derive information, such as what
"is" the real
		>> "target" for the request, from the History-Info
header.   So, IMHO,
		>> it's a matter of the target-uri document specifying
that the last
		>> entry tagged by whatever name we come up with "is"
the real "target"
		>> for the request.  I believe it's up to the
applications to describe
		>> how they make use of the History-Info and not for the
History-Info to
	=09
		>> describe how applications can use it.  History-Info
purely reflects
		>> what happened from a SIP Protocol perspective and
should not imply
		>> any
		>>
		>
		>
		>> application semantics. Indeed the intent of section 5
of 4244 was to
		>> inform the applications how they should decribe their
usage of the
		>> header.
		>>
		>> Mary.
		>> Note: I changed the subject to hopefully make this
topic easier to
		>> keep track of.
		>>
		>>
---------------------------------------------------------------------
		>> -
		>> --
		>> *From:* Hans Erik van Elburg
[mailto:ietf.hanserik@gmail.com]
		>> *Sent:* Wednesday, March 11, 2009 3:59 AM
		>> *To:* Shida Schubert
		>> *Cc:* sip@ietf.org List; Barnes, Mary (RICH2:AR00)
		>> *Subject:* Re: [Sip] Fwd: I-D
		>> ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
		>>
		>> One of the problems that made the discussion quite
tedious is the
		>> complete inversed meaning of the terms "retarget" and
"reroute"
		>> between 4244bis and target-uri factions have.
		>>
		>> Conclusion was that the terminology need to be
properly defined and
		>> some attempt was made to start such activity.
		>>
		>> On 3. I have strong concerns when we start tagging
the URI as to how
		>> they come about "retarget"/"mapped conficuration"
etc. It is better
		>> to
		>>
		>
		>
		>> embellish them with a tag that just represents there
meaning for
		>> example "istarget" or "hop". For the following
reasons:
		>> 1. It is much more intuitive for the user of this
information, its
		>> meaning can basically be guessed.
		>> 2. Such meaning also probably survives application in
other problem
		>> spaces that we had not foreseen when introducing the
concept.
		>>
		>> The solution that is  described now in the target-URI
delivery draft
		>> is not yet complete, it does not solve the freephone
use case for
		>> example and in its current form it will deliver
exactly the same
		>> target-URI as that which would be delivered by the
P-Called-Party-ID
	=09
		>> header. Contrary to what the draft says. So some work
is needed
		still.
		>>
		>> /Hans Erik van Elburg
		>>
		>>
		>> On Wed, Mar 11, 2009 at 7:13 AM, Shida Schubert
<shida@ntt-at.com
		>> <mailto:shida@ntt-at.com>> wrote:
		>>
		>>
		>>     We have submitted the updated target-uri draft
based on the
		>>
		> comments
		>
		>>     submitted to the list and comments received at
IETF73.
		>>
		>>     I have taken over as editor as Jonathan didn't
have the cycles to
	=09
		>>     update the draft, with Francois, Christer and
Hans Erick as
		>>     additional
		>>     co-authors and great deal of help from Mary.
		>>
		>>     The following summarizes the changes made to the
target-uri
		>>
		> document
		>
		>>     1. Added use-case for toll-free number back
		>>     2. Added definition of "retarget" operation.
		>>     3. Removed a reference to URN
		>>     4. Added a text discussing the difference to
P-Called-Party-Id
		>>     5. Changed parameter name from "target" to
"istarget"
		>>
		>>     Note, that the target-uri document still contains
the normative
		>>     text for the
		>>     History-Info header.
		>>
		>>     In addition, Mary (with Francois as co-author)
has submitted a
		>>     rfc4244bis, with the following changes:
		>>     1. Incorporated the normative aspects of the
target-uri document
		>>     into the existing normative text in RFC 4244 -
the functionality
		>>
		> is
		>
		>>     virtually identical (as is some of the text) as
the HI based
		>>
		> solution
		>
		>>     described in the target-uri document.  It's
important that the
		>>     solution
		>>     be integrated into RFC 4244 as it MUST work and
be based on the
		>>     normative
		>>     aspects of RFC 4244.
		>>     2.  Added the use cases from target-uri the the
summary
		>>     in the overview of rfc4244bis.
		>>     3. Added an additional requirement to capture the
"target-uri"
		>>     information.
		>>     4. Fixed an error in the RFC 4244 ABNF and added
"retarget"
		>>
		> parameter.
		>
		>>     5. Added a more simplified example.
		>>
		>>
		>>     We had some very long offline exchanges as to the
best way
		>> forward
		>>
		> and
		>
		>>     remaining work for both documents.
		>>
		>>     Some of the issues identified are:
		>>
		>>     ::Issues::
		>>        1. Should we remove the normative text from
target-uri and
		>>
		> progress
		>
		>>            4244bis along with the target-uri document
to meet the
		>>     chartered
		>>           SIP WG milestone?
		>>
		>>
		>>        2. Name of the parameter.
		>>           At the last meeting, parameter "target" was
said
		>>
		> inappropriate
		>
		>>           because voicemail-uri spec already defines
a parameter
		>>
		> called
		>
		>>           "target" which also can be found in
hi-entry, thus
		>>
		> potentially
		>
		>>           causing confusion.
		>>
		>>           Currently the target-uri draft uses
"istarget" and 4244bis
		>>
		> uses
		>
		>>           "retarget"  but we could never come to
		>>           a consensus on what name is appropriate.
Other suggestions
		>>
		> have
		>
		>>           included the following:
		>>           "target-identity" (someone didn't like that
"identity" is
		>>     also a SIP header)
		>>           "reg-uri"  (can be paired with "mapped-uri"
for item 3
		>>
		> below)
		>
		>>           "aor"
		>>           "jibberish"
		>>           etc.
		>>
		>>           One reason this is so difficult relates to
the problem
		>>     statement in target-uri in that
		>>           RFC 3261 doesn't differentiate the
mechanism by which the
		>>
		> new
		>
		>>           (target) Request-URI is selected.  Another
issue is that
		>>     some of the terminology in
		>>           RFC 3261 is overloaded - e.g., "forwarding"
refers both to
		a
		>>     Proxy
		>>           which does not have responsibility for the
domain of the
		>>     request-URI
		>>           in the incoming request, thus the proxy
just "forwards" the
		>>     request to
		>>           the next hop AND "forwarding" is used to
describe the
		>>     process whereby
		>>           the outgoing request is built and
"forwarded" to the next
		>>     hop at which
		>>           point the proxy does not know how the new
request-uri was
		>>     selected.
		>>           RFC 4244 has attempted to clarify the terms
and attempts to
		>>     use "forward"
		>>           in the context of the former situation and
"retarget" for
		>>     the case whereby
		>>           a proxy is responsible for the domain and
thus can use a
		>>     number of
		>>           mechanism to select the new target for the
request - e.g.,
		a
		>>     REGISTRAR,
		>>           configured data, etc.
		>>
		>>      3.  Related to the last point in item 2 above,
it has been
		>>     proposed that
		>>           we differentiate the hi-entries even more
by defining
		>>     separate parameters
		>>          for registered and configured/mapped
contacts.
		>>          Currently when the R-URI is translated to a
URI which is
		>>     either derived
		>>          from location service lookup(registered by
UA) or from
		>>
		> mapping
		>
		>>          table, there is no differentiation as to how
the URI was
		>>     derived once it is
		>>          added to the list of potential targets.
		>>
		>>          The general consensus of the authors of the
two documents
		was
		>>     that it may
		>>          be useful for some services to have the
hi-entries tagged
		>>     with the
		>>          more specific information.
		>>
		>>          And, of course, this gets us into another
naming contest. In
		>>     the end, the naming
		>>          is not so important as long as the term
isn't too overloaded
		>>     and it is defined
		>>          precisely in the document(s).
		>>
		>>     We would appreciate WG feedback on these issues
and any other
		>>     comments on
		>>     the two documents prior to IETF-74.
		>>
		>>     Regards,
		>>     Shida and Mary.
		>>
		>>
		>>
		>>     Begin forwarded message:
		>>
		>>>     *From: *Internet-Drafts@ietf.org
		>>>
		> <mailto:Internet-Drafts@ietf.org>
		>
		>>>     *Date: *March 10, 2009 2:30:01 AM JST
		>>>     *To: *i-d-announce@ietf.org
<mailto:i-d-announce@ietf.org>
		>>>     *Subject: **I-D
		>>>
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt *
		>>>     *Reply-To: *internet-drafts@ietf.org
		>>>     <mailto:internet-drafts@ietf.org>
		>>>
		>>>     A New Internet-Draft is available from the
on-line
		>>>
		> Internet-Drafts
		>
		>>>     directories.
		>>>
		>>>
		>>>     Title
		>>>     : Delivery of Request-URI Targets to User Agents
		>>>     Author(s)
		>>>     : J. Rosenberg, H. van Elburg, C. Holmberg, F.
Audet, S.
		Schubert
		>>>     Filename :
draft-rosenberg-sip-target-uri-delivery-01.txt
		>>>     Pages
		>>>     : 16
		>>>     Date
		>>>     : 2009-3-9
		>>>
		>>>     When a Session Initiation Protocol (SIP) proxy
receives a
		request
		>>>       targeted at a URI identifying a user or
resource it is
		>>>
		> responsible
		>
		>>>       for, the proxy translates the URI to a
registered or
		configured
		>>>       contact URI of an agent representing that user
or resource.
		>>> In
		>>>
		> the
		>
		>>>       process, the original URI is removed from the
request.
		>>>      Numerous use
		>>>       cases have arisen which require this
information to be
		>>>
		> delivered to
		>
		>>>       the user agent.  This document describes these
use cases and
		>>>     defines
		>>>       an extension to the History-Info header field
which allows it
		>>>
		> to be
		>
		>>>       used to support those cases.
		>>>
		>>>     A URL for this Internet-Draft is:
		>>>
		>>>
http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri-d
		>>> e
		>>> livery-01.txt
		>>>
		>>>     Internet-Drafts are also available by anonymous
FTP at:
		>>>     ftp://ftp.ietf.org/internet-drafts/
		>>>
		>>>     Below is the data which will enable a MIME
compliant mail reader
		>>>     implementation to automatically retrieve the
ASCII version of
		the
		>>>     Internet-Draft.
		>>>
		>>     Content-Type: text/plain<BR>Content-ID:
		>>     &lt;2009-3-9102756.I-D@ietf.org
<mailto:lt%3B2009-3-9102756.I-D@ietf.org>=20
		>>     <mailto:2009-3-9102756.I-D@ietf.org>&gt;<BR><BR>
		>>
		>>>     _______________________________________________
		>>>     I-D-Announce mailing list
		>>>     I-D-Announce@ietf.org
<mailto:I-D-Announce@ietf.org>
		>>>
https://www.ietf.org/mailman/listinfo/i-d-announce
		>>>     Internet-Draft directories:
http://www.ietf.org/shadow.html
		>>>     or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
		>>>
		>>     _______________________________________________
		>>     Sip mailing list
https://www.ietf.org/mailman/listinfo/sip
		>>     This list is for NEW development of the core SIP
Protocol
		>>     Use sip-implementors@cs.columbia.edu
		>>     <mailto:sip-implementors@cs.columbia.edu> for
questions on
		>> current
		>>
		> sip
		>
		>>     Use sipping@ietf.org <mailto:sipping@ietf.org>
for new
		>>     developments on the application of sip
		>>
		>>
		>>
	=09



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Date: Thu, 12 Mar 2009 12:48:57 +0100
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Thread-Topic: RE: One chart  //"UPDATE during Re-INVITE" discussion
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References: <CA9998CD4A020D418654FCDEF4E707DF0BA2CB0C@esealmw113.eemea.ericsson.se> <OFE3E9D5E4.964E749A-ON48257577.0029CDAD-48257577.002A3529@zte.com.cn>
From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: <gao.yang2@zte.com.cn>
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Cc: SIP <sip@ietf.org>, SIPPING <sipping@ietf.org>, Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
Subject: Re: [Sip] One chart  //"UPDATE during Re-INVITE" discussion
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Hi,
=20
If the UAC receives the 487 and can't fallback to the pre-reINVITE =
state, it sends a new re-INVITE with something that it DOES supports. =
That is also described in Gonzalo's draft, I belive.
=20
And, you could have the same problem if the target refresh was done =
already in the re-INVITE/18x. If the UAC then sends CANCEL, and can't =
fall back, it will have to send a new re-INVITE.
=20
I also think it is bad practise to do a target refresh in a nested =
UPDATE, but that's a separate issue.
=20
Regards,
=20
Christer
=20

________________________________

	From: gao.yang2@zte.com.cn [mailto:gao.yang2@zte.com.cn]=20
	Sent: 12. maaliskuuta 2009 9:41
	To: Christer Holmberg
	Cc: Gonzalo Camarillo; SIP; SIPPING
	Subject: =B4=F0=B8=B4: RE: One chart //"UPDATE during Re-INVITE" =
discussion
=09
=09

	Hi,=20
=09
	I just mean that if we treat F3 as sub-transaction of F1, then after =
F7(ACK), UAC can discard refreshed dialog state in F3.=20
=09
	And it may make the two parts can not communicate anymore in current =
dialog.=20
=09
	And IMO, this is a counterevidence for treating all UPDATE(during =
Re-INVITE) as sub-transaction of Re-INVITE.=20
=09
	Gao=20
=09
=09
=09
=09
=09
"Christer Holmberg" <christer.holmberg@ericsson.com>=20

2009-03-12 15:31=20

=CA=D5=BC=FE=C8=CB
<gao.yang2@zte.com.cn>, "SIP" <sip@ietf.org>, "SIPPING" =
<sipping@ietf.org>=20
=B3=AD=CB=CD
"Gonzalo Camarillo" <gonzalo.camarillo@ericsson.com>=20
=D6=F7=CC=E2
RE: One chart  //"UPDATE during Re-INVITE" discussion

=09




	In my opinion 487 should trigger whatever actions/rollback that any =
other non-200 response sent by the UAS would.=20
	 =20
	Regards,=20
	 =20
	Christer=20
=09
=09
________________________________

	From: gao.yang2@zte.com.cn [mailto:gao.yang2@zte.com.cn]=20
	Sent: 12. maaliskuuta 2009 5:47
	To: SIP; SIPPING
	Cc: Christer Holmberg; Gonzalo Camarillo
	Subject: One chart //"UPDATE during Re-INVITE" discussion
=09
=09
	     UAC                   UAS=20
	      | session established |=20
	      |<=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D>|=20
	      |                     |=20
	      | F1  re-INVITE (SDP) |=20
	      |-------------------->|=20
	      | F2 1xx-rel (SDP)    |=20
	      |<--------------------|=20
	      | F3   UPDATE         |=20
	      |<--------------------| Target refreshing=20
	      | F4 2xx UPT          |=20
	      |-------------------->|=20
	      |                     |=20
	      | F5/6 Cancel/200OK   |=20
	      |-------------------->|=20
	      | F7    487 INV       |=20
	      |<--------------------|=20
	      | F7     ACK          |=20
	      |-------------------->|=20
	      | F8  re-INVITE (SDP) |=20
	      |-------------------->| Using which dialog state=20
=09
	F3/F4 is about target refreshing.=20
=09
	And if UAC sends Cancel=A3=AC UAS sends 487. The first Re-INVITE is =
discarded.=20
=09
	But if UAC sends another Re-INVITE, which dialog state should it use?=20
=09
	I think is the one refreshed by F3. That's the merit of regarding =
UPDATE/200OK's atomicity.=20
=09
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dgb2312">
<META content=3D"MSHTML 6.00.6000.16809" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D169564411-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Hi,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D169564411-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D169564411-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>If the UAC receives the 487 and can't fallback =
to the=20
pre-reINVITE state, it sends a new re-INVITE with something that it=20
DOES&nbsp;supports. That is also described in Gonzalo's draft, I=20
belive.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D169564411-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D169564411-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>And, you could have the same problem if the =
target refresh=20
was done already in the re-INVITE/18x. If the UAC then sends CANCEL, and =
can't=20
fall back, it will have to send a new re-INVITE.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D169564411-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D169564411-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>I also think it is bad practise to do a target =
refresh in a=20
nested UPDATE, but that's a separate issue.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D169564411-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D169564411-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D169564411-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D169564411-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Christer</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D169564411-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> gao.yang2@zte.com.cn=20
  [mailto:gao.yang2@zte.com.cn] <BR><B>Sent:</B> 12. maaliskuuta 2009=20
  9:41<BR><B>To:</B> Christer Holmberg<BR><B>Cc:</B> Gonzalo Camarillo; =
SIP;=20
  SIPPING<BR><B>Subject:</B> =B4=F0=B8=B4: RE: One chart //"UPDATE =
during Re-INVITE"=20
  discussion<BR></FONT><BR></DIV>
  <DIV></DIV><BR><FONT face=3Dsans-serif size=3D2>Hi,</FONT> =
<BR><BR><FONT=20
  face=3Dsans-serif size=3D2>I just mean that if we treat F3 as =
sub-transaction of=20
  F1, then after F7(ACK), UAC can discard refreshed dialog state in =
F3.</FONT>=20
  <BR><BR><FONT face=3Dsans-serif size=3D2>And it may make the two parts =
can not=20
  communicate anymore in current dialog.</FONT> <BR><BR><FONT =
face=3Dsans-serif=20
  size=3D2>And IMO, this is a counterevidence for treating all =
UPDATE(during=20
  Re-INVITE) as sub-transaction of Re-INVITE.</FONT> <BR><BR><FONT=20
  face=3Dsans-serif size=3D2>Gao</FONT> <BR><BR><BR><BR><BR>
  <TABLE width=3D"100%">
    <TBODY>
    <TR vAlign=3Dtop>
      <TD width=3D"26%"><FONT face=3Dsans-serif size=3D1><B>"Christer =
Holmberg"=20
        &lt;christer.holmberg@ericsson.com&gt;</B> </FONT>
        <P><FONT face=3Dsans-serif size=3D1>2009-03-12 15:31</FONT> </P>
      <TD width=3D"73%">
        <TABLE width=3D"100%">
          <TBODY>
          <TR vAlign=3Dtop>
            <TD>
              <DIV align=3Dright><FONT face=3Dsans-serif =
size=3D1>=CA=D5=BC=FE=C8=CB</FONT></DIV>
            <TD><FONT face=3Dsans-serif =
size=3D1>&lt;gao.yang2@zte.com.cn&gt;,=20
              "SIP" &lt;sip@ietf.org&gt;, "SIPPING"=20
              &lt;sipping@ietf.org&gt;</FONT>=20
          <TR vAlign=3Dtop>
            <TD>
              <DIV align=3Dright><FONT face=3Dsans-serif =
size=3D1>=B3=AD=CB=CD</FONT></DIV>
            <TD><FONT face=3Dsans-serif size=3D1>"Gonzalo Camarillo"=20
              &lt;gonzalo.camarillo@ericsson.com&gt;</FONT>=20
          <TR vAlign=3Dtop>
            <TD>
              <DIV align=3Dright><FONT face=3Dsans-serif =
size=3D1>=D6=F7=CC=E2</FONT></DIV>
            <TD><FONT face=3Dsans-serif size=3D1>RE: One chart =
&nbsp;//"UPDATE=20
              during Re-INVITE" =
discussion</FONT></TR></TBODY></TABLE><BR>
        <TABLE>
          <TBODY>
          <TR vAlign=3Dtop>
            <TD>
            =
<TD></TR></TBODY></TABLE><BR></TR></TBODY></TABLE><BR><BR><BR><FONT =
face=3DArial=20
  color=3Dblue size=3D2>In my opinion 487 should trigger whatever =
actions/rollback=20
  that any other non-200 response sent by the UAS would.</FONT> =
<BR><FONT=20
  face=3Dsans-serif size=3D3>&nbsp;</FONT> <BR><FONT face=3DArial =
color=3Dblue=20
  size=3D2>Regards,</FONT> <BR><FONT face=3Dsans-serif =
size=3D3>&nbsp;</FONT>=20
  <BR><FONT face=3DArial color=3Dblue size=3D2>Christer</FONT> <BR><BR>
  <HR>
  <FONT face=3DTahoma size=3D2><B>From:</B> gao.yang2@zte.com.cn=20
  [mailto:gao.yang2@zte.com.cn] <B><BR>Sent:</B> 12. maaliskuuta 2009=20
  5:47<B><BR>To:</B> SIP; SIPPING<B><BR>Cc:</B> Christer Holmberg; =
Gonzalo=20
  Camarillo<B><BR>Subject:</B> One chart //"UPDATE during Re-INVITE"=20
  discussion</FONT><FONT face=3Dsans-serif size=3D3><BR></FONT><BR><FONT =

  face=3Dsans-serif size=3D2><BR>&nbsp; &nbsp; &nbsp;UAC &nbsp; &nbsp; =
&nbsp; &nbsp;=20
  &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; UAS</FONT><FONT face=3Dsans-serif =
size=3D3>=20
  </FONT><FONT face=3Dsans-serif size=3D2><BR>&nbsp; &nbsp; &nbsp; | =
session=20
  established |</FONT><FONT face=3Dsans-serif size=3D3> </FONT><FONT =
face=3Dsans-serif=20
  size=3D2><BR>&nbsp; &nbsp; &nbsp; =
|&lt;=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D&gt;|</FONT=
><FONT=20
  face=3Dsans-serif size=3D3> </FONT><FONT face=3Dsans-serif =
size=3D2><BR>&nbsp; &nbsp;=20
  &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
  |</FONT><FONT face=3Dsans-serif size=3D3> </FONT><FONT =
face=3Dsans-serif=20
  size=3D2><BR>&nbsp; &nbsp; &nbsp; | F1 &nbsp;re-INVITE (SDP) =
|</FONT><FONT=20
  face=3Dsans-serif size=3D3> </FONT><FONT face=3Dsans-serif =
size=3D2><BR>&nbsp; &nbsp;=20
  &nbsp; |--------------------&gt;|</FONT><FONT face=3Dsans-serif =
size=3D3>=20
  </FONT><FONT face=3Dsans-serif size=3D2><BR>&nbsp; &nbsp; &nbsp; | F2 =
1xx-rel=20
  (SDP) &nbsp; &nbsp;|</FONT><FONT face=3Dsans-serif size=3D3> =
</FONT><FONT=20
  face=3Dsans-serif size=3D2><BR>&nbsp; &nbsp; &nbsp;=20
  |&lt;--------------------|</FONT><FONT face=3Dsans-serif size=3D3> =
</FONT><FONT=20
  face=3Dsans-serif size=3D2><BR>&nbsp; &nbsp; &nbsp; | F3 &nbsp; UPDATE =
&nbsp;=20
  &nbsp; &nbsp; &nbsp; |</FONT><FONT face=3Dsans-serif size=3D3> =
</FONT><FONT=20
  face=3Dsans-serif size=3D2><BR>&nbsp; &nbsp; &nbsp; =
|&lt;--------------------|=20
  Target refreshing</FONT><FONT face=3Dsans-serif size=3D3> </FONT><FONT =

  face=3Dsans-serif size=3D2><BR>&nbsp; &nbsp; &nbsp; | F4 2xx UPT =
&nbsp; &nbsp;=20
  &nbsp; &nbsp; &nbsp;|</FONT><FONT face=3Dsans-serif size=3D3> =
</FONT><FONT=20
  face=3Dsans-serif size=3D2><BR>&nbsp; &nbsp; &nbsp;=20
  |--------------------&gt;|</FONT><FONT face=3Dsans-serif size=3D3> =
</FONT><FONT=20
  face=3Dsans-serif size=3D2><BR>&nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; =
&nbsp; &nbsp;=20
  &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</FONT><FONT =
face=3Dsans-serif=20
  size=3D3> </FONT><FONT face=3Dsans-serif size=3D2><BR>&nbsp; &nbsp; =
&nbsp; | F5/6=20
  Cancel/200OK &nbsp; |</FONT><FONT face=3Dsans-serif size=3D3> =
</FONT><FONT=20
  face=3Dsans-serif size=3D2><BR>&nbsp; &nbsp; &nbsp;=20
  |--------------------&gt;|</FONT><FONT face=3Dsans-serif size=3D3> =
</FONT><FONT=20
  face=3Dsans-serif size=3D2><BR>&nbsp; &nbsp; &nbsp; | F7 &nbsp; =
&nbsp;487 INV=20
  &nbsp; &nbsp; &nbsp; |</FONT><FONT face=3Dsans-serif size=3D3> =
</FONT><FONT=20
  face=3Dsans-serif size=3D2><BR>&nbsp; &nbsp; &nbsp;=20
  |&lt;--------------------|</FONT><FONT face=3Dsans-serif size=3D3> =
</FONT><FONT=20
  face=3Dsans-serif size=3D2><BR>&nbsp; &nbsp; &nbsp; | F7 &nbsp; &nbsp; =
ACK &nbsp;=20
  &nbsp; &nbsp; &nbsp; &nbsp;|</FONT><FONT face=3Dsans-serif size=3D3> =
</FONT><FONT=20
  face=3Dsans-serif size=3D2><BR>&nbsp; &nbsp; &nbsp;=20
  |--------------------&gt;|</FONT><FONT face=3Dsans-serif size=3D3> =
</FONT><FONT=20
  face=3Dsans-serif size=3D2><BR>&nbsp; &nbsp; &nbsp; | F8 =
&nbsp;re-INVITE (SDP)=20
  |</FONT><FONT face=3Dsans-serif size=3D3> </FONT><FONT =
face=3Dsans-serif=20
  size=3D2><BR>&nbsp; &nbsp; &nbsp; |--------------------&gt;| Using =
which dialog=20
  state</FONT><FONT face=3Dsans-serif size=3D3> <BR></FONT><FONT =
face=3Dsans-serif=20
  size=3D2><BR>F3/F4 is about target refreshing.</FONT><FONT =
face=3Dsans-serif=20
  size=3D3> <BR></FONT><FONT face=3Dsans-serif size=3D2><BR>And if UAC =
sends Cancel=A3=AC=20
  UAS sends 487. The first Re-INVITE is discarded.</FONT><FONT =
face=3Dsans-serif=20
  size=3D3> <BR></FONT><FONT face=3Dsans-serif size=3D2><BR>But if UAC =
sends another=20
  Re-INVITE, which dialog state should it use?</FONT><FONT =
face=3Dsans-serif=20
  size=3D3> <BR></FONT><FONT face=3Dsans-serif size=3D2><BR>I think is =
the one=20
  refreshed by F3. That's the merit of regarding UPDATE/200OK's=20
  atomicity.</FONT><FONT face=3Dsans-serif size=3D3> =
<BR></FONT><BR><FONT=20
  =
size=3D3><TT>--------------------------------------------------------<BR>=
ZTE=20
  Information Security Notice: The information contained in this mail is =
solely=20
  property of the sender's organization. This mail communication is=20
  confidential. Recipients named above are obligated to maintain secrecy =
and are=20
  not permitted to disclose the contents of this communication to=20
  others.<BR>This email and any files transmitted with it are =
confidential and=20
  intended solely for the use of the individual or entity to whom they =
are=20
  addressed. If you have received this email in error please notify the=20
  originator of the message. Any views expressed in this message are =
those of=20
  the individual sender.<BR>This message has been scanned for viruses =
and Spam=20
  by ZTE Anti-Spam =
system.<BR></TT></FONT><BR><BR><PRE>-------------------------------------=
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Date: Thu, 12 Mar 2009 11:52:03 +0000
From: "Elwell, John" <john.elwell@siemens.com>
In-reply-to: <29682BF0-403F-426F-A635-FC3F94BBEA1C@cisco.com>
To: Cullen Jennings <fluffy@cisco.com>
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Thread-Topic: [Sip] Seeking consensus on draft-elwell-sip-e2e-identity-important
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Cc: SIP List <sip@ietf.org>, Jon Peterson <jon.peterson@neustar.biz>
Subject: Re: [Sip] Seeking consensus on draft-elwell-sip-e2e-identity-important
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Cullen,

> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]=20
> Sent: 11 March 2009 23:07
> To: Elwell, John
> Cc: SIP List; Jon Peterson
> Subject: Re: [Sip] Seeking consensus on=20
> draft-elwell-sip-e2e-identity-important
>=20
>=20
> I continue to believe that we need to to come to some common =20
> understanding of the security problem we are trying to solve and the =20
> constraints on that solution. On one of the close to last=20
> phones calls =20
> involving this draft I asked everyone one on the call what they =20
> thought the problem was. I got no common answer.
>=20
> There is a lot of good information in this draft related to the =20
> problem and solutions, but I don't believe this draft gets to what =20
> problem we are trying to solve thus making it very hard to answer a =20
> question like if the solution should involve changing m line or not. =20
> If we all agreed the problem was, just for example, stop MITM from =20
> listening to phone calls while still allowing intermediaries to do =20
> media steering, then I think we could get there.
[JRE] That may be a good way to express it. At present, in section 9, it
expresses the problem in terms of:
"Loss of end-to-end authenticated identification caused by the
      breaking of SIP Identity signatures on non-E.164-based SIP URIs by
      B2BUAs in intermediate domains performing legitimate functions
      such as media steering."
and a slightly different statement for the E.164-based case. Then in
section 10 it tries to state the requirement in terms of a third party,
user 3, which is the MitM in your formulation. So that is not 100 miles
away from your suggestion.

When working on slides for IETF74, I will try to take this into account
and provide a better problems statement. Other input welcome.

John


> I know the=20
> problem is =20
> much more complicated than I just said - I'm not suggesting that my =20
> above statement is the right problem statement but just that it is =20
> closer to level we should be looking at the problem at.
>=20
> Cullen
>=20
>=20
> On Mar 3, 2009, at 9:46 AM, Elwell, John wrote:
>=20
> > Draft 03:
> >=20
> http://tools.ietf.org/html/draft-elwell-sip-e2e-identity-important-03
> > includes for the first time an analysis of changes to SIP=20
> requests by
> > B2BUAs (section 8). Also the Conclusions (section 9) have been =20
> > extended.
> >
> > I and a number of people who have assisted in revising this draft =20
> > would
> > like to seek consensus on a number of the points. We aim to=20
> ask these
> > questions in San Francisco, but obviously it would help to get =20
> > opinions
> > from the list in advance.
> >
> > 1. Tolerance to changes of certain elements as a SIP request is
> > forwarded through intermediate domains.
> >
> > Section 8 analyses a number of elements of SIP messages that are =20
> > signed
> > by the present RFC 4474 solution to end-to-end (or near end-to-end)
> > authenticated identification. For some elements it claims that any
> > solution for end-to-end authenticated identification needs to be
> > tolerant of changes by intermediate domains, since there=20
> appear to be
> > valid reasons why B2BUAs (including but not limited to SBCs) change
> > these elements. Picking on some of these:
> >
> > A) Is there agreement that a solution needs to be tolerant of =20
> > changes to
> > IP addresses and ports in c- and m-lines in SDP? In other=20
> words, that
> > intermediate domains can legitimately modify these fields (e.g., for
> > media steering) and a solution is needed for end-to-end=20
> authenticated
> > identification in such circumstances?
> >
> > B) Is there agreement that a solution needs to be tolerant of =20
> > changes to
> > Contact (addr-spec), Call-Id (call-id) and CSeq (digit)=20
> header fields?
> > In other words, that intermediate domains can legitimately modify =20
> > these
> > fields (e.g., for topology hiding) and a solution is needed for
> > end-to-end authenticated identification in such circumstances?
> >
> > C) Is there agreement that a solution does not need to be=20
> tolerant of
> > changes to the addr-spec of From and To header fields? In=20
> other words,
> > that intermediate domains should not modify these fields and can =20
> > expect
> > to break end-to-end authenticated identification if they do so?
> >
> >
> > 2. Are people in broad agreement with the conclusions in section 9
> > concerning the importance of end-to-end identification and in =20
> > particular
> > end-to-end authenticated identification?
> >
> > John
> >
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
>=20
>=20

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From: Brett Tate <brett@broadsoft.com>
To: "gao.yang2@zte.com.cn" <gao.yang2@zte.com.cn>, SIP <sip@ietf.org>, SIPPING <sipping@ietf.org>
Date: Thu, 12 Mar 2009 04:57:06 -0700
Thread-Topic: [Sip] One chart  //"UPDATE during Re-INVITE" discussion
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Data (like Contact, Session-Expires, and etcetera) that was successfully up=
dated by F3/F4 still apply after F7.  Thus F8 data should reflect (and can =
attempt to modify) adjustments made during F3/F4.


From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of gao.y=
ang2@zte.com.cn
Sent: Wednesday, March 11, 2009 11:47 PM
To: SIP; SIPPING
Cc: Gonzalo Camarillo; Christer Holmberg
Subject: [Sip] One chart //"UPDATE during Re-INVITE" discussion


      UAC                   UAS
       | session established |
       |<=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D>|
       |                     |
       | F1  re-INVITE (SDP) |
       |-------------------->|
       | F2 1xx-rel (SDP)    |
       |<--------------------|
       | F3   UPDATE         |
       |<--------------------| Target refreshing
       | F4 2xx UPT          |
       |-------------------->|
       |                     |
       | F5/6 Cancel/200OK   |
       |-------------------->|
       | F7    487 INV       |
       |<--------------------|
       | F7     ACK          |
       |-------------------->|
       | F8  re-INVITE (SDP) |
       |-------------------->| Using which dialog state

F3/F4 is about target refreshing.

And if UAC sends Cancel=1B$B!$=1B(B UAS sends 487. The first Re-INVITE is d=
iscarded.

But if UAC sends another Re-INVITE, which dialog state should it use?

I think is the one refreshed by F3. That's the merit of regarding UPDATE/20=
0OK's atomicity.



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<div class=3DSection1>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew";
color:#1F497D'>Data (like Contact, Session-Expires, and etcetera) that was =
successfully
updated by F3/F4 still apply after F7.&nbsp; Thus F8 data should reflect (a=
nd can
attempt to modify) adjustments made during F3/F4.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in'>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma=
","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> sip-bounces@i=
etf.org
[mailto:sip-bounces@ietf.org] <b>On Behalf Of </b>gao.yang2@zte.com.cn<br>
<b>Sent:</b> Wednesday, March 11, 2009 11:47 PM<br>
<b>To:</b> SIP; SIPPING<br>
<b>Cc:</b> Gonzalo Camarillo; Christer Holmberg<br>
<b>Subject:</b> [Sip] One chart //&quot;UPDATE during Re-INVITE&quot;
discussion<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; UAC &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; U=
AS</span>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; &nbsp;| session established |</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; &nbsp;|&lt;=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
&gt;|</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p;
&nbsp; |</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; &nbsp;| F1 &nbsp;re-INVITE (SDP) |</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; &nbsp;|--------------------&gt;|</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; &nbsp;| F2 1xx-rel (SDP) &nbsp; &nbsp;|</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; &nbsp;|&lt;--------------------|</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; &nbsp;| F3 &nbsp; UPDATE &nbsp; &nbsp; &nbsp; &nbsp; |</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; &nbsp;|&lt;--------------------| Target refreshing</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; &nbsp;| F4 2xx UPT &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; &nbsp;|--------------------&gt;|</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p;
&nbsp; |</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; &nbsp;| F5/6 Cancel/200OK &nbsp; |</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; &nbsp;|--------------------&gt;|</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; &nbsp;| F7 &nbsp; &nbsp;487 INV &nbsp; &nbsp; &nbsp; |</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; &nbsp;|&lt;--------------------|</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; &nbsp;| F7 &nbsp; &nbsp; ACK &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|</sp=
an> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; &nbsp;|--------------------&gt;|</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; &nbsp;| F8 &nbsp;re-INVITE (SDP) |</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp; &n=
bsp;
&nbsp; &nbsp;|--------------------&gt;| Using which dialog state</span> <br=
>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>F3/F4 is =
about
target refreshing.</span> <br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>And if UA=
C sends
Cancel</span><span lang=3DZH-CN style=3D'font-size:10.0pt'>=1B$B!$=1B(J</sp=
an><span
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'> UAS sends 487.=
 The
first Re-INVITE is discarded.</span> <br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>But if UA=
C
sends another Re-INVITE, which dialog state should it use?</span> <br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>I think i=
s the
one refreshed by F3. That's the merit of regarding UPDATE/200OK's atomicity=
.</span>
<o:p></o:p></p>

<pre><o:p>&nbsp;</o:p></pre></div>

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Date: Thu, 12 Mar 2009 13:56:35 +0100
Message-ID: <9ae56b1e0903120556t73208627v39b3c6221098cc44@mail.gmail.com>
From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: "Elwell, John" <john.elwell@siemens.com>
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Cc: sip@ietf.org
Subject: Re: [Sip] Terminology (was RE: Fwd: I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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But that does not solve the freephone service use case. Freephone represents
a use case where the R-URI is replaced with an AoR of user B - not with the
contact of user B.

A calls Bfreephonenumber routed the B (AOR) routed to B's contact.

The target-uri draft is interested in obtaining: Bfreephonenumber as that is
what A used to reach B.

So what we like to see that B UA receives is something like
    H-I:  Bfreephone; "istarget",\
        B,\
        contact-B

So you want to get Bfreephonenumber and not B.

In certain situations you are also interested in B, I guess that  PBX's
would interested in that to deliver to the correct user. We could
distinguish them like:
    H-I:  Bfreephone; "istarget",\
        B,"aor"\
        contact-B

BR,
/Hans Erik van Elburg


On Thu, Mar 12, 2009 at 12:08 PM, Elwell, John <john.elwell@siemens.com>wrote:

> If 'istarget' is used only to identify the value that was overwritten in
> the Request-URI by a contact URI, an alternative approach would be to
> flag the contact URI H-I entry as 'iscontact'. Then the UAS would just
> need to look for the most recent H-I entry that is not marked
> 'iscontact'.
>
> John
>
>
> ________________________________
>
>        From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> Behalf Of Hans Erik van Elburg
>        Sent: 12 March 2009 08:35
>        To: Mary Barnes
>        Cc: sip@ietf.org
>        Subject: Re: [Sip] Terminology (was RE: Fwd:
> I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>
>
>        Yes, because you are using the 3261 use of target and the
> 4244bis introduced definition of retarget. I thought it was clear that
> we need other words as those definitions don't match the target-uri
> drafts use of the terms. Also they do not suffice to provide a solution
> for the use cases in the target-uri draft.
>
>        The 3261 text you refer to is exactly about the case where the
> home proxy overwrites the Request-URI with a new target. This target is
> teh registered contact address. And hence this would be what target-uri
> calls a hop or a route. This case and this is where it gets confusing is
> not a "retarget" in the target-uri draft use of the term.
>
>        The target-uri draft states:
>
>           "To avoid confusion, we
>           refer to a SIP URI that is an address for a user or resource
> as a
>           "target" and a SIP URI that is a hop for reaching that user
> as a
>
>           "hop".
>
>
>        Apparently that does not suffice to avoid confusion.
>
>        As for the tagging, speaking about the solution before agreeing
> on the terminology and the problem it should solve is meaningless.
>
>        /Hans Erik van Elburg
>
>
>
>        On Thu, Mar 12, 2009 at 2:33 AM, Mary Barnes
> <mary.barnes@nortel.com> wrote:
>
>
>                Responses below [MB].
>
>
>                -----Original Message-----
>                From: Hans Erik van Elburg
> [mailto:ietf.hanserik@gmail.com]
>
>                Sent: Wednesday, March 11, 2009 6:42 PM
>                To: Barnes, Mary (RICH2:AR00)
>                Cc: Shida Schubert; sip@ietf.org
>                Subject: Re: Terminology (was RE: [Sip] Fwd: I-D
>                ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>
>
>                I was talking about the concept.
>
>                The use cases only describe  cases where the target-uri
> (lets call that
>                current target from now) has been lost when an initial
> request for a
>                dialog or standalone request arrives at the UAS.
>
>                [MB] And I think this is where the terminology confusion
> starts - I
>                don't think of the "lost" target-uri as being the
> "current" target. In
>                my mind, the "current" target is reflected by the last
> hi-entry and the
>                request-uri in the incoming request at the UAS. If the
> entity that sent
>                the request was the entity that added the last hi-entry,
> then the uri in
>                that hi-entry is the same as the request-URI in the SIP
> request that
>                arrives at the UAS. I refer to the "lost" uri as the one
> that was
>                "retargeted" - that's the one the UAS wants to pull from
> the hi-entries
>                in the incoming request. That hi-entry was not the one
> that was just
>                added by the entity that built the request just received
> by the UAS.
>                That hi-entry is tagged with whatever name we are going
> to tag it with
>                BEFORE the request is forwarded (using the term forward
> per section 16.6
>                of RFC 3261).  That tag is added once the target set of
> potential
>                candidates for the new request uri are determined in
> section 16.5 of
>                3261 (with "target set" being a 3261 term), just before
> the request is
>                forwarded in section 16.6 to one of those targets.  A
> new hi-entry
>                (which will be the last hi-entry in the request received
> by the UAS) is
>                added in section 16.6 of 3261 as the request is
> forwarded.  At this
>                point in time, the lost information is in the previous
> hi-entry when the
>                outgoing request is sent. [/MB]
>
>
>                What has been lost is the current target of the request:
>
>                [MB] Right - at which point in my mind, it's no longer
> the current
>                target ;) Maybe we call it "lost". [/MB]
>
>
>
>                     Current target
>
>                  The current target of an initial request for a dialog
> or standalone
>                  request is the name or address to which the request is
> targeted, i.e.
>                  either the initial target inserted in the Request-URI
> by the UAC that
>                  originates the request, or when a retarget occurred,
> the target
>                  provided in that retarget operation.  Reroute and
> translation
>                  operations never change the current target.
>
>
>                [MB] I don't think this definition fits what you want -
> i.e., if there
>                is no retargeting, then none of the hi-entries are
> tagged - i.e., you
>                won't have your concept of current. The way it works is
> that if no
>                hi-entries are tagged, then you know that there was no
> retargeting, thus
>                you know the request-uri has not been lost. [/MB]
>
>
>                This defintion only makes sense when the following
> definitions are used:
>
>                Name:
>
>                  A name is a moniker for an entity which refers to it
> in a way which
>
>                  reveals nothing about where it is in a network.  In
> SIP, tel URI
>
>                  which doesn't represent the location of the entity is
> a name.
>
>
>                Address:
>
>                  An address is an identifier for an entity which
> describes it by its
>
>                  location on the network.  In SIP, the SIP URI itself
> is a form of
>
>                  address because the host part of the URI, the only
> mandatory part of
>
>                  the URI besides the scheme itself, indicates the
> location of a SIP
>
>                  server that can be used to handle the request.
>
>                Route:
>
>                  Finally, a route is a   sequence of SIP entities
> (including the UA
>                itself!) which are
>
>                 traversed in order to forward a request to an address
> or name.
>
>                Retarget (other term might be needed, as this is highly
> confusing):
>
>                   A Request-URI rewrite operation that changes the
> target identity of
>                the request.
>
>                Reroute (other term might be needed):
>
>                   A Request-URI rewrite operation that does not change
> the current
>                target of the request, but     determines the route/next
> hop taken to
>                reach the target-identity.
>
>                /Hans Erik
>
>                Mary Barnes wrote:
>                > So, by "target-uri concept" are you referring to the
> solution in the
>                > current target-uri draft or the application usage as
> such in the
>                > target-uri document (which is what I was referring
> to)?
>                >
>                > Yes, the problem is that SIP doesn't differentiate the
> various
>                > mechanisms by which a request-URI may be changed,
> which is what some
>                > of the updated text in 4244bis is trying to do and why
> we need precise
>
>                > terms. I think we all agree that.
>                >
>                > I honestly don't care what we call the tag, as long as
> we're clear
>                > about the functionality.
>                >
>                > Mary.
>                >
>                > -----Original Message-----
>                > From: Hans Erik van Elburg
> [mailto:ietf.hanserik@gmail.com]
>                > Sent: Wednesday, March 11, 2009 4:40 PM
>                > To: Barnes, Mary (RICH2:AR00)
>                > Cc: Shida Schubert; sip@ietf.org
>                > Subject: Re: Terminology (was RE: [Sip] Fwd: I-D
>                > ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>                >
>                > The target-uri concept when defined properly is
> entirely application
>                > agnostic.
>                >
>                > When one uses History-Info or another header as a
> vehicle for this
>                > information then it still can be considered
> application agnostic as
>                > any application can use this information as it likes.
>                >
>                >  From SIP perspective all the Request-URI rewrites
> look the same, that
>
>                > is what brought us discussing this in the first place.
>                >
>                > /Hans Erik
>                >
>                > Mary Barnes wrote:
>                >
>                >> And, I think the big issue with the terminology is
> that 4244bis is
>                >> application agnostic, so it is tagging what happens
> to the URIs
>                >> (why/how the specific URI was overwritten).  Whereas,
> specific
>                >> applications can derive information, such as what
> "is" the real
>                >> "target" for the request, from the History-Info
> header.   So, IMHO,
>                >> it's a matter of the target-uri document specifying
> that the last
>                >> entry tagged by whatever name we come up with "is"
> the real "target"
>                >> for the request.  I believe it's up to the
> applications to describe
>                >> how they make use of the History-Info and not for the
> History-Info to
>
>                >> describe how applications can use it.  History-Info
> purely reflects
>                >> what happened from a SIP Protocol perspective and
> should not imply
>                >> any
>                >>
>                >
>                >
>                >> application semantics. Indeed the intent of section 5
> of 4244 was to
>                >> inform the applications how they should decribe their
> usage of the
>                >> header.
>                >>
>                >> Mary.
>                >> Note: I changed the subject to hopefully make this
> topic easier to
>                >> keep track of.
>                >>
>                >>
> ---------------------------------------------------------------------
>                >> -
>                >> --
>                >> *From:* Hans Erik van Elburg
> [mailto:ietf.hanserik@gmail.com]
>                >> *Sent:* Wednesday, March 11, 2009 3:59 AM
>                >> *To:* Shida Schubert
>                >> *Cc:* sip@ietf.org List; Barnes, Mary (RICH2:AR00)
>                >> *Subject:* Re: [Sip] Fwd: I-D
>                >> ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>                >>
>                >> One of the problems that made the discussion quite
> tedious is the
>                >> complete inversed meaning of the terms "retarget" and
> "reroute"
>                >> between 4244bis and target-uri factions have.
>                >>
>                >> Conclusion was that the terminology need to be
> properly defined and
>                >> some attempt was made to start such activity.
>                >>
>                >> On 3. I have strong concerns when we start tagging
> the URI as to how
>                >> they come about "retarget"/"mapped conficuration"
> etc. It is better
>                >> to
>                >>
>                >
>                >
>                >> embellish them with a tag that just represents there
> meaning for
>                >> example "istarget" or "hop". For the following
> reasons:
>                >> 1. It is much more intuitive for the user of this
> information, its
>                >> meaning can basically be guessed.
>                >> 2. Such meaning also probably survives application in
> other problem
>                >> spaces that we had not foreseen when introducing the
> concept.
>                >>
>                >> The solution that is  described now in the target-URI
> delivery draft
>                >> is not yet complete, it does not solve the freephone
> use case for
>                >> example and in its current form it will deliver
> exactly the same
>                >> target-URI as that which would be delivered by the
> P-Called-Party-ID
>
>                >> header. Contrary to what the draft says. So some work
> is needed
>                still.
>                >>
>                >> /Hans Erik van Elburg
>                >>
>                >>
>                >> On Wed, Mar 11, 2009 at 7:13 AM, Shida Schubert
> <shida@ntt-at.com
>                >> <mailto:shida@ntt-at.com>> wrote:
>                >>
>                >>
>                >>     We have submitted the updated target-uri draft
> based on the
>                >>
>                > comments
>                >
>                >>     submitted to the list and comments received at
> IETF73.
>                >>
>                >>     I have taken over as editor as Jonathan didn't
> have the cycles to
>
>                >>     update the draft, with Francois, Christer and
> Hans Erick as
>                >>     additional
>                >>     co-authors and great deal of help from Mary.
>                >>
>                >>     The following summarizes the changes made to the
> target-uri
>                >>
>                > document
>                >
>                >>     1. Added use-case for toll-free number back
>                >>     2. Added definition of "retarget" operation.
>                >>     3. Removed a reference to URN
>                >>     4. Added a text discussing the difference to
> P-Called-Party-Id
>                >>     5. Changed parameter name from "target" to
> "istarget"
>                >>
>                >>     Note, that the target-uri document still contains
> the normative
>                >>     text for the
>                >>     History-Info header.
>                >>
>                >>     In addition, Mary (with Francois as co-author)
> has submitted a
>                >>     rfc4244bis, with the following changes:
>                >>     1. Incorporated the normative aspects of the
> target-uri document
>                >>     into the existing normative text in RFC 4244 -
> the functionality
>                >>
>                > is
>                >
>                >>     virtually identical (as is some of the text) as
> the HI based
>                >>
>                > solution
>                >
>                >>     described in the target-uri document.  It's
> important that the
>                >>     solution
>                >>     be integrated into RFC 4244 as it MUST work and
> be based on the
>                >>     normative
>                >>     aspects of RFC 4244.
>                >>     2.  Added the use cases from target-uri the the
> summary
>                >>     in the overview of rfc4244bis.
>                >>     3. Added an additional requirement to capture the
> "target-uri"
>                >>     information.
>                >>     4. Fixed an error in the RFC 4244 ABNF and added
> "retarget"
>                >>
>                > parameter.
>                >
>                >>     5. Added a more simplified example.
>                >>
>                >>
>                >>     We had some very long offline exchanges as to the
> best way
>                >> forward
>                >>
>                > and
>                >
>                >>     remaining work for both documents.
>                >>
>                >>     Some of the issues identified are:
>                >>
>                >>     ::Issues::
>                >>        1. Should we remove the normative text from
> target-uri and
>                >>
>                > progress
>                >
>                >>            4244bis along with the target-uri document
> to meet the
>                >>     chartered
>                >>           SIP WG milestone?
>                >>
>                >>
>                >>        2. Name of the parameter.
>                >>           At the last meeting, parameter "target" was
> said
>                >>
>                > inappropriate
>                >
>                >>           because voicemail-uri spec already defines
> a parameter
>                >>
>                > called
>                >
>                >>           "target" which also can be found in
> hi-entry, thus
>                >>
>                > potentially
>                >
>                >>           causing confusion.
>                >>
>                >>           Currently the target-uri draft uses
> "istarget" and 4244bis
>                >>
>                > uses
>                >
>                >>           "retarget"  but we could never come to
>                >>           a consensus on what name is appropriate.
> Other suggestions
>                >>
>                > have
>                >
>                >>           included the following:
>                >>           "target-identity" (someone didn't like that
> "identity" is
>                >>     also a SIP header)
>                >>           "reg-uri"  (can be paired with "mapped-uri"
> for item 3
>                >>
>                > below)
>                >
>                >>           "aor"
>                >>           "jibberish"
>                >>           etc.
>                >>
>                >>           One reason this is so difficult relates to
> the problem
>                >>     statement in target-uri in that
>                >>           RFC 3261 doesn't differentiate the
> mechanism by which the
>                >>
>                > new
>                >
>                >>           (target) Request-URI is selected.  Another
> issue is that
>                >>     some of the terminology in
>                >>           RFC 3261 is overloaded - e.g., "forwarding"
> refers both to
>                a
>                >>     Proxy
>                >>           which does not have responsibility for the
> domain of the
>                >>     request-URI
>                >>           in the incoming request, thus the proxy
> just "forwards" the
>                >>     request to
>                >>           the next hop AND "forwarding" is used to
> describe the
>                >>     process whereby
>                >>           the outgoing request is built and
> "forwarded" to the next
>                >>     hop at which
>                >>           point the proxy does not know how the new
> request-uri was
>                >>     selected.
>                >>           RFC 4244 has attempted to clarify the terms
> and attempts to
>                >>     use "forward"
>                >>           in the context of the former situation and
> "retarget" for
>                >>     the case whereby
>                >>           a proxy is responsible for the domain and
> thus can use a
>                >>     number of
>                >>           mechanism to select the new target for the
> request - e.g.,
>                a
>                >>     REGISTRAR,
>                >>           configured data, etc.
>                >>
>                >>      3.  Related to the last point in item 2 above,
> it has been
>                >>     proposed that
>                >>           we differentiate the hi-entries even more
> by defining
>                >>     separate parameters
>                >>          for registered and configured/mapped
> contacts.
>                >>          Currently when the R-URI is translated to a
> URI which is
>                >>     either derived
>                >>          from location service lookup(registered by
> UA) or from
>                >>
>                > mapping
>                >
>                >>          table, there is no differentiation as to how
> the URI was
>                >>     derived once it is
>                >>          added to the list of potential targets.
>                >>
>                >>          The general consensus of the authors of the
> two documents
>                was
>                >>     that it may
>                >>          be useful for some services to have the
> hi-entries tagged
>                >>     with the
>                >>          more specific information.
>                >>
>                >>          And, of course, this gets us into another
> naming contest. In
>                >>     the end, the naming
>                >>          is not so important as long as the term
> isn't too overloaded
>                >>     and it is defined
>                >>          precisely in the document(s).
>                >>
>                >>     We would appreciate WG feedback on these issues
> and any other
>                >>     comments on
>                >>     the two documents prior to IETF-74.
>                >>
>                >>     Regards,
>                >>     Shida and Mary.
>                >>
>                >>
>                >>
>                >>     Begin forwarded message:
>                >>
>                >>>     *From: *Internet-Drafts@ietf.org
>                >>>
>                > <mailto:Internet-Drafts@ietf.org>
>                >
>                >>>     *Date: *March 10, 2009 2:30:01 AM JST
>                >>>     *To: *i-d-announce@ietf.org
> <mailto:i-d-announce@ietf.org>
>                >>>     *Subject: **I-D
>                >>>
> ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt *
>                >>>     *Reply-To: *internet-drafts@ietf.org
>                >>>     <mailto:internet-drafts@ietf.org>
>                >>>
>                >>>     A New Internet-Draft is available from the
> on-line
>                >>>
>                > Internet-Drafts
>                >
>                >>>     directories.
>                >>>
>                >>>
>                >>>     Title
>                >>>     : Delivery of Request-URI Targets to User Agents
>                >>>     Author(s)
>                >>>     : J. Rosenberg, H. van Elburg, C. Holmberg, F.
> Audet, S.
>                Schubert
>                >>>     Filename :
> draft-rosenberg-sip-target-uri-delivery-01.txt
>                >>>     Pages
>                >>>     : 16
>                >>>     Date
>                >>>     : 2009-3-9
>                >>>
>                >>>     When a Session Initiation Protocol (SIP) proxy
> receives a
>                request
>                >>>       targeted at a URI identifying a user or
> resource it is
>                >>>
>                > responsible
>                >
>                >>>       for, the proxy translates the URI to a
> registered or
>                configured
>                >>>       contact URI of an agent representing that user
> or resource.
>                >>> In
>                >>>
>                > the
>                >
>                >>>       process, the original URI is removed from the
> request.
>                >>>      Numerous use
>                >>>       cases have arisen which require this
> information to be
>                >>>
>                > delivered to
>                >
>                >>>       the user agent.  This document describes these
> use cases and
>                >>>     defines
>                >>>       an extension to the History-Info header field
> which allows it
>                >>>
>                > to be
>                >
>                >>>       used to support those cases.
>                >>>
>                >>>     A URL for this Internet-Draft is:
>                >>>
>                >>>
> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri-d
>                >>> e
>                >>> livery-01.txt
>                >>>
>                >>>     Internet-Drafts are also available by anonymous
> FTP at:
>                >>>     ftp://ftp.ietf.org/internet-drafts/
>                >>>
>                >>>     Below is the data which will enable a MIME
> compliant mail reader
>                >>>     implementation to automatically retrieve the
> ASCII version of
>                the
>                >>>     Internet-Draft.
>                >>>
>                >>     Content-Type: text/plain<BR>Content-ID:
>                >>     &lt;2009-3-9102756.I-D@ietf.org<lt%3B2009-3-9102756.I-D@ietf.org>
> <mailto:lt%3B2009-3-9102756.I-D@ietf.org<lt%253B2009-3-9102756.I-D@ietf.org>
> >
>                 >>     <mailto:2009-3-9102756.I-D@ietf.org>&gt;<BR><BR>
>                >>
>                >>>     _______________________________________________
>                >>>     I-D-Announce mailing list
>                >>>     I-D-Announce@ietf.org
> <mailto:I-D-Announce@ietf.org>
>                >>>
> https://www.ietf.org/mailman/listinfo/i-d-announce
>                >>>     Internet-Draft directories:
> http://www.ietf.org/shadow.html
>                >>>     or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>                >>>
>                >>     _______________________________________________
>                >>     Sip mailing list
> https://www.ietf.org/mailman/listinfo/sip
>                >>     This list is for NEW development of the core SIP
> Protocol
>                >>     Use sip-implementors@cs.columbia.edu
>                >>     <mailto:sip-implementors@cs.columbia.edu> for
> questions on
>                >> current
>                >>
>                > sip
>                >
>                >>     Use sipping@ietf.org <mailto:sipping@ietf.org>
> for new
>                >>     developments on the application of sip
>                >>
>                >>
>                >>
>
>
>
>

--0015174be732a3eec90464eb8327
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Content-Transfer-Encoding: quoted-printable

But that does not solve the freephone service use case. <style type=3D"text=
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<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DUTF-8">Fr=
eephone represents a use case where the R-URI is replaced with an AoR of us=
er B - not with the contact of user B. <br><br>A calls Bfreephonenumber rou=
ted the B (AOR) routed to B&#39;s contact.<br>

<br>The target-uri draft is interested in obtaining: Bfreephonenumber as th=
at is what A used to reach B.<br><br>So what we like to see that B UA recei=
ves is something like<br>=A0=A0=A0 H-I:=A0 Bfreephone; &quot;istarget&quot;=
,\<br>

=A0=A0=A0 =A0=A0=A0 B,\<br>=A0=A0=A0 =A0=A0=A0 contact-B<br><br>So you want=
 to get Bfreephonenumber and not B.<br><br>In
certain situations you are also interested in B, I guess that=A0 PBX&#39;s
would interested in that to deliver to the correct user. We could
distinguish them like:<br>
=A0=A0=A0 H-I:=A0 Bfreephone; &quot;istarget&quot;,\<br>
=A0=A0=A0 =A0=A0=A0 B,&quot;aor&quot;\<br>
=A0=A0=A0 =A0=A0=A0 contact-B<br>
<br>BR,<br clear=3D"all">/Hans Erik van Elburg<br>
<br><br><div class=3D"gmail_quote">On Thu, Mar 12, 2009 at 12:08 PM, Elwell=
, John <span dir=3D"ltr">&lt;<a href=3D"mailto:john.elwell@siemens.com">joh=
n.elwell@siemens.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_qu=
ote" style=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0p=
t 0.8ex; padding-left: 1ex;">
If &#39;istarget&#39; is used only to identify the value that was overwritt=
en in<br>
the Request-URI by a contact URI, an alternative approach would be to<br>
flag the contact URI H-I entry as &#39;iscontact&#39;. Then the UAS would j=
ust<br>
need to look for the most recent H-I entry that is not marked<br>
&#39;iscontact&#39;.<br>
<br>
John<br>
<br>
<br>
________________________________<br>
<br>
 =A0 =A0 =A0 =A0From: <a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@i=
etf.org</a> [mailto:<a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@iet=
f.org</a>] On<br>
Behalf Of Hans Erik van Elburg<br>
 =A0 =A0 =A0 =A0Sent: 12 March 2009 08:35<br>
 =A0 =A0 =A0 =A0To: Mary Barnes<br>
 =A0 =A0 =A0 =A0Cc: <a href=3D"mailto:sip@ietf.org">sip@ietf.org</a><br>
 =A0 =A0 =A0 =A0Subject: Re: [Sip] Terminology (was RE: Fwd:<br>
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br>
<div><div></div><div class=3D"h5"><br>
<br>
 =A0 =A0 =A0 =A0Yes, because you are using the 3261 use of target and the<b=
r>
4244bis introduced definition of retarget. I thought it was clear that<br>
we need other words as those definitions don&#39;t match the target-uri<br>
drafts use of the terms. Also they do not suffice to provide a solution<br>
for the use cases in the target-uri draft.<br>
<br>
 =A0 =A0 =A0 =A0The 3261 text you refer to is exactly about the case where =
the<br>
home proxy overwrites the Request-URI with a new target. This target is<br>
teh registered contact address. And hence this would be what target-uri<br>
calls a hop or a route. This case and this is where it gets confusing is<br=
>
not a &quot;retarget&quot; in the target-uri draft use of the term.<br>
<br>
 =A0 =A0 =A0 =A0The target-uri draft states:<br>
<br>
 =A0 =A0 =A0 =A0 =A0 &quot;To avoid confusion, we<br>
 =A0 =A0 =A0 =A0 =A0 refer to a SIP URI that is an address for a user or re=
source<br>
as a<br>
 =A0 =A0 =A0 =A0 =A0 &quot;target&quot; and a SIP URI that is a hop for rea=
ching that user<br>
as a<br>
<br>
 =A0 =A0 =A0 =A0 =A0 &quot;hop&quot;.<br>
<br>
<br>
 =A0 =A0 =A0 =A0Apparently that does not suffice to avoid confusion.<br>
<br>
 =A0 =A0 =A0 =A0As for the tagging, speaking about the solution before agre=
eing<br>
on the terminology and the problem it should solve is meaningless.<br>
<br>
 =A0 =A0 =A0 =A0/Hans Erik van Elburg<br>
<br>
<br>
<br>
 =A0 =A0 =A0 =A0On Thu, Mar 12, 2009 at 2:33 AM, Mary Barnes<br>
&lt;<a href=3D"mailto:mary.barnes@nortel.com">mary.barnes@nortel.com</a>&gt=
; wrote:<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Responses below [MB].<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0-----Original Message-----<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0From: Hans Erik van Elburg<br>
[mailto:<a href=3D"mailto:ietf.hanserik@gmail.com">ietf.hanserik@gmail.com<=
/a>]<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Sent: Wednesday, March 11, 2009 6:42 PM<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0To: Barnes, Mary (RICH2:AR00)<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Cc: Shida Schubert; <a href=3D"mailto:sip@i=
etf.org">sip@ietf.org</a><br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Subject: Re: Terminology (was RE: [Sip] Fwd=
: I-D<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0ACTION:draft-rosenberg-sip-target-uri-deliv=
ery-01.txt<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0I was talking about the concept.<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0The use cases only describe =A0cases where =
the target-uri<br>
(lets call that<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0current target from now) has been lost when=
 an initial<br>
request for a<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0dialog or standalone request arrives at the=
 UAS.<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0[MB] And I think this is where the terminol=
ogy confusion<br>
starts - I<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0don&#39;t think of the &quot;lost&quot; tar=
get-uri as being the<br>
&quot;current&quot; target. In<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0my mind, the &quot;current&quot; target is =
reflected by the last<br>
hi-entry and the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0request-uri in the incoming request at the =
UAS. If the<br>
entity that sent<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the request was the entity that added the l=
ast hi-entry,<br>
then the uri in<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0that hi-entry is the same as the request-UR=
I in the SIP<br>
request that<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0arrives at the UAS. I refer to the &quot;lo=
st&quot; uri as the one<br>
that was<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;retargeted&quot; - that&#39;s the one=
 the UAS wants to pull from<br>
the hi-entries<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0in the incoming request. That hi-entry was =
not the one<br>
that was just<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0added by the entity that built the request =
just received<br>
by the UAS.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0That hi-entry is tagged with whatever name =
we are going<br>
to tag it with<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0BEFORE the request is forwarded (using the =
term forward<br>
per section 16.6<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0of RFC 3261). =A0That tag is added once the=
 target set of<br>
potential<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0candidates for the new request uri are dete=
rmined in<br>
section 16.5 of<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A03261 (with &quot;target set&quot; being a 3=
261 term), just before<br>
the request is<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0forwarded in section 16.6 to one of those t=
argets. =A0A<br>
new hi-entry<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0(which will be the last hi-entry in the req=
uest received<br>
by the UAS) is<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0added in section 16.6 of 3261 as the reques=
t is<br>
forwarded. =A0At this<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0point in time, the lost information is in t=
he previous<br>
hi-entry when the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0outgoing request is sent. [/MB]<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0What has been lost is the current target of=
 the request:<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0[MB] Right - at which point in my mind, it&=
#39;s no longer<br>
the current<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0target ;) Maybe we call it &quot;lost&quot;=
. [/MB]<br>
<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Current target<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0The current target of an initial reques=
t for a dialog<br>
or standalone<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0request is the name or address to which=
 the request is<br>
targeted, i.e.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0either the initial target inserted in t=
he Request-URI<br>
by the UAC that<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0originates the request, or when a retar=
get occurred,<br>
the target<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0provided in that retarget operation. =
=A0Reroute and<br>
translation<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0operations never change the current tar=
get.<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0[MB] I don&#39;t think this definition fits=
 what you want -<br>
i.e., if there<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0is no retargeting, then none of the hi-entr=
ies are<br>
tagged - i.e., you<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0won&#39;t have your concept of current. The=
 way it works is<br>
that if no<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0hi-entries are tagged, then you know that t=
here was no<br>
retargeting, thus<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0you know the request-uri has not been lost.=
 [/MB]<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0This defintion only makes sense when the fo=
llowing<br>
definitions are used:<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Name:<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0A name is a moniker for an entity which=
 refers to it<br>
in a way which<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0reveals nothing about where it is in a =
network. =A0In<br>
SIP, tel URI<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0which doesn&#39;t represent the locatio=
n of the entity is<br>
a name.<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Address:<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0An address is an identifier for an enti=
ty which<br>
describes it by its<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0location on the network. =A0In SIP, the=
 SIP URI itself<br>
is a form of<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0address because the host part of the UR=
I, the only<br>
mandatory part of<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the URI besides the scheme itself, indi=
cates the<br>
location of a SIP<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0server that can be used to handle the r=
equest.<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Route:<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Finally, a route is a =A0 sequence of S=
IP entities<br>
(including the UA<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0itself!) which are<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 traversed in order to forward a request to=
 an address<br>
or name.<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Retarget (other term might be needed, as th=
is is highly<br>
confusing):<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 A Request-URI rewrite operation that c=
hanges the<br>
target identity of<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the request.<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Reroute (other term might be needed):<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 A Request-URI rewrite operation that d=
oes not change<br>
the current<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0target of the request, but =A0 =A0 determin=
es the route/next<br>
hop taken to<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0reach the target-identity.<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0/Hans Erik<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Mary Barnes wrote:<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; So, by &quot;target-uri concept&quot; =
are you referring to the<br>
solution in the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; current target-uri draft or the applic=
ation usage as<br>
such in the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; target-uri document (which is what I w=
as referring<br>
to)?<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; Yes, the problem is that SIP doesn&#39=
;t differentiate the<br>
various<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; mechanisms by which a request-URI may =
be changed,<br>
which is what some<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; of the updated text in 4244bis is tryi=
ng to do and why<br>
we need precise<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; terms. I think we all agree that.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; I honestly don&#39;t care what we call=
 the tag, as long as<br>
we&#39;re clear<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; about the functionality.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; Mary.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; -----Original Message-----<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; From: Hans Erik van Elburg<br>
[mailto:<a href=3D"mailto:ietf.hanserik@gmail.com">ietf.hanserik@gmail.com<=
/a>]<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; Sent: Wednesday, March 11, 2009 4:40 P=
M<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; To: Barnes, Mary (RICH2:AR00)<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; Cc: Shida Schubert; <a href=3D"mailto:=
sip@ietf.org">sip@ietf.org</a><br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; Subject: Re: Terminology (was RE: [Sip=
] Fwd: I-D<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; ACTION:draft-rosenberg-sip-target-uri-=
delivery-01.txt<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; The target-uri concept when defined pr=
operly is<br>
entirely application<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; agnostic.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; When one uses History-Info or another =
header as a<br>
vehicle for this<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; information then it still can be consi=
dered<br>
application agnostic as<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; any application can use this informati=
on as it likes.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; =A0From SIP perspective all the Reques=
t-URI rewrites<br>
look the same, that<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; is what brought us discussing this in =
the first place.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; /Hans Erik<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; Mary Barnes wrote:<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; And, I think the big issue with th=
e terminology is<br>
that 4244bis is<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; application agnostic, so it is tag=
ging what happens<br>
to the URIs<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; (why/how the specific URI was over=
written). =A0Whereas,<br>
specific<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; applications can derive informatio=
n, such as what<br>
&quot;is&quot; the real<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; &quot;target&quot; for the request=
, from the History-Info<br>
header. =A0 So, IMHO,<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; it&#39;s a matter of the target-ur=
i document specifying<br>
that the last<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; entry tagged by whatever name we c=
ome up with &quot;is&quot;<br>
the real &quot;target&quot;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; for the request. =A0I believe it&#=
39;s up to the<br>
applications to describe<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; how they make use of the History-I=
nfo and not for the<br>
History-Info to<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; describe how applications can use =
it. =A0History-Info<br>
purely reflects<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; what happened from a SIP Protocol =
perspective and<br>
should not imply<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; any<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; application semantics. Indeed the =
intent of section 5<br>
of 4244 was to<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; inform the applications how they s=
hould decribe their<br>
usage of the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; header.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; Mary.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; Note: I changed the subject to hop=
efully make this<br>
topic easier to<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; keep track of.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
---------------------------------------------------------------------<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; -<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; --<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; *From:* Hans Erik van Elburg<br>
[mailto:<a href=3D"mailto:ietf.hanserik@gmail.com">ietf.hanserik@gmail.com<=
/a>]<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; *Sent:* Wednesday, March 11, 2009 =
3:59 AM<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; *To:* Shida Schubert<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; *Cc:* <a href=3D"mailto:sip@ietf.o=
rg">sip@ietf.org</a> List; Barnes, Mary (RICH2:AR00)<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; *Subject:* Re: [Sip] Fwd: I-D<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; ACTION:draft-rosenberg-sip-target-=
uri-delivery-01.txt<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; One of the problems that made the =
discussion quite<br>
tedious is the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; complete inversed meaning of the t=
erms &quot;retarget&quot; and<br>
&quot;reroute&quot;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; between 4244bis and target-uri fac=
tions have.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; Conclusion was that the terminolog=
y need to be<br>
properly defined and<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; some attempt was made to start suc=
h activity.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; On 3. I have strong concerns when =
we start tagging<br>
the URI as to how<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; they come about &quot;retarget&quo=
t;/&quot;mapped conficuration&quot;<br>
etc. It is better<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; to<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; embellish them with a tag that jus=
t represents there<br>
meaning for<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; example &quot;istarget&quot; or &q=
uot;hop&quot;. For the following<br>
reasons:<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; 1. It is much more intuitive for t=
he user of this<br>
information, its<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; meaning can basically be guessed.<=
br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; 2. Such meaning also probably surv=
ives application in<br>
other problem<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; spaces that we had not foreseen wh=
en introducing the<br>
concept.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; The solution that is =A0described =
now in the target-URI<br>
delivery draft<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; is not yet complete, it does not s=
olve the freephone<br>
use case for<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; example and in its current form it=
 will deliver<br>
exactly the same<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; target-URI as that which would be =
delivered by the<br>
P-Called-Party-ID<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; header. Contrary to what the draft=
 says. So some work<br>
is needed<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0still.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; /Hans Erik van Elburg<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; On Wed, Mar 11, 2009 at 7:13 AM, S=
hida Schubert<br>
&lt;<a href=3D"mailto:shida@ntt-at.com">shida@ntt-at.com</a><br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; &lt;mailto:<a href=3D"mailto:shida=
@ntt-at.com">shida@ntt-at.com</a>&gt;&gt; wrote:<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 We have submitted the upda=
ted target-uri draft<br>
based on the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; comments<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 submitted to the list and =
comments received at<br>
IETF73.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 I have taken over as edito=
r as Jonathan didn&#39;t<br>
have the cycles to<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 update the draft, with Fra=
ncois, Christer and<br>
Hans Erick as<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 additional<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 co-authors and great deal =
of help from Mary.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 The following summarizes t=
he changes made to the<br>
target-uri<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; document<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 1. Added use-case for toll=
-free number back<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 2. Added definition of &qu=
ot;retarget&quot; operation.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 3. Removed a reference to =
URN<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 4. Added a text discussing=
 the difference to<br>
P-Called-Party-Id<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 5. Changed parameter name =
from &quot;target&quot; to<br>
&quot;istarget&quot;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 Note, that the target-uri =
document still contains<br>
the normative<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 text for the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 History-Info header.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 In addition, Mary (with Fr=
ancois as co-author)<br>
has submitted a<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 rfc4244bis, with the follo=
wing changes:<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 1. Incorporated the normat=
ive aspects of the<br>
target-uri document<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 into the existing normativ=
e text in RFC 4244 -<br>
the functionality<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; is<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 virtually identical (as is=
 some of the text) as<br>
the HI based<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; solution<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 described in the target-ur=
i document. =A0It&#39;s<br>
important that the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 solution<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 be integrated into RFC 424=
4 as it MUST work and<br>
be based on the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 normative<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 aspects of RFC 4244.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 2. =A0Added the use cases =
from target-uri the the<br>
summary<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 in the overview of rfc4244=
bis.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 3. Added an additional req=
uirement to capture the<br>
&quot;target-uri&quot;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 information.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 4. Fixed an error in the R=
FC 4244 ABNF and added<br>
&quot;retarget&quot;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; parameter.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 5. Added a more simplified=
 example.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 We had some very long offl=
ine exchanges as to the<br>
best way<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; forward<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; and<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 remaining work for both do=
cuments.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 Some of the issues identif=
ied are:<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 ::Issues::<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A01. Should we remove=
 the normative text from<br>
target-uri and<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; progress<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 =A04244bis alo=
ng with the target-uri document<br>
to meet the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 chartered<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 SIP WG milesto=
ne?<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A02. Name of the para=
meter.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 At the last me=
eting, parameter &quot;target&quot; was<br>
said<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; inappropriate<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 because voicem=
ail-uri spec already defines<br>
a parameter<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; called<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 &quot;target&q=
uot; which also can be found in<br>
hi-entry, thus<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; potentially<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 causing confus=
ion.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 Currently the =
target-uri draft uses<br>
&quot;istarget&quot; and 4244bis<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; uses<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 &quot;retarget=
&quot; =A0but we could never come to<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 a consensus on=
 what name is appropriate.<br>
Other suggestions<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; have<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 included the f=
ollowing:<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 &quot;target-i=
dentity&quot; (someone didn&#39;t like that<br>
&quot;identity&quot; is<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 also a SIP header)<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 &quot;reg-uri&=
quot; =A0(can be paired with &quot;mapped-uri&quot;<br>
for item 3<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; below)<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 &quot;aor&quot=
;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 &quot;jibberis=
h&quot;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 etc.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 One reason thi=
s is so difficult relates to<br>
the problem<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 statement in target-uri in=
 that<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 RFC 3261 doesn=
&#39;t differentiate the<br>
mechanism by which the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; new<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 (target) Reque=
st-URI is selected. =A0Another<br>
issue is that<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 some of the terminology in=
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 RFC 3261 is ov=
erloaded - e.g., &quot;forwarding&quot;<br>
refers both to<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0a<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 Proxy<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 which does not=
 have responsibility for the<br>
domain of the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 request-URI<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 in the incomin=
g request, thus the proxy<br>
just &quot;forwards&quot; the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 request to<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 the next hop A=
ND &quot;forwarding&quot; is used to<br>
describe the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 process whereby<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 the outgoing r=
equest is built and<br>
&quot;forwarded&quot; to the next<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 hop at which<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 point the prox=
y does not know how the new<br>
request-uri was<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 selected.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 RFC 4244 has a=
ttempted to clarify the terms<br>
and attempts to<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 use &quot;forward&quot;<br=
>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 in the context=
 of the former situation and<br>
&quot;retarget&quot; for<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 the case whereby<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 a proxy is res=
ponsible for the domain and<br>
thus can use a<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 number of<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 mechanism to s=
elect the new target for the<br>
request - e.g.,<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0a<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 REGISTRAR,<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 configured dat=
a, etc.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A03. =A0Related to the la=
st point in item 2 above,<br>
it has been<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 proposed that<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0 we differentia=
te the hi-entries even more<br>
by defining<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 separate parameters<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0for registered =
and configured/mapped<br>
contacts.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0Currently when =
the R-URI is translated to a<br>
URI which is<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 either derived<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0from location s=
ervice lookup(registered by<br>
UA) or from<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; mapping<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0table, there is=
 no differentiation as to how<br>
the URI was<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 derived once it is<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0added to the li=
st of potential targets.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0The general con=
sensus of the authors of the<br>
two documents<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0was<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 that it may<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0be useful for s=
ome services to have the<br>
hi-entries tagged<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 with the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0more specific i=
nformation.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0And, of course,=
 this gets us into another<br>
naming contest. In<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 the end, the naming<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0is not so impor=
tant as long as the term<br>
isn&#39;t too overloaded<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 and it is defined<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 =A0 =A0 =A0precisely in th=
e document(s).<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 We would appreciate WG fee=
dback on these issues<br>
and any other<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 comments on<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 the two documents prior to=
 IETF-74.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 Regards,<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 Shida and Mary.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 Begin forwarded message:<b=
r>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 *From: *<a href=3D"mai=
lto:Internet-Drafts@ietf.org">Internet-Drafts@ietf.org</a><br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; &lt;mailto:<a href=3D"mailto:Internet-=
Drafts@ietf.org">Internet-Drafts@ietf.org</a>&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 *Date: *March 10, 2009=
 2:30:01 AM JST<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 *To: *<a href=3D"mailt=
o:i-d-announce@ietf.org">i-d-announce@ietf.org</a><br>
&lt;mailto:<a href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</=
a>&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 *Subject: **I-D<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt;<br>
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt *<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 *Reply-To: *<a href=3D=
"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 &lt;mailto:<a href=3D"=
mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a>&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 A New Internet-Draft i=
s available from the<br>
on-line<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; Internet-Drafts<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 directories.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 Title<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 : Delivery of Request-=
URI Targets to User Agents<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 Author(s)<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 : J. Rosenberg, H. van=
 Elburg, C. Holmberg, F.<br>
Audet, S.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Schubert<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 Filename :<br>
draft-rosenberg-sip-target-uri-delivery-01.txt<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 Pages<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 : 16<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 Date<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 : 2009-3-9<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 When a Session Initiat=
ion Protocol (SIP) proxy<br>
receives a<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0request<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 =A0 targeted at a URI =
identifying a user or<br>
resource it is<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; responsible<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 =A0 for, the proxy tra=
nslates the URI to a<br>
registered or<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0configured<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 =A0 contact URI of an =
agent representing that user<br>
or resource.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; In<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 =A0 process, the origi=
nal URI is removed from the<br>
request.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 =A0Numerous use<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 =A0 cases have arisen =
which require this<br>
information to be<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; delivered to<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 =A0 the user agent. =
=A0This document describes these<br>
use cases and<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 defines<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 =A0 an extension to th=
e History-Info header field<br>
which allows it<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; to be<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 =A0 used to support th=
ose cases.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 A URL for this Interne=
t-Draft is:<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt;<br>
<a href=3D"http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-u=
ri-d" target=3D"_blank">http://www.ietf.org/internet-drafts/draft-rosenberg=
-sip-target-uri-d</a><br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; e<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; livery-01.txt<br>
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so available by anonymous<br>
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etf.org/internet-drafts/" target=3D"_blank">ftp://ftp.ietf.org/internet-dra=
fts/</a><br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt;<br>
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h will enable a MIME<br>
compliant mail reader<br>
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matically retrieve the<br>
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t;BR&gt;Content-ID:<br>
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3B2009-3-9102756.I-D@ietf.org">lt;2009-3-9102756.I-D@ietf.org</a><br>
</div></div>&lt;mailto:<a href=3D"mailto:lt%253B2009-3-9102756.I-D@ietf.org=
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<div><div></div><div class=3D"h5"> =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =
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 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 ______________________=
_________________________<br>
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ist<br>
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Announce@ietf.org">I-D-Announce@ietf.org</a><br>
&lt;mailto:<a href=3D"mailto:I-D-Announce@ietf.org">I-D-Announce@ietf.org</=
a>&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt;<br>
<a href=3D"https://www.ietf.org/mailman/listinfo/i-d-announce" target=3D"_b=
lank">https://www.ietf.org/mailman/listinfo/i-d-announce</a><br>
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ries:<br>
<a href=3D"http://www.ietf.org/shadow.html" target=3D"_blank">http://www.ie=
tf.org/shadow.html</a><br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt; =A0 =A0 or <a href=3D"ftp://ft=
p.ietf.org/ietf/1shadow-sites.txt" target=3D"_blank">ftp://ftp.ietf.org/iet=
f/1shadow-sites.txt</a><br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 __________________________=
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 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 This list is for NEW devel=
opment of the core SIP<br>
Protocol<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 Use <a href=3D"mailto:sip-=
implementors@cs.columbia.edu">sip-implementors@cs.columbia.edu</a><br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 &lt;mailto:<a href=3D"mail=
to:sip-implementors@cs.columbia.edu">sip-implementors@cs.columbia.edu</a>&g=
t; for<br>
questions on<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; current<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt; sip<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 Use <a href=3D"mailto:sipp=
ing@ietf.org">sipping@ietf.org</a> &lt;mailto:<a href=3D"mailto:sipping@iet=
f.org">sipping@ietf.org</a>&gt;<br>
for new<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt; =A0 =A0 developments on the applic=
ation of sip<br>
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 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&gt;&gt;<br>
<br>
<br>
<br>
</div></div></blockquote></div><br>

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I am Gao. And comments inline.
 


Date: Thu, 12 Mar 2009 12:48:57 +0100
From: christer.holmberg@ericsson.com
To: gao.yang2@zte.com.cn
CC: sip@ietf.org; sipping@ietf.org; gonzalo.camarillo@ericsson.com
Subject: Re: [Sipping] One chart //"UPDATE during Re-INVITE" discussion


Hi,
 
If the UAC receives the 487 and can't fallback to the pre-reINVITE state, it sends a new re-INVITE with something that it DOES supports. That is also described in Gonzalo's draft, I belive.
 
[Gao] I just mean that the UAC can only use the old dialog state by Gonzalo's draft. Do you think so?
And Re-INVITE using the old dialog state can not be processed by UAS. It is a serious problem.
 
 
And, you could have the same problem if the target refresh was done already in the re-INVITE/18x. If the UAC then sends CANCEL, and can't fall back, it will have to send a new re-INVITE.
 
[Gao] If the target fresh is in Re-INVITE(and there is no F3/F4), the new dialog state is rejected by UAS. And F8 MUST uses old dialog state. It is clear.
 
I also think it is bad practise to do a target refresh in a nested UPDATE, but that's a separate issue.
 
[Gao] When UAC send Re-INVITE and UAS moves to a new access network while Re-INVITE in pending state, UAS will send the target refresh UPDATE. It is a real situation.
 
Regards,
 
Christer
 



From: gao.yang2@zte.com.cn [mailto:gao.yang2@zte.com.cn] 
Sent: 12. maaliskuuta 2009 9:41
To: Christer Holmberg
Cc: Gonzalo Camarillo; SIP; SIPPING
Subject: ´ð¸´: RE: One chart //"UPDATE during Re-INVITE" discussion



Hi, 

I just mean that if we treat F3 as sub-transaction of F1, then after F7(ACK), UAC can discard refreshed dialog state in F3. 

And it may make the two parts can not communicate anymore in current dialog. 

And IMO, this is a counterevidence for treating all UPDATE(during Re-INVITE) as sub-transaction of Re-INVITE. 

Gao 








"Christer Holmberg" <christer.holmberg@ericsson.com> 
2009-03-12 15:31 






ÊÕ¼þÈË
<gao.yang2@zte.com.cn>, "SIP" <sip@ietf.org>, "SIPPING" <sipping@ietf.org> 


³­ËÍ
"Gonzalo Camarillo" <gonzalo.camarillo@ericsson.com> 


Ö÷Ìâ
RE: One chart  //"UPDATE during Re-INVITE" discussion









In my opinion 487 should trigger whatever actions/rollback that any other non-200 response sent by the UAS would. 
  
Regards, 
  
Christer 



From: gao.yang2@zte.com.cn [mailto:gao.yang2@zte.com.cn] 
Sent: 12. maaliskuuta 2009 5:47
To: SIP; SIPPING
Cc: Christer Holmberg; Gonzalo Camarillo
Subject: One chart //"UPDATE during Re-INVITE" discussion


     UAC                   UAS 
      | session established | 
      |<===================>| 
      |                     | 
      | F1  re-INVITE (SDP) | 
      |-------------------->| 
      | F2 1xx-rel (SDP)    | 
      |<--------------------| 
      | F3   UPDATE         | 
      |<--------------------| Target refreshing 
      | F4 2xx UPT          | 
      |-------------------->| 
      |                     | 
      | F5/6 Cancel/200OK   | 
      |-------------------->| 
      | F7    487 INV       | 
      |<--------------------| 
      | F7     ACK          | 
      |-------------------->| 
      | F8  re-INVITE (SDP) | 
      |-------------------->| Using which dialog state 

F3/F4 is about target refreshing. 

And if UAC sends Cancel£¬ UAS sends 487. The first Re-INVITE is discarded. 

But if UAC sends another Re-INVITE, which dialog state should it use? 

I think is the one refreshed by F3. That's the merit of regarding UPDATE/200OK's atomicity. 

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I am&nbsp;Gao. And comments inline.<BR>&nbsp;<BR>
<HR id=stopSpelling>
Date: Thu, 12 Mar 2009 12:48:57 +0100<BR>From: christer.holmberg@ericsson.com<BR>To: gao.yang2@zte.com.cn<BR>CC: sip@ietf.org; sipping@ietf.org; gonzalo.camarillo@ericsson.com<BR>Subject: Re: [Sipping] One chart //"UPDATE during Re-INVITE" discussion<BR><BR>
<DIV dir=ltr align=left><SPAN class=EC_169564411-12032009><FONT face=Arial color=#0000ff size=2>Hi,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=EC_169564411-12032009><FONT face=Arial color=#0000ff size=2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=ltr align=left><SPAN class=EC_169564411-12032009><FONT face=Arial color=#0000ff size=2>If the UAC receives the 487 and can't fallback to the pre-reINVITE state, it sends a new re-INVITE with something that it DOES&nbsp;supports. That is also described in Gonzalo's draft, I belive.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=EC_169564411-12032009><FONT face=Arial color=#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV dir=ltr align=left><SPAN class=EC_169564411-12032009><FONT face=Arial color=#0000ff>[Gao] I just mean that the UAC can only use the old dialog state by Gonzalo's draft. Do you think so?</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=EC_169564411-12032009><FONT face=Arial color=#0000ff>And Re-INVITE using the old dialog state can not be processed by UAS. It is a serious problem.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=EC_169564411-12032009><FONT face=Arial color=#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV dir=ltr align=left><SPAN class=EC_169564411-12032009><FONT face=Arial color=#0000ff></FONT></SPAN><SPAN class=EC_169564411-12032009><FONT face=Arial color=#0000ff size=2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=ltr align=left><SPAN class=EC_169564411-12032009><FONT face=Arial color=#0000ff size=2>And, you could have the same problem if the target refresh was done already in the re-INVITE/18x. If the UAC then sends CANCEL, and can't fall back, it will have to send a new re-INVITE.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=EC_169564411-12032009></SPAN>&nbsp;</DIV>
<DIV dir=ltr align=left><SPAN class=EC_169564411-12032009>[Gao] If the target fresh is in Re-INVITE(and there is no F3/F4), the new dialog state is rejected by UAS. And F8 MUST uses old dialog state. It is clear.</SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=EC_169564411-12032009><FONT face=Arial color=#0000ff size=2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=ltr align=left><SPAN class=EC_169564411-12032009><FONT face=Arial color=#0000ff size=2>I also think it is bad practise to do a target refresh in a nested UPDATE, but that's a separate issue.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=EC_169564411-12032009><FONT face=Arial color=#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV dir=ltr align=left><SPAN class=EC_169564411-12032009><FONT face=Arial color=#0000ff>[Gao] When UAC send Re-INVITE and UAS moves to a new access network while Re-INVITE in pending state, UAS will send the target refresh UPDATE. It is a real situation.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=EC_169564411-12032009><FONT face=Arial color=#0000ff size=2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=ltr align=left><SPAN class=EC_169564411-12032009><FONT face=Arial color=#0000ff size=2>Regards,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=EC_169564411-12032009><FONT face=Arial color=#0000ff size=2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=ltr align=left><SPAN class=EC_169564411-12032009><FONT face=Arial color=#0000ff size=2>Christer</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=EC_169564411-12032009><FONT face=Arial color=#0000ff size=2></FONT></SPAN>&nbsp;</DIV>
<BLOCKQUOTE dir=ltr style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
<DIV class=EC_OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR>
<FONT face=Tahoma size=2><B>From:</B> gao.yang2@zte.com.cn [mailto:gao.yang2@zte.com.cn] <BR><B>Sent:</B> 12. maaliskuuta 2009 9:41<BR><B>To:</B> Christer Holmberg<BR><B>Cc:</B> Gonzalo Camarillo; SIP; SIPPING<BR><B>Subject:</B> ´ð¸´: RE: One chart //"UPDATE during Re-INVITE" discussion<BR></FONT><BR></DIV>
<DIV></DIV><BR><FONT face=sans-serif size=2>Hi,</FONT> <BR><BR><FONT face=sans-serif size=2>I just mean that if we treat F3 as sub-transaction of F1, then after F7(ACK), UAC can discard refreshed dialog state in F3.</FONT> <BR><BR><FONT face=sans-serif size=2>And it may make the two parts can not communicate anymore in current dialog.</FONT> <BR><BR><FONT face=sans-serif size=2>And IMO, this is a counterevidence for treating all UPDATE(during Re-INVITE) as sub-transaction of Re-INVITE.</FONT> <BR><BR><FONT face=sans-serif size=2>Gao</FONT> <BR><BR><BR><BR><BR>
<TABLE width="100%">
<TBODY>
<TR vAlign=top>
<TD width="26%"><FONT face=sans-serif size=1><B>"Christer Holmberg" &lt;christer.holmberg@ericsson.com&gt;</B> </FONT>
<FONT face=sans-serif size=1>2009-03-12 15:31</FONT> <BR>
<TD width="73%">
<TABLE width="100%">
<TBODY>
<TR vAlign=top>
<TD>
<DIV align=right><FONT face=sans-serif size=1>ÊÕ¼þÈË</FONT></DIV>
<TD><FONT face=sans-serif size=1>&lt;gao.yang2@zte.com.cn&gt;, "SIP" &lt;sip@ietf.org&gt;, "SIPPING" &lt;sipping@ietf.org&gt;</FONT> 
<TR vAlign=top>
<TD>
<DIV align=right><FONT face=sans-serif size=1>³­ËÍ</FONT></DIV>
<TD><FONT face=sans-serif size=1>"Gonzalo Camarillo" &lt;gonzalo.camarillo@ericsson.com&gt;</FONT> 
<TR vAlign=top>
<TD>
<DIV align=right><FONT face=sans-serif size=1>Ö÷Ìâ</FONT></DIV>
<TD><FONT face=sans-serif size=1>RE: One chart &nbsp;//"UPDATE during Re-INVITE" discussion</FONT></TD></TR></TBODY></TABLE><BR>
<TABLE>
<TBODY>
<TR vAlign=top>
<TD>
<TD></TD></TR></TBODY></TABLE><BR></TD></TR></TBODY></TABLE><BR><BR><BR><FONT face=Arial color=blue size=2>In my opinion 487 should trigger whatever actions/rollback that any other non-200 response sent by the UAS would.</FONT> <BR><FONT face=sans-serif size=3>&nbsp;</FONT> <BR><FONT face=Arial color=blue size=2>Regards,</FONT> <BR><FONT face=sans-serif size=3>&nbsp;</FONT> <BR><FONT face=Arial color=blue size=2>Christer</FONT> <BR><BR>
<HR>
<FONT face=Tahoma size=2><B>From:</B> gao.yang2@zte.com.cn [mailto:gao.yang2@zte.com.cn] <B><BR>Sent:</B> 12. maaliskuuta 2009 5:47<B><BR>To:</B> SIP; SIPPING<B><BR>Cc:</B> Christer Holmberg; Gonzalo Camarillo<B><BR>Subject:</B> One chart //"UPDATE during Re-INVITE" discussion</FONT><FONT face=sans-serif size=3><BR></FONT><BR><FONT face=sans-serif size=2><BR>&nbsp; &nbsp; &nbsp;UAC &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; UAS</FONT><FONT face=sans-serif size=3> </FONT><FONT face=sans-serif size=2><BR>&nbsp; &nbsp; &nbsp; | session established |</FONT><FONT face=sans-serif size=3> </FONT><FONT face=sans-serif size=2><BR>&nbsp; &nbsp; &nbsp; |&lt;===================&gt;|</FONT><FONT face=sans-serif size=3> </FONT><FONT face=sans-serif size=2><BR>&nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</FONT><FONT face=sans-serif size=3> </FONT><FONT face=sans-serif size=2><BR>&nbsp; &nbsp; &nbsp; | F1 &nbsp;re-INVITE (
 SDP) |</FONT><FONT face=sans-serif size=3> </FONT><FONT face=sans-serif size=2><BR>&nbsp; &nbsp; &nbsp; |--------------------&gt;|</FONT><FONT face=sans-serif size=3> </FONT><FONT face=sans-serif size=2><BR>&nbsp; &nbsp; &nbsp; | F2 1xx-rel (SDP) &nbsp; &nbsp;|</FONT><FONT face=sans-serif size=3> </FONT><FONT face=sans-serif size=2><BR>&nbsp; &nbsp; &nbsp; |&lt;--------------------|</FONT><FONT face=sans-serif size=3> </FONT><FONT face=sans-serif size=2><BR>&nbsp; &nbsp; &nbsp; | F3 &nbsp; UPDATE &nbsp; &nbsp; &nbsp; &nbsp; |</FONT><FONT face=sans-serif size=3> </FONT><FONT face=sans-serif size=2><BR>&nbsp; &nbsp; &nbsp; |&lt;--------------------| Target refreshing</FONT><FONT face=sans-serif size=3> </FONT><FONT face=sans-serif size=2><BR>&nbsp; &nbsp; &nbsp; | F4 2xx UPT &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|</FONT><FONT face=sans-serif size=3> </FONT><FONT face=sans-serif size=2><BR>&nbsp; &nbsp; &nbsp; |--------------------&gt;|</FONT><FONT face=sans-serif size=3> </FONT><F
 ONT face=sans-serif size=2><BR>&nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</FONT><FONT face=sans-serif size=3> </FONT><FONT face=sans-serif size=2><BR>&nbsp; &nbsp; &nbsp; | F5/6 Cancel/200OK &nbsp; |</FONT><FONT face=sans-serif size=3> </FONT><FONT face=sans-serif size=2><BR>&nbsp; &nbsp; &nbsp; |--------------------&gt;|</FONT><FONT face=sans-serif size=3> </FONT><FONT face=sans-serif size=2><BR>&nbsp; &nbsp; &nbsp; | F7 &nbsp; &nbsp;487 INV &nbsp; &nbsp; &nbsp; |</FONT><FONT face=sans-serif size=3> </FONT><FONT face=sans-serif size=2><BR>&nbsp; &nbsp; &nbsp; |&lt;--------------------|</FONT><FONT face=sans-serif size=3> </FONT><FONT face=sans-serif size=2><BR>&nbsp; &nbsp; &nbsp; | F7 &nbsp; &nbsp; ACK &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|</FONT><FONT face=sans-serif size=3> </FONT><FONT face=sans-serif size=2><BR>&nbsp; &nbsp; &nbsp; |--------------------&gt;|</FONT><FONT face=sans-serif size=3> </FONT><FONT face=sans-seri
 f size=2><BR>&nbsp; &nbsp; &nbsp; | F8 &nbsp;re-INVITE (SDP) |</FONT><FONT face=sans-serif size=3> </FONT><FONT face=sans-serif size=2><BR>&nbsp; &nbsp; &nbsp; |--------------------&gt;| Using which dialog state</FONT><FONT face=sans-serif size=3> <BR></FONT><FONT face=sans-serif size=2><BR>F3/F4 is about target refreshing.</FONT><FONT face=sans-serif size=3> <BR></FONT><FONT face=sans-serif size=2><BR>And if UAC sends Cancel£¬ UAS sends 487. The first Re-INVITE is discarded.</FONT><FONT face=sans-serif size=3> <BR></FONT><FONT face=sans-serif size=2><BR>But if UAC sends another Re-INVITE, which dialog state should it use?</FONT><FONT face=sans-serif size=3> <BR></FONT><FONT face=sans-serif size=2><BR>I think is the one refreshed by F3. That's the merit of regarding UPDATE/200OK's atomicity.</FONT><FONT face=sans-serif size=3> <BR></FONT><BR><FONT size=3><TT>--------------------------------------------------------<BR>ZTE Information Security Notice: The information contained
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From gao.yang.seu@hotmail.com  Thu Mar 12 06:35:44 2009
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Hi,

 

I have same thought about it as yours. 

 

But in Ganzalo's draft, the new dialog state MUST be discarded by 4xx of Re-INVITE.


The main branching is about the relationship of Re-INVITE and UPDATE. I have seen your supporting of view 2. Abd we need more people to show their standpoint.

 

Gao

 


From: brett@broadsoft.com
To: gao.yang2@zte.com.cn; sip@ietf.org; sipping@ietf.org
Date: Thu, 12 Mar 2009 04:57:06 -0700
Subject: Re: [Sipping] [Sip] One chart //"UPDATE during Re-INVITE" discussion





Data (like Contact, Session-Expires, and etcetera) that was successfully updated by F3/F4 still apply after F7.  Thus F8 data should reflect (and can attempt to modify) adjustments made during F3/F4.
 
 



From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of gao.yang2@zte.com.cn
Sent: Wednesday, March 11, 2009 11:47 PM
To: SIP; SIPPING
Cc: Gonzalo Camarillo; Christer Holmberg
Subject: [Sip] One chart //"UPDATE during Re-INVITE" discussion
 

      UAC                   UAS 
       | session established | 
       |<===================>| 
       |                     | 
       | F1  re-INVITE (SDP) | 
       |-------------------->| 
       | F2 1xx-rel (SDP)    | 
       |<--------------------| 
       | F3   UPDATE         | 
       |<--------------------| Target refreshing 
       | F4 2xx UPT          | 
       |-------------------->| 
       |                     | 
       | F5/6 Cancel/200OK   | 
       |-------------------->| 
       | F7    487 INV       | 
       |<--------------------| 
       | F7     ACK          | 
       |-------------------->| 
       | F8  re-INVITE (SDP) | 
       |-------------------->| Using which dialog state 

F3/F4 is about target refreshing. 

And if UAC sends Cancel£¬ UAS sends 487. The first Re-INVITE is discarded. 

But if UAC sends another Re-INVITE, which dialog state should it use? 

I think is the one refreshed by F3. That's the merit of regarding UPDATE/200OK's atomicity.  
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&nbsp;<BR>
Hi,<BR>
&nbsp;<BR>
I&nbsp;have same thought&nbsp;about it as yours. <BR>
&nbsp;<BR>
But&nbsp;in Ganzalo's draft,&nbsp;the new dialog state&nbsp;MUST be discarded by 4xx of Re-INVITE.<BR><BR>
The main branching is about the relationship of Re-INVITE and UPDATE. I have seen your supporting of view 2. Abd we need more people to show their standpoint.<BR>
&nbsp;<BR>
Gao<BR>
&nbsp;<BR>
<HR id=stopSpelling>
From: brett@broadsoft.com<BR>To: gao.yang2@zte.com.cn; sip@ietf.org; sipping@ietf.org<BR>Date: Thu, 12 Mar 2009 04:57:06 -0700<BR>Subject: Re: [Sipping] [Sip] One chart //"UPDATE during Re-INVITE" discussion<BR><BR>
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<DIV class=EC_Section1>
<P class=EC_MsoNormal><SPAN style="FONT-SIZE: 10pt; COLOR: #1f497d; FONT-FAMILY: 'Courier New'">Data (like Contact, Session-Expires, and etcetera) that was successfully updated by F3/F4 still apply after F7.&nbsp; Thus F8 data should reflect (and can attempt to modify) adjustments made during F3/F4.</SPAN></P>
<P class=EC_MsoNormal><SPAN style="FONT-SIZE: 10pt; COLOR: #1f497d; FONT-FAMILY: 'Courier New'">&nbsp;</SPAN></P>
<P class=EC_MsoNormal><SPAN style="FONT-SIZE: 10pt; COLOR: #1f497d; FONT-FAMILY: 'Courier New'">&nbsp;</SPAN></P>
<DIV style="BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: medium none; PADDING-LEFT: 4pt; PADDING-BOTTOM: 0in; BORDER-LEFT: blue 1.5pt solid; PADDING-TOP: 0in; BORDER-BOTTOM: medium none">
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<DIV style="BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid; PADDING-LEFT: 0in; PADDING-BOTTOM: 0in; BORDER-LEFT: medium none; PADDING-TOP: 3pt; BORDER-BOTTOM: medium none">
<P class=EC_MsoNormal><B><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'">From:</SPAN></B><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'"> sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>gao.yang2@zte.com.cn<BR><B>Sent:</B> Wednesday, March 11, 2009 11:47 PM<BR><B>To:</B> SIP; SIPPING<BR><B>Cc:</B> Gonzalo Camarillo; Christer Holmberg<BR><B>Subject:</B> [Sip] One chart //"UPDATE during Re-INVITE" discussion</SPAN></P></DIV></DIV>
<P class=EC_MsoNormal>&nbsp;</P>
<P class=EC_MsoNormal style="MARGIN-BOTTOM: 12pt"><BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp; &nbsp; &nbsp; UAC &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; UAS</SPAN> <BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp; &nbsp; &nbsp; &nbsp;| session established |</SPAN> <BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp; &nbsp; &nbsp; &nbsp;|&lt;===================&gt;|</SPAN> <BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</SPAN> <BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp; &nbsp; &nbsp; &nbsp;| F1 &nbsp;re-INVITE (SDP) |</SPAN> <BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp; &nbsp; &nbsp; &nbsp;|--------------------&gt;|</SPAN> <BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbs
 p; &nbsp; &nbsp; &nbsp;| F2 1xx-rel (SDP) &nbsp; &nbsp;|</SPAN> <BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp; &nbsp; &nbsp; &nbsp;|&lt;--------------------|</SPAN> <BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp; &nbsp; &nbsp; &nbsp;| F3 &nbsp; UPDATE &nbsp; &nbsp; &nbsp; &nbsp; |</SPAN> <BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp; &nbsp; &nbsp; &nbsp;|&lt;--------------------| Target refreshing</SPAN> <BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp; &nbsp; &nbsp; &nbsp;| F4 2xx UPT &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|</SPAN> <BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp; &nbsp; &nbsp; &nbsp;|--------------------&gt;|</SPAN> <BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</SPAN> <BR><SPAN style="FONT-SIZE: 10pt; FO
 NT-FAMILY: 'Arial','sans-serif'">&nbsp; &nbsp; &nbsp; &nbsp;| F5/6 Cancel/200OK &nbsp; |</SPAN> <BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp; &nbsp; &nbsp; &nbsp;|--------------------&gt;|</SPAN> <BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp; &nbsp; &nbsp; &nbsp;| F7 &nbsp; &nbsp;487 INV &nbsp; &nbsp; &nbsp; |</SPAN> <BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp; &nbsp; &nbsp; &nbsp;|&lt;--------------------|</SPAN> <BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp; &nbsp; &nbsp; &nbsp;| F7 &nbsp; &nbsp; ACK &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|</SPAN> <BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp; &nbsp; &nbsp; &nbsp;|--------------------&gt;|</SPAN> <BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp; &nbsp; &nbsp; &nbsp;| F8 &nbsp;re-INVITE (SDP) |</SPAN> <BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','s
 ans-serif'">&nbsp; &nbsp; &nbsp; &nbsp;|--------------------&gt;| Using which dialog state</SPAN> <BR><BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">F3/F4 is about target refreshing.</SPAN> <BR><BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">And if UAC sends Cancel</SPAN><SPAN lang=ZH-CN style="FONT-SIZE: 10pt">£¬</SPAN><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'"> UAS sends 487. The first Re-INVITE is discarded.</SPAN> <BR><BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">But if UAC sends another Re-INVITE, which dialog state should it use?</SPAN> <BR><BR><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">I think is the one refreshed by F3. That's the merit of regarding UPDATE/200OK's atomicity.</SPAN> </P><PRE>&nbsp;</PRE></DIV></DIV><br /><hr />Á¢¿ÌÏÂÔØ MSN ±£»¤¶Ü£¬±£ÕÏ MSN °²È«ÎÈ¶¨£¡ <a href='http://im.live.cn/safe/' target='_new'>ÏÖÔÚ¾ÍÏÂÔØ£¡</a></body>
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Subject: Re: [Sip] Terminology (was RE: Fwd: I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Hans Erik,
=20
The way I interpreted the 01 draft, you would not get the freephone
number anyway. If that is a requirement, then I accept my suggestion
won't suffice.
=20
John


________________________________

	From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
	Sent: 12 March 2009 12:57
	To: Elwell, John
	Cc: sip@ietf.org
	Subject: Re: [Sip] Terminology (was RE: Fwd:
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
=09
=09
	But that does not solve the freephone service use case.
Freephone represents a use case where the R-URI is replaced with an AoR
of user B - not with the contact of user B.=20
=09
	A calls Bfreephonenumber routed the B (AOR) routed to B's
contact.
=09
	The target-uri draft is interested in obtaining:
Bfreephonenumber as that is what A used to reach B.
=09
	So what we like to see that B UA receives is something like
	    H-I:  Bfreephone; "istarget",\
	        B,\
	        contact-B
=09
	So you want to get Bfreephonenumber and not B.
=09
	In certain situations you are also interested in B, I guess that
PBX's would interested in that to deliver to the correct user. We could
distinguish them like:
	    H-I:  Bfreephone; "istarget",\
	        B,"aor"\
	        contact-B
=09
	BR,
	/Hans Erik van Elburg
=09
=09
=09
	On Thu, Mar 12, 2009 at 12:08 PM, Elwell, John
<john.elwell@siemens.com> wrote:
=09

		If 'istarget' is used only to identify the value that
was overwritten in
		the Request-URI by a contact URI, an alternative
approach would be to
		flag the contact URI H-I entry as 'iscontact'. Then the
UAS would just
		need to look for the most recent H-I entry that is not
marked
		'iscontact'.
	=09
		John
	=09
	=09
		________________________________
	=09
		       From: sip-bounces@ietf.org
[mailto:sip-bounces@ietf.org] On
		Behalf Of Hans Erik van Elburg
		       Sent: 12 March 2009 08:35
		       To: Mary Barnes
		       Cc: sip@ietf.org
		       Subject: Re: [Sip] Terminology (was RE: Fwd:
		I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
	=09


		       Yes, because you are using the 3261 use of target
and the
		4244bis introduced definition of retarget. I thought it
was clear that
		we need other words as those definitions don't match the
target-uri
		drafts use of the terms. Also they do not suffice to
provide a solution
		for the use cases in the target-uri draft.
	=09
		       The 3261 text you refer to is exactly about the
case where the
		home proxy overwrites the Request-URI with a new target.
This target is
		teh registered contact address. And hence this would be
what target-uri
		calls a hop or a route. This case and this is where it
gets confusing is
		not a "retarget" in the target-uri draft use of the
term.
	=09
		       The target-uri draft states:
	=09
		          "To avoid confusion, we
		          refer to a SIP URI that is an address for a
user or resource
		as a
		          "target" and a SIP URI that is a hop for
reaching that user
		as a
	=09
		          "hop".
	=09
	=09
		       Apparently that does not suffice to avoid
confusion.
	=09
		       As for the tagging, speaking about the solution
before agreeing
		on the terminology and the problem it should solve is
meaningless.
	=09
		       /Hans Erik van Elburg
	=09
	=09
	=09
		       On Thu, Mar 12, 2009 at 2:33 AM, Mary Barnes
		<mary.barnes@nortel.com> wrote:
	=09
	=09
		               Responses below [MB].
	=09
	=09
		               -----Original Message-----
		               From: Hans Erik van Elburg
		[mailto:ietf.hanserik@gmail.com]
	=09
		               Sent: Wednesday, March 11, 2009 6:42 PM
		               To: Barnes, Mary (RICH2:AR00)
		               Cc: Shida Schubert; sip@ietf.org
		               Subject: Re: Terminology (was RE: [Sip]
Fwd: I-D
=09
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
	=09
	=09
		               I was talking about the concept.
	=09
		               The use cases only describe  cases where
the target-uri
		(lets call that
		               current target from now) has been lost
when an initial
		request for a
		               dialog or standalone request arrives at
the UAS.
	=09
		               [MB] And I think this is where the
terminology confusion
		starts - I
		               don't think of the "lost" target-uri as
being the
		"current" target. In
		               my mind, the "current" target is
reflected by the last
		hi-entry and the
		               request-uri in the incoming request at
the UAS. If the
		entity that sent
		               the request was the entity that added the
last hi-entry,
		then the uri in
		               that hi-entry is the same as the
request-URI in the SIP
		request that
		               arrives at the UAS. I refer to the "lost"
uri as the one
		that was
		               "retargeted" - that's the one the UAS
wants to pull from
		the hi-entries
		               in the incoming request. That hi-entry
was not the one
		that was just
		               added by the entity that built the
request just received
		by the UAS.
		               That hi-entry is tagged with whatever
name we are going
		to tag it with
		               BEFORE the request is forwarded (using
the term forward
		per section 16.6
		               of RFC 3261).  That tag is added once the
target set of
		potential
		               candidates for the new request uri are
determined in
		section 16.5 of
		               3261 (with "target set" being a 3261
term), just before
		the request is
		               forwarded in section 16.6 to one of those
targets.  A
		new hi-entry
		               (which will be the last hi-entry in the
request received
		by the UAS) is
		               added in section 16.6 of 3261 as the
request is
		forwarded.  At this
		               point in time, the lost information is in
the previous
		hi-entry when the
		               outgoing request is sent. [/MB]
	=09
	=09
		               What has been lost is the current target
of the request:
	=09
		               [MB] Right - at which point in my mind,
it's no longer
		the current
		               target ;) Maybe we call it "lost". [/MB]
	=09
	=09
	=09
		                    Current target
	=09
		                 The current target of an initial
request for a dialog
		or standalone
		                 request is the name or address to which
the request is
		targeted, i.e.
		                 either the initial target inserted in
the Request-URI
		by the UAC that
		                 originates the request, or when a
retarget occurred,
		the target
		                 provided in that retarget operation.
Reroute and
		translation
		                 operations never change the current
target.
	=09
	=09
		               [MB] I don't think this definition fits
what you want -
		i.e., if there
		               is no retargeting, then none of the
hi-entries are
		tagged - i.e., you
		               won't have your concept of current. The
way it works is
		that if no
		               hi-entries are tagged, then you know that
there was no
		retargeting, thus
		               you know the request-uri has not been
lost. [/MB]
	=09
	=09
		               This defintion only makes sense when the
following
		definitions are used:
	=09
		               Name:
	=09
		                 A name is a moniker for an entity which
refers to it
		in a way which
	=09
		                 reveals nothing about where it is in a
network.  In
		SIP, tel URI
	=09
		                 which doesn't represent the location of
the entity is
		a name.
	=09
	=09
		               Address:
	=09
		                 An address is an identifier for an
entity which
		describes it by its
	=09
		                 location on the network.  In SIP, the
SIP URI itself
		is a form of
	=09
		                 address because the host part of the
URI, the only
		mandatory part of
	=09
		                 the URI besides the scheme itself,
indicates the
		location of a SIP
	=09
		                 server that can be used to handle the
request.
	=09
		               Route:
	=09
		                 Finally, a route is a   sequence of SIP
entities
		(including the UA
		               itself!) which are
	=09
		                traversed in order to forward a request
to an address
		or name.
	=09
		               Retarget (other term might be needed, as
this is highly
		confusing):
	=09
		                  A Request-URI rewrite operation that
changes the
		target identity of
		               the request.
	=09
		               Reroute (other term might be needed):
	=09
		                  A Request-URI rewrite operation that
does not change
		the current
		               target of the request, but     determines
the route/next
		hop taken to
		               reach the target-identity.
	=09
		               /Hans Erik
	=09
		               Mary Barnes wrote:
		               > So, by "target-uri concept" are you
referring to the
		solution in the
		               > current target-uri draft or the
application usage as
		such in the
		               > target-uri document (which is what I
was referring
		to)?
		               >
		               > Yes, the problem is that SIP doesn't
differentiate the
		various
		               > mechanisms by which a request-URI may
be changed,
		which is what some
		               > of the updated text in 4244bis is
trying to do and why
		we need precise
	=09
		               > terms. I think we all agree that.
		               >
		               > I honestly don't care what we call the
tag, as long as
		we're clear
		               > about the functionality.
		               >
		               > Mary.
		               >
		               > -----Original Message-----
		               > From: Hans Erik van Elburg
		[mailto:ietf.hanserik@gmail.com]
		               > Sent: Wednesday, March 11, 2009 4:40 PM
		               > To: Barnes, Mary (RICH2:AR00)
		               > Cc: Shida Schubert; sip@ietf.org
		               > Subject: Re: Terminology (was RE: [Sip]
Fwd: I-D
		               >
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
		               >
		               > The target-uri concept when defined
properly is
		entirely application
		               > agnostic.
		               >
		               > When one uses History-Info or another
header as a
		vehicle for this
		               > information then it still can be
considered
		application agnostic as
		               > any application can use this
information as it likes.
		               >
		               >  From SIP perspective all the
Request-URI rewrites
		look the same, that
	=09
		               > is what brought us discussing this in
the first place.
		               >
		               > /Hans Erik
		               >
		               > Mary Barnes wrote:
		               >
		               >> And, I think the big issue with the
terminology is
		that 4244bis is
		               >> application agnostic, so it is tagging
what happens
		to the URIs
		               >> (why/how the specific URI was
overwritten).  Whereas,
		specific
		               >> applications can derive information,
such as what
		"is" the real
		               >> "target" for the request, from the
History-Info
		header.   So, IMHO,
		               >> it's a matter of the target-uri
document specifying
		that the last
		               >> entry tagged by whatever name we come
up with "is"
		the real "target"
		               >> for the request.  I believe it's up to
the
		applications to describe
		               >> how they make use of the History-Info
and not for the
		History-Info to
	=09
		               >> describe how applications can use it.
History-Info
		purely reflects
		               >> what happened from a SIP Protocol
perspective and
		should not imply
		               >> any
		               >>
		               >
		               >
		               >> application semantics. Indeed the
intent of section 5
		of 4244 was to
		               >> inform the applications how they
should decribe their
		usage of the
		               >> header.
		               >>
		               >> Mary.
		               >> Note: I changed the subject to
hopefully make this
		topic easier to
		               >> keep track of.
		               >>
		               >>
=09
---------------------------------------------------------------------
		               >> -
		               >> --
		               >> *From:* Hans Erik van Elburg
		[mailto:ietf.hanserik@gmail.com]
		               >> *Sent:* Wednesday, March 11, 2009 3:59
AM
		               >> *To:* Shida Schubert
		               >> *Cc:* sip@ietf.org List; Barnes, Mary
(RICH2:AR00)
		               >> *Subject:* Re: [Sip] Fwd: I-D
		               >>
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
		               >>
		               >> One of the problems that made the
discussion quite
		tedious is the
		               >> complete inversed meaning of the terms
"retarget" and
		"reroute"
		               >> between 4244bis and target-uri
factions have.
		               >>
		               >> Conclusion was that the terminology
need to be
		properly defined and
		               >> some attempt was made to start such
activity.
		               >>
		               >> On 3. I have strong concerns when we
start tagging
		the URI as to how
		               >> they come about "retarget"/"mapped
conficuration"
		etc. It is better
		               >> to
		               >>
		               >
		               >
		               >> embellish them with a tag that just
represents there
		meaning for
		               >> example "istarget" or "hop". For the
following
		reasons:
		               >> 1. It is much more intuitive for the
user of this
		information, its
		               >> meaning can basically be guessed.
		               >> 2. Such meaning also probably survives
application in
		other problem
		               >> spaces that we had not foreseen when
introducing the
		concept.
		               >>
		               >> The solution that is  described now in
the target-URI
		delivery draft
		               >> is not yet complete, it does not solve
the freephone
		use case for
		               >> example and in its current form it
will deliver
		exactly the same
		               >> target-URI as that which would be
delivered by the
		P-Called-Party-ID
	=09
		               >> header. Contrary to what the draft
says. So some work
		is needed
		               still.
		               >>
		               >> /Hans Erik van Elburg
		               >>
		               >>
		               >> On Wed, Mar 11, 2009 at 7:13 AM, Shida
Schubert
		<shida@ntt-at.com
		               >> <mailto:shida@ntt-at.com>> wrote:
		               >>
		               >>
		               >>     We have submitted the updated
target-uri draft
		based on the
		               >>
		               > comments
		               >
		               >>     submitted to the list and comments
received at
		IETF73.
		               >>
		               >>     I have taken over as editor as
Jonathan didn't
		have the cycles to
	=09
		               >>     update the draft, with Francois,
Christer and
		Hans Erick as
		               >>     additional
		               >>     co-authors and great deal of help
from Mary.
		               >>
		               >>     The following summarizes the
changes made to the
		target-uri
		               >>
		               > document
		               >
		               >>     1. Added use-case for toll-free
number back
		               >>     2. Added definition of "retarget"
operation.
		               >>     3. Removed a reference to URN
		               >>     4. Added a text discussing the
difference to
		P-Called-Party-Id
		               >>     5. Changed parameter name from
"target" to
		"istarget"
		               >>
		               >>     Note, that the target-uri document
still contains
		the normative
		               >>     text for the
		               >>     History-Info header.
		               >>
		               >>     In addition, Mary (with Francois
as co-author)
		has submitted a
		               >>     rfc4244bis, with the following
changes:
		               >>     1. Incorporated the normative
aspects of the
		target-uri document
		               >>     into the existing normative text
in RFC 4244 -
		the functionality
		               >>
		               > is
		               >
		               >>     virtually identical (as is some of
the text) as
		the HI based
		               >>
		               > solution
		               >
		               >>     described in the target-uri
document.  It's
		important that the
		               >>     solution
		               >>     be integrated into RFC 4244 as it
MUST work and
		be based on the
		               >>     normative
		               >>     aspects of RFC 4244.
		               >>     2.  Added the use cases from
target-uri the the
		summary
		               >>     in the overview of rfc4244bis.
		               >>     3. Added an additional requirement
to capture the
		"target-uri"
		               >>     information.
		               >>     4. Fixed an error in the RFC 4244
ABNF and added
		"retarget"
		               >>
		               > parameter.
		               >
		               >>     5. Added a more simplified
example.
		               >>
		               >>
		               >>     We had some very long offline
exchanges as to the
		best way
		               >> forward
		               >>
		               > and
		               >
		               >>     remaining work for both documents.
		               >>
		               >>     Some of the issues identified are:
		               >>
		               >>     ::Issues::
		               >>        1. Should we remove the
normative text from
		target-uri and
		               >>
		               > progress
		               >
		               >>            4244bis along with the
target-uri document
		to meet the
		               >>     chartered
		               >>           SIP WG milestone?
		               >>
		               >>
		               >>        2. Name of the parameter.
		               >>           At the last meeting,
parameter "target" was
		said
		               >>
		               > inappropriate
		               >
		               >>           because voicemail-uri spec
already defines
		a parameter
		               >>
		               > called
		               >
		               >>           "target" which also can be
found in
		hi-entry, thus
		               >>
		               > potentially
		               >
		               >>           causing confusion.
		               >>
		               >>           Currently the target-uri
draft uses
		"istarget" and 4244bis
		               >>
		               > uses
		               >
		               >>           "retarget"  but we could
never come to
		               >>           a consensus on what name is
appropriate.
		Other suggestions
		               >>
		               > have
		               >
		               >>           included the following:
		               >>           "target-identity" (someone
didn't like that
		"identity" is
		               >>     also a SIP header)
		               >>           "reg-uri"  (can be paired
with "mapped-uri"
		for item 3
		               >>
		               > below)
		               >
		               >>           "aor"
		               >>           "jibberish"
		               >>           etc.
		               >>
		               >>           One reason this is so
difficult relates to
		the problem
		               >>     statement in target-uri in that
		               >>           RFC 3261 doesn't
differentiate the
		mechanism by which the
		               >>
		               > new
		               >
		               >>           (target) Request-URI is
selected.  Another
		issue is that
		               >>     some of the terminology in
		               >>           RFC 3261 is overloaded -
e.g., "forwarding"
		refers both to
		               a
		               >>     Proxy
		               >>           which does not have
responsibility for the
		domain of the
		               >>     request-URI
		               >>           in the incoming request,
thus the proxy
		just "forwards" the
		               >>     request to
		               >>           the next hop AND
"forwarding" is used to
		describe the
		               >>     process whereby
		               >>           the outgoing request is
built and
		"forwarded" to the next
		               >>     hop at which
		               >>           point the proxy does not
know how the new
		request-uri was
		               >>     selected.
		               >>           RFC 4244 has attempted to
clarify the terms
		and attempts to
		               >>     use "forward"
		               >>           in the context of the former
situation and
		"retarget" for
		               >>     the case whereby
		               >>           a proxy is responsible for
the domain and
		thus can use a
		               >>     number of
		               >>           mechanism to select the new
target for the
		request - e.g.,
		               a
		               >>     REGISTRAR,
		               >>           configured data, etc.
		               >>
		               >>      3.  Related to the last point in
item 2 above,
		it has been
		               >>     proposed that
		               >>           we differentiate the
hi-entries even more
		by defining
		               >>     separate parameters
		               >>          for registered and
configured/mapped
		contacts.
		               >>          Currently when the R-URI is
translated to a
		URI which is
		               >>     either derived
		               >>          from location service
lookup(registered by
		UA) or from
		               >>
		               > mapping
		               >
		               >>          table, there is no
differentiation as to how
		the URI was
		               >>     derived once it is
		               >>          added to the list of
potential targets.
		               >>
		               >>          The general consensus of the
authors of the
		two documents
		               was
		               >>     that it may
		               >>          be useful for some services
to have the
		hi-entries tagged
		               >>     with the
		               >>          more specific information.
		               >>
		               >>          And, of course, this gets us
into another
		naming contest. In
		               >>     the end, the naming
		               >>          is not so important as long
as the term
		isn't too overloaded
		               >>     and it is defined
		               >>          precisely in the document(s).
		               >>
		               >>     We would appreciate WG feedback on
these issues
		and any other
		               >>     comments on
		               >>     the two documents prior to
IETF-74.
		               >>
		               >>     Regards,
		               >>     Shida and Mary.
		               >>
		               >>
		               >>
		               >>     Begin forwarded message:
		               >>
		               >>>     *From: *Internet-Drafts@ietf.org
		               >>>
		               > <mailto:Internet-Drafts@ietf.org>
		               >
		               >>>     *Date: *March 10, 2009 2:30:01 AM
JST
		               >>>     *To: *i-d-announce@ietf.org
		<mailto:i-d-announce@ietf.org>
		               >>>     *Subject: **I-D
		               >>>
		ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt *
		               >>>     *Reply-To:
*internet-drafts@ietf.org
		               >>>     <mailto:internet-drafts@ietf.org>
		               >>>
		               >>>     A New Internet-Draft is available
from the
		on-line
		               >>>
		               > Internet-Drafts
		               >
		               >>>     directories.
		               >>>
		               >>>
		               >>>     Title
		               >>>     : Delivery of Request-URI Targets
to User Agents
		               >>>     Author(s)
		               >>>     : J. Rosenberg, H. van Elburg, C.
Holmberg, F.
		Audet, S.
		               Schubert
		               >>>     Filename :
		draft-rosenberg-sip-target-uri-delivery-01.txt
		               >>>     Pages
		               >>>     : 16
		               >>>     Date
		               >>>     : 2009-3-9
		               >>>
		               >>>     When a Session Initiation
Protocol (SIP) proxy
		receives a
		               request
		               >>>       targeted at a URI identifying a
user or
		resource it is
		               >>>
		               > responsible
		               >
		               >>>       for, the proxy translates the
URI to a
		registered or
		               configured
		               >>>       contact URI of an agent
representing that user
		or resource.
		               >>> In
		               >>>
		               > the
		               >
		               >>>       process, the original URI is
removed from the
		request.
		               >>>      Numerous use
		               >>>       cases have arisen which require
this
		information to be
		               >>>
		               > delivered to
		               >
		               >>>       the user agent.  This document
describes these
		use cases and
		               >>>     defines
		               >>>       an extension to the
History-Info header field
		which allows it
		               >>>
		               > to be
		               >
		               >>>       used to support those cases.
		               >>>
		               >>>     A URL for this Internet-Draft is:
		               >>>
		               >>>
=09
http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri-d
		               >>> e
		               >>> livery-01.txt
		               >>>
		               >>>     Internet-Drafts are also
available by anonymous
		FTP at:
		               >>>
ftp://ftp.ietf.org/internet-drafts/
		               >>>
		               >>>     Below is the data which will
enable a MIME
		compliant mail reader
		               >>>     implementation to automatically
retrieve the
		ASCII version of
		               the
		               >>>     Internet-Draft.
		               >>>
		               >>     Content-Type:
text/plain<BR>Content-ID:
		               >>     &lt;2009-3-9102756.I-D@ietf.org
<mailto:lt%3B2009-3-9102756.I-D@ietf.org>=20
	=09
		<mailto:lt%3B2009-3-9102756.I-D@ietf.org
<mailto:lt%253B2009-3-9102756.I-D@ietf.org> >
	=09
		               >>
<mailto:2009-3-9102756.I-D@ietf.org>&gt;<BR><BR>
		               >>
		               >>>
_______________________________________________
		               >>>     I-D-Announce mailing list
		               >>>     I-D-Announce@ietf.org
		<mailto:I-D-Announce@ietf.org>
		               >>>
		https://www.ietf.org/mailman/listinfo/i-d-announce
		               >>>     Internet-Draft directories:
		http://www.ietf.org/shadow.html
		               >>>     or
ftp://ftp.ietf.org/ietf/1shadow-sites.txt
		               >>>
		               >>
_______________________________________________
		               >>     Sip mailing list
		https://www.ietf.org/mailman/listinfo/sip
		               >>     This list is for NEW development
of the core SIP
		Protocol
		               >>     Use
sip-implementors@cs.columbia.edu
		               >>
<mailto:sip-implementors@cs.columbia.edu> for
		questions on
		               >> current
		               >>
		               > sip
		               >
		               >>     Use sipping@ietf.org
<mailto:sipping@ietf.org>
		for new
		               >>     developments on the application of
sip
		               >>
		               >>
		               >>
	=09
	=09
	=09
	=09



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<DIV dir=3Dltr align=3Dleft>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D441300613-12032009><FONT =
face=3DArial=20
color=3D#000080>Hans Erik,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D441300613-12032009><FONT =
face=3DArial=20
color=3D#000080></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D441300613-12032009><FONT =
face=3DArial><FONT=20
color=3D#000080>The way I interpreted the 01 draft, you would not get =
the=20
freephone number anyway. If that is a requirement, then I accept my =
suggestion=20
won't suffice<SPAN =
class=3D125123013-12032009>.</SPAN></FONT></FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D441300613-12032009><FONT =
face=3DArial=20
color=3D#000080></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D441300613-12032009><FONT =
face=3DArial=20
color=3D#000080>John</FONT></SPAN></DIV></DIV><BR>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000080 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Hans Erik van Elburg=20
  [mailto:ietf.hanserik@gmail.com] <BR><B>Sent:</B> 12 March 2009=20
  12:57<BR><B>To:</B> Elwell, John<BR><B>Cc:</B> =
sip@ietf.org<BR><B>Subject:</B>=20
  Re: [Sip] Terminology (was RE: Fwd:=20
  =
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR></FONT><BR></=
DIV>
  <DIV></DIV>But that does not solve the freephone service use case.=20
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.tsDisplay { display:block }</STYLE>
  Freephone represents a use case where the R-URI is replaced with an =
AoR of=20
  user B - not with the contact of user B. <BR><BR>A calls =
Bfreephonenumber=20
  routed the B (AOR) routed to B's contact.<BR><BR>The target-uri draft =
is=20
  interested in obtaining: Bfreephonenumber as that is what A used to =
reach=20
  B.<BR><BR>So what we like to see that B UA receives is something=20
  like<BR>&nbsp;&nbsp;&nbsp; H-I:&nbsp; Bfreephone;=20
  "istarget",\<BR>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;=20
  B,\<BR>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; contact-B<BR><BR>So you =
want to=20
  get Bfreephonenumber and not B.<BR><BR>In certain situations you are =
also=20
  interested in B, I guess that&nbsp; PBX's would interested in that to =
deliver=20
  to the correct user. We could distinguish them =
like:<BR>&nbsp;&nbsp;&nbsp;=20
  H-I:&nbsp; Bfreephone; "istarget",\<BR>&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
  B,"aor"\<BR>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
contact-B<BR><BR>BR,<BR=20
  clear=3Dall>/Hans Erik van Elburg<BR><BR><BR>
  <DIV class=3Dgmail_quote>On Thu, Mar 12, 2009 at 12:08 PM, Elwell, =
John <SPAN=20
  dir=3Dltr>&lt;<A=20
  =
href=3D"mailto:john.elwell@siemens.com">john.elwell@siemens.com</A>&gt;</=
SPAN>=20
  wrote:<BR>
  <BLOCKQUOTE class=3Dgmail_quote=20
  style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">If=20
    'istarget' is used only to identify the value that was overwritten =
in<BR>the=20
    Request-URI by a contact URI, an alternative approach would be =
to<BR>flag=20
    the contact URI H-I entry as 'iscontact'. Then the UAS would =
just<BR>need to=20
    look for the most recent H-I entry that is not=20
    =
marked<BR>'iscontact'.<BR><BR>John<BR><BR><BR>___________________________=
_____<BR><BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp;From: <A=20
    href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</A> =
[mailto:<A=20
    href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</A>] =
On<BR>Behalf Of=20
    Hans Erik van Elburg<BR>&nbsp; &nbsp; &nbsp; &nbsp;Sent: 12 March =
2009=20
    08:35<BR>&nbsp; &nbsp; &nbsp; &nbsp;To: Mary Barnes<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp;Cc: <A href=3D"mailto:sip@ietf.org">sip@ietf.org</A><BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp;Subject: Re: [Sip] Terminology (was RE:=20
    Fwd:<BR>I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR>
    <DIV>
    <DIV></DIV>
    <DIV class=3Dh5><BR><BR>&nbsp; &nbsp; &nbsp; &nbsp;Yes, because you =
are using=20
    the 3261 use of target and the<BR>4244bis introduced definition of =
retarget.=20
    I thought it was clear that<BR>we need other words as those =
definitions=20
    don't match the target-uri<BR>drafts use of the terms. Also they do =
not=20
    suffice to provide a solution<BR>for the use cases in the target-uri =

    draft.<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp;The 3261 text you refer to =
is=20
    exactly about the case where the<BR>home proxy overwrites the =
Request-URI=20
    with a new target. This target is<BR>teh registered contact address. =
And=20
    hence this would be what target-uri<BR>calls a hop or a route. This =
case and=20
    this is where it gets confusing is<BR>not a "retarget" in the =
target-uri=20
    draft use of the term.<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp;The =
target-uri=20
    draft states:<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; "To avoid =
confusion,=20
    we<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; refer to a SIP URI that is =
an=20
    address for a user or resource<BR>as a<BR>&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
    "target" and a SIP URI that is a hop for reaching that user<BR>as=20
    a<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; "hop".<BR><BR><BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp;Apparently that does not suffice to avoid=20
    confusion.<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp;As for the tagging, =
speaking=20
    about the solution before agreeing<BR>on the terminology and the =
problem it=20
    should solve is meaningless.<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp;/Hans =
Erik=20
    van Elburg<BR><BR><BR><BR>&nbsp; &nbsp; &nbsp; &nbsp;On Thu, Mar 12, =
2009 at=20
    2:33 AM, Mary Barnes<BR>&lt;<A=20
    =
href=3D"mailto:mary.barnes@nortel.com">mary.barnes@nortel.com</A>&gt;=20
    wrote:<BR><BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;Responses below [MB].<BR><BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;-----Original Message-----<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;From: Hans Erik van =
Elburg<BR>[mailto:<A=20
    =
href=3D"mailto:ietf.hanserik@gmail.com">ietf.hanserik@gmail.com</A>]<BR><=
BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Sent: Wednesday, =
March 11,=20
    2009 6:42 PM<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;To:=20
    Barnes, Mary (RICH2:AR00)<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;Cc: Shida Schubert; <A=20
    href=3D"mailto:sip@ietf.org">sip@ietf.org</A><BR>&nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;Subject: Re: Terminology (was RE: [Sip] =
Fwd:=20
    I-D<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    =
&nbsp;ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR><BR><BR>&n=
bsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;I was talking about =
the=20
    concept.<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;The=20
    use cases only describe &nbsp;cases where the target-uri<BR>(lets =
call=20
    that<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;current=20
    target from now) has been lost when an initial<BR>request for =
a<BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;dialog or standalone =
request=20
    arrives at the UAS.<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;[MB] And I think this is where the terminology =
confusion<BR>starts -=20
    I<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;don't =
think of=20
    the "lost" target-uri as being the<BR>"current" target. In<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;my mind, the "current" =
target is=20
    reflected by the last<BR>hi-entry and the<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;request-uri in the incoming request at =
the UAS.=20
    If the<BR>entity that sent<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;the request was the entity that added the last=20
    hi-entry,<BR>then the uri in<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;that hi-entry is the same as the request-URI in the=20
    SIP<BR>request that<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;arrives at the UAS. I refer to the "lost" uri as the =
one<BR>that=20
    was<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;"retargeted" -=20
    that's the one the UAS wants to pull from<BR>the =
hi-entries<BR>&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;in the incoming request. =
That=20
    hi-entry was not the one<BR>that was just<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;added by the entity that built the =
request just=20
    received<BR>by the UAS.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;That hi-entry is tagged with whatever name we are going<BR>to =
tag it=20
    with<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;BEFORE the=20
    request is forwarded (using the term forward<BR>per section =
16.6<BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;of RFC 3261). =
&nbsp;That tag=20
    is added once the target set of<BR>potential<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;candidates for the new request uri are =
determined=20
    in<BR>section 16.5 of<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;3261 (with "target set" being a 3261 term), just before<BR>the =
request=20
    is<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;forwarded in=20
    section 16.6 to one of those targets. &nbsp;A<BR>new =
hi-entry<BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;(which will be the =
last=20
    hi-entry in the request received<BR>by the UAS) is<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;added in section 16.6 of 3261 as =
the=20
    request is<BR>forwarded. &nbsp;At this<BR>&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp;point in time, the lost information is in the=20
    previous<BR>hi-entry when the<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;outgoing request is sent. [/MB]<BR><BR><BR>&nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;What has been lost is the current =
target=20
    of the request:<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;[MB] Right - at which point in my mind, it's no longer<BR>the=20
    current<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;target ;)=20
    Maybe we call it "lost". [/MB]<BR><BR><BR><BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Current =
target<BR><BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;The current =
target of=20
    an initial request for a dialog<BR>or standalone<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;request is the name or =
address to=20
    which the request is<BR>targeted, i.e.<BR>&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;either the initial target inserted in the =

    Request-URI<BR>by the UAC that<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;originates the request, or when a retarget=20
    occurred,<BR>the target<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;provided in that retarget operation. &nbsp;Reroute=20
    and<BR>translation<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;operations never change the current =
target.<BR><BR><BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;[MB] I don't think =
this=20
    definition fits what you want -<BR>i.e., if there<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;is no retargeting, then none of =
the=20
    hi-entries are<BR>tagged - i.e., you<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;won't have your concept of current. The way it =
works=20
    is<BR>that if no<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

    &nbsp;hi-entries are tagged, then you know that there was =
no<BR>retargeting,=20
    thus<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;you =
know the=20
    request-uri has not been lost. [/MB]<BR><BR><BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;This defintion only makes sense when the=20
    following<BR>definitions are used:<BR><BR>&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp;Name:<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;A name is a moniker for an entity which refers =
to=20
    it<BR>in a way which<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp;reveals nothing about where it is in a network.=20
    &nbsp;In<BR>SIP, tel URI<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;which doesn't represent the location of the =
entity=20
    is<BR>a name.<BR><BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;Address:<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;An address is an identifier for an entity =
which<BR>describes it=20
    by its<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;location on the network. &nbsp;In SIP, the SIP URI =
itself<BR>is a form=20
    of<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;address because the host part of the URI, the =
only<BR>mandatory part=20
    of<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;the=20
    URI besides the scheme itself, indicates the<BR>location of a=20
    SIP<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;server that can be used to handle the request.<BR><BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Route:<BR><BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Finally, a route is a =
&nbsp;=20
    sequence of SIP entities<BR>(including the UA<BR>&nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;itself!) which are<BR><BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; traversed in order to forward a =
request=20
    to an address<BR>or name.<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;Retarget (other term might be needed, as this is=20
    highly<BR>confusing):<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; A Request-URI rewrite operation that changes=20
    the<BR>target identity of<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;the request.<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;Reroute (other term might be needed):<BR><BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; A Request-URI =
rewrite=20
    operation that does not change<BR>the current<BR>&nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;target of the request, but &nbsp; &nbsp;=20
    determines the route/next<BR>hop taken to<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;reach the target-identity.<BR><BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;/Hans Erik<BR><BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Mary Barnes =
wrote:<BR>&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt; So, by "target-uri =
concept"=20
    are you referring to the<BR>solution in the<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt; current target-uri draft or the =
application=20
    usage as<BR>such in the<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt; target-uri document (which is what I was=20
    referring<BR>to)?<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;=20
    Yes, the problem is that SIP doesn't differentiate =
the<BR>various<BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt; mechanisms by =
which a=20
    request-URI may be changed,<BR>which is what some<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt; of the updated text in =
4244bis is=20
    trying to do and why<BR>we need precise<BR><BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt; terms. I think we all agree =
that.<BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt; I honestly don't care what we =
call=20
    the tag, as long as<BR>we're clear<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt; about the functionality.<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt; Mary.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt; -----Original Message-----<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt; From: Hans Erik van Elburg<BR>[mailto:<A=20
    =
href=3D"mailto:ietf.hanserik@gmail.com">ietf.hanserik@gmail.com</A>]<BR>&=
nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt; Sent: =
Wednesday, March=20
    11, 2009 4:40 PM<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

    &nbsp;&gt; To: Barnes, Mary (RICH2:AR00)<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt; Cc: Shida Schubert; <A=20
    href=3D"mailto:sip@ietf.org">sip@ietf.org</A><BR>&nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt; Subject: Re: Terminology (was RE: =
[Sip] Fwd:=20
    I-D<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;=20
    ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt; The target-uri concept when defined =
properly=20
    is<BR>entirely application<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt; agnostic.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt; When one uses History-Info or another header as =
a<BR>vehicle for=20
    this<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;=20
    information then it still can be considered<BR>application agnostic=20
    as<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt; =
any=20
    application can use this information as it likes.<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt; &nbsp;From SIP perspective all the =
Request-URI=20
    rewrites<BR>look the same, that<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt; is what brought us discussing this in the =
first=20
    place.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;=20
    /Hans Erik<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;=20
    Mary Barnes wrote:<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; And, I think the big issue with the terminology =
is<BR>that=20
    4244bis is<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; application agnostic, so it is tagging what =
happens<BR>to the=20
    URIs<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
    (why/how the specific URI was overwritten).=20
    &nbsp;Whereas,<BR>specific<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt; applications can derive information, such as=20
    what<BR>"is" the real<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt; "target" for the request, from the =
History-Info<BR>header.=20
    &nbsp; So, IMHO,<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

    &nbsp;&gt;&gt; it's a matter of the target-uri document =
specifying<BR>that=20
    the last<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
    entry tagged by whatever name we come up with "is"<BR>the real=20
    "target"<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
    for the request. &nbsp;I believe it's up to the<BR>applications to=20
    describe<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
    how they make use of the History-Info and not for =
the<BR>History-Info=20
    to<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
    describe how applications can use it. &nbsp;History-Info<BR>purely=20
    reflects<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
    what happened from a SIP Protocol perspective and<BR>should not=20
    imply<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
    any<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; application semantics. Indeed the intent of section =
5<BR>of=20
    4244 was to<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; inform the applications how they should decribe=20
    their<BR>usage of the<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt; header.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; Mary.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt; Note: I changed the subject to hopefully make =
this<BR>topic=20
    easier to<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
    keep track of.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    =
&nbsp;&gt;&gt;<BR>-------------------------------------------------------=
--------------<BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; -<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; --<BR>&nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; *From:* Hans Erik van=20
    Elburg<BR>[mailto:<A=20
    =
href=3D"mailto:ietf.hanserik@gmail.com">ietf.hanserik@gmail.com</A>]<BR>&=
nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; *Sent:* =
Wednesday,=20
    March 11, 2009 3:59 AM<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt; *To:* Shida Schubert<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt;&gt; *Cc:* <A=20
    href=3D"mailto:sip@ietf.org">sip@ietf.org</A> List; Barnes, Mary=20
    (RICH2:AR00)<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; *Subject:* Re: [Sip] Fwd: I-D<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;=20
    ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; One of the problems that =
made the=20
    discussion quite<BR>tedious is the<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt;&gt; complete inversed meaning of the terms=20
    "retarget" and<BR>"reroute"<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt; between 4244bis and target-uri factions=20
    have.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; Conclusion was that the terminology need to =
be<BR>properly=20
    defined and<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; some attempt was made to start such =
activity.<BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; On 3. I have =
strong=20
    concerns when we start tagging<BR>the URI as to how<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; they come about=20
    "retarget"/"mapped conficuration"<BR>etc. It is better<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; to<BR>&nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt; embellish them with a tag that just represents=20
    there<BR>meaning for<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt; example "istarget" or "hop". For the=20
    following<BR>reasons:<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt; 1. It is much more intuitive for the user of=20
    this<BR>information, its<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp;&gt;&gt; meaning can basically be guessed.<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; 2. Such meaning also =
probably=20
    survives application in<BR>other problem<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; spaces that we had not foreseen =
when=20
    introducing the<BR>concept.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt; The solution that is &nbsp;described now in the=20
    target-URI<BR>delivery draft<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt; is not yet complete, it does not solve the=20
    freephone<BR>use case for<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt; example and in its current form it will=20
    deliver<BR>exactly the same<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt; target-URI as that which would be delivered by =

    the<BR>P-Called-Party-ID<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt; header. Contrary to what the draft says. So =
some=20
    work<BR>is needed<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;still.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; /Hans Erik van Elburg<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt; On Wed, Mar 11, 2009 at 7:13 AM, Shida =
Schubert<BR>&lt;<A=20
    href=3D"mailto:shida@ntt-at.com">shida@ntt-at.com</A><BR>&nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &lt;mailto:<A=20
    href=3D"mailto:shida@ntt-at.com">shida@ntt-at.com</A>&gt;&gt; =
wrote:<BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; We have =
submitted=20
    the updated target-uri draft<BR>based on the<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt; comments<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; submitted to the list and comments =
received=20
    at<BR>IETF73.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; I have taken over as editor as Jonathan =

    didn't<BR>have the cycles to<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; update the draft, with =
Francois,=20
    Christer and<BR>Hans Erick as<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; additional<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; co-authors =
and=20
    great deal of help from Mary.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; The following summarizes the changes =
made to=20
    the<BR>target-uri<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt; document<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; 1. Added use-case for toll-free number=20
    back<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
    &nbsp; &nbsp; 2. Added definition of "retarget" operation.<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; 3. =
Removed a=20
    reference to URN<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

    &nbsp;&gt;&gt; &nbsp; &nbsp; 4. Added a text discussing the =
difference=20
    to<BR>P-Called-Party-Id<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; 5. Changed parameter name from "target" =

    to<BR>"istarget"<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

    &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; Note, that the target-uri document =
still=20
    contains<BR>the normative<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; text for the<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; =
History-Info=20
    header.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; In addition, Mary (with Francois as=20
    co-author)<BR>has submitted a<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; rfc4244bis, with the following=20
    changes:<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
    &nbsp; &nbsp; 1. Incorporated the normative aspects of =
the<BR>target-uri=20
    document<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
    &nbsp; &nbsp; into the existing normative text in RFC 4244 -<BR>the=20
    functionality<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt; is<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; virtually identical (as is some of the =
text)=20
    as<BR>the HI based<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt; solution<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; described in the target-uri document.=20
    &nbsp;It's<BR>important that the<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; solution<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; be =
integrated into=20
    RFC 4244 as it MUST work and<BR>be based on the<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; =
normative<BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; =
&nbsp;=20
    aspects of RFC 4244.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; 2. &nbsp;Added the use cases from =
target-uri=20
    the the<BR>summary<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; in the overview of =
rfc4244bis.<BR>&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; 3. =
Added an=20
    additional requirement to capture the<BR>"target-uri"<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp;=20
    information.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; 4. Fixed an error in the RFC 4244 ABNF =
and=20
    added<BR>"retarget"<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt; parameter.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; 5. Added a more simplified =
example.<BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; We had some =
very=20
    long offline exchanges as to the<BR>best way<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; forward<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt; and<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; remaining work for both =
documents.<BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; Some =
of the=20
    issues identified are:<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; ::Issues::<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp;1. =
Should we=20
    remove the normative text from<BR>target-uri and<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt; progress<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;4244bis along=20
    with the target-uri document<BR>to meet the<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; =
chartered<BR>&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; SIP WG milestone?<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp;2. Name of the=20
    parameter.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; At the last =
meeting,=20
    parameter "target" was<BR>said<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt; inappropriate<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; because =
voicemail-uri spec=20
    already defines<BR>a parameter<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt; called<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; "target" which =
also can be=20
    found in<BR>hi-entry, thus<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt; potentially<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; causing=20
    confusion.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Currently the =
target-uri=20
    draft uses<BR>"istarget" and 4244bis<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt; uses<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; "retarget" =
&nbsp;but we=20
    could never come to<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; a consensus on =
what name=20
    is appropriate.<BR>Other suggestions<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt; have<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; included the=20
    following:<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; "target-identity" =
(someone=20
    didn't like that<BR>"identity" is<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; also a SIP =
header)<BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; "reg-uri" &nbsp;(can be paired with =
"mapped-uri"<BR>for=20
    item 3<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt; below)<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; "aor"<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; "jibberish"<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
etc.<BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; One reason this is so difficult relates to<BR>the=20
    problem<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
    &nbsp; &nbsp; statement in target-uri in that<BR>&nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; RFC=20
    3261 doesn't differentiate the<BR>mechanism by which the<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt; new<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; (target)=20
    Request-URI is selected. &nbsp;Another<BR>issue is that<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; some =
of the=20
    terminology in<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; RFC 3261 is =
overloaded -=20
    e.g., "forwarding"<BR>refers both to<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;a<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; Proxy<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
which does=20
    not have responsibility for the<BR>domain of the<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp;=20
    request-URI<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; in the incoming =
request,=20
    thus the proxy<BR>just "forwards" the<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; request to<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; the next hop AND "forwarding" is used to<BR>describe=20
    the<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
    &nbsp; &nbsp; process whereby<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; the =
outgoing=20
    request is built and<BR>"forwarded" to the next<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; hop at=20
    which<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; point the proxy does not know how =
the=20
    new<BR>request-uri was<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; selected.<BR>&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; RFC =
4244 has=20
    attempted to clarify the terms<BR>and attempts to<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; use=20
    "forward"<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; in the context of the former =
situation=20
    and<BR>"retarget" for<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; the case whereby<BR>&nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; a=20
    proxy is responsible for the domain and<BR>thus can use a<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; =
number=20
    of<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt; &nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; mechanism to select the new target for=20
    the<BR>request - e.g.,<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;a<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
    &nbsp; &nbsp; REGISTRAR,<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; configured data,=20
    etc.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp;3. &nbsp;Related to the last =
point in=20
    item 2 above,<BR>it has been<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; proposed that<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; we differentiate the hi-entries even more<BR>by =
defining<BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; =
&nbsp;=20
    separate parameters<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;for registered and=20
    configured/mapped<BR>contacts.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Currently =
when the=20
    R-URI is translated to a<BR>URI which is<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; either =
derived<BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;from location service lookup(registered =
by<BR>UA) or=20
    from<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt; mapping<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;table, there is no=20
    differentiation as to how<BR>the URI was<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; derived once it=20
    is<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt; &nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;added to the list of potential =
targets.<BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;The general consensus of the authors of the<BR>two=20
    documents<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;was<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
    &nbsp; &nbsp; that it may<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;be useful =
for some=20
    services to have the<BR>hi-entries tagged<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; with the<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;more specific information.<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;And, =
of=20
    course, this gets us into another<BR>naming contest. In<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; the =
end, the=20
    naming<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;is not so important as long as the =

    term<BR>isn't too overloaded<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; and it is defined<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;precisely in the document(s).<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; We would appreciate WG =
feedback=20
    on these issues<BR>and any other<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; comments on<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; the =
two=20
    documents prior to IETF-74.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt; &nbsp; &nbsp; Regards,<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; Shida and Mary.<BR>&nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; Begin forwarded=20
    message:<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; *From: *<A=20
    =
href=3D"mailto:Internet-Drafts@ietf.org">Internet-Drafts@ietf.org</A><BR>=
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;&gt;<BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt; &lt;mailto:<A=20
    =
href=3D"mailto:Internet-Drafts@ietf.org">Internet-Drafts@ietf.org</A>&gt;=
<BR>&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; *Date: =
*March=20
    10, 2009 2:30:01 AM JST<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; *To: *<A=20
    =
href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</A><BR>&lt;ma=
ilto:<A=20
    =
href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</A>&gt;<BR>&n=
bsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; &nbsp; =
&nbsp;=20
    *Subject: **I-D<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    =
&nbsp;&gt;&gt;&gt;<BR>ACTION:draft-rosenberg-sip-target-uri-delivery-01.t=
xt=20
    *<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;&gt;=20
    &nbsp; &nbsp; *Reply-To: *<A=20
    =
href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</A><BR>=
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; &nbsp; =
&nbsp;=20
    &lt;mailto:<A=20
    =
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    &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; : J. Rosenberg, H. van Elburg, C. =
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    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; &nbsp; &nbsp;=20
    defines<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
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arget-uri-d</A><BR>&nbsp;=20
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From ian.elz@ericsson.com  Thu Mar 12 06:40:51 2009
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From: "Ian Elz" <ian.elz@ericsson.com>
To: "Mary Barnes" <mary.barnes@nortel.com>
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Cc: sip@ietf.org, Francois Audet <audet@nortel.com>
Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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Mary,

(Written after reading your discussion with Hans Erik.)

First an apology for not making it clear in my original email that a
value of "none" in a Privacy header parameter in a History-Info entry
was only applicable to the uri contained in the hi-targeted-to-uri.

I only suggested using the value "none" as this is a value defined for
the Privacy header which explicitly indicates that privacy is not
required. Any other value with the meaning of explicitly indicating that
there is no privacy required for the specific uri could be used.

With regard to your second paragraph below I understand that the SIP and
PSTN models do not align. You are correct in your assumption that this
problem is not unique for History-Info. The problem also arises with the
Referred-By header.=20

I have raised this for History-Info at this time as your new draft
presents an opportunity to correct this problem for the H-I case.

The general SIP privacy issue arises as when the Privacy header was
defined the concept of carrying third party identities in the SIP
messages was not defined. This came later with the REFER Request and the
associated Referred-By header and the History-Info header.=20

We should take what opportunities arise to discuss and correct this if
there is a will to make these changes.

Ian Elz

Office: + 44 24 764 35256
ian.elz@ericsson.com


-----Original Message-----
From: Mary Barnes [mailto:mary.barnes@nortel.com]=20
Sent: 11 March 2009 16:53
To: Ian Elz; Francois Audet
Cc: sip@ietf.org
Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

Ian,

This is an interesting question. I need to think about it some more, but
I don't like the "none" idea as it really must be up to the entity that
added the Privacy header to the request, as to whether it wants the HI
entries that it adds to go outside a domain. My initial thought is that
we can't overrule the overall Privacy header. The thing is that the
Privacy header doesn't preclude gathering the information and using
within a domain AND if it were to not include an hi-entry when the
request leaves the domain for which the proxy is responsible, the
recipient can add the hi-entry for the current request-uri before it
adds the new hi-entry. =20

Your PSTN example doesn't strictly map obviously to a SIP model and I
would think you might consider the PSTN hop to be within the same domain
for which the proxy (or gw, I guess) is responsible OR consider this a
walled garden whereby the Privacy header doesn't apply. Which brings me
to a more general question as to how you all deal with other headers
when you do your mapping to the PSTN when there is a Privacy header?  It
would seem this problem isn't unique to History-Info, although I know
little about the details of PSTN I/W.

Regards,
Mary.=20

-----Original Message-----
From: Ian Elz [mailto:ian.elz@ericsson.com]=20
Sent: Wednesday, March 11, 2009 4:54 AM
To: Barnes, Mary (RICH2:AR00); Audet, Francois (SC100:3055)
Cc: sip@ietf.org
Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

Mary, Francois,

Section 4.1 of the draft allows for the addition of the Privacy header
parameter with a value of "history" to be included with the
hi-targeted-to-uri.

Can this be extended to also allow the value "none".

A node adding a H-I entry is allowed to specify the privacy value
"history" either in the Privacy header or as a header parameter
associated with the hi-targeted-to-uri. If the former action is taken by
one node this results in privacy for all history info header entries
even if this is not required. There is no way for privacy of an
individual H-I entry to be set to none if no Privacy is required for the
uri.

This creates a specific problem when interworking with the PSTN where
the privacy is associated with each E.164 number included in the
protocol. If an INVITE is being mapped to ISUP and the H-I entries are
being used to map to redirection numbers then a single Privacy header
with the value "history" will result in all numbers being restricted.

If the Privacy header parameter could include the value "none" then this
would explicitly indicate that the associated uri was allowed to be
presented.

The ability to explicitly allow presentation of specific H-I entries may
also be useful in pure sip implementations.

I don't believe that this change would create any backward compatibility
issues. The existing implementations will continue as deployed but new
implementations could explicitly set no privacy for a specific uri.

There are no security issues other than those already defined in the
draft that an intervening node could modify an existing H-I entry. The
Privacy header value of "none" should only be included by the node in
the same manner as the existing inclusion of the "history" value.

Ian Elz


-----Original Message-----
From: i-d-announce-bounces@ietf.org
[mailto:i-d-announce-bounces@ietf.org] On Behalf Of
Internet-Drafts@ietf.org
Sent: 04 March 2009 15:15
To: i-d-announce@ietf.org
Subject: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

A New Internet-Draft is available from the on-line Internet-Drafts
directories.

	Title           : An Extension to the Session Initiation
Protocol (SIP) for Request History Information
	Author(s)       : M. Barnes, F. Audet
	Filename        : draft-barnes-sip-rfc4244bis-00.txt
	Pages           : 49
	Date            : 2009-03-04

This document defines a standard mechanism for capturing the history
information associated with a Session Initiation Protocol (SIP) request.
This capability enables many enhanced services by providing the
information as to how and why a call arrives at a specific application
or user.  This document defines a new optional SIP header, History-Info,
for capturing the history information in requests.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-barnes-sip-rfc4244bis-00.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

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From mary.barnes@nortel.com  Thu Mar 12 06:43:06 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Elwell, John" <john.elwell@siemens.com>, "Hans Erik van Elburg" <ietf.hanserik@gmail.com>
Cc: sip@ietf.org
Subject: Re: [Sip] Terminology (was RE: Fwd: I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Yeah, we did looked at that approach, BUT there are two problems:
1) It doesn't take into account the case where it's just a proxy hop to
a new domain  - i.e., those also won't be tagged. We also considered
just flagging those proxy hops, but then you still get into trouble with
the last entry as that isn't what you want - you typically want the
previous entry, which takes us to the next problem. =20
2) You are then back to assuming that the one you want is one back from
the current hi-entry. I thought the whole point was to have it obviously
tagged.=20

I am struggling with why tagging the ones, that we truly know are lost,
at the point we know they're lost won't work?  Isn't that the
information that is really desired? =20

I think the 'iscontact' tag is a fine name - I just think we need to use
that in the manner currently specified in 4244bis for the "retarget"
flag.=20

As far as Hans Erik's point about not discussing implementation, I don't
think not discussing implementation is at all helpful. I think the
problem statement is already fairly well laid out in the target-uri
draft. And, if you plan on using History-Info as the solution, we MUST
understand how it works now and we MUST understand how the proposed
solution works within the existing History-Info processing structure -
it's the just the same as we do with any new SIP header. And, indeed it
was the same thing that was done for the History-Info header to make
sure it was consistent with RFC 3261. The terminology is an issue and I
just think it takes precise wording ONCE we all understand the
implementation - it's got to be compatible with 3261 processing (now, we
can suggest some clarifications to that text and again, the definitions
that are currently in 4244bis are an attempt to do so).

That all said, we can certainly name the tag now and the 'iscontact'
does work for me.

Mary.=20

-----Original Message-----
From: Elwell, John [mailto:john.elwell@siemens.com]=20
Sent: Thursday, March 12, 2009 6:09 AM
To: Hans Erik van Elburg; Barnes, Mary (RICH2:AR00)
Cc: sip@ietf.org
Subject: RE: [Sip] Terminology (was RE: Fwd:
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt

If 'istarget' is used only to identify the value that was overwritten in
the Request-URI by a contact URI, an alternative approach would be to
flag the contact URI H-I entry as 'iscontact'. Then the UAS would just
need to look for the most recent H-I entry that is not marked
'iscontact'.

John


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Hans Erik van Elburg
	Sent: 12 March 2009 08:35
	To: Mary Barnes
	Cc: sip@ietf.org
	Subject: Re: [Sip] Terminology (was RE: Fwd:
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
=09
=09
	Yes, because you are using the 3261 use of target and the
4244bis introduced definition of retarget. I thought it was clear that
we need other words as those definitions don't match the target-uri
drafts use of the terms. Also they do not suffice to provide a solution
for the use cases in the target-uri draft.
=09
	The 3261 text you refer to is exactly about the case where the
home proxy overwrites the Request-URI with a new target. This target is
teh registered contact address. And hence this would be what target-uri
calls a hop or a route. This case and this is where it gets confusing is
not a "retarget" in the target-uri draft use of the term.
=09
	The target-uri draft states:
=09
	   "To avoid confusion, we
	   refer to a SIP URI that is an address for a user or resource
as a
	   "target" and a SIP URI that is a hop for reaching that user
as a
=09
	   "hop".
=09

	Apparently that does not suffice to avoid confusion.
=09
	As for the tagging, speaking about the solution before agreeing
on the terminology and the problem it should solve is meaningless.
=09
	/Hans Erik van Elburg
=09
=09
=09
	On Thu, Mar 12, 2009 at 2:33 AM, Mary Barnes
<mary.barnes@nortel.com> wrote:
=09

		Responses below [MB].
	=09

		-----Original Message-----
		From: Hans Erik van Elburg
[mailto:ietf.hanserik@gmail.com]
	=09
		Sent: Wednesday, March 11, 2009 6:42 PM
		To: Barnes, Mary (RICH2:AR00)
		Cc: Shida Schubert; sip@ietf.org
		Subject: Re: Terminology (was RE: [Sip] Fwd: I-D
		ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
	=09
	=09
		I was talking about the concept.
	=09
		The use cases only describe  cases where the target-uri
(lets call that
		current target from now) has been lost when an initial
request for a
		dialog or standalone request arrives at the UAS.
	=09
		[MB] And I think this is where the terminology confusion
starts - I
		don't think of the "lost" target-uri as being the
"current" target. In
		my mind, the "current" target is reflected by the last
hi-entry and the
		request-uri in the incoming request at the UAS. If the
entity that sent
		the request was the entity that added the last hi-entry,
then the uri in
		that hi-entry is the same as the request-URI in the SIP
request that
		arrives at the UAS. I refer to the "lost" uri as the one
that was
		"retargeted" - that's the one the UAS wants to pull from
the hi-entries
		in the incoming request. That hi-entry was not the one
that was just
		added by the entity that built the request just received
by the UAS.
		That hi-entry is tagged with whatever name we are going
to tag it with
		BEFORE the request is forwarded (using the term forward
per section 16.6
		of RFC 3261).  That tag is added once the target set of
potential
		candidates for the new request uri are determined in
section 16.5 of
		3261 (with "target set" being a 3261 term), just before
the request is
		forwarded in section 16.6 to one of those targets.  A
new hi-entry
		(which will be the last hi-entry in the request received
by the UAS) is
		added in section 16.6 of 3261 as the request is
forwarded.  At this
		point in time, the lost information is in the previous
hi-entry when the
		outgoing request is sent. [/MB]
	=09

		What has been lost is the current target of the request:
	=09
		[MB] Right - at which point in my mind, it's no longer
the current
		target ;) Maybe we call it "lost". [/MB]
	=09


		     Current target
	=09
		  The current target of an initial request for a dialog
or standalone
		  request is the name or address to which the request is
targeted, i.e.
		  either the initial target inserted in the Request-URI
by the UAC that
		  originates the request, or when a retarget occurred,
the target
		  provided in that retarget operation.  Reroute and
translation
		  operations never change the current target.
	=09
	=09
		[MB] I don't think this definition fits what you want -
i.e., if there
		is no retargeting, then none of the hi-entries are
tagged - i.e., you
		won't have your concept of current. The way it works is
that if no
		hi-entries are tagged, then you know that there was no
retargeting, thus
		you know the request-uri has not been lost. [/MB]
	=09

		This defintion only makes sense when the following
definitions are used:
	=09
		Name:
	=09
		  A name is a moniker for an entity which refers to it
in a way which
	=09
		  reveals nothing about where it is in a network.  In
SIP, tel URI
	=09
		  which doesn't represent the location of the entity is
a name.
	=09
	=09
		Address:
	=09
		  An address is an identifier for an entity which
describes it by its
	=09
		  location on the network.  In SIP, the SIP URI itself
is a form of
	=09
		  address because the host part of the URI, the only
mandatory part of
	=09
		  the URI besides the scheme itself, indicates the
location of a SIP
	=09
		  server that can be used to handle the request.
	=09
		Route:
	=09
		  Finally, a route is a   sequence of SIP entities
(including the UA
		itself!) which are
	=09
		 traversed in order to forward a request to an address
or name.
	=09
		Retarget (other term might be needed, as this is highly
confusing):
	=09
		   A Request-URI rewrite operation that changes the
target identity of
		the request.
	=09
		Reroute (other term might be needed):
	=09
		   A Request-URI rewrite operation that does not change
the current
		target of the request, but     determines the route/next
hop taken to
		reach the target-identity.
	=09
		/Hans Erik
	=09
		Mary Barnes wrote:
		> So, by "target-uri concept" are you referring to the
solution in the
		> current target-uri draft or the application usage as
such in the
		> target-uri document (which is what I was referring
to)?
		>
		> Yes, the problem is that SIP doesn't differentiate the
various
		> mechanisms by which a request-URI may be changed,
which is what some
		> of the updated text in 4244bis is trying to do and why
we need precise
	=09
		> terms. I think we all agree that.
		>
		> I honestly don't care what we call the tag, as long as
we're clear
		> about the functionality.
		>
		> Mary.
		>
		> -----Original Message-----
		> From: Hans Erik van Elburg
[mailto:ietf.hanserik@gmail.com]
		> Sent: Wednesday, March 11, 2009 4:40 PM
		> To: Barnes, Mary (RICH2:AR00)
		> Cc: Shida Schubert; sip@ietf.org
		> Subject: Re: Terminology (was RE: [Sip] Fwd: I-D
		> ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
		>
		> The target-uri concept when defined properly is
entirely application
		> agnostic.
		>
		> When one uses History-Info or another header as a
vehicle for this
		> information then it still can be considered
application agnostic as
		> any application can use this information as it likes.
		>
		>  From SIP perspective all the Request-URI rewrites
look the same, that
	=09
		> is what brought us discussing this in the first place.
		>
		> /Hans Erik
		>
		> Mary Barnes wrote:
		>
		>> And, I think the big issue with the terminology is
that 4244bis is
		>> application agnostic, so it is tagging what happens
to the URIs
		>> (why/how the specific URI was overwritten).  Whereas,
specific
		>> applications can derive information, such as what
"is" the real
		>> "target" for the request, from the History-Info
header.   So, IMHO,
		>> it's a matter of the target-uri document specifying
that the last
		>> entry tagged by whatever name we come up with "is"
the real "target"
		>> for the request.  I believe it's up to the
applications to describe
		>> how they make use of the History-Info and not for the
History-Info to
	=09
		>> describe how applications can use it.  History-Info
purely reflects
		>> what happened from a SIP Protocol perspective and
should not imply
		>> any
		>>
		>
		>
		>> application semantics. Indeed the intent of section 5
of 4244 was to
		>> inform the applications how they should decribe their
usage of the
		>> header.
		>>
		>> Mary.
		>> Note: I changed the subject to hopefully make this
topic easier to
		>> keep track of.
		>>
		>>
---------------------------------------------------------------------
		>> -
		>> --
		>> *From:* Hans Erik van Elburg
[mailto:ietf.hanserik@gmail.com]
		>> *Sent:* Wednesday, March 11, 2009 3:59 AM
		>> *To:* Shida Schubert
		>> *Cc:* sip@ietf.org List; Barnes, Mary (RICH2:AR00)
		>> *Subject:* Re: [Sip] Fwd: I-D
		>> ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
		>>
		>> One of the problems that made the discussion quite
tedious is the
		>> complete inversed meaning of the terms "retarget" and
"reroute"
		>> between 4244bis and target-uri factions have.
		>>
		>> Conclusion was that the terminology need to be
properly defined and
		>> some attempt was made to start such activity.
		>>
		>> On 3. I have strong concerns when we start tagging
the URI as to how
		>> they come about "retarget"/"mapped conficuration"
etc. It is better
		>> to
		>>
		>
		>
		>> embellish them with a tag that just represents there
meaning for
		>> example "istarget" or "hop". For the following
reasons:
		>> 1. It is much more intuitive for the user of this
information, its
		>> meaning can basically be guessed.
		>> 2. Such meaning also probably survives application in
other problem
		>> spaces that we had not foreseen when introducing the
concept.
		>>
		>> The solution that is  described now in the target-URI
delivery draft
		>> is not yet complete, it does not solve the freephone
use case for
		>> example and in its current form it will deliver
exactly the same
		>> target-URI as that which would be delivered by the
P-Called-Party-ID
	=09
		>> header. Contrary to what the draft says. So some work
is needed
		still.
		>>
		>> /Hans Erik van Elburg
		>>
		>>
		>> On Wed, Mar 11, 2009 at 7:13 AM, Shida Schubert
<shida@ntt-at.com
		>> <mailto:shida@ntt-at.com>> wrote:
		>>
		>>
		>>     We have submitted the updated target-uri draft
based on the
		>>
		> comments
		>
		>>     submitted to the list and comments received at
IETF73.
		>>
		>>     I have taken over as editor as Jonathan didn't
have the cycles to
	=09
		>>     update the draft, with Francois, Christer and
Hans Erick as
		>>     additional
		>>     co-authors and great deal of help from Mary.
		>>
		>>     The following summarizes the changes made to the
target-uri
		>>
		> document
		>
		>>     1. Added use-case for toll-free number back
		>>     2. Added definition of "retarget" operation.
		>>     3. Removed a reference to URN
		>>     4. Added a text discussing the difference to
P-Called-Party-Id
		>>     5. Changed parameter name from "target" to
"istarget"
		>>
		>>     Note, that the target-uri document still contains
the normative
		>>     text for the
		>>     History-Info header.
		>>
		>>     In addition, Mary (with Francois as co-author)
has submitted a
		>>     rfc4244bis, with the following changes:
		>>     1. Incorporated the normative aspects of the
target-uri document
		>>     into the existing normative text in RFC 4244 -
the functionality
		>>
		> is
		>
		>>     virtually identical (as is some of the text) as
the HI based
		>>
		> solution
		>
		>>     described in the target-uri document.  It's
important that the
		>>     solution
		>>     be integrated into RFC 4244 as it MUST work and
be based on the
		>>     normative
		>>     aspects of RFC 4244.
		>>     2.  Added the use cases from target-uri the the
summary
		>>     in the overview of rfc4244bis.
		>>     3. Added an additional requirement to capture the
"target-uri"
		>>     information.
		>>     4. Fixed an error in the RFC 4244 ABNF and added
"retarget"
		>>
		> parameter.
		>
		>>     5. Added a more simplified example.
		>>
		>>
		>>     We had some very long offline exchanges as to the
best way
		>> forward
		>>
		> and
		>
		>>     remaining work for both documents.
		>>
		>>     Some of the issues identified are:
		>>
		>>     ::Issues::
		>>        1. Should we remove the normative text from
target-uri and
		>>
		> progress
		>
		>>            4244bis along with the target-uri document
to meet the
		>>     chartered
		>>           SIP WG milestone?
		>>
		>>
		>>        2. Name of the parameter.
		>>           At the last meeting, parameter "target" was
said
		>>
		> inappropriate
		>
		>>           because voicemail-uri spec already defines
a parameter
		>>
		> called
		>
		>>           "target" which also can be found in
hi-entry, thus
		>>
		> potentially
		>
		>>           causing confusion.
		>>
		>>           Currently the target-uri draft uses
"istarget" and 4244bis
		>>
		> uses
		>
		>>           "retarget"  but we could never come to
		>>           a consensus on what name is appropriate.
Other suggestions
		>>
		> have
		>
		>>           included the following:
		>>           "target-identity" (someone didn't like that
"identity" is
		>>     also a SIP header)
		>>           "reg-uri"  (can be paired with "mapped-uri"
for item 3
		>>
		> below)
		>
		>>           "aor"
		>>           "jibberish"
		>>           etc.
		>>
		>>           One reason this is so difficult relates to
the problem
		>>     statement in target-uri in that
		>>           RFC 3261 doesn't differentiate the
mechanism by which the
		>>
		> new
		>
		>>           (target) Request-URI is selected.  Another
issue is that
		>>     some of the terminology in
		>>           RFC 3261 is overloaded - e.g., "forwarding"
refers both to
		a
		>>     Proxy
		>>           which does not have responsibility for the
domain of the
		>>     request-URI
		>>           in the incoming request, thus the proxy
just "forwards" the
		>>     request to
		>>           the next hop AND "forwarding" is used to
describe the
		>>     process whereby
		>>           the outgoing request is built and
"forwarded" to the next
		>>     hop at which
		>>           point the proxy does not know how the new
request-uri was
		>>     selected.
		>>           RFC 4244 has attempted to clarify the terms
and attempts to
		>>     use "forward"
		>>           in the context of the former situation and
"retarget" for
		>>     the case whereby
		>>           a proxy is responsible for the domain and
thus can use a
		>>     number of
		>>           mechanism to select the new target for the
request - e.g.,
		a
		>>     REGISTRAR,
		>>           configured data, etc.
		>>
		>>      3.  Related to the last point in item 2 above,
it has been
		>>     proposed that
		>>           we differentiate the hi-entries even more
by defining
		>>     separate parameters
		>>          for registered and configured/mapped
contacts.
		>>          Currently when the R-URI is translated to a
URI which is
		>>     either derived
		>>          from location service lookup(registered by
UA) or from
		>>
		> mapping
		>
		>>          table, there is no differentiation as to how
the URI was
		>>     derived once it is
		>>          added to the list of potential targets.
		>>
		>>          The general consensus of the authors of the
two documents
		was
		>>     that it may
		>>          be useful for some services to have the
hi-entries tagged
		>>     with the
		>>          more specific information.
		>>
		>>          And, of course, this gets us into another
naming contest. In
		>>     the end, the naming
		>>          is not so important as long as the term
isn't too overloaded
		>>     and it is defined
		>>          precisely in the document(s).
		>>
		>>     We would appreciate WG feedback on these issues
and any other
		>>     comments on
		>>     the two documents prior to IETF-74.
		>>
		>>     Regards,
		>>     Shida and Mary.
		>>
		>>
		>>
		>>     Begin forwarded message:
		>>
		>>>     *From: *Internet-Drafts@ietf.org
		>>>
		> <mailto:Internet-Drafts@ietf.org>
		>
		>>>     *Date: *March 10, 2009 2:30:01 AM JST
		>>>     *To: *i-d-announce@ietf.org
<mailto:i-d-announce@ietf.org>
		>>>     *Subject: **I-D
		>>>
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt *
		>>>     *Reply-To: *internet-drafts@ietf.org
		>>>     <mailto:internet-drafts@ietf.org>
		>>>
		>>>     A New Internet-Draft is available from the
on-line
		>>>
		> Internet-Drafts
		>
		>>>     directories.
		>>>
		>>>
		>>>     Title
		>>>     : Delivery of Request-URI Targets to User Agents
		>>>     Author(s)
		>>>     : J. Rosenberg, H. van Elburg, C. Holmberg, F.
Audet, S.
		Schubert
		>>>     Filename :
draft-rosenberg-sip-target-uri-delivery-01.txt
		>>>     Pages
		>>>     : 16
		>>>     Date
		>>>     : 2009-3-9
		>>>
		>>>     When a Session Initiation Protocol (SIP) proxy
receives a
		request
		>>>       targeted at a URI identifying a user or
resource it is
		>>>
		> responsible
		>
		>>>       for, the proxy translates the URI to a
registered or
		configured
		>>>       contact URI of an agent representing that user
or resource.
		>>> In
		>>>
		> the
		>
		>>>       process, the original URI is removed from the
request.
		>>>      Numerous use
		>>>       cases have arisen which require this
information to be
		>>>
		> delivered to
		>
		>>>       the user agent.  This document describes these
use cases and
		>>>     defines
		>>>       an extension to the History-Info header field
which allows it
		>>>
		> to be
		>
		>>>       used to support those cases.
		>>>
		>>>     A URL for this Internet-Draft is:
		>>>
		>>>
http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri-d
		>>> e
		>>> livery-01.txt
		>>>
		>>>     Internet-Drafts are also available by anonymous
FTP at:
		>>>     ftp://ftp.ietf.org/internet-drafts/
		>>>
		>>>     Below is the data which will enable a MIME
compliant mail reader
		>>>     implementation to automatically retrieve the
ASCII version of
		the
		>>>     Internet-Draft.
		>>>
		>>     Content-Type: text/plain<BR>Content-ID:
		>>     &lt;2009-3-9102756.I-D@ietf.org
<mailto:lt%3B2009-3-9102756.I-D@ietf.org>=20
		>>     <mailto:2009-3-9102756.I-D@ietf.org>&gt;<BR><BR>
		>>
		>>>     _______________________________________________
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		>>>
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		>>>
		>>     _______________________________________________
		>>     Sip mailing list
https://www.ietf.org/mailman/listinfo/sip
		>>     This list is for NEW development of the core SIP
Protocol
		>>     Use sip-implementors@cs.columbia.edu
		>>     <mailto:sip-implementors@cs.columbia.edu> for
questions on
		>> current
		>>
		> sip
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		>>     Use sipping@ietf.org <mailto:sipping@ietf.org>
for new
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		>>
		>>
		>>
	=09



From john.elwell@siemens.com  Thu Mar 12 06:43:49 2009
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Date: Thu, 12 Mar 2009 13:44:23 +0000
From: "Elwell, John" <john.elwell@siemens.com>
In-reply-to: <9ae56b1e0903120556t73208627v39b3c6221098cc44@mail.gmail.com>
To: Hans Erik van Elburg <ietf.hanserik@gmail.com>
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Cc: sip@ietf.org
Subject: Re: [Sip] Terminology (was RE: Fwd: I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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(Old messages stripped to prevent blocking)

Hans Erik,
=20
The way I interpreted the 01 draft, you would not get the freephone
number anyway. If that is a requirement, then I accept my suggestion
won't suffice.
=20
John


________________________________

	From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
	Sent: 12 March 2009 12:57
	To: Elwell, John
	Cc: sip@ietf.org
	Subject: Re: [Sip] Terminology (was RE: Fwd:
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
=09
=09
	But that does not solve the freephone service use case.
Freephone represents a use case where the R-URI is replaced with an AoR
of user B - not with the contact of user B.=20
=09
	A calls Bfreephonenumber routed the B (AOR) routed to B's
contact.
=09
	The target-uri draft is interested in obtaining:
Bfreephonenumber as that is what A used to reach B.
=09
	So what we like to see that B UA receives is something like
	    H-I:  Bfreephone; "istarget",\
	        B,\
	        contact-B
=09
	So you want to get Bfreephonenumber and not B.
=09
	In certain situations you are also interested in B, I guess that
PBX's would interested in that to deliver to the correct user. We could
distinguish them like:
	    H-I:  Bfreephone; "istarget",\
	        B,"aor"\
	        contact-B
=09
	BR,
	/Hans Erik van Elburg
=09

From mary.barnes@nortel.com  Thu Mar 12 06:51:35 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Ian Elz" <ian.elz@ericsson.com>
Cc: sip@ietf.org, Francois Audet <audet@nortel.com>
Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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Ian,

Okay, so I think I might understand now - you want to use the "none" to
override the Privacy header that would be (in current 4244/4244bis)
applied to the whole message as it traverses across the n/w from the UAC
to the UAS. Is that correct? That then though changes the meaning of the
Privacy header, as I understand it - i.e., the "none" I thought had to
do with specifying that an intermediary MUST NOT change the value of the
Privacy header. But, I guess you are saying we specify a slightly
different usage in 4244bis such that when we add it to an hi-entry, it
specifies that that hi-entry MUST NOT be dropped by an intermediary.
That might work, although, we would need verbage about the "risks" of
doing so.

Thanks,
Mary.=20

-----Original Message-----
From: Ian Elz [mailto:ian.elz@ericsson.com]=20
Sent: Thursday, March 12, 2009 8:43 AM
To: Barnes, Mary (RICH2:AR00)
Cc: sip@ietf.org; Audet, Francois (SC100:3055)
Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

Mary,

(Written after reading your discussion with Hans Erik.)

First an apology for not making it clear in my original email that a
value of "none" in a Privacy header parameter in a History-Info entry
was only applicable to the uri contained in the hi-targeted-to-uri.

I only suggested using the value "none" as this is a value defined for
the Privacy header which explicitly indicates that privacy is not
required. Any other value with the meaning of explicitly indicating that
there is no privacy required for the specific uri could be used.

With regard to your second paragraph below I understand that the SIP and
PSTN models do not align. You are correct in your assumption that this
problem is not unique for History-Info. The problem also arises with the
Referred-By header.=20

I have raised this for History-Info at this time as your new draft
presents an opportunity to correct this problem for the H-I case.

The general SIP privacy issue arises as when the Privacy header was
defined the concept of carrying third party identities in the SIP
messages was not defined. This came later with the REFER Request and the
associated Referred-By header and the History-Info header.=20

We should take what opportunities arise to discuss and correct this if
there is a will to make these changes.

Ian Elz

Office: + 44 24 764 35256
ian.elz@ericsson.com


-----Original Message-----
From: Mary Barnes [mailto:mary.barnes@nortel.com]
Sent: 11 March 2009 16:53
To: Ian Elz; Francois Audet
Cc: sip@ietf.org
Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

Ian,

This is an interesting question. I need to think about it some more, but
I don't like the "none" idea as it really must be up to the entity that
added the Privacy header to the request, as to whether it wants the HI
entries that it adds to go outside a domain. My initial thought is that
we can't overrule the overall Privacy header. The thing is that the
Privacy header doesn't preclude gathering the information and using
within a domain AND if it were to not include an hi-entry when the
request leaves the domain for which the proxy is responsible, the
recipient can add the hi-entry for the current request-uri before it
adds the new hi-entry. =20

Your PSTN example doesn't strictly map obviously to a SIP model and I
would think you might consider the PSTN hop to be within the same domain
for which the proxy (or gw, I guess) is responsible OR consider this a
walled garden whereby the Privacy header doesn't apply. Which brings me
to a more general question as to how you all deal with other headers
when you do your mapping to the PSTN when there is a Privacy header?  It
would seem this problem isn't unique to History-Info, although I know
little about the details of PSTN I/W.

Regards,
Mary.=20

-----Original Message-----
From: Ian Elz [mailto:ian.elz@ericsson.com]
Sent: Wednesday, March 11, 2009 4:54 AM
To: Barnes, Mary (RICH2:AR00); Audet, Francois (SC100:3055)
Cc: sip@ietf.org
Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

Mary, Francois,

Section 4.1 of the draft allows for the addition of the Privacy header
parameter with a value of "history" to be included with the
hi-targeted-to-uri.

Can this be extended to also allow the value "none".

A node adding a H-I entry is allowed to specify the privacy value
"history" either in the Privacy header or as a header parameter
associated with the hi-targeted-to-uri. If the former action is taken by
one node this results in privacy for all history info header entries
even if this is not required. There is no way for privacy of an
individual H-I entry to be set to none if no Privacy is required for the
uri.

This creates a specific problem when interworking with the PSTN where
the privacy is associated with each E.164 number included in the
protocol. If an INVITE is being mapped to ISUP and the H-I entries are
being used to map to redirection numbers then a single Privacy header
with the value "history" will result in all numbers being restricted.

If the Privacy header parameter could include the value "none" then this
would explicitly indicate that the associated uri was allowed to be
presented.

The ability to explicitly allow presentation of specific H-I entries may
also be useful in pure sip implementations.

I don't believe that this change would create any backward compatibility
issues. The existing implementations will continue as deployed but new
implementations could explicitly set no privacy for a specific uri.

There are no security issues other than those already defined in the
draft that an intervening node could modify an existing H-I entry. The
Privacy header value of "none" should only be included by the node in
the same manner as the existing inclusion of the "history" value.

Ian Elz


-----Original Message-----
From: i-d-announce-bounces@ietf.org
[mailto:i-d-announce-bounces@ietf.org] On Behalf Of
Internet-Drafts@ietf.org
Sent: 04 March 2009 15:15
To: i-d-announce@ietf.org
Subject: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

A New Internet-Draft is available from the on-line Internet-Drafts
directories.

	Title           : An Extension to the Session Initiation
Protocol (SIP) for Request History Information
	Author(s)       : M. Barnes, F. Audet
	Filename        : draft-barnes-sip-rfc4244bis-00.txt
	Pages           : 49
	Date            : 2009-03-04

This document defines a standard mechanism for capturing the history
information associated with a Session Initiation Protocol (SIP) request.
This capability enables many enhanced services by providing the
information as to how and why a call arrives at a specific application
or user.  This document defines a new optional SIP header, History-Info,
for capturing the history information in requests.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-barnes-sip-rfc4244bis-00.txt

Internet-Drafts are also available by anonymous FTP at:
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From christer.holmberg@ericsson.com  Thu Mar 12 06:49:48 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens.com>, "Hans Erik van Elburg" <ietf.hanserik@gmail.com>
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=20
Hi,
=20
IF we would remove that requierment, and ONLY care about the case when
the R-URI is replaced with the CONTACT, I don't think we need to do
anything. I believe P-Called-Party-ID already solves that.
=20
Regards,
=20
Christer


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Elwell, John
	Sent: 12. maaliskuuta 2009 15:31
	To: Hans Erik van Elburg
	Cc: sip@ietf.org
	Subject: Re: [Sip] Terminology (was RE: Fwd:
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
=09
=09
	Hans Erik,
	=20
	The way I interpreted the 01 draft, you would not get the
freephone number anyway. If that is a requirement, then I accept my
suggestion won't suffice.
	=20
	John


________________________________

		From: Hans Erik van Elburg
[mailto:ietf.hanserik@gmail.com]=20
		Sent: 12 March 2009 12:57
		To: Elwell, John
		Cc: sip@ietf.org
		Subject: Re: [Sip] Terminology (was RE: Fwd:
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
	=09
	=09
		But that does not solve the freephone service use case.
Freephone represents a use case where the R-URI is replaced with an AoR
of user B - not with the contact of user B.=20
	=09
		A calls Bfreephonenumber routed the B (AOR) routed to
B's contact.
	=09
		The target-uri draft is interested in obtaining:
Bfreephonenumber as that is what A used to reach B.
	=09
		So what we like to see that B UA receives is something
like
		    H-I:  Bfreephone; "istarget",\
		        B,\
		        contact-B
	=09
		So you want to get Bfreephonenumber and not B.
	=09
		In certain situations you are also interested in B, I
guess that  PBX's would interested in that to deliver to the correct
user. We could distinguish them like:
		    H-I:  Bfreephone; "istarget",\
		        B,"aor"\
		        contact-B
	=09
		BR,
		/Hans Erik van Elburg
	=09
	=09
	=09
		On Thu, Mar 12, 2009 at 12:08 PM, Elwell, John
<john.elwell@siemens.com> wrote:
	=09

			If 'istarget' is used only to identify the value
that was overwritten in
			the Request-URI by a contact URI, an alternative
approach would be to
			flag the contact URI H-I entry as 'iscontact'.
Then the UAS would just
			need to look for the most recent H-I entry that
is not marked
			'iscontact'.
		=09
			John
		=09
		=09
			________________________________
		=09
			       From: sip-bounces@ietf.org
[mailto:sip-bounces@ietf.org] On
			Behalf Of Hans Erik van Elburg
			       Sent: 12 March 2009 08:35
			       To: Mary Barnes
			       Cc: sip@ietf.org
			       Subject: Re: [Sip] Terminology (was RE:
Fwd:
=09
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
		=09


			       Yes, because you are using the 3261 use
of target and the
			4244bis introduced definition of retarget. I
thought it was clear that
			we need other words as those definitions don't
match the target-uri
			drafts use of the terms. Also they do not
suffice to provide a solution
			for the use cases in the target-uri draft.
		=09
			       The 3261 text you refer to is exactly
about the case where the
			home proxy overwrites the Request-URI with a new
target. This target is
			teh registered contact address. And hence this
would be what target-uri
			calls a hop or a route. This case and this is
where it gets confusing is
			not a "retarget" in the target-uri draft use of
the term.
		=09
			       The target-uri draft states:
		=09
			          "To avoid confusion, we
			          refer to a SIP URI that is an address
for a user or resource
			as a
			          "target" and a SIP URI that is a hop
for reaching that user
			as a
		=09
			          "hop".
		=09
		=09
			       Apparently that does not suffice to avoid
confusion.
		=09
			       As for the tagging, speaking about the
solution before agreeing
			on the terminology and the problem it should
solve is meaningless.
		=09
			       /Hans Erik van Elburg
		=09
		=09
		=09
			       On Thu, Mar 12, 2009 at 2:33 AM, Mary
Barnes
			<mary.barnes@nortel.com> wrote:
		=09
		=09
			               Responses below [MB].
		=09
		=09
			               -----Original Message-----
			               From: Hans Erik van Elburg
			[mailto:ietf.hanserik@gmail.com]
		=09
			               Sent: Wednesday, March 11, 2009
6:42 PM
			               To: Barnes, Mary (RICH2:AR00)
			               Cc: Shida Schubert; sip@ietf.org
			               Subject: Re: Terminology (was RE:
[Sip] Fwd: I-D
=09
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
		=09
		=09
			               I was talking about the concept.
		=09
			               The use cases only describe
cases where the target-uri
			(lets call that
			               current target from now) has been
lost when an initial
			request for a
			               dialog or standalone request
arrives at the UAS.
		=09
			               [MB] And I think this is where
the terminology confusion
			starts - I
			               don't think of the "lost"
target-uri as being the
			"current" target. In
			               my mind, the "current" target is
reflected by the last
			hi-entry and the
			               request-uri in the incoming
request at the UAS. If the
			entity that sent
			               the request was the entity that
added the last hi-entry,
			then the uri in
			               that hi-entry is the same as the
request-URI in the SIP
			request that
			               arrives at the UAS. I refer to
the "lost" uri as the one
			that was
			               "retargeted" - that's the one the
UAS wants to pull from
			the hi-entries
			               in the incoming request. That
hi-entry was not the one
			that was just
			               added by the entity that built
the request just received
			by the UAS.
			               That hi-entry is tagged with
whatever name we are going
			to tag it with
			               BEFORE the request is forwarded
(using the term forward
			per section 16.6
			               of RFC 3261).  That tag is added
once the target set of
			potential
			               candidates for the new request
uri are determined in
			section 16.5 of
			               3261 (with "target set" being a
3261 term), just before
			the request is
			               forwarded in section 16.6 to one
of those targets.  A
			new hi-entry
			               (which will be the last hi-entry
in the request received
			by the UAS) is
			               added in section 16.6 of 3261 as
the request is
			forwarded.  At this
			               point in time, the lost
information is in the previous
			hi-entry when the
			               outgoing request is sent. [/MB]
		=09
		=09
			               What has been lost is the current
target of the request:
		=09
			               [MB] Right - at which point in my
mind, it's no longer
			the current
			               target ;) Maybe we call it
"lost". [/MB]
		=09
		=09
		=09
			                    Current target
		=09
			                 The current target of an
initial request for a dialog
			or standalone
			                 request is the name or address
to which the request is
			targeted, i.e.
			                 either the initial target
inserted in the Request-URI
			by the UAC that
			                 originates the request, or when
a retarget occurred,
			the target
			                 provided in that retarget
operation.  Reroute and
			translation
			                 operations never change the
current target.
		=09
		=09
			               [MB] I don't think this
definition fits what you want -
			i.e., if there
			               is no retargeting, then none of
the hi-entries are
			tagged - i.e., you
			               won't have your concept of
current. The way it works is
			that if no
			               hi-entries are tagged, then you
know that there was no
			retargeting, thus
			               you know the request-uri has not
been lost. [/MB]
		=09
		=09
			               This defintion only makes sense
when the following
			definitions are used:
		=09
			               Name:
		=09
			                 A name is a moniker for an
entity which refers to it
			in a way which
		=09
			                 reveals nothing about where it
is in a network.  In
			SIP, tel URI
		=09
			                 which doesn't represent the
location of the entity is
			a name.
		=09
		=09
			               Address:
		=09
			                 An address is an identifier for
an entity which
			describes it by its
		=09
			                 location on the network.  In
SIP, the SIP URI itself
			is a form of
		=09
			                 address because the host part
of the URI, the only
			mandatory part of
		=09
			                 the URI besides the scheme
itself, indicates the
			location of a SIP
		=09
			                 server that can be used to
handle the request.
		=09
			               Route:
		=09
			                 Finally, a route is a
sequence of SIP entities
			(including the UA
			               itself!) which are
		=09
			                traversed in order to forward a
request to an address
			or name.
		=09
			               Retarget (other term might be
needed, as this is highly
			confusing):
		=09
			                  A Request-URI rewrite
operation that changes the
			target identity of
			               the request.
		=09
			               Reroute (other term might be
needed):
		=09
			                  A Request-URI rewrite
operation that does not change
			the current
			               target of the request, but
determines the route/next
			hop taken to
			               reach the target-identity.
		=09
			               /Hans Erik
		=09
			               Mary Barnes wrote:
			               > So, by "target-uri concept" are
you referring to the
			solution in the
			               > current target-uri draft or the
application usage as
			such in the
			               > target-uri document (which is
what I was referring
			to)?
			               >
			               > Yes, the problem is that SIP
doesn't differentiate the
			various
			               > mechanisms by which a
request-URI may be changed,
			which is what some
			               > of the updated text in 4244bis
is trying to do and why
			we need precise
		=09
			               > terms. I think we all agree
that.
			               >
			               > I honestly don't care what we
call the tag, as long as
			we're clear
			               > about the functionality.
			               >
			               > Mary.
			               >
			               > -----Original Message-----
			               > From: Hans Erik van Elburg
			[mailto:ietf.hanserik@gmail.com]
			               > Sent: Wednesday, March 11, 2009
4:40 PM
			               > To: Barnes, Mary (RICH2:AR00)
			               > Cc: Shida Schubert;
sip@ietf.org
			               > Subject: Re: Terminology (was
RE: [Sip] Fwd: I-D
			               >
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
			               >
			               > The target-uri concept when
defined properly is
			entirely application
			               > agnostic.
			               >
			               > When one uses History-Info or
another header as a
			vehicle for this
			               > information then it still can
be considered
			application agnostic as
			               > any application can use this
information as it likes.
			               >
			               >  From SIP perspective all the
Request-URI rewrites
			look the same, that
		=09
			               > is what brought us discussing
this in the first place.
			               >
			               > /Hans Erik
			               >
			               > Mary Barnes wrote:
			               >
			               >> And, I think the big issue
with the terminology is
			that 4244bis is
			               >> application agnostic, so it is
tagging what happens
			to the URIs
			               >> (why/how the specific URI was
overwritten).  Whereas,
			specific
			               >> applications can derive
information, such as what
			"is" the real
			               >> "target" for the request, from
the History-Info
			header.   So, IMHO,
			               >> it's a matter of the
target-uri document specifying
			that the last
			               >> entry tagged by whatever name
we come up with "is"
			the real "target"
			               >> for the request.  I believe
it's up to the
			applications to describe
			               >> how they make use of the
History-Info and not for the
			History-Info to
		=09
			               >> describe how applications can
use it.  History-Info
			purely reflects
			               >> what happened from a SIP
Protocol perspective and
			should not imply
			               >> any
			               >>
			               >
			               >
			               >> application semantics. Indeed
the intent of section 5
			of 4244 was to
			               >> inform the applications how
they should decribe their
			usage of the
			               >> header.
			               >>
			               >> Mary.
			               >> Note: I changed the subject to
hopefully make this
			topic easier to
			               >> keep track of.
			               >>
			               >>
=09
---------------------------------------------------------------------
			               >> -
			               >> --
			               >> *From:* Hans Erik van Elburg
			[mailto:ietf.hanserik@gmail.com]
			               >> *Sent:* Wednesday, March 11,
2009 3:59 AM
			               >> *To:* Shida Schubert
			               >> *Cc:* sip@ietf.org List;
Barnes, Mary (RICH2:AR00)
			               >> *Subject:* Re: [Sip] Fwd: I-D
			               >>
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
			               >>
			               >> One of the problems that made
the discussion quite
			tedious is the
			               >> complete inversed meaning of
the terms "retarget" and
			"reroute"
			               >> between 4244bis and target-uri
factions have.
			               >>
			               >> Conclusion was that the
terminology need to be
			properly defined and
			               >> some attempt was made to start
such activity.
			               >>
			               >> On 3. I have strong concerns
when we start tagging
			the URI as to how
			               >> they come about
"retarget"/"mapped conficuration"
			etc. It is better
			               >> to
			               >>
			               >
			               >
			               >> embellish them with a tag that
just represents there
			meaning for
			               >> example "istarget" or "hop".
For the following
			reasons:
			               >> 1. It is much more intuitive
for the user of this
			information, its
			               >> meaning can basically be
guessed.
			               >> 2. Such meaning also probably
survives application in
			other problem
			               >> spaces that we had not
foreseen when introducing the
			concept.
			               >>
			               >> The solution that is
described now in the target-URI
			delivery draft
			               >> is not yet complete, it does
not solve the freephone
			use case for
			               >> example and in its current
form it will deliver
			exactly the same
			               >> target-URI as that which would
be delivered by the
			P-Called-Party-ID
		=09
			               >> header. Contrary to what the
draft says. So some work
			is needed
			               still.
			               >>
			               >> /Hans Erik van Elburg
			               >>
			               >>
			               >> On Wed, Mar 11, 2009 at 7:13
AM, Shida Schubert
			<shida@ntt-at.com
			               >> <mailto:shida@ntt-at.com>>
wrote:
			               >>
			               >>
			               >>     We have submitted the
updated target-uri draft
			based on the
			               >>
			               > comments
			               >
			               >>     submitted to the list and
comments received at
			IETF73.
			               >>
			               >>     I have taken over as
editor as Jonathan didn't
			have the cycles to
		=09
			               >>     update the draft, with
Francois, Christer and
			Hans Erick as
			               >>     additional
			               >>     co-authors and great deal
of help from Mary.
			               >>
			               >>     The following summarizes
the changes made to the
			target-uri
			               >>
			               > document
			               >
			               >>     1. Added use-case for
toll-free number back
			               >>     2. Added definition of
"retarget" operation.
			               >>     3. Removed a reference to
URN
			               >>     4. Added a text discussing
the difference to
			P-Called-Party-Id
			               >>     5. Changed parameter name
from "target" to
			"istarget"
			               >>
			               >>     Note, that the target-uri
document still contains
			the normative
			               >>     text for the
			               >>     History-Info header.
			               >>
			               >>     In addition, Mary (with
Francois as co-author)
			has submitted a
			               >>     rfc4244bis, with the
following changes:
			               >>     1. Incorporated the
normative aspects of the
			target-uri document
			               >>     into the existing
normative text in RFC 4244 -
			the functionality
			               >>
			               > is
			               >
			               >>     virtually identical (as is
some of the text) as
			the HI based
			               >>
			               > solution
			               >
			               >>     described in the
target-uri document.  It's
			important that the
			               >>     solution
			               >>     be integrated into RFC
4244 as it MUST work and
			be based on the
			               >>     normative
			               >>     aspects of RFC 4244.
			               >>     2.  Added the use cases
from target-uri the the
			summary
			               >>     in the overview of
rfc4244bis.
			               >>     3. Added an additional
requirement to capture the
			"target-uri"
			               >>     information.
			               >>     4. Fixed an error in the
RFC 4244 ABNF and added
			"retarget"
			               >>
			               > parameter.
			               >
			               >>     5. Added a more simplified
example.
			               >>
			               >>
			               >>     We had some very long
offline exchanges as to the
			best way
			               >> forward
			               >>
			               > and
			               >
			               >>     remaining work for both
documents.
			               >>
			               >>     Some of the issues
identified are:
			               >>
			               >>     ::Issues::
			               >>        1. Should we remove the
normative text from
			target-uri and
			               >>
			               > progress
			               >
			               >>            4244bis along with
the target-uri document
			to meet the
			               >>     chartered
			               >>           SIP WG milestone?
			               >>
			               >>
			               >>        2. Name of the
parameter.
			               >>           At the last meeting,
parameter "target" was
			said
			               >>
			               > inappropriate
			               >
			               >>           because
voicemail-uri spec already defines
			a parameter
			               >>
			               > called
			               >
			               >>           "target" which also
can be found in
			hi-entry, thus
			               >>
			               > potentially
			               >
			               >>           causing confusion.
			               >>
			               >>           Currently the
target-uri draft uses
			"istarget" and 4244bis
			               >>
			               > uses
			               >
			               >>           "retarget"  but we
could never come to
			               >>           a consensus on what
name is appropriate.
			Other suggestions
			               >>
			               > have
			               >
			               >>           included the
following:
			               >>           "target-identity"
(someone didn't like that
			"identity" is
			               >>     also a SIP header)
			               >>           "reg-uri"  (can be
paired with "mapped-uri"
			for item 3
			               >>
			               > below)
			               >
			               >>           "aor"
			               >>           "jibberish"
			               >>           etc.
			               >>
			               >>           One reason this is
so difficult relates to
			the problem
			               >>     statement in target-uri in
that
			               >>           RFC 3261 doesn't
differentiate the
			mechanism by which the
			               >>
			               > new
			               >
			               >>           (target) Request-URI
is selected.  Another
			issue is that
			               >>     some of the terminology in
			               >>           RFC 3261 is
overloaded - e.g., "forwarding"
			refers both to
			               a
			               >>     Proxy
			               >>           which does not have
responsibility for the
			domain of the
			               >>     request-URI
			               >>           in the incoming
request, thus the proxy
			just "forwards" the
			               >>     request to
			               >>           the next hop AND
"forwarding" is used to
			describe the
			               >>     process whereby
			               >>           the outgoing request
is built and
			"forwarded" to the next
			               >>     hop at which
			               >>           point the proxy does
not know how the new
			request-uri was
			               >>     selected.
			               >>           RFC 4244 has
attempted to clarify the terms
			and attempts to
			               >>     use "forward"
			               >>           in the context of
the former situation and
			"retarget" for
			               >>     the case whereby
			               >>           a proxy is
responsible for the domain and
			thus can use a
			               >>     number of
			               >>           mechanism to select
the new target for the
			request - e.g.,
			               a
			               >>     REGISTRAR,
			               >>           configured data,
etc.
			               >>
			               >>      3.  Related to the last
point in item 2 above,
			it has been
			               >>     proposed that
			               >>           we differentiate the
hi-entries even more
			by defining
			               >>     separate parameters
			               >>          for registered and
configured/mapped
			contacts.
			               >>          Currently when the
R-URI is translated to a
			URI which is
			               >>     either derived
			               >>          from location service
lookup(registered by
			UA) or from
			               >>
			               > mapping
			               >
			               >>          table, there is no
differentiation as to how
			the URI was
			               >>     derived once it is
			               >>          added to the list of
potential targets.
			               >>
			               >>          The general consensus
of the authors of the
			two documents
			               was
			               >>     that it may
			               >>          be useful for some
services to have the
			hi-entries tagged
			               >>     with the
			               >>          more specific
information.
			               >>
			               >>          And, of course, this
gets us into another
			naming contest. In
			               >>     the end, the naming
			               >>          is not so important
as long as the term
			isn't too overloaded
			               >>     and it is defined
			               >>          precisely in the
document(s).
			               >>
			               >>     We would appreciate WG
feedback on these issues
			and any other
			               >>     comments on
			               >>     the two documents prior to
IETF-74.
			               >>
			               >>     Regards,
			               >>     Shida and Mary.
			               >>
			               >>
			               >>
			               >>     Begin forwarded message:
			               >>
			               >>>     *From:
*Internet-Drafts@ietf.org
			               >>>
			               >
<mailto:Internet-Drafts@ietf.org>
			               >
			               >>>     *Date: *March 10, 2009
2:30:01 AM JST
			               >>>     *To:
*i-d-announce@ietf.org
			<mailto:i-d-announce@ietf.org>
			               >>>     *Subject: **I-D
			               >>>
=09
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt *
			               >>>     *Reply-To:
*internet-drafts@ietf.org
			               >>>
<mailto:internet-drafts@ietf.org>
			               >>>
			               >>>     A New Internet-Draft is
available from the
			on-line
			               >>>
			               > Internet-Drafts
			               >
			               >>>     directories.
			               >>>
			               >>>
			               >>>     Title
			               >>>     : Delivery of Request-URI
Targets to User Agents
			               >>>     Author(s)
			               >>>     : J. Rosenberg, H. van
Elburg, C. Holmberg, F.
			Audet, S.
			               Schubert
			               >>>     Filename :
			draft-rosenberg-sip-target-uri-delivery-01.txt
			               >>>     Pages
			               >>>     : 16
			               >>>     Date
			               >>>     : 2009-3-9
			               >>>
			               >>>     When a Session Initiation
Protocol (SIP) proxy
			receives a
			               request
			               >>>       targeted at a URI
identifying a user or
			resource it is
			               >>>
			               > responsible
			               >
			               >>>       for, the proxy
translates the URI to a
			registered or
			               configured
			               >>>       contact URI of an agent
representing that user
			or resource.
			               >>> In
			               >>>
			               > the
			               >
			               >>>       process, the original
URI is removed from the
			request.
			               >>>      Numerous use
			               >>>       cases have arisen which
require this
			information to be
			               >>>
			               > delivered to
			               >
			               >>>       the user agent.  This
document describes these
			use cases and
			               >>>     defines
			               >>>       an extension to the
History-Info header field
			which allows it
			               >>>
			               > to be
			               >
			               >>>       used to support those
cases.
			               >>>
			               >>>     A URL for this
Internet-Draft is:
			               >>>
			               >>>
=09
http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri-d
			               >>> e
			               >>> livery-01.txt
			               >>>
			               >>>     Internet-Drafts are also
available by anonymous
			FTP at:
			               >>>
ftp://ftp.ietf.org/internet-drafts/
			               >>>
			               >>>     Below is the data which
will enable a MIME
			compliant mail reader
			               >>>     implementation to
automatically retrieve the
			ASCII version of
			               the
			               >>>     Internet-Draft.
			               >>>
			               >>     Content-Type:
text/plain<BR>Content-ID:
			               >>
&lt;2009-3-9102756.I-D@ietf.org
<mailto:lt%3B2009-3-9102756.I-D@ietf.org>=20
		=09
			<mailto:lt%3B2009-3-9102756.I-D@ietf.org
<mailto:lt%253B2009-3-9102756.I-D@ietf.org> >
		=09
			               >>
<mailto:2009-3-9102756.I-D@ietf.org>&gt;<BR><BR>
			               >>
			               >>>
_______________________________________________
			               >>>     I-D-Announce mailing list
			               >>>     I-D-Announce@ietf.org
			<mailto:I-D-Announce@ietf.org>
			               >>>
=09
https://www.ietf.org/mailman/listinfo/i-d-announce
			               >>>     Internet-Draft
directories:
			http://www.ietf.org/shadow.html
			               >>>     or
ftp://ftp.ietf.org/ietf/1shadow-sites.txt
			               >>>
			               >>
_______________________________________________
			               >>     Sip mailing list
			https://www.ietf.org/mailman/listinfo/sip
			               >>     This list is for NEW
development of the core SIP
			Protocol
			               >>     Use
sip-implementors@cs.columbia.edu
			               >>
<mailto:sip-implementors@cs.columbia.edu> for
			questions on
			               >> current
			               >>
			               > sip
			               >
			               >>     Use sipping@ietf.org
<mailto:sipping@ietf.org>
			for new
			               >>     developments on the
application of sip
			               >>
			               >>
			               >>
		=09
		=09
		=09
		=09



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<BODY>
<DIV dir=3Dltr align=3Dleft>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN =
class=3D034104813-12032009>Hi,</SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN =
class=3D034104813-12032009></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D034104813-12032009>IF we would =
remove that=20
requierment, and&nbsp;ONLY care about the case when the R-URI is =
replaced with=20
the CONTACT, I don't think we need to do anything. I believe =
P-Called-Party-ID=20
already solves that.</SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN =
class=3D034104813-12032009></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN =
class=3D034104813-12032009>Regards,</SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN =
class=3D034104813-12032009></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN =
class=3D034104813-12032009>Christer</SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Elwell, =
John<BR><B>Sent:</B>=20
  12. maaliskuuta 2009 15:31<BR><B>To:</B> Hans Erik van =
Elburg<BR><B>Cc:</B>=20
  sip@ietf.org<BR><B>Subject:</B> Re: [Sip] Terminology (was RE: Fwd:=20
  =
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR></FONT><BR></=
DIV>
  <DIV></DIV>
  <DIV dir=3Dltr align=3Dleft>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D441300613-12032009><FONT =
face=3DArial=20
  color=3D#000080>Hans Erik,</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D441300613-12032009><FONT =
face=3DArial=20
  color=3D#000080></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D441300613-12032009><FONT =
face=3DArial><FONT=20
  color=3D#000080>The way I interpreted the 01 draft, you would not get =
the=20
  freephone number anyway. If that is a requirement, then I accept my =
suggestion=20
  won't suffice<SPAN=20
class=3D125123013-12032009>.</SPAN></FONT></FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D441300613-12032009><FONT =
face=3DArial=20
  color=3D#000080></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D441300613-12032009><FONT =
face=3DArial=20
  color=3D#000080>John</FONT></SPAN></DIV></DIV><BR>
  <BLOCKQUOTE=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000080 2px =
solid; MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft>
    <HR tabIndex=3D-1>
    <FONT face=3DTahoma size=3D2><B>From:</B> Hans Erik van Elburg=20
    [mailto:ietf.hanserik@gmail.com] <BR><B>Sent:</B> 12 March 2009=20
    12:57<BR><B>To:</B> Elwell, John<BR><B>Cc:</B>=20
    sip@ietf.org<BR><B>Subject:</B> Re: [Sip] Terminology (was RE: Fwd:=20
    =
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR></FONT><BR></=
DIV>
    <DIV></DIV>But that does not solve the freephone service use case.=20
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</STYLE>
    Freephone represents a use case where the R-URI is replaced with an =
AoR of=20
    user B - not with the contact of user B. <BR><BR>A calls =
Bfreephonenumber=20
    routed the B (AOR) routed to B's contact.<BR><BR>The target-uri =
draft is=20
    interested in obtaining: Bfreephonenumber as that is what A used to =
reach=20
    B.<BR><BR>So what we like to see that B UA receives is something=20
    like<BR>&nbsp;&nbsp;&nbsp; H-I:&nbsp; Bfreephone;=20
    "istarget",\<BR>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;=20
    B,\<BR>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; contact-B<BR><BR>So you =
want to=20
    get Bfreephonenumber and not B.<BR><BR>In certain situations you are =
also=20
    interested in B, I guess that&nbsp; PBX's would interested in that =
to=20
    deliver to the correct user. We could distinguish them=20
    like:<BR>&nbsp;&nbsp;&nbsp; H-I:&nbsp; Bfreephone;=20
    "istarget",\<BR>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;=20
    B,"aor"\<BR>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
contact-B<BR><BR>BR,<BR=20
    clear=3Dall>/Hans Erik van Elburg<BR><BR><BR>
    <DIV class=3Dgmail_quote>On Thu, Mar 12, 2009 at 12:08 PM, Elwell, =
John <SPAN=20
    dir=3Dltr>&lt;<A=20
    =
href=3D"mailto:john.elwell@siemens.com">john.elwell@siemens.com</A>&gt;</=
SPAN>=20
    wrote:<BR>
    <BLOCKQUOTE class=3Dgmail_quote=20
    style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">If=20
      'istarget' is used only to identify the value that was overwritten =

      in<BR>the Request-URI by a contact URI, an alternative approach =
would be=20
      to<BR>flag the contact URI H-I entry as 'iscontact'. Then the UAS =
would=20
      just<BR>need to look for the most recent H-I entry that is not=20
      =
marked<BR>'iscontact'.<BR><BR>John<BR><BR><BR>___________________________=
_____<BR><BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp;From: <A=20
      href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</A> =
[mailto:<A=20
      href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</A>] =
On<BR>Behalf=20
      Of Hans Erik van Elburg<BR>&nbsp; &nbsp; &nbsp; &nbsp;Sent: 12 =
March 2009=20
      08:35<BR>&nbsp; &nbsp; &nbsp; &nbsp;To: Mary Barnes<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp;Cc: <A =
href=3D"mailto:sip@ietf.org">sip@ietf.org</A><BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp;Subject: Re: [Sip] Terminology (was RE:=20
      =
Fwd:<BR>I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR>
      <DIV>
      <DIV></DIV>
      <DIV class=3Dh5><BR><BR>&nbsp; &nbsp; &nbsp; &nbsp;Yes, because =
you are=20
      using the 3261 use of target and the<BR>4244bis introduced =
definition of=20
      retarget. I thought it was clear that<BR>we need other words as =
those=20
      definitions don't match the target-uri<BR>drafts use of the terms. =
Also=20
      they do not suffice to provide a solution<BR>for the use cases in =
the=20
      target-uri draft.<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp;The 3261 text =
you=20
      refer to is exactly about the case where the<BR>home proxy =
overwrites the=20
      Request-URI with a new target. This target is<BR>teh registered =
contact=20
      address. And hence this would be what target-uri<BR>calls a hop or =
a=20
      route. This case and this is where it gets confusing is<BR>not a=20
      "retarget" in the target-uri draft use of the term.<BR><BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp;The target-uri draft states:<BR><BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; "To avoid confusion, we<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; refer to a SIP URI that is an address for a user or =
resource<BR>as=20
      a<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; "target" and a SIP URI =
that is a=20
      hop for reaching that user<BR>as a<BR><BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; "hop".<BR><BR><BR>&nbsp; &nbsp; &nbsp; &nbsp;Apparently =
that does=20
      not suffice to avoid confusion.<BR><BR>&nbsp; &nbsp; &nbsp; =
&nbsp;As for=20
      the tagging, speaking about the solution before agreeing<BR>on the =

      terminology and the problem it should solve is =
meaningless.<BR><BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp;/Hans Erik van Elburg<BR><BR><BR><BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp;On Thu, Mar 12, 2009 at 2:33 AM, Mary =
Barnes<BR>&lt;<A=20
      =
href=3D"mailto:mary.barnes@nortel.com">mary.barnes@nortel.com</A>&gt;=20
      wrote:<BR><BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;Responses below [MB].<BR><BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;-----Original Message-----<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;From: Hans Erik van =
Elburg<BR>[mailto:<A=20
      =
href=3D"mailto:ietf.hanserik@gmail.com">ietf.hanserik@gmail.com</A>]<BR><=
BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Sent: Wednesday, =
March 11,=20
      2009 6:42 PM<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;To:=20
      Barnes, Mary (RICH2:AR00)<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;Cc: Shida Schubert; <A=20
      href=3D"mailto:sip@ietf.org">sip@ietf.org</A><BR>&nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;Subject: Re: Terminology (was RE: [Sip] =
Fwd:=20
      I-D<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      =
&nbsp;ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR><BR><BR>&n=
bsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;I was talking =
about the=20
      concept.<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;The=20
      use cases only describe &nbsp;cases where the target-uri<BR>(lets =
call=20
      that<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;current=20
      target from now) has been lost when an initial<BR>request for =
a<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;dialog or =
standalone=20
      request arrives at the UAS.<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;[MB] And I think this is where the terminology =

      confusion<BR>starts - I<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;don't think of the "lost" target-uri as being=20
      the<BR>"current" target. In<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;my mind, the "current" target is reflected by the=20
      last<BR>hi-entry and the<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;request-uri in the incoming request at the UAS. If=20
      the<BR>entity that sent<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;the request was the entity that added the last=20
      hi-entry,<BR>then the uri in<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;that hi-entry is the same as the request-URI in the=20
      SIP<BR>request that<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;arrives at the UAS. I refer to the "lost" uri as the =
one<BR>that=20
      was<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;"retargeted"=20
      - that's the one the UAS wants to pull from<BR>the =
hi-entries<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;in the incoming =
request.=20
      That hi-entry was not the one<BR>that was just<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;added by the entity that built =
the=20
      request just received<BR>by the UAS.<BR>&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp; &nbsp; &nbsp;That hi-entry is tagged with whatever name we =
are=20
      going<BR>to tag it with<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;BEFORE the request is forwarded (using the term=20
      forward<BR>per section 16.6<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;of RFC 3261). &nbsp;That tag is added once the target =
set=20
      of<BR>potential<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;candidates for the new request uri are determined =
in<BR>section 16.5=20
      of<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;3261 =
(with=20
      "target set" being a 3261 term), just before<BR>the request =
is<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;forwarded in =
section 16.6=20
      to one of those targets. &nbsp;A<BR>new hi-entry<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;(which will be the last hi-entry =
in the=20
      request received<BR>by the UAS) is<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;added in section 16.6 of 3261 as the request=20
      is<BR>forwarded. &nbsp;At this<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;point in time, the lost information is in the=20
      previous<BR>hi-entry when the<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp; &nbsp;outgoing request is sent. [/MB]<BR><BR><BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;What has been lost is the =
current=20
      target of the request:<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;[MB] Right - at which point in my mind, it's no =
longer<BR>the=20
      current<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;target=20
      ;) Maybe we call it "lost". [/MB]<BR><BR><BR><BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Current=20
      target<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;The current target of an initial request for a dialog<BR>or=20
      standalone<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;request is the name or address to which the request =
is<BR>targeted,=20
      i.e.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;either the initial target inserted in the Request-URI<BR>by =
the UAC=20
      that<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;originates the request, or when a retarget occurred,<BR>the=20
      target<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;provided in that retarget operation. &nbsp;Reroute=20
      and<BR>translation<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;operations never change the current =
target.<BR><BR><BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;[MB] I don't think =
this=20
      definition fits what you want -<BR>i.e., if there<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;is no retargeting, then none of =
the=20
      hi-entries are<BR>tagged - i.e., you<BR>&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp; &nbsp; &nbsp;won't have your concept of current. The way it =
works=20
      is<BR>that if no<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;hi-entries are tagged, then you know that there was=20
      no<BR>retargeting, thus<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;you know the request-uri has not been lost.=20
      [/MB]<BR><BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;This defintion only makes sense when the =
following<BR>definitions=20
      are used:<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;Name:<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp;A name is a moniker for an entity which refers to it<BR>in a =
way=20
      which<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;reveals nothing about where it is in a network. =
&nbsp;In<BR>SIP, tel=20
      URI<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;which doesn't represent the location of the entity is<BR>a=20
      name.<BR><BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;Address:<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;An address is an identifier for an entity =
which<BR>describes=20
      it by its<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;location on the network. &nbsp;In SIP, the SIP URI =
itself<BR>is a=20
      form of<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;address because the host part of the URI, the =
only<BR>mandatory part=20
      of<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;the URI besides the scheme itself, indicates the<BR>location =
of a=20
      SIP<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;server that can be used to handle the request.<BR><BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Route:<BR><BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Finally, a route =
is a=20
      &nbsp; sequence of SIP entities<BR>(including the UA<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;itself!) which =
are<BR><BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; traversed in =
order to=20
      forward a request to an address<BR>or name.<BR><BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Retarget (other term might be =
needed, as=20
      this is highly<BR>confusing):<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; A Request-URI rewrite operation that =
changes=20
      the<BR>target identity of<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;the request.<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp; &nbsp;Reroute (other term might be needed):<BR><BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; A Request-URI =
rewrite=20
      operation that does not change<BR>the current<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;target of the request, but =
&nbsp; &nbsp;=20
      determines the route/next<BR>hop taken to<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;reach the =
target-identity.<BR><BR>&nbsp; &nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;/Hans Erik<BR><BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Mary Barnes =
wrote:<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt; So, by =
"target-uri=20
      concept" are you referring to the<BR>solution in the<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt; current target-uri =
draft or=20
      the application usage as<BR>such in the<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt; target-uri document (which is what =
I was=20
      referring<BR>to)?<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;=20
      Yes, the problem is that SIP doesn't differentiate=20
      the<BR>various<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt; mechanisms by which a request-URI may be =
changed,<BR>which is=20
      what some<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;=20
      of the updated text in 4244bis is trying to do and why<BR>we need=20
      precise<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;=20
      terms. I think we all agree that.<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt; I honestly don't care what we call the tag, as =
long=20
      as<BR>we're clear<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt; about the functionality.<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt; Mary.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt; -----Original Message-----<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt; From: Hans Erik van =
Elburg<BR>[mailto:<A=20
      =
href=3D"mailto:ietf.hanserik@gmail.com">ietf.hanserik@gmail.com</A>]<BR>&=
nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt; Sent: =
Wednesday,=20
      March 11, 2009 4:40 PM<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp;&gt; To: Barnes, Mary (RICH2:AR00)<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt; Cc: Shida Schubert; <A=20
      href=3D"mailto:sip@ietf.org">sip@ietf.org</A><BR>&nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt; Subject: Re: Terminology (was RE: =
[Sip]=20
      Fwd: I-D<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;=20
      ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt; The target-uri concept when =
defined=20
      properly is<BR>entirely application<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt; agnostic.<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt; When one uses History-Info or another header as=20
      a<BR>vehicle for this<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt; information then it still can be =
considered<BR>application=20
      agnostic as<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;=20
      any application can use this information as it likes.<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt; &nbsp;From SIP perspective =
all the=20
      Request-URI rewrites<BR>look the same, that<BR><BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt; is what brought us =
discussing this=20
      in the first place.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;=20
      /Hans Erik<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;=20
      Mary Barnes wrote:<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; And, I think the big issue with the terminology =
is<BR>that=20
      4244bis is<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; application agnostic, so it is tagging what =
happens<BR>to=20
      the URIs<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; (why/how the specific URI was overwritten).=20
      &nbsp;Whereas,<BR>specific<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt; applications can derive information, such as =

      what<BR>"is" the real<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt; "target" for the request, from the =
History-Info<BR>header.=20
      &nbsp; So, IMHO,<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt; it's a matter of the target-uri document =
specifying<BR>that=20
      the last<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; entry tagged by whatever name we come up with =
"is"<BR>the=20
      real "target"<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; for the request. &nbsp;I believe it's up to=20
      the<BR>applications to describe<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt; how they make use of the History-Info =
and not=20
      for the<BR>History-Info to<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt; describe how applications can use it. =

      &nbsp;History-Info<BR>purely reflects<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; what happened from a SIP =
Protocol=20
      perspective and<BR>should not imply<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt; any<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; application semantics. Indeed the intent of section =
5<BR>of=20
      4244 was to<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; inform the applications how they should decribe=20
      their<BR>usage of the<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt; header.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt;&gt; Mary.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt; Note: I changed the subject to hopefully make =
this<BR>topic=20
      easier to<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; keep track of.<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      =
&nbsp;&gt;&gt;<BR>-------------------------------------------------------=
--------------<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; =
-<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; =
--<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; *From:* =
Hans Erik=20
      van Elburg<BR>[mailto:<A=20
      =
href=3D"mailto:ietf.hanserik@gmail.com">ietf.hanserik@gmail.com</A>]<BR>&=
nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; *Sent:*=20
      Wednesday, March 11, 2009 3:59 AM<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt; *To:* Shida Schubert<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; *Cc:* <A=20
      href=3D"mailto:sip@ietf.org">sip@ietf.org</A> List; Barnes, Mary=20
      (RICH2:AR00)<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; *Subject:* Re: [Sip] Fwd: I-D<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;=20
      ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; One of the problems =
that made=20
      the discussion quite<BR>tedious is the<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; complete inversed meaning of =
the terms=20
      "retarget" and<BR>"reroute"<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt; between 4244bis and target-uri factions=20
      have.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt;&gt; Conclusion was that the terminology need to =
be<BR>properly=20
      defined and<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; some attempt was made to start such =
activity.<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; On 3. I have =
strong=20
      concerns when we start tagging<BR>the URI as to how<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; they come about=20
      "retarget"/"mapped conficuration"<BR>etc. It is better<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; to<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt; embellish them with a tag that just=20
      represents there<BR>meaning for<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt; example "istarget" or "hop". For the=20
      following<BR>reasons:<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt; 1. It is much more intuitive for the user of=20
      this<BR>information, its<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt; meaning can basically be guessed.<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; 2. Such meaning =
also=20
      probably survives application in<BR>other problem<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; spaces that we had not =
foreseen=20
      when introducing the<BR>concept.<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt; The solution that is &nbsp;described now in =
the=20
      target-URI<BR>delivery draft<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt; is not yet complete, it does not solve the=20
      freephone<BR>use case for<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt; example and in its current form it will=20
      deliver<BR>exactly the same<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt; target-URI as that which would be delivered =
by=20
      the<BR>P-Called-Party-ID<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt; header. Contrary to what the draft says. So =
some=20
      work<BR>is needed<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;still.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt;&gt; /Hans Erik van Elburg<BR>&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt; On Wed, Mar 11, 2009 at 7:13 AM, Shida =
Schubert<BR>&lt;<A=20
      href=3D"mailto:shida@ntt-at.com">shida@ntt-at.com</A><BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &lt;mailto:<A=20
      href=3D"mailto:shida@ntt-at.com">shida@ntt-at.com</A>&gt;&gt;=20
      wrote:<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt;&gt; &nbsp; &nbsp; We have submitted the updated =
target-uri=20
      draft<BR>based on the<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt; comments<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; submitted to the list and comments =
received=20
      at<BR>IETF73.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt;&gt; &nbsp; &nbsp; I have taken over as editor as =
Jonathan=20
      didn't<BR>have the cycles to<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; update the draft, with=20
      Francois, Christer and<BR>Hans Erick as<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; =
additional<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; =
&nbsp;=20
      co-authors and great deal of help from Mary.<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; The following =
summarizes=20
      the changes made to the<BR>target-uri<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt; document<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; 1. Added use-case for =
toll-free number=20
      back<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
      &nbsp; &nbsp; 2. Added definition of "retarget" =
operation.<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; =
&nbsp; 3.=20
      Removed a reference to URN<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; 4. Added a text discussing the =

      difference to<BR>P-Called-Party-Id<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; 5. Changed parameter =
name from=20
      "target" to<BR>"istarget"<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; Note, that the target-uri document =
still=20
      contains<BR>the normative<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; text for the<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; =
History-Info=20
      header.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt;&gt; &nbsp; &nbsp; In addition, Mary (with Francois as=20
      co-author)<BR>has submitted a<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; rfc4244bis, with the following =

      changes:<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; 1. Incorporated the normative aspects =
of=20
      the<BR>target-uri document<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; into the existing normative =
text in=20
      RFC 4244 -<BR>the functionality<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt; is<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; virtually identical (as is some of =
the text)=20
      as<BR>the HI based<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt; solution<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; described in the target-uri document. =

      &nbsp;It's<BR>important that the<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; solution<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; be =

      integrated into RFC 4244 as it MUST work and<BR>be based on =
the<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; =
&nbsp;=20
      normative<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; aspects of RFC 4244.<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; 2. =
&nbsp;Added=20
      the use cases from target-uri the the<BR>summary<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; in the =
overview=20
      of rfc4244bis.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt;&gt; &nbsp; &nbsp; 3. Added an additional requirement to =
capture=20
      the<BR>"target-uri"<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; information.<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; 4. Fixed an =
error in the=20
      RFC 4244 ABNF and added<BR>"retarget"<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt; parameter.<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; 5. Added a more simplified=20
      example.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt;&gt; &nbsp; &nbsp; We had some very long offline =
exchanges as to=20
      the<BR>best way<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt; forward<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt; and<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; remaining work for both =
documents.<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; =
Some of=20
      the issues identified are:<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; ::Issues::<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp;1. =
Should=20
      we remove the normative text from<BR>target-uri and<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt; progress<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;4244bis along with the target-uri document<BR>to meet =
the<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; =
&nbsp;=20
      chartered<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; SIP WG=20
      milestone?<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp;2. Name of the=20
      parameter.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; At the last =
meeting,=20
      parameter "target" was<BR>said<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt; inappropriate<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; because=20
      voicemail-uri spec already defines<BR>a parameter<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt; called<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
"target"=20
      which also can be found in<BR>hi-entry, thus<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt; potentially<BR>&nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
causing=20
      confusion.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Currently the =
target-uri=20
      draft uses<BR>"istarget" and 4244bis<BR>&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt; uses<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; "retarget" =
&nbsp;but we=20
      could never come to<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; a consensus on =
what name=20
      is appropriate.<BR>Other suggestions<BR>&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt; have<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; included the=20
      following:<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
"target-identity"=20
      (someone didn't like that<BR>"identity" is<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; also a SIP=20
      header)<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; "reg-uri" &nbsp;(can be paired =
with=20
      "mapped-uri"<BR>for item 3<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt; below)<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; "aor"<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; "jibberish"<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
etc.<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; One reason this is so difficult relates to<BR>the=20
      problem<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
      &nbsp; &nbsp; statement in target-uri in that<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; RFC 3261 doesn't differentiate the<BR>mechanism by which=20
      the<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt; new<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; (target) =
Request-URI is=20
      selected. &nbsp;Another<BR>issue is that<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; some of the =
terminology=20
      in<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; RFC 3261 is overloaded - e.g.,=20
      "forwarding"<BR>refers both to<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;a<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; Proxy<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
which does=20
      not have responsibility for the<BR>domain of the<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp;=20
      request-URI<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; in the incoming =
request,=20
      thus the proxy<BR>just "forwards" the<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; request =
to<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; the next hop AND "forwarding" is used =
to<BR>describe=20
      the<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
      &nbsp; &nbsp; process whereby<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; the =
outgoing=20
      request is built and<BR>"forwarded" to the next<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; hop at=20
      which<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; point the proxy does not know =
how the=20
      new<BR>request-uri was<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; selected.<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; RFC=20
      4244 has attempted to clarify the terms<BR>and attempts =
to<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; =
&nbsp; use=20
      "forward"<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; in the context =
of the=20
      former situation and<BR>"retarget" for<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; the case=20
      whereby<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; a proxy is responsible for the =
domain=20
      and<BR>thus can use a<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; number of<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      mechanism to select the new target for the<BR>request - =
e.g.,<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;a<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp;=20
      REGISTRAR,<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; configured data, =

      etc.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp;3. &nbsp;Related to the last =
point in=20
      item 2 above,<BR>it has been<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; proposed that<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; we differentiate the hi-entries even more<BR>by =
defining<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; =
&nbsp;=20
      separate parameters<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;for registered =
and=20
      configured/mapped<BR>contacts.<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;Currently=20
      when the R-URI is translated to a<BR>URI which is<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; either=20
      derived<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;from location service =
lookup(registered=20
      by<BR>UA) or from<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt; mapping<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;table, there is =
no=20
      differentiation as to how<BR>the URI was<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; derived once it=20
      is<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;added to the list of potential=20
      targets.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;The general =
consensus of=20
      the authors of the<BR>two documents<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;was<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; that it may<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;be useful for some services to have the<BR>hi-entries=20
      tagged<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
      &nbsp; &nbsp; with the<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;more specific=20
      information.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;And, of course, =
this gets=20
      us into another<BR>naming contest. In<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; the end, the=20
      naming<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;is not so important as long as =
the=20
      term<BR>isn't too overloaded<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; and it is defined<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;precisely in the document(s).<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; We would appreciate WG =
feedback=20
      on these issues<BR>and any other<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; comments on<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; =
the two=20
      documents prior to IETF-74.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; Regards,<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; Shida and=20
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      &nbsp;&gt;&gt; &nbsp; &nbsp; Begin forwarded message:<BR>&nbsp; =
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href=3D"mailto:Internet-Drafts@ietf.org">Internet-Drafts@ietf.org</A><BR>=
&nbsp;=20
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      =
href=3D"mailto:Internet-Drafts@ietf.org">Internet-Drafts@ietf.org</A>&gt;=
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      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; =
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      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; &nbsp; =
&nbsp; *Date:=20
      *March 10, 2009 2:30:01 AM JST<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
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      &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; *To: *<A=20
      =
href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</A><BR>&lt;ma=
ilto:<A=20
      =
href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</A>&gt;<BR>&n=
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      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; =
&nbsp; &nbsp;=20
      *Subject: **I-D<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      =
&nbsp;&gt;&gt;&gt;<BR>ACTION:draft-rosenberg-sip-target-uri-delivery-01.t=
xt=20
      *<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
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      &nbsp; &nbsp; *Reply-To: *<A=20
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      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; =
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&nbsp;&gt;&gt;&gt;<BR>&nbsp;=20
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      A New Internet-Draft is available from the<BR>on-line<BR>&nbsp; =
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      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt;<BR>&nbsp; =
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      directories.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
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      Request-URI Targets to User Agents<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
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      &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; Author(s)<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; &nbsp; =
&nbsp; : J.=20
      Rosenberg, H. van Elburg, C. Holmberg, F.<BR>Audet, S.<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Schubert<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; =
Filename=20
      :<BR>draft-rosenberg-sip-target-uri-delivery-01.txt<BR>&nbsp; =
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      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; &nbsp; =
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      Pages<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; : 16<BR>&nbsp; &nbsp; &nbsp; =
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      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; =
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      proxy<BR>receives a<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
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      &nbsp;request<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; &nbsp; targeted at a URI =
identifying a=20
      user or<BR>resource it is<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
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      &nbsp; &nbsp;&gt;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
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      &nbsp; &nbsp;&gt; responsible<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
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      &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
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      &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; &nbsp; for, the proxy translates =
the URI=20
      to a<BR>registered or<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;configured<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; &nbsp; contact URI of an agent=20
      representing that user<BR>or resource.<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; In<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; =
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      &nbsp; &nbsp; &nbsp; &nbsp;&gt; the<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
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      &nbsp; &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; &nbsp; process, the =
original URI=20
      is removed from the<BR>request.<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; &nbsp;Numerous=20
      use<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;&gt;=20
      &nbsp; &nbsp; &nbsp; cases have arisen which require =
this<BR>information=20
      to be<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt; delivered to<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; &nbsp; the user agent. &nbsp;This =

      document describes these<BR>use cases and<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; =
defines<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; =
&nbsp; &nbsp;=20
      &nbsp; an extension to the History-Info header field<BR>which =
allows=20
      it<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
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      &nbsp;&gt; to be<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; &nbsp; used to support those=20
      cases.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; A URL for this Internet-Draft=20
      is:<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt;&gt;<BR><A=20
      =
href=3D"http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-ur=
i-d"=20
      =
target=3D_blank>http://www.ietf.org/internet-drafts/draft-rosenberg-sip-t=
arget-uri-d</A><BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; =
e<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt;=20
      livery-01.txt<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; Internet-Drafts are also =
available by=20
      anonymous<BR>FTP at:<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; <A=20
      href=3D"ftp://ftp.ietf.org/internet-drafts/"=20
      target=3D_blank>ftp://ftp.ietf.org/internet-drafts/</A><BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt;<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; &nbsp; =
&nbsp; Below=20
      is the data which will enable a MIME<BR>compliant mail =
reader<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; =
&nbsp; &nbsp;=20
      implementation to automatically retrieve the<BR>ASCII version =
of<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;the<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; &nbsp; =
&nbsp;=20
      Internet-Draft.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; Content-Type:=20
      text/plain&lt;BR&gt;Content-ID:<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; &amp;<A=20
      =
href=3D"mailto:lt%3B2009-3-9102756.I-D@ietf.org">lt;2009-3-9102756.I-D@ie=
tf.org</A><BR></DIV></DIV>&lt;mailto:<A=20
      =
href=3D"mailto:lt%253B2009-3-9102756.I-D@ietf.org">lt%3B2009-3-9102756.I-=
D@ietf.org</A>&gt;<BR>
      <DIV>
      <DIV></DIV>
      <DIV class=3Dh5>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; &lt;mailto:<A=20
      =
href=3D"mailto:2009-3-9102756.I-D@ietf.org">2009-3-9102756.I-D@ietf.org</=
A>&gt;&amp;gt;&lt;BR&gt;&lt;BR&gt;<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; &nbsp; =
&nbsp;=20
      _______________________________________________<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; =
I-D-Announce=20
      mailing list<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
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      =
href=3D"mailto:I-D-Announce@ietf.org">I-D-Announce@ietf.org</A><BR>&lt;ma=
ilto:<A=20
      =
href=3D"mailto:I-D-Announce@ietf.org">I-D-Announce@ietf.org</A>&gt;<BR>&n=
bsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt;<BR><A =

      href=3D"https://www.ietf.org/mailman/listinfo/i-d-announce"=20
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target=3D_blank>https://www.ietf.org/mailman/listinfo/i-d-announce</A><BR=
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href=3D"http://www.ietf.org/shadow.html"=20
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      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;&gt; &nbsp; &nbsp; or <A =

      href=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt"=20
      =
target=3D_blank>ftp://ftp.ietf.org/ietf/1shadow-sites.txt</A><BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;&gt;<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; =
&nbsp;=20
      _______________________________________________<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; &nbsp; Sip =
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      list<BR><A href=3D"https://www.ietf.org/mailman/listinfo/sip"=20
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      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; =
&nbsp;=20
      This list is for NEW development of the core =
SIP<BR>Protocol<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; =
&nbsp; Use=20
      <A=20
      =
href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.colu=
mbia.edu</A><BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt; &nbsp; =
&nbsp;=20
      &lt;mailto:<A=20
      =
href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.colu=
mbia.edu</A>&gt;=20
      for<BR>questions on<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;&gt;&gt; current<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt; sip<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;&gt;&gt; &nbsp; &nbsp; Use <A=20
      href=3D"mailto:sipping@ietf.org">sipping@ietf.org</A> =
&lt;mailto:<A=20
      href=3D"mailto:sipping@ietf.org">sipping@ietf.org</A>&gt;<BR>for=20
      new<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&gt;&gt;=20
      &nbsp; &nbsp; developments on the application of sip<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&gt;&gt;<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;=20
    =
&nbsp;&gt;&gt;<BR><BR><BR><BR></DIV></DIV></BLOCKQUOTE></DIV><BR></BLOCKQ=
UOTE></BLOCKQUOTE></BODY></HTML>

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Thread-Topic: [Sip] Terminology (was RE: Fwd: I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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References: <899F1518-6872-442E-AD3E-D3F343A605DC@ntt-at.com><9ae56b1e0903110158m1e6fb07o66110501e8581971@mail.gmail.com><1ECE0EB50388174790F9694F77522CCF1CCA681D@zrc2hxm0.corp.nortel.com><49B82FB4.4030002@gmail.com><1ECE0EB50388174790F9694F77522CCF1CCF97F5@zrc2hxm0.corp.nortel.com><49B84C55.5070107@gmail.com><1ECE0EB50388174790F9694F77522CCF1CCF9A06@zrc2hxm0.corp.nortel.com><9ae56b1e0903120134i76a6a92brb183205cd66e841f@mail.gmail.com><0D5F89FAC29E2C41B98A6A762007F5D001A27AF1@GBNTHT12009MSX.gb002.siemens.net><9ae56b1e0903120556t73208627v39b3c6221098cc44@mail.gmail.com> <0D5F89FAC29E2C41B98A6A762007F5D001A27C50@GBNTHT12009MSX.gb002.siemens.net>
From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens.com>, "Hans Erik van Elburg" <ietf.hanserik@gmail.com>
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Subject: Re: [Sip] Terminology (was RE: Fwd: I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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=20
Hi,
=20
IF we would remove that requierment, and ONLY care about the case when
the R-URI is replaced with the CONTACT, I don't think we need to do
anything. I believe P-Called-Party-ID already solves that.
=20
Regards,
=20
Christer


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Elwell, John
	Sent: 12. maaliskuuta 2009 15:31
	To: Hans Erik van Elburg
	Cc: sip@ietf.org
	Subject: Re: [Sip] Terminology (was RE: Fwd:
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
=09
=09
	Hans Erik,
	=20
	The way I interpreted the 01 draft, you would not get the
freephone number anyway. If that is a requirement, then I accept my
suggestion won't suffice.
	=20
	John


________________________________

		From: Hans Erik van Elburg
[mailto:ietf.hanserik@gmail.com]=20
		Sent: 12 March 2009 12:57
		To: Elwell, John
		Cc: sip@ietf.org
		Subject: Re: [Sip] Terminology (was RE: Fwd:
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
	=09
	=09
		But that does not solve the freephone service use case.
Freephone represents a use case where the R-URI is replaced with an AoR
of user B - not with the contact of user B.=20
	=09
		A calls Bfreephonenumber routed the B (AOR) routed to
B's contact.
	=09
		The target-uri draft is interested in obtaining:
Bfreephonenumber as that is what A used to reach B.
	=09
		So what we like to see that B UA receives is something
like
		    H-I:  Bfreephone; "istarget",\
		        B,\
		        contact-B
	=09
		So you want to get Bfreephonenumber and not B.
	=09
		In certain situations you are also interested in B, I
guess that  PBX's would interested in that to deliver to the correct
user. We could distinguish them like:
		    H-I:  Bfreephone; "istarget",\
		        B,"aor"\
		        contact-B
	=09
		BR,
		/Hans Erik van Elburg
	=09
	=09
	=09
		On Thu, Mar 12, 2009 at 12:08 PM, Elwell, John
<john.elwell@siemens.com> wrote:
	=09

			If 'istarget' is used only to identify the value
that was overwritten in
			the Request-URI by a contact URI, an alternative
approach would be to
			flag the contact URI H-I entry as 'iscontact'.
Then the UAS would just
			need to look for the most recent H-I entry that
is not marked
			'iscontact'.
		=09
			John
		=09
		=09
			________________________________
		=09
			       From: sip-bounces@ietf.org
[mailto:sip-bounces@ietf.org] On
			Behalf Of Hans Erik van Elburg
			       Sent: 12 March 2009 08:35
			       To: Mary Barnes
			       Cc: sip@ietf.org
			       Subject: Re: [Sip] Terminology (was RE:
Fwd:
=09
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
		=09


			       Yes, because you are using the 3261 use
of target and the
			4244bis introduced definition of retarget. I
thought it was clear that
			we need other words as those definitions don't
match the target-uri
			drafts use of the terms. Also they do not
suffice to provide a solution
			for the use cases in the target-uri draft.
		=09
			       The 3261 text you refer to is exactly
about the case where the
			home proxy overwrites the Request-URI with a new
target. This target is
			teh registered contact address. And hence this
would be what target-uri
			calls a hop or a route. This case and this is
where it gets confusing is
			not a "retarget" in the target-uri draft use of
the term.
		=09
			       The target-uri draft states:
		=09
			          "To avoid confusion, we
			          refer to a SIP URI that is an address
for a user or resource
			as a
			          "target" and a SIP URI that is a hop
for reaching that user
			as a
		=09
			          "hop".
		=09
		=09
			       Apparently that does not suffice to avoid
confusion.
		=09
			       As for the tagging, speaking about the
solution before agreeing
			on the terminology and the problem it should
solve is meaningless.
		=09
			       /Hans Erik van Elburg
		=09
		=09
		=09
			       On Thu, Mar 12, 2009 at 2:33 AM, Mary
Barnes
			<mary.barnes@nortel.com> wrote:
		=09
		=09
			               Responses below [MB].
		=09
		=09
			               -----Original Message-----
			               From: Hans Erik van Elburg
			[mailto:ietf.hanserik@gmail.com]
		=09
			               Sent: Wednesday, March 11, 2009
6:42 PM
			               To: Barnes, Mary (RICH2:AR00)
			               Cc: Shida Schubert; sip@ietf.org
			               Subject: Re: Terminology (was RE:
[Sip] Fwd: I-D
=09
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
		=09
		=09
			               I was talking about the concept.
		=09
			               The use cases only describe
cases where the target-uri
			(lets call that
			               current target from now) has been
lost when an initial
			request for a
			               dialog or standalone request
arrives at the UAS.
		=09
			               [MB] And I think this is where
the terminology confusion
			starts - I
			               don't think of the "lost"
target-uri as being the
			"current" target. In
			               my mind, the "current" target is
reflected by the last
			hi-entry and the
			               request-uri in the incoming
request at the UAS. If the
			entity that sent
			               the request was the entity that
added the last hi-entry,
			then the uri in
			               that hi-entry is the same as the
request-URI in the SIP
			request that
			               arrives at the UAS. I refer to
the "lost" uri as the one
			that was
			               "retargeted" - that's the one the
UAS wants to pull from
			the hi-entries
			               in the incoming request. That
hi-entry was not the one
			that was just
			               added by the entity that built
the request just received
			by the UAS.
			               That hi-entry is tagged with
whatever name we are going
			to tag it with
			               BEFORE the request is forwarded
(using the term forward
			per section 16.6
			               of RFC 3261).  That tag is added
once the target set of
			potential
			               candidates for the new request
uri are determined in
			section 16.5 of
			               3261 (with "target set" being a
3261 term), just before
			the request is
			               forwarded in section 16.6 to one
of those targets.  A
			new hi-entry
			               (which will be the last hi-entry
in the request received
			by the UAS) is
			               added in section 16.6 of 3261 as
the request is
			forwarded.  At this
			               point in time, the lost
information is in the previous
			hi-entry when the
			               outgoing request is sent. [/MB]
		=09
		=09
			               What has been lost is the current
target of the request:
		=09
			               [MB] Right - at which point in my
mind, it's no longer
			the current
			               target ;) Maybe we call it
"lost". [/MB]
		=09
		=09
		=09
			                    Current target
		=09
			                 The current target of an
initial request for a dialog
			or standalone
			                 request is the name or address
to which the request is
			targeted, i.e.
			                 either the initial target
inserted in the Request-URI
			by the UAC that
			                 originates the request, or when
a retarget occurred,
			the target
			                 provided in that retarget
operation.  Reroute and
			translation
			                 operations never change the
current target.
		=09
		=09
			               [MB] I don't think this
definition fits what you want -
			i.e., if there
			               is no retargeting, then none of
the hi-entries are
			tagged - i.e., you
			               won't have your concept of
current. The way it works is
			that if no
			               hi-entries are tagged, then you
know that there was no
			retargeting, thus
			               you know the request-uri has not
been lost. [/MB]
		=09
		=09
			               This defintion only makes sense
when the following
			definitions are used:
		=09
			               Name:
		=09
			                 A name is a moniker for an
entity which refers to it
			in a way which
		=09
			                 reveals nothing about where it
is in a network.  In
			SIP, tel URI
		=09
			                 which doesn't represent the
location of the entity is
			a name.
		=09
		=09
			               Address:
		=09
			                 An address is an identifier for
an entity which
			describes it by its
		=09
			                 location on the network.  In
SIP, the SIP URI itself
			is a form of
		=09
			                 address because the host part
of the URI, the only
			mandatory part of
		=09
			                 the URI besides the scheme
itself, indicates the
			location of a SIP
		=09
			                 server that can be used to
handle the request.
		=09
			               Route:
		=09
			                 Finally, a route is a
sequence of SIP entities
			(including the UA
			               itself!) which are
		=09
			                traversed in order to forward a
request to an address
			or name.
		=09
			               Retarget (other term might be
needed, as this is highly
			confusing):
		=09
			                  A Request-URI rewrite
operation that changes the
			target identity of
			               the request.
		=09
			               Reroute (other term might be
needed):
		=09
			                  A Request-URI rewrite
operation that does not change
			the current
			               target of the request, but
determines the route/next
			hop taken to
			               reach the target-identity.
		=09
			               /Hans Erik
		=09
		=09


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<BODY>
<DIV dir=3Dltr align=3Dleft>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft>
<DIV dir=3Dltr align=3Dleft><SPAN =
class=3D034104813-12032009>Hi,</SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN =
class=3D034104813-12032009></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D034104813-12032009>IF we would =
remove that=20
requierment, and&nbsp;ONLY care about the case when the R-URI is =
replaced with=20
the CONTACT, I don't think we need to do anything. I believe =
P-Called-Party-ID=20
already solves that.</SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN =
class=3D034104813-12032009></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN =
class=3D034104813-12032009>Regards,</SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN =
class=3D034104813-12032009></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN=20
class=3D034104813-12032009>Christer</SPAN></DIV></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Elwell, =
John<BR><B>Sent:</B>=20
  12. maaliskuuta 2009 15:31<BR><B>To:</B> Hans Erik van =
Elburg<BR><B>Cc:</B>=20
  sip@ietf.org<BR><B>Subject:</B> Re: [Sip] Terminology (was RE: Fwd:=20
  =
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR></FONT><BR></=
DIV>
  <DIV></DIV>
  <DIV dir=3Dltr align=3Dleft>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D441300613-12032009><FONT =
face=3DArial=20
  color=3D#000080>Hans Erik,</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D441300613-12032009><FONT =
face=3DArial=20
  color=3D#000080></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D441300613-12032009><FONT =
face=3DArial><FONT=20
  color=3D#000080>The way I interpreted the 01 draft, you would not get =
the=20
  freephone number anyway. If that is a requirement, then I accept my =
suggestion=20
  won't suffice<SPAN=20
class=3D125123013-12032009>.</SPAN></FONT></FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D441300613-12032009><FONT =
face=3DArial=20
  color=3D#000080></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D441300613-12032009><FONT =
face=3DArial=20
  color=3D#000080>John</FONT></SPAN></DIV></DIV><BR>
  <BLOCKQUOTE=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000080 2px =
solid; MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft>
    <HR tabIndex=3D-1>
    <FONT face=3DTahoma size=3D2><B>From:</B> Hans Erik van Elburg=20
    [mailto:ietf.hanserik@gmail.com] <BR><B>Sent:</B> 12 March 2009=20
    12:57<BR><B>To:</B> Elwell, John<BR><B>Cc:</B>=20
    sip@ietf.org<BR><B>Subject:</B> Re: [Sip] Terminology (was RE: Fwd:=20
    =
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR></FONT><BR></=
DIV>
    <DIV></DIV>But that does not solve the freephone service use case.=20
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</STYLE>
    Freephone represents a use case where the R-URI is replaced with an =
AoR of=20
    user B - not with the contact of user B. <BR><BR>A calls =
Bfreephonenumber=20
    routed the B (AOR) routed to B's contact.<BR><BR>The target-uri =
draft is=20
    interested in obtaining: Bfreephonenumber as that is what A used to =
reach=20
    B.<BR><BR>So what we like to see that B UA receives is something=20
    like<BR>&nbsp;&nbsp;&nbsp; H-I:&nbsp; Bfreephone;=20
    "istarget",\<BR>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;=20
    B,\<BR>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; contact-B<BR><BR>So you =
want to=20
    get Bfreephonenumber and not B.<BR><BR>In certain situations you are =
also=20
    interested in B, I guess that&nbsp; PBX's would interested in that =
to=20
    deliver to the correct user. We could distinguish them=20
    like:<BR>&nbsp;&nbsp;&nbsp; H-I:&nbsp; Bfreephone;=20
    "istarget",\<BR>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;=20
    B,"aor"\<BR>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
contact-B<BR><BR>BR,<BR=20
    clear=3Dall>/Hans Erik van Elburg<BR><BR><BR>
    <DIV class=3Dgmail_quote>On Thu, Mar 12, 2009 at 12:08 PM, Elwell, =
John <SPAN=20
    dir=3Dltr>&lt;<A=20
    =
href=3D"mailto:john.elwell@siemens.com">john.elwell@siemens.com</A>&gt;</=
SPAN>=20
    wrote:<BR>
    <BLOCKQUOTE class=3Dgmail_quote=20
    style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">If=20
      'istarget' is used only to identify the value that was overwritten =

      in<BR>the Request-URI by a contact URI, an alternative approach =
would be=20
      to<BR>flag the contact URI H-I entry as 'iscontact'. Then the UAS =
would=20
      just<BR>need to look for the most recent H-I entry that is not=20
      =
marked<BR>'iscontact'.<BR><BR>John<BR><BR><BR>___________________________=
_____<BR><BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp;From: <A=20
      href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</A> =
[mailto:<A=20
      href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</A>] =
On<BR>Behalf=20
      Of Hans Erik van Elburg<BR>&nbsp; &nbsp; &nbsp; &nbsp;Sent: 12 =
March 2009=20
      08:35<BR>&nbsp; &nbsp; &nbsp; &nbsp;To: Mary Barnes<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp;Cc: <A =
href=3D"mailto:sip@ietf.org">sip@ietf.org</A><BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp;Subject: Re: [Sip] Terminology (was RE:=20
      =
Fwd:<BR>I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR>
      <DIV>
      <DIV></DIV>
      <DIV class=3Dh5><BR><BR>&nbsp; &nbsp; &nbsp; &nbsp;Yes, because =
you are=20
      using the 3261 use of target and the<BR>4244bis introduced =
definition of=20
      retarget. I thought it was clear that<BR>we need other words as =
those=20
      definitions don't match the target-uri<BR>drafts use of the terms. =
Also=20
      they do not suffice to provide a solution<BR>for the use cases in =
the=20
      target-uri draft.<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp;The 3261 text =
you=20
      refer to is exactly about the case where the<BR>home proxy =
overwrites the=20
      Request-URI with a new target. This target is<BR>teh registered =
contact=20
      address. And hence this would be what target-uri<BR>calls a hop or =
a=20
      route. This case and this is where it gets confusing is<BR>not a=20
      "retarget" in the target-uri draft use of the term.<BR><BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp;The target-uri draft states:<BR><BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; "To avoid confusion, we<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; refer to a SIP URI that is an address for a user or =
resource<BR>as=20
      a<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; "target" and a SIP URI =
that is a=20
      hop for reaching that user<BR>as a<BR><BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; "hop".<BR><BR><BR>&nbsp; &nbsp; &nbsp; &nbsp;Apparently =
that does=20
      not suffice to avoid confusion.<BR><BR>&nbsp; &nbsp; &nbsp; =
&nbsp;As for=20
      the tagging, speaking about the solution before agreeing<BR>on the =

      terminology and the problem it should solve is =
meaningless.<BR><BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp;/Hans Erik van Elburg<BR><BR><BR><BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp;On Thu, Mar 12, 2009 at 2:33 AM, Mary =
Barnes<BR>&lt;<A=20
      =
href=3D"mailto:mary.barnes@nortel.com">mary.barnes@nortel.com</A>&gt;=20
      wrote:<BR><BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;Responses below [MB].<BR><BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;-----Original Message-----<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;From: Hans Erik van =
Elburg<BR>[mailto:<A=20
      =
href=3D"mailto:ietf.hanserik@gmail.com">ietf.hanserik@gmail.com</A>]<BR><=
BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Sent: Wednesday, =
March 11,=20
      2009 6:42 PM<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;To:=20
      Barnes, Mary (RICH2:AR00)<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;Cc: Shida Schubert; <A=20
      href=3D"mailto:sip@ietf.org">sip@ietf.org</A><BR>&nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;Subject: Re: Terminology (was RE: [Sip] =
Fwd:=20
      I-D<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      =
&nbsp;ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR><BR><BR>&n=
bsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;I was talking =
about the=20
      concept.<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;The=20
      use cases only describe &nbsp;cases where the target-uri<BR>(lets =
call=20
      that<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;current=20
      target from now) has been lost when an initial<BR>request for =
a<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;dialog or =
standalone=20
      request arrives at the UAS.<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;[MB] And I think this is where the terminology =

      confusion<BR>starts - I<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;don't think of the "lost" target-uri as being=20
      the<BR>"current" target. In<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;my mind, the "current" target is reflected by the=20
      last<BR>hi-entry and the<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;request-uri in the incoming request at the UAS. If=20
      the<BR>entity that sent<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;the request was the entity that added the last=20
      hi-entry,<BR>then the uri in<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;that hi-entry is the same as the request-URI in the=20
      SIP<BR>request that<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;arrives at the UAS. I refer to the "lost" uri as the =
one<BR>that=20
      was<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;"retargeted"=20
      - that's the one the UAS wants to pull from<BR>the =
hi-entries<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;in the incoming =
request.=20
      That hi-entry was not the one<BR>that was just<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;added by the entity that built =
the=20
      request just received<BR>by the UAS.<BR>&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp; &nbsp; &nbsp;That hi-entry is tagged with whatever name we =
are=20
      going<BR>to tag it with<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;BEFORE the request is forwarded (using the term=20
      forward<BR>per section 16.6<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;of RFC 3261). &nbsp;That tag is added once the target =
set=20
      of<BR>potential<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;candidates for the new request uri are determined =
in<BR>section 16.5=20
      of<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;3261 =
(with=20
      "target set" being a 3261 term), just before<BR>the request =
is<BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;forwarded in =
section 16.6=20
      to one of those targets. &nbsp;A<BR>new hi-entry<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;(which will be the last hi-entry =
in the=20
      request received<BR>by the UAS) is<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;added in section 16.6 of 3261 as the request=20
      is<BR>forwarded. &nbsp;At this<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp;point in time, the lost information is in the=20
      previous<BR>hi-entry when the<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp; &nbsp;outgoing request is sent. [/MB]<BR><BR><BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;What has been lost is the =
current=20
      target of the request:<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;[MB] Right - at which point in my mind, it's no =
longer<BR>the=20
      current<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;target=20
      ;) Maybe we call it "lost". [/MB]<BR><BR><BR><BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Current=20
      target<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;The current target of an initial request for a dialog<BR>or=20
      standalone<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;request is the name or address to which the request =
is<BR>targeted,=20
      i.e.<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;either the initial target inserted in the Request-URI<BR>by =
the UAC=20
      that<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;originates the request, or when a retarget occurred,<BR>the=20
      target<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;provided in that retarget operation. &nbsp;Reroute=20
      and<BR>translation<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;operations never change the current =
target.<BR><BR><BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;[MB] I don't think =
this=20
      definition fits what you want -<BR>i.e., if there<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;is no retargeting, then none of =
the=20
      hi-entries are<BR>tagged - i.e., you<BR>&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp; &nbsp; &nbsp;won't have your concept of current. The way it =
works=20
      is<BR>that if no<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;hi-entries are tagged, then you know that there was=20
      no<BR>retargeting, thus<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;you know the request-uri has not been lost.=20
      [/MB]<BR><BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;This defintion only makes sense when the =
following<BR>definitions=20
      are used:<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;Name:<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp;A name is a moniker for an entity which refers to it<BR>in a =
way=20
      which<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;reveals nothing about where it is in a network. =
&nbsp;In<BR>SIP, tel=20
      URI<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;which doesn't represent the location of the entity is<BR>a=20
      name.<BR><BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;Address:<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;An address is an identifier for an entity =
which<BR>describes=20
      it by its<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;location on the network. &nbsp;In SIP, the SIP URI =
itself<BR>is a=20
      form of<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp;address because the host part of the URI, the =
only<BR>mandatory part=20
      of<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
      &nbsp;the URI besides the scheme itself, indicates the<BR>location =
of a=20
      SIP<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =

      &nbsp;server that can be used to handle the request.<BR><BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Route:<BR><BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Finally, a route =
is a=20
      &nbsp; sequence of SIP entities<BR>(including the UA<BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;itself!) which =
are<BR><BR>&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; traversed in =
order to=20
      forward a request to an address<BR>or name.<BR><BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Retarget (other term might be =
needed, as=20
      this is highly<BR>confusing):<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; A Request-URI rewrite operation that =
changes=20
      the<BR>target identity of<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp;the request.<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
      &nbsp; &nbsp;Reroute (other term might be needed):<BR><BR>&nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; A Request-URI =
rewrite=20
      operation that does not change<BR>the current<BR>&nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;target of the request, but =
&nbsp; &nbsp;=20
      determines the route/next<BR>hop taken to<BR>&nbsp; &nbsp; &nbsp; =
&nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp;reach the =
target-identity.<BR><BR>&nbsp; &nbsp;=20
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;/Hans=20
    =
Erik<BR><BR></DIV></DIV></BLOCKQUOTE></DIV></BLOCKQUOTE></BLOCKQUOTE></BO=
DY></HTML>

------_=_NextPart_001_01C9A31B.AF9F7A6C--

From drage@alcatel-lucent.com  Thu Mar 12 07:07:00 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: SIP IETF <sip@ietf.org>
Date: Thu, 12 Mar 2009 15:07:28 +0100
Thread-Topic: Long posts from long threads
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Subject: [Sip] Long posts from long threads
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(As WG chair)

There is a length limitation in bytes on the messages posted to the SIP exp=
loder, which I for one am glad of given the amount of time I have to spend =
keeping my inbox from overflowing.

If your message exceeds that length, it is held for moderation until Dean o=
r myself get round to spending the cycles to unblock it. Many times we have=
 better things to do with life.

If you blindly reply to existing messages including all the content from th=
e thread, then you will rapidly reach this limit.=20

Posting non-text mails (e.g. RTF or HTML) also consumes more space and ther=
efore will more rapidly exceed this limit. You do not need to use the MS ou=
tlook default of not sending text - please change these defaults to text/pl=
ain when sending to IETF lists.

regards

Keith=

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Date: Thu, 12 Mar 2009 15:54:52 +0100
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From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: Mary Barnes <mary.barnes@nortel.com>
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Subject: Re: [Sip] Terminology (was RE: Fwd: I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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--0015174bea24a8efb40464ed2a94
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I'm fine with discussing implementation if everyone is meaning the same if
they use the same words.

As  to the "retarget" term such does not seem to be the case.

/Hans Erik van Elburg

--0015174bea24a8efb40464ed2a94
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I&#39;m fine with discussing implementation if everyone is meaning the same=
 if they use the same words.<br><br>As=A0 to the &quot;retarget&quot; term =
such does not seem to be the case.<br><br clear=3D"all">/Hans Erik van Elbu=
rg<br>

<br>

--0015174bea24a8efb40464ed2a94--

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From: "Ian Elz" <ian.elz@ericsson.com>
To: "Mary Barnes" <mary.barnes@nortel.com>
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Cc: sip@ietf.org, Francois Audet <audet@nortel.com>
Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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Mary,

I was only considering the specific domain case as included in paragraph
2 of section 3.3 in the draft:

                                                  Thus, local policy
   MAY also be used to determine whether to include the History-Info
   header at all, whether to capture a specific Request-URI in the
   header, or whether it be included only in the Request as it is
   retargeted within a specific domain (PRIV-req-3).  In the latter
   case, this is accomplished by adding a new priv-value, history, to
   the Privacy header [RFC3323] indicating whether any or a specific
   History-Info header(s) SHOULD be forwarded.

And from Section 4.1=20

   o  Privacy: An optional parameter for History-Info, reflected in the
      History-Info header field values by including the Privacy Header
      [RFC3323] with a priv-value of "history" escaped in the hi-
      targeted-to-uri or by adding the Privacy header with a priv-value
      of "history" to the Request.  The use of the Privacy Header with a
      priv-value of "history" indicates whether a specific or all
      History-Info headers should not be forwarded.

(There appears to be an inconsistency here as sections 3.3 and 4.1
appear to contradict. [Perhaps it just needs clarification that section
3.3 allows forwarding within the domain and 4.1 prohibits forwarding
outside the domain?])

Assuming that the value 'history' means that presentation and forwarding
should not occur.

In this case the inclusion of the value 'history' in the Privacy header
impacts all H-I entries. If one node includes the value 'history' in the
Privacy header then there is no way for another node to specifically
allow the uri included in the hi-targeted-to-uri to not be restricted.
To allow forwarding of a specific H-I entry there needs to be another
value which can be included in the Privacy header parameter of for the
hi-targeted-to-uri.

If the value "none" has specific meanings which will cause confusion
then another value should be used. =20

More comments in-line

Nit in A.3

In both the sequence and the text you indicate that the INVITE to UA-B
will retransmit. This should not occur if a 100 Trying has been received
from UA-B. This means that the retransmit timer cannot be used to
determine if UA-B does not respond. There will be a separate timer
specified for determining no answer after the 180 has been received.

Ian Elz

Office: + 44 24 764 35256
ian.elz@ericsson.com


-----Original Message-----
From: Mary Barnes [mailto:mary.barnes@nortel.com]=20
Sent: 12 March 2009 13:55
To: Ian Elz
Cc: sip@ietf.org; Francois Audet
Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

Ian,

Okay, so I think I might understand now - you want to use the "none" to
override the Privacy header that would be (in current 4244/4244bis)
applied to the whole message as it traverses across the n/w from the UAC
to the UAS. Is that correct?=20
=20
[ige] Yes - currently a Privacy header with value 'history' will prevent
any H-I entries from being passed. If the Privacy header does not
contain the value 'history' then individual H-I entries may have privacy
associated. Those which do will not be forwarded but those with no
privacy will be forwarded.

That then though changes the meaning of the
Privacy header, as I understand it - i.e., the "none" I thought had to
do with specifying that an intermediary MUST NOT change the value of the
Privacy header. But, I guess you are saying we specify a slightly
different usage in 4244bis such that when we add it to an hi-entry, it
specifies that that hi-entry MUST NOT be dropped by an intermediary.
That might work, although, we would need verbage about the "risks" of
doing so.
=20
[ige] I would not be as strong as MUST NOT. There may be local policy
which will override what is included. I would prefer the words NEED NOT
be removed {Scott B will hate me for that.} or perhaps MAY be forwarded
based upon local policy.=20

Thanks,
Mary.=20

-----Original Message-----
From: Ian Elz [mailto:ian.elz@ericsson.com]=20
Sent: Thursday, March 12, 2009 8:43 AM
To: Barnes, Mary (RICH2:AR00)
Cc: sip@ietf.org; Audet, Francois (SC100:3055)
Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

Mary,

(Written after reading your discussion with Hans Erik.)

First an apology for not making it clear in my original email that a
value of "none" in a Privacy header parameter in a History-Info entry
was only applicable to the uri contained in the hi-targeted-to-uri.

I only suggested using the value "none" as this is a value defined for
the Privacy header which explicitly indicates that privacy is not
required. Any other value with the meaning of explicitly indicating that
there is no privacy required for the specific uri could be used.

With regard to your second paragraph below I understand that the SIP and
PSTN models do not align. You are correct in your assumption that this
problem is not unique for History-Info. The problem also arises with the
Referred-By header.=20

I have raised this for History-Info at this time as your new draft
presents an opportunity to correct this problem for the H-I case.

The general SIP privacy issue arises as when the Privacy header was
defined the concept of carrying third party identities in the SIP
messages was not defined. This came later with the REFER Request and the
associated Referred-By header and the History-Info header.=20

We should take what opportunities arise to discuss and correct this if
there is a will to make these changes.

Ian Elz

Office: + 44 24 764 35256
ian.elz@ericsson.com


-----Original Message-----
From: Mary Barnes [mailto:mary.barnes@nortel.com]
Sent: 11 March 2009 16:53
To: Ian Elz; Francois Audet
Cc: sip@ietf.org
Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

Ian,

This is an interesting question. I need to think about it some more, but
I don't like the "none" idea as it really must be up to the entity that
added the Privacy header to the request, as to whether it wants the HI
entries that it adds to go outside a domain. My initial thought is that
we can't overrule the overall Privacy header. The thing is that the
Privacy header doesn't preclude gathering the information and using
within a domain AND if it were to not include an hi-entry when the
request leaves the domain for which the proxy is responsible, the
recipient can add the hi-entry for the current request-uri before it
adds the new hi-entry. =20

Your PSTN example doesn't strictly map obviously to a SIP model and I
would think you might consider the PSTN hop to be within the same domain
for which the proxy (or gw, I guess) is responsible OR consider this a
walled garden whereby the Privacy header doesn't apply. Which brings me
to a more general question as to how you all deal with other headers
when you do your mapping to the PSTN when there is a Privacy header?  It
would seem this problem isn't unique to History-Info, although I know
little about the details of PSTN I/W.

Regards,
Mary.=20

-----Original Message-----
From: Ian Elz [mailto:ian.elz@ericsson.com]
Sent: Wednesday, March 11, 2009 4:54 AM
To: Barnes, Mary (RICH2:AR00); Audet, Francois (SC100:3055)
Cc: sip@ietf.org
Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

Mary, Francois,

Section 4.1 of the draft allows for the addition of the Privacy header
parameter with a value of "history" to be included with the
hi-targeted-to-uri.

Can this be extended to also allow the value "none".

A node adding a H-I entry is allowed to specify the privacy value
"history" either in the Privacy header or as a header parameter
associated with the hi-targeted-to-uri. If the former action is taken by
one node this results in privacy for all history info header entries
even if this is not required. There is no way for privacy of an
individual H-I entry to be set to none if no Privacy is required for the
uri.

This creates a specific problem when interworking with the PSTN where
the privacy is associated with each E.164 number included in the
protocol. If an INVITE is being mapped to ISUP and the H-I entries are
being used to map to redirection numbers then a single Privacy header
with the value "history" will result in all numbers being restricted.

If the Privacy header parameter could include the value "none" then this
would explicitly indicate that the associated uri was allowed to be
presented.

The ability to explicitly allow presentation of specific H-I entries may
also be useful in pure sip implementations.

I don't believe that this change would create any backward compatibility
issues. The existing implementations will continue as deployed but new
implementations could explicitly set no privacy for a specific uri.

There are no security issues other than those already defined in the
draft that an intervening node could modify an existing H-I entry. The
Privacy header value of "none" should only be included by the node in
the same manner as the existing inclusion of the "history" value.

Ian Elz


-----Original Message-----
From: i-d-announce-bounces@ietf.org
[mailto:i-d-announce-bounces@ietf.org] On Behalf Of
Internet-Drafts@ietf.org
Sent: 04 March 2009 15:15
To: i-d-announce@ietf.org
Subject: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

A New Internet-Draft is available from the on-line Internet-Drafts
directories.

	Title           : An Extension to the Session Initiation
Protocol (SIP) for Request History Information
	Author(s)       : M. Barnes, F. Audet
	Filename        : draft-barnes-sip-rfc4244bis-00.txt
	Pages           : 49
	Date            : 2009-03-04

This document defines a standard mechanism for capturing the history
information associated with a Session Initiation Protocol (SIP) request.
This capability enables many enhanced services by providing the
information as to how and why a call arrives at a specific application
or user.  This document defines a new optional SIP header, History-Info,
for capturing the history information in requests.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-barnes-sip-rfc4244bis-00.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

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Internet-Draft.

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<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Mary,</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">I =
was only considering the specific domain case as included in paragraph 2 =
of</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier =
New">section 3.3 in</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New"></FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 =
FACE=3D"Courier New">the dra</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D2 FACE=3D"Courier New">ft:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 =
FACE=3D"Courier New">Thus, local policy</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; MAY also be used to determine whether to include the =
History-Info</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; header at all, whether to capture a specific =
Request-URI in the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; header, or whether it be included only in the Request =
as it is</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; retargeted within a specific domain =
(PRIV-req-3).&nbsp; In the latter</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; case, this is accomplished by adding a new priv-value, =
history, to</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; the Privacy header [RFC3323] indicating whether any or =
a specific</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; History-Info header(s)</FONT></SPAN><SPAN =
LANG=3D"en-us"><B> <FONT SIZE=3D2 FACE=3D"Courier New">SHOULD be =
forwarded</FONT></B></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New">.</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">And from Section 4.1 </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; o&nbsp; Privacy: An optional parameter for =
History-Info, reflected in the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; History-Info header field values by =
including the Privacy Header</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [RFC3323] with a priv-value of =
&quot;history&quot; escaped in the hi-</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; targeted-to-uri or by adding the =
Privacy header with a priv-value</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; of &quot;history&quot; to the =
Request.&nbsp; The use of the Privacy Header with a</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; priv-value of &quot;history&quot; =
indicates whether a specific or all</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; History-Info =
headers</FONT></SPAN><SPAN LANG=3D"en-us"><B> <FONT SIZE=3D2 =
FACE=3D"Courier New">should not be forwarded</FONT></B></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">.</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">(</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">There appears to be an inconsistency here as =
section</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">s</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New"> 3.3 and 4.1 appear to contradict.</FONT></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New"></FONT></SPAN><SPAN =
LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier New">[</FONT></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">Perhaps it just =
need</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">s</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New"> clarification that section 3.3 allows forwarding within the domain =
and 4.1 prohi</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New">b</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New">its forwarding</FONT></SPAN><SPAN LANG=3D"en-us"> =
<FONT SIZE=3D2 FACE=3D"Courier New">outside the</FONT></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New"></FONT></SPAN><SPAN =
LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier =
New">domain?]</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New">)</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Assuming that the value 'history'</FONT></SPAN><SPAN =
LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier New">means that =
presentation and forwarding should not occur.</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">In =
this case</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 =
FACE=3D"Courier New">the</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D2 FACE=3D"Courier New"> inclusion of</FONT></SPAN><SPAN =
LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier =
New">the</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New"></FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 =
FACE=3D"Courier New">value 'history' in the Privacy header impacts all =
H-I entries. If one node includes the value 'history'</FONT></SPAN><SPAN =
LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier New">in the Privacy =
header then there is no way for another node to specifically =
allow</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New"> the uri included in the hi-targeted-to-uri to not be =
restricted.</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New"> To allow</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT =
SIZE=3D2 FACE=3D"Courier New">forwarding of a specific H-I =
entry</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New"> there needs to be another value</FONT></SPAN><SPAN LANG=3D"en-us"> =
<FONT SIZE=3D2 FACE=3D"Courier New">which can be included in =
the</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier =
New">Privacy header parameter of for</FONT></SPAN><SPAN LANG=3D"en-us"> =
<FONT SIZE=3D2 FACE=3D"Courier New">the</FONT></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New"> =
hi-targeted-to-uri</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New">.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">If =
the value &quot;none&quot;</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT =
SIZE=3D2 FACE=3D"Courier New">has specific meanings =
which</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier =
New">will</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 =
FACE=3D"Courier New">cause confusion then another value should be =
used.</FONT></SPAN><SPAN LANG=3D"en-us">&nbsp;<FONT SIZE=3D2 =
FACE=3D"Courier New"></FONT></SPAN><SPAN LANG=3D"en-us"> </SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">More comments in-line</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Nit in A.</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New">3</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">In =
both the sequence and</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 =
FACE=3D"Courier New">the</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D2 FACE=3D"Courier New"> text you indicate that</FONT></SPAN><SPAN =
LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier =
New">the</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New"></FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 =
FACE=3D"Courier New">INVITE to UA-B will retransmit. This should not =
occur if a 100 Trying has been received</FONT></SPAN><SPAN =
LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier New">from UA-B. This =
means that the retransmit timer cannot be used</FONT></SPAN><SPAN =
LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier New">to determine if =
UA-B</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier =
New">does not respond. There</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT =
SIZE=3D2 FACE=3D"Courier New">will</FONT></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New"> be =
a</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier =
New">separate</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New"></FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 =
FACE=3D"Courier New">timer specified for determining no =
answer</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New"> after</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 =
FACE=3D"Courier New">the</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D2 FACE=3D"Courier New"> 180 has been received</FONT></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">.</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"></SPAN><A =
NAME=3D""><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">Ian =
Elz</FONT></SPAN></A></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Office: + 44 24 764 35256</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">ian.elz@ericsson.com</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">-----Original Message-----<BR>
From: Mary Barnes [<A =
HREF=3D"mailto:mary.barnes@nortel.com">mailto:mary.barnes@nortel.com</A>]=
<BR>
Sent: 12 March 2009 13:55<BR>
To: Ian Elz<BR>
Cc: sip@ietf.org; Francois Audet<BR>
Subject: RE: I-D =
Action:draft-barnes-sip-rfc4244bis-00.txt</FONT></SPAN><SPAN =
LANG=3D"en-us"> </SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Ian,</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Okay, so I think I might understand now - you want to use the =
&quot;none&quot; to</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">override the Privacy header that would be (in current =
4244/4244bis)</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">applied to the whole message as it traverses across the n/w from =
the UAC</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">to =
the UAS. Is that correct?</FONT></SPAN><SPAN LANG=3D"en-us"> </SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><B><I><FONT SIZE=3D2 FACE=3D"Courier =
New"></FONT></I></B></SPAN><SPAN LANG=3D"en-us">&nbsp;</SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><B><I><FONT SIZE=3D2 FACE=3D"Courier =
New">[ige]</FONT></I></B></SPAN><SPAN LANG=3D"en-us"><B><I> <FONT =
SIZE=3D2 FACE=3D"Courier New">Yes - currently</FONT></I></B></SPAN><SPAN =
LANG=3D"en-us"><B><I> <FONT SIZE=3D2 FACE=3D"Courier New">a Privacy =
header with value 'history' will prevent any H-I entries from being =
passed. If the</FONT></I></B></SPAN><SPAN LANG=3D"en-us"><B><I> <FONT =
SIZE=3D2 FACE=3D"Courier New">P</FONT></I></B></SPAN><SPAN =
LANG=3D"en-us"><B><I><FONT SIZE=3D2 FACE=3D"Courier New">rivacy header =
does</FONT></I></B></SPAN><SPAN LANG=3D"en-us"><B><I> <FONT SIZE=3D2 =
FACE=3D"Courier New">not contain the value 'history' then individual H-I =
entries may have privacy associated</FONT></I></B></SPAN><SPAN =
LANG=3D"en-us"><B><I><FONT SIZE=3D2 FACE=3D"Courier New">. Those which =
do will not be forwarded but those with no privacy will be =
forwarded.</FONT></I></B></SPAN><SPAN =
LANG=3D"en-us"><B><I></I></B></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><B><I></I></B></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">That then though changes the meaning of the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Privacy header, as I understand it - i.e., the &quot;none&quot; I =
thought had to</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">do =
with specifying that an intermediary MUST NOT change the value of =
the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Privacy header. But, I guess you are saying we specify a =
slightly</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">different usage in 4244bis such that when we add it to an hi-entry, =
it</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">specifies that that hi-entry MUST NOT be dropped by an =
intermediary.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">That might work, although, we would need verbage about the =
&quot;risks&quot; of</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">doing so.</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><B><I><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Courier New"></FONT></I></B></SPAN><SPAN =
LANG=3D"en-us">&nbsp;</SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><B><I><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Courier New">[ige]</FONT></I></B></SPAN><SPAN =
LANG=3D"en-us"><B><I> <FONT COLOR=3D"#000000" SIZE=3D2 FACE=3D"Courier =
New">I would not be as strong as MUST NOT. There may be local policy =
which will override what is</FONT></I></B></SPAN><SPAN =
LANG=3D"en-us"><B><I> <FONT COLOR=3D"#000000" SIZE=3D2 FACE=3D"Courier =
New">included</FONT></I></B></SPAN><SPAN LANG=3D"en-us"><B><I><FONT =
COLOR=3D"#000000" SIZE=3D2 FACE=3D"Courier =
New">.</FONT></I></B></SPAN><SPAN LANG=3D"en-us"><B><I><FONT =
COLOR=3D"#000000" SIZE=3D2 FACE=3D"Courier New"> I would =
prefer</FONT></I></B></SPAN><SPAN LANG=3D"en-us"><B><I> <FONT =
COLOR=3D"#000000" SIZE=3D2 FACE=3D"Courier =
New">the</FONT></I></B></SPAN><SPAN LANG=3D"en-us"><B><I><FONT =
COLOR=3D"#000000" SIZE=3D2 FACE=3D"Courier New"> words NEED NOT be =
removed {Scott B will hate me for that.}</FONT></I></B></SPAN><SPAN =
LANG=3D"en-us"><B><I> <FONT COLOR=3D"#000000" SIZE=3D2 FACE=3D"Courier =
New">or perhaps MAY be forwarded based upon local =
policy.</FONT></I></B></SPAN><SPAN LANG=3D"en-us"> </SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Thanks,</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Mary. </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">-----Original Message-----</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">From: Ian Elz [<A =
HREF=3D"mailto:ian.elz@ericsson.com">mailto:ian.elz@ericsson.com</A>] =
</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Sent: Thursday, March 12, 2009 8:43 AM</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">To: Barnes, Mary (RICH2:AR00)</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Cc: sip@ietf.org; Audet, Francois (SC100:3055)</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt =
</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Mary,</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">(Written after reading your discussion with Hans =
Erik.)</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">First an apology for not making it clear in my original email that =
a</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">value of &quot;none&quot; in a Privacy header parameter in a =
History-Info entry</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">was only applicable to the uri contained in the =
hi-targeted-to-uri.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">I =
only suggested using the value &quot;none&quot; as this is a value =
defined for</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">the Privacy header which explicitly indicates that privacy is =
not</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">required. Any other value with the meaning of explicitly indicating =
that</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">there is no privacy required for the specific uri could be =
used.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">With regard to your second paragraph below I understand that the =
SIP and</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">PSTN models do not align. You are correct in your assumption that =
this</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">problem is not unique for History-Info. The problem also arises =
with the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Referred-By header. </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">I =
have raised this for History-Info at this time as your new =
draft</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">presents an opportunity to correct this problem for the H-I =
case.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">The general SIP privacy issue arises as when the Privacy header =
was</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">defined the concept of carrying third party identities in the =
SIP</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">messages was not defined. This came later with the REFER Request =
and the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">associated Referred-By header and the History-Info header. =
</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">We =
should take what opportunities arise to discuss and correct this =
if</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">there is a will to make these changes.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Ian Elz</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Office: + 44 24 764 35256</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">ian.elz@ericsson.com</FONT></SPAN></P>
<BR>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">-----Original Message-----</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">From: Mary Barnes [<A =
HREF=3D"mailto:mary.barnes@nortel.com">mailto:mary.barnes@nortel.com</A>]=
</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Sent: 11 March 2009 16:53</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">To: Ian Elz; Francois Audet</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Cc: sip@ietf.org</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt =
</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Ian,</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">This is an interesting question. I need to think about it some =
more, but</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">I =
don't like the &quot;none&quot; idea as it really must be up to the =
entity that</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">added the Privacy header to the request, as to whether it wants the =
HI</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">entries that it adds to go outside a domain. My initial thought is =
that</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">we =
can't overrule the overall Privacy header. The thing is that =
the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Privacy header doesn't preclude gathering the information and =
using</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">within a domain AND if it were to not include an hi-entry when =
the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">request leaves the domain for which the proxy is responsible, =
the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">recipient can add the hi-entry for the current request-uri before =
it</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">adds the new hi-entry.&nbsp; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Your PSTN example doesn't strictly map obviously to a SIP model and =
I</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">would think you might consider the PSTN hop to be within the same =
domain</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">for which the proxy (or gw, I guess) is responsible OR consider =
this a</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">walled garden whereby the Privacy header doesn't apply. Which =
brings me</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">to =
a more general question as to how you all deal with other =
headers</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">when you do your mapping to the PSTN when there is a Privacy =
header?&nbsp; It</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">would seem this problem isn't unique to History-Info, although I =
know</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">little about the details of PSTN I/W.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Regards,</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Mary. </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">-----Original Message-----</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">From: Ian Elz [<A =
HREF=3D"mailto:ian.elz@ericsson.com">mailto:ian.elz@ericsson.com</A>]</FO=
NT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Sent: Wednesday, March 11, 2009 4:54 AM</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">To: Barnes, Mary (RICH2:AR00); Audet, Francois =
(SC100:3055)</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Cc: sip@ietf.org</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt =
</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Mary, Francois,</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Section 4.1 of the draft allows for the addition of the Privacy =
header</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">parameter with a value of &quot;history&quot; to be included with =
the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">hi-targeted-to-uri.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Can this be extended to also allow the value =
&quot;none&quot;.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">A =
node adding a H-I entry is allowed to specify the privacy =
value</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&quot;history&quot; either in the Privacy header or as a header =
parameter</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">associated with the hi-targeted-to-uri. If the former action is =
taken by</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">one node this results in privacy for all history info header =
entries</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">even if this is not required. There is no way for privacy of =
an</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">individual H-I entry to be set to none if no Privacy is required =
for the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">uri.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">This creates a specific problem when interworking with the PSTN =
where</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">the privacy is associated with each E.164 number included in =
the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">protocol. If an INVITE is being mapped to ISUP and the H-I entries =
are</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">being used to map to redirection numbers then a single Privacy =
header</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">with the value &quot;history&quot; will result in all numbers being =
restricted.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">If =
the Privacy header parameter could include the value &quot;none&quot; =
then this</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">would explicitly indicate that the associated uri was allowed to =
be</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">presented.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">The ability to explicitly allow presentation of specific H-I =
entries may</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">also be useful in pure sip implementations.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">I =
don't believe that this change would create any backward =
compatibility</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">issues. The existing implementations will continue as deployed but =
new</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">implementations could explicitly set no privacy for a specific =
uri.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">There are no security issues other than those already defined in =
the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">draft that an intervening node could modify an existing H-I entry. =
The</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Privacy header value of &quot;none&quot; should only be included by =
the node in</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">the same manner as the existing inclusion of the =
&quot;history&quot; value.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Ian Elz</FONT></SPAN></P>
<BR>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">-----Original Message-----</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">From: i-d-announce-bounces@ietf.org</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">[<A =
HREF=3D"mailto:i-d-announce-bounces@ietf.org">mailto:i-d-announce-bounces=
@ietf.org</A>] On Behalf Of</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Internet-Drafts@ietf.org</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Sent: 04 March 2009 15:15</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">To: i-d-announce@ietf.org</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Subject: I-D Action:draft-barnes-sip-rfc4244bis-00.txt =
</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">A =
New Internet-Draft is available from the on-line =
Internet-Drafts</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">directories.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN =
LANG=3D"en-us">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Courier =
New">Title&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
An Extension to the Session Initiation</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Protocol (SIP) for Request History Information</FONT></SPAN></P>

<P DIR=3DLTR><SPAN =
LANG=3D"en-us">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Courier New">Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : M. =
Barnes, F. Audet</FONT></SPAN></P>

<P DIR=3DLTR><SPAN =
LANG=3D"en-us">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Courier New">Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
: draft-barnes-sip-rfc4244bis-00.txt</FONT></SPAN></P>

<P DIR=3DLTR><SPAN =
LANG=3D"en-us">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Courier =
New">Pages&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
49</FONT></SPAN></P>

<P DIR=3DLTR><SPAN =
LANG=3D"en-us">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Courier =
New">Date&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; : 2009-03-04</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">This document defines a standard mechanism for capturing the =
history</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">information associated with a Session Initiation Protocol (SIP) =
request.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">This capability enables many enhanced services by providing =
the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">information as to how and why a call arrives at a specific =
application</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">or =
user.&nbsp; This document defines a new optional SIP header, =
History-Info,</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">for capturing the history information in =
requests.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">A =
URL for this Internet-Draft is:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New"><A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-barnes-sip-rfc4244bis-0=
0.txt">http://www.ietf.org/internet-drafts/draft-barnes-sip-rfc4244bis-00=
.txt</A></FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Internet-Drafts are also available by anonymous FTP =
at:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New"><A =
HREF=3D"ftp://ftp.ietf.org/internet-drafts/">ftp://ftp.ietf.org/internet-=
drafts/</A></FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Below is the data which will enable a MIME compliant mail =
reader</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">implementation to automatically retrieve the ASCII version of =
the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Internet-Draft.</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

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From: "Francois Audet" <audet@nortel.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>
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thread-topic: [Sip] Terminology (was RE: Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Cc: sip@ietf.org, "Elwell, John" <john.elwell@siemens.com>
Subject: Re: [Sip] Terminology (was RE: Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Does p-called-header include URI parameters?


On Mar 12, 2009, at 7:07, "Christer Holmberg" <christer.holmberg@ericsson.com 
 > wrote:

>
> Hi,
>
> IF we would remove that requierment, and ONLY care about the case  
> when the R-URI is replaced with the CONTACT, I don't think we need  
> to do anything. I believe P-Called-Party-ID already solves that.
>
> Regards,
>
> Christer
>
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf  
> Of Elwell, John
> Sent: 12. maaliskuuta 2009 15:31
> To: Hans Erik van Elburg
> Cc: sip@ietf.org
> Subject: Re: [Sip] Terminology (was RE: Fwd: I-DACTION:draft- 
> rosenberg-sip-target-uri-delivery-01.txt
>
> Hans Erik,
>
> The way I interpreted the 01 draft, you would not get the freephone  
> number anyway. If that is a requirement, then I accept my suggestion  
> won't suffice.
>
> John
>
> From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
> Sent: 12 March 2009 12:57
> To: Elwell, John
> Cc: sip@ietf.org
> Subject: Re: [Sip] Terminology (was RE: Fwd: I-DACTION:draft- 
> rosenberg-sip-target-uri-delivery-01.txt
>
> But that does not solve the freephone service use case. Freephone  
> represents a use case where the R-URI is replaced with an AoR of  
> user B - not with the contact of user B.
>
> A calls Bfreephonenumber routed the B (AOR) routed to B's contact.
>
> The target-uri draft is interested in obtaining: Bfreephonenumber as  
> that is what A used to reach B.
>
> So what we like to see that B UA receives is something like
>     H-I:  Bfreephone; "istarget",\
>         B,\
>         contact-B
>
> So you want to get Bfreephonenumber and not B.
>
> In certain situations you are also interested in B, I guess that   
> PBX's would interested in that to deliver to the correct user. We  
> could distinguish them like:
>     H-I:  Bfreephone; "istarget",\
>         B,"aor"\
>         contact-B
>
> BR,
> /Hans Erik van Elburg
>
>
> On Thu, Mar 12, 2009 at 12:08 PM, Elwell, John <john.elwell@siemens.com 
> > wrote:
> If 'istarget' is used only to identify the value that was  
> overwritten in
> the Request-URI by a contact URI, an alternative approach would be to
> flag the contact URI H-I entry as 'iscontact'. Then the UAS would just
> need to look for the most recent H-I entry that is not marked
> 'iscontact'.
>
> John
>
>
> ________________________________
>
>        From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> Behalf Of Hans Erik van Elburg
>        Sent: 12 March 2009 08:35
>        To: Mary Barnes
>        Cc: sip@ietf.org
>        Subject: Re: [Sip] Terminology (was RE: Fwd:
> I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>
>
>        Yes, because you are using the 3261 use of target and the
> 4244bis introduced definition of        retarget. I thought it was  
> clear that
> we need other words as those definitions don't match the target-uri
> drafts use of the terms. Also they do not suffice to provide a  
> solution
> for the use cases in the target-uri draft.
>
>        The 3261 text you refer to is exactly about the case where the
> home proxy overwrites the Request-URI with a new target. This target  
> is
> teh registered contact address. And hence this would be what target- 
> uri
> calls a hop or a route. This case and this is where it gets  
> confusing is
> not a "retarget" in the target-uri draft use of the term.
>
>        The target-uri draft states:
>
>           "To avoid confusion, we
>           refer to a SIP URI that is an address for a user or resource
> as a
>           "target" and a SIP URI that is a hop for reaching that user
> as a
>
>           "hop".
>
>
>        Apparently that does not suffice to avoid confusion.
>
>        As for the tagging, speaking about the solution before agreeing
> on the terminology and the problem it should solve is meaningless.
>
>        /Hans Erik van Elburg
>
>
>
>        On Thu, Mar 12, 2009 at 2:33 AM, Mary Barnes
> <mary.barnes@nortel.com> wrote:
>
>
>                Responses below [MB].
>
>
>                -----Original Message-----
>                From: Hans Erik van Elburg
> [mailto:ietf.hanserik@gmail.com]
>
>                Sent: Wednesday, March 11, 2009 6:42 PM
>                To: Barnes, Mary (RICH2:AR00)
>                Cc: Shida Schubert; sip@ietf.org
>                Subject: Re: Terminology (was RE: [Sip] Fwd: I-D
>                ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>
>
>                I was talking about the concept.
>
>                The use cases only describe  cases where the target-uri
> (lets call that
>                current target from now) has been lost when an initial
> request for a
>                dialog or standalone request arrives at the UAS.
>
>                [MB] And I think this is where the terminology  
> confusion
> starts - I
>                don't think of the "lost" target-uri as being the
> "current" target. In
>                my mind, the "current" target is reflected by the last
> hi-entry and the
>                request-uri in the incoming request at the UAS.  
> If        the
> entity that sent
>                the request was the entity that added the last hi- 
> entry,
> then the uri in
>                that hi-entry is the same as the request-URI in  
> the        SIP
> request that
>                arrives at the UAS. I refer to the "lost" uri as the  
> one
> that was
>                "retargeted" - that's the one the UAS wants to pull  
> from
> the hi-entries
>                in the incoming request. That hi-entry was not the one
> that was just
>                added by the entity that built the request just  
> received
> by the UAS.
>                That hi-entry is tagged with whatever name we are going
> to tag it with
>                BEFORE the request is forwarded (using the term forward
> per section 16.6
>                of RFC 3261).  That tag is added once the target  
> set        of
> potential
>                candidates for the new request uri are determined in
> section 16.5 of
>                3261 (with "target set" being a 3261 term), just before
> the request is
>                forwarded in section 16.6 to one of those targets.  A
> new hi-entry
>                (which will be the last hi-entry in the request  
> received
> by the UAS) is
>                added in section 16.6 of 3261 as the request is
> forwarded.  At this
>                point in time, the lost information is in the previous
> hi-entry when the
>                outgoing request is sent. [/MB]
>
>
>                What has been lost is the current target of the  
> request:
>
>                [MB] Right - at which point in my mind, it's no longer
> the current
>                target ;) Maybe we call it "lost". [/MB]
>
>
>
>                     Current target
>
>                  The current target of an initial request for a dialog
> or standalone
>                  request is the name or address to which the request  
> is
> targeted, i.e.
>                  either the initial target inserted in the Request-URI
> by the UAC that
>                  originates the request, or when a retarget occurred,
> the target
>                  provided in that retarget operation.  Reroute and
> translation
>                  operations never change the current target.
>
>
>                [MB] I don't think this        definition fits what  
> you want -
> i.e., if there
>                is no retargeting, then none of the hi-entries are
> tagged - i.e., you
>                won't have your concept of current. The way it works is
> that if no
>                hi-entries are tagged, then you know that there was no
> retargeting, thus
>                you know the request-uri has not been lost. [/MB]
>
>
>                This defintion only makes sense when the following
> definitions are used:
>
>                Name:
>
>                  A name is a moniker for an entity which refers to it
> in a way which
>
>                  reveals nothing about where it is in a network.  In
> SIP, tel URI
>
>                  which doesn't represent the location of the entity is
> a name.
>
>
>                Address:
>
>                  An address is an identifier for an entity which
> describes it by its
>
>                  location on the network.  In SIP, the SIP URI itself
> is a form of
>
>                  address because the host part of the URI, the only
> mandatory part of
>
>                  the URI besides the scheme itself, indicates the
> location of a SIP
>
>                  server that can be used to handle the request.
>
>                Route:
>
>                  Finally, a route is a   sequence of SIP entities
> (including the UA
>                itself!) which are
>
>                 traversed in order to        forward a request to an  
> address
> or name.
>
>                Retarget (other term might be needed, as this is highly
> confusing):
>
>                   A Request-URI rewrite operation that changes the
> target identity of
>                the request.
>
>                Reroute (other term might be needed):
>
>                   A Request-URI rewrite operation that does not change
> the current
>                target of the request, but     determines the route/ 
> next
> hop taken to
>                reach the target-identity.
>
>                /Hans Erik
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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<html><body bgcolor="#FFFFFF"><div>Does p-called-header include URI parameters?<br><br></div><div><br>On Mar 12, 2009, at 7:07, "Christer Holmberg" &lt;<a href="mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson.com</a>> wrote:<br><br></div><div></div><blockquote type="cite"><div>

<div dir="ltr" align="left">&nbsp;</div>
<div dir="ltr" align="left">
<div dir="ltr" align="left"><span class="034104813-12032009">Hi,</span></div>
<div dir="ltr" align="left"><span class="034104813-12032009"></span>&nbsp;</div>
<div dir="ltr" align="left"><span class="034104813-12032009">IF we would remove that 
requierment, and&nbsp;ONLY care about the case when the R-URI is replaced with 
the CONTACT, I don't think we need to do anything. I believe P-Called-Party-ID 
already solves that.</span></div>
<div dir="ltr" align="left"><span class="034104813-12032009"></span>&nbsp;</div>
<div dir="ltr" align="left"><span class="034104813-12032009">Regards,</span></div>
<div dir="ltr" align="left"><span class="034104813-12032009"></span>&nbsp;</div>
<div dir="ltr" align="left"><span class="034104813-12032009">Christer</span></div></div><br>
<blockquote dir="ltr" style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
  <div class="OutlookMessageHeader" lang="en-us" dir="ltr" align="left">
  <hr tabindex="-1">
  <font face="Tahoma" size="2"><b>From:</b> <a href="mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a> 
  [<a href="mailto:sip-bounces@ietf.org">mailto:sip-bounces@ietf.org</a>] <b>On Behalf Of </b>Elwell, John<br><b>Sent:</b> 
  12. maaliskuuta 2009 15:31<br><b>To:</b> Hans Erik van Elburg<br><b>Cc:</b> 
  <a href="mailto:sip@ietf.org"><a href="mailto:sip@ietf.org">sip@ietf.org</a></a><br><b>Subject:</b> Re: [Sip] Terminology (was RE: Fwd: 
  I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br></font><br></div>
  <div></div>
  <div dir="ltr" align="left">
  <div dir="ltr" align="left"><span class="441300613-12032009"><font face="Arial" color="#000080">Hans Erik,</font></span></div>
  <div dir="ltr" align="left"><span class="441300613-12032009"><font face="Arial" color="#000080"></font></span>&nbsp;</div>
  <div dir="ltr" align="left"><span class="441300613-12032009"><font face="Arial"><font color="#000080">The way I interpreted the 01 draft, you would not get the 
  freephone number anyway. If that is a requirement, then I accept my suggestion 
  won't suffice<span class="125123013-12032009">.</span></font></font></span></div>
  <div dir="ltr" align="left"><span class="441300613-12032009"><font face="Arial" color="#000080"></font></span>&nbsp;</div>
  <div dir="ltr" align="left"><span class="441300613-12032009"><font face="Arial" color="#000080">John</font></span></div></div><br>
  <blockquote style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000080 2px solid; MARGIN-RIGHT: 0px">
    <div class="OutlookMessageHeader" lang="en-us" dir="ltr" align="left">
    <hr tabindex="-1">
    <font face="Tahoma" size="2"><b>From:</b> Hans Erik van Elburg 
    [<a href="mailto:ietf.hanserik@gmail.com">mailto:ietf.hanserik@gmail.com</a>] <br><b>Sent:</b> 12 March 2009 
    12:57<br><b>To:</b> Elwell, John<br><b>Cc:</b> 
    <a href="mailto:sip@ietf.org"><a href="mailto:sip@ietf.org">sip@ietf.org</a></a><br><b>Subject:</b> Re: [Sip] Terminology (was RE: Fwd: 
    I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br></font><br></div>
    <div></div>But that does not solve the freephone service use case. 
    
    Freephone represents a use case where the R-URI is replaced with an AoR of 
    user B - not with the contact of user B. <br><br>A calls Bfreephonenumber 
    routed the B (AOR) routed to B's contact.<br><br>The target-uri draft is 
    interested in obtaining: Bfreephonenumber as that is what A used to reach 
    B.<br><br>So what we like to see that B UA receives is something 
    like<br>&nbsp;&nbsp;&nbsp; H-I:&nbsp; Bfreephone; 
    "istarget",\<br>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 
    B,\<br>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; contact-B<br><br>So you want to 
    get Bfreephonenumber and not B.<br><br>In certain situations you are also 
    interested in B, I guess that&nbsp; PBX's would interested in that to 
    deliver to the correct user. We could distinguish them 
    like:<br>&nbsp;&nbsp;&nbsp; H-I:&nbsp; Bfreephone; 
    "istarget",\<br>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 
    B,"aor"\<br>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; contact-B<br><br>BR,<br clear="all">/Hans Erik van Elburg<br><br><br>
    <div class="gmail_quote">On Thu, Mar 12, 2009 at 12:08 PM, Elwell, John <span dir="ltr">&lt;<a href="mailto:john.elwell@siemens.com"><a href="mailto:john.elwell@siemens.com">john.elwell@siemens.com</a></a>></span> 
    wrote:<br>
    <blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">If 
      'istarget' is used only to identify the value that was overwritten 
      in<br>the Request-URI by a contact URI, an alternative approach would be 
      to<br>flag the contact URI H-I entry as 'iscontact'. Then the UAS would 
      just<br>need to look for the most recent H-I entry that is not 
      marked<br>'iscontact'.<br><br>John<br><br><br>________________________________<br><br>&nbsp; 
      &nbsp; &nbsp; &nbsp;From: <a href="mailto:sip-bounces@ietf.org"><a href="mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a></a> [mailto:<a href="mailto:sip-bounces@ietf.org"><a href="mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a></a>] On<br>Behalf 
      Of Hans Erik van Elburg<br>&nbsp; &nbsp; &nbsp; &nbsp;Sent: 12 March 2009 
      08:35<br>&nbsp; &nbsp; &nbsp; &nbsp;To: Mary Barnes<br>&nbsp; &nbsp; 
      &nbsp; &nbsp;Cc: <a href="mailto:sip@ietf.org"><a href="mailto:sip@ietf.org">sip@ietf.org</a></a><br>&nbsp; 
      &nbsp; &nbsp; &nbsp;Subject: Re: [Sip] Terminology (was RE: 
      Fwd:<br>I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br>
      <div>
      <div></div>
      <div class="h5"><br><br>&nbsp; &nbsp; &nbsp; &nbsp;Yes, because you are 
      using the 3261 use of target and the<br>4244bis introduced definition of 
      retarget. I thought it was clear that<br>we need other words as those 
      definitions don't match the target-uri<br>drafts use of the terms. Also 
      they do not suffice to provide a solution<br>for the use cases in the 
      target-uri draft.<br><br>&nbsp; &nbsp; &nbsp; &nbsp;The 3261 text you 
      refer to is exactly about the case where the<br>home proxy overwrites the 
      Request-URI with a new target. This target is<br>teh registered contact 
      address. And hence this would be what target-uri<br>calls a hop or a 
      route. This case and this is where it gets confusing is<br>not a 
      "retarget" in the target-uri draft use of the term.<br><br>&nbsp; &nbsp; 
      &nbsp; &nbsp;The target-uri draft states:<br><br>&nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp; "To avoid confusion, we<br>&nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; refer to a SIP URI that is an address for a user or resource<br>as 
      a<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; "target" and a SIP URI that is a 
      hop for reaching that user<br>as a<br><br>&nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; "hop".<br><br><br>&nbsp; &nbsp; &nbsp; &nbsp;Apparently that does 
      not suffice to avoid confusion.<br><br>&nbsp; &nbsp; &nbsp; &nbsp;As for 
      the tagging, speaking about the solution before agreeing<br>on the 
      terminology and the problem it should solve is meaningless.<br><br>&nbsp; 
      &nbsp; &nbsp; &nbsp;/Hans Erik van Elburg<br><br><br><br>&nbsp; &nbsp; 
      &nbsp; &nbsp;On Thu, Mar 12, 2009 at 2:33 AM, Mary Barnes<br>&lt;<a href="mailto:mary.barnes@nortel.com"><a href="mailto:mary.barnes@nortel.com">mary.barnes@nortel.com</a></a>> 
      wrote:<br><br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp;Responses below [MB].<br><br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp;-----Original Message-----<br>&nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;From: Hans Erik van Elburg<br>[mailto:<a href="mailto:ietf.hanserik@gmail.com"><a href="mailto:ietf.hanserik@gmail.com">ietf.hanserik@gmail.com</a></a>]<br><br>&nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Sent: Wednesday, March 11, 
      2009 6:42 PM<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;To: 
      Barnes, Mary (RICH2:AR00)<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp;Cc: Shida Schubert; <a href="mailto:sip@ietf.org"><a href="mailto:sip@ietf.org">sip@ietf.org</a></a><br>&nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp;Subject: Re: Terminology (was RE: [Sip] Fwd: 
      I-D<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp;ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br><br><br>&nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;I was talking about the 
      concept.<br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;The 
      use cases only describe &nbsp;cases where the target-uri<br>(lets call 
      that<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;current 
      target from now) has been lost when an initial<br>request for a<br>&nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;dialog or standalone 
      request arrives at the UAS.<br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp;[MB] And I think this is where the terminology 
      confusion<br>starts - I<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp;don't think of the "lost" target-uri as being 
      the<br>"current" target. In<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp;my mind, the "current" target is reflected by the 
      last<br>hi-entry and the<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp;request-uri in the incoming request at the UAS. If 
      the<br>entity that sent<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp;the request was the entity that added the last 
      hi-entry,<br>then the uri in<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp;that hi-entry is the same as the request-URI in the 
      SIP<br>request that<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp;arrives at the UAS. I refer to the "lost" uri as the one<br>that 
      was<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;"retargeted" 
      - that's the one the UAS wants to pull from<br>the hi-entries<br>&nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;in the incoming request. 
      That hi-entry was not the one<br>that was just<br>&nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;added by the entity that built the 
      request just received<br>by the UAS.<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp;That hi-entry is tagged with whatever name we are 
      going<br>to tag it with<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp;BEFORE the request is forwarded (using the term 
      forward<br>per section 16.6<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp;of RFC 3261). &nbsp;That tag is added once the target set 
      of<br>potential<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp;candidates for the new request uri are determined in<br>section 16.5 
      of<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;3261 (with 
      "target set" being a 3261 term), just before<br>the request is<br>&nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;forwarded in section 16.6 
      to one of those targets. &nbsp;A<br>new hi-entry<br>&nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;(which will be the last hi-entry in the 
      request received<br>by the UAS) is<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp;added in section 16.6 of 3261 as the request 
      is<br>forwarded. &nbsp;At this<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp;point in time, the lost information is in the 
      previous<br>hi-entry when the<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp;outgoing request is sent. [/MB]<br><br><br>&nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;What has been lost is the current 
      target of the request:<br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp;[MB] Right - at which point in my mind, it's no longer<br>the 
      current<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;target 
      ;) Maybe we call it "lost". [/MB]<br><br><br><br>&nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Current 
      target<br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp;The current target of an initial request for a dialog<br>or 
      standalone<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp;request is the name or address to which the request is<br>targeted, 
      i.e.<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp;either the initial target inserted in the Request-URI<br>by the UAC 
      that<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp;originates the request, or when a retarget occurred,<br>the 
      target<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp;provided in that retarget operation. &nbsp;Reroute 
      and<br>translation<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp;operations never change the current target.<br><br><br>&nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;[MB] I don't think this 
      definition fits what you want -<br>i.e., if there<br>&nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;is no retargeting, then none of the 
      hi-entries are<br>tagged - i.e., you<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp;won't have your concept of current. The way it works 
      is<br>that if no<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp;hi-entries are tagged, then you know that there was 
      no<br>retargeting, thus<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp;you know the request-uri has not been lost. 
      [/MB]<br><br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp;This defintion only makes sense when the following<br>definitions 
      are used:<br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp;Name:<br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp;A name is a moniker for an entity which refers to it<br>in a way 
      which<br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp;reveals nothing about where it is in a network. &nbsp;In<br>SIP, tel 
      URI<br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp;which doesn't represent the location of the entity is<br>a 
      name.<br><br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp;Address:<br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp;An address is an identifier for an entity which<br>describes 
      it by its<br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp;location on the network. &nbsp;In SIP, the SIP URI itself<br>is a 
      form of<br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp;address because the host part of the URI, the only<br>mandatory part 
      of<br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp;the URI besides the scheme itself, indicates the<br>location of a 
      SIP<br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp;server that can be used to handle the request.<br><br>&nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Route:<br><br>&nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Finally, a route is a 
      &nbsp; sequence of SIP entities<br>(including the UA<br>&nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;itself!) which are<br><br>&nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; traversed in order to 
      forward a request to an address<br>or name.<br><br>&nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Retarget (other term might be needed, as 
      this is highly<br>confusing):<br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; A Request-URI rewrite operation that changes 
      the<br>target identity of<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp;the request.<br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp;Reroute (other term might be needed):<br><br>&nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; A Request-URI rewrite 
      operation that does not change<br>the current<br>&nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;target of the request, but &nbsp; &nbsp; 
      determines the route/next<br>hop taken to<br>&nbsp; &nbsp; &nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp;reach the target-identity.<br><br>&nbsp; &nbsp; 
      &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;/Hans 
    Erik<br><br></div></div></blockquote></div></blockquote></blockquote>
</div></blockquote><blockquote type="cite"><div><span>_______________________________________________</span><br><span>Sip mailing list &nbsp;<a href="https://www.ietf.org/mailman/listinfo/sip"><a href="https://www.ietf.org/mailman/listinfo/sip">https://www.ietf.org/mailman/listinfo/sip</a></a></span><br><span>This list is for NEW development of the core SIP Protocol</span><br><span>Use <a href="mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.columbia.edu</a> for questions on current sip</span><br><span>Use <a href="mailto:sipping@ietf.org">sipping@ietf.org</a> for new developments on the application of sip</span></div></blockquote></body></html>
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Hans Erik van Elburg" <ietf.hanserik@gmail.com>
Cc: sip@ietf.org, "Elwell, John" <john.elwell@siemens.com>
Subject: Re: [Sip] Terminology (was RE: Fwd: I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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When I use "retarget" I am using the definition that I put into 4244bis.
So, if you could ignore the tag or change it to 'iscontact' or whatever,
what would you change in that definition? Also, consider that if you
were to change everyplace that I use
"retarget"/"retargeted"/"retargeting" in 4244bis with 'iscontact' the
text would not make sense. So, even if we change the name of the tag, we
still need to understand the concept or define another term that means
that the request-uri in the incoming request (which should be the same
as the last hi-entry in the request) is being chnaged due to service
lookup, configuration, blah, blah, blah and that to meet the
requirements in the target-uri document, we want to tag this entry so
that this request-uri is not "lost" - i.e., this is the one the services
identified in the target-uri document care about.=20
=20
So, right now, here's what's in 4244bis:
=20
   The term "retarget" is used in this document to refer to the process
   of a Proxy Server/User Agent Client (UAC) changing a Uniform Resource
   Identifier (URI) in a request based on a lookup in a location service
   and thus changing the target of the request.
=20
We have previously discussed that we do need to expand the "based on a
lookup in a location service" to include configuration, etc., so let's
not debate that.=20
=20
But, what explicitly is the concern with that definition?
=20
Mary.=20
=20

________________________________

From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
Sent: Thursday, March 12, 2009 9:55 AM
To: Barnes, Mary (RICH2:AR00)
Cc: Elwell, John; sip@ietf.org
Subject: Re: [Sip] Terminology (was RE: Fwd:
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


I'm fine with discussing implementation if everyone is meaning the same
if they use the same words.

As  to the "retarget" term such does not seem to be the case.

/Hans Erik van Elburg



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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.3492" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D421131315-12032009>When I=20
use "retarget" I am using the definition that I put into 4244bis.&nbsp; =
So, if=20
you could ignore the tag or change it to 'iscontact' or whatever, what =
would you=20
change in that definition? Also, consider that if you were to change =
everyplace=20
that I use "retarget"/"retargeted"/"retargeting" in 4244bis with =
'iscontact' the=20
text would not make sense. So, even if we change the name of the tag, we =
still=20
need to understand the concept or define another term that means that =
the=20
request-uri in the incoming request (which should be the same as the =
last=20
hi-entry in the request) is being chnaged due to service lookup, =
configuration,=20
blah, blah, blah and that to meet the requirements in the target-uri =
document,=20
we want to tag this entry so that this request-uri is not "lost" - i.e., =
this is=20
the one the services identified in the target-uri document care about.=20
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D421131315-12032009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D421131315-12032009>So,=20
right now, here's what's in 4244bis:</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D421131315-12032009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D421131315-12032009>&nbsp;&nbsp; The term "retarget" is used in =
this=20
document to refer to the process<BR>&nbsp;&nbsp; of a Proxy Server/User =
Agent=20
Client (UAC) changing a Uniform Resource<BR>&nbsp;&nbsp; Identifier =
(URI) in a=20
request based on a lookup in a location service</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D421131315-12032009>&nbsp;&nbsp; and thus changing the target of =
the=20
request.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D421131315-12032009>We=20
have previously discussed that we do need to expand the "based on a =
lookup in a=20
location service" to include configuration, etc., so let's not debate =
that.=20
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D421131315-12032009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D421131315-12032009>But,=20
what explicitly is the concern with that definition?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D421131315-12032009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D421131315-12032009>Mary.=20
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><BR></DIV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Hans Erik van Elburg=20
[mailto:ietf.hanserik@gmail.com] <BR><B>Sent:</B> Thursday, March 12, =
2009 9:55=20
AM<BR><B>To:</B> Barnes, Mary (RICH2:AR00)<BR><B>Cc:</B> Elwell, John;=20
sip@ietf.org<BR><B>Subject:</B> Re: [Sip] Terminology (was RE: Fwd:=20
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR></FONT><BR></=
DIV>
<DIV></DIV>I'm fine with discussing implementation if everyone is =
meaning the=20
same if they use the same words.<BR><BR>As&nbsp; to the "retarget" term =
such=20
does not seem to be the case.<BR><BR clear=3Dall>/Hans Erik van=20
Elburg<BR><BR></BODY></HTML>

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From mary.barnes@nortel.com  Thu Mar 12 08:49:57 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Ian Elz" <ian.elz@ericsson.com>
Cc: sip@ietf.org, Francois Audet <audet@nortel.com>
Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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Ian,
=20
I do see your point now - the text does need clarification.  Detailed
comments are below [MB].
=20
Thanks,
Mary.=20

________________________________

From: Ian Elz [mailto:ian.elz@ericsson.com]=20
Sent: Thursday, March 12, 2009 10:12 AM
To: Barnes, Mary (RICH2:AR00)
Cc: sip@ietf.org; Audet, Francois (SC100:3055)
Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20



Mary,

I was only considering the specific domain case as included in paragraph
2 of section 3.3 in the draft:

                                                  Thus, local policy

   MAY also be used to determine whether to include the History-Info

   header at all, whether to capture a specific Request-URI in the

   header, or whether it be included only in the Request as it is

   retargeted within a specific domain (PRIV-req-3).  In the latter

   case, this is accomplished by adding a new priv-value, history, to

   the Privacy header [RFC3323] indicating whether any or a specific

   History-Info header(s) SHOULD be forwarded.=20

[MB] Yes, this is a little fuzzy. It is trying to make a very general
statement that we are adding this new "priv-value" to the  Privacy
header so that we can control whether the entries are sent outside the
proxy.  I think that sentence is okay if we remove the normative
(SHOULD) and just add a note that the details of the use of the privacy
header are provided in section 4.1., something like the following:=20
   In the latter

   case, this is accomplished by adding a new priv-value, history, to

   the Privacy header [RFC3323] indicating whether or not any or a
specific

   History-Info header(s) are forwarded outside a specific domain for
which the proxy is responsible.=20
=20
   The details as to the use of the new priv-value with the Privacy
header are provided in section 4.1.

 [/MB]

=20

    o  Privacy: An optional parameter for History-Info, reflected in the

      History-Info header field values by including the Privacy Header

      [RFC3323] with a priv-value of "history" escaped in the hi-

      targeted-to-uri or by adding the Privacy header with a priv-value

      of "history" to the Request.  The use of the Privacy Header with a

      priv-value of "history" indicates whether a specific or all

      History-Info headers should not be forwarded.

(There appears to be an inconsistency here as sections 3.3 and 4.1
appear to contradict. [Perhaps it just needs clarification that section
3.3 allows forwarding within the domain and 4.1 prohibits forwarding
outside the domain?])=20

[MB] Yes, this is also fuzzy. It should really read (and the SHOULD NOT
ought to be caps):

 Privacy: An optional parameter for History-Info, reflected in the

      History-Info header field values by including the Privacy Header

      [RFC3323] with a priv-value of "history" escaped in the hi-

      targeted-to-uri or by adding the Privacy header with a priv-value

      of "history" to the Request.   The latter case indicates that =20

      none of the History-Info headers should be forwarded , whereas the
use of

      the Privacy header escaped in the hi-targeted-to-uri means that a
specific hi-entry SHOULD NOT be forwarded.=20
[/MB]

Assuming that the value 'history' means that presentation and forwarding
should not occur.

In this case the inclusion of the value 'history' in the Privacy header
impacts all H-I entries. If one node includes the value 'history' in the
Privacy header then there is no way for another node to specifically
allow the uri included in the hi-targeted-to-uri to not be restricted.
To allow forwarding of a specific H-I entry there needs to be another
value which can be included in the Privacy header parameter of for the
hi-targeted-to-uri.

[MB] Exactly - that's why I had suggested earlier that we perhaps
RECOMMEND that the Privacy header only be used in the escaped form
associated with a specific hi-entry. There is text elsewhere explaining
that you do have potential loss of functionality when you use the
Privacy header. Perhaps, we need a stronger warning. [/MB]

If the value "none" has specific meanings which will cause confusion
then another value should be used. =20

More comments in-line

Nit in A.3

In both the sequence and the text you indicate that the INVITE to UA-B
will retransmit. This should not occur if a 100 Trying has been received
from UA-B. This means that the retransmit timer cannot be used to
determine if UA-B does not respond. There will be a separate timer
specified for determining no answer after the 180 has been received.

[MB] You are correct - that flow is buggy and there are bugs in the
flows and we will rework these and/or remove some of them.[/MB]

Ian Elz

Office: + 44 24 764 35256

ian.elz@ericsson.com

-----Original Message-----
From: Mary Barnes [mailto:mary.barnes@nortel.com]
Sent: 12 March 2009 13:55
To: Ian Elz
Cc: sip@ietf.org; Francois Audet
Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

Ian,

Okay, so I think I might understand now - you want to use the "none" to

override the Privacy header that would be (in current 4244/4244bis)

applied to the whole message as it traverses across the n/w from the UAC

to the UAS. Is that correct?=20

=20

[ige] Yes - currently a Privacy header with value 'history' will prevent
any H-I entries from being passed. If the Privacy header does not
contain the value 'history' then individual H-I entries may have privacy
associated. Those which do will not be forwarded but those with no
privacy will be forwarded.

That then though changes the meaning of the

Privacy header, as I understand it - i.e., the "none" I thought had to

do with specifying that an intermediary MUST NOT change the value of the

Privacy header. But, I guess you are saying we specify a slightly

different usage in 4244bis such that when we add it to an hi-entry, it

specifies that that hi-entry MUST NOT be dropped by an intermediary.

That might work, although, we would need verbage about the "risks" of

doing so.

=20

[ige] I would not be as strong as MUST NOT. There may be local policy
which will override what is included. I would prefer the words NEED NOT
be removed {Scott B will hate me for that.} or perhaps MAY be forwarded
based upon local policy.=20

Thanks,

Mary.=20

-----Original Message-----

From: Ian Elz [mailto:ian.elz@ericsson.com]=20

Sent: Thursday, March 12, 2009 8:43 AM

To: Barnes, Mary (RICH2:AR00)

Cc: sip@ietf.org; Audet, Francois (SC100:3055)

Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

Mary,

(Written after reading your discussion with Hans Erik.)

First an apology for not making it clear in my original email that a

value of "none" in a Privacy header parameter in a History-Info entry

was only applicable to the uri contained in the hi-targeted-to-uri.

I only suggested using the value "none" as this is a value defined for

the Privacy header which explicitly indicates that privacy is not

required. Any other value with the meaning of explicitly indicating that

there is no privacy required for the specific uri could be used.

With regard to your second paragraph below I understand that the SIP and

PSTN models do not align. You are correct in your assumption that this

problem is not unique for History-Info. The problem also arises with the

Referred-By header.=20

I have raised this for History-Info at this time as your new draft

presents an opportunity to correct this problem for the H-I case.

The general SIP privacy issue arises as when the Privacy header was

defined the concept of carrying third party identities in the SIP

messages was not defined. This came later with the REFER Request and the

associated Referred-By header and the History-Info header.=20

We should take what opportunities arise to discuss and correct this if

there is a will to make these changes.

Ian Elz

Office: + 44 24 764 35256

ian.elz@ericsson.com


-----Original Message-----

From: Mary Barnes [mailto:mary.barnes@nortel.com]

Sent: 11 March 2009 16:53

To: Ian Elz; Francois Audet

Cc: sip@ietf.org

Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

Ian,

This is an interesting question. I need to think about it some more, but

I don't like the "none" idea as it really must be up to the entity that

added the Privacy header to the request, as to whether it wants the HI

entries that it adds to go outside a domain. My initial thought is that

we can't overrule the overall Privacy header. The thing is that the

Privacy header doesn't preclude gathering the information and using

within a domain AND if it were to not include an hi-entry when the

request leaves the domain for which the proxy is responsible, the

recipient can add the hi-entry for the current request-uri before it

adds the new hi-entry. =20

Your PSTN example doesn't strictly map obviously to a SIP model and I

would think you might consider the PSTN hop to be within the same domain

for which the proxy (or gw, I guess) is responsible OR consider this a

walled garden whereby the Privacy header doesn't apply. Which brings me

to a more general question as to how you all deal with other headers

when you do your mapping to the PSTN when there is a Privacy header?  It

would seem this problem isn't unique to History-Info, although I know

little about the details of PSTN I/W.

Regards,

Mary.=20

-----Original Message-----

From: Ian Elz [mailto:ian.elz@ericsson.com]

Sent: Wednesday, March 11, 2009 4:54 AM

To: Barnes, Mary (RICH2:AR00); Audet, Francois (SC100:3055)

Cc: sip@ietf.org

Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

Mary, Francois,

Section 4.1 of the draft allows for the addition of the Privacy header

parameter with a value of "history" to be included with the

hi-targeted-to-uri.

Can this be extended to also allow the value "none".

A node adding a H-I entry is allowed to specify the privacy value

"history" either in the Privacy header or as a header parameter

associated with the hi-targeted-to-uri. If the former action is taken by

one node this results in privacy for all history info header entries

even if this is not required. There is no way for privacy of an

individual H-I entry to be set to none if no Privacy is required for the

uri.

This creates a specific problem when interworking with the PSTN where

the privacy is associated with each E.164 number included in the

protocol. If an INVITE is being mapped to ISUP and the H-I entries are

being used to map to redirection numbers then a single Privacy header

with the value "history" will result in all numbers being restricted.

If the Privacy header parameter could include the value "none" then this

would explicitly indicate that the associated uri was allowed to be

presented.

The ability to explicitly allow presentation of specific H-I entries may

also be useful in pure sip implementations.

I don't believe that this change would create any backward compatibility

issues. The existing implementations will continue as deployed but new

implementations could explicitly set no privacy for a specific uri.

There are no security issues other than those already defined in the

draft that an intervening node could modify an existing H-I entry. The

Privacy header value of "none" should only be included by the node in

the same manner as the existing inclusion of the "history" value.

Ian Elz


-----Original Message-----

From: i-d-announce-bounces@ietf.org

[mailto:i-d-announce-bounces@ietf.org] On Behalf Of

Internet-Drafts@ietf.org

Sent: 04 March 2009 15:15

To: i-d-announce@ietf.org

Subject: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

A New Internet-Draft is available from the on-line Internet-Drafts

directories.

        Title           : An Extension to the Session Initiation

Protocol (SIP) for Request History Information

        Author(s)       : M. Barnes, F. Audet

        Filename        : draft-barnes-sip-rfc4244bis-00.txt

        Pages           : 49

        Date            : 2009-03-04

This document defines a standard mechanism for capturing the history

information associated with a Session Initiation Protocol (SIP) request.

This capability enables many enhanced services by providing the

information as to how and why a call arrives at a specific application

or user.  This document defines a new optional SIP header, History-Info,

for capturing the history information in requests.

A URL for this Internet-Draft is:

http://www.ietf.org/internet-drafts/draft-barnes-sip-rfc4244bis-00.txt

Internet-Drafts are also available by anonymous FTP at:

ftp://ftp.ietf.org/internet-drafts/

Below is the data which will enable a MIME compliant mail reader

implementation to automatically retrieve the ASCII version of the

Internet-Draft.


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Thread-Topic: [Sip] Terminology (was RE: Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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References: <899F1518-6872-442E-AD3E-D3F343A605DC@ntt-at.com><9ae56b1e0903110158m1e6fb07o66110501e8581971@mail.gmail.com><1ECE0EB50388174790F9694F77522CCF1CCA681D@zrc2hxm0.corp.nortel.com><49B82FB4.4030002@gmail.com><1ECE0EB50388174790F9694F77522CCF1CCF97F5@zrc2hxm0.corp.nortel.com><49B84C55.5070107@gmail.com><1ECE0EB50388174790F9694F77522CCF1CCF9A06@zrc2hxm0.corp.nortel.com><9ae56b1e0903120134i76a6a92brb183205cd66e841f@mail.gmail.com><0D5F89FAC29E2C41B98A6A762007F5D001A27AF1@GBNTHT12009MSX.gb002.siemens.net><1ECE0EB50388174790F9694F77522CCF1CCF9DC8@zrc2hxm0.corp.nortel.com><9ae56b1e0903120754m66c0b7c5y225ce0f9193dba28@mail.gmail.com> <1ECE0EB50388174790F9694F77522CCF1CD4B01D@zrc2hxm0.corp.nortel.com>
From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Mary Barnes" <mary.barnes@nortel.com>, "Hans Erik van Elburg" <ietf.hanserik@gmail.com>
Cc: sip@ietf.org, "Elwell, John" <john.elwell@siemens.com>
Subject: Re: [Sip] Terminology (was RE: Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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And, just to clarify the phrase=20
"...changing a Uniform Resource Identifier (URI) in a request..." should
perhaps be worded more specifically as:
" ...changing the Request-URI in a SIP request..."

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Barnes, Mary (RICH2:AR00)
Sent: Thursday, March 12, 2009 10:20 AM
To: Hans Erik van Elburg
Cc: sip@ietf.org; Elwell, John
Subject: Re: [Sip] Terminology (was RE:
Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


When I use "retarget" I am using the definition that I put into 4244bis.
So, if you could ignore the tag or change it to 'iscontact' or whatever,
what would you change in that definition? Also, consider that if you
were to change everyplace that I use
"retarget"/"retargeted"/"retargeting" in 4244bis with 'iscontact' the
text would not make sense. So, even if we change the name of the tag, we
still need to understand the concept or define another term that means
that the request-uri in the incoming request (which should be the same
as the last hi-entry in the request) is being chnaged due to service
lookup, configuration, blah, blah, blah and that to meet the
requirements in the target-uri document, we want to tag this entry so
that this request-uri is not "lost" - i.e., this is the one the services
identified in the target-uri document care about.=20
=20
So, right now, here's what's in 4244bis:
=20
   The term "retarget" is used in this document to refer to the process
   of a Proxy Server/User Agent Client (UAC) changing a Uniform Resource
   Identifier (URI) in a request based on a lookup in a location service
   and thus changing the target of the request.
=20
We have previously discussed that we do need to expand the "based on a
lookup in a location service" to include configuration, etc., so let's
not debate that.=20
=20
But, what explicitly is the concern with that definition?
=20
Mary.=20
=20

________________________________

From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
Sent: Thursday, March 12, 2009 9:55 AM
To: Barnes, Mary (RICH2:AR00)
Cc: Elwell, John; sip@ietf.org
Subject: Re: [Sip] Terminology (was RE: Fwd:
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


I'm fine with discussing implementation if everyone is meaning the same
if they use the same words.

As  to the "retarget" term such does not seem to be the case.

/Hans Erik van Elburg



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Date: Thu, 12 Mar 2009 17:17:02 +0100
Message-ID: <9ae56b1e0903120917u3472f27bxc1366a9192473783@mail.gmail.com>
From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: Francois Audet <audet@nortel.com>
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Cc: sip@ietf.org, "Elwell, John" <john.elwell@siemens.com>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] Terminology (was RE: Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Yes. RFC3455, section 4.2.2.2.

/Hans Erik van Elburg


On Thu, Mar 12, 2009 at 4:11 PM, Francois Audet <audet@nortel.com> wrote:

> Does p-called-header include URI parameters?
>
>
> On Mar 12, 2009, at 7:07, "Christer Holmberg" <
> christer.holmberg@ericsson.com> wrote:
>
>
>  Hi,
>
> IF we would remove that requierment, and ONLY care about the case when the
> R-URI is replaced with the CONTACT, I don't think we need to do anything. I
> believe P-Called-Party-ID already solves that.
>
> Regards,
>
> Christer
>
>  ------------------------------
> *From:* sip-bounces@ietf.org [mailto:sip-bounces@ietf.org<sip-bounces@ietf.org>]
> *On Behalf Of *Elwell, John
> *Sent:* 12. maaliskuuta 2009 15:31
> *To:* Hans Erik van Elburg
> *Cc:* <sip@ietf.org>sip@ietf.org
> *Subject:* Re: [Sip] Terminology (was RE: Fwd:
> I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>
>  Hans Erik,
>
> The way I interpreted the 01 draft, you would not get the freephone number
> anyway. If that is a requirement, then I accept my suggestion won't suffice
> .
>
> John
>
>  ------------------------------
> *From:* Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com<ietf.hanserik@gmail.com>]
>
> *Sent:* 12 March 2009 12:57
> *To:* Elwell, John
> *Cc:* <sip@ietf.org>sip@ietf.org
> *Subject:* Re: [Sip] Terminology (was RE: Fwd:
> I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>
> But that does not solve the freephone service use case. Freephone
> represents a use case where the R-URI is replaced with an AoR of user B -
> not with the contact of user B.
>
> A calls Bfreephonenumber routed the B (AOR) routed to B's contact.
>
> The target-uri draft is interested in obtaining: Bfreephonenumber as that
> is what A used to reach B.
>
> So what we like to see that B UA receives is something like
>     H-I:  Bfreephone; "istarget",\
>         B,\
>         contact-B
>
> So you want to get Bfreephonenumber and not B.
>
> In certain situations you are also interested in B, I guess that  PBX's
> would interested in that to deliver to the correct user. We could
> distinguish them like:
>     H-I:  Bfreephone; "istarget",\
>         B,"aor"\
>         contact-B
>
> BR,
> /Hans Erik van Elburg
>
>
> On Thu, Mar 12, 2009 at 12:08 PM, Elwell, John < <john.elwell@siemens.com>
> john.elwell@siemens.com> wrote:
>
>> If 'istarget' is used only to identify the value that was overwritten in
>> the Request-URI by a contact URI, an alternative approach would be to
>> flag the contact URI H-I entry as 'iscontact'. Then the UAS would just
>> need to look for the most recent H-I entry that is not marked
>> 'iscontact'.
>>
>> John
>>
>>
>> ________________________________
>>
>>        From: <sip-bounces@ietf.org>sip-bounces@ietf.org [mailto:<sip-bounces@ietf.org>
>> sip-bounces@ietf.org] On
>> Behalf Of Hans Erik van Elburg
>>        Sent: 12 March 2009 08:35
>>        To: Mary Barnes
>>        Cc: <sip@ietf.org>sip@ietf.org
>>        Subject: Re: [Sip] Terminology (was RE: Fwd:
>> I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>>
>>
>>        Yes, because you are using the 3261 use of target and the
>> 4244bis introduced definition of retarget. I thought it was clear that
>> we need other words as those definitions don't match the target-uri
>> drafts use of the terms. Also they do not suffice to provide a solution
>> for the use cases in the target-uri draft.
>>
>>        The 3261 text you refer to is exactly about the case where the
>> home proxy overwrites the Request-URI with a new target. This target is
>> teh registered contact address. And hence this would be what target-uri
>> calls a hop or a route. This case and this is where it gets confusing is
>> not a "retarget" in the target-uri draft use of the term.
>>
>>        The target-uri draft states:
>>
>>           "To avoid confusion, we
>>           refer to a SIP URI that is an address for a user or resource
>> as a
>>           "target" and a SIP URI that is a hop for reaching that user
>> as a
>>
>>           "hop".
>>
>>
>>        Apparently that does not suffice to avoid confusion.
>>
>>        As for the tagging, speaking about the solution before agreeing
>> on the terminology and the problem it should solve is meaningless.
>>
>>        /Hans Erik van Elburg
>>
>>
>>
>>        On Thu, Mar 12, 2009 at 2:33 AM, Mary Barnes
>> < <mary.barnes@nortel.com>mary.barnes@nortel.com> wrote:
>>
>>
>>                Responses below [MB].
>>
>>
>>                -----Original Message-----
>>                From: Hans Erik van Elburg
>> [mailto: <ietf.hanserik@gmail.com>ietf.hanserik@gmail.com]
>>
>>                Sent: Wednesday, March 11, 2009 6:42 PM
>>                To: Barnes, Mary (RICH2:AR00)
>>                Cc: Shida Schubert; <sip@ietf.org>sip@ietf.org
>>                Subject: Re: Terminology (was RE: [Sip] Fwd: I-D
>>                ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>>
>>
>>                I was talking about the concept.
>>
>>                The use cases only describe  cases where the target-uri
>> (lets call that
>>                current target from now) has been lost when an initial
>> request for a
>>                dialog or standalone request arrives at the UAS.
>>
>>                [MB] And I think this is where the terminology confusion
>> starts - I
>>                don't think of the "lost" target-uri as being the
>> "current" target. In
>>                my mind, the "current" target is reflected by the last
>> hi-entry and the
>>                request-uri in the incoming request at the UAS. If the
>> entity that sent
>>                the request was the entity that added the last hi-entry,
>> then the uri in
>>                that hi-entry is the same as the request-URI in the SIP
>> request that
>>                arrives at the UAS. I refer to the "lost" uri as the one
>> that was
>>                "retargeted" - that's the one the UAS wants to pull from
>> the hi-entries
>>                in the incoming request. That hi-entry was not the one
>> that was just
>>                added by the entity that built the request just received
>> by the UAS.
>>                That hi-entry is tagged with whatever name we are going
>> to tag it with
>>                BEFORE the request is forwarded (using the term forward
>> per section 16.6
>>                of RFC 3261).  That tag is added once the target set of
>> potential
>>                candidates for the new request uri are determined in
>> section 16.5 of
>>                3261 (with "target set" being a 3261 term), just before
>> the request is
>>                forwarded in section 16.6 to one of those targets.  A
>> new hi-entry
>>                (which will be the last hi-entry in the request received
>> by the UAS) is
>>                added in section 16.6 of 3261 as the request is
>> forwarded.  At this
>>                point in time, the lost information is in the previous
>> hi-entry when the
>>                outgoing request is sent. [/MB]
>>
>>
>>                What has been lost is the current target of the request:
>>
>>                [MB] Right - at which point in my mind, it's no longer
>> the current
>>                target ;) Maybe we call it "lost". [/MB]
>>
>>
>>
>>                     Current target
>>
>>                  The current target of an initial request for a dialog
>> or standalone
>>                  request is the name or address to which the request is
>> targeted, i.e.
>>                  either the initial target inserted in the Request-URI
>> by the UAC that
>>                  originates the request, or when a retarget occurred,
>> the target
>>                  provided in that retarget operation.  Reroute and
>> translation
>>                  operations never change the current target.
>>
>>
>>                [MB] I don't think this definition fits what you want -
>> i.e., if there
>>                is no retargeting, then none of the hi-entries are
>> tagged - i.e., you
>>                won't have your concept of current. The way it works is
>> that if no
>>                hi-entries are tagged, then you know that there was no
>> retargeting, thus
>>                you know the request-uri has not been lost. [/MB]
>>
>>
>>                This defintion only makes sense when the following
>> definitions are used:
>>
>>                Name:
>>
>>                  A name is a moniker for an entity which refers to it
>> in a way which
>>
>>                  reveals nothing about where it is in a network.  In
>> SIP, tel URI
>>
>>                  which doesn't represent the location of the entity is
>> a name.
>>
>>
>>                Address:
>>
>>                  An address is an identifier for an entity which
>> describes it by its
>>
>>                  location on the network.  In SIP, the SIP URI itself
>> is a form of
>>
>>                  address because the host part of the URI, the only
>> mandatory part of
>>
>>                  the URI besides the scheme itself, indicates the
>> location of a SIP
>>
>>                  server that can be used to handle the request.
>>
>>                Route:
>>
>>                  Finally, a route is a   sequence of SIP entities
>> (including the UA
>>                itself!) which are
>>
>>                 traversed in order to forward a request to an address
>> or name.
>>
>>                Retarget (other term might be needed, as this is highly
>> confusing):
>>
>>                   A Request-URI rewrite operation that changes the
>> target identity of
>>                the request.
>>
>>                Reroute (other term might be needed):
>>
>>                   A Request-URI rewrite operation that does not change
>> the current
>>                target of the request, but     determines the route/next
>> hop taken to
>>                reach the target-identity.
>>
>>                /Hans Erik
>>
>> _______________________________________________
> Sip mailing list   <https://www.ietf.org/mailman/listinfo/sip>
> https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
>

--0015174bde448785340464ee50d5
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br>Yes. RFC3455, section 4.2.2.2.<br><br clear=3D"all">/Hans Erik van Elbu=
rg<br>
<br><br><div class=3D"gmail_quote">On Thu, Mar 12, 2009 at 4:11 PM, Francoi=
s Audet <span dir=3D"ltr">&lt;<a href=3D"mailto:audet@nortel.com">audet@nor=
tel.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D=
"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padd=
ing-left: 1ex;">
<div bgcolor=3D"#FFFFFF"><div>Does p-called-header include URI parameters?<=
br><br></div><div><div></div><div class=3D"h5"><div><br>On Mar 12, 2009, at=
 7:07, &quot;Christer Holmberg&quot; &lt;<a href=3D"mailto:christer.holmber=
g@ericsson.com" target=3D"_blank">christer.holmberg@ericsson.com</a>&gt; wr=
ote:<br>
<br></div><div></div><blockquote type=3D"cite"><div>

<div dir=3D"ltr" align=3D"left">=A0</div>
<div dir=3D"ltr" align=3D"left">
<div dir=3D"ltr" align=3D"left"><span>Hi,</span></div>
<div dir=3D"ltr" align=3D"left"><span></span>=A0</div>
<div dir=3D"ltr" align=3D"left"><span>IF we would remove that=20
requierment, and=A0ONLY care about the case when the R-URI is replaced with=
=20
the CONTACT, I don&#39;t think we need to do anything. I believe P-Called-P=
arty-ID=20
already solves that.</span></div>
<div dir=3D"ltr" align=3D"left"><span></span>=A0</div>
<div dir=3D"ltr" align=3D"left"><span>Regards,</span></div>
<div dir=3D"ltr" align=3D"left"><span></span>=A0</div>
<div dir=3D"ltr" align=3D"left"><span>Christer</span></div></div><br>
<blockquote dir=3D"ltr" style=3D"border-left: 2px solid rgb(0, 0, 0); paddi=
ng-left: 5px; margin-left: 5px; margin-right: 0px;">
  <div dir=3D"ltr" align=3D"left" lang=3D"en-us">
  <hr>
  <font size=3D"2" face=3D"Tahoma"><b>From:</b> <a href=3D"mailto:sip-bounc=
es@ietf.org" target=3D"_blank">sip-bounces@ietf.org</a>=20
  [<a href=3D"mailto:sip-bounces@ietf.org" target=3D"_blank">mailto:sip-bou=
nces@ietf.org</a>] <b>On Behalf Of </b>Elwell, John<br><b>Sent:</b>=20
  12. maaliskuuta 2009 15:31<br><b>To:</b> Hans Erik van Elburg<br><b>Cc:</=
b>=20
  <a href=3D"mailto:sip@ietf.org" target=3D"_blank"></a><a href=3D"mailto:s=
ip@ietf.org" target=3D"_blank">sip@ietf.org</a><br><b>Subject:</b> Re: [Sip=
] Terminology (was RE: Fwd:=20
  I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br></font><br></=
div>
  <div></div>
  <div dir=3D"ltr" align=3D"left">
  <div dir=3D"ltr" align=3D"left"><span><font color=3D"#000080" face=3D"Ari=
al">Hans Erik,</font></span></div>
  <div dir=3D"ltr" align=3D"left"><span><font color=3D"#000080" face=3D"Ari=
al"></font></span>=A0</div>
  <div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial"><font color=3D=
"#000080">The way I interpreted the 01 draft, you would not get the=20
  freephone number anyway. If that is a requirement, then I accept my sugge=
stion=20
  won&#39;t suffice<span>.</span></font></font></span></div>
  <div dir=3D"ltr" align=3D"left"><span><font color=3D"#000080" face=3D"Ari=
al"></font></span>=A0</div>
  <div dir=3D"ltr" align=3D"left"><span><font color=3D"#000080" face=3D"Ari=
al">John</font></span></div></div><br>
  <blockquote style=3D"border-left: 2px solid rgb(0, 0, 128); padding-left:=
 5px; margin-left: 5px; margin-right: 0px;">
    <div dir=3D"ltr" align=3D"left" lang=3D"en-us">
    <hr>
    <font size=3D"2" face=3D"Tahoma"><b>From:</b> Hans Erik van Elburg=20
    [<a href=3D"mailto:ietf.hanserik@gmail.com" target=3D"_blank">mailto:ie=
tf.hanserik@gmail.com</a>] <br><b>Sent:</b> 12 March 2009=20
    12:57<br><b>To:</b> Elwell, John<br><b>Cc:</b>=20
    <a href=3D"mailto:sip@ietf.org" target=3D"_blank"></a><a href=3D"mailto=
:sip@ietf.org" target=3D"_blank">sip@ietf.org</a><br><b>Subject:</b> Re: [S=
ip] Terminology (was RE: Fwd:=20
    I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br></font><br>=
</div>
    <div></div>But that does not solve the freephone service use case.=20
   =20
    Freephone represents a use case where the R-URI is replaced with an AoR=
 of=20
    user B - not with the contact of user B. <br><br>A calls Bfreephonenumb=
er=20
    routed the B (AOR) routed to B&#39;s contact.<br><br>The target-uri dra=
ft is=20
    interested in obtaining: Bfreephonenumber as that is what A used to rea=
ch=20
    B.<br><br>So what we like to see that B UA receives is something=20
    like<br>=A0=A0=A0 H-I:=A0 Bfreephone;=20
    &quot;istarget&quot;,\<br>=A0=A0=A0 =A0=A0=A0=20
    B,\<br>=A0=A0=A0 =A0=A0=A0 contact-B<br><br>So you want to=20
    get Bfreephonenumber and not B.<br><br>In certain situations you are al=
so=20
    interested in B, I guess that=A0 PBX&#39;s would interested in that to=
=20
    deliver to the correct user. We could distinguish them=20
    like:<br>=A0=A0=A0 H-I:=A0 Bfreephone;=20
    &quot;istarget&quot;,\<br>=A0=A0=A0 =A0=A0=A0=20
    B,&quot;aor&quot;\<br>=A0=A0=A0 =A0=A0=A0 contact-B<br><br>BR,<br clear=
=3D"all">/Hans Erik van Elburg<br><br><br>
    <div class=3D"gmail_quote">On Thu, Mar 12, 2009 at 12:08 PM, Elwell, Jo=
hn <span dir=3D"ltr">&lt;<a href=3D"mailto:john.elwell@siemens.com" target=
=3D"_blank"></a><a href=3D"mailto:john.elwell@siemens.com" target=3D"_blank=
">john.elwell@siemens.com</a>&gt;</span>=20
    wrote:<br>
    <blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(2=
04, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">If=20
      &#39;istarget&#39; is used only to identify the value that was overwr=
itten=20
      in<br>the Request-URI by a contact URI, an alternative approach would=
 be=20
      to<br>flag the contact URI H-I entry as &#39;iscontact&#39;. Then the=
 UAS would=20
      just<br>need to look for the most recent H-I entry that is not=20
      marked<br>&#39;iscontact&#39;.<br><br>John<br><br><br>_______________=
_________________<br><br>=A0=20
      =A0 =A0 =A0From: <a href=3D"mailto:sip-bounces@ietf.org" target=3D"_b=
lank"></a><a href=3D"mailto:sip-bounces@ietf.org" target=3D"_blank">sip-bou=
nces@ietf.org</a> [mailto:<a href=3D"mailto:sip-bounces@ietf.org" target=3D=
"_blank"></a><a href=3D"mailto:sip-bounces@ietf.org" target=3D"_blank">sip-=
bounces@ietf.org</a>] On<br>
Behalf=20
      Of Hans Erik van Elburg<br>=A0 =A0 =A0 =A0Sent: 12 March 2009=20
      08:35<br>=A0 =A0 =A0 =A0To: Mary Barnes<br>=A0 =A0=20
      =A0 =A0Cc: <a href=3D"mailto:sip@ietf.org" target=3D"_blank"></a><a h=
ref=3D"mailto:sip@ietf.org" target=3D"_blank">sip@ietf.org</a><br>=A0=20
      =A0 =A0 =A0Subject: Re: [Sip] Terminology (was RE:=20
      Fwd:<br>I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br>
      <div>
      <div></div>
      <div><br><br>=A0 =A0 =A0 =A0Yes, because you are=20
      using the 3261 use of target and the<br>4244bis introduced definition=
 of=20
      retarget. I thought it was clear that<br>we need other words as those=
=20
      definitions don&#39;t match the target-uri<br>drafts use of the terms=
. Also=20
      they do not suffice to provide a solution<br>for the use cases in the=
=20
      target-uri draft.<br><br>=A0 =A0 =A0 =A0The 3261 text you=20
      refer to is exactly about the case where the<br>home proxy overwrites=
 the=20
      Request-URI with a new target. This target is<br>teh registered conta=
ct=20
      address. And hence this would be what target-uri<br>calls a hop or a=
=20
      route. This case and this is where it gets confusing is<br>not a=20
      &quot;retarget&quot; in the target-uri draft use of the term.<br><br>=
=A0 =A0=20
      =A0 =A0The target-uri draft states:<br><br>=A0 =A0 =A0=20
      =A0 =A0 &quot;To avoid confusion, we<br>=A0 =A0 =A0 =A0=20
      =A0 refer to a SIP URI that is an address for a user or resource<br>a=
s=20
      a<br>=A0 =A0 =A0 =A0 =A0 &quot;target&quot; and a SIP URI that is a=
=20
      hop for reaching that user<br>as a<br><br>=A0 =A0 =A0 =A0=20
      =A0 &quot;hop&quot;.<br><br><br>=A0 =A0 =A0 =A0Apparently that does=
=20
      not suffice to avoid confusion.<br><br>=A0 =A0 =A0 =A0As for=20
      the tagging, speaking about the solution before agreeing<br>on the=20
      terminology and the problem it should solve is meaningless.<br><br>=
=A0=20
      =A0 =A0 =A0/Hans Erik van Elburg<br><br><br><br>=A0 =A0=20
      =A0 =A0On Thu, Mar 12, 2009 at 2:33 AM, Mary Barnes<br>&lt;<a href=3D=
"mailto:mary.barnes@nortel.com" target=3D"_blank"></a><a href=3D"mailto:mar=
y.barnes@nortel.com" target=3D"_blank">mary.barnes@nortel.com</a>&gt;=20
      wrote:<br><br><br>=A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0Responses below [MB].<br><br><br>=A0 =A0 =A0 =A0 =A0=20
      =A0 =A0 =A0-----Original Message-----<br>=A0 =A0 =A0=20
      =A0 =A0 =A0 =A0 =A0From: Hans Erik van Elburg<br>[mailto:<a href=3D"m=
ailto:ietf.hanserik@gmail.com" target=3D"_blank"></a><a href=3D"mailto:ietf=
.hanserik@gmail.com" target=3D"_blank">ietf.hanserik@gmail.com</a>]<br><br>=
=A0=20
      =A0 =A0 =A0 =A0 =A0 =A0 =A0Sent: Wednesday, March 11,=20
      2009 6:42 PM<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0To:=20
      Barnes, Mary (RICH2:AR00)<br>=A0 =A0 =A0 =A0 =A0 =A0=20
      =A0 =A0Cc: Shida Schubert; <a href=3D"mailto:sip@ietf.org" target=3D"=
_blank"></a><a href=3D"mailto:sip@ietf.org" target=3D"_blank">sip@ietf.org<=
/a><br>=A0 =A0 =A0 =A0=20
      =A0 =A0 =A0 =A0Subject: Re: Terminology (was RE: [Sip] Fwd:=20
      I-D<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br><br><br>=
=A0=20
      =A0 =A0 =A0 =A0 =A0 =A0 =A0I was talking about the=20
      concept.<br><br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0The=20
      use cases only describe =A0cases where the target-uri<br>(lets call=
=20
      that<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0current=20
      target from now) has been lost when an initial<br>request for a<br>=
=A0=20
      =A0 =A0 =A0 =A0 =A0 =A0 =A0dialog or standalone=20
      request arrives at the UAS.<br><br>=A0 =A0 =A0 =A0 =A0=20
      =A0 =A0 =A0[MB] And I think this is where the terminology=20
      confusion<br>starts - I<br>=A0 =A0 =A0 =A0 =A0 =A0=20
      =A0 =A0don&#39;t think of the &quot;lost&quot; target-uri as being=20
      the<br>&quot;current&quot; target. In<br>=A0 =A0 =A0 =A0 =A0 =A0=20
      =A0 =A0my mind, the &quot;current&quot; target is reflected by the=20
      last<br>hi-entry and the<br>=A0 =A0 =A0 =A0 =A0 =A0=20
      =A0 =A0request-uri in the incoming request at the UAS. If=20
      the<br>entity that sent<br>=A0 =A0 =A0 =A0 =A0 =A0=20
      =A0 =A0the request was the entity that added the last=20
      hi-entry,<br>then the uri in<br>=A0 =A0 =A0 =A0 =A0 =A0=20
      =A0 =A0that hi-entry is the same as the request-URI in the=20
      SIP<br>request that<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0arrives at the UAS. I refer to the &quot;lost&quot; uri as the one=
<br>that=20
      was<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;retargeted&quot;=20
      - that&#39;s the one the UAS wants to pull from<br>the hi-entries<br>=
=A0=20
      =A0 =A0 =A0 =A0 =A0 =A0 =A0in the incoming request.=20
      That hi-entry was not the one<br>that was just<br>=A0 =A0 =A0=20
      =A0 =A0 =A0 =A0 =A0added by the entity that built the=20
      request just received<br>by the UAS.<br>=A0 =A0 =A0 =A0 =A0=20
      =A0 =A0 =A0That hi-entry is tagged with whatever name we are=20
      going<br>to tag it with<br>=A0 =A0 =A0 =A0 =A0 =A0=20
      =A0 =A0BEFORE the request is forwarded (using the term=20
      forward<br>per section 16.6<br>=A0 =A0 =A0 =A0 =A0 =A0=20
      =A0 =A0of RFC 3261). =A0That tag is added once the target set=20
      of<br>potential<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0candidates for the new request uri are determined in<br>section 16=
.5=20
      of<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A03261 (with=20
      &quot;target set&quot; being a 3261 term), just before<br>the request=
 is<br>=A0=20
      =A0 =A0 =A0 =A0 =A0 =A0 =A0forwarded in section 16.6=20
      to one of those targets. =A0A<br>new hi-entry<br>=A0 =A0 =A0=20
      =A0 =A0 =A0 =A0 =A0(which will be the last hi-entry in the=20
      request received<br>by the UAS) is<br>=A0 =A0 =A0 =A0 =A0=20
      =A0 =A0 =A0added in section 16.6 of 3261 as the request=20
      is<br>forwarded. =A0At this<br>=A0 =A0 =A0 =A0 =A0=20
      =A0 =A0 =A0point in time, the lost information is in the=20
      previous<br>hi-entry when the<br>=A0 =A0 =A0 =A0 =A0 =A0=20
      =A0 =A0outgoing request is sent. [/MB]<br><br><br>=A0 =A0=20
      =A0 =A0 =A0 =A0 =A0 =A0What has been lost is the current=20
      target of the request:<br><br>=A0 =A0 =A0 =A0 =A0 =A0=20
      =A0 =A0[MB] Right - at which point in my mind, it&#39;s no longer<br>=
the=20
      current<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0target=20
      ;) Maybe we call it &quot;lost&quot;. [/MB]<br><br><br><br>=A0 =A0 =
=A0=20
      =A0 =A0 =A0 =A0 =A0 =A0 =A0 Current=20
      target<br><br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0The current target of an initial request for a dialog<br>or=20
      standalone<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0request is the name or address to which the request is<br>targeted=
,=20
      i.e.<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0either the initial target inserted in the Request-URI<br>by the UA=
C=20
      that<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0originates the request, or when a retarget occurred,<br>the=20
      target<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0provided in that retarget operation. =A0Reroute=20
      and<br>translation<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0 =A0operations never change the current target.<br><br><br>=A0=20
      =A0 =A0 =A0 =A0 =A0 =A0 =A0[MB] I don&#39;t think this=20
      definition fits what you want -<br>i.e., if there<br>=A0 =A0 =A0=20
      =A0 =A0 =A0 =A0 =A0is no retargeting, then none of the=20
      hi-entries are<br>tagged - i.e., you<br>=A0 =A0 =A0 =A0 =A0=20
      =A0 =A0 =A0won&#39;t have your concept of current. The way it works=
=20
      is<br>that if no<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0hi-entries are tagged, then you know that there was=20
      no<br>retargeting, thus<br>=A0 =A0 =A0 =A0 =A0 =A0=20
      =A0 =A0you know the request-uri has not been lost.=20
      [/MB]<br><br><br>=A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0This defintion only makes sense when the following<br>definitions=
=20
      are used:<br><br>=A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0Name:<br><br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0A name is a moniker for an entity which refers to it<br>in a way=
=20
      which<br><br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0reveals nothing about where it is in a network. =A0In<br>SIP, tel=
=20
      URI<br><br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0which doesn&#39;t represent the location of the entity is<br>a=20
      name.<br><br><br>=A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0Address:<br><br>=A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0 =A0An address is an identifier for an entity which<br>describes=
=20
      it by its<br><br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0location on the network. =A0In SIP, the SIP URI itself<br>is a=20
      form of<br><br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0address because the host part of the URI, the only<br>mandatory pa=
rt=20
      of<br><br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0the URI besides the scheme itself, indicates the<br>location of a=
=20
      SIP<br><br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
      =A0server that can be used to handle the request.<br><br>=A0 =A0=20
      =A0 =A0 =A0 =A0 =A0 =A0Route:<br><br>=A0 =A0=20
      =A0 =A0 =A0 =A0 =A0 =A0 =A0Finally, a route is a=20
      =A0 sequence of SIP entities<br>(including the UA<br>=A0 =A0=20
      =A0 =A0 =A0 =A0 =A0 =A0itself!) which are<br><br>=A0=20
      =A0 =A0 =A0 =A0 =A0 =A0 =A0 traversed in order to=20
      forward a request to an address<br>or name.<br><br>=A0 =A0 =A0=20
      =A0 =A0 =A0 =A0 =A0Retarget (other term might be needed, as=20
      this is highly<br>confusing):<br><br>=A0 =A0 =A0 =A0 =A0=20
      =A0 =A0 =A0 =A0 A Request-URI rewrite operation that changes=20
      the<br>target identity of<br>=A0 =A0 =A0 =A0 =A0 =A0=20
      =A0 =A0the request.<br><br>=A0 =A0 =A0 =A0 =A0 =A0=20
      =A0 =A0Reroute (other term might be needed):<br><br>=A0 =A0=20
      =A0 =A0 =A0 =A0 =A0 =A0 =A0 A Request-URI rewrite=20
      operation that does not change<br>the current<br>=A0 =A0 =A0=20
      =A0 =A0 =A0 =A0 =A0target of the request, but =A0 =A0=20
      determines the route/next<br>hop taken to<br>=A0 =A0 =A0 =A0=20
      =A0 =A0 =A0 =A0reach the target-identity.<br><br>=A0 =A0=20
      =A0 =A0 =A0 =A0 =A0 =A0/Hans=20
    Erik<br><br></div></div></blockquote></div></blockquote></blockquote>
</div></blockquote></div></div><blockquote type=3D"cite"><div><div class=3D=
"im"><span>_______________________________________________</span><br><span>=
Sip mailing list =A0<a href=3D"https://www.ietf.org/mailman/listinfo/sip" t=
arget=3D"_blank"></a><a href=3D"https://www.ietf.org/mailman/listinfo/sip" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a></span><br>
<span>This list is for NEW development of the core SIP Protocol</span><br><=
/div><span>Use <a href=3D"mailto:sip-implementors@cs.columbia.edu" target=
=3D"_blank">sip-implementors@cs.columbia.edu</a> for questions on current s=
ip</span><br>
<span>Use <a href=3D"mailto:sipping@ietf.org" target=3D"_blank">sipping@iet=
f.org</a> for new developments on the application of sip</span></div></bloc=
kquote></div></blockquote></div><br>

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To: Mary Barnes <mary.barnes@nortel.com>, Dean Willis <dean.willis@softarmor.com>, Shida Schubert <shida@ntt-at.com>
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Which is where I was coming from on my offline email to you, i.e. that we s=
hould start recording our position of any changes required to RFC 3261 in a=
n appendix. That way we know what changes we need to make to RFC 3261 when =
we get to make them, even if they are only for alignment of terminology. We=
 can decide later whether the appendix gets published with the update to ei=
ther RFC 4244 or target-URI or some combined document.

Keith=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Mary Barnes
> Sent: Wednesday, March 11, 2009 4:27 PM
> To: Dean Willis; Shida Schubert
> Cc: sip@ietf.org
> Subject: Re: [Sip] Fwd: I-D=20
> ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>=20
> I don't think it should cause so much concern as I think it=20
> just requires clarifying text and we just need to agree to=20
> more precise
> terms: e.g., qualifying the use of "forward" with either=20
> "next hop forwarding" or "request forwarding" for this=20
> particular case. As long as we can be precise in 4244bis,=20
> then that should suffice for now, but whenever 3261 is=20
> updated or bised, then I think the clarifications should be included.
>=20
> Mary.=20
>=20
> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Wednesday, March 11, 2009 1:38 AM
> To: Shida Schubert
> Cc: sip@ietf.org List; Barnes, Mary (RICH2:AR00)
> Subject: Re: [Sip] Fwd: I-D
> ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>=20
>=20
> On Mar 11, 2009, at 1:13 AM, Shida Schubert wrote:
>=20
> >
> >       One reason this is so difficult relates to the=20
> problem statement
>=20
> > in target-uri in that
> >       RFC 3261 doesn't differentiate the mechanism by which the new
> >       (target) Request-URI is selected.  Another issue is=20
> that some of
>=20
> > the terminology in
> >       RFC 3261 is overloaded - e.g., "forwarding" refers both to a=20
> > Proxy
> >       which does not have responsibility for the domain of the=20
> > request-URI
> >       in the incoming request, thus the proxy just "forwards" the=20
> > request to
> >       the next hop AND "forwarding" is used to describe the process=20
> > whereby
> >       the outgoing request is built and "forwarded" to the=20
> next hop at
>=20
> > which
> >       point the proxy does not know how the new request-uri was=20
> > selected.
> >       RFC 4244 has attempted to clarify the terms and=20
> attempts to use=20
> > "forward"
> >       in the context of the former situation and "retarget" for the=20
> > case whereby
> >       a proxy is responsible for the domain and thus can=20
> use a number=20
> > of
> >       mechanism to select the new target for the request - e.g., a=20
> > REGISTRAR,
> >       configured data, etc.
> >
>=20
> Thanks, Shida.
>=20
> I personally would not be averse to an Essential Correction that
> corrects the terminology in RFC 3261, but I suspect that will=20
> make many
> heads spin around elsewhere in the WG.
>=20
> --
> Dean
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> =

From ian.elz@ericsson.com  Thu Mar 12 09:50:40 2009
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From: "Ian Elz" <ian.elz@ericsson.com>
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Mary,

Thanks.

RFC4244 was always considered a little fuzzy in some areas but the
intent was generally understood.

We may be able to use this opportunity to clarify some additional parts.

I will read through the full draft with this purpose to see what may
need clarification.

Ian Elz

System Manager
DUCI LDC UK
(Lucid Duck)

Office: + 44 24 764 35256
ian.elz@ericsson.com


-----Original Message-----
From: Mary Barnes [mailto:mary.barnes@nortel.com]=20
Sent: 12 March 2009 15:53
To: Ian Elz
Cc: sip@ietf.org; Francois Audet
Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

Ian,
=20
I do see your point now - the text does need clarification.  Detailed
comments are below [MB].
=20
Thanks,
Mary.=20

________________________________

From: Ian Elz [mailto:ian.elz@ericsson.com]=20
Sent: Thursday, March 12, 2009 10:12 AM
To: Barnes, Mary (RICH2:AR00)
Cc: sip@ietf.org; Audet, Francois (SC100:3055)
Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20



Mary,

I was only considering the specific domain case as included in paragraph
2 of section 3.3 in the draft:

                                                  Thus, local policy

   MAY also be used to determine whether to include the History-Info

   header at all, whether to capture a specific Request-URI in the

   header, or whether it be included only in the Request as it is

   retargeted within a specific domain (PRIV-req-3).  In the latter

   case, this is accomplished by adding a new priv-value, history, to

   the Privacy header [RFC3323] indicating whether any or a specific

   History-Info header(s) SHOULD be forwarded.=20

[MB] Yes, this is a little fuzzy. It is trying to make a very general
statement that we are adding this new "priv-value" to the  Privacy
header so that we can control whether the entries are sent outside the
proxy.  I think that sentence is okay if we remove the normative
(SHOULD) and just add a note that the details of the use of the privacy
header are provided in section 4.1., something like the following:=20
   In the latter

   case, this is accomplished by adding a new priv-value, history, to

   the Privacy header [RFC3323] indicating whether or not any or a
specific

   History-Info header(s) are forwarded outside a specific domain for
which the proxy is responsible.=20
=20
   The details as to the use of the new priv-value with the Privacy
header are provided in section 4.1.

 [/MB]

=20

    o  Privacy: An optional parameter for History-Info, reflected in the

      History-Info header field values by including the Privacy Header

      [RFC3323] with a priv-value of "history" escaped in the hi-

      targeted-to-uri or by adding the Privacy header with a priv-value

      of "history" to the Request.  The use of the Privacy Header with a

      priv-value of "history" indicates whether a specific or all

      History-Info headers should not be forwarded.

(There appears to be an inconsistency here as sections 3.3 and 4.1
appear to contradict. [Perhaps it just needs clarification that section
3.3 allows forwarding within the domain and 4.1 prohibits forwarding
outside the domain?])=20

[MB] Yes, this is also fuzzy. It should really read (and the SHOULD NOT
ought to be caps):

 Privacy: An optional parameter for History-Info, reflected in the

      History-Info header field values by including the Privacy Header

      [RFC3323] with a priv-value of "history" escaped in the hi-

      targeted-to-uri or by adding the Privacy header with a priv-value

      of "history" to the Request.   The latter case indicates that =20

      none of the History-Info headers should be forwarded , whereas the
use of

      the Privacy header escaped in the hi-targeted-to-uri means that a
specific hi-entry SHOULD NOT be forwarded.=20
[/MB]

Assuming that the value 'history' means that presentation and forwarding
should not occur.

In this case the inclusion of the value 'history' in the Privacy header
impacts all H-I entries. If one node includes the value 'history' in the
Privacy header then there is no way for another node to specifically
allow the uri included in the hi-targeted-to-uri to not be restricted.
To allow forwarding of a specific H-I entry there needs to be another
value which can be included in the Privacy header parameter of for the
hi-targeted-to-uri.

[MB] Exactly - that's why I had suggested earlier that we perhaps
RECOMMEND that the Privacy header only be used in the escaped form
associated with a specific hi-entry. There is text elsewhere explaining
that you do have potential loss of functionality when you use the
Privacy header. Perhaps, we need a stronger warning. [/MB]

If the value "none" has specific meanings which will cause confusion
then another value should be used. =20

More comments in-line

Nit in A.3

In both the sequence and the text you indicate that the INVITE to UA-B
will retransmit. This should not occur if a 100 Trying has been received
from UA-B. This means that the retransmit timer cannot be used to
determine if UA-B does not respond. There will be a separate timer
specified for determining no answer after the 180 has been received.

[MB] You are correct - that flow is buggy and there are bugs in the
flows and we will rework these and/or remove some of them.[/MB]

Ian Elz

Office: + 44 24 764 35256

ian.elz@ericsson.com

-----Original Message-----
From: Mary Barnes [mailto:mary.barnes@nortel.com]
Sent: 12 March 2009 13:55
To: Ian Elz
Cc: sip@ietf.org; Francois Audet
Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

Ian,

Okay, so I think I might understand now - you want to use the "none" to

override the Privacy header that would be (in current 4244/4244bis)

applied to the whole message as it traverses across the n/w from the UAC

to the UAS. Is that correct?=20

=20

[ige] Yes - currently a Privacy header with value 'history' will prevent
any H-I entries from being passed. If the Privacy header does not
contain the value 'history' then individual H-I entries may have privacy
associated. Those which do will not be forwarded but those with no
privacy will be forwarded.

That then though changes the meaning of the

Privacy header, as I understand it - i.e., the "none" I thought had to

do with specifying that an intermediary MUST NOT change the value of the

Privacy header. But, I guess you are saying we specify a slightly

different usage in 4244bis such that when we add it to an hi-entry, it

specifies that that hi-entry MUST NOT be dropped by an intermediary.

That might work, although, we would need verbage about the "risks" of

doing so.

=20

[ige] I would not be as strong as MUST NOT. There may be local policy
which will override what is included. I would prefer the words NEED NOT
be removed {Scott B will hate me for that.} or perhaps MAY be forwarded
based upon local policy.=20

Thanks,

Mary.=20

-----Original Message-----

From: Ian Elz [mailto:ian.elz@ericsson.com]=20

Sent: Thursday, March 12, 2009 8:43 AM

To: Barnes, Mary (RICH2:AR00)

Cc: sip@ietf.org; Audet, Francois (SC100:3055)

Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

Mary,

(Written after reading your discussion with Hans Erik.)

First an apology for not making it clear in my original email that a

value of "none" in a Privacy header parameter in a History-Info entry

was only applicable to the uri contained in the hi-targeted-to-uri.

I only suggested using the value "none" as this is a value defined for

the Privacy header which explicitly indicates that privacy is not

required. Any other value with the meaning of explicitly indicating that

there is no privacy required for the specific uri could be used.

With regard to your second paragraph below I understand that the SIP and

PSTN models do not align. You are correct in your assumption that this

problem is not unique for History-Info. The problem also arises with the

Referred-By header.=20

I have raised this for History-Info at this time as your new draft

presents an opportunity to correct this problem for the H-I case.

The general SIP privacy issue arises as when the Privacy header was

defined the concept of carrying third party identities in the SIP

messages was not defined. This came later with the REFER Request and the

associated Referred-By header and the History-Info header.=20

We should take what opportunities arise to discuss and correct this if

there is a will to make these changes.

Ian Elz

Office: + 44 24 764 35256

ian.elz@ericsson.com


-----Original Message-----

From: Mary Barnes [mailto:mary.barnes@nortel.com]

Sent: 11 March 2009 16:53

To: Ian Elz; Francois Audet

Cc: sip@ietf.org

Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

Ian,

This is an interesting question. I need to think about it some more, but

I don't like the "none" idea as it really must be up to the entity that

added the Privacy header to the request, as to whether it wants the HI

entries that it adds to go outside a domain. My initial thought is that

we can't overrule the overall Privacy header. The thing is that the

Privacy header doesn't preclude gathering the information and using

within a domain AND if it were to not include an hi-entry when the

request leaves the domain for which the proxy is responsible, the

recipient can add the hi-entry for the current request-uri before it

adds the new hi-entry. =20

Your PSTN example doesn't strictly map obviously to a SIP model and I

would think you might consider the PSTN hop to be within the same domain

for which the proxy (or gw, I guess) is responsible OR consider this a

walled garden whereby the Privacy header doesn't apply. Which brings me

to a more general question as to how you all deal with other headers

when you do your mapping to the PSTN when there is a Privacy header?  It

would seem this problem isn't unique to History-Info, although I know

little about the details of PSTN I/W.

Regards,

Mary.=20

-----Original Message-----

From: Ian Elz [mailto:ian.elz@ericsson.com]

Sent: Wednesday, March 11, 2009 4:54 AM

To: Barnes, Mary (RICH2:AR00); Audet, Francois (SC100:3055)

Cc: sip@ietf.org

Subject: RE: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

Mary, Francois,

Section 4.1 of the draft allows for the addition of the Privacy header

parameter with a value of "history" to be included with the

hi-targeted-to-uri.

Can this be extended to also allow the value "none".

A node adding a H-I entry is allowed to specify the privacy value

"history" either in the Privacy header or as a header parameter

associated with the hi-targeted-to-uri. If the former action is taken by

one node this results in privacy for all history info header entries

even if this is not required. There is no way for privacy of an

individual H-I entry to be set to none if no Privacy is required for the

uri.

This creates a specific problem when interworking with the PSTN where

the privacy is associated with each E.164 number included in the

protocol. If an INVITE is being mapped to ISUP and the H-I entries are

being used to map to redirection numbers then a single Privacy header

with the value "history" will result in all numbers being restricted.

If the Privacy header parameter could include the value "none" then this

would explicitly indicate that the associated uri was allowed to be

presented.

The ability to explicitly allow presentation of specific H-I entries may

also be useful in pure sip implementations.

I don't believe that this change would create any backward compatibility

issues. The existing implementations will continue as deployed but new

implementations could explicitly set no privacy for a specific uri.

There are no security issues other than those already defined in the

draft that an intervening node could modify an existing H-I entry. The

Privacy header value of "none" should only be included by the node in

the same manner as the existing inclusion of the "history" value.

Ian Elz


-----Original Message-----

From: i-d-announce-bounces@ietf.org

[mailto:i-d-announce-bounces@ietf.org] On Behalf Of

Internet-Drafts@ietf.org

Sent: 04 March 2009 15:15

To: i-d-announce@ietf.org

Subject: I-D Action:draft-barnes-sip-rfc4244bis-00.txt=20

A New Internet-Draft is available from the on-line Internet-Drafts

directories.

        Title           : An Extension to the Session Initiation

Protocol (SIP) for Request History Information

        Author(s)       : M. Barnes, F. Audet

        Filename        : draft-barnes-sip-rfc4244bis-00.txt

        Pages           : 49

        Date            : 2009-03-04

This document defines a standard mechanism for capturing the history

information associated with a Session Initiation Protocol (SIP) request.

This capability enables many enhanced services by providing the

information as to how and why a call arrives at a specific application

or user.  This document defines a new optional SIP header, History-Info,

for capturing the history information in requests.

A URL for this Internet-Draft is:

http://www.ietf.org/internet-drafts/draft-barnes-sip-rfc4244bis-00.txt

Internet-Drafts are also available by anonymous FTP at:

ftp://ftp.ietf.org/internet-drafts/

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On Wed, Mar 11, 2009 at 2:49 PM, Nils Ohlmeier <lists@ohlmeier.org> wrote:
> Sure to create a Contact binding via an authenticated REGISTER is one
> possible solution.
> But I think this only works because you rely on the fact that REGISTERs
> are not being retargeted like INVITEs. Which then in the end means you
> simply can forget about authenticating all other requests, as long as the
> Contact or IP/port was successfully authenticated via the "one hop"
> REGISTER method (refer-to: IMS).
> The advantage of this solution is clearly that it can be easily deployed
> by hopefully most of the service providers today.
>
> But I believe the proper technical solution would be that the server
> authenticates itself in the challenge, plus adding protected informations
> to the challenge which allows the receiver of the challenge to verify tha=
t
> this challenge was/is targeted to himself.
> The dis-advantage is clearly that this would be only possible with
> extensions of the existing protocols. But we would/might gain other
> benefits by such a solution as well.

I did not follow the discussion on draft-dotson-sip-mutual-auth. Has
this work been discontinued?

http://tools.ietf.org/id/draft-dotson-sip-mutual-auth-03.txt

Cheers,
--=20
Victor Pascual =C3=81vila

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Subject: [Sip] Fwd: Internet Society Panel on IPv6 Deployment during IETF 74
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If you have opinions about IPv6, or NAT, you might be interested in  
the following.

Begin forwarded message:

> From: Russ Housley <housley@vigilsec.com>
> Date: March 12, 2009 10:55:26 AM MDT (CA)
> To: all74@ietf.org
> Cc: iab@iab.org, iesg@ietf.org
> Subject: Internet Society Panel on IPv6 Deployment during IETF 74
> Reply-To: Russ Housley <housley@vigilsec.com>
>
> IETF 74 Attendees:
>
> I am writing to make sure that you are aware of the Internet Society  
> (ISOC) briefing panel focused on the real world experiences  
> associated with deploying IPv6.  While not part of the IETF agenda,  
> the panel session will be coincident with the IETF 74 meeting in San  
> Francisco.
>
> "The Seven Stages of IPv6" will review the opportunities of IPv6  
> from the perspective of network citizens in the seven stages of  
> dealing
> with massive change.  Leslie Daigle, ISOC Chief Internet Technology  
> Officer, will moderate the panel.  The panel session will be held at  
> the Hilton San Francisco on Tuesday, March 24th at 11:30am - 12:45pm  
> PDT.
>
> Lunch will be provided; however, seating is limited and pre- 
> registration is required before
> Wednesday, March 18th at 5:00pm PDT.  Registration is free.  The  
> panel session is open
> to interested IETF participants, ISOC members, and journalists.
>
> Register online at:
> http://www.isoc.org/isoc/conferences/registration/20090324.shtml


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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><div><br></div>If you =
have&nbsp;opinions about&nbsp;IPv6, or NAT, you might be interested in =
the&nbsp;following.&nbsp;<br><div><br><div>Begin forwarded =
message:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><font face=3D"Helvetica" =
size=3D"3" color=3D"#000000" style=3D"font: 12.0px Helvetica; color: =
#000000"><b>From: </b></font><font face=3D"Helvetica" size=3D"3" =
style=3D"font: 12.0px Helvetica">Russ Housley &lt;<a =
href=3D"mailto:housley@vigilsec.com">housley@vigilsec.com</a>></font></div=
><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><b>Date: =
</b></font><font face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px =
Helvetica">March 12, 2009 10:55:26 AM MDT (CA)</font></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><b>To: </b></font><font =
face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px Helvetica"><a =
href=3D"mailto:all74@ietf.org">all74@ietf.org</a></font></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><b>Cc: </b></font><font =
face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px Helvetica"><a =
href=3D"mailto:iab@iab.org">iab@iab.org</a>, <a =
href=3D"mailto:iesg@ietf.org">iesg@ietf.org</a></font></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><b>Subject: =
</b></font><font face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px =
Helvetica"><b>Internet Society Panel on IPv6 Deployment during IETF =
74</b></font></div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><font face=3D"Helvetica" =
size=3D"3" color=3D"#000000" style=3D"font: 12.0px Helvetica; color: =
#000000"><b>Reply-To: </b></font><font face=3D"Helvetica" size=3D"3" =
style=3D"font: 12.0px Helvetica">Russ Housley &lt;<a =
href=3D"mailto:housley@vigilsec.com">housley@vigilsec.com</a>></font></div=
><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; min-height: 14px; "><br></div> </div><font =
face=3D"Courier New">IETF 74 Attendees:<br><br>I am writing to make sure =
that you are aware of the Internet Society (ISOC)&nbsp;briefing panel =
focused on the real world experiences associated with deploying =
IPv6.&nbsp; While not part of the IETF agenda, the panel session will be =
coincident with the IETF 74 meeting in San Francisco.<br><br>"The Seven =
Stages of IPv6"&nbsp;will review the opportunities of IPv6 from the =
perspective of network citizens in the seven stages of dealing <br>with =
massive change.&nbsp;&nbsp;Leslie Daigle, ISOC Chief Internet Technology =
Officer, will moderate the panel.&nbsp; The panel session&nbsp;will be =
held at the Hilton San Francisco on Tuesday,&nbsp;March&nbsp;24th =
at&nbsp;11:30am - 12:45pm PDT.<br><br>Lunch will be provided; however, =
seating is limited and pre-registration is required before<br>Wednesday, =
March 18th at 5:00pm PDT.&nbsp; Registration is free.&nbsp; The panel =
session is open <br>to interested IETF participants, ISOC members, and =
journalists.<br><br>Register online at:<br></font><a class=3D"autolink" =
href=3D"http://www.isoc.org/isoc/conferences/registration/20090324.shtml" =
target=3D"_blank"><font face=3D"Courier =
New">http://www.isoc.org/isoc/conferences/registration/20090324.shtml</fon=
t></a> </blockquote></div><br></body></html>=

--Apple-Mail-1-872646857--

From mary.barnes@nortel.com  Thu Mar 12 11:08:55 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Hans Erik van Elburg" <hanserik.van.elburg@ericsson.com>, <sip@ietf.org>
Subject: [Sip] Issue 3 - two tags (was RE: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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I think there is another solution proposal that we should pursue for
issue 3 identified in the email summary sent by Shida earlier this week
- I've snipped all the text except for that issue and issue 2 in the
summary below.  I do think the discussion has confounded the two issues
and I think it ought to be clarified that if we agree to the two tags
per issue 3, that the options for the terminology MUST change. And,
perhaps this is why the other discussion threads aren't progressing
well.
=20
So, Hans Erick, I would like clarification on as to your view on the
solution:
=20
In general, ISTM that one of the issues we're having in this terminology
thread is that you are considering the solution to be that the
hi-entries are tagged as they=20
are added to the request.  And, just for clarification from my
perspective, that is NOT the solution in either 4244bis or target-uri.=20
=20
I don't view it as a problem necessarily if you'd like to pursue an
alternate solution, we just need to be clear that's what you're
advocating. In that case, I would agree the "retarget" is not an
appropriate tag for the entries, as you don't know at the point in time
that you add an hi-entry that it will be retargeted. In the case of this
approach to the solution, you do have to alter the mechanism for
determining the hi-entry that your services would want to use. You would
need to skip the most recent/last hi-entry (as that would contain the
same as the Request-URI in the incoming request), since it would be
tagged if you use this solution approach. Thus, you'd have to take the
next hi-entry that you find that has the desired "tag".  I will posit
that we could accomplish this solution without any changes to 4244 by
just registering new SIP URI parameter(s).   And,  that's not to say we
shouldn't do a 4244bis as there are other issues to address in that
document, but that allows the target-uri to describe a solution that
makes use of 4244bis "as is".
=20
So, feedback on this specific point (from Hans Erik and others) would be
appreciated.=20
=20
Mary.
=20
=20
=20

________________________________

From: Shida Schubert [mailto:shida@ntt-at.com]=20
Sent: Wednesday, March 11, 2009 1:13 AM
To: sip@ietf.org List
Cc: Barnes, Mary (RICH2:AR00)
Subject: Fwd: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt=20



We have submitted the updated target-uri draft based on the comments=20
submitted to the list and comments received at IETF73.=20

I have taken over as editor as Jonathan didn't have the cycles to=20
update the draft, with Francois, Christer and Hans Erick as additional=20
co-authors and great deal of help from Mary. =20
=20
The following summarizes the changes made to the target-uri document
1. Added use-case for toll-free number back
2. Added definition of "retarget" operation.=20
3. Removed a reference to URN
4. Added a text discussing the difference to P-Called-Party-Id
5. Changed parameter name from "target" to "istarget"
=20
Note, that the target-uri document still contains the normative text for
the
History-Info header.
=20
In addition, Mary (with Francois as co-author) has submitted a
rfc4244bis, with the following changes:
1. Incorporated the normative aspects of the target-uri document
into the existing normative text in RFC 4244 - the functionality is=20
virtually identical (as is some of the text) as the HI based solution=20
described in the target-uri document.  It's important that the solution
be integrated into RFC 4244 as it MUST work and be based on the
normative
aspects of RFC 4244.
2.  Added the use cases from target-uri the the summary
in the overview of rfc4244bis.
3. Added an additional requirement to capture the "target-uri"
information.
4. Fixed an error in the RFC 4244 ABNF and added "retarget" parameter.
5. Added a more simplified example.
=20
=20
We had some very long offline exchanges as to the best way forward and
remaining work for both documents.
=20
Some of the issues identified are:

::Issues::
   1. Should we remove the normative text from target-uri and progress
       4244bis along with the target-uri document to meet the chartered
      SIP WG milestone?=20
=20
=20
   2. Name of the parameter.=20
      At the last meeting, parameter "target" was said inappropriate=20
      because voicemail-uri spec already defines a parameter called=20
      "target" which also can be found in hi-entry, thus potentially=20
      causing confusion.
    =20
      Currently the target-uri draft uses "istarget" and 4244bis uses
      "retarget"  but we could never come to=20
      a consensus on what name is appropriate.  Other suggestions have
      included the following:
      "target-identity" (someone didn't like that "identity" is also a
SIP header)
      "reg-uri"  (can be paired with "mapped-uri" for item 3 below)
      "aor"
      "jibberish"=20
      etc.
=20
      One reason this is so difficult relates to the problem statement
in target-uri in that
      RFC 3261 doesn't differentiate the mechanism by which the new
      (target) Request-URI is selected.  Another issue is that some of
the terminology in
      RFC 3261 is overloaded - e.g., "forwarding" refers both to a Proxy
      which does not have responsibility for the domain of the
request-URI
      in the incoming request, thus the proxy just "forwards" the
request to
      the next hop AND "forwarding" is used to describe the process
whereby
      the outgoing request is built and "forwarded" to the next hop at
which
      point the proxy does not know how the new request-uri was
selected.
      RFC 4244 has attempted to clarify the terms and attempts to use
"forward"=20
      in the context of the former situation and "retarget" for the case
whereby
      a proxy is responsible for the domain and thus can use a number of

      mechanism to select the new target for the request - e.g., a
REGISTRAR,=20
      configured data, etc. =20

 3.  Related to the last point in item 2 above, it has been proposed
that
      we differentiate the hi-entries even more by defining separate
parameters
     for registered and configured/mapped contacts.=20
     Currently when the R-URI is translated to a URI which is either
derived=20
     from location service lookup(registered by UA) or from mapping=20
     table, there is no differentiation as to how the URI was derived
once it is
     added to the list of potential targets.
    =20
     The general consensus of the authors of the two documents was that
it may=20
     be useful for some services to have the hi-entries tagged with the
     more specific information.=20
=20
     And, of course, this gets us into another naming contest. In the
end, the naming
     is not so important as long as the term isn't too overloaded and it
is defined
     precisely in the document(s).
=20
We would appreciate WG feedback on these issues and any other comments
on
the two documents prior to IETF-74.
=20
Regards,
Shida and Mary.



Begin forwarded message:

	From: Internet-Drafts@ietf.org
	Date: March 10, 2009 2:30:01 AM JST
	To: i-d-announce@ietf.org
	Subject: I-D
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt=20
	Reply-To: internet-drafts@ietf.org

	A New Internet-Draft is available from the on-line
Internet-Drafts=20
	directories.
=09
=09
	Title : Delivery of Request-URI Targets to User Agents
	Author(s) : J. Rosenberg, H. van Elburg, C. Holmberg, F. Audet,
S. Schubert
	Filename : draft-rosenberg-sip-target-uri-delivery-01.txt
	Pages : 16
	Date : 2009-3-9
=09
	When a Session Initiation Protocol (SIP) proxy receives a
request
	  targeted at a URI identifying a user or resource it is
responsible
	  for, the proxy translates the URI to a registered or
configured
	  contact URI of an agent representing that user or resource.
In the
	  process, the original URI is removed from the request.
Numerous use
	  cases have arisen which require this information to be
delivered to
	  the user agent.  This document describes these use cases and
defines
	  an extension to the History-Info header field which allows it
to be
	  used to support those cases.
=09
	A URL for this Internet-Draft is:
=09
http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri-deliv
ery-01.txt
=09
	Internet-Drafts are also available by anonymous FTP at:
	ftp://ftp.ietf.org/internet-drafts/
=09
	Below is the data which will enable a MIME compliant mail reader
	implementation to automatically retrieve the ASCII version of
the
	Internet-Draft.
=09

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	_______________________________________________
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<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D578425417-12032009>I=20
think there is another solution proposal that we should pursue for issue =
3=20
identified in the email summary sent by Shida earlier this week - I've =
snipped=20
all the text except for that issue and issue 2 in the summary =
below.&nbsp; I do=20
think the discussion has confounded the two issues and I think it ought =
to be=20
clarified that if we agree to the two tags per issue 3, that the options =
for the=20
terminology MUST change. And, perhaps this is why the other discussion =
threads=20
aren't progressing well.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D578425417-12032009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D578425417-12032009>So,=20
Hans Erick, I would like clarification on as to your view on the=20
solution:</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D578425417-12032009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D578425417-12032009>In=20
general, ISTM that one of the issues we're having in this terminology =
thread is=20
that you are considering the solution to be that the hi-entries are =
tagged as=20
they </SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D578425417-12032009>are=20
added to the request.&nbsp; And, just for clarification from my =
perspective,=20
that is NOT the solution in either 4244bis or target-uri. =
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D578425417-12032009></SPAN></FONT><FONT face=3DArial =
color=3D#0000ff=20
size=3D2><SPAN class=3D578425417-12032009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D578425417-12032009>I=20
don't view it as a problem necessarily if you'd like to pursue an =
alternate=20
solution, we just need to be clear that's what you're advocating. In =
that case,=20
I would agree the "retarget" is not an appropriate tag for the entries, =
as you=20
don't know at the point in time that you add an hi-entry that it will be =

retargeted. In the case of this approach to the solution, you do have to =
alter=20
the mechanism for determining the hi-entry that your services would want =
to use.=20
You would need to skip the most recent/last hi-entry (as that =
would&nbsp;contain=20
the same as the Request-URI in the incoming request), since it would be =
tagged=20
if you use this solution approach. Thus, you'd have to take the next =
hi-entry=20
that you find that has the desired "tag".&nbsp; I will posit that we =
could=20
accomplish this solution without any changes to 4244 by just registering =

new&nbsp;SIP URI parameter(s).&nbsp;&nbsp;&nbsp;And, &nbsp;that's not to =
say we=20
shouldn't do a 4244bis as there are other issues to address in that =
document,=20
but that allows the target-uri to describe a solution that makes use of =
4244bis=20
"as is".</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D578425417-12032009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D578425417-12032009>So,=20
feedback on this specific point (from Hans Erik and others) would be=20
appreciated. </SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D578425417-12032009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D578425417-12032009>Mary.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><BR></DIV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Shida Schubert =
[mailto:shida@ntt-at.com]=20
<BR><B>Sent:</B> Wednesday, March 11, 2009 1:13 AM<BR><B>To:</B> =
sip@ietf.org=20
List<BR><B>Cc:</B> Barnes, Mary (RICH2:AR00)<BR><B>Subject:</B> Fwd: I-D =

ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt =
<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><BR></DIV>
<DIV>We have submitted the updated target-uri draft based on the=20
comments&nbsp;</DIV>
<DIV>submitted to the list and comments received at IETF73.&nbsp;</DIV>
<DIV><BR></DIV>
<DIV>I have taken over as editor as Jonathan didn't have the cycles=20
to&nbsp;</DIV>
<DIV>update the draft, with Francois, Christer and Hans Erick as=20
additional&nbsp;</DIV>
<DIV>co-authors and great deal of help from Mary. &nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>The following summarizes the changes made to the target-uri =
document</DIV>
<DIV>1. Added use-case for toll-free number back</DIV>
<DIV>2. Added definition of "retarget" operation.&nbsp;</DIV>
<DIV>3. Removed a reference to URN</DIV>
<DIV>4. Added a text discussing the difference to =
P-Called-Party-Id</DIV>
<DIV>5. Changed parameter name from "target" to "istarget"</DIV>
<DIV>&nbsp;</DIV>
<DIV>Note, that the target-uri document still contains the normative =
text for=20
the</DIV>
<DIV>History-Info header.</DIV>
<DIV>&nbsp;</DIV>
<DIV>In addition, Mary (with Francois as co-author) has submitted =
a</DIV>
<DIV>rfc4244bis, with the following changes:</DIV>
<DIV>1. Incorporated the normative aspects of the target-uri =
document</DIV>
<DIV>into the existing normative text in RFC 4244 - the functionality=20
is&nbsp;</DIV>
<DIV>virtually identical (as is some of the text) as the HI based=20
solution&nbsp;</DIV>
<DIV>described in the target-uri document. &nbsp;It's important that the =

solution</DIV>
<DIV>be integrated into RFC 4244 as it MUST work and be based on the=20
normative</DIV>
<DIV>aspects of RFC 4244.</DIV>
<DIV>2. &nbsp;Added the use cases from target-uri the the summary</DIV>
<DIV>in the overview of rfc4244bis.</DIV>
<DIV>3. Added an additional requirement to capture the "target-uri"=20
information.</DIV>
<DIV>4. Fixed an error in the RFC 4244 ABNF and added "retarget"=20
parameter.</DIV>
<DIV>5. Added a more simplified example.</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>We had some very long offline exchanges as to the best way forward=20
and</DIV>
<DIV>remaining work for both documents.</DIV>
<DIV>&nbsp;</DIV>
<DIV>Some of the issues identified are:</DIV>
<DIV><BR></DIV>
<DIV>::Issues::</DIV>
<DIV>&nbsp;&nbsp; 1. Should we remove the normative text from target-uri =
and=20
progress</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp; 4244bis along with the target-uri =
document to=20
meet the chartered</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;SIP WG milestone?&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;&nbsp; 2. Name of the parameter.&nbsp;</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;At the last meeting, parameter "target" =
was said=20
inappropriate&nbsp;</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;because voicemail-uri spec already =
defines a=20
parameter called&nbsp;</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;"target" which also can be found in =
hi-entry,=20
thus potentially&nbsp;</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;causing confusion.</DIV>
<DIV>&nbsp;&nbsp; &nbsp;&nbsp;</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;Currently the target-uri draft uses =
"istarget"=20
and 4244bis uses</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;"retarget" &nbsp;but we could never come=20
to&nbsp;</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;a consensus on what name is appropriate.=20
&nbsp;Other suggestions have</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;included the following:</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;"target-identity" (someone didn't like =
that=20
"identity" is also a SIP header)</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;"reg-uri" &nbsp;(can be paired with =
"mapped-uri"=20
for item 3 below)</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;"aor"</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;"jibberish"&nbsp;</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;etc.</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;One reason this is so difficult relates =
to the=20
problem statement in target-uri in that</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;RFC 3261 doesn't differentiate the =
mechanism by=20
which the new</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;(target) Request-URI is selected. =
&nbsp;Another=20
issue is that some of the terminology in</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;RFC 3261 is overloaded - e.g., =
"forwarding"=20
refers both to a Proxy</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;which does not have responsibility for =
the domain=20
of the request-URI</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;in the incoming request, thus the proxy =
just=20
"forwards" the request to</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;the next hop AND "forwarding" is used to =
describe=20
the process whereby</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;the outgoing request is built and =
"forwarded" to=20
the next hop at which</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;point the proxy does not know how the new =

request-uri was selected.</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;RFC 4244 has attempted to clarify the =
terms and=20
attempts to use "forward"&nbsp;</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;in the context of the former situation =
and=20
"retarget" for the case whereby</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;a proxy is responsible for the domain and =
thus=20
can use a number of&nbsp;</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;mechanism to select the new target for =
the=20
request - e.g., a REGISTRAR,&nbsp;</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;configured data, etc. &nbsp;</DIV>
<DIV><BR></DIV>
<DIV>&nbsp;3. &nbsp;Related to the last point in item 2 above, it has =
been=20
proposed that</DIV>
<DIV>&nbsp;&nbsp; &nbsp; &nbsp;we differentiate the hi-entries even more =
by=20
defining separate parameters</DIV>
<DIV>&nbsp;&nbsp; &nbsp; for registered and configured/mapped=20
contacts.&nbsp;</DIV>
<DIV>&nbsp;&nbsp; &nbsp; Currently when the R-URI is translated to a URI =
which=20
is either derived&nbsp;</DIV>
<DIV>&nbsp;&nbsp; &nbsp; from location service lookup(registered by UA) =
or from=20
mapping&nbsp;</DIV>
<DIV>&nbsp;&nbsp; &nbsp; table, there is no differentiation as to how =
the URI=20
was derived once it is</DIV>
<DIV>&nbsp;&nbsp; &nbsp; added to the list of potential targets.</DIV>
<DIV>&nbsp;&nbsp; &nbsp;&nbsp;</DIV>
<DIV>&nbsp;&nbsp; &nbsp; The general consensus of the authors of the two =

documents was that it may&nbsp;</DIV>
<DIV>&nbsp;&nbsp; &nbsp; be useful for some services to have the =
hi-entries=20
tagged with the</DIV>
<DIV>&nbsp;&nbsp; &nbsp; more specific information.&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;&nbsp; &nbsp; And, of course, this gets us into another =
naming=20
contest. In the end, the naming</DIV>
<DIV>&nbsp;&nbsp; &nbsp; is not so important as long as the term isn't =
too=20
overloaded and it is defined</DIV>
<DIV>&nbsp;&nbsp; &nbsp; precisely in the document(s).</DIV>
<DIV>&nbsp;</DIV>
<DIV>We would appreciate WG feedback on these issues and any other =
comments=20
on</DIV>
<DIV>the two documents prior to IETF-74.</DIV>
<DIV>&nbsp;</DIV>
<DIV>Regards,</DIV>
<DIV>Shida and Mary.</DIV>
<DIV><BR></DIV>
<DIV><BR></DIV>
<DIV>
<DIV>
<DIV>
<DIV><BR class=3DApple-interchange-newline>Begin forwarded =
message:</DIV>
<BLOCKQUOTE type=3D"cite">
  <DIV>
  <DIV style=3D"MARGIN: 0px"><FONT style=3D"FONT: 12px Helvetica; COLOR: =
rgb(0,0,0)"=20
  face=3DHelvetica color=3D#000000 =
size=3D3><B>From:&nbsp;</B></FONT><FONT=20
  style=3D"FONT: 12px Helvetica" face=3DHelvetica size=3D3><A=20
  =
href=3D"mailto:Internet-Drafts@ietf.org">Internet-Drafts@ietf.org</A></FO=
NT></DIV>
  <DIV style=3D"MARGIN: 0px"><FONT style=3D"FONT: 12px Helvetica; COLOR: =
rgb(0,0,0)"=20
  face=3DHelvetica color=3D#000000 =
size=3D3><B>Date:&nbsp;</B></FONT><FONT=20
  style=3D"FONT: 12px Helvetica" face=3DHelvetica size=3D3>March 10, =
2009 2:30:01 AM=20
  JST</FONT></DIV>
  <DIV style=3D"MARGIN: 0px"><FONT style=3D"FONT: 12px Helvetica; COLOR: =
rgb(0,0,0)"=20
  face=3DHelvetica color=3D#000000 size=3D3><B>To:&nbsp;</B></FONT><FONT =

  style=3D"FONT: 12px Helvetica" face=3DHelvetica size=3D3><A=20
  =
href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</A></FONT></D=
IV>
  <DIV style=3D"MARGIN: 0px"><FONT style=3D"FONT: 12px Helvetica; COLOR: =
rgb(0,0,0)"=20
  face=3DHelvetica color=3D#000000 =
size=3D3><B>Subject:&nbsp;</B></FONT><FONT=20
  style=3D"FONT: 12px Helvetica" face=3DHelvetica size=3D3><B>I-D=20
  =
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt&nbsp;</B></FONT></D=
IV>
  <DIV style=3D"MARGIN: 0px"><FONT style=3D"FONT: 12px Helvetica; COLOR: =
rgb(0,0,0)"=20
  face=3DHelvetica color=3D#000000 =
size=3D3><B>Reply-To:&nbsp;</B></FONT><FONT=20
  style=3D"FONT: 12px Helvetica" face=3DHelvetica size=3D3><A=20
  =
href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</A></FO=
NT></DIV>
  <DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px"><BR></DIV></DIV>
  <DIV>A New Internet-Draft is available from the on-line=20
  Internet-Drafts&nbsp;<BR>directories.<BR><BR><BR><SPAN =
class=3DApple-tab-span=20
  style=3D"WHITE-SPACE: pre"></SPAN>Title<SPAN class=3DApple-tab-span=20
  style=3D"WHITE-SPACE: pre"> </SPAN><SPAN class=3DApple-tab-span=20
  style=3D"WHITE-SPACE: pre"></SPAN>: Delivery of Request-URI Targets to =
User=20
  Agents<BR><SPAN class=3DApple-tab-span=20
  style=3D"WHITE-SPACE: pre"></SPAN>Author(s)<SPAN =
class=3DApple-tab-span=20
  style=3D"WHITE-SPACE: pre"> </SPAN>: J. Rosenberg, H. van Elburg, C. =
Holmberg,=20
  F. Audet, S. Schubert<BR><SPAN class=3DApple-tab-span=20
  style=3D"WHITE-SPACE: pre"></SPAN>Filename<SPAN class=3DApple-tab-span =

  style=3D"WHITE-SPACE: pre"> </SPAN>:=20
  draft-rosenberg-sip-target-uri-delivery-01.txt<BR><SPAN =
class=3DApple-tab-span=20
  style=3D"WHITE-SPACE: pre"></SPAN>Pages<SPAN class=3DApple-tab-span=20
  style=3D"WHITE-SPACE: pre"> </SPAN><SPAN class=3DApple-tab-span=20
  style=3D"WHITE-SPACE: pre"></SPAN>: 16<BR><SPAN class=3DApple-tab-span =

  style=3D"WHITE-SPACE: pre"></SPAN>Date<SPAN class=3DApple-tab-span=20
  style=3D"WHITE-SPACE: pre"> </SPAN><SPAN class=3DApple-tab-span=20
  style=3D"WHITE-SPACE: pre"></SPAN>: 2009-3-9<BR><SPAN =
class=3DApple-tab-span=20
  style=3D"WHITE-SPACE: pre"></SPAN><BR>When a Session Initiation =
Protocol (SIP)=20
  proxy receives a request<BR>&nbsp;&nbsp;targeted at a URI identifying =
a user=20
  or resource it is responsible<BR>&nbsp;&nbsp;for, the proxy translates =
the URI=20
  to a registered or configured<BR>&nbsp;&nbsp;contact URI of an agent=20
  representing that user or resource. &nbsp;In =
the<BR>&nbsp;&nbsp;process, the=20
  original URI is removed from the request. &nbsp;Numerous=20
  use<BR>&nbsp;&nbsp;cases have arisen which require this information to =
be=20
  delivered to<BR>&nbsp;&nbsp;the user agent. &nbsp;This document =
describes=20
  these use cases and defines<BR>&nbsp;&nbsp;an extension to the =
History-Info=20
  header field which allows it to be<BR>&nbsp;&nbsp;used to support =
those=20
  cases.<BR><BR>A URL for this Internet-Draft is:<BR><A=20
  =
href=3D"http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-ur=
i-delivery-01.txt">http://www.ietf.org/internet-drafts/draft-rosenberg-si=
p-target-uri-delivery-01.txt</A><BR><BR>Internet-Drafts=20
  are also available by anonymous FTP at:<BR><A=20
  =
href=3D"ftp://ftp.ietf.org/internet-drafts/">ftp://ftp.ietf.org/internet-=
drafts/</A><BR><BR>Below=20
  is the data which will enable a MIME compliant mail =
reader<BR>implementation=20
  to automatically retrieve the ASCII version of=20
the<BR>Internet-Draft.<BR></DIV></BLOCKQUOTE></DIV></DIV>Content-Type:=20
text/plain&lt;BR&gt;Content-ID: &amp;lt;<A=20
href=3D"mailto:2009-3-9102756.I-D@ietf.org">2009-3-9102756.I-D@ietf.org</=
A>&amp;gt;&lt;BR&gt;&lt;BR&gt;
<DIV>
<DIV>
<BLOCKQUOTE type=3D"cite">
  <DIV>_______________________________________________<BR>I-D-Announce =
mailing=20
  list<BR><A =
href=3D"mailto:I-D-Announce@ietf.org">I-D-Announce@ietf.org</A><BR><A=20
  =
href=3D"https://www.ietf.org/mailman/listinfo/i-d-announce">https://www.i=
etf.org/mailman/listinfo/i-d-announce</A><BR>Internet-Draft=20
  directories:&nbsp;<A=20
  =
href=3D"http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html<=
/A><BR>or&nbsp;<A=20
  =
href=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/iet=
f/1shadow-sites.txt</A></DIV></BLOCKQUOTE></DIV></DIV></DIV></BODY></HTML=
>

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Yes, we can plan on doing so in the revisions, once we actually agree to
where we're going with both docs in terms of supporting the target-uri
solution requirements.

Mary.=20

-----Original Message-----
From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]=20
Sent: Thursday, March 12, 2009 11:41 AM
To: Barnes, Mary (RICH2:AR00); Dean Willis; Shida Schubert
Cc: sip@ietf.org
Subject: RE: [Sip] Fwd:
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt

Which is where I was coming from on my offline email to you, i.e. that
we should start recording our position of any changes required to RFC
3261 in an appendix. That way we know what changes we need to make to
RFC 3261 when we get to make them, even if they are only for alignment
of terminology. We can decide later whether the appendix gets published
with the update to either RFC 4244 or target-URI or some combined
document.

Keith=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of=20
> Mary Barnes
> Sent: Wednesday, March 11, 2009 4:27 PM
> To: Dean Willis; Shida Schubert
> Cc: sip@ietf.org
> Subject: Re: [Sip] Fwd: I-D
> ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>=20
> I don't think it should cause so much concern as I think it just=20
> requires clarifying text and we just need to agree to more precise
> terms: e.g., qualifying the use of "forward" with either "next hop=20
> forwarding" or "request forwarding" for this particular case. As long=20
> as we can be precise in 4244bis, then that should suffice for now, but

> whenever 3261 is updated or bised, then I think the clarifications=20
> should be included.
>=20
> Mary.=20
>=20
> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Wednesday, March 11, 2009 1:38 AM
> To: Shida Schubert
> Cc: sip@ietf.org List; Barnes, Mary (RICH2:AR00)
> Subject: Re: [Sip] Fwd: I-D
> ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>=20
>=20
> On Mar 11, 2009, at 1:13 AM, Shida Schubert wrote:
>=20
> >
> >       One reason this is so difficult relates to the
> problem statement
>=20
> > in target-uri in that
> >       RFC 3261 doesn't differentiate the mechanism by which the new
> >       (target) Request-URI is selected.  Another issue is
> that some of
>=20
> > the terminology in
> >       RFC 3261 is overloaded - e.g., "forwarding" refers both to a=20
> > Proxy
> >       which does not have responsibility for the domain of the=20
> > request-URI
> >       in the incoming request, thus the proxy just "forwards" the=20
> > request to
> >       the next hop AND "forwarding" is used to describe the process=20
> > whereby
> >       the outgoing request is built and "forwarded" to the
> next hop at
>=20
> > which
> >       point the proxy does not know how the new request-uri was=20
> > selected.
> >       RFC 4244 has attempted to clarify the terms and
> attempts to use
> > "forward"
> >       in the context of the former situation and "retarget" for the=20
> > case whereby
> >       a proxy is responsible for the domain and thus can
> use a number
> > of
> >       mechanism to select the new target for the request - e.g., a=20
> > REGISTRAR,
> >       configured data, etc.
> >
>=20
> Thanks, Shida.
>=20
> I personally would not be averse to an Essential Correction that=20
> corrects the terminology in RFC 3261, but I suspect that will make=20
> many heads spin around elsewhere in the WG.
>=20
> --
> Dean
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> sipping@ietf.org for new developments on the application of sip
>=20

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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Hans Erik van Elburg" <hanserik.van.elburg@ericsson.com>, <sip@ietf.org>
Subject: [Sip] Snipped/text version- Issue 3 - two tags (was RE: I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Sorry Keith ... I forgot to snip and transform to text.=20

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Barnes, Mary (RICH2:AR00)
Sent: Thursday, March 12, 2009 1:10 PM
To: Hans Erik van Elburg; sip@ietf.org
Subject: [Sip] Issue 3 - two tags (was RE:
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


I think there is another solution proposal that we should pursue for
issue 3 identified in the email summary sent by Shida earlier this week
- I've snipped all the text except for that issue and issue 2 in the
summary below.  I do think the discussion has confounded the two issues
and I think it ought to be clarified that if we agree to the two tags
per issue 3, that the options for the terminology MUST change. And,
perhaps this is why the other discussion threads aren't progressing
well.
=20
So, Hans Erick, I would like clarification on as to your view on the
solution:
=20
In general, ISTM that one of the issues we're having in this terminology
thread is that you are considering the solution to be that the
hi-entries are tagged as they=20
are added to the request.  And, just for clarification from my
perspective, that is NOT the solution in either 4244bis or target-uri.=20
=20
I don't view it as a problem necessarily if you'd like to pursue an
alternate solution, we just need to be clear that's what you're
advocating. In that case, I would agree the "retarget" is not an
appropriate tag for the entries, as you don't know at the point in time
that you add an hi-entry that it will be retargeted. In the case of this
approach to the solution, you do have to alter the mechanism for
determining the hi-entry that your services would want to use. You would
need to skip the most recent/last hi-entry (as that would contain the
same as the Request-URI in the incoming request), since it would be
tagged if you use this solution approach. Thus, you'd have to take the
next hi-entry that you find that has the desired "tag".  I will posit
that we could accomplish this solution without any changes to 4244 by
just registering new SIP URI parameter(s).   And,  that's not to say we
shouldn't do a 4244bis as there are other issues to address in that
document, but that allows the target-uri to describe a solution that
makes use of 4244bis "as is".
=20
So, feedback on this specific point (from Hans Erik and others) would be
appreciated.=20
=20
Mary.
=20
=20
=20

________________________________

From: Shida Schubert [mailto:shida@ntt-at.com]=20
Sent: Wednesday, March 11, 2009 1:13 AM
To: sip@ietf.org List
Cc: Barnes, Mary (RICH2:AR00)
Subject: Fwd: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt=20



We have submitted the updated target-uri draft based on the comments=20
submitted to the list and comments received at IETF73.=20

....snipped...
   2. Name of the parameter.=20
      At the last meeting, parameter "target" was said inappropriate=20
      because voicemail-uri spec already defines a parameter called=20
      "target" which also can be found in hi-entry, thus potentially=20
      causing confusion.
    =20
      Currently the target-uri draft uses "istarget" and 4244bis uses
      "retarget"  but we could never come to=20
      a consensus on what name is appropriate.  Other suggestions have
      included the following:
      "target-identity" (someone didn't like that "identity" is also a
SIP header)
      "reg-uri"  (can be paired with "mapped-uri" for item 3 below)
      "aor"
      "jibberish"=20
      etc.
=20
      One reason this is so difficult relates to the problem statement
in target-uri in that
      RFC 3261 doesn't differentiate the mechanism by which the new
      (target) Request-URI is selected.  Another issue is that some of
the terminology in
      RFC 3261 is overloaded - e.g., "forwarding" refers both to a Proxy
      which does not have responsibility for the domain of the
request-URI
      in the incoming request, thus the proxy just "forwards" the
request to
      the next hop AND "forwarding" is used to describe the process
whereby
      the outgoing request is built and "forwarded" to the next hop at
which
      point the proxy does not know how the new request-uri was
selected.
      RFC 4244 has attempted to clarify the terms and attempts to use
"forward"=20
      in the context of the former situation and "retarget" for the case
whereby
      a proxy is responsible for the domain and thus can use a number of

      mechanism to select the new target for the request - e.g., a
REGISTRAR,=20
      configured data, etc. =20

 3.  Related to the last point in item 2 above, it has been proposed
that
      we differentiate the hi-entries even more by defining separate
parameters
     for registered and configured/mapped contacts.=20
     Currently when the R-URI is translated to a URI which is either
derived=20
     from location service lookup(registered by UA) or from mapping=20
     table, there is no differentiation as to how the URI was derived
once it is
     added to the list of potential targets.
    =20
     The general consensus of the authors of the two documents was that
it may=20
     be useful for some services to have the hi-entries tagged with the
     more specific information.=20
=20
     And, of course, this gets us into another naming contest. In the
end, the naming
     is not so important as long as the term isn't too overloaded and it
is defined
     precisely in the document(s).
=20
We would appreciate WG feedback on these issues and any other comments
on
the two documents prior to IETF-74.
=20
Regards,
Shida and Mary.



Begin forwarded message:

	From: Internet-Drafts@ietf.org
	Date: March 10, 2009 2:30:01 AM JST
	To: i-d-announce@ietf.org
	Subject: I-D
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt=20
	Reply-To: internet-drafts@ietf.org

	A New Internet-Draft is available from the on-line
Internet-Drafts=20
	directories.
=09
=09
	Title : Delivery of Request-URI Targets to User Agents
	Author(s) : J. Rosenberg, H. van Elburg, C. Holmberg, F. Audet,
S. Schubert
	Filename : draft-rosenberg-sip-target-uri-delivery-01.txt
	Pages : 16
	Date : 2009-3-9
=09
	When a Session Initiation Protocol (SIP) proxy receives a
request
	  targeted at a URI identifying a user or resource it is
responsible
	  for, the proxy translates the URI to a registered or
configured
	  contact URI of an agent representing that user or resource.
In the
	  process, the original URI is removed from the request.
Numerous use
	  cases have arisen which require this information to be
delivered to
	  the user agent.  This document describes these use cases and
defines
	  an extension to the History-Info header field which allows it
to be
	  used to support those cases.
=09
	A URL for this Internet-Draft is:
=09
http://www.ietf.org/internet-drafts/draft-rosenberg-sip-target-uri-deliv
ery-01.txt
=09
	Internet-Drafts are also available by anonymous FTP at:
	ftp://ftp.ietf.org/internet-drafts/
=09
	Below is the data which will enable a MIME compliant mail reader
	implementation to automatically retrieve the ASCII version of
the
	Internet-Draft.
=09

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References: <899F1518-6872-442E-AD3E-D3F343A605DC@ntt-at.com><9ae56b1e0903110158m1e6fb07o66110501e8581971@mail.gmail.com><1ECE0EB50388174790F9694F77522CCF1CCA681D@zrc2hxm0.corp.nortel.com><49B82FB4.4030002@gmail.com><1ECE0EB50388174790F9694F77522CCF1CCF97F5@zrc2hxm0.corp.nortel.com><49B84C55.5070107@gmail.com><1ECE0EB50388174790F9694F77522CCF1CCF9A06@zrc2hxm0.corp.nortel.com><9ae56b1e0903120134i76a6a92brb183205cd66e841f@mail.gmail.com><0D5F89FAC29E2C41B98A6A762007F5D001A27AF1@GBNTHT12009MSX.gb002.siemens.net><1ECE0EB50388174790F9694F77522CCF1CCF9DC8@zrc2hxm0.corp.nortel.com><9ae56b1e0903120754m66c0b7c5y225ce0f9193dba28@mail.gmail.com> <1ECE0EB50388174790F9694F77522CCF1CD4B01D@zrc2hxm0.corp.nortel.com>
From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Mary Barnes" <mary.barnes@nortel.com>, "Hans Erik van Elburg" <ietf.hanserik@gmail.com>
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Subject: Re: [Sip] Terminology (was RE: Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Hi Mary,
=20
A couple of question:
=20
1. Does the 4244bis definition of "retarget" cover the freephone case,
when the R-URI is replaced with the AoR of B - not with the contact of B
- no matter whether it's done based on location service, configuration
or whatever?
=20
2. Related to the first question, when you say "...and thus chaning the
target of the request", what is the defintion of "target"?
=20
Regards,
=20
Christer

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Mary Barnes
Sent: Thursday, March 12, 2009 5:20 PM
To: Hans Erik van Elburg
Cc: sip@ietf.org; Elwell, John
Subject: Re: [Sip] Terminology (was RE:
Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


When I use "retarget" I am using the definition that I put into 4244bis.
So, if you could ignore the tag or change it to 'iscontact' or whatever,
what would you change in that definition? Also, consider that if you
were to change everyplace that I use
"retarget"/"retargeted"/"retargeting" in 4244bis with 'iscontact' the
text would not make sense. So, even if we change the name of the tag, we
still need to understand the concept or define another term that means
that the request-uri in the incoming request (which should be the same
as the last hi-entry in the request) is being chnaged due to service
lookup, configuration, blah, blah, blah and that to meet the
requirements in the target-uri document, we want to tag this entry so
that this request-uri is not "lost" - i.e., this is the one the services
identified in the target-uri document care about.=20
=20
So, right now, here's what's in 4244bis:
=20
   The term "retarget" is used in this document to refer to the process
   of a Proxy Server/User Agent Client (UAC) changing a Uniform Resource
   Identifier (URI) in a request based on a lookup in a location service
   and thus changing the target of the request.
=20
We have previously discussed that we do need to expand the "based on a
lookup in a location service" to include configuration, etc., so let's
not debate that.=20
=20
But, what explicitly is the concern with that definition?
=20
Mary.=20
=20

________________________________

From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
Sent: Thursday, March 12, 2009 9:55 AM
To: Barnes, Mary (RICH2:AR00)
Cc: Elwell, John; sip@ietf.org
Subject: Re: [Sip] Terminology (was RE: Fwd:
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


I'm fine with discussing implementation if everyone is meaning the same
if they use the same words.

As  to the "retarget" term such does not seem to be the case.

/Hans Erik van Elburg



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<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D732481218-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Hi Mary,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D732481218-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D732481218-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>A couple of question:</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D732481218-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D732481218-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>1. Does the 4244bis definition of "retarget" =
cover the=20
freephone case, when the R-URI is replaced with the AoR of B - not with =
the=20
contact of B - no matter whether it's done based on location service,=20
configuration or whatever?</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D732481218-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D732481218-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>2. Related to the first question, when you say =
"...and thus=20
chaning the target of the request", what is the defintion of=20
"target"?</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D732481218-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D732481218-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D732481218-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D732481218-12032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Christer</FONT></SPAN></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Mary =
Barnes<BR><B>Sent:</B>=20
Thursday, March 12, 2009 5:20 PM<BR><B>To:</B> Hans Erik van=20
Elburg<BR><B>Cc:</B> sip@ietf.org; Elwell, John<BR><B>Subject:</B> Re: =
[Sip]=20
Terminology (was RE:=20
Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR></FONT><B=
R></DIV>
<DIV></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D421131315-12032009>When I=20
use "retarget" I am using the definition that I put into 4244bis.&nbsp; =
So, if=20
you could ignore the tag or change it to 'iscontact' or whatever, what =
would you=20
change in that definition? Also, consider that if you were to change =
everyplace=20
that I use "retarget"/"retargeted"/"retargeting" in 4244bis with =
'iscontact' the=20
text would not make sense. So, even if we change the name of the tag, we =
still=20
need to understand the concept or define another term that means that =
the=20
request-uri in the incoming request (which should be the same as the =
last=20
hi-entry in the request) is being chnaged due to service lookup, =
configuration,=20
blah, blah, blah and that to meet the requirements in the target-uri =
document,=20
we want to tag this entry so that this request-uri is not "lost" - i.e., =
this is=20
the one the services identified in the target-uri document care about.=20
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D421131315-12032009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D421131315-12032009>So,=20
right now, here's what's in 4244bis:</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D421131315-12032009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D421131315-12032009>&nbsp;&nbsp; The term "retarget" is used in =
this=20
document to refer to the process<BR>&nbsp;&nbsp; of a Proxy Server/User =
Agent=20
Client (UAC) changing a Uniform Resource<BR>&nbsp;&nbsp; Identifier =
(URI) in a=20
request based on a lookup in a location service</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D421131315-12032009>&nbsp;&nbsp; and thus changing the target of =
the=20
request.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D421131315-12032009>We=20
have previously discussed that we do need to expand the "based on a =
lookup in a=20
location service" to include configuration, etc., so let's not debate =
that.=20
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D421131315-12032009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D421131315-12032009>But,=20
what explicitly is the concern with that definition?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D421131315-12032009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D421131315-12032009>Mary.=20
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><BR></DIV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Hans Erik van Elburg=20
[mailto:ietf.hanserik@gmail.com] <BR><B>Sent:</B> Thursday, March 12, =
2009 9:55=20
AM<BR><B>To:</B> Barnes, Mary (RICH2:AR00)<BR><B>Cc:</B> Elwell, John;=20
sip@ietf.org<BR><B>Subject:</B> Re: [Sip] Terminology (was RE: Fwd:=20
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR></FONT><BR></=
DIV>
<DIV></DIV>I'm fine with discussing implementation if everyone is =
meaning the=20
same if they use the same words.<BR><BR>As&nbsp; to the "retarget" term =
such=20
does not seem to be the case.<BR><BR clear=3Dall>/Hans Erik van=20
Elburg<BR><BR></BODY></HTML>

------_=_NextPart_001_01C9A33E.E9E13BBA--

From mary.barnes@nortel.com  Thu Mar 12 11:33:45 2009
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References: <899F1518-6872-442E-AD3E-D3F343A605DC@ntt-at.com><9ae56b1e0903110158m1e6fb07o66110501e8581971@mail.gmail.com><1ECE0EB50388174790F9694F77522CCF1CCA681D@zrc2hxm0.corp.nortel.com><49B82FB4.4030002@gmail.com><1ECE0EB50388174790F9694F77522CCF1CCF97F5@zrc2hxm0.corp.nortel.com><49B84C55.5070107@gmail.com><1ECE0EB50388174790F9694F77522CCF1CCF9A06@zrc2hxm0.corp.nortel.com><9ae56b1e0903120134i76a6a92brb183205cd66e841f@mail.gmail.com><0D5F89FAC29E2C41B98A6A762007F5D001A27AF1@GBNTHT12009MSX.gb002.siemens.net><1ECE0EB50388174790F9694F77522CCF1CCF9DC8@zrc2hxm0.corp.nortel.com><9ae56b1e0903120754m66c0b7c5y225ce0f9193dba28@mail.gmail.com> <1ECE0EB50388174790F9694F77522CCF1CD4B01D@zrc2hxm0.corp.nortel.com> <CA9998CD4A020D418654FCDEF4E707DF0B167FF7@esealmw113.eemea.ericsson.se>
From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>, "Hans Erik van Elburg" <ietf.hanserik@gmail.com>
Cc: sip@ietf.org, "Elwell, John" <john.elwell@siemens.com>
Subject: Re: [Sip] Terminology (was RE: Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Responses below [MB].

________________________________

From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
Sent: Thursday, March 12, 2009 1:18 PM
To: Barnes, Mary (RICH2:AR00); Hans Erik van Elburg
Cc: sip@ietf.org; Elwell, John
Subject: RE: [Sip] Terminology (was RE:
Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


Hi Mary,
=20
A couple of question:
=20
1. Does the 4244bis definition of "retarget" cover the freephone case,
when the R-URI is replaced with the AoR of B - not with the contact of B
- no matter whether it's done based on location service, configuration
or whatever?
[MB] That was the intention, however, the current text needs to be made
more general (i.e., I was using location service to mean more than
registration based contacts - another thing that's not quite so clear in
3261). [/MB]
=20
2. Related to the first question, when you say "...and thus chaning the
target of the request", what is the defintion of "target"?
[MB] The definition of target matches that used in section 16.5 of 3261.
A target is an entry in the target list, which is populated using
information from a location service, configuration, etc. I can clarify
that as well and as you previously suggested make a more comprehensive
glossary in the more conventional format - e.g., "Target: definition
text, blah, blah, blah...] So, I believe it is the same use of target as
in the target-uri draft. [/MB]
=20
Regards,
=20
Christer

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Mary Barnes
Sent: Thursday, March 12, 2009 5:20 PM
To: Hans Erik van Elburg
Cc: sip@ietf.org; Elwell, John
Subject: Re: [Sip] Terminology (was RE:
Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


When I use "retarget" I am using the definition that I put into 4244bis.
So, if you could ignore the tag or change it to 'iscontact' or whatever,
what would you change in that definition? Also, consider that if you
were to change everyplace that I use
"retarget"/"retargeted"/"retargeting" in 4244bis with 'iscontact' the
text would not make sense. So, even if we change the name of the tag, we
still need to understand the concept or define another term that means
that the request-uri in the incoming request (which should be the same
as the last hi-entry in the request) is being chnaged due to service
lookup, configuration, blah, blah, blah and that to meet the
requirements in the target-uri document, we want to tag this entry so
that this request-uri is not "lost" - i.e., this is the one the services
identified in the target-uri document care about.=20
=20
So, right now, here's what's in 4244bis:
=20
   The term "retarget" is used in this document to refer to the process
   of a Proxy Server/User Agent Client (UAC) changing a Uniform Resource
   Identifier (URI) in a request based on a lookup in a location service
   and thus changing the target of the request.
=20
We have previously discussed that we do need to expand the "based on a
lookup in a location service" to include configuration, etc., so let's
not debate that.=20
=20
But, what explicitly is the concern with that definition?
=20
Mary.=20
=20

________________________________

From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
Sent: Thursday, March 12, 2009 9:55 AM
To: Barnes, Mary (RICH2:AR00)
Cc: Elwell, John; sip@ietf.org
Subject: Re: [Sip] Terminology (was RE: Fwd:
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


I'm fine with discussing implementation if everyone is meaning the same
if they use the same words.

As  to the "retarget" term such does not seem to be the case.

/Hans Erik van Elburg



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Date: Thu, 12 Mar 2009 19:39:13 +0100 (CET)
From: "Nils Ohlmeier" <lists@ohlmeier.org>
To: Victor Pascual =?iso-8859-15?Q?=C1vila?= <victor.pascual.avila@gmail.com>
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Cc: Nils Ohlmeier <lists@ohlmeier.org>, "sip@ietf.org" <sip@ietf.org>
Subject: Re: [Sip] draft-state-sip-relay-attack-00
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> On Wed, Mar 11, 2009 at 2:49 PM, Nils Ohlmeier <lists@ohlmeier.org> wrote:
>> Sure to create a Contact binding via an authenticated REGISTER is one
>> possible solution.
>> But I think this only works because you rely on the fact that REGISTERs
>> are not being retargeted like INVITEs. Which then in the end means you
>> simply can forget about authenticating all other requests, as long as
>> the
>> Contact or IP/port was successfully authenticated via the "one hop"
>> REGISTER method (refer-to: IMS).
>> The advantage of this solution is clearly that it can be easily deployed
>> by hopefully most of the service providers today.
>>
>> But I believe the proper technical solution would be that the server
>> authenticates itself in the challenge, plus adding protected
>> informations
>> to the challenge which allows the receiver of the challenge to verify
>> that
>> this challenge was/is targeted to himself.
>> The dis-advantage is clearly that this would be only possible with
>> extensions of the existing protocols. But we would/might gain other
>> benefits by such a solution as well.
>
> I did not follow the discussion on draft-dotson-sip-mutual-auth. Has
> this work been discontinued?
>
> http://tools.ietf.org/id/draft-dotson-sip-mutual-auth-03.txt

I can not answer your question.

But this draft does not solve the discussed attack, because the server is
authenticated after the victim sends its credentials to the attacker
alerady. To have a chance to solve the attack scenario the
server/challenger would have to prove its identity together with the
challenge to prevent that the victim sends out its credentials to the
attacker at all.

Regards
  Nils Ohlmeier


From lists@ohlmeier.org  Thu Mar 12 11:49:27 2009
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From: "Nils Ohlmeier" <lists@ohlmeier.org>
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Cc: s.hoggan@cablelabs.com, steve.dotson@cox.com
Subject: [Sip] Question regarding draft-dotson-sip-mutual-auth-03
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Hello,

after reading the mutual auth draft:

http://tools.ietf.org/id/draft-dotson-sip-mutual-auth-03.txt

I have an open question:
what should the client do if the server send authentication informations
in a Proxy-Authentication-Info header back in a let say 200 response, but
when the client computes response it comes to a different result (e.g.
because man in the middle changed something in the messages)?

In chapter 5 of your draft you are simply referring to RFC3261 for more
details regarding the implementation of the UAC. But I failed to find any
information about the UAC handling of this header in 3261. Even RFC2617
gives no hints, at least I did not found any, what a client should do when
the server authentication fails.
So it is probably not your fault, but still an interesting question I
think. Especially because the client has already send its credentials when
the check of the server authentication fails.

Best regards
  Nils Ohlmeier


From dean.willis@softarmor.com  Thu Mar 12 12:27:30 2009
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Subject: Re: [Sip] Terminology (was RE: Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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On Mar 12, 2009, at 1:18 PM, Christer Holmberg wrote:

> A couple of question:
>
> 1. Does the 4244bis definition of "retarget" cover the freephone  
> case, when the R-URI is replaced with the AoR of B - not with the  
> contact of B - no matter whether it's done based on location  
> service, configuration or whatever?

By the way, I believe that the way I use the words, this is a  
"reroute" and not a "retarget". In the freephone case, I would still  
expect the destination to have an awareness that it is the target of  
freephone calls and have credentials for the freephone number such  
that it could appropriately authenticate its responses if we had a  
means to do so.

>
> 2. Related to the first question, when you say "...and thus chaning  
> the target of the request", what is the defintion of "target"?
>


My definition is that a "retarget" changes the expected identity of  
the expected responder, introducing the possibility of an  
"unanticipated respondent" scenario.

This is, AFAIK, very different from the 4244 terminology, which I've  
always held to be not particularly useful.

--
Dean


From sumanth@cablelabs.com  Thu Mar 12 13:24:02 2009
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Nils,

In the case you describe, the UAC cannot trust the response. So it can
take actions similar to when it does not receive a valid response: e.g.,
it can reasonably retry, log the error, and/or any other appropriate
behavior specified by the architecture.=20

RFC3261 supports the Authentication-Info header (used in registration)
and not the Proxy-Authentication-Info header; which is why this I-D was
authored in the first place. The reference was to ensure that the
behavior is kept consistent, and available irrespective of registration
and non-registration messages.=20

In any case, please refer to the following small thread (3 emails) for
the feedback I received during offline discussions at the last IETF:

http://www.ietf.org/mail-archive/web/sip/current/msg25882.html

=09
- S

-----Original Message-----
From: Nils Ohlmeier [mailto:lists@ohlmeier.org]=20
Sent: Thursday, March 12, 2009 12:50 PM
To: sip@ietf.org
Cc: steve.dotson@cox.com; Stuart Hoggan; Sumanth Channabasappa
Subject: Question regarding draft-dotson-sip-mutual-auth-03

Hello,

after reading the mutual auth draft:

http://tools.ietf.org/id/draft-dotson-sip-mutual-auth-03.txt

I have an open question:
what should the client do if the server send authentication informations
in a Proxy-Authentication-Info header back in a let say 200 response,
but
when the client computes response it comes to a different result (e.g.
because man in the middle changed something in the messages)?

In chapter 5 of your draft you are simply referring to RFC3261 for more
details regarding the implementation of the UAC. But I failed to find
any
information about the UAC handling of this header in 3261. Even RFC2617
gives no hints, at least I did not found any, what a client should do
when
the server authentication fails.
So it is probably not your fault, but still an interesting question I
think. Especially because the client has already send its credentials
when
the check of the server authentication fails.

Best regards
  Nils Ohlmeier


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From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: Mary Barnes <mary.barnes@nortel.com>
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Subject: Re: [Sip] Issue 3 - two tags (was RE: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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--0015174c0cec362eb10464f2a1cd
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 Inline...

/Hans Erik van Elburg


On Thu, Mar 12, 2009 at 7:10 PM, Mary Barnes <mary.barnes@nortel.com> wrote:

>  I think there is another solution proposal that we should pursue for
> issue 3 identified in the email summary sent by Shida earlier this week -
> I've snipped all the text except for that issue and issue 2 in the summary
> below.  I do think the discussion has confounded the two issues and I think
> it ought to be clarified that if we agree to the two tags per issue 3, that
> the options for the terminology MUST change. And, perhaps this is why the
> other discussion threads aren't progressing well.
>

Agree.


>
> So, Hans Erick, I would like clarification on as to your view on the
> solution:
>
> In general, ISTM that one of the issues we're having in this terminology
> thread is that you are considering the solution to be that the hi-entries
> are tagged as they
> are added to the request.  And, just for clarification from my perspective,
> that is NOT the solution in either 4244bis or target-uri.
>
>
The principle should be that the node that has the information about what
type of rewrite is performed should add the tag. (This is done by the
current target-uri draft allthough the solution is not complete.)

For the freephone service, that would be the freephone service. For the AOR
that is the home proxy.

Assume that the freephone service receives an INVITE  request that either
1. received request contains History-Info header that contains the freephone
number (as in R-URI) as last entry, THEN when rewriting the Request-URI it
has to tag that existing received entry AND add the new entry with the new
value of the Request-URI. (I think that was how History-Info works right.)
2. received request contains History-Info header that does not contain the
freephone number as last entry, THEN when rewriting the Request-URI it has
add an entry with the freephone number and tag that entry AND add the new
entry with the new value of the Request-URI.
3. received request does not contain a History-Info header, THEN when
rewriting the Request-URI it has to add an entry with the freephone number
and tag that entry AND add the new entry with the new value of the
Request-URI.

After forwarding this request it arrives at the home proxy:
1. received request contains History-Info header that contains the AOR (as
in R-URI) as last entry, THEN when rewriting the Request-URI it has to tag
that existing received entry AND add the new entry with the new value of the
Request-URI i.e. the registered contact. (I think that was how History-Info
works right.)
2. received request contains History-Info header that does not contain the
AOR (as in R-URI) as last entry, THEN when rewriting the Request-URI it has
add an entry with the AOR and tag that entry AND add the new entry with the
new value of the Request-URI i.e. the registered contact.
3. received request does not contain a History-Info header, THEN when
rewriting the Request-URI it has to add an entry with the AOR as in the
Request-URI and tag that entry AND add the new entry with the new value of
the Request-URI i.e. the registered contact..


>  I don't view it as a problem necessarily if you'd like to pursue an
> alternate solution, we just need to be clear that's what you're advocating.
> In that case, I would agree the "retarget" is not an appropriate tag for the
> entries, as you don't know at the point in time that you add an hi-entry
> that it will be retargeted. In the case of this approach to the solution,
> you do have to alter the mechanism for determining the hi-entry that your
> services would want to use. You would need to skip the most recent/last
> hi-entry (as that would contain the same as the Request-URI in the incoming
> request), since it would be tagged if you use this solution approach.
>

Yes, but only if the previous proxy added it. As History-Info is optional
one never knows really.


> Thus, you'd have to take the next hi-entry that you find that has the
> desired "tag".  I will posit that we could accomplish this solution without
> any changes to 4244 by just registering new SIP URI parameter(s).   And,
>  that's not to say we shouldn't do a 4244bis as there are other issues to
> address in that document, but that allows the target-uri to describe a
> solution that makes use of 4244bis "as is".
>
>
Are you proposing to decouple the target-uri and the 4244bis discussion?


> So, feedback on this specific point (from Hans Erik and others) would be
> appreciated.
>
>
>  Mary.
>
>

--0015174c0cec362eb10464f2a1cd
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable





<div bgcolor=3D"#ffffff" text=3D"#000000">
Inline...<br>
<br clear=3D"all">
/Hans Erik van Elburg<br>
<br>
<br>
<div class=3D"gmail_quote">On Thu, Mar 12, 2009 at 7:10 PM, Mary Barnes <sp=
an dir=3D"ltr">&lt;<a href=3D"mailto:mary.barnes@nortel.com" target=3D"_bla=
nk">mary.barnes@nortel.com</a>&gt;</span>
wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
  <div>
  <div><font size=3D"2" color=3D"#0000ff" face=3D"Arial"><span>I think ther=
e
is another solution proposal that we should pursue for issue 3
identified in the email summary sent by Shida earlier this week - I&#39;ve
snipped all the text except for that issue and issue 2 in the summary
below.=A0 I do think the discussion has confounded the two issues and I
think it ought to be clarified that if we agree to the two tags per
issue 3, that the options for the terminology MUST change. And, perhaps
this is why the other discussion threads aren&#39;t progressing well.</span=
></font></div>
  </div>
</blockquote>
<div><br>
Agree.<br>
=A0</div>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
  <div>
  <div><font size=3D"2" color=3D"#0000ff" face=3D"Arial"><span></span></fon=
t></div>
  <div><font size=3D"2" color=3D"#0000ff" face=3D"Arial"><span></span></fon=
t>=A0</div>
  <div><font size=3D"2" color=3D"#0000ff" face=3D"Arial"><span>So, Hans
Erick, I would like clarification on as to your view on the solution:</span=
></font></div>
  <div><font size=3D"2" color=3D"#0000ff" face=3D"Arial"><span></span></fon=
t>=A0</div>
  <div><font size=3D"2" color=3D"#0000ff" face=3D"Arial"><span>In general,
ISTM that one of the issues we&#39;re having in this terminology thread is
that you are considering the solution to be that the hi-entries are
tagged as they </span></font></div>
  <div><font size=3D"2" color=3D"#0000ff" face=3D"Arial"><span>are added to
the request.=A0 And, just for clarification from my perspective, that is
NOT the solution in either 4244bis or target-uri. </span></font></div>
  <div><font size=3D"2" color=3D"#0000ff" face=3D"Arial"><span></span></fon=
t><font size=3D"2" color=3D"#0000ff" face=3D"Arial"><span></span></font>=A0=
</div>
  </div>
</blockquote>
<div>The principle should be that the node that has the information
about what type of rewrite is performed should add the tag. (This is done b=
y the current target-uri draft allthough the solution is not complete.)<br>
<br>
For the freephone service, that would be the freephone service. For the AOR=
 that is the home proxy.<br>
<br>
Assume that the freephone service receives an INVITE=A0 request that
either=A0 <br>
1. received request contains History-Info header that contains the
freephone number (as in R-URI) as last entry, THEN when rewriting the
Request-URI it has to tag that existing received entry AND add the new
entry with the new value of the Request-URI. (I think that was how
History-Info works right.)<br>
2. received request contains History-Info header that does not contain
the freephone number as last
entry, THEN when rewriting the Request-URI it has add an entry with the
freephone number and tag that entry AND add the new entry with the new
value of the Request-URI.<br>
3. received request does not contain a History-Info header, THEN when
rewriting the Request-URI it has to add an entry with the
freephone number and tag that entry AND add the new entry with the new
value of the Request-URI.<br>
<br>
After forwarding this request it arrives at the home proxy:<br>
1. received request contains History-Info header that contains the AOR
(as in R-URI) as last entry, THEN when rewriting the
Request-URI it has to tag that existing received entry AND add the new
entry with the new value of the Request-URI i.e. the registered
contact. (I think that was how
History-Info works right.)<br>
2. received request contains History-Info header that does not contain
the AOR (as in R-URI) as last entry, THEN when rewriting the
Request-URI it has add an entry with the AOR and tag that entry AND add
the new entry with the new
value of the Request-URI i.e. the registered contact.<br>
3. received request does not contain a History-Info header, THEN when
rewriting the Request-URI it has to add an entry with the AOR as in the
Request-URI and tag that entry AND add the new entry with the new
value of the Request-URI i.e. the registered contact..<br>
=A0</div>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
  <div>
  <div><font size=3D"2" color=3D"#0000ff" face=3D"Arial"><span>I don&#39;t =
view
it as a problem necessarily if you&#39;d like to pursue an alternate
solution, we just need to be clear that&#39;s what you&#39;re advocating. I=
n
that case, I would agree the &quot;retarget&quot; is not an appropriate tag=
 for
the entries, as you don&#39;t know at the point in time that you add an
hi-entry that it will be retargeted. In the case of this approach to
the solution, you do have to alter the mechanism for determining the
hi-entry that your services would want to use. You would need to skip
the most recent/last hi-entry (as that would=A0contain the same as the
Request-URI in the incoming request), since it would be tagged if you
use this solution approach.</span></font></div></div></blockquote><div><br>=
Yes, but only if the previous proxy added it. As History-Info is optional o=
ne never knows really.<br>=A0<br></div><blockquote class=3D"gmail_quote" st=
yle=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex=
; padding-left: 1ex;">
<div><div><font size=3D"2" color=3D"#0000ff" face=3D"Arial"><span> Thus, yo=
u&#39;d have to take the next hi-entry
that you find that has the desired &quot;tag&quot;.=A0 I will posit that we=
 could
accomplish this solution without any changes to 4244 by just
registering new=A0SIP URI parameter(s).=A0=A0=A0And, =A0that&#39;s not to s=
ay we
shouldn&#39;t do a 4244bis as there are other issues to address in that
document, but that allows the target-uri to describe a solution that
makes use of 4244bis &quot;as is&quot;.<br>
  </span></font></div>
  <div><font size=3D"2" color=3D"#0000ff" face=3D"Arial"><span></span></fon=
t>=A0</div></div></blockquote><div>Are you proposing to decouple the target=
-uri and the 4244bis discussion?<br>=A0</div><blockquote class=3D"gmail_quo=
te" style=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt=
 0.8ex; padding-left: 1ex;">
<div><div></div>
  <div><font size=3D"2" color=3D"#0000ff" face=3D"Arial"><span>So, feedback
on this specific point (from Hans Erik and others) would be
appreciated. </span></font></div>
  <div><font size=3D"2" color=3D"#0000ff" face=3D"Arial"><span></span></fon=
t><br>
  <br>
  </div>
  <div><font size=3D"2" color=3D"#0000ff" face=3D"Arial"><span>Mary.</span>=
</font></div>
  <div> <br>
  </div>
  </div>
</blockquote>
</div>
<br>
<br>
</div>



--0015174c0cec362eb10464f2a1cd--

From christer.holmberg@ericsson.com  Thu Mar 12 14:56:39 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Mary Barnes" <mary.barnes@nortel.com>, "Hans Erik van Elburg" <ietf.hanserik@gmail.com>
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Cc: sip@ietf.org, "Elwell, John" <john.elwell@siemens.com>
Subject: Re: [Sip] Terminology (was RE: Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Hi Mary,=20

>A couple of question:
>=20
>1. Does the 4244bis definition of "retarget" cover the freephone case,
>when the R-URI is replaced with the AoR of B - not with the contact of
B
>- no matter whether it's done based on location service, configuration
>or whatever?
>[MB] That was the intention, however, the current text needs to be made
>more general (i.e., I was using location service to mean more than
>registration based contacts - another thing that's not quite so clear
in
>3261). [/MB]

[CH] We can fix the text. I just wanted to make sure that you agree on
the principle.

=20
>2. Related to the first question, when you say "...and thus chaning the
>target of the request", what is the defintion of "target"?
>[MB] The definition of target matches that used in section 16.5 of
3261.
>A target is an entry in the target list, which is populated using
>information from a location service, configuration, etc. I can clarify
>that as well and as you previously suggested make a more comprehensive
>glossary in the more conventional format - e.g., "Target: definition
>text, blah, blah, blah...] So, I believe it is the same use of target
as
>in the target-uri draft. [/MB]

[CH] Ok.


Regards,
=20
Christer

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Mary Barnes
Sent: Thursday, March 12, 2009 5:20 PM
To: Hans Erik van Elburg
Cc: sip@ietf.org; Elwell, John
Subject: Re: [Sip] Terminology (was RE:
Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


When I use "retarget" I am using the definition that I put into 4244bis.
So, if you could ignore the tag or change it to 'iscontact' or whatever,
what would you change in that definition? Also, consider that if you
were to change everyplace that I use
"retarget"/"retargeted"/"retargeting" in 4244bis with 'iscontact' the
text would not make sense. So, even if we change the name of the tag, we
still need to understand the concept or define another term that means
that the request-uri in the incoming request (which should be the same
as the last hi-entry in the request) is being chnaged due to service
lookup, configuration, blah, blah, blah and that to meet the
requirements in the target-uri document, we want to tag this entry so
that this request-uri is not "lost" - i.e., this is the one the services
identified in the target-uri document care about.=20
=20
So, right now, here's what's in 4244bis:
=20
   The term "retarget" is used in this document to refer to the process
   of a Proxy Server/User Agent Client (UAC) changing a Uniform Resource
   Identifier (URI) in a request based on a lookup in a location service
   and thus changing the target of the request.
=20
We have previously discussed that we do need to expand the "based on a
lookup in a location service" to include configuration, etc., so let's
not debate that.=20
=20
But, what explicitly is the concern with that definition?
=20
Mary.=20
=20

________________________________

From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
Sent: Thursday, March 12, 2009 9:55 AM
To: Barnes, Mary (RICH2:AR00)
Cc: Elwell, John; sip@ietf.org
Subject: Re: [Sip] Terminology (was RE: Fwd:
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


I'm fine with discussing implementation if everyone is meaning the same
if they use the same words.

As  to the "retarget" term such does not seem to be the case.

/Hans Erik van Elburg



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From: =?UTF-8?Q?Victor_Pascual_=C3=81vila?= <victor.pascual.avila@gmail.com>
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Subject: Re: [Sip] Question regarding draft-dotson-sip-mutual-auth-03
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On Thu, Mar 12, 2009 at 9:24 PM, Sumanth Channabasappa
<sumanth@cablelabs.com> wrote:
(snip)
> In any case, please refer to the following small thread (3 emails) for
> the feedback I received during offline discussions at the last IETF:
>
> http://www.ietf.org/mail-archive/web/sip/current/msg25882.html

>From [1]: "the UA trusts its connection with the next-hop and does not
need to authenticate any elements beyond the next-hop"

[1] http://www.ietf.org/mail-archive/web/sip/current/msg25890.html

Could you please develop this assumption?

Thanks,
--=20
Victor Pascual =C3=81vila

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From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
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Subject: Re: [Sip] Terminology (was RE: Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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--0015174beb3ad629910464fd6c61
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Hi Dean,

Yes this was exactly how we used/defined the terms at the time we where
discussing the ua-loose-route vs Target-header solutions.

I still think that this way they are most intuitive and closest to how you
would use the terms in natural language.

/Hans Erik van Elburg

On Thu, Mar 12, 2009 at 8:27 PM, Dean Willis <dean.willis@softarmor.com>wrote:

>
> On Mar 12, 2009, at 1:18 PM, Christer Holmberg wrote:
>
>  A couple of question:
>>
>> 1. Does the 4244bis definition of "retarget" cover the freephone case,
>> when the R-URI is replaced with the AoR of B - not with the contact of B -
>> no matter whether it's done based on location service, configuration or
>> whatever?
>>
>
> By the way, I believe that the way I use the words, this is a "reroute" and
> not a "retarget". In the freephone case, I would still expect the
> destination to have an awareness that it is the target of freephone calls
> and have credentials for the freephone number such that it could
> appropriately authenticate its responses if we had a means to do so.
>
>
>> 2. Related to the first question, when you say "...and thus chaning the
>> target of the request", what is the defintion of "target"?
>>
>>
>
> My definition is that a "retarget" changes the expected identity of the
> expected responder, introducing the possibility of an "unanticipated
> respondent" scenario.
>
> This is, AFAIK, very different from the 4244 terminology, which I've always
> held to be not particularly useful.
>
> --
> Dean
>
>

--0015174beb3ad629910464fd6c61
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hi Dean,<br><br>Yes this was exactly how we used/defined the terms at the t=
ime we where discussing the ua-loose-route vs Target-header solutions.<br><=
br>I still think that this way they are most intuitive and closest to how y=
ou would use the terms in natural language. <br>
<br clear=3D"all">/Hans Erik van Elburg<br>
<br><div class=3D"gmail_quote">On Thu, Mar 12, 2009 at 8:27 PM, Dean Willis=
 <span dir=3D"ltr">&lt;<a href=3D"mailto:dean.willis@softarmor.com">dean.wi=
llis@softarmor.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quot=
e" style=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt =
0.8ex; padding-left: 1ex;">
<div class=3D"im"><br>
On Mar 12, 2009, at 1:18 PM, Christer Holmberg wrote:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
A couple of question:<br>
<br>
1. Does the 4244bis definition of &quot;retarget&quot; cover the freephone =
case, when the R-URI is replaced with the AoR of B - not with the contact o=
f B - no matter whether it&#39;s done based on location service, configurat=
ion or whatever?<br>

</blockquote>
<br></div>
By the way, I believe that the way I use the words, this is a &quot;reroute=
&quot; and not a &quot;retarget&quot;. In the freephone case, I would still=
 expect the destination to have an awareness that it is the target of freep=
hone calls and have credentials for the freephone number such that it could=
 appropriately authenticate its responses if we had a means to do so.<div c=
lass=3D"im">
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
2. Related to the first question, when you say &quot;...and thus chaning th=
e target of the request&quot;, what is the defintion of &quot;target&quot;?=
<br>
<br>
</blockquote>
<br>
<br></div>
My definition is that a &quot;retarget&quot; changes the expected identity =
of the expected responder, introducing the possibility of an &quot;unantici=
pated respondent&quot; scenario.<br>
<br>
This is, AFAIK, very different from the 4244 terminology, which I&#39;ve al=
ways held to be not particularly useful.<br>
<br>
--<br><font color=3D"#888888">
Dean<br>
<br>
</font></blockquote></div><br>

--0015174beb3ad629910464fd6c61--

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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Hans Erik van Elburg" <ietf.hanserik@gmail.com>, "Dean Willis" <dean.willis@softarmor.com>
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Subject: Re: [Sip] Terminology (was RE: Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Hi,
=20
I think we should use those definitions as base. That way I think we
would be able to progress faster.
=20
Because, I think we really need to agree on the definitions FIRST. There
is no idea on discussing detailed wording in 4244bis if we don't have a
common understanding of what we mean.
=20
Regards,
=20
Christer
=20


________________________________

	From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
	Sent: 13. maaliskuuta 2009 12:19
	To: Dean Willis
	Cc: Christer Holmberg; Mary Barnes; sip@ietf.org; Elwell, John
	Subject: Re: [Sip] Terminology (was RE:
Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
=09
=09
	Hi Dean,
=09
	Yes this was exactly how we used/defined the terms at the time
we where discussing the ua-loose-route vs Target-header solutions.
=09
	I still think that this way they are most intuitive and closest
to how you would use the terms in natural language.=20
=09
	/Hans Erik van Elburg
=09
=09
	On Thu, Mar 12, 2009 at 8:27 PM, Dean Willis
<dean.willis@softarmor.com> wrote:
=09


		On Mar 12, 2009, at 1:18 PM, Christer Holmberg wrote:
	=09
	=09

			A couple of question:
		=09
			1. Does the 4244bis definition of "retarget"
cover the freephone case, when the R-URI is replaced with the AoR of B -
not with the contact of B - no matter whether it's done based on
location service, configuration or whatever?
		=09


		By the way, I believe that the way I use the words, this
is a "reroute" and not a "retarget". In the freephone case, I would
still expect the destination to have an awareness that it is the target
of freephone calls and have credentials for the freephone number such
that it could appropriately authenticate its responses if we had a means
to do so.=20




			2. Related to the first question, when you say
"...and thus chaning the target of the request", what is the defintion
of "target"?
		=09
		=09



		My definition is that a "retarget" changes the expected
identity of the expected responder, introducing the possibility of an
"unanticipated respondent" scenario.
	=09
		This is, AFAIK, very different from the 4244
terminology, which I've always held to be not particularly useful.
	=09
		--
		Dean
	=09
	=09



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<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D198453811-13032009>Hi,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D198453811-13032009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D198453811-13032009>I think we should use those definitions as =
base. That=20
way I think we would be able to progress faster.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D198453811-13032009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D198453811-13032009>Because, I think we really need to agree on =
the=20
definitions FIRST. There is no idea on discussing detailed wording in =
4244bis if=20
we don't have a common understanding of what we =
mean.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D198453811-13032009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D198453811-13032009>Regards,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D198453811-13032009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D198453811-13032009>Christer</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV><BR>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Hans Erik van Elburg=20
  [mailto:ietf.hanserik@gmail.com] <BR><B>Sent:</B> 13. maaliskuuta 2009 =

  12:19<BR><B>To:</B> Dean Willis<BR><B>Cc:</B> Christer Holmberg; Mary =
Barnes;=20
  sip@ietf.org; Elwell, John<BR><B>Subject:</B> Re: [Sip] Terminology =
(was RE:=20
  =
Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<BR></FONT><B=
R></DIV>
  <DIV></DIV>Hi Dean,<BR><BR>Yes this was exactly how we used/defined =
the terms=20
  at the time we where discussing the ua-loose-route vs Target-header=20
  solutions.<BR><BR>I still think that this way they are most intuitive =
and=20
  closest to how you would use the terms in natural language. <BR><BR=20
  clear=3Dall>/Hans Erik van Elburg<BR><BR>
  <DIV class=3Dgmail_quote>On Thu, Mar 12, 2009 at 8:27 PM, Dean Willis =
<SPAN=20
  dir=3Dltr>&lt;<A=20
  =
href=3D"mailto:dean.willis@softarmor.com">dean.willis@softarmor.com</A>&g=
t;</SPAN>=20
  wrote:<BR>
  <BLOCKQUOTE class=3Dgmail_quote=20
  style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">
    <DIV class=3Dim><BR>On Mar 12, 2009, at 1:18 PM, Christer Holmberg=20
    wrote:<BR><BR>
    <BLOCKQUOTE class=3Dgmail_quote=20
    style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">A=20
      couple of question:<BR><BR>1. Does the 4244bis definition of =
"retarget"=20
      cover the freephone case, when the R-URI is replaced with the AoR =
of B -=20
      not with the contact of B - no matter whether it's done based on =
location=20
      service, configuration or whatever?<BR></BLOCKQUOTE><BR></DIV>By =
the way, I=20
    believe that the way I use the words, this is a "reroute" and not a=20
    "retarget". In the freephone case, I would still expect the =
destination to=20
    have an awareness that it is the target of freephone calls and have=20
    credentials for the freephone number such that it could =
appropriately=20
    authenticate its responses if we had a means to do so.
    <DIV class=3Dim><BR><BR>
    <BLOCKQUOTE class=3Dgmail_quote=20
    style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid"><BR>2.=20
      Related to the first question, when you say "...and thus chaning =
the=20
      target of the request", what is the defintion of=20
    "target"?<BR><BR></BLOCKQUOTE><BR><BR></DIV>My definition is that a=20
    "retarget" changes the expected identity of the expected responder,=20
    introducing the possibility of an "unanticipated respondent"=20
    scenario.<BR><BR>This is, AFAIK, very different from the 4244 =
terminology,=20
    which I've always held to be not particularly =
useful.<BR><BR>--<BR><FONT=20
    =
color=3D#888888>Dean<BR><BR></FONT></BLOCKQUOTE></DIV><BR></BLOCKQUOTE></=
BODY></HTML>

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Thread-Topic: [Sipping] Pls clarify on session timer
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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "Prakash Ganesan" <vgprakashin@gmail.com>, <sip@ietf.org>, <sipping@ietf.org>
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Hi Prakash
=20
The Server sends a 420 when it doesn't support "timer" option-tag. so
its working fine when you remove it from your request.
=20
As per section 7.1 of RFC 4028, the UAC SHALL include a Require header
with option-tag as "time" in the initial INVITE when it wants the server
to support session timer - but this does not necessarily mean that UAC
wants the server to perform refresh - but only wants the server to
support the session timer.
=20
but if the UAC includes a Proxy-Require header with value as "timer", in
the initial INVITE then the server MUST support "session-timer" in order
to process the request.
=20
But again as per RFC Require and Proxy-Require headers are not
recommended to be used - but a UAC needs to include "Session-Expires"
header field to indicate to the server to apply Session Timer.

Regards=20
Ranjit=20
=20

________________________________

From: sipping-bounces@ietf.org [mailto:sipping-bounces@ietf.org] On
Behalf Of Prakash Ganesan
Sent: Friday, March 13, 2009 5:08 PM
To: sip@ietf.org; sipping@ietf.org
Subject: [Sipping] Pls clarify on session timer


Hi All,
I am facing a problem with the session timer.
=20
The refesh re-invite contains header field Require: timer and my server
is rejecting the request with 420 bad extension. When I remove the
header field Reqire: timer, it is working fine.=20
I am bit confused with the section 7.1 Generating an Initial Session
Refresh Request, in RFC 4028 it is referring to the first SIP invite or
the first session refresher (re-Invite)


--=20
Cheers
Prakash


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<DIV dir=3Dltr align=3Dleft><SPAN class=3D824334311-13032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Hi Prakash</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D824334311-13032009><FONT =
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<DIV dir=3Dltr align=3Dleft><SPAN class=3D824334311-13032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>The Server sends a 420 when it doesn't support =
"timer"=20
option-tag. so its working fine when you remove it from your=20
request.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D824334311-13032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D824334311-13032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>As per section 7.1 of RFC 4028, the UAC SHALL =
include a=20
Require header with option-tag as "time" in the initial INVITE when it =
wants the=20
server to support session timer - but this does not necessarily mean =
that UAC=20
wants the server to perform refresh - but only wants the server to =
support the=20
session timer.</FONT></SPAN></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D824334311-13032009><FONT face=3DArial color=3D#0000ff =
size=3D2>but if=20
the UAC includes a Proxy-Require header with value as "timer", in the =
initial=20
INVITE then the server MUST support "session-timer" in order to process =
the=20
request.</FONT></SPAN></DIV>
<DIV><SPAN class=3D824334311-13032009><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D824334311-13032009><FONT face=3DArial color=3D#0000ff =
size=3D2>But=20
again as per RFC Require and Proxy-Require headers are not recommended =
to be=20
used - but a UAC needs to include "Session-Expires" header field to =
indicate to=20
the server to apply Session Timer.</FONT></SPAN><BR =
clear=3Dall><BR><SPAN=20
lang=3Den-us><FONT face=3DArial color=3D#0000ff =
size=3D2>Regards</FONT></SPAN> <BR><SPAN=20
lang=3Den-us><FONT face=3DArial color=3D#0000ff =
size=3D2>Ranjit</FONT></SPAN>=20
</DIV></DIV>
<DIV>&nbsp;</DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> sipping-bounces@ietf.org=20
[mailto:sipping-bounces@ietf.org] <B>On Behalf Of </B>Prakash=20
Ganesan<BR><B>Sent:</B> Friday, March 13, 2009 5:08 PM<BR><B>To:</B>=20
sip@ietf.org; sipping@ietf.org<BR><B>Subject:</B> [Sipping] Pls clarify =
on=20
session timer<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><FONT face=3Dverdana,sans-serif>Hi All,</FONT></DIV>
<DIV><FONT face=3Dverdana,sans-serif>I am facing a problem with the =
session=20
timer.</FONT></DIV>
<DIV><FONT face=3Dverdana,sans-serif></FONT>&nbsp;</DIV>
<DIV><FONT face=3Dverdana,sans-serif>The refesh re-invite contains =
header field=20
Require: timer and my server is rejecting the request with 420 bad =
extension.=20
When I remove the header field Reqire: timer, it is working fine. =
</FONT></DIV>
<DIV><FONT face=3Dverdana,sans-serif>I am bit confused&nbsp;with the =
section 7.1=20
<SPAN lang=3DEN-US>Generating an Initial Session Refresh Request, in RFC =
4028 it=20
is referring to the first SIP invite or the first session refresher=20
(re-Invite)</SPAN></FONT></DIV>
<DIV><BR clear=3Dall><BR>-- =
<BR>Cheers<BR>Prakash<BR></DIV></BODY></HTML>

------_=_NextPart_001_01C9A3D2.438C6B88--

From lists@ohlmeier.org  Fri Mar 13 05:15:02 2009
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Date: Fri, 13 Mar 2009 13:15:28 +0100 (CET)
From: "Nils Ohlmeier" <lists@ohlmeier.org>
To: Victor Pascual =?iso-8859-15?Q?=C1vila?= <victor.pascual.avila@gmail.com>
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Cc: Nils Ohlmeier <lists@ohlmeier.org>, sip@ietf.org, Stuart Hoggan <s.hoggan@cablelabs.com>, steve.dotson@cox.com
Subject: Re: [Sip] Question regarding draft-dotson-sip-mutual-auth-03
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> On Thu, Mar 12, 2009 at 9:24 PM, Sumanth Channabasappa
> <sumanth@cablelabs.com> wrote:
> (snip)
>> In any case, please refer to the following small thread (3 emails) for
>> the feedback I received during offline discussions at the last IETF:
>>
>> http://www.ietf.org/mail-archive/web/sip/current/msg25882.html
>
>>From [1]: "the UA trusts its connection with the next-hop and does not
> need to authenticate any elements beyond the next-hop"
>
> [1] http://www.ietf.org/mail-archive/web/sip/current/msg25890.html
>
> Could you please develop this assumption?

I also do not understand this sentense. It sounds like the IP connection
to the next hop is so trustworthy that we could use clear text passwords
instead of digest auth?!
But I think that such conditions would only be given when we use TLS or
IPSec, in which case digest mutual auth makes no real sense any more.

Regards
  Nils


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Date: Fri, 13 Mar 2009 13:37:29 +0100 (CET)
From: "Nils Ohlmeier" <lists@ohlmeier.org>
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Cc: Nils Ohlmeier <lists@ohlmeier.org>, sip@ietf.org, Stuart Hoggan <s.hoggan@cablelabs.com>, steve.dotson@cox.com
Subject: Re: [Sip] Question regarding draft-dotson-sip-mutual-auth-03
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Hi Sumanth,

> In the case you describe, the UAC cannot trust the response. So it can
> take actions similar to when it does not receive a valid response: e.g.,
> it can reasonably retry, log the error, and/or any other appropriate
> behavior specified by the architecture.

Ok, but the point is that in most of the cases it will be too late already.
Maybe an example is helpfull:
I try to register via proxy.hotel.com to my registrar at example.com.
Note: I'm using the open WLAN of hotel of cours. Now proxy.hotel.com
challenges my initial REGISTER request, which my UA replies to with the
password which I got at the receiption. Now this second REGISTER request
is forwarded to example.com and replied with a 200 OK.
When my UA receives the 200 OK it checks the Proxy-Authentication-Info
header and detects an error (maybe because a man-in-the-midlle modified
something). Upppss!
What should the UA do here? Display me a warning "It looks like you
successfully registered to example.com. Unfortunately something went
wrong. But your credentials where send already to example.com... Maybe now
someone else is registered with your account to example.com?!"

> RFC3261 supports the Authentication-Info header (used in registration)
> and not the Proxy-Authentication-Info header; which is why this I-D was
> authored in the first place. The reference was to ensure that the
> behavior is kept consistent, and available irrespective of registration
> and non-registration messages.

I understand that your draft is only adding the Proxy-Authentication-Info
header.
On the other hand I doubt that the whole mutual digest auth like it is
described in 2617 and 3261 is usuable for SIP at all.

> In any case, please refer to the following small thread (3 emails) for
> the feedback I received during offline discussions at the last IETF:
>
> http://www.ietf.org/mail-archive/web/sip/current/msg25882.html

>From that small thread I can only ask the same question as Victor did
already: what is meant by this trusted connection to the next hop?

And another issue in your draft in my opinion is what you describe in
second paragraph of section 10:
If someone is going to use mutual digest authentication it can not be
optional! As recent "demonstrations" on HTTPS and SMTP with TLS have shown
security "upgrades" of an IP connection are not optional. If you know that
your server (the hotel proxy in the example above) supports mutual auth,
you should configure your UA to require mutual auth. If the mutual auth is
not there or failed, something is really wrong in the network and you
should definately not proceed with whatever you wanted to do.

If you insist on the Proxy-Authorization-Info header being optional is
will be of no use at all. Because in all positiv working scenario it will
be present and working. But in all scenarios with broken, tapped or
whatever networks the header will be missing and your UA should send any
message with credentials in first place.

Best regards
  Nils Ohlmeier

>
> - S
>
> -----Original Message-----
> From: Nils Ohlmeier [mailto:lists@ohlmeier.org]
> Sent: Thursday, March 12, 2009 12:50 PM
> To: sip@ietf.org
> Cc: steve.dotson@cox.com; Stuart Hoggan; Sumanth Channabasappa
> Subject: Question regarding draft-dotson-sip-mutual-auth-03
>
> Hello,
>
> after reading the mutual auth draft:
>
> http://tools.ietf.org/id/draft-dotson-sip-mutual-auth-03.txt
>
> I have an open question:
> what should the client do if the server send authentication informations
> in a Proxy-Authentication-Info header back in a let say 200 response,
> but
> when the client computes response it comes to a different result (e.g.
> because man in the middle changed something in the messages)?
>
> In chapter 5 of your draft you are simply referring to RFC3261 for more
> details regarding the implementation of the UAC. But I failed to find
> any
> information about the UAC handling of this header in 3261. Even RFC2617
> gives no hints, at least I did not found any, what a client should do
> when
> the server authentication fails.
> So it is probably not your fault, but still an interesting question I
> think. Especially because the client has already send its credentials
> when
> the check of the server authentication fails.
>
> Best regards
>   Nils Ohlmeier
>
>



From john.elwell@siemens.com  Fri Mar 13 07:24:04 2009
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Date: Fri, 13 Mar 2009 14:24:30 +0000
From: "Elwell, John" <john.elwell@siemens.com>
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Subject: Re: [Sip] Issue 3 - two tags (was RE: I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Hans Erik,

Following your detailed description, it seems that the UAS serving a
freephone number could receive one of the following (assuming H-I is
supported by all relevant nodes):

1) H-I containing:
- the freephone number (tagged);
- the AoR (tagged);
- the contact URI (not tagged).

Or

2) H-I containing:=20
- the freephone number (tagged);
- the contact URI (not tagged).

The latter would occur when the same proxy translates the freephone
number (via the AoR) to the contact directly.

So how does this help the UAS to figure out the freephone number? It
seems the UAS would have to search back through the tagged entries
looking for one that matches a freephone number it recognises? Does the
tagging really add much value?

John


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Hans Erik van Elburg
	Sent: 12 March 2009 21:26
	To: Mary Barnes
	Cc: Hans Erik van Elburg; sip@ietf.org
	Subject: Re: [Sip] Issue 3 - two tags (was RE:
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
=09
=09
	Inline...
=09
	/Hans Erik van Elburg
=09
=09
=09
	On Thu, Mar 12, 2009 at 7:10 PM, Mary Barnes
<mary.barnes@nortel.com> wrote:
=09

		I think there is another solution proposal that we
should pursue for issue 3 identified in the email summary sent by Shida
earlier this week - I've snipped all the text except for that issue and
issue 2 in the summary below.  I do think the discussion has confounded
the two issues and I think it ought to be clarified that if we agree to
the two tags per issue 3, that the options for the terminology MUST
change. And, perhaps this is why the other discussion threads aren't
progressing well.


	Agree.
	=20

				=20
		So, Hans Erick, I would like clarification on as to your
view on the solution:
		=20
		In general, ISTM that one of the issues we're having in
this terminology thread is that you are considering the solution to be
that the hi-entries are tagged as they=20
		are added to the request.  And, just for clarification
from my perspective, that is NOT the solution in either 4244bis or
target-uri.=20
		=20

	The principle should be that the node that has the information
about what type of rewrite is performed should add the tag. (This is
done by the current target-uri draft allthough the solution is not
complete.)
=09
	For the freephone service, that would be the freephone service.
For the AOR that is the home proxy.
=09
	Assume that the freephone service receives an INVITE  request
that either =20
	1. received request contains History-Info header that contains
the freephone number (as in R-URI) as last entry, THEN when rewriting
the Request-URI it has to tag that existing received entry AND add the
new entry with the new value of the Request-URI. (I think that was how
History-Info works right.)
	2. received request contains History-Info header that does not
contain the freephone number as last entry, THEN when rewriting the
Request-URI it has add an entry with the freephone number and tag that
entry AND add the new entry with the new value of the Request-URI.
	3. received request does not contain a History-Info header, THEN
when rewriting the Request-URI it has to add an entry with the freephone
number and tag that entry AND add the new entry with the new value of
the Request-URI.
=09
	After forwarding this request it arrives at the home proxy:
	1. received request contains History-Info header that contains
the AOR (as in R-URI) as last entry, THEN when rewriting the Request-URI
it has to tag that existing received entry AND add the new entry with
the new value of the Request-URI i.e. the registered contact. (I think
that was how History-Info works right.)
	2. received request contains History-Info header that does not
contain the AOR (as in R-URI) as last entry, THEN when rewriting the
Request-URI it has add an entry with the AOR and tag that entry AND add
the new entry with the new value of the Request-URI i.e. the registered
contact.
	3. received request does not contain a History-Info header, THEN
when rewriting the Request-URI it has to add an entry with the AOR as in
the Request-URI and tag that entry AND add the new entry with the new
value of the Request-URI i.e. the registered contact..
	=20

		I don't view it as a problem necessarily if you'd like
to pursue an alternate solution, we just need to be clear that's what
you're advocating. In that case, I would agree the "retarget" is not an
appropriate tag for the entries, as you don't know at the point in time
that you add an hi-entry that it will be retargeted. In the case of this
approach to the solution, you do have to alter the mechanism for
determining the hi-entry that your services would want to use. You would
need to skip the most recent/last hi-entry (as that would contain the
same as the Request-URI in the incoming request), since it would be
tagged if you use this solution approach.


	Yes, but only if the previous proxy added it. As History-Info is
optional one never knows really.
	=20
=09

		Thus, you'd have to take the next hi-entry that you find
that has the desired "tag".  I will posit that we could accomplish this
solution without any changes to 4244 by just registering new SIP URI
parameter(s).   And,  that's not to say we shouldn't do a 4244bis as
there are other issues to address in that document, but that allows the
target-uri to describe a solution that makes use of 4244bis "as is".
	=09
		=20

	Are you proposing to decouple the target-uri and the 4244bis
discussion?
	=20

		So, feedback on this specific point (from Hans Erik and
others) would be appreciated.=20
	=09
	=09
	=09
		Mary.





From mary.barnes@nortel.com  Fri Mar 13 07:27:35 2009
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Thread-Topic: [Sip] Terminology (was RE: Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
Thread-Index: AcmjxRhZ819f4W9wQ5iV2x9RFuWOtAACyv+AAAXQ7aA=
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>, "Hans Erik van Elburg" <ietf.hanserik@gmail.com>, "Dean Willis" <dean.willis@softarmor.com>
Cc: sip@ietf.org, "Elwell, John" <john.elwell@siemens.com>
Subject: Re: [Sip] Terminology (was RE: Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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That's fine - we just need to be careful with the wording in 4244 so
that the existing normative functionality isn't so impacted. I'll look
at the detailed impact of the changes as retarget is quite prevalent in
that document. It could be that we can leave it in the early sections
(and define it to be the same as the term we agree) and just keep it out
of the normative sections - right now, it is only used in a few places
other than describing the new parameter in the normative sections.
=20
Mary.

________________________________

From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
Sent: Friday, March 13, 2009 6:42 AM
To: Hans Erik van Elburg; Dean Willis
Cc: Barnes, Mary (RICH2:AR00); sip@ietf.org; Elwell, John
Subject: RE: [Sip] Terminology (was RE:
Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


Hi,
=20
I think we should use those definitions as base. That way I think we
would be able to progress faster.
=20
Because, I think we really need to agree on the definitions FIRST. There
is no idea on discussing detailed wording in 4244bis if we don't have a
common understanding of what we mean.
=20
Regards,
=20
Christer
=20


________________________________

	From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
	Sent: 13. maaliskuuta 2009 12:19
	To: Dean Willis
	Cc: Christer Holmberg; Mary Barnes; sip@ietf.org; Elwell, John
	Subject: Re: [Sip] Terminology (was RE:
Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
=09
=09
	Hi Dean,
=09
	Yes this was exactly how we used/defined the terms at the time
we where discussing the ua-loose-route vs Target-header solutions.
=09
	I still think that this way they are most intuitive and closest
to how you would use the terms in natural language.=20
=09
	/Hans Erik van Elburg
=09
=09
	On Thu, Mar 12, 2009 at 8:27 PM, Dean Willis
<dean.willis@softarmor.com> wrote:
=09


		On Mar 12, 2009, at 1:18 PM, Christer Holmberg wrote:
	=09
	=09

			A couple of question:
		=09
			1. Does the 4244bis definition of "retarget"
cover the freephone case, when the R-URI is replaced with the AoR of B -
not with the contact of B - no matter whether it's done based on
location service, configuration or whatever?
		=09


		By the way, I believe that the way I use the words, this
is a "reroute" and not a "retarget". In the freephone case, I would
still expect the destination to have an awareness that it is the target
of freephone calls and have credentials for the freephone number such
that it could appropriately authenticate its responses if we had a means
to do so.=20




			2. Related to the first question, when you say
"...and thus chaning the target of the request", what is the defintion
of "target"?
		=09
		=09



		My definition is that a "retarget" changes the expected
identity of the expected responder, introducing the possibility of an
"unanticipated respondent" scenario.
	=09
		This is, AFAIK, very different from the 4244
terminology, which I've always held to be not particularly useful.
	=09
		--
		Dean
	=09
	=09



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Date: Fri, 13 Mar 2009 15:33:16 +0100
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From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: "Elwell, John" <john.elwell@siemens.com>
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Subject: Re: [Sip] Issue 3 - two tags (was RE: I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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--0015174be6ea3ff725046500fba1
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Hi John,

Well that is simple one would use different tags for different type of
entries, like in the example I already gave:

   H-I:  Bfreephone; "istarget",\
        B,"aor"\
        contact-B

Where the UAS interested in the initial target with which it was addressed
it would obtain the "istarget" tagged entry.
Where the UAS interested in the AOR  that was used to address it would
obtain the "aor" tagged entry.

In simpler cases it could be that one entry holds the "aor" as well as the
"istarget" tag.

Of course the algorithm needs some work to make it complete.

/Hans Erik van Elburg


On Fri, Mar 13, 2009 at 3:24 PM, Elwell, John <john.elwell@siemens.com>wrote:

> Hans Erik,
>
> Following your detailed description, it seems that the UAS serving a
> freephone number could receive one of the following (assuming H-I is
> supported by all relevant nodes):
>
> 1) H-I containing:
> - the freephone number (tagged);
> - the AoR (tagged);
> - the contact URI (not tagged).
>
> Or
>
> 2) H-I containing:
> - the freephone number (tagged);
> - the contact URI (not tagged).
>
> The latter would occur when the same proxy translates the freephone
> number (via the AoR) to the contact directly.
>
> So how does this help the UAS to figure out the freephone number? It
> seems the UAS would have to search back through the tagged entries
> looking for one that matches a freephone number it recognises? Does the
> tagging really add much value?
>
> John
>
>
> ________________________________
>
>        From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> Behalf Of Hans Erik van Elburg
>        Sent: 12 March 2009 21:26
>        To: Mary Barnes
>        Cc: Hans Erik van Elburg; sip@ietf.org
>        Subject: Re: [Sip] Issue 3 - two tags (was RE:
> I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>
>
>        Inline...
>
>        /Hans Erik van Elburg
>
>
>
>        On Thu, Mar 12, 2009 at 7:10 PM, Mary Barnes
> <mary.barnes@nortel.com> wrote:
>
>
>                I think there is another solution proposal that we
> should pursue for issue 3 identified in the email summary sent by Shida
> earlier this week - I've snipped all the text except for that issue and
> issue 2 in the summary below.  I do think the discussion has confounded
> the two issues and I think it ought to be clarified that if we agree to
> the two tags per issue 3, that the options for the terminology MUST
> change. And, perhaps this is why the other discussion threads aren't
> progressing well.
>
>
>        Agree.
>
>
>
>                So, Hans Erick, I would like clarification on as to your
> view on the solution:
>
>                In general, ISTM that one of the issues we're having in
> this terminology thread is that you are considering the solution to be
> that the hi-entries are tagged as they
>                are added to the request.  And, just for clarification
> from my perspective, that is NOT the solution in either 4244bis or
> target-uri.
>
>
>        The principle should be that the node that has the information
> about what type of rewrite is performed should add the tag. (This is
> done by the current target-uri draft allthough the solution is not
> complete.)
>
>        For the freephone service, that would be the freephone service.
> For the AOR that is the home proxy.
>
>        Assume that the freephone service receives an INVITE  request
> that either
>        1. received request contains History-Info header that contains
> the freephone number (as in R-URI) as last entry, THEN when rewriting
> the Request-URI it has to tag that existing received entry AND add the
> new entry with the new value of the Request-URI. (I think that was how
> History-Info works right.)
>        2. received request contains History-Info header that does not
> contain the freephone number as last entry, THEN when rewriting the
> Request-URI it has add an entry with the freephone number and tag that
> entry AND add the new entry with the new value of the Request-URI.
>        3. received request does not contain a History-Info header, THEN
> when rewriting the Request-URI it has to add an entry with the freephone
> number and tag that entry AND add the new entry with the new value of
> the Request-URI.
>
>        After forwarding this request it arrives at the home proxy:
>        1. received request contains History-Info header that contains
> the AOR (as in R-URI) as last entry, THEN when rewriting the Request-URI
> it has to tag that existing received entry AND add the new entry with
> the new value of the Request-URI i.e. the registered contact. (I think
> that was how History-Info works right.)
>        2. received request contains History-Info header that does not
> contain the AOR (as in R-URI) as last entry, THEN when rewriting the
> Request-URI it has add an entry with the AOR and tag that entry AND add
> the new entry with the new value of the Request-URI i.e. the registered
> contact.
>        3. received request does not contain a History-Info header, THEN
> when rewriting the Request-URI it has to add an entry with the AOR as in
> the Request-URI and tag that entry AND add the new entry with the new
> value of the Request-URI i.e. the registered contact..
>
>
>                I don't view it as a problem necessarily if you'd like
> to pursue an alternate solution, we just need to be clear that's what
> you're advocating. In that case, I would agree the "retarget" is not an
> appropriate tag for the entries, as you don't know at the point in time
> that you add an hi-entry that it will be retargeted. In the case of this
> approach to the solution, you do have to alter the mechanism for
> determining the hi-entry that your services would want to use. You would
> need to skip the most recent/last hi-entry (as that would contain the
> same as the Request-URI in the incoming request), since it would be
> tagged if you use this solution approach.
>
>
>        Yes, but only if the previous proxy added it. As History-Info is
> optional one never knows really.
>
>
>
>                Thus, you'd have to take the next hi-entry that you find
> that has the desired "tag".  I will posit that we could accomplish this
> solution without any changes to 4244 by just registering new SIP URI
> parameter(s).   And,  that's not to say we shouldn't do a 4244bis as
> there are other issues to address in that document, but that allows the
> target-uri to describe a solution that makes use of 4244bis "as is".
>
>
>
>        Are you proposing to decouple the target-uri and the 4244bis
> discussion?
>
>
>                So, feedback on this specific point (from Hans Erik and
> others) would be appreciated.
>
>
>
>                Mary.
>
>
>
>
>

--0015174be6ea3ff725046500fba1
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hi John,<br><br>Well that is simple one would use different tags for differ=
ent type of entries, like in the example I already gave:<br><br>=A0=A0 H-I:=
=A0 Bfreephone; &quot;istarget&quot;,\<br>
=A0=A0=A0 =A0=A0=A0 B,&quot;aor&quot;\<br>
=A0=A0=A0 =A0=A0=A0 contact-B<br><br>Where the UAS interested in the initia=
l target with which it was addressed it would obtain the &quot;istarget&quo=
t; tagged entry.<br>Where the UAS interested in the AOR=A0 that was used to=
 address it would obtain the &quot;aor&quot; tagged entry.<br>
=A0<br>In simpler cases it could be that one entry holds the &quot;aor&quot=
; as well as the &quot;istarget&quot; tag.<br><br>Of course the algorithm n=
eeds some work to make it complete.<br><br>/Hans Erik van Elburg<br>
<br><br><div class=3D"gmail_quote">On Fri, Mar 13, 2009 at 3:24 PM, Elwell,=
 John <span dir=3D"ltr">&lt;<a href=3D"mailto:john.elwell@siemens.com">john=
.elwell@siemens.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quo=
te" style=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt=
 0.8ex; padding-left: 1ex;">
Hans Erik,<br>
<br>
Following your detailed description, it seems that the UAS serving a<br>
freephone number could receive one of the following (assuming H-I is<br>
supported by all relevant nodes):<br>
<br>
1) H-I containing:<br>
- the freephone number (tagged);<br>
- the AoR (tagged);<br>
- the contact URI (not tagged).<br>
<br>
Or<br>
<br>
2) H-I containing:<br>
- the freephone number (tagged);<br>
- the contact URI (not tagged).<br>
<br>
The latter would occur when the same proxy translates the freephone<br>
number (via the AoR) to the contact directly.<br>
<br>
So how does this help the UAS to figure out the freephone number? It<br>
seems the UAS would have to search back through the tagged entries<br>
looking for one that matches a freephone number it recognises? Does the<br>
tagging really add much value?<br>
<br>
John<br>
<br>
<br>
________________________________<br>
<br>
 =A0 =A0 =A0 =A0From: <a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@i=
etf.org</a> [mailto:<a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@iet=
f.org</a>] On<br>
Behalf Of Hans Erik van Elburg<br>
 =A0 =A0 =A0 =A0Sent: 12 March 2009 21:26<br>
 =A0 =A0 =A0 =A0To: Mary Barnes<br>
 =A0 =A0 =A0 =A0Cc: Hans Erik van Elburg; <a href=3D"mailto:sip@ietf.org">s=
ip@ietf.org</a><br>
 =A0 =A0 =A0 =A0Subject: Re: [Sip] Issue 3 - two tags (was RE:<br>
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br>
<div><div></div><div class=3D"h5"><br>
<br>
 =A0 =A0 =A0 =A0Inline...<br>
<br>
 =A0 =A0 =A0 =A0/Hans Erik van Elburg<br>
<br>
<br>
<br>
 =A0 =A0 =A0 =A0On Thu, Mar 12, 2009 at 7:10 PM, Mary Barnes<br>
&lt;<a href=3D"mailto:mary.barnes@nortel.com">mary.barnes@nortel.com</a>&gt=
; wrote:<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0I think there is another solution proposal =
that we<br>
should pursue for issue 3 identified in the email summary sent by Shida<br>
earlier this week - I&#39;ve snipped all the text except for that issue and=
<br>
issue 2 in the summary below. =A0I do think the discussion has confounded<b=
r>
the two issues and I think it ought to be clarified that if we agree to<br>
the two tags per issue 3, that the options for the terminology MUST<br>
change. And, perhaps this is why the other discussion threads aren&#39;t<br=
>
progressing well.<br>
<br>
<br>
 =A0 =A0 =A0 =A0Agree.<br>
<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0So, Hans Erick, I would like clarification =
on as to your<br>
view on the solution:<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0In general, ISTM that one of the issues we&=
#39;re having in<br>
this terminology thread is that you are considering the solution to be<br>
that the hi-entries are tagged as they<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0are added to the request. =A0And, just for =
clarification<br>
from my perspective, that is NOT the solution in either 4244bis or<br>
target-uri.<br>
<br>
<br>
 =A0 =A0 =A0 =A0The principle should be that the node that has the informat=
ion<br>
about what type of rewrite is performed should add the tag. (This is<br>
done by the current target-uri draft allthough the solution is not<br>
complete.)<br>
<br>
 =A0 =A0 =A0 =A0For the freephone service, that would be the freephone serv=
ice.<br>
For the AOR that is the home proxy.<br>
<br>
 =A0 =A0 =A0 =A0Assume that the freephone service receives an INVITE =A0req=
uest<br>
that either<br>
 =A0 =A0 =A0 =A01. received request contains History-Info header that conta=
ins<br>
the freephone number (as in R-URI) as last entry, THEN when rewriting<br>
the Request-URI it has to tag that existing received entry AND add the<br>
new entry with the new value of the Request-URI. (I think that was how<br>
History-Info works right.)<br>
 =A0 =A0 =A0 =A02. received request contains History-Info header that does =
not<br>
contain the freephone number as last entry, THEN when rewriting the<br>
Request-URI it has add an entry with the freephone number and tag that<br>
entry AND add the new entry with the new value of the Request-URI.<br>
 =A0 =A0 =A0 =A03. received request does not contain a History-Info header,=
 THEN<br>
when rewriting the Request-URI it has to add an entry with the freephone<br=
>
number and tag that entry AND add the new entry with the new value of<br>
the Request-URI.<br>
<br>
 =A0 =A0 =A0 =A0After forwarding this request it arrives at the home proxy:=
<br>
 =A0 =A0 =A0 =A01. received request contains History-Info header that conta=
ins<br>
the AOR (as in R-URI) as last entry, THEN when rewriting the Request-URI<br=
>
it has to tag that existing received entry AND add the new entry with<br>
the new value of the Request-URI i.e. the registered contact. (I think<br>
that was how History-Info works right.)<br>
 =A0 =A0 =A0 =A02. received request contains History-Info header that does =
not<br>
contain the AOR (as in R-URI) as last entry, THEN when rewriting the<br>
Request-URI it has add an entry with the AOR and tag that entry AND add<br>
the new entry with the new value of the Request-URI i.e. the registered<br>
contact.<br>
 =A0 =A0 =A0 =A03. received request does not contain a History-Info header,=
 THEN<br>
when rewriting the Request-URI it has to add an entry with the AOR as in<br=
>
the Request-URI and tag that entry AND add the new entry with the new<br>
value of the Request-URI i.e. the registered contact..<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0I don&#39;t view it as a problem necessaril=
y if you&#39;d like<br>
to pursue an alternate solution, we just need to be clear that&#39;s what<b=
r>
you&#39;re advocating. In that case, I would agree the &quot;retarget&quot;=
 is not an<br>
appropriate tag for the entries, as you don&#39;t know at the point in time=
<br>
that you add an hi-entry that it will be retargeted. In the case of this<br=
>
approach to the solution, you do have to alter the mechanism for<br>
determining the hi-entry that your services would want to use. You would<br=
>
need to skip the most recent/last hi-entry (as that would contain the<br>
same as the Request-URI in the incoming request), since it would be<br>
tagged if you use this solution approach.<br>
<br>
<br>
 =A0 =A0 =A0 =A0Yes, but only if the previous proxy added it. As History-In=
fo is<br>
optional one never knows really.<br>
<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Thus, you&#39;d have to take the next hi-en=
try that you find<br>
that has the desired &quot;tag&quot;. =A0I will posit that we could accompl=
ish this<br>
solution without any changes to 4244 by just registering new SIP URI<br>
parameter(s). =A0 And, =A0that&#39;s not to say we shouldn&#39;t do a 4244b=
is as<br>
there are other issues to address in that document, but that allows the<br>
target-uri to describe a solution that makes use of 4244bis &quot;as is&quo=
t;.<br>
<br>
<br>
<br>
 =A0 =A0 =A0 =A0Are you proposing to decouple the target-uri and the 4244bi=
s<br>
discussion?<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0So, feedback on this specific point (from H=
ans Erik and<br>
others) would be appreciated.<br>
<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Mary.<br>
<br>
<br>
<br>
<br>
</div></div></blockquote></div><br>

--0015174be6ea3ff725046500fba1--

From mary.barnes@nortel.com  Fri Mar 13 07:36:25 2009
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Thread-Topic: [Sip] Issue 3 - two tags (was RE: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Hans Erik van Elburg" <ietf.hanserik@gmail.com>
Cc: Hans Erik van Elburg <hanserik.van.elburg@ericsson.com>, sip@ietf.org
Subject: Re: [Sip] Issue 3 - two tags (was RE: I-D ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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So, it appears we are both talking about the solutions that are
currently described in 4244bis and target-uri (they are the same and one
in 4244bis is more complete than the one in target-uri for obvious
reasons). I view this solution to be tagging the entries with the
"action" (whatever we call it) that resulted in the uri reflected in the
hi-entry in the incoming request being overwritten by a "new"
Request-URI in the outgoing request.  Whereas, I had been concerned that
the solution you had in mind was one whereby we were tagging the "new"
Request-URI with the action that resulted in the "new" request URI.  So
your response clears my confusion, so we'll ignore the alternative
suggestion (as that would only apply to that latter view of the solution
- i.e., the one suggested below).
=20
Thanks,
Mary.=20

________________________________

From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
Sent: Thursday, March 12, 2009 4:26 PM
To: Barnes, Mary (RICH2:AR00)
Cc: Hans Erik van Elburg; sip@ietf.org
Subject: Re: [Sip] Issue 3 - two tags (was RE: I-D
ACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


Inline...

/Hans Erik van Elburg



On Thu, Mar 12, 2009 at 7:10 PM, Mary Barnes <mary.barnes@nortel.com>
wrote:


>	I think there is another solution proposal that we should pursue
for issue 3 identified in the email=20
> summary sent by Shida earlier this week - I've snipped all the text
except for that issue and issue 2 in the=20
> summary below.  I do think the discussion has confounded the two
issues and I think it ought to be clarified=20
> that if we agree to the two tags per issue 3, that the options for the
terminology MUST change. And, perhaps=20
> this is why the other discussion threads aren't progressing well.


Agree.
=20

		=20
> 	So, Hans Erick, I would like clarification on as to your view on
the solution:
	=20
> 	In general, ISTM that one of the issues we're having in this
terminology thread is that you are considering the=20
> solution to be that the hi-entries are tagged as they=20
> are added to the request.  And, just for clarification from my
perspective, that is NOT the solution in either
>  4244bis or target-uri.=20
	=20

The principle should be that the node that has the information about
what type of rewrite is performed should add the tag. (This is done by
the current target-uri draft allthough the solution is not complete.)

For the freephone service, that would be the freephone service. For the
AOR that is the home proxy.

Assume that the freephone service receives an INVITE  request that
either =20
1. received request contains History-Info header that contains the
freephone number (as in R-URI) as last entry, THEN when rewriting the
Request-URI it has to tag that existing received entry AND add the new
entry with the new value of the Request-URI. (I think that was how
History-Info works right.)
2. received request contains History-Info header that does not contain
the freephone number as last entry, THEN when rewriting the Request-URI
it has add an entry with the freephone number and tag that entry AND add
the new entry with the new value of the Request-URI.
3. received request does not contain a History-Info header, THEN when
rewriting the Request-URI it has to add an entry with the freephone
number and tag that entry AND add the new entry with the new value of
the Request-URI.

After forwarding this request it arrives at the home proxy:
1. received request contains History-Info header that contains the AOR
(as in R-URI) as last entry, THEN when rewriting the Request-URI it has
to tag that existing received entry AND add the new entry with the new
value of the Request-URI i.e. the registered contact. (I think that was
how History-Info works right.)
2. received request contains History-Info header that does not contain
the AOR (as in R-URI) as last entry, THEN when rewriting the Request-URI
it has add an entry with the AOR and tag that entry AND add the new
entry with the new value of the Request-URI i.e. the registered contact.
3. received request does not contain a History-Info header, THEN when
rewriting the Request-URI it has to add an entry with the AOR as in the
Request-URI and tag that entry AND add the new entry with the new value
of the Request-URI i.e. the registered contact..
=20

>	I don't view it as a problem necessarily if you'd like to pursue
an alternate solution, we just need to be clear=20
> that's what you're advocating. In that case, I would agree the
"retarget" is not an appropriate tag for the entries,
> as you don't know at the point in time that you add an hi-entry that
it will be retargeted. In the case of this=20
> approach to the solution, you do have to alter the mechanism for
determining the hi-entry that your services would=20
> want to use. You would need to skip the most recent/last hi-entry (as
that would contain the same as the Request-URI
 > in the incoming request), since it would be tagged if you use this
solution approach.


Yes, but only if the previous proxy added it. As History-Info is
optional one never knows really.
=20


>	Thus, you'd have to take the next hi-entry that you find that
has the desired "tag".  I will posit that we could=20
> accomplish this solution without any changes to 4244 by just
registering new SIP URI parameter(s).   And,  that's not
> to say we shouldn't do a 4244bis as there are other issues to address
in that document, but that allows the target-uri
>  to describe a solution that makes use of 4244bis "as is".
=09
	=20

Are you proposing to decouple the target-uri and the 4244bis discussion?
=20

>	So, feedback on this specific point (from Hans Erik and others)
would be appreciated.=20
=09
=09
=09
	Mary.





From john.elwell@siemens.com  Fri Mar 13 07:39:27 2009
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Date: Fri, 13 Mar 2009 14:39:48 +0000
From: "Elwell, John" <john.elwell@siemens.com>
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 Hans Erik,

But how does a proxy know what to use? How does it know whether the
received Request-URI, which it uses to query the LS, is a target or an
AoR? Does it need to scan back up the H-I chain to see whether there is
already an 'istarget' present, and if so use 'aor' instead of
'istarget'?

John


________________________________

	From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
	Sent: 13 March 2009 14:33
	To: Elwell, John
	Cc: sip@ietf.org
	Subject: Re: [Sip] Issue 3 - two tags (was RE:
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
=09
=09
	Hi John,
=09
	Well that is simple one would use different tags for different
type of entries, like in the example I already gave:
=09
	   H-I:  Bfreephone; "istarget",\
	        B,"aor"\
	        contact-B
=09
	Where the UAS interested in the initial target with which it was
addressed it would obtain the "istarget" tagged entry.
	Where the UAS interested in the AOR  that was used to address it
would obtain the "aor" tagged entry.
	=20
	In simpler cases it could be that one entry holds the "aor" as
well as the "istarget" tag.
=09
	Of course the algorithm needs some work to make it complete.
=09
	/Hans Erik van Elburg
=09
=09
=09
	On Fri, Mar 13, 2009 at 3:24 PM, Elwell, John
<john.elwell@siemens.com> wrote:
=09

		Hans Erik,
	=09
		Following your detailed description, it seems that the
UAS serving a
		freephone number could receive one of the following
(assuming H-I is
		supported by all relevant nodes):
	=09
		1) H-I containing:
		- the freephone number (tagged);
		- the AoR (tagged);
		- the contact URI (not tagged).
	=09
		Or
	=09
		2) H-I containing:
		- the freephone number (tagged);
		- the contact URI (not tagged).
	=09
		The latter would occur when the same proxy translates
the freephone
		number (via the AoR) to the contact directly.
	=09
		So how does this help the UAS to figure out the
freephone number? It
		seems the UAS would have to search back through the
tagged entries
		looking for one that matches a freephone number it
recognises? Does the
		tagging really add much value?
	=09
		John
	=09
	=09
		________________________________
	=09
		       From: sip-bounces@ietf.org
[mailto:sip-bounces@ietf.org] On
		Behalf Of Hans Erik van Elburg
		       Sent: 12 March 2009 21:26
		       To: Mary Barnes
		       Cc: Hans Erik van Elburg; sip@ietf.org
		       Subject: Re: [Sip] Issue 3 - two tags (was RE:
		I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
	=09


		       Inline...
	=09
		       /Hans Erik van Elburg
	=09
	=09
	=09
		       On Thu, Mar 12, 2009 at 7:10 PM, Mary Barnes
		<mary.barnes@nortel.com> wrote:
	=09
	=09
		               I think there is another solution
proposal that we
		should pursue for issue 3 identified in the email
summary sent by Shida
		earlier this week - I've snipped all the text except for
that issue and
		issue 2 in the summary below.  I do think the discussion
has confounded
		the two issues and I think it ought to be clarified that
if we agree to
		the two tags per issue 3, that the options for the
terminology MUST
		change. And, perhaps this is why the other discussion
threads aren't
		progressing well.
	=09
	=09
		       Agree.
	=09
	=09
	=09
		               So, Hans Erick, I would like
clarification on as to your
		view on the solution:
	=09
		               In general, ISTM that one of the issues
we're having in
		this terminology thread is that you are considering the
solution to be
		that the hi-entries are tagged as they
		               are added to the request.  And, just for
clarification
		from my perspective, that is NOT the solution in either
4244bis or
		target-uri.
	=09
	=09
		       The principle should be that the node that has
the information
		about what type of rewrite is performed should add the
tag. (This is
		done by the current target-uri draft allthough the
solution is not
		complete.)
	=09
		       For the freephone service, that would be the
freephone service.
		For the AOR that is the home proxy.
	=09
		       Assume that the freephone service receives an
INVITE  request
		that either
		       1. received request contains History-Info header
that contains
		the freephone number (as in R-URI) as last entry, THEN
when rewriting
		the Request-URI it has to tag that existing received
entry AND add the
		new entry with the new value of the Request-URI. (I
think that was how
		History-Info works right.)
		       2. received request contains History-Info header
that does not
		contain the freephone number as last entry, THEN when
rewriting the
		Request-URI it has add an entry with the freephone
number and tag that
		entry AND add the new entry with the new value of the
Request-URI.
		       3. received request does not contain a
History-Info header, THEN
		when rewriting the Request-URI it has to add an entry
with the freephone
		number and tag that entry AND add the new entry with the
new value of
		the Request-URI.
	=09
		       After forwarding this request it arrives at the
home proxy:
		       1. received request contains History-Info header
that contains
		the AOR (as in R-URI) as last entry, THEN when rewriting
the Request-URI
		it has to tag that existing received entry AND add the
new entry with
		the new value of the Request-URI i.e. the registered
contact. (I think
		that was how History-Info works right.)
		       2. received request contains History-Info header
that does not
		contain the AOR (as in R-URI) as last entry, THEN when
rewriting the
		Request-URI it has add an entry with the AOR and tag
that entry AND add
		the new entry with the new value of the Request-URI i.e.
the registered
		contact.
		       3. received request does not contain a
History-Info header, THEN
		when rewriting the Request-URI it has to add an entry
with the AOR as in
		the Request-URI and tag that entry AND add the new entry
with the new
		value of the Request-URI i.e. the registered contact..
	=09
	=09
		               I don't view it as a problem necessarily
if you'd like
		to pursue an alternate solution, we just need to be
clear that's what
		you're advocating. In that case, I would agree the
"retarget" is not an
		appropriate tag for the entries, as you don't know at
the point in time
		that you add an hi-entry that it will be retargeted. In
the case of this
		approach to the solution, you do have to alter the
mechanism for
		determining the hi-entry that your services would want
to use. You would
		need to skip the most recent/last hi-entry (as that
would contain the
		same as the Request-URI in the incoming request), since
it would be
		tagged if you use this solution approach.
	=09
	=09
		       Yes, but only if the previous proxy added it. As
History-Info is
		optional one never knows really.
	=09
	=09
	=09
		               Thus, you'd have to take the next
hi-entry that you find
		that has the desired "tag".  I will posit that we could
accomplish this
		solution without any changes to 4244 by just registering
new SIP URI
		parameter(s).   And,  that's not to say we shouldn't do
a 4244bis as
		there are other issues to address in that document, but
that allows the
		target-uri to describe a solution that makes use of
4244bis "as is".
	=09
	=09
	=09
		       Are you proposing to decouple the target-uri and
the 4244bis
		discussion?
	=09
	=09
		               So, feedback on this specific point (from
Hans Erik and
		others) would be appreciated.
	=09
	=09
	=09
		               Mary.
	=09
	=09
	=09
	=09
	=09



From ietf.hanserik@gmail.com  Fri Mar 13 07:51:28 2009
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Date: Fri, 13 Mar 2009 15:51:59 +0100
Message-ID: <9ae56b1e0903130751i5c9718abwbd44a6371130d258@mail.gmail.com>
From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: "Elwell, John" <john.elwell@siemens.com>
Content-Type: multipart/alternative; boundary=0015174bde8432cd740465013eae
Cc: sip@ietf.org
Subject: Re: [Sip] Issue 3 - two tags (was RE: I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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--0015174bde8432cd740465013eae
Content-Type: text/plain; charset=ISO-8859-1
Content-Transfer-Encoding: 7bit

It knows when it rewrites the Request-URI due to a location lookup --> tag
"aor".
It knows (is configured to) when it rewrites the Request-URI with a value
that leads the request to target with a different identity then the one that
has been replaced --> tag new entry "istarget".
It knows when it rewrites the Request-URI with a value and this leads to the
first History-Info entry being recorded --> tag "istarget".

/Hans Erik van Elburg


On Fri, Mar 13, 2009 at 3:39 PM, Elwell, John <john.elwell@siemens.com>wrote:

>  Hans Erik,
>
> But how does a proxy know what to use? How does it know whether the
> received Request-URI, which it uses to query the LS, is a target or an
> AoR? Does it need to scan back up the H-I chain to see whether there is
> already an 'istarget' present, and if so use 'aor' instead of
> 'istarget'?
>
> John
>
>
> ________________________________
>
>        From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
>         Sent: 13 March 2009 14:33
>        To: Elwell, John
>        Cc: sip@ietf.org
>         Subject: Re: [Sip] Issue 3 - two tags (was RE:
> I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>
>
>        Hi John,
>
>        Well that is simple one would use different tags for different
> type of entries, like in the example I already gave:
>
>           H-I:  Bfreephone; "istarget",\
>                B,"aor"\
>                contact-B
>
>        Where the UAS interested in the initial target with which it was
> addressed it would obtain the "istarget" tagged entry.
>        Where the UAS interested in the AOR  that was used to address it
> would obtain the "aor" tagged entry.
>
>        In simpler cases it could be that one entry holds the "aor" as
> well as the "istarget" tag.
>
>        Of course the algorithm needs some work to make it complete.
>
>        /Hans Erik van Elburg
>
>
>
>        On Fri, Mar 13, 2009 at 3:24 PM, Elwell, John
> <john.elwell@siemens.com> wrote:
>
>
>                Hans Erik,
>
>                Following your detailed description, it seems that the
> UAS serving a
>                freephone number could receive one of the following
> (assuming H-I is
>                supported by all relevant nodes):
>
>                1) H-I containing:
>                - the freephone number (tagged);
>                - the AoR (tagged);
>                - the contact URI (not tagged).
>
>                Or
>
>                2) H-I containing:
>                - the freephone number (tagged);
>                - the contact URI (not tagged).
>
>                The latter would occur when the same proxy translates
> the freephone
>                number (via the AoR) to the contact directly.
>
>                So how does this help the UAS to figure out the
> freephone number? It
>                seems the UAS would have to search back through the
> tagged entries
>                looking for one that matches a freephone number it
> recognises? Does the
>                tagging really add much value?
>
>                John
>
>
>                ________________________________
>
>                       From: sip-bounces@ietf.org
> [mailto:sip-bounces@ietf.org] On
>                Behalf Of Hans Erik van Elburg
>                       Sent: 12 March 2009 21:26
>                       To: Mary Barnes
>                       Cc: Hans Erik van Elburg; sip@ietf.org
>                       Subject: Re: [Sip] Issue 3 - two tags (was RE:
>                I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>
>
>
>                       Inline...
>
>                       /Hans Erik van Elburg
>
>
>
>                       On Thu, Mar 12, 2009 at 7:10 PM, Mary Barnes
>                <mary.barnes@nortel.com> wrote:
>
>
>                               I think there is another solution
> proposal that we
>                should pursue for issue 3 identified in the email
> summary sent by Shida
>                earlier this week - I've snipped all the text except for
> that issue and
>                issue 2 in the summary below.  I do think the discussion
> has confounded
>                the two issues and I think it ought to be clarified that
> if we agree to
>                the two tags per issue 3, that the options for the
> terminology MUST
>                change. And, perhaps this is why the other discussion
> threads aren't
>                progressing well.
>
>
>                       Agree.
>
>
>
>                               So, Hans Erick, I would like
> clarification on as to your
>                view on the solution:
>
>                               In general, ISTM that one of the issues
> we're having in
>                this terminology thread is that you are considering the
> solution to be
>                that the hi-entries are tagged as they
>                               are added to the request.  And, just for
> clarification
>                from my perspective, that is NOT the solution in either
> 4244bis or
>                target-uri.
>
>
>                       The principle should be that the node that has
> the information
>                about what type of rewrite is performed should add the
> tag. (This is
>                done by the current target-uri draft allthough the
> solution is not
>                complete.)
>
>                       For the freephone service, that would be the
> freephone service.
>                For the AOR that is the home proxy.
>
>                       Assume that the freephone service receives an
> INVITE  request
>                that either
>                       1. received request contains History-Info header
> that contains
>                the freephone number (as in R-URI) as last entry, THEN
> when rewriting
>                the Request-URI it has to tag that existing received
> entry AND add the
>                new entry with the new value of the Request-URI. (I
> think that was how
>                History-Info works right.)
>                       2. received request contains History-Info header
> that does not
>                contain the freephone number as last entry, THEN when
> rewriting the
>                Request-URI it has add an entry with the freephone
> number and tag that
>                entry AND add the new entry with the new value of the
> Request-URI.
>                       3. received request does not contain a
> History-Info header, THEN
>                when rewriting the Request-URI it has to add an entry
> with the freephone
>                number and tag that entry AND add the new entry with the
> new value of
>                the Request-URI.
>
>                       After forwarding this request it arrives at the
> home proxy:
>                       1. received request contains History-Info header
> that contains
>                the AOR (as in R-URI) as last entry, THEN when rewriting
> the Request-URI
>                it has to tag that existing received entry AND add the
> new entry with
>                the new value of the Request-URI i.e. the registered
> contact. (I think
>                that was how History-Info works right.)
>                       2. received request contains History-Info header
> that does not
>                contain the AOR (as in R-URI) as last entry, THEN when
> rewriting the
>                Request-URI it has add an entry with the AOR and tag
> that entry AND add
>                the new entry with the new value of the Request-URI i.e.
> the registered
>                contact.
>                       3. received request does not contain a
> History-Info header, THEN
>                when rewriting the Request-URI it has to add an entry
> with the AOR as in
>                the Request-URI and tag that entry AND add the new entry
> with the new
>                value of the Request-URI i.e. the registered contact..
>
>
>                               I don't view it as a problem necessarily
> if you'd like
>                to pursue an alternate solution, we just need to be
> clear that's what
>                you're advocating. In that case, I would agree the
> "retarget" is not an
>                appropriate tag for the entries, as you don't know at
> the point in time
>                that you add an hi-entry that it will be retargeted. In
> the case of this
>                approach to the solution, you do have to alter the
> mechanism for
>                determining the hi-entry that your services would want
> to use. You would
>                need to skip the most recent/last hi-entry (as that
> would contain the
>                same as the Request-URI in the incoming request), since
> it would be
>                tagged if you use this solution approach.
>
>
>                       Yes, but only if the previous proxy added it. As
> History-Info is
>                optional one never knows really.
>
>
>
>                               Thus, you'd have to take the next
> hi-entry that you find
>                that has the desired "tag".  I will posit that we could
> accomplish this
>                solution without any changes to 4244 by just registering
> new SIP URI
>                parameter(s).   And,  that's not to say we shouldn't do
> a 4244bis as
>                there are other issues to address in that document, but
> that allows the
>                target-uri to describe a solution that makes use of
> 4244bis "as is".
>
>
>
>                       Are you proposing to decouple the target-uri and
> the 4244bis
>                discussion?
>
>
>                               So, feedback on this specific point (from
> Hans Erik and
>                others) would be appreciated.
>
>
>
>                               Mary.
>
>
>
>
>
>
>
>

--0015174bde8432cd740465013eae
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

It knows when it rewrites the Request-URI due to a location lookup --&gt; t=
ag &quot;aor&quot;.<br>It knows (is configured to) when it rewrites the Req=
uest-URI with a value that leads the request to target with a different ide=
ntity then the one that has been replaced --&gt; tag new entry &quot;istarg=
et&quot;.<br>
It knows when it rewrites the Request-URI with a value and this leads to th=
e first History-Info entry being recorded --&gt; tag &quot;istarget&quot;.<=
br><br clear=3D"all">/Hans Erik van Elburg<br>
<br><br><div class=3D"gmail_quote">On Fri, Mar 13, 2009 at 3:39 PM, Elwell,=
 John <span dir=3D"ltr">&lt;<a href=3D"mailto:john.elwell@siemens.com">john=
.elwell@siemens.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quo=
te" style=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt=
 0.8ex; padding-left: 1ex;">
=A0Hans Erik,<br>
<br>
But how does a proxy know what to use? How does it know whether the<br>
received Request-URI, which it uses to query the LS, is a target or an<br>
AoR? Does it need to scan back up the H-I chain to see whether there is<br>
already an &#39;istarget&#39; present, and if so use &#39;aor&#39; instead =
of<br>
&#39;istarget&#39;?<br>
<br>
John<br>
<div class=3D"im"><br>
<br>
________________________________<br>
<br>
 =A0 =A0 =A0 =A0From: Hans Erik van Elburg [mailto:<a href=3D"mailto:ietf.h=
anserik@gmail.com">ietf.hanserik@gmail.com</a>]<br>
</div> =A0 =A0 =A0 =A0Sent: 13 March 2009 14:33<br>
 =A0 =A0 =A0 =A0To: Elwell, John<br>
 =A0 =A0 =A0 =A0Cc: <a href=3D"mailto:sip@ietf.org">sip@ietf.org</a><br>
<div><div></div><div class=3D"h5"> =A0 =A0 =A0 =A0Subject: Re: [Sip] Issue =
3 - two tags (was RE:<br>
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br>
<br>
<br>
 =A0 =A0 =A0 =A0Hi John,<br>
<br>
 =A0 =A0 =A0 =A0Well that is simple one would use different tags for differ=
ent<br>
type of entries, like in the example I already gave:<br>
<br>
 =A0 =A0 =A0 =A0 =A0 H-I: =A0Bfreephone; &quot;istarget&quot;,\<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0B,&quot;aor&quot;\<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0contact-B<br>
<br>
 =A0 =A0 =A0 =A0Where the UAS interested in the initial target with which i=
t was<br>
addressed it would obtain the &quot;istarget&quot; tagged entry.<br>
 =A0 =A0 =A0 =A0Where the UAS interested in the AOR =A0that was used to add=
ress it<br>
would obtain the &quot;aor&quot; tagged entry.<br>
<br>
 =A0 =A0 =A0 =A0In simpler cases it could be that one entry holds the &quot=
;aor&quot; as<br>
well as the &quot;istarget&quot; tag.<br>
<br>
 =A0 =A0 =A0 =A0Of course the algorithm needs some work to make it complete=
.<br>
<br>
 =A0 =A0 =A0 =A0/Hans Erik van Elburg<br>
<br>
<br>
<br>
 =A0 =A0 =A0 =A0On Fri, Mar 13, 2009 at 3:24 PM, Elwell, John<br>
&lt;<a href=3D"mailto:john.elwell@siemens.com">john.elwell@siemens.com</a>&=
gt; wrote:<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Hans Erik,<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Following your detailed description, it see=
ms that the<br>
UAS serving a<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0freephone number could receive one of the f=
ollowing<br>
(assuming H-I is<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0supported by all relevant nodes):<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A01) H-I containing:<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0- the freephone number (tagged);<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0- the AoR (tagged);<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0- the contact URI (not tagged).<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Or<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A02) H-I containing:<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0- the freephone number (tagged);<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0- the contact URI (not tagged).<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0The latter would occur when the same proxy =
translates<br>
the freephone<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0number (via the AoR) to the contact directl=
y.<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0So how does this help the UAS to figure out=
 the<br>
freephone number? It<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0seems the UAS would have to search back thr=
ough the<br>
tagged entries<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0looking for one that matches a freephone nu=
mber it<br>
recognises? Does the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0tagging really add much value?<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0John<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0________________________________<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 From: <a href=3D"mailto:sip-bo=
unces@ietf.org">sip-bounces@ietf.org</a><br>
[mailto:<a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a>] O=
n<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Behalf Of Hans Erik van Elburg<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Sent: 12 March 2009 21:26<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 To: Mary Barnes<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Cc: Hans Erik van Elburg; <a h=
ref=3D"mailto:sip@ietf.org">sip@ietf.org</a><br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Subject: Re: [Sip] Issue 3 - t=
wo tags (was RE:<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0I-DACTION:draft-rosenberg-sip-target-uri-de=
livery-01.txt<br>
<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Inline...<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 /Hans Erik van Elburg<br>
<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 On Thu, Mar 12, 2009 at 7:10 P=
M, Mary Barnes<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&lt;<a href=3D"mailto:mary.barnes@nortel.co=
m">mary.barnes@nortel.com</a>&gt; wrote:<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 I think there =
is another solution<br>
proposal that we<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0should pursue for issue 3 identified in the=
 email<br>
summary sent by Shida<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0earlier this week - I&#39;ve snipped all th=
e text except for<br>
that issue and<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0issue 2 in the summary below. =A0I do think=
 the discussion<br>
has confounded<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the two issues and I think it ought to be c=
larified that<br>
if we agree to<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the two tags per issue 3, that the options =
for the<br>
terminology MUST<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0change. And, perhaps this is why the other =
discussion<br>
threads aren&#39;t<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0progressing well.<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Agree.<br>
<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 So, Hans Erick=
, I would like<br>
clarification on as to your<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0view on the solution:<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 In general, IS=
TM that one of the issues<br>
we&#39;re having in<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0this terminology thread is that you are con=
sidering the<br>
solution to be<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0that the hi-entries are tagged as they<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 are added to t=
he request. =A0And, just for<br>
clarification<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0from my perspective, that is NOT the soluti=
on in either<br>
4244bis or<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0target-uri.<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 The principle should be that t=
he node that has<br>
the information<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0about what type of rewrite is performed sho=
uld add the<br>
tag. (This is<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0done by the current target-uri draft alltho=
ugh the<br>
solution is not<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0complete.)<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 For the freephone service, tha=
t would be the<br>
freephone service.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0For the AOR that is the home proxy.<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Assume that the freephone serv=
ice receives an<br>
INVITE =A0request<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0that either<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 1. received request contains H=
istory-Info header<br>
that contains<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the freephone number (as in R-URI) as last =
entry, THEN<br>
when rewriting<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the Request-URI it has to tag that existing=
 received<br>
entry AND add the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0new entry with the new value of the Request=
-URI. (I<br>
think that was how<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0History-Info works right.)<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 2. received request contains H=
istory-Info header<br>
that does not<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0contain the freephone number as last entry,=
 THEN when<br>
rewriting the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Request-URI it has add an entry with the fr=
eephone<br>
number and tag that<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0entry AND add the new entry with the new va=
lue of the<br>
Request-URI.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 3. received request does not c=
ontain a<br>
History-Info header, THEN<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0when rewriting the Request-URI it has to ad=
d an entry<br>
with the freephone<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0number and tag that entry AND add the new e=
ntry with the<br>
new value of<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the Request-URI.<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 After forwarding this request =
it arrives at the<br>
home proxy:<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 1. received request contains H=
istory-Info header<br>
that contains<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the AOR (as in R-URI) as last entry, THEN w=
hen rewriting<br>
the Request-URI<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0it has to tag that existing received entry =
AND add the<br>
new entry with<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the new value of the Request-URI i.e. the r=
egistered<br>
contact. (I think<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0that was how History-Info works right.)<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 2. received request contains H=
istory-Info header<br>
that does not<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0contain the AOR (as in R-URI) as last entry=
, THEN when<br>
rewriting the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Request-URI it has add an entry with the AO=
R and tag<br>
that entry AND add<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the new entry with the new value of the Req=
uest-URI i.e.<br>
the registered<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0contact.<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 3. received request does not c=
ontain a<br>
History-Info header, THEN<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0when rewriting the Request-URI it has to ad=
d an entry<br>
with the AOR as in<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the Request-URI and tag that entry AND add =
the new entry<br>
with the new<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0value of the Request-URI i.e. the registere=
d contact..<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 I don&#39;t vi=
ew it as a problem necessarily<br>
if you&#39;d like<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0to pursue an alternate solution, we just ne=
ed to be<br>
clear that&#39;s what<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0you&#39;re advocating. In that case, I woul=
d agree the<br>
&quot;retarget&quot; is not an<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0appropriate tag for the entries, as you don=
&#39;t know at<br>
the point in time<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0that you add an hi-entry that it will be re=
targeted. In<br>
the case of this<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0approach to the solution, you do have to al=
ter the<br>
mechanism for<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0determining the hi-entry that your services=
 would want<br>
to use. You would<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0need to skip the most recent/last hi-entry =
(as that<br>
would contain the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0same as the Request-URI in the incoming req=
uest), since<br>
it would be<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0tagged if you use this solution approach.<b=
r>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Yes, but only if the previous =
proxy added it. As<br>
History-Info is<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0optional one never knows really.<br>
<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Thus, you&#39;=
d have to take the next<br>
hi-entry that you find<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0that has the desired &quot;tag&quot;. =A0I =
will posit that we could<br>
accomplish this<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0solution without any changes to 4244 by jus=
t registering<br>
new SIP URI<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0parameter(s). =A0 And, =A0that&#39;s not to=
 say we shouldn&#39;t do<br>
a 4244bis as<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0there are other issues to address in that d=
ocument, but<br>
that allows the<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0target-uri to describe a solution that make=
s use of<br>
4244bis &quot;as is&quot;.<br>
<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Are you proposing to decouple =
the target-uri and<br>
the 4244bis<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0discussion?<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 So, feedback o=
n this specific point (from<br>
Hans Erik and<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0others) would be appreciated.<br>
<br>
<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Mary.<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
</div></div></blockquote></div><br>

--0015174bde8432cd740465013eae--

From john.elwell@siemens.com  Fri Mar 13 09:34:08 2009
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From: "Elwell, John" <john.elwell@siemens.com>
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Subject: Re: [Sip] Issue 3 - two tags (was RE: I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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=20

> -----Original Message-----
> From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
> Sent: 13 March 2009 14:52
> To: Elwell, John
> Cc: sip@ietf.org
> Subject: Re: [Sip] Issue 3 - two tags (was RE:=20
> I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>=20
> It knows when it rewrites the Request-URI due to a location=20
> lookup --> tag "aor".
> It knows (is configured to) when it rewrites the Request-URI=20
> with a value that leads the request to target with a=20
> different identity then the one that has been replaced -->=20
> tag new entry "istarget".
> It knows when it rewrites the Request-URI with a value and=20
> this leads to the first History-Info entry being recorded -->=20
> tag "istarget".
[JRE] It does not know that this is the first target, because upstream
nodes may have retargeted without inserting H-I. In any case, the To
header field URI is the first target, so we don't need that anyway.

I can accept that a proxy might know, having done a LS look-up, whether
the new URI is a registered contact or some other URI, and on that basis
it could choose whether or not to insert a single tag. I thought that
was the basis on which we agreed to go ahead with this updated to RFC
4424, rather than following the loose route draft. It is not clear to me
that a proxy would have enough information to make a 3-way choice.

John


>=20
> /Hans Erik van Elburg
>=20
>=20
>=20
> On Fri, Mar 13, 2009 at 3:39 PM, Elwell, John=20
> <john.elwell@siemens.com> wrote:
>=20
>=20
> 	 Hans Erik,
> =09
> 	But how does a proxy know what to use? How does it know=20
> whether the
> 	received Request-URI, which it uses to query the LS, is=20
> a target or an
> 	AoR? Does it need to scan back up the H-I chain to see=20
> whether there is
> 	already an 'istarget' present, and if so use 'aor' instead of
> 	'istarget'?
> =09
> 	John
> =09
>=20
>=20
> 	________________________________
> =09
> 	       From: Hans Erik van Elburg=20
> [mailto:ietf.hanserik@gmail.com]
> =09
> 	       Sent: 13 March 2009 14:33
> 	       To: Elwell, John
> 	       Cc: sip@ietf.org
> =09
> 	       Subject: Re: [Sip] Issue 3 - two tags (was RE:
> 	I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
> =09
> =09
> 	       Hi John,
> =09
> 	       Well that is simple one would use different tags=20
> for different
> 	type of entries, like in the example I already gave:
> =09
> 	          H-I:  Bfreephone; "istarget",\
> 	               B,"aor"\
> 	               contact-B
> =09
> 	       Where the UAS interested in the initial target=20
> with which it was
> 	addressed it would obtain the "istarget" tagged entry.
> 	       Where the UAS interested in the AOR  that was=20
> used to address it
> 	would obtain the "aor" tagged entry.
> =09
> 	       In simpler cases it could be that one entry=20
> holds the "aor" as
> 	well as the "istarget" tag.
> =09
> 	       Of course the algorithm needs some work to make=20
> it complete.
> =09
> 	       /Hans Erik van Elburg
> =09
> =09
> =09
> 	       On Fri, Mar 13, 2009 at 3:24 PM, Elwell, John
> 	<john.elwell@siemens.com> wrote:
> =09
> =09
> 	               Hans Erik,
> =09
> 	               Following your detailed description, it=20
> seems that the
> 	UAS serving a
> 	               freephone number could receive one of=20
> the following
> 	(assuming H-I is
> 	               supported by all relevant nodes):
> =09
> 	               1) H-I containing:
> 	               - the freephone number (tagged);
> 	               - the AoR (tagged);
> 	               - the contact URI (not tagged).
> =09
> 	               Or
> =09
> 	               2) H-I containing:
> 	               - the freephone number (tagged);
> 	               - the contact URI (not tagged).
> =09
> 	               The latter would occur when the same=20
> proxy translates
> 	the freephone
> 	               number (via the AoR) to the contact directly.
> =09
> 	               So how does this help the UAS to figure out the
> 	freephone number? It
> 	               seems the UAS would have to search back=20
> through the
> 	tagged entries
> 	               looking for one that matches a freephone=20
> number it
> 	recognises? Does the
> 	               tagging really add much value?
> =09
> 	               John
> =09
> =09
> 	               ________________________________
> =09
> 	                      From: sip-bounces@ietf.org
> 	[mailto:sip-bounces@ietf.org] On
> 	               Behalf Of Hans Erik van Elburg
> 	                      Sent: 12 March 2009 21:26
> 	                      To: Mary Barnes
> 	                      Cc: Hans Erik van Elburg; sip@ietf.org
> 	                      Subject: Re: [Sip] Issue 3 - two=20
> tags (was RE:
> 	              =20
> I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
> =09
> =09
> =09
> 	                      Inline...
> =09
> 	                      /Hans Erik van Elburg
> =09
> =09
> =09
> 	                      On Thu, Mar 12, 2009 at 7:10 PM,=20
> Mary Barnes
> 	               <mary.barnes@nortel.com> wrote:
> =09
> =09
> 	                              I think there is another solution
> 	proposal that we
> 	               should pursue for issue 3 identified in the email
> 	summary sent by Shida
> 	               earlier this week - I've snipped all the=20
> text except for
> 	that issue and
> 	               issue 2 in the summary below.  I do=20
> think the discussion
> 	has confounded
> 	               the two issues and I think it ought to=20
> be clarified that
> 	if we agree to
> 	               the two tags per issue 3, that the=20
> options for the
> 	terminology MUST
> 	               change. And, perhaps this is why the=20
> other discussion
> 	threads aren't
> 	               progressing well.
> =09
> =09
> 	                      Agree.
> =09
> =09
> =09
> 	                              So, Hans Erick, I would like
> 	clarification on as to your
> 	               view on the solution:
> =09
> 	                              In general, ISTM that one=20
> of the issues
> 	we're having in
> 	               this terminology thread is that you are=20
> considering the
> 	solution to be
> 	               that the hi-entries are tagged as they
> 	                              are added to the request.=20
>  And, just for
> 	clarification
> 	               from my perspective, that is NOT the=20
> solution in either
> 	4244bis or
> 	               target-uri.
> =09
> =09
> 	                      The principle should be that the=20
> node that has
> 	the information
> 	               about what type of rewrite is performed=20
> should add the
> 	tag. (This is
> 	               done by the current target-uri draft=20
> allthough the
> 	solution is not
> 	               complete.)
> =09
> 	                      For the freephone service, that=20
> would be the
> 	freephone service.
> 	               For the AOR that is the home proxy.
> =09
> 	                      Assume that the freephone service=20
> receives an
> 	INVITE  request
> 	               that either
> 	                      1. received request contains=20
> History-Info header
> 	that contains
> 	               the freephone number (as in R-URI) as=20
> last entry, THEN
> 	when rewriting
> 	               the Request-URI it has to tag that=20
> existing received
> 	entry AND add the
> 	               new entry with the new value of the=20
> Request-URI. (I
> 	think that was how
> 	               History-Info works right.)
> 	                      2. received request contains=20
> History-Info header
> 	that does not
> 	               contain the freephone number as last=20
> entry, THEN when
> 	rewriting the
> 	               Request-URI it has add an entry with the=20
> freephone
> 	number and tag that
> 	               entry AND add the new entry with the new=20
> value of the
> 	Request-URI.
> 	                      3. received request does not contain a
> 	History-Info header, THEN
> 	               when rewriting the Request-URI it has to=20
> add an entry
> 	with the freephone
> 	               number and tag that entry AND add the=20
> new entry with the
> 	new value of
> 	               the Request-URI.
> =09
> 	                      After forwarding this request it=20
> arrives at the
> 	home proxy:
> 	                      1. received request contains=20
> History-Info header
> 	that contains
> 	               the AOR (as in R-URI) as last entry,=20
> THEN when rewriting
> 	the Request-URI
> 	               it has to tag that existing received=20
> entry AND add the
> 	new entry with
> 	               the new value of the Request-URI i.e.=20
> the registered
> 	contact. (I think
> 	               that was how History-Info works right.)
> 	                      2. received request contains=20
> History-Info header
> 	that does not
> 	               contain the AOR (as in R-URI) as last=20
> entry, THEN when
> 	rewriting the
> 	               Request-URI it has add an entry with the=20
> AOR and tag
> 	that entry AND add
> 	               the new entry with the new value of the=20
> Request-URI i.e.
> 	the registered
> 	               contact.
> 	                      3. received request does not contain a
> 	History-Info header, THEN
> 	               when rewriting the Request-URI it has to=20
> add an entry
> 	with the AOR as in
> 	               the Request-URI and tag that entry AND=20
> add the new entry
> 	with the new
> 	               value of the Request-URI i.e. the=20
> registered contact..
> =09
> =09
> 	                              I don't view it as a=20
> problem necessarily
> 	if you'd like
> 	               to pursue an alternate solution, we just=20
> need to be
> 	clear that's what
> 	               you're advocating. In that case, I would=20
> agree the
> 	"retarget" is not an
> 	               appropriate tag for the entries, as you=20
> don't know at
> 	the point in time
> 	               that you add an hi-entry that it will be=20
> retargeted. In
> 	the case of this
> 	               approach to the solution, you do have to=20
> alter the
> 	mechanism for
> 	               determining the hi-entry that your=20
> services would want
> 	to use. You would
> 	               need to skip the most recent/last=20
> hi-entry (as that
> 	would contain the
> 	               same as the Request-URI in the incoming=20
> request), since
> 	it would be
> 	               tagged if you use this solution approach.
> =09
> =09
> 	                      Yes, but only if the previous=20
> proxy added it. As
> 	History-Info is
> 	               optional one never knows really.
> =09
> =09
> =09
> 	                              Thus, you'd have to take the next
> 	hi-entry that you find
> 	               that has the desired "tag".  I will=20
> posit that we could
> 	accomplish this
> 	               solution without any changes to 4244 by=20
> just registering
> 	new SIP URI
> 	               parameter(s).   And,  that's not to say=20
> we shouldn't do
> 	a 4244bis as
> 	               there are other issues to address in=20
> that document, but
> 	that allows the
> 	               target-uri to describe a solution that=20
> makes use of
> 	4244bis "as is".
> =09
> =09
> =09
> 	                      Are you proposing to decouple the=20
> target-uri and
> 	the 4244bis
> 	               discussion?
> =09
> =09
> 	                              So, feedback on this=20
> specific point (from
> 	Hans Erik and
> 	               others) would be appreciated.
> =09
> =09
> =09
> 	                              Mary.
> =09
> =09
> =09
> =09
> =09
> =09
> =09
> =09
>=20
>=20
>=20

From ietf.hanserik@gmail.com  Fri Mar 13 09:58:37 2009
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Date: Fri, 13 Mar 2009 17:59:12 +0100
Message-ID: <9ae56b1e0903130959h5a142c55v335600dc93041fe6@mail.gmail.com>
From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: "Elwell, John" <john.elwell@siemens.com>
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Cc: sip@ietf.org
Subject: Re: [Sip] Issue 3 - two tags (was RE: I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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3-way choice?

A proxy that has retargeting logic knows when it is doing so.
A proxy that has rerouting logic to registered contact knows when it is
doing so.

No 3-way magic there.

And yes the History-Info is optional so you never know if all nodes in the
chain do there bit. But that is fact of life. As long as you can ensure that
in your domain it is supported you will receive  correct information.

/Hans Erik van Elburg


On Fri, Mar 13, 2009 at 5:31 PM, Elwell, John <john.elwell@siemens.com>wrote:

>
>
> > -----Original Message-----
> > From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
> > Sent: 13 March 2009 14:52
> > To: Elwell, John
> > Cc: sip@ietf.org
> > Subject: Re: [Sip] Issue 3 - two tags (was RE:
> > I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
> >
> > It knows when it rewrites the Request-URI due to a location
> > lookup --> tag "aor".
> > It knows (is configured to) when it rewrites the Request-URI
> > with a value that leads the request to target with a
> > different identity then the one that has been replaced -->
> > tag new entry "istarget".
> > It knows when it rewrites the Request-URI with a value and
> > this leads to the first History-Info entry being recorded -->
> > tag "istarget".
> [JRE] It does not know that this is the first target, because upstream
> nodes may have retargeted without inserting H-I. In any case, the To
> header field URI is the first target, so we don't need that anyway.
>
> I can accept that a proxy might know, having done a LS look-up, whether
> the new URI is a registered contact or some other URI, and on that basis
> it could choose whether or not to insert a single tag. I thought that
> was the basis on which we agreed to go ahead with this updated to RFC
> 4424, rather than following the loose route draft. It is not clear to me
> that a proxy would have enough information to make a 3-way choice.
>
> John
>
>
> >
> > /Hans Erik van Elburg
> >
> >
> >
> > On Fri, Mar 13, 2009 at 3:39 PM, Elwell, John
> > <john.elwell@siemens.com> wrote:
> >
> >
> >        Hans Erik,
> >
> >       But how does a proxy know what to use? How does it know
> > whether the
> >       received Request-URI, which it uses to query the LS, is
> > a target or an
> >       AoR? Does it need to scan back up the H-I chain to see
> > whether there is
> >       already an 'istarget' present, and if so use 'aor' instead of
> >       'istarget'?
> >
> >       John
> >
> >
> >
> >       ________________________________
> >
> >              From: Hans Erik van Elburg
> > [mailto:ietf.hanserik@gmail.com]
> >
> >              Sent: 13 March 2009 14:33
> >              To: Elwell, John
> >              Cc: sip@ietf.org
> >
> >              Subject: Re: [Sip] Issue 3 - two tags (was RE:
> >       I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
> >
> >
> >              Hi John,
> >
> >              Well that is simple one would use different tags
> > for different
> >       type of entries, like in the example I already gave:
> >
> >                 H-I:  Bfreephone; "istarget",\
> >                      B,"aor"\
> >                      contact-B
> >
> >              Where the UAS interested in the initial target
> > with which it was
> >       addressed it would obtain the "istarget" tagged entry.
> >              Where the UAS interested in the AOR  that was
> > used to address it
> >       would obtain the "aor" tagged entry.
> >
> >              In simpler cases it could be that one entry
> > holds the "aor" as
> >       well as the "istarget" tag.
> >
> >              Of course the algorithm needs some work to make
> > it complete.
> >
> >              /Hans Erik van Elburg
> >
> >
> >
> >              On Fri, Mar 13, 2009 at 3:24 PM, Elwell, John
> >       <john.elwell@siemens.com> wrote:
> >
> >
> >                      Hans Erik,
> >
> >                      Following your detailed description, it
> > seems that the
> >       UAS serving a
> >                      freephone number could receive one of
> > the following
> >       (assuming H-I is
> >                      supported by all relevant nodes):
> >
> >                      1) H-I containing:
> >                      - the freephone number (tagged);
> >                      - the AoR (tagged);
> >                      - the contact URI (not tagged).
> >
> >                      Or
> >
> >                      2) H-I containing:
> >                      - the freephone number (tagged);
> >                      - the contact URI (not tagged).
> >
> >                      The latter would occur when the same
> > proxy translates
> >       the freephone
> >                      number (via the AoR) to the contact directly.
> >
> >                      So how does this help the UAS to figure out the
> >       freephone number? It
> >                      seems the UAS would have to search back
> > through the
> >       tagged entries
> >                      looking for one that matches a freephone
> > number it
> >       recognises? Does the
> >                      tagging really add much value?
> >
> >                      John
> >
> >
> >                      ________________________________
> >
> >                             From: sip-bounces@ietf.org
> >       [mailto:sip-bounces@ietf.org] On
> >                      Behalf Of Hans Erik van Elburg
> >                             Sent: 12 March 2009 21:26
> >                             To: Mary Barnes
> >                             Cc: Hans Erik van Elburg; sip@ietf.org
> >                             Subject: Re: [Sip] Issue 3 - two
> > tags (was RE:
> >
> > I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
> >
> >
> >
> >                             Inline...
> >
> >                             /Hans Erik van Elburg
> >
> >
> >
> >                             On Thu, Mar 12, 2009 at 7:10 PM,
> > Mary Barnes
> >                      <mary.barnes@nortel.com> wrote:
> >
> >
> >                                     I think there is another solution
> >       proposal that we
> >                      should pursue for issue 3 identified in the email
> >       summary sent by Shida
> >                      earlier this week - I've snipped all the
> > text except for
> >       that issue and
> >                      issue 2 in the summary below.  I do
> > think the discussion
> >       has confounded
> >                      the two issues and I think it ought to
> > be clarified that
> >       if we agree to
> >                      the two tags per issue 3, that the
> > options for the
> >       terminology MUST
> >                      change. And, perhaps this is why the
> > other discussion
> >       threads aren't
> >                      progressing well.
> >
> >
> >                             Agree.
> >
> >
> >
> >                                     So, Hans Erick, I would like
> >       clarification on as to your
> >                      view on the solution:
> >
> >                                     In general, ISTM that one
> > of the issues
> >       we're having in
> >                      this terminology thread is that you are
> > considering the
> >       solution to be
> >                      that the hi-entries are tagged as they
> >                                     are added to the request.
> >  And, just for
> >       clarification
> >                      from my perspective, that is NOT the
> > solution in either
> >       4244bis or
> >                      target-uri.
> >
> >
> >                             The principle should be that the
> > node that has
> >       the information
> >                      about what type of rewrite is performed
> > should add the
> >       tag. (This is
> >                      done by the current target-uri draft
> > allthough the
> >       solution is not
> >                      complete.)
> >
> >                             For the freephone service, that
> > would be the
> >       freephone service.
> >                      For the AOR that is the home proxy.
> >
> >                             Assume that the freephone service
> > receives an
> >       INVITE  request
> >                      that either
> >                             1. received request contains
> > History-Info header
> >       that contains
> >                      the freephone number (as in R-URI) as
> > last entry, THEN
> >       when rewriting
> >                      the Request-URI it has to tag that
> > existing received
> >       entry AND add the
> >                      new entry with the new value of the
> > Request-URI. (I
> >       think that was how
> >                      History-Info works right.)
> >                             2. received request contains
> > History-Info header
> >       that does not
> >                      contain the freephone number as last
> > entry, THEN when
> >       rewriting the
> >                      Request-URI it has add an entry with the
> > freephone
> >       number and tag that
> >                      entry AND add the new entry with the new
> > value of the
> >       Request-URI.
> >                             3. received request does not contain a
> >       History-Info header, THEN
> >                      when rewriting the Request-URI it has to
> > add an entry
> >       with the freephone
> >                      number and tag that entry AND add the
> > new entry with the
> >       new value of
> >                      the Request-URI.
> >
> >                             After forwarding this request it
> > arrives at the
> >       home proxy:
> >                             1. received request contains
> > History-Info header
> >       that contains
> >                      the AOR (as in R-URI) as last entry,
> > THEN when rewriting
> >       the Request-URI
> >                      it has to tag that existing received
> > entry AND add the
> >       new entry with
> >                      the new value of the Request-URI i.e.
> > the registered
> >       contact. (I think
> >                      that was how History-Info works right.)
> >                             2. received request contains
> > History-Info header
> >       that does not
> >                      contain the AOR (as in R-URI) as last
> > entry, THEN when
> >       rewriting the
> >                      Request-URI it has add an entry with the
> > AOR and tag
> >       that entry AND add
> >                      the new entry with the new value of the
> > Request-URI i.e.
> >       the registered
> >                      contact.
> >                             3. received request does not contain a
> >       History-Info header, THEN
> >                      when rewriting the Request-URI it has to
> > add an entry
> >       with the AOR as in
> >                      the Request-URI and tag that entry AND
> > add the new entry
> >       with the new
> >                      value of the Request-URI i.e. the
> > registered contact..
> >
> >
> >                                     I don't view it as a
> > problem necessarily
> >       if you'd like
> >                      to pursue an alternate solution, we just
> > need to be
> >       clear that's what
> >                      you're advocating. In that case, I would
> > agree the
> >       "retarget" is not an
> >                      appropriate tag for the entries, as you
> > don't know at
> >       the point in time
> >                      that you add an hi-entry that it will be
> > retargeted. In
> >       the case of this
> >                      approach to the solution, you do have to
> > alter the
> >       mechanism for
> >                      determining the hi-entry that your
> > services would want
> >       to use. You would
> >                      need to skip the most recent/last
> > hi-entry (as that
> >       would contain the
> >                      same as the Request-URI in the incoming
> > request), since
> >       it would be
> >                      tagged if you use this solution approach.
> >
> >
> >                             Yes, but only if the previous
> > proxy added it. As
> >       History-Info is
> >                      optional one never knows really.
> >
> >
> >
> >                                     Thus, you'd have to take the next
> >       hi-entry that you find
> >                      that has the desired "tag".  I will
> > posit that we could
> >       accomplish this
> >                      solution without any changes to 4244 by
> > just registering
> >       new SIP URI
> >                      parameter(s).   And,  that's not to say
> > we shouldn't do
> >       a 4244bis as
> >                      there are other issues to address in
> > that document, but
> >       that allows the
> >                      target-uri to describe a solution that
> > makes use of
> >       4244bis "as is".
> >
> >
> >
> >                             Are you proposing to decouple the
> > target-uri and
> >       the 4244bis
> >                      discussion?
> >
> >
> >                                     So, feedback on this
> > specific point (from
> >       Hans Erik and
> >                      others) would be appreciated.
> >
> >
> >
> >                                     Mary.
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
>

--0015174c3f902308e9046503053b
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

3-way choice?<br><br>A proxy that has retargeting logic knows when it is do=
ing so.<br>A proxy that has rerouting logic to registered contact knows whe=
n it is doing so.<br><br>No 3-way magic there.<br><br>And yes the History-I=
nfo is optional so you never know if all nodes in the chain do there bit. B=
ut that is fact of life. As long as you can ensure that in your domain it i=
s supported you will receive=A0 correct information.<br>
<br clear=3D"all">/Hans Erik van Elburg<br>
<br><br><div class=3D"gmail_quote">On Fri, Mar 13, 2009 at 5:31 PM, Elwell,=
 John <span dir=3D"ltr">&lt;<a href=3D"mailto:john.elwell@siemens.com">john=
.elwell@siemens.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quo=
te" style=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt=
 0.8ex; padding-left: 1ex;">
<div class=3D"im"><br>
<br>
&gt; -----Original Message-----<br>
&gt; From: Hans Erik van Elburg [mailto:<a href=3D"mailto:ietf.hanserik@gma=
il.com">ietf.hanserik@gmail.com</a>]<br>
</div><div class=3D"im">&gt; Sent: 13 March 2009 14:52<br>
&gt; To: Elwell, John<br>
&gt; Cc: <a href=3D"mailto:sip@ietf.org">sip@ietf.org</a><br>
&gt; Subject: Re: [Sip] Issue 3 - two tags (was RE:<br>
&gt; I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br>
&gt;<br>
</div><div class=3D"im">&gt; It knows when it rewrites the Request-URI due =
to a location<br>
&gt; lookup --&gt; tag &quot;aor&quot;.<br>
&gt; It knows (is configured to) when it rewrites the Request-URI<br>
&gt; with a value that leads the request to target with a<br>
&gt; different identity then the one that has been replaced --&gt;<br>
&gt; tag new entry &quot;istarget&quot;.<br>
&gt; It knows when it rewrites the Request-URI with a value and<br>
&gt; this leads to the first History-Info entry being recorded --&gt;<br>
&gt; tag &quot;istarget&quot;.<br>
</div>[JRE] It does not know that this is the first target, because upstrea=
m<br>
nodes may have retargeted without inserting H-I. In any case, the To<br>
header field URI is the first target, so we don&#39;t need that anyway.<br>
<br>
I can accept that a proxy might know, having done a LS look-up, whether<br>
the new URI is a registered contact or some other URI, and on that basis<br=
>
it could choose whether or not to insert a single tag. I thought that<br>
was the basis on which we agreed to go ahead with this updated to RFC<br>
4424, rather than following the loose route draft. It is not clear to me<br=
>
that a proxy would have enough information to make a 3-way choice.<br>
<font color=3D"#888888"><br>
John<br>
</font><div><div></div><div class=3D"h5"><br>
<br>
&gt;<br>
&gt; /Hans Erik van Elburg<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On Fri, Mar 13, 2009 at 3:39 PM, Elwell, John<br>
&gt; &lt;<a href=3D"mailto:john.elwell@siemens.com">john.elwell@siemens.com=
</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0Hans Erik,<br>
&gt;<br>
&gt; =A0 =A0 =A0 But how does a proxy know what to use? How does it know<br=
>
&gt; whether the<br>
&gt; =A0 =A0 =A0 received Request-URI, which it uses to query the LS, is<br=
>
&gt; a target or an<br>
&gt; =A0 =A0 =A0 AoR? Does it need to scan back up the H-I chain to see<br>
&gt; whether there is<br>
&gt; =A0 =A0 =A0 already an &#39;istarget&#39; present, and if so use &#39;=
aor&#39; instead of<br>
&gt; =A0 =A0 =A0 &#39;istarget&#39;?<br>
&gt;<br>
&gt; =A0 =A0 =A0 John<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 ________________________________<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0From: Hans Erik van Elburg<br>
&gt; [mailto:<a href=3D"mailto:ietf.hanserik@gmail.com">ietf.hanserik@gmail=
.com</a>]<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0Sent: 13 March 2009 14:33<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0To: Elwell, John<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0Cc: <a href=3D"mailto:sip@ietf.org">sip@iet=
f.org</a><br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0Subject: Re: [Sip] Issue 3 - two tags (was =
RE:<br>
&gt; =A0 =A0 =A0 I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<b=
r>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0Hi John,<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0Well that is simple one would use different=
 tags<br>
&gt; for different<br>
&gt; =A0 =A0 =A0 type of entries, like in the example I already gave:<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 H-I: =A0Bfreephone; &quot;istarget&quo=
t;,\<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0B,&quot;aor&quot;\<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0contact-B<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0Where the UAS interested in the initial tar=
get<br>
&gt; with which it was<br>
&gt; =A0 =A0 =A0 addressed it would obtain the &quot;istarget&quot; tagged =
entry.<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0Where the UAS interested in the AOR =A0that=
 was<br>
&gt; used to address it<br>
&gt; =A0 =A0 =A0 would obtain the &quot;aor&quot; tagged entry.<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0In simpler cases it could be that one entry=
<br>
&gt; holds the &quot;aor&quot; as<br>
&gt; =A0 =A0 =A0 well as the &quot;istarget&quot; tag.<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0Of course the algorithm needs some work to =
make<br>
&gt; it complete.<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0/Hans Erik van Elburg<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0On Fri, Mar 13, 2009 at 3:24 PM, Elwell, Jo=
hn<br>
&gt; =A0 =A0 =A0 &lt;<a href=3D"mailto:john.elwell@siemens.com">john.elwell=
@siemens.com</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Hans Erik,<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Following your detailed des=
cription, it<br>
&gt; seems that the<br>
&gt; =A0 =A0 =A0 UAS serving a<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0freephone number could rece=
ive one of<br>
&gt; the following<br>
&gt; =A0 =A0 =A0 (assuming H-I is<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0supported by all relevant n=
odes):<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A01) H-I containing:<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0- the freephone number (tag=
ged);<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0- the AoR (tagged);<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0- the contact URI (not tagg=
ed).<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Or<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A02) H-I containing:<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0- the freephone number (tag=
ged);<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0- the contact URI (not tagg=
ed).<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0The latter would occur when=
 the same<br>
&gt; proxy translates<br>
&gt; =A0 =A0 =A0 the freephone<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0number (via the AoR) to the=
 contact directly.<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0So how does this help the U=
AS to figure out the<br>
&gt; =A0 =A0 =A0 freephone number? It<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0seems the UAS would have to=
 search back<br>
&gt; through the<br>
&gt; =A0 =A0 =A0 tagged entries<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0looking for one that matche=
s a freephone<br>
&gt; number it<br>
&gt; =A0 =A0 =A0 recognises? Does the<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0tagging really add much val=
ue?<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0John<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0___________________________=
_____<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 From: <a href=
=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a><br>
&gt; =A0 =A0 =A0 [mailto:<a href=3D"mailto:sip-bounces@ietf.org">sip-bounce=
s@ietf.org</a>] On<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Behalf Of Hans Erik van Elb=
urg<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Sent: 12 March=
 2009 21:26<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 To: Mary Barne=
s<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Cc: Hans Erik =
van Elburg; <a href=3D"mailto:sip@ietf.org">sip@ietf.org</a><br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Subject: Re: [=
Sip] Issue 3 - two<br>
&gt; tags (was RE:<br>
&gt;<br>
&gt; I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Inline...<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 /Hans Erik van=
 Elburg<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 On Thu, Mar 12=
, 2009 at 7:10 PM,<br>
&gt; Mary Barnes<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&lt;<a href=3D"mailto:mary.=
barnes@nortel.com">mary.barnes@nortel.com</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 I think there is another solution<br>
&gt; =A0 =A0 =A0 proposal that we<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0should pursue for issue 3 i=
dentified in the email<br>
&gt; =A0 =A0 =A0 summary sent by Shida<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0earlier this week - I&#39;v=
e snipped all the<br>
&gt; text except for<br>
&gt; =A0 =A0 =A0 that issue and<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0issue 2 in the summary belo=
w. =A0I do<br>
&gt; think the discussion<br>
&gt; =A0 =A0 =A0 has confounded<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the two issues and I think =
it ought to<br>
&gt; be clarified that<br>
&gt; =A0 =A0 =A0 if we agree to<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the two tags per issue 3, t=
hat the<br>
&gt; options for the<br>
&gt; =A0 =A0 =A0 terminology MUST<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0change. And, perhaps this i=
s why the<br>
&gt; other discussion<br>
&gt; =A0 =A0 =A0 threads aren&#39;t<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0progressing well.<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Agree.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 So, Hans Erick, I would like<br>
&gt; =A0 =A0 =A0 clarification on as to your<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0view on the solution:<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 In general, ISTM that one<br>
&gt; of the issues<br>
&gt; =A0 =A0 =A0 we&#39;re having in<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0this terminology thread is =
that you are<br>
&gt; considering the<br>
&gt; =A0 =A0 =A0 solution to be<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0that the hi-entries are tag=
ged as they<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 are added to the request.<br>
&gt; =A0And, just for<br>
&gt; =A0 =A0 =A0 clarification<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0from my perspective, that i=
s NOT the<br>
&gt; solution in either<br>
&gt; =A0 =A0 =A0 4244bis or<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0target-uri.<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 The principle =
should be that the<br>
&gt; node that has<br>
&gt; =A0 =A0 =A0 the information<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0about what type of rewrite =
is performed<br>
&gt; should add the<br>
&gt; =A0 =A0 =A0 tag. (This is<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0done by the current target-=
uri draft<br>
&gt; allthough the<br>
&gt; =A0 =A0 =A0 solution is not<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0complete.)<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 For the freeph=
one service, that<br>
&gt; would be the<br>
&gt; =A0 =A0 =A0 freephone service.<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0For the AOR that is the hom=
e proxy.<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Assume that th=
e freephone service<br>
&gt; receives an<br>
&gt; =A0 =A0 =A0 INVITE =A0request<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0that either<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 1. received re=
quest contains<br>
&gt; History-Info header<br>
&gt; =A0 =A0 =A0 that contains<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the freephone number (as in=
 R-URI) as<br>
&gt; last entry, THEN<br>
&gt; =A0 =A0 =A0 when rewriting<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the Request-URI it has to t=
ag that<br>
&gt; existing received<br>
&gt; =A0 =A0 =A0 entry AND add the<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0new entry with the new valu=
e of the<br>
&gt; Request-URI. (I<br>
&gt; =A0 =A0 =A0 think that was how<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0History-Info works right.)<=
br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 2. received re=
quest contains<br>
&gt; History-Info header<br>
&gt; =A0 =A0 =A0 that does not<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0contain the freephone numbe=
r as last<br>
&gt; entry, THEN when<br>
&gt; =A0 =A0 =A0 rewriting the<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Request-URI it has add an e=
ntry with the<br>
&gt; freephone<br>
&gt; =A0 =A0 =A0 number and tag that<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0entry AND add the new entry=
 with the new<br>
&gt; value of the<br>
&gt; =A0 =A0 =A0 Request-URI.<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 3. received re=
quest does not contain a<br>
&gt; =A0 =A0 =A0 History-Info header, THEN<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0when rewriting the Request-=
URI it has to<br>
&gt; add an entry<br>
&gt; =A0 =A0 =A0 with the freephone<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0number and tag that entry A=
ND add the<br>
&gt; new entry with the<br>
&gt; =A0 =A0 =A0 new value of<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the Request-URI.<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 After forwardi=
ng this request it<br>
&gt; arrives at the<br>
&gt; =A0 =A0 =A0 home proxy:<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 1. received re=
quest contains<br>
&gt; History-Info header<br>
&gt; =A0 =A0 =A0 that contains<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the AOR (as in R-URI) as la=
st entry,<br>
&gt; THEN when rewriting<br>
&gt; =A0 =A0 =A0 the Request-URI<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0it has to tag that existing=
 received<br>
&gt; entry AND add the<br>
&gt; =A0 =A0 =A0 new entry with<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the new value of the Reques=
t-URI i.e.<br>
&gt; the registered<br>
&gt; =A0 =A0 =A0 contact. (I think<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0that was how History-Info w=
orks right.)<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 2. received re=
quest contains<br>
&gt; History-Info header<br>
&gt; =A0 =A0 =A0 that does not<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0contain the AOR (as in R-UR=
I) as last<br>
&gt; entry, THEN when<br>
&gt; =A0 =A0 =A0 rewriting the<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Request-URI it has add an e=
ntry with the<br>
&gt; AOR and tag<br>
&gt; =A0 =A0 =A0 that entry AND add<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the new entry with the new =
value of the<br>
&gt; Request-URI i.e.<br>
&gt; =A0 =A0 =A0 the registered<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0contact.<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 3. received re=
quest does not contain a<br>
&gt; =A0 =A0 =A0 History-Info header, THEN<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0when rewriting the Request-=
URI it has to<br>
&gt; add an entry<br>
&gt; =A0 =A0 =A0 with the AOR as in<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the Request-URI and tag tha=
t entry AND<br>
&gt; add the new entry<br>
&gt; =A0 =A0 =A0 with the new<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0value of the Request-URI i.=
e. the<br>
&gt; registered contact..<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 I don&#39;t view it as a<br>
&gt; problem necessarily<br>
&gt; =A0 =A0 =A0 if you&#39;d like<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0to pursue an alternate solu=
tion, we just<br>
&gt; need to be<br>
&gt; =A0 =A0 =A0 clear that&#39;s what<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0you&#39;re advocating. In t=
hat case, I would<br>
&gt; agree the<br>
&gt; =A0 =A0 =A0 &quot;retarget&quot; is not an<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0appropriate tag for the ent=
ries, as you<br>
&gt; don&#39;t know at<br>
&gt; =A0 =A0 =A0 the point in time<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0that you add an hi-entry th=
at it will be<br>
&gt; retargeted. In<br>
&gt; =A0 =A0 =A0 the case of this<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0approach to the solution, y=
ou do have to<br>
&gt; alter the<br>
&gt; =A0 =A0 =A0 mechanism for<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0determining the hi-entry th=
at your<br>
&gt; services would want<br>
&gt; =A0 =A0 =A0 to use. You would<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0need to skip the most recen=
t/last<br>
&gt; hi-entry (as that<br>
&gt; =A0 =A0 =A0 would contain the<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0same as the Request-URI in =
the incoming<br>
&gt; request), since<br>
&gt; =A0 =A0 =A0 it would be<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0tagged if you use this solu=
tion approach.<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Yes, but only =
if the previous<br>
&gt; proxy added it. As<br>
&gt; =A0 =A0 =A0 History-Info is<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0optional one never knows re=
ally.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 Thus, you&#39;d have to take the next<br>
&gt; =A0 =A0 =A0 hi-entry that you find<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0that has the desired &quot;=
tag&quot;. =A0I will<br>
&gt; posit that we could<br>
&gt; =A0 =A0 =A0 accomplish this<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0solution without any change=
s to 4244 by<br>
&gt; just registering<br>
&gt; =A0 =A0 =A0 new SIP URI<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0parameter(s). =A0 And, =A0t=
hat&#39;s not to say<br>
&gt; we shouldn&#39;t do<br>
&gt; =A0 =A0 =A0 a 4244bis as<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0there are other issues to a=
ddress in<br>
&gt; that document, but<br>
&gt; =A0 =A0 =A0 that allows the<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0target-uri to describe a so=
lution that<br>
&gt; makes use of<br>
&gt; =A0 =A0 =A0 4244bis &quot;as is&quot;.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Are you propos=
ing to decouple the<br>
&gt; target-uri and<br>
&gt; =A0 =A0 =A0 the 4244bis<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0discussion?<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 So, feedback on this<br>
&gt; specific point (from<br>
&gt; =A0 =A0 =A0 Hans Erik and<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0others) would be appreciate=
d.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 Mary.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
</div></div></blockquote></div><br>

--0015174c3f902308e9046503053b--

From sumanth@cablelabs.com  Fri Mar 13 10:07:36 2009
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From: "Sumanth Channabasappa" <sumanth@cablelabs.com>
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Nils,


Ok, but the point is that in most of the cases it will be too late
already.
Maybe an example is helpfull:
I try to register via proxy.hotel.com to my registrar at example.com.
Note: I'm using the open WLAN of hotel of cours. Now proxy.hotel.com
challenges my initial REGISTER request, which my UA replies to with the
password which I got at the receiption. Now this second REGISTER request
is forwarded to example.com and replied with a 200 OK.
When my UA receives the 200 OK it checks the Proxy-Authentication-Info
header and detects an error (maybe because a man-in-the-midlle modified
something). Upppss!
What should the UA do here? Display me a warning "It looks like you
successfully registered to example.com. Unfortunately something went
wrong. But your credentials where send already to example.com... Maybe
now
someone else is registered with your account to example.com?!"

[S] In the registration example you provide, the SIP UA will need to
verify the Authentication-Info header (per RFC 3261), which is specific
to the UA-to-registrar interactions.

However, the example can be easily generalized for UA-to-Proxy
interactions, where the Proxy-Authentication-Info header is used. When
the response (say 200 OK) cannot be authenticated, then the UA probably
cannot ascertain as to what happened; for example, the proxy could have
been an imposter who generated a phony response (i.e., it never reached
the intended proxy) or there was a M-i-M who modified a valid response
(i.e., it did reach the intended proxy). What the UA can ascertain is
that something is wrong! This is similar to the case where it does not
receive a response, i.e., the UA does not know if the other end did
receive it and the response was lost (in your registration example, the
device will be registered without the UA knowing about it), or the other
end did not ever receive it (and registration request never made it
through).=20

By rejecting an unauthenticated response, the UA can potentially
minimize some of the threats (e.g., a hijack of the UA by a phony
proxy). Further, actions such as a retry may help. For example, if the
proxy sees a retry after it sent a response, then it can ascertain that
something went wrong (e.g., the packet was dropped) and clear up
associated resources. [Obvious caution: Too many retries can lead to
dictionary attacks.]

[Of course, if the MiM did not manipulate the parts of the response that
are authenticated by the Proxy-Authentication-Info header, then the
threat never surfaces. This is a drawback/limitation.]


I understand that your draft is only adding the
Proxy-Authentication-Info
header.
On the other hand I doubt that the whole mutual digest auth like it is
described in 2617 and 3261 is usuable for SIP at all.
[S] Well, digest is being used by quite a few implementations :)! And in
the presence of multiple proxies, digest is an option to authenticate
(or authorize) end-to-end. Thus, even when TLS is established with the
next-hop, it can be a useful tool. Is it a replacement for SIPS? Perhaps
not! But what are the other options?=20


> In any case, please refer to the following small thread (3 emails) for
> the feedback I received during offline discussions at the last IETF:
>
> http://www.ietf.org/mail-archive/web/sip/current/msg25882.html

>From that small thread I can only ask the same question as Victor did
already: what is meant by this trusted connection to the next hop?
[S] The feedback I received (as I understood) was that once you
establish mutually-authenticated-TLS with the next-hop, you trust any
communication over that link (even to registrars and proxies outside of
the next-hop). This also meant that the Authentication-Info header (for
registration) is redundant (in RFC3261).


And another issue in your draft in my opinion is what you describe in
second paragraph of section 10:
If someone is going to use mutual digest authentication it can not be
optional! As recent "demonstrations" on HTTPS and SMTP with TLS have
shown
security "upgrades" of an IP connection are not optional. If you know
that
your server (the hotel proxy in the example above) supports mutual auth,
you should configure your UA to require mutual auth. If the mutual auth
is
not there or failed, something is really wrong in the network and you
should definately not proceed with whatever you wanted to do.
[S] The reason it is optional is primarily due to WG feedback that it
may not be required in all cases. For example, if you use SIPS or your
access network is always integrity-protected (e.g., Cable access with
DOCSIS BPI+). As Section 10 tries to say, use of this header is not a
replacement for TLS. It is only useful in architectures where digest is
used (with or without TLS) and you prefer mutual authentication to
minimize some threats. I say minimize since some threats still exist
(e.g., a transparent MiM who is an active or passive participant).

- S


From dworley@nortel.com  Fri Mar 13 10:52:53 2009
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From: "Dale Worley" <dworley@nortel.com>
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Subject: [Sip] draft-ietf-sip-199-06
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I don't know if this has been mentioned, but I skimmed
draft-ietf-sip-199-06 and couldn't find it:

One thing that I've always felt was a weakness of the "199" proposal was
that it was not possible to associate the early dialog of the 199 with
the particular media streams of that early dialog, which makes it
difficult for a UAC to use a 199 response to free media-processing
resources.

However, if the UAS that creates an early media stream sends a 1xx
(e.g., 183) *reliably*, then the UAC is certain to be able to associate
the to-tag of the 1xx (which is the same as the to-tag of the 199) with
the early media source address/port.

Why isn't this fact put in the draft?  I suppose it isn't a
*requirement* for 199, but it seems like a very strong suggestion for
UAS behavior to enable the UAC to take advantage of 199.

Dale



From mary.barnes@nortel.com  Fri Mar 13 11:41:06 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Elwell, John" <john.elwell@siemens.com>, "Hans Erik van Elburg" <ietf.hanserik@gmail.com>
Cc: sip@ietf.org
Subject: Re: [Sip] Issue 3 - two tags (was RE: I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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=20
Hi John,

Thanks for all your feedback thus far on this topic. I have a couple
comments below [MB].

Mary.

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Elwell, John
Sent: Friday, March 13, 2009 11:31 AM
To: Hans Erik van Elburg
Cc: sip@ietf.org
Subject: Re: [Sip] Issue 3 - two tags (was RE:
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


=20

> -----Original Message-----
> From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
> Sent: 13 March 2009 14:52
> To: Elwell, John
> Cc: sip@ietf.org
> Subject: Re: [Sip] Issue 3 - two tags (was RE:=20
> I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>=20
> It knows when it rewrites the Request-URI due to a location lookup -->

> tag "aor".
> It knows (is configured to) when it rewrites the Request-URI with a=20
> value that leads the request to target with a different identity then=20
> the one that has been replaced --> tag new entry "istarget".
> It knows when it rewrites the Request-URI with a value and this leads=20
> to the first History-Info entry being recorded --> tag "istarget".
[JRE] It does not know that this is the first target, because upstream
nodes may have retargeted without inserting H-I. In any case, the To
header field URI is the first target, so we don't need that anyway.
[MB] John is correct. There is no rule about tagging the first
History-Info entry. It may be tagged if the next node changes the
request-URI via a service lookup or some other manner such as
configuration.[/MB]

I can accept that a proxy might know, having done a LS look-up, whether
the new URI is a registered contact or some other URI, and on that basis
it could choose whether or not to insert a single tag. I thought that
was the basis on which we agreed to go ahead with this updated to RFC
4424, rather than following the loose route draft. It is not clear to me
that a proxy would have enough information to make a 3-way choice.
[MB] My understanding is that there are just two ways in which the
request-URI changes if you are using loose routing. It's either changed
due to a lookup in a location server, whose information was populated by
the UA during registration, OR it is changed due to configuration data
(either system, user, etc.). In both cases, obviously, the proxy that is
doing this is responsible for the domain of the request-URI in the
incoming request. If it were not the case, then all the proxy can do is
forward the outgoing request to the next hop. So, I think that may be
where the concept of a 3rd choice comes in. But, as far as tagging, we
(as I understand it) at least were discussing the potential of two tags
only - i.e., further differentiating the cases that are currently all
captured by the "retarget" as defined in 4244bis, which is the same
mechanism that is specified in the target-uri document EXCEPT that
document was only capturing the service lookups, which we believe is
incorrect. [/MB]

John


>=20
> /Hans Erik van Elburg
>=20
>=20
>=20
> On Fri, Mar 13, 2009 at 3:39 PM, Elwell, John=20
> <john.elwell@siemens.com> wrote:
>=20
>=20
> 	 Hans Erik,
> =09
> 	But how does a proxy know what to use? How does it know whether
the
> 	received Request-URI, which it uses to query the LS, is a target
or=20
> an
> 	AoR? Does it need to scan back up the H-I chain to see whether
there=20
> is
> 	already an 'istarget' present, and if so use 'aor' instead of
> 	'istarget'?
> =09
> 	John
> =09
>=20
>=20
> 	________________________________
> =09
> 	       From: Hans Erik van Elburg
> [mailto:ietf.hanserik@gmail.com]
> =09
> 	       Sent: 13 March 2009 14:33
> 	       To: Elwell, John
> 	       Cc: sip@ietf.org
> =09
> 	       Subject: Re: [Sip] Issue 3 - two tags (was RE:
> 	I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
> =09
> =09
> 	       Hi John,
> =09
> 	       Well that is simple one would use different tags for
different
> 	type of entries, like in the example I already gave:
> =09
> 	          H-I:  Bfreephone; "istarget",\
> 	               B,"aor"\
> 	               contact-B
> =09
> 	       Where the UAS interested in the initial target with which
it=20
> was
> 	addressed it would obtain the "istarget" tagged entry.
> 	       Where the UAS interested in the AOR  that was used to
address=20
> it
> 	would obtain the "aor" tagged entry.
> =09
> 	       In simpler cases it could be that one entry holds the
"aor" as
> 	well as the "istarget" tag.
> =09
> 	       Of course the algorithm needs some work to make it
complete.
> =09
> 	       /Hans Erik van Elburg
> =09
> =09
> =09
> 	       On Fri, Mar 13, 2009 at 3:24 PM, Elwell, John
> 	<john.elwell@siemens.com> wrote:
> =09
> =09
> 	               Hans Erik,
> =09
> 	               Following your detailed description, it seems
that the
> 	UAS serving a
> 	               freephone number could receive one of the
following
> 	(assuming H-I is
> 	               supported by all relevant nodes):
> =09
> 	               1) H-I containing:
> 	               - the freephone number (tagged);
> 	               - the AoR (tagged);
> 	               - the contact URI (not tagged).
> =09
> 	               Or
> =09
> 	               2) H-I containing:
> 	               - the freephone number (tagged);
> 	               - the contact URI (not tagged).
> =09
> 	               The latter would occur when the same proxy
translates
> 	the freephone
> 	               number (via the AoR) to the contact directly.
> =09
> 	               So how does this help the UAS to figure out the
> 	freephone number? It
> 	               seems the UAS would have to search back through
the
> 	tagged entries
> 	               looking for one that matches a freephone number
it
> 	recognises? Does the
> 	               tagging really add much value?
> =09
> 	               John
> =09
> =09
> 	               ________________________________
> =09
> 	                      From: sip-bounces@ietf.org
> 	[mailto:sip-bounces@ietf.org] On
> 	               Behalf Of Hans Erik van Elburg
> 	                      Sent: 12 March 2009 21:26
> 	                      To: Mary Barnes
> 	                      Cc: Hans Erik van Elburg; sip@ietf.org
> 	                      Subject: Re: [Sip] Issue 3 - two tags (was
RE:
> 	              =20
> I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
> =09
> =09
> =09
> 	                      Inline...
> =09
> 	                      /Hans Erik van Elburg
> =09
> =09
> =09
> 	                      On Thu, Mar 12, 2009 at 7:10 PM, Mary
Barnes
> 	               <mary.barnes@nortel.com> wrote:
> =09
> =09
> 	                              I think there is another solution
> 	proposal that we
> 	               should pursue for issue 3 identified in the email
> 	summary sent by Shida
> 	               earlier this week - I've snipped all the text
except=20
> for
> 	that issue and
> 	               issue 2 in the summary below.  I do think the=20
> discussion
> 	has confounded
> 	               the two issues and I think it ought to be
clarified=20
> that
> 	if we agree to
> 	               the two tags per issue 3, that the options for
the
> 	terminology MUST
> 	               change. And, perhaps this is why the other
discussion
> 	threads aren't
> 	               progressing well.
> =09
> =09
> 	                      Agree.
> =09
> =09
> =09
> 	                              So, Hans Erick, I would like
> 	clarification on as to your
> 	               view on the solution:
> =09
> 	                              In general, ISTM that one of the
issues
> 	we're having in
> 	               this terminology thread is that you are
considering=20
> the
> 	solution to be
> 	               that the hi-entries are tagged as they
> 	                              are added to the request.=20
>  And, just for
> 	clarification
> 	               from my perspective, that is NOT the solution in=20
> either
> 	4244bis or
> 	               target-uri.
> =09
> =09
> 	                      The principle should be that the node that
has
> 	the information
> 	               about what type of rewrite is performed should
add the
> 	tag. (This is
> 	               done by the current target-uri draft allthough
the
> 	solution is not
> 	               complete.)
> =09
> 	                      For the freephone service, that would be
the
> 	freephone service.
> 	               For the AOR that is the home proxy.
> =09
> 	                      Assume that the freephone service receives
an
> 	INVITE  request
> 	               that either
> 	                      1. received request contains History-Info=20
> header
> 	that contains
> 	               the freephone number (as in R-URI) as last entry,
THEN
> 	when rewriting
> 	               the Request-URI it has to tag that existing
received
> 	entry AND add the
> 	               new entry with the new value of the Request-URI.
(I
> 	think that was how
> 	               History-Info works right.)
> 	                      2. received request contains History-Info=20
> header
> 	that does not
> 	               contain the freephone number as last entry, THEN
when
> 	rewriting the
> 	               Request-URI it has add an entry with the
freephone
> 	number and tag that
> 	               entry AND add the new entry with the new value of
the
> 	Request-URI.
> 	                      3. received request does not contain a
> 	History-Info header, THEN
> 	               when rewriting the Request-URI it has to add an
entry
> 	with the freephone
> 	               number and tag that entry AND add the new entry
with=20
> the
> 	new value of
> 	               the Request-URI.
> =09
> 	                      After forwarding this request it arrives
at the
> 	home proxy:
> 	                      1. received request contains History-Info=20
> header
> 	that contains
> 	               the AOR (as in R-URI) as last entry, THEN when=20
> rewriting
> 	the Request-URI
> 	               it has to tag that existing received entry AND
add the
> 	new entry with
> 	               the new value of the Request-URI i.e.=20
> the registered
> 	contact. (I think
> 	               that was how History-Info works right.)
> 	                      2. received request contains History-Info=20
> header
> 	that does not
> 	               contain the AOR (as in R-URI) as last entry, THEN
when
> 	rewriting the
> 	               Request-URI it has add an entry with the AOR and
tag
> 	that entry AND add
> 	               the new entry with the new value of the
Request-URI=20
> i.e.
> 	the registered
> 	               contact.
> 	                      3. received request does not contain a
> 	History-Info header, THEN
> 	               when rewriting the Request-URI it has to add an
entry
> 	with the AOR as in
> 	               the Request-URI and tag that entry AND add the
new=20
> entry
> 	with the new
> 	               value of the Request-URI i.e. the registered
contact..
> =09
> =09
> 	                              I don't view it as a problem=20
> necessarily
> 	if you'd like
> 	               to pursue an alternate solution, we just need to
be
> 	clear that's what
> 	               you're advocating. In that case, I would agree
the
> 	"retarget" is not an
> 	               appropriate tag for the entries, as you don't
know at
> 	the point in time
> 	               that you add an hi-entry that it will be
retargeted.=20
> In
> 	the case of this
> 	               approach to the solution, you do have to alter
the
> 	mechanism for
> 	               determining the hi-entry that your services would
want
> 	to use. You would
> 	               need to skip the most recent/last hi-entry (as
that
> 	would contain the
> 	               same as the Request-URI in the incoming request),

> since
> 	it would be
> 	               tagged if you use this solution approach.
> =09
> =09
> 	                      Yes, but only if the previous proxy added
it.=20
> As
> 	History-Info is
> 	               optional one never knows really.
> =09
> =09
> =09
> 	                              Thus, you'd have to take the next
> 	hi-entry that you find
> 	               that has the desired "tag".  I will posit that we

> could
> 	accomplish this
> 	               solution without any changes to 4244 by just=20
> registering
> 	new SIP URI
> 	               parameter(s).   And,  that's not to say=20
> we shouldn't do
> 	a 4244bis as
> 	               there are other issues to address in that
document,=20
> but
> 	that allows the
> 	               target-uri to describe a solution that makes use
of
> 	4244bis "as is".
> =09
> =09
> =09
> 	                      Are you proposing to decouple the
target-uri=20
> and
> 	the 4244bis
> 	               discussion?
> =09
> =09
> 	                              So, feedback on this specific
point=20
> (from
> 	Hans Erik and
> 	               others) would be appreciated.
> =09
> =09
> =09
> 	                              Mary.
> =09
> =09
> =09
> =09
> =09
> =09
> =09
> =09
>=20
>=20
>=20
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Date: Fri, 13 Mar 2009 22:17:18 +0100
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From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: Mary Barnes <mary.barnes@nortel.com>
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Cc: sip@ietf.org, "Elwell, John" <john.elwell@siemens.com>
Subject: Re: [Sip] Issue 3 - two tags (was RE: I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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I guess you mean that you agree with John that such rule can not be reliably
construed.
I dont think I have seen convincing proof that such is the case.

Are you also saying that we may only consider loose routing being used?

Further for "configured mappings" for which we never saw a proper
definition. But assuming that this is a proxy that can be configured to do a
Request-URI  rewrite based on some configured setting.

In that case there are two variants of such mapping:
1. A mapping that changes the identity of the target to be reached. (This is
what call forwarding or diversion does normally.)
2. A mapping that only changes  the next hop route taken towards the target
identity that has not been changed.

For the UAS it only matters that the home domain does the tagging
consistently and correctly if it wants to rely on such information.

Otherwise we would have to start adding information about the domain that
added a specific hi-entry, if one wants to be sure who added the particular
hi-entry.

/Hans Erik van Elburg


On Fri, Mar 13, 2009 at 7:43 PM, Mary Barnes <mary.barnes@nortel.com> wrote:

> Hi John,
>
> Thanks for all your feedback thus far on this topic. I have a couple
> comments below [MB].
>
> Mary.
>
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Elwell, John
> Sent: Friday, March 13, 2009 11:31 AM
> To: Hans Erik van Elburg
> Cc: sip@ietf.org
> Subject: Re: [Sip] Issue 3 - two tags (was RE:
> I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>
>
>
>
> > -----Original Message-----
> > From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
> > Sent: 13 March 2009 14:52
> > To: Elwell, John
> > Cc: sip@ietf.org
> > Subject: Re: [Sip] Issue 3 - two tags (was RE:
> > I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
> >
> > It knows when it rewrites the Request-URI due to a location lookup -->
>
> > tag "aor".
> > It knows (is configured to) when it rewrites the Request-URI with a
> > value that leads the request to target with a different identity then
> > the one that has been replaced --> tag new entry "istarget".
> > It knows when it rewrites the Request-URI with a value and this leads
> > to the first History-Info entry being recorded --> tag "istarget".
> [JRE] It does not know that this is the first target, because upstream
> nodes may have retargeted without inserting H-I. In any case, the To
> header field URI is the first target, so we don't need that anyway.
> [MB] John is correct. There is no rule about tagging the first
> History-Info entry. It may be tagged if the next node changes the
> request-URI via a service lookup or some other manner such as
> configuration.[/MB]
>
> I can accept that a proxy might know, having done a LS look-up, whether
> the new URI is a registered contact or some other URI, and on that basis
> it could choose whether or not to insert a single tag. I thought that
> was the basis on which we agreed to go ahead with this updated to RFC
> 4424, rather than following the loose route draft. It is not clear to me
> that a proxy would have enough information to make a 3-way choice.
> [MB] My understanding is that there are just two ways in which the
> request-URI changes if you are using loose routing. It's either changed
> due to a lookup in a location server, whose information was populated by
> the UA during registration, OR it is changed due to configuration data
> (either system, user, etc.). In both cases, obviously, the proxy that is
> doing this is responsible for the domain of the request-URI in the
> incoming request. If it were not the case, then all the proxy can do is
> forward the outgoing request to the next hop. So, I think that may be
> where the concept of a 3rd choice comes in. But, as far as tagging, we
> (as I understand it) at least were discussing the potential of two tags
> only - i.e., further differentiating the cases that are currently all
> captured by the "retarget" as defined in 4244bis, which is the same
> mechanism that is specified in the target-uri document EXCEPT that
> document was only capturing the service lookups, which we believe is
> incorrect. [/MB]
>
> John
>
>
> >
> > /Hans Erik van Elburg
> >
> >
> >
> > On Fri, Mar 13, 2009 at 3:39 PM, Elwell, John
> > <john.elwell@siemens.com> wrote:
> >
> >
> >        Hans Erik,
> >
> >       But how does a proxy know what to use? How does it know whether
> the
> >       received Request-URI, which it uses to query the LS, is a target
> or
> > an
> >       AoR? Does it need to scan back up the H-I chain to see whether
> there
> > is
> >       already an 'istarget' present, and if so use 'aor' instead of
> >       'istarget'?
> >
> >       John
> >
> >
> >
> >       ________________________________
> >
> >              From: Hans Erik van Elburg
> > [mailto:ietf.hanserik@gmail.com]
> >
> >              Sent: 13 March 2009 14:33
> >              To: Elwell, John
> >              Cc: sip@ietf.org
> >
> >              Subject: Re: [Sip] Issue 3 - two tags (was RE:
> >       I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
> >
> >
> >              Hi John,
> >
> >              Well that is simple one would use different tags for
> different
> >       type of entries, like in the example I already gave:
> >
> >                 H-I:  Bfreephone; "istarget",\
> >                      B,"aor"\
> >                      contact-B
> >
> >              Where the UAS interested in the initial target with which
> it
> > was
> >       addressed it would obtain the "istarget" tagged entry.
> >              Where the UAS interested in the AOR  that was used to
> address
> > it
> >       would obtain the "aor" tagged entry.
> >
> >              In simpler cases it could be that one entry holds the
> "aor" as
> >       well as the "istarget" tag.
> >
> >              Of course the algorithm needs some work to make it
> complete.
> >
> >              /Hans Erik van Elburg
> >
> >
> >
> >              On Fri, Mar 13, 2009 at 3:24 PM, Elwell, John
> >       <john.elwell@siemens.com> wrote:
> >
> >
> >                      Hans Erik,
> >
> >                      Following your detailed description, it seems
> that the
> >       UAS serving a
> >                      freephone number could receive one of the
> following
> >       (assuming H-I is
> >                      supported by all relevant nodes):
> >
> >                      1) H-I containing:
> >                      - the freephone number (tagged);
> >                      - the AoR (tagged);
> >                      - the contact URI (not tagged).
> >
> >                      Or
> >
> >                      2) H-I containing:
> >                      - the freephone number (tagged);
> >                      - the contact URI (not tagged).
> >
> >                      The latter would occur when the same proxy
> translates
> >       the freephone
> >                      number (via the AoR) to the contact directly.
> >
> >                      So how does this help the UAS to figure out the
> >       freephone number? It
> >                      seems the UAS would have to search back through
> the
> >       tagged entries
> >                      looking for one that matches a freephone number
> it
> >       recognises? Does the
> >                      tagging really add much value?
> >
> >                      John
> >
> >
> >                      ________________________________
> >
> >                             From: sip-bounces@ietf.org
> >       [mailto:sip-bounces@ietf.org] On
> >                      Behalf Of Hans Erik van Elburg
> >                             Sent: 12 March 2009 21:26
> >                             To: Mary Barnes
> >                             Cc: Hans Erik van Elburg; sip@ietf.org
> >                             Subject: Re: [Sip] Issue 3 - two tags (was
> RE:
> >
> > I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
> >
> >
> >
> >                             Inline...
> >
> >                             /Hans Erik van Elburg
> >
> >
> >
> >                             On Thu, Mar 12, 2009 at 7:10 PM, Mary
> Barnes
> >                      <mary.barnes@nortel.com> wrote:
> >
> >
> >                                     I think there is another solution
> >       proposal that we
> >                      should pursue for issue 3 identified in the email
> >       summary sent by Shida
> >                      earlier this week - I've snipped all the text
> except
> > for
> >       that issue and
> >                      issue 2 in the summary below.  I do think the
> > discussion
> >       has confounded
> >                      the two issues and I think it ought to be
> clarified
> > that
> >       if we agree to
> >                      the two tags per issue 3, that the options for
> the
> >       terminology MUST
> >                      change. And, perhaps this is why the other
> discussion
> >       threads aren't
> >                      progressing well.
> >
> >
> >                             Agree.
> >
> >
> >
> >                                     So, Hans Erick, I would like
> >       clarification on as to your
> >                      view on the solution:
> >
> >                                     In general, ISTM that one of the
> issues
> >       we're having in
> >                      this terminology thread is that you are
> considering
> > the
> >       solution to be
> >                      that the hi-entries are tagged as they
> >                                     are added to the request.
> >  And, just for
> >       clarification
> >                      from my perspective, that is NOT the solution in
> > either
> >       4244bis or
> >                      target-uri.
> >
> >
> >                             The principle should be that the node that
> has
> >       the information
> >                      about what type of rewrite is performed should
> add the
> >       tag. (This is
> >                      done by the current target-uri draft allthough
> the
> >       solution is not
> >                      complete.)
> >
> >                             For the freephone service, that would be
> the
> >       freephone service.
> >                      For the AOR that is the home proxy.
> >
> >                             Assume that the freephone service receives
> an
> >       INVITE  request
> >                      that either
> >                             1. received request contains History-Info
> > header
> >       that contains
> >                      the freephone number (as in R-URI) as last entry,
> THEN
> >       when rewriting
> >                      the Request-URI it has to tag that existing
> received
> >       entry AND add the
> >                      new entry with the new value of the Request-URI.
> (I
> >       think that was how
> >                      History-Info works right.)
> >                             2. received request contains History-Info
> > header
> >       that does not
> >                      contain the freephone number as last entry, THEN
> when
> >       rewriting the
> >                      Request-URI it has add an entry with the
> freephone
> >       number and tag that
> >                      entry AND add the new entry with the new value of
> the
> >       Request-URI.
> >                             3. received request does not contain a
> >       History-Info header, THEN
> >                      when rewriting the Request-URI it has to add an
> entry
> >       with the freephone
> >                      number and tag that entry AND add the new entry
> with
> > the
> >       new value of
> >                      the Request-URI.
> >
> >                             After forwarding this request it arrives
> at the
> >       home proxy:
> >                             1. received request contains History-Info
> > header
> >       that contains
> >                      the AOR (as in R-URI) as last entry, THEN when
> > rewriting
> >       the Request-URI
> >                      it has to tag that existing received entry AND
> add the
> >       new entry with
> >                      the new value of the Request-URI i.e.
> > the registered
> >       contact. (I think
> >                      that was how History-Info works right.)
> >                             2. received request contains History-Info
> > header
> >       that does not
> >                      contain the AOR (as in R-URI) as last entry, THEN
> when
> >       rewriting the
> >                      Request-URI it has add an entry with the AOR and
> tag
> >       that entry AND add
> >                      the new entry with the new value of the
> Request-URI
> > i.e.
> >       the registered
> >                      contact.
> >                             3. received request does not contain a
> >       History-Info header, THEN
> >                      when rewriting the Request-URI it has to add an
> entry
> >       with the AOR as in
> >                      the Request-URI and tag that entry AND add the
> new
> > entry
> >       with the new
> >                      value of the Request-URI i.e. the registered
> contact..
> >
> >
> >                                     I don't view it as a problem
> > necessarily
> >       if you'd like
> >                      to pursue an alternate solution, we just need to
> be
> >       clear that's what
> >                      you're advocating. In that case, I would agree
> the
> >       "retarget" is not an
> >                      appropriate tag for the entries, as you don't
> know at
> >       the point in time
> >                      that you add an hi-entry that it will be
> retargeted.
> > In
> >       the case of this
> >                      approach to the solution, you do have to alter
> the
> >       mechanism for
> >                      determining the hi-entry that your services would
> want
> >       to use. You would
> >                      need to skip the most recent/last hi-entry (as
> that
> >       would contain the
> >                      same as the Request-URI in the incoming request),
>
> > since
> >       it would be
> >                      tagged if you use this solution approach.
> >
> >
> >                             Yes, but only if the previous proxy added
> it.
> > As
> >       History-Info is
> >                      optional one never knows really.
> >
> >
> >
> >                                     Thus, you'd have to take the next
> >       hi-entry that you find
> >                      that has the desired "tag".  I will posit that we
>
> > could
> >       accomplish this
> >                      solution without any changes to 4244 by just
> > registering
> >       new SIP URI
> >                      parameter(s).   And,  that's not to say
> > we shouldn't do
> >       a 4244bis as
> >                      there are other issues to address in that
> document,
> > but
> >       that allows the
> >                      target-uri to describe a solution that makes use
> of
> >       4244bis "as is".
> >
> >
> >
> >                             Are you proposing to decouple the
> target-uri
> > and
> >       the 4244bis
> >                      discussion?
> >
> >
> >                                     So, feedback on this specific
> point
> > (from
> >       Hans Erik and
> >                      others) would be appreciated.
> >
> >
> >
> >                                     Mary.
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use
> sip-implementors@cs.columbia.edu for questions on current sip Use
> sipping@ietf.org for new developments on the application of sip
>

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I guess you mean that you agree with John that such rule can not be reliabl=
y construed.<br>I dont think I have seen convincing proof that such is the =
case.<br><br>Are you also saying that we may only consider loose routing be=
ing used?<br>
<br>Further for &quot;configured mappings&quot; for which we never saw a pr=
oper definition. But assuming that this is a proxy that can be configured t=
o do a Request-URI=A0 rewrite based on some configured setting.<br><br>In t=
hat case there are two variants of such mapping:<br>
1. A mapping that changes the identity of the target to be reached. (This i=
s what call forwarding or diversion does normally.)<br>2. A mapping that on=
ly changes=A0 the next hop route taken towards the target identity that has=
 not been changed. <br>
<br>For the UAS it only matters that the home domain does the tagging consi=
stently and correctly if it wants to rely on such information.<br><br>Other=
wise we would have to start adding information about the domain that added =
a specific hi-entry, if one wants to be sure who added the particular hi-en=
try.<br>
<br clear=3D"all">/Hans Erik van Elburg<br>
<br><br><div class=3D"gmail_quote">On Fri, Mar 13, 2009 at 7:43 PM, Mary Ba=
rnes <span dir=3D"ltr">&lt;<a href=3D"mailto:mary.barnes@nortel.com">mary.b=
arnes@nortel.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote"=
 style=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.=
8ex; padding-left: 1ex;">
<div id=3D":g7" class=3D"ii gt">Hi John,<br>
<br>
Thanks for all your feedback thus far on this topic. I have a couple<br>
comments below [MB].<br>
<br>
Mary.<br>
<div class=3D"im"><br>
-----Original Message-----<br>
From: <a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a> [mai=
lto:<a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a>] On Be=
half Of<br>
</div><div class=3D"im">Elwell, John<br>
Sent: Friday, March 13, 2009 11:31 AM<br>
To: Hans Erik van Elburg<br>
</div><div><div></div><div class=3D"h5">Cc: <a href=3D"mailto:sip@ietf.org"=
>sip@ietf.org</a><br>
Subject: Re: [Sip] Issue 3 - two tags (was RE:<br>
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br>
<br>
<br>
<br>
<br>
&gt; -----Original Message-----<br>
&gt; From: Hans Erik van Elburg [mailto:<a href=3D"mailto:ietf.hanserik@gma=
il.com">ietf.hanserik@gmail.com</a>]<br>
&gt; Sent: 13 March 2009 14:52<br>
&gt; To: Elwell, John<br>
&gt; Cc: <a href=3D"mailto:sip@ietf.org">sip@ietf.org</a><br>
&gt; Subject: Re: [Sip] Issue 3 - two tags (was RE:<br>
&gt; I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br>
&gt;<br>
&gt; It knows when it rewrites the Request-URI due to a location lookup --&=
gt;<br>
<br>
&gt; tag &quot;aor&quot;.<br>
&gt; It knows (is configured to) when it rewrites the Request-URI with a<br=
>
&gt; value that leads the request to target with a different identity then<=
br>
&gt; the one that has been replaced --&gt; tag new entry &quot;istarget&quo=
t;.<br>
&gt; It knows when it rewrites the Request-URI with a value and this leads<=
br>
&gt; to the first History-Info entry being recorded --&gt; tag &quot;istarg=
et&quot;.<br>
[JRE] It does not know that this is the first target, because upstream<br>
nodes may have retargeted without inserting H-I. In any case, the To<br>
header field URI is the first target, so we don&#39;t need that anyway.<br>
</div></div>[MB] John is correct. There is no rule about tagging the first<=
br>
History-Info entry. It may be tagged if the next node changes the<br>
request-URI via a service lookup or some other manner such as<br>
configuration.[/MB]<br>
<div class=3D"im"><br>
I can accept that a proxy might know, having done a LS look-up, whether<br>
the new URI is a registered contact or some other URI, and on that basis<br=
>
it could choose whether or not to insert a single tag. I thought that<br>
was the basis on which we agreed to go ahead with this updated to RFC<br>
4424, rather than following the loose route draft. It is not clear to me<br=
>
that a proxy would have enough information to make a 3-way choice.<br>
</div>[MB] My understanding is that there are just two ways in which the<br=
>
request-URI changes if you are using loose routing. It&#39;s either changed=
<br>
due to a lookup in a location server, whose information was populated by<br=
>
the UA during registration, OR it is changed due to configuration data<br>
(either system, user, etc.). In both cases, obviously, the proxy that is<br=
>
doing this is responsible for the domain of the request-URI in the<br>
incoming request. If it were not the case, then all the proxy can do is<br>
forward the outgoing request to the next hop. So, I think that may be<br>
where the concept of a 3rd choice comes in. But, as far as tagging, we<br>
(as I understand it) at least were discussing the potential of two tags<br>
only - i.e., further differentiating the cases that are currently all<br>
captured by the &quot;retarget&quot; as defined in 4244bis, which is the sa=
me<br>
mechanism that is specified in the target-uri document EXCEPT that<br>
document was only capturing the service lookups, which we believe is<br>
incorrect. [/MB]<br>
<div><div></div><div class=3D"h5"><br>
John<br>
<br>
<br>
&gt;<br>
&gt; /Hans Erik van Elburg<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On Fri, Mar 13, 2009 at 3:39 PM, Elwell, John<br>
&gt; &lt;<a href=3D"mailto:john.elwell@siemens.com">john.elwell@siemens.com=
</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0Hans Erik,<br>
&gt;<br>
&gt; =A0 =A0 =A0 But how does a proxy know what to use? How does it know wh=
ether<br>
the<br>
&gt; =A0 =A0 =A0 received Request-URI, which it uses to query the LS, is a =
target<br>
or<br>
&gt; an<br>
&gt; =A0 =A0 =A0 AoR? Does it need to scan back up the H-I chain to see whe=
ther<br>
there<br>
&gt; is<br>
&gt; =A0 =A0 =A0 already an &#39;istarget&#39; present, and if so use &#39;=
aor&#39; instead of<br>
&gt; =A0 =A0 =A0 &#39;istarget&#39;?<br>
&gt;<br>
&gt; =A0 =A0 =A0 John<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 ________________________________<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0From: Hans Erik van Elburg<br>
&gt; [mailto:<a href=3D"mailto:ietf.hanserik@gmail.com">ietf.hanserik@gmail=
.com</a>]<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0Sent: 13 March 2009 14:33<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0To: Elwell, John<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0Cc: <a href=3D"mailto:sip@ietf.org">sip@iet=
f.org</a><br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0Subject: Re: [Sip] Issue 3 - two tags (was =
RE:<br>
&gt; =A0 =A0 =A0 I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<b=
r>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0Hi John,<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0Well that is simple one would use different=
 tags for<br>
different<br>
&gt; =A0 =A0 =A0 type of entries, like in the example I already gave:<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 H-I: =A0Bfreephone; &quot;istarget&quo=
t;,\<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0B,&quot;aor&quot;\<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0contact-B<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0Where the UAS interested in the initial tar=
get with which<br>
it<br>
&gt; was<br>
&gt; =A0 =A0 =A0 addressed it would obtain the &quot;istarget&quot; tagged =
entry.<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0Where the UAS interested in the AOR =A0that=
 was used to<br>
address<br>
&gt; it<br>
&gt; =A0 =A0 =A0 would obtain the &quot;aor&quot; tagged entry.<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0In simpler cases it could be that one entry=
 holds the<br>
&quot;aor&quot; as<br>
&gt; =A0 =A0 =A0 well as the &quot;istarget&quot; tag.<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0Of course the algorithm needs some work to =
make it<br>
complete.<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0/Hans Erik van Elburg<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0On Fri, Mar 13, 2009 at 3:24 PM, Elwell, Jo=
hn<br>
&gt; =A0 =A0 =A0 &lt;<a href=3D"mailto:john.elwell@siemens.com">john.elwell=
@siemens.com</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Hans Erik,<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Following your detailed des=
cription, it seems<br>
that the<br>
&gt; =A0 =A0 =A0 UAS serving a<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0freephone number could rece=
ive one of the<br>
following<br>
&gt; =A0 =A0 =A0 (assuming H-I is<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0supported by all relevant n=
odes):<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A01) H-I containing:<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0- the freephone number (tag=
ged);<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0- the AoR (tagged);<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0- the contact URI (not tagg=
ed).<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Or<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A02) H-I containing:<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0- the freephone number (tag=
ged);<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0- the contact URI (not tagg=
ed).<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0The latter would occur when=
 the same proxy<br>
translates<br>
&gt; =A0 =A0 =A0 the freephone<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0number (via the AoR) to the=
 contact directly.<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0So how does this help the U=
AS to figure out the<br>
&gt; =A0 =A0 =A0 freephone number? It<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0seems the UAS would have to=
 search back through<br>
the<br>
&gt; =A0 =A0 =A0 tagged entries<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0looking for one that matche=
s a freephone number<br>
it<br>
&gt; =A0 =A0 =A0 recognises? Does the<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0tagging really add much val=
ue?<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0John<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0___________________________=
_____<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 From: <a href=
=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a><br>
&gt; =A0 =A0 =A0 [mailto:<a href=3D"mailto:sip-bounces@ietf.org">sip-bounce=
s@ietf.org</a>] On<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Behalf Of Hans Erik van Elb=
urg<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Sent: 12 March=
 2009 21:26<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 To: Mary Barne=
s<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Cc: Hans Erik =
van Elburg; <a href=3D"mailto:sip@ietf.org">sip@ietf.org</a><br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Subject: Re: [=
Sip] Issue 3 - two tags (was<br>
RE:<br>
&gt;<br>
&gt; I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Inline...<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 /Hans Erik van=
 Elburg<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 On Thu, Mar 12=
, 2009 at 7:10 PM, Mary<br>
Barnes<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&lt;<a href=3D"mailto:mary.=
barnes@nortel.com">mary.barnes@nortel.com</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 I think there is another solution<br>
&gt; =A0 =A0 =A0 proposal that we<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0should pursue for issue 3 i=
dentified in the email<br>
&gt; =A0 =A0 =A0 summary sent by Shida<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0earlier this week - I&#39;v=
e snipped all the text<br>
except<br>
&gt; for<br>
&gt; =A0 =A0 =A0 that issue and<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0issue 2 in the summary belo=
w. =A0I do think the<br>
&gt; discussion<br>
&gt; =A0 =A0 =A0 has confounded<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the two issues and I think =
it ought to be<br>
clarified<br>
&gt; that<br>
&gt; =A0 =A0 =A0 if we agree to<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the two tags per issue 3, t=
hat the options for<br>
the<br>
&gt; =A0 =A0 =A0 terminology MUST<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0change. And, perhaps this i=
s why the other<br>
discussion<br>
&gt; =A0 =A0 =A0 threads aren&#39;t<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0progressing well.<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Agree.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 So, Hans Erick, I would like<br>
&gt; =A0 =A0 =A0 clarification on as to your<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0view on the solution:<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 In general, ISTM that one of the<br>
issues<br>
&gt; =A0 =A0 =A0 we&#39;re having in<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0this terminology thread is =
that you are<br>
considering<br>
&gt; the<br>
&gt; =A0 =A0 =A0 solution to be<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0that the hi-entries are tag=
ged as they<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 are added to the request.<br>
&gt; =A0And, just for<br>
&gt; =A0 =A0 =A0 clarification<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0from my perspective, that i=
s NOT the solution in<br>
&gt; either<br>
&gt; =A0 =A0 =A0 4244bis or<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0target-uri.<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 The principle =
should be that the node that<br>
has<br>
&gt; =A0 =A0 =A0 the information<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0about what type of rewrite =
is performed should<br>
add the<br>
&gt; =A0 =A0 =A0 tag. (This is<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0done by the current target-=
uri draft allthough<br>
the<br>
&gt; =A0 =A0 =A0 solution is not<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0complete.)<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 For the freeph=
one service, that would be<br>
the<br>
&gt; =A0 =A0 =A0 freephone service.<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0For the AOR that is the hom=
e proxy.<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Assume that th=
e freephone service receives<br>
an<br>
&gt; =A0 =A0 =A0 INVITE =A0request<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0that either<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 1. received re=
quest contains History-Info<br>
&gt; header<br>
&gt; =A0 =A0 =A0 that contains<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the freephone number (as in=
 R-URI) as last entry,<br>
THEN<br>
&gt; =A0 =A0 =A0 when rewriting<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the Request-URI it has to t=
ag that existing<br>
received<br>
&gt; =A0 =A0 =A0 entry AND add the<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0new entry with the new valu=
e of the Request-URI.<br>
(I<br>
&gt; =A0 =A0 =A0 think that was how<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0History-Info works right.)<=
br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 2. received re=
quest contains History-Info<br>
&gt; header<br>
&gt; =A0 =A0 =A0 that does not<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0contain the freephone numbe=
r as last entry, THEN<br>
when<br>
&gt; =A0 =A0 =A0 rewriting the<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Request-URI it has add an e=
ntry with the<br>
freephone<br>
&gt; =A0 =A0 =A0 number and tag that<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0entry AND add the new entry=
 with the new value of<br>
the<br>
&gt; =A0 =A0 =A0 Request-URI.<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 3. received re=
quest does not contain a<br>
&gt; =A0 =A0 =A0 History-Info header, THEN<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0when rewriting the Request-=
URI it has to add an<br>
entry<br>
&gt; =A0 =A0 =A0 with the freephone<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0number and tag that entry A=
ND add the new entry<br>
with<br>
&gt; the<br>
&gt; =A0 =A0 =A0 new value of<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the Request-URI.<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 After forwardi=
ng this request it arrives<br>
at the<br>
&gt; =A0 =A0 =A0 home proxy:<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 1. received re=
quest contains History-Info<br>
&gt; header<br>
&gt; =A0 =A0 =A0 that contains<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the AOR (as in R-URI) as la=
st entry, THEN when<br>
&gt; rewriting<br>
&gt; =A0 =A0 =A0 the Request-URI<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0it has to tag that existing=
 received entry AND<br>
add the<br>
&gt; =A0 =A0 =A0 new entry with<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the new value of the Reques=
t-URI i.e.<br>
&gt; the registered<br>
&gt; =A0 =A0 =A0 contact. (I think<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0that was how History-Info w=
orks right.)<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 2. received re=
quest contains History-Info<br>
&gt; header<br>
&gt; =A0 =A0 =A0 that does not<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0contain the AOR (as in R-UR=
I) as last entry, THEN<br>
when<br>
&gt; =A0 =A0 =A0 rewriting the<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Request-URI it has add an e=
ntry with the AOR and<br>
tag<br>
&gt; =A0 =A0 =A0 that entry AND add<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the new entry with the new =
value of the<br>
Request-URI<br>
&gt; i.e.<br>
&gt; =A0 =A0 =A0 the registered<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0contact.<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 3. received re=
quest does not contain a<br>
&gt; =A0 =A0 =A0 History-Info header, THEN<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0when rewriting the Request-=
URI it has to add an<br>
entry<br>
&gt; =A0 =A0 =A0 with the AOR as in<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0the Request-URI and tag tha=
t entry AND add the<br>
new<br>
&gt; entry<br>
&gt; =A0 =A0 =A0 with the new<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0value of the Request-URI i.=
e. the registered<br>
contact..<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 I don&#39;t view it as a problem<br>
&gt; necessarily<br>
&gt; =A0 =A0 =A0 if you&#39;d like<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0to pursue an alternate solu=
tion, we just need to<br>
be<br>
&gt; =A0 =A0 =A0 clear that&#39;s what<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0you&#39;re advocating. In t=
hat case, I would agree<br>
the<br>
&gt; =A0 =A0 =A0 &quot;retarget&quot; is not an<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0appropriate tag for the ent=
ries, as you don&#39;t<br>
know at<br>
&gt; =A0 =A0 =A0 the point in time<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0that you add an hi-entry th=
at it will be<br>
retargeted.<br>
&gt; In<br>
&gt; =A0 =A0 =A0 the case of this<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0approach to the solution, y=
ou do have to alter<br>
the<br>
&gt; =A0 =A0 =A0 mechanism for<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0determining the hi-entry th=
at your services would<br>
want<br>
&gt; =A0 =A0 =A0 to use. You would<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0need to skip the most recen=
t/last hi-entry (as<br>
that<br>
&gt; =A0 =A0 =A0 would contain the<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0same as the Request-URI in =
the incoming request),<br>
<br>
&gt; since<br>
&gt; =A0 =A0 =A0 it would be<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0tagged if you use this solu=
tion approach.<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Yes, but only =
if the previous proxy added<br>
it.<br>
&gt; As<br>
&gt; =A0 =A0 =A0 History-Info is<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0optional one never knows re=
ally.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 Thus, you&#39;d have to take the next<br>
&gt; =A0 =A0 =A0 hi-entry that you find<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0that has the desired &quot;=
tag&quot;. =A0I will posit that we<br>
<br>
&gt; could<br>
&gt; =A0 =A0 =A0 accomplish this<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0solution without any change=
s to 4244 by just<br>
&gt; registering<br>
&gt; =A0 =A0 =A0 new SIP URI<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0parameter(s). =A0 And, =A0t=
hat&#39;s not to say<br>
&gt; we shouldn&#39;t do<br>
&gt; =A0 =A0 =A0 a 4244bis as<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0there are other issues to a=
ddress in that<br>
document,<br>
&gt; but<br>
&gt; =A0 =A0 =A0 that allows the<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0target-uri to describe a so=
lution that makes use<br>
of<br>
&gt; =A0 =A0 =A0 4244bis &quot;as is&quot;.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Are you propos=
ing to decouple the<br>
target-uri<br>
&gt; and<br>
&gt; =A0 =A0 =A0 the 4244bis<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0discussion?<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 So, feedback on this specific<br>
point<br>
&gt; (from<br>
&gt; =A0 =A0 =A0 Hans Erik and<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0others) would be appreciate=
d.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 Mary.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
</div></div>_______________________________________________<br>
Sip mailing list =A0<a href=3D"https://www.ietf.org/mailman/listinfo/sip" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
This list is for NEW development of the core SIP Protocol Use<br>
<a href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.col=
umbia.edu</a> for questions on current sip Use<br>
<a href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a> for new developmen=
ts on the application of sip</div></blockquote></div><br>

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From mary.barnes@nortel.com  Fri Mar 13 14:50:31 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Hans Erik van Elburg" <ietf.hanserik@gmail.com>
Cc: sip@ietf.org, "Elwell, John" <john.elwell@siemens.com>
Subject: Re: [Sip] Issue 3 - two tags (was RE: I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Responses below [MB].

________________________________

From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
Sent: Friday, March 13, 2009 4:17 PM
To: Barnes, Mary (RICH2:AR00)
Cc: Elwell, John; sip@ietf.org
Subject: Re: [Sip] Issue 3 - two tags (was RE:
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


I guess you mean that you agree with John that such rule can not be
reliably construed.=20
[MB] Which rule?  I fully agree we can discern the 3 cases as I
explained below. Per 16.5 (Determining Request Targets) in RFC 3261,
these are unique and recognized situations. The reason I refer to loose
routing is that it's my understanding it is far more commonly used and
RFC 3261 doesn't even describe strict routing - it focuses on loose
routing and strict routing support is for 2543 backwards compatibility
as I understood. But, actually it makes no difference to History-Info
processing.  The important thing really, is that you MUST tag the old
entry as you are doing the processing per 16.5 as that information is
lost by the time you get around to adding the next History-Info header
in section 16.6 ( Request Forwarding). I think one thing that might be
being missed in this thread is that I have been very careful to rely
explicitly on what is described in 16.5 - that a must read to understand
what I am saying - again there is a lack of clarity in that section and
we can suggest clarifications once we all can finally agree appropriate
terms. If you'd like me to pull the explicit text that supports
everything I've been saying, I can do so, but it's somewhat a waste of
time. [/MB]
=20
I dont think I have seen convincing proof that such is the case.=20
[MB] And, I'm not saying that you can't construct such a rule - indeed I
am saying that you certainly should be able to per RFC 3261 section
16.5.  Can you point out what text below that suggests I am saying that?
[/MB]=20

Are you also saying that we may only consider loose routing being used?=20
[MB] Nope. [/MB]=20

Further for "configured mappings" for which we never saw a proper
definition. But assuming that this is a proxy that can be configured to
do a Request-URI  rewrite based on some configured setting.=20
[MB] The concept of "configured mappings" is described in RFC 3261 -
perhaps not with the level of granularity or specificity that some might
desire, but it's there - it's initially described as an abstract
location service (1st Paragraph section 16.5).  And, then later it
specifies that a proxy can determine "targets" (3261 term not mine) just
about any ole way it wants, so I don't know that we need to define a new
term for "configured mappings" per se - there is no limitation as to the
mechanism one can use to determine the request-URI for the outgoing
request (once it is determined that the Proxy is ALLOWED  to overwrite
the request-URI - the right domain, etc.),  This is a fundamental SIP
concept that many products use to give a (human) user flexibility in
configuring where an incoming call might terminate. So, I see no issue
at all is specifying two tags to capture this rewrite as it's already in
section 16.5 that this is information that a Proxy should be able to
derive. [/MB]=20

In that case there are two variants of such mapping:
1. A mapping that changes the identity of the target to be reached.
(This is what call forwarding or diversion does normally.)
2. A mapping that only changes  the next hop route taken towards the
target identity that has not been changed.  =20

For the UAS it only matters that the home domain does the tagging
consistently and correctly if it wants to rely on such information.=20
[MB] I don't know that I agree it's the home domain - it's a proxy that
has responsibility for the domain in the request-URI that is able to do
the service lookup or using any other sort of mapping to determine the
new URI for the request-URI in the outgoing request.  I don't know that
we want to restrict that to the "home" domain. If we restrict it to the
home domain, then that implies that we are only tagging the entries that
are determined based on service lookup in a Registrar (per RFC 3261's
definition of a "home domain").  [/MB]

Otherwise we would have to start adding information about the domain
that added a specific hi-entry, if one wants to be sure who added the
particular hi-entry.=20
[MB] I'm not sure why this would be needed and I wasn't espousing that
it would - we know by definition that the domain that mucked with a URI
is the domain associated with the URI that was overwritten/lost, etc.
[/MB]=20

/Hans Erik van Elburg



On Fri, Mar 13, 2009 at 7:43 PM, Mary Barnes <mary.barnes@nortel.com>
wrote:


	Hi John,
=09
	Thanks for all your feedback thus far on this topic. I have a
couple
	comments below [MB].
=09
	Mary.
=09

	-----Original Message-----
	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of
=09
	Elwell, John
	Sent: Friday, March 13, 2009 11:31 AM
	To: Hans Erik van Elburg
=09
	Cc: sip@ietf.org
	Subject: Re: [Sip] Issue 3 - two tags (was RE:
	I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
=09
=09
=09
=09
	> -----Original Message-----
	> From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
	> Sent: 13 March 2009 14:52
	> To: Elwell, John
	> Cc: sip@ietf.org
	> Subject: Re: [Sip] Issue 3 - two tags (was RE:
	> I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
	>
	> It knows when it rewrites the Request-URI due to a location
lookup -->
=09
	> tag "aor".
	> It knows (is configured to) when it rewrites the Request-URI
with a
	> value that leads the request to target with a different
identity then
	> the one that has been replaced --> tag new entry "istarget".
	> It knows when it rewrites the Request-URI with a value and
this leads
	> to the first History-Info entry being recorded --> tag
"istarget".
	[JRE] It does not know that this is the first target, because
upstream
	nodes may have retargeted without inserting H-I. In any case,
the To
	header field URI is the first target, so we don't need that
anyway.
=09
	[MB] John is correct. There is no rule about tagging the first
	History-Info entry. It may be tagged if the next node changes
the
	request-URI via a service lookup or some other manner such as
	configuration.[/MB]
=09

	I can accept that a proxy might know, having done a LS look-up,
whether
	the new URI is a registered contact or some other URI, and on
that basis
	it could choose whether or not to insert a single tag. I thought
that
	was the basis on which we agreed to go ahead with this updated
to RFC
	4424, rather than following the loose route draft. It is not
clear to me
	that a proxy would have enough information to make a 3-way
choice.
=09
	[MB] My understanding is that there are just two ways in which
the
	request-URI changes if you are using loose routing. It's either
changed
	due to a lookup in a location server, whose information was
populated by
	the UA during registration, OR it is changed due to
configuration data
	(either system, user, etc.). In both cases, obviously, the proxy
that is
	doing this is responsible for the domain of the request-URI in
the
	incoming request. If it were not the case, then all the proxy
can do is
	forward the outgoing request to the next hop. So, I think that
may be
	where the concept of a 3rd choice comes in. But, as far as
tagging, we
	(as I understand it) at least were discussing the potential of
two tags
	only - i.e., further differentiating the cases that are
currently all
	captured by the "retarget" as defined in 4244bis, which is the
same
	mechanism that is specified in the target-uri document EXCEPT
that
	document was only capturing the service lookups, which we
believe is
	incorrect. [/MB]
=09

	John
=09
=09
	>
---remainder of theread snipped by Mary. --------

From christer.holmberg@ericsson.com  Fri Mar 13 15:26:36 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Mary Barnes" <mary.barnes@nortel.com>, "Hans Erik van Elburg" <ietf.hanserik@gmail.com>, "Dean Willis" <dean.willis@softarmor.com>
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Cc: sip@ietf.org, "Elwell, John" <john.elwell@siemens.com>
Subject: Re: [Sip] Terminology (was RE: Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Hi,

I think we shall try to agree on the definitions first - then we can
decide where to put it.

But, wherever we put it, I think there shall be a separate and clear
definitions chapter. Other chapters then simply refers to that.

Regards,

Christer=20

-----Original Message-----
From: Mary Barnes [mailto:mary.barnes@nortel.com]=20
Sent: Friday, March 13, 2009 4:31 PM
To: Christer Holmberg; Hans Erik van Elburg; Dean Willis
Cc: sip@ietf.org; Elwell, John
Subject: RE: [Sip] Terminology (was RE:
Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt

That's fine - we just need to be careful with the wording in 4244 so
that the existing normative functionality isn't so impacted. I'll look
at the detailed impact of the changes as retarget is quite prevalent in
that document. It could be that we can leave it in the early sections
(and define it to be the same as the term we agree) and just keep it out
of the normative sections - right now, it is only used in a few places
other than describing the new parameter in the normative sections.
=20
Mary.

________________________________

From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
Sent: Friday, March 13, 2009 6:42 AM
To: Hans Erik van Elburg; Dean Willis
Cc: Barnes, Mary (RICH2:AR00); sip@ietf.org; Elwell, John
Subject: RE: [Sip] Terminology (was RE:
Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


Hi,
=20
I think we should use those definitions as base. That way I think we
would be able to progress faster.
=20
Because, I think we really need to agree on the definitions FIRST. There
is no idea on discussing detailed wording in 4244bis if we don't have a
common understanding of what we mean.
=20
Regards,
=20
Christer
=20


________________________________

	From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
	Sent: 13. maaliskuuta 2009 12:19
	To: Dean Willis
	Cc: Christer Holmberg; Mary Barnes; sip@ietf.org; Elwell, John
	Subject: Re: [Sip] Terminology (was RE:
Fwd:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
=09
=09
	Hi Dean,
=09
	Yes this was exactly how we used/defined the terms at the time
we where discussing the ua-loose-route vs Target-header solutions.
=09
	I still think that this way they are most intuitive and closest
to how you would use the terms in natural language.=20
=09
	/Hans Erik van Elburg
=09
=09
	On Thu, Mar 12, 2009 at 8:27 PM, Dean Willis
<dean.willis@softarmor.com> wrote:
=09


		On Mar 12, 2009, at 1:18 PM, Christer Holmberg wrote:
	=09
	=09

			A couple of question:
		=09
			1. Does the 4244bis definition of "retarget"
cover the freephone case, when the R-URI is replaced with the AoR of B -
not with the contact of B - no matter whether it's done based on
location service, configuration or whatever?
		=09


		By the way, I believe that the way I use the words, this
is a "reroute" and not a "retarget". In the freephone case, I would
still expect the destination to have an awareness that it is the target
of freephone calls and have credentials for the freephone number such
that it could appropriately authenticate its responses if we had a means
to do so.=20




			2. Related to the first question, when you say
"...and thus chaning the target of the request", what is the defintion
of "target"?
		=09
		=09



		My definition is that a "retarget" changes the expected
identity of the expected responder, introducing the possibility of an
"unanticipated respondent" scenario.
	=09
		This is, AFAIK, very different from the 4244
terminology, which I've always held to be not particularly useful.
	=09
		--
		Dean
	=09
	=09



From christer.holmberg@ericsson.com  Fri Mar 13 15:45:37 2009
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Hi Dale,=20

>I don't know if this has been mentioned, but I skimmed
draft-ietf-sip-199-06 and couldn't find it:
>
>One thing that I've always felt was a weakness of the "199" proposal
was that it was not possible to associate the early dialog of the 199
with the particular media streams of that early dialog, which=20
>makes it difficult for a UAC to use a 199 response to free
media-processing resources.

It is true that the 199 spec doesn't define mechanisms how to do the
association. I think we a long time ago agreed that it is outside the
scope of the document.

But, as indicated in the draft, even if the UAC is not able to associate
the dialog with a particular strem, when it receives 199 it can e.g.
remove media resources (codecs etc) which have only been reserved for
that particular dialgo.=20

>However, if the UAS that creates an early media stream sends a 1xx
(e.g., 183) *reliably*, then the UAC is certain to be able to associate
the to-tag of the 1xx (which is the same as the to-tag of the=20
>199) with the early media source address/port.
>
>Why isn't this fact put in the draft?  I suppose it isn't a
*requirement* for 199, but it seems like a very strong suggestion for
UAS behavior to enable the UAC to take advantage of 199.

Are you saying that if the UAS sends a reliable 18x with SDP, the UAC
will then be able to use the SDP information to associate the media with
the dialog?

That works with the following assumptions:

1. The UAS uses symmetric media, ie it sends media from the same
port/address where it wants to receive media (keep in mind that SDP only
indicates where you want to receive media).
2. There are no NATs in the media path.

Also, the draft doesn't say anything about whether to send 18x responses
reliably or not. If the UAS wants to make sure the UAC gets the SDP, it
should send the 18x reliably - no matter if he supports 199 or not.

Regards,

Christer


From HKaplan@acmepacket.com  Fri Mar 13 20:34:40 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Elwell, John" <john.elwell@siemens.com>, "sip@ietf.org" <sip@ietf.org>
Date: Fri, 13 Mar 2009 23:35:15 -0400
Thread-Topic: [Sip] FW: I-D Action:draft-kaplan-sip-asserter-identity-01.txt
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Hey John,
Thanks for reviewing the draft!  I was thinking that basically the asserter=
 should decide what it wants to sign, but I think you're right that a defau=
lt list would be good.=20

-hadriel

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com]
> Sent: Tuesday, March 10, 2009 5:29 AM
>=20
> An interesting draft, perhaps useful in two ways: first in the way you
> intend (as asserter ID in closed domains or federations of closed
> domains), but secondly some aspects might be relevant for inclusion in
> an eventual solution to the SIP-Identity problem, notably the means of
> selecting particular body parts and SDP attributes to sign.
>=20
> On that particular issue, I am not sure whether the asserter is in a
> position to decide on a request-by-request basis which body parts and
> SDP attributes to sign, since it cannot know what downstream entities
> are likely to need to do to those fields, so we may need some guidance
> (e.g., a default list of things to sign).


From ietf.hanserik@gmail.com  Sat Mar 14 00:54:07 2009
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Date: Sat, 14 Mar 2009 08:54:42 +0100
Message-ID: <9ae56b1e0903140054l467fcd8eu50676ea31c914734@mail.gmail.com>
From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: "sip@ietf.org List" <sip@ietf.org>
Content-Type: multipart/alternative; boundary=0015174c17a8b72bea04650f87ec
Subject: Re: [Sip] Issue 3 - two tags (was RE: I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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--0015174c17a8b72bea04650f87ec
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I just realised that there is an additional problem that makes it even
harder to do any distinction.

A UA may register as a contact an AOR of another user or resource.
Effectively this causes the request to be forwarded to the other user by
overwriting the Request-URI.

So the fact that the Request-URI was overwritten due to a location lookup,
doesn't tell you a bit about the nature of the rewrite.

/Hans Erik van Elburg


On Fri, Mar 13, 2009 at 10:53 PM, Mary Barnes <mary.barnes@nortel.com>wrote:

> Responses below [MB].
>
> ________________________________
>
> From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
> Sent: Friday, March 13, 2009 4:17 PM
> To: Barnes, Mary (RICH2:AR00)
> Cc: Elwell, John; sip@ietf.org
> Subject: Re: [Sip] Issue 3 - two tags (was RE:
> I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>
>
> I guess you mean that you agree with John that such rule can not be
> reliably construed.
> [MB] Which rule?  I fully agree we can discern the 3 cases as I
> explained below. Per 16.5 (Determining Request Targets) in RFC 3261,
> these are unique and recognized situations. The reason I refer to loose
> routing is that it's my understanding it is far more commonly used and
> RFC 3261 doesn't even describe strict routing - it focuses on loose
> routing and strict routing support is for 2543 backwards compatibility
> as I understood. But, actually it makes no difference to History-Info
> processing.  The important thing really, is that you MUST tag the old
> entry as you are doing the processing per 16.5 as that information is
> lost by the time you get around to adding the next History-Info header
> in section 16.6 ( Request Forwarding). I think one thing that might be
> being missed in this thread is that I have been very careful to rely
> explicitly on what is described in 16.5 - that a must read to understand
> what I am saying - again there is a lack of clarity in that section and
> we can suggest clarifications once we all can finally agree appropriate
> terms. If you'd like me to pull the explicit text that supports
> everything I've been saying, I can do so, but it's somewhat a waste of
> time. [/MB]
>
> I dont think I have seen convincing proof that such is the case.
> [MB] And, I'm not saying that you can't construct such a rule - indeed I
> am saying that you certainly should be able to per RFC 3261 section
> 16.5.  Can you point out what text below that suggests I am saying that?
> [/MB]
>
> Are you also saying that we may only consider loose routing being used?
> [MB] Nope. [/MB]
>
> Further for "configured mappings" for which we never saw a proper
> definition. But assuming that this is a proxy that can be configured to
> do a Request-URI  rewrite based on some configured setting.
> [MB] The concept of "configured mappings" is described in RFC 3261 -
> perhaps not with the level of granularity or specificity that some might
> desire, but it's there - it's initially described as an abstract
> location service (1st Paragraph section 16.5).  And, then later it
> specifies that a proxy can determine "targets" (3261 term not mine) just
> about any ole way it wants, so I don't know that we need to define a new
> term for "configured mappings" per se - there is no limitation as to the
> mechanism one can use to determine the request-URI for the outgoing
> request (once it is determined that the Proxy is ALLOWED  to overwrite
> the request-URI - the right domain, etc.),  This is a fundamental SIP
> concept that many products use to give a (human) user flexibility in
> configuring where an incoming call might terminate. So, I see no issue
> at all is specifying two tags to capture this rewrite as it's already in
> section 16.5 that this is information that a Proxy should be able to
> derive. [/MB]
>
> In that case there are two variants of such mapping:
> 1. A mapping that changes the identity of the target to be reached.
> (This is what call forwarding or diversion does normally.)
> 2. A mapping that only changes  the next hop route taken towards the
> target identity that has not been changed.
>
> For the UAS it only matters that the home domain does the tagging
> consistently and correctly if it wants to rely on such information.
> [MB] I don't know that I agree it's the home domain - it's a proxy that
> has responsibility for the domain in the request-URI that is able to do
> the service lookup or using any other sort of mapping to determine the
> new URI for the request-URI in the outgoing request.  I don't know that
> we want to restrict that to the "home" domain. If we restrict it to the
> home domain, then that implies that we are only tagging the entries that
> are determined based on service lookup in a Registrar (per RFC 3261's
> definition of a "home domain").  [/MB]
>
> Otherwise we would have to start adding information about the domain
> that added a specific hi-entry, if one wants to be sure who added the
> particular hi-entry.
> [MB] I'm not sure why this would be needed and I wasn't espousing that
> it would - we know by definition that the domain that mucked with a URI
> is the domain associated with the URI that was overwritten/lost, etc.
> [/MB]
>
> /Hans Erik van Elburg
>
>
>
> On Fri, Mar 13, 2009 at 7:43 PM, Mary Barnes <mary.barnes@nortel.com>
> wrote:
>
>
>        Hi John,
>
>        Thanks for all your feedback thus far on this topic. I have a
> couple
>        comments below [MB].
>
>        Mary.
>
>
>        -----Original Message-----
>        From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> Behalf Of
>
>        Elwell, John
>        Sent: Friday, March 13, 2009 11:31 AM
>        To: Hans Erik van Elburg
>
>        Cc: sip@ietf.org
>        Subject: Re: [Sip] Issue 3 - two tags (was RE:
>        I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>
>
>
>
>        > -----Original Message-----
>        > From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
>        > Sent: 13 March 2009 14:52
>        > To: Elwell, John
>        > Cc: sip@ietf.org
>        > Subject: Re: [Sip] Issue 3 - two tags (was RE:
>        > I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>        >
>        > It knows when it rewrites the Request-URI due to a location
> lookup -->
>
>        > tag "aor".
>        > It knows (is configured to) when it rewrites the Request-URI
> with a
>        > value that leads the request to target with a different
> identity then
>        > the one that has been replaced --> tag new entry "istarget".
>        > It knows when it rewrites the Request-URI with a value and
> this leads
>        > to the first History-Info entry being recorded --> tag
> "istarget".
>        [JRE] It does not know that this is the first target, because
> upstream
>        nodes may have retargeted without inserting H-I. In any case,
> the To
>        header field URI is the first target, so we don't need that
> anyway.
>
>        [MB] John is correct. There is no rule about tagging the first
>        History-Info entry. It may be tagged if the next node changes
> the
>        request-URI via a service lookup or some other manner such as
>        configuration.[/MB]
>
>
>        I can accept that a proxy might know, having done a LS look-up,
> whether
>        the new URI is a registered contact or some other URI, and on
> that basis
>        it could choose whether or not to insert a single tag. I thought
> that
>        was the basis on which we agreed to go ahead with this updated
> to RFC
>        4424, rather than following the loose route draft. It is not
> clear to me
>        that a proxy would have enough information to make a 3-way
> choice.
>
>        [MB] My understanding is that there are just two ways in which
> the
>        request-URI changes if you are using loose routing. It's either
> changed
>        due to a lookup in a location server, whose information was
> populated by
>        the UA during registration, OR it is changed due to
> configuration data
>        (either system, user, etc.). In both cases, obviously, the proxy
> that is
>        doing this is responsible for the domain of the request-URI in
> the
>        incoming request. If it were not the case, then all the proxy
> can do is
>        forward the outgoing request to the next hop. So, I think that
> may be
>        where the concept of a 3rd choice comes in. But, as far as
> tagging, we
>        (as I understand it) at least were discussing the potential of
> two tags
>        only - i.e., further differentiating the cases that are
> currently all
>        captured by the "retarget" as defined in 4244bis, which is the
> same
>        mechanism that is specified in the target-uri document EXCEPT
> that
>        document was only capturing the service lookups, which we
> believe is
>        incorrect. [/MB]
>
>
>        John
>
>
>        >
> ---remainder of theread snipped by Mary. --------
>

--0015174c17a8b72bea04650f87ec
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

I just realised that there is an additional problem that makes it even hard=
er to do any distinction.<br><br>A UA may register as a contact an AOR of a=
nother user or resource. Effectively this causes the request to be forwarde=
d to the other user by overwriting the Request-URI.<br>
<br>So the fact that the Request-URI was overwritten due to a location look=
up, doesn&#39;t tell you a bit about the nature of the rewrite.<br><br clea=
r=3D"all">/Hans Erik van Elburg<br>
<br><br><div class=3D"gmail_quote">On Fri, Mar 13, 2009 at 10:53 PM, Mary B=
arnes <span dir=3D"ltr">&lt;<a href=3D"mailto:mary.barnes@nortel.com">mary.=
barnes@nortel.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote=
" style=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0=
.8ex; padding-left: 1ex;">
Responses below [MB].<br>
<br>
________________________________<br>
<div class=3D"im"><br>
From: Hans Erik van Elburg [mailto:<a href=3D"mailto:ietf.hanserik@gmail.co=
m">ietf.hanserik@gmail.com</a>]<br>
</div>Sent: Friday, March 13, 2009 4:17 PM<br>
<div class=3D"im">To: Barnes, Mary (RICH2:AR00)<br>
</div>Cc: Elwell, John; <a href=3D"mailto:sip@ietf.org">sip@ietf.org</a><br=
>
<div class=3D"im">Subject: Re: [Sip] Issue 3 - two tags (was RE:<br>
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br>
<br>
<br>
</div><div class=3D"im">I guess you mean that you agree with John that such=
 rule can not be<br>
reliably construed.<br>
</div>[MB] Which rule? =A0I fully agree we can discern the 3 cases as I<br>
explained below. Per 16.5 (Determining Request Targets) in RFC 3261,<br>
these are unique and recognized situations. The reason I refer to loose<br>
routing is that it&#39;s my understanding it is far more commonly used and<=
br>
RFC 3261 doesn&#39;t even describe strict routing - it focuses on loose<br>
routing and strict routing support is for 2543 backwards compatibility<br>
as I understood. But, actually it makes no difference to History-Info<br>
processing. =A0The important thing really, is that you MUST tag the old<br>
entry as you are doing the processing per 16.5 as that information is<br>
lost by the time you get around to adding the next History-Info header<br>
in section 16.6 ( Request Forwarding). I think one thing that might be<br>
being missed in this thread is that I have been very careful to rely<br>
explicitly on what is described in 16.5 - that a must read to understand<br=
>
what I am saying - again there is a lack of clarity in that section and<br>
we can suggest clarifications once we all can finally agree appropriate<br>
terms. If you&#39;d like me to pull the explicit text that supports<br>
everything I&#39;ve been saying, I can do so, but it&#39;s somewhat a waste=
 of<br>
time. [/MB]<br>
<div class=3D"im"><br>
I dont think I have seen convincing proof that such is the case.<br>
</div>[MB] And, I&#39;m not saying that you can&#39;t construct such a rule=
 - indeed I<br>
am saying that you certainly should be able to per RFC 3261 section<br>
16.5. =A0Can you point out what text below that suggests I am saying that?<=
br>
[/MB]<br>
<div class=3D"im"><br>
Are you also saying that we may only consider loose routing being used?<br>
</div>[MB] Nope. [/MB]<br>
<div class=3D"im"><br>
Further for &quot;configured mappings&quot; for which we never saw a proper=
<br>
definition. But assuming that this is a proxy that can be configured to<br>
do a Request-URI =A0rewrite based on some configured setting.<br>
</div>[MB] The concept of &quot;configured mappings&quot; is described in R=
FC 3261 -<br>
perhaps not with the level of granularity or specificity that some might<br=
>
desire, but it&#39;s there - it&#39;s initially described as an abstract<br=
>
location service (1st Paragraph section 16.5). =A0And, then later it<br>
specifies that a proxy can determine &quot;targets&quot; (3261 term not min=
e) just<br>
about any ole way it wants, so I don&#39;t know that we need to define a ne=
w<br>
term for &quot;configured mappings&quot; per se - there is no limitation as=
 to the<br>
mechanism one can use to determine the request-URI for the outgoing<br>
request (once it is determined that the Proxy is ALLOWED =A0to overwrite<br=
>
the request-URI - the right domain, etc.), =A0This is a fundamental SIP<br>
concept that many products use to give a (human) user flexibility in<br>
configuring where an incoming call might terminate. So, I see no issue<br>
at all is specifying two tags to capture this rewrite as it&#39;s already i=
n<br>
section 16.5 that this is information that a Proxy should be able to<br>
derive. [/MB]<br>
<div class=3D"im"><br>
In that case there are two variants of such mapping:<br>
1. A mapping that changes the identity of the target to be reached.<br>
(This is what call forwarding or diversion does normally.)<br>
2. A mapping that only changes =A0the next hop route taken towards the<br>
target identity that has not been changed.<br>
<br>
For the UAS it only matters that the home domain does the tagging<br>
consistently and correctly if it wants to rely on such information.<br>
</div>[MB] I don&#39;t know that I agree it&#39;s the home domain - it&#39;=
s a proxy that<br>
has responsibility for the domain in the request-URI that is able to do<br>
the service lookup or using any other sort of mapping to determine the<br>
new URI for the request-URI in the outgoing request. =A0I don&#39;t know th=
at<br>
we want to restrict that to the &quot;home&quot; domain. If we restrict it =
to the<br>
home domain, then that implies that we are only tagging the entries that<br=
>
are determined based on service lookup in a Registrar (per RFC 3261&#39;s<b=
r>
definition of a &quot;home domain&quot;). =A0[/MB]<br>
<div class=3D"im"><br>
Otherwise we would have to start adding information about the domain<br>
that added a specific hi-entry, if one wants to be sure who added the<br>
particular hi-entry.<br>
</div>[MB] I&#39;m not sure why this would be needed and I wasn&#39;t espou=
sing that<br>
it would - we know by definition that the domain that mucked with a URI<br>
is the domain associated with the URI that was overwritten/lost, etc.<br>
[/MB]<br>
<div><div></div><div class=3D"h5"><br>
/Hans Erik van Elburg<br>
<br>
<br>
<br>
On Fri, Mar 13, 2009 at 7:43 PM, Mary Barnes &lt;<a href=3D"mailto:mary.bar=
nes@nortel.com">mary.barnes@nortel.com</a>&gt;<br>
wrote:<br>
<br>
<br>
 =A0 =A0 =A0 =A0Hi John,<br>
<br>
 =A0 =A0 =A0 =A0Thanks for all your feedback thus far on this topic. I have=
 a<br>
couple<br>
 =A0 =A0 =A0 =A0comments below [MB].<br>
<br>
 =A0 =A0 =A0 =A0Mary.<br>
<br>
<br>
 =A0 =A0 =A0 =A0-----Original Message-----<br>
 =A0 =A0 =A0 =A0From: <a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@i=
etf.org</a> [mailto:<a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@iet=
f.org</a>] On<br>
Behalf Of<br>
<br>
 =A0 =A0 =A0 =A0Elwell, John<br>
 =A0 =A0 =A0 =A0Sent: Friday, March 13, 2009 11:31 AM<br>
 =A0 =A0 =A0 =A0To: Hans Erik van Elburg<br>
<br>
 =A0 =A0 =A0 =A0Cc: <a href=3D"mailto:sip@ietf.org">sip@ietf.org</a><br>
 =A0 =A0 =A0 =A0Subject: Re: [Sip] Issue 3 - two tags (was RE:<br>
 =A0 =A0 =A0 =A0I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br=
>
<br>
<br>
<br>
<br>
 =A0 =A0 =A0 =A0&gt; -----Original Message-----<br>
 =A0 =A0 =A0 =A0&gt; From: Hans Erik van Elburg [mailto:<a href=3D"mailto:i=
etf.hanserik@gmail.com">ietf.hanserik@gmail.com</a>]<br>
 =A0 =A0 =A0 =A0&gt; Sent: 13 March 2009 14:52<br>
 =A0 =A0 =A0 =A0&gt; To: Elwell, John<br>
 =A0 =A0 =A0 =A0&gt; Cc: <a href=3D"mailto:sip@ietf.org">sip@ietf.org</a><b=
r>
 =A0 =A0 =A0 =A0&gt; Subject: Re: [Sip] Issue 3 - two tags (was RE:<br>
 =A0 =A0 =A0 =A0&gt; I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.t=
xt<br>
 =A0 =A0 =A0 =A0&gt;<br>
 =A0 =A0 =A0 =A0&gt; It knows when it rewrites the Request-URI due to a loc=
ation<br>
lookup --&gt;<br>
<br>
 =A0 =A0 =A0 =A0&gt; tag &quot;aor&quot;.<br>
 =A0 =A0 =A0 =A0&gt; It knows (is configured to) when it rewrites the Reque=
st-URI<br>
with a<br>
 =A0 =A0 =A0 =A0&gt; value that leads the request to target with a differen=
t<br>
identity then<br>
 =A0 =A0 =A0 =A0&gt; the one that has been replaced --&gt; tag new entry &q=
uot;istarget&quot;.<br>
 =A0 =A0 =A0 =A0&gt; It knows when it rewrites the Request-URI with a value=
 and<br>
this leads<br>
 =A0 =A0 =A0 =A0&gt; to the first History-Info entry being recorded --&gt; =
tag<br>
&quot;istarget&quot;.<br>
 =A0 =A0 =A0 =A0[JRE] It does not know that this is the first target, becau=
se<br>
upstream<br>
 =A0 =A0 =A0 =A0nodes may have retargeted without inserting H-I. In any cas=
e,<br>
the To<br>
 =A0 =A0 =A0 =A0header field URI is the first target, so we don&#39;t need =
that<br>
anyway.<br>
<br>
 =A0 =A0 =A0 =A0[MB] John is correct. There is no rule about tagging the fi=
rst<br>
 =A0 =A0 =A0 =A0History-Info entry. It may be tagged if the next node chang=
es<br>
the<br>
 =A0 =A0 =A0 =A0request-URI via a service lookup or some other manner such =
as<br>
 =A0 =A0 =A0 =A0configuration.[/MB]<br>
<br>
<br>
 =A0 =A0 =A0 =A0I can accept that a proxy might know, having done a LS look=
-up,<br>
whether<br>
 =A0 =A0 =A0 =A0the new URI is a registered contact or some other URI, and =
on<br>
that basis<br>
 =A0 =A0 =A0 =A0it could choose whether or not to insert a single tag. I th=
ought<br>
that<br>
 =A0 =A0 =A0 =A0was the basis on which we agreed to go ahead with this upda=
ted<br>
to RFC<br>
 =A0 =A0 =A0 =A04424, rather than following the loose route draft. It is no=
t<br>
clear to me<br>
 =A0 =A0 =A0 =A0that a proxy would have enough information to make a 3-way<=
br>
choice.<br>
<br>
 =A0 =A0 =A0 =A0[MB] My understanding is that there are just two ways in wh=
ich<br>
the<br>
 =A0 =A0 =A0 =A0request-URI changes if you are using loose routing. It&#39;=
s either<br>
changed<br>
 =A0 =A0 =A0 =A0due to a lookup in a location server, whose information was=
<br>
populated by<br>
 =A0 =A0 =A0 =A0the UA during registration, OR it is changed due to<br>
configuration data<br>
 =A0 =A0 =A0 =A0(either system, user, etc.). In both cases, obviously, the =
proxy<br>
that is<br>
 =A0 =A0 =A0 =A0doing this is responsible for the domain of the request-URI=
 in<br>
the<br>
 =A0 =A0 =A0 =A0incoming request. If it were not the case, then all the pro=
xy<br>
can do is<br>
 =A0 =A0 =A0 =A0forward the outgoing request to the next hop. So, I think t=
hat<br>
may be<br>
 =A0 =A0 =A0 =A0where the concept of a 3rd choice comes in. But, as far as<=
br>
tagging, we<br>
 =A0 =A0 =A0 =A0(as I understand it) at least were discussing the potential=
 of<br>
two tags<br>
 =A0 =A0 =A0 =A0only - i.e., further differentiating the cases that are<br>
currently all<br>
 =A0 =A0 =A0 =A0captured by the &quot;retarget&quot; as defined in 4244bis,=
 which is the<br>
same<br>
 =A0 =A0 =A0 =A0mechanism that is specified in the target-uri document EXCE=
PT<br>
that<br>
 =A0 =A0 =A0 =A0document was only capturing the service lookups, which we<b=
r>
believe is<br>
 =A0 =A0 =A0 =A0incorrect. [/MB]<br>
<br>
<br>
 =A0 =A0 =A0 =A0John<br>
<br>
<br>
 =A0 =A0 =A0 =A0&gt;<br>
</div></div>---remainder of theread snipped by Mary. --------<br>
</blockquote></div><br>

--0015174c17a8b72bea04650f87ec--

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Rafael and others,

Thanks for putting this together.

Considering for a moment figure 1, this attack is possible only if UAs 
accept incoming requests from any place, and not just their proxy. In 
practice, this is seldom allowed. Indeed, if it is allowed, there are 
worse attacks than this which can be launched (free phone calls, spam 
calling, etc.). Using the SIP recommended TLS between UA and proxy also 
mitigates that.

So really figure 2 is the interesting one. However, this attack assumes 
that Alice has credentials on multiple systems. Again, in practice, this 
is extremely uncommon. Certainly none of the existing deployed 
enterprise or service provider consumer deployments are of this nature.

As such, I dont think this attack is likely in practice. However, in 
theory it is possible. The essence of the attack is that the victim is 
providing credentials to an unauthenticated server (since the attacker 
is acting like a server, asking for credentials). In that way, as others 
have pointed out, it is similar to baiting attacks that have been 
previously documented. With SIP it is most easily remedied by a rule 
which says, 'don't pass credentials for domain X to a server that is not 
domain X'. Server identity can be verified by normal server-only auth 
between a client and its server, but even that is not needed. A client 
will know which domain its proxy is representing, and once connected, it 
only provides credentials for that domain.

-Jonathan R.


Raphael Coeffic wrote:
> Hello,
> 
> a new internet draft has been published concerning the relay attack on 
> digest authentication and SIP. The attack itself has been first 
> disclosed 2 years ago by the maydnes team from the french INRIA. Until 
> now, no document has been pushlished that documents the attack and 
> provides guidance to SIP operators or handset manufacturers.
> 
> http://tools.ietf.org/html/draft-state-sip-relay-attack-00
> 
> The appropriate mitigations of problem resolutions are still not 100% 
> clear. We hope that this draft can help start a discussion on how to 
> best resolve this problem.
> 
> 
> Regards,
> 
> Raphael Coeffic.
> (on behalf of all the authors of this draft)
> 
> --------------------------------------------------------------------------------------------------- 
> 
> 
> Filename:       draft-state-sip-relay-attack
> Version:       00
> Staging URL:       
> http://www3.ietf.org/proceedings/staging/draft-state-sip-relay-attack-00.txt 
> 
> Title:           SIP digest authentication relay attack
> Creation_date:       2009-03-02
> WG ID:           Indvidual Submission
> Number_of_pages: 18
> Abstract:
> The Session Initiation Protocol (SIP [RFC3261]) provides a mechanism
> for creating, modifying, and terminating sessions with one or more
> participants.  This document describes a vulnerability of SIP
> combined with HTTP Digest Access Authentication [RFC2617] through
> which an attacker can leverage the victim's credentials to send
> authenticated requests on his behalf.  This attack is different from
> the man-in-the-middle (MITM) attack and does not require any
> eavesdropping, DNS or IP spoofing.
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
Cisco Fellow                                   Iselin, NJ 08830
Cisco, Voice Technology Group
jdrosen@cisco.com
http://www.jdrosen.net                         PHONE: (408) 902-3084
http://www.cisco.com

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Subject: [Sip] comments on draft-zourzouvillys-sip-via-cookie-02
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Theo,

Thanks for writing this up.

A few comments.

Firstly, it seems that the easiest solution is just 'use tcp'. This is 
one of many reasons (and growing) why we need to be pushing 
implementations to tcp.

Secondly, if you want to do this for UDP, you could instead use 
anonymous authentication. Then, the existing nonce takes on the role of 
a cookie. I suspect that anonymous authentication is not widely 
implemented, but neither is a new protocol. That said, I think the right 
answer still is 'use tcp'.

Finally, the media portion of this attack, as you point out, is far more 
disruptive. That attack does not require spoofing even; just listing the 
IP address/port of the target in the SDP of the INVITE. We were calling 
this the 'voice hammer' attack, originally documented here:

http://tools.ietf.org/html/draft-rosenberg-mmusic-rtp-denialofservice-00

and described in Section 18.5.1 of ICE. The via cookie mechanism you 
propose, SIP-over-TCP, or anonymous authentication, none of them fix 
this attack. AFAIK, ICE is the only remedy.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
Cisco Fellow                                   Iselin, NJ 08830
Cisco, Voice Technology Group
jdrosen@cisco.com
http://www.jdrosen.net                         PHONE: (408) 902-3084
http://www.cisco.com

From theo@crazygreek.co.uk  Sat Mar 14 07:05:26 2009
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Hi Jonathan,

Thank you very much for your comments.

On Sat, Mar 14, 2009 at 1:38 PM, Jonathan Rosenberg <jdrosen@cisco.com> wrote:

> Firstly, it seems that the easiest solution is just 'use tcp'. This is one
> of many reasons (and growing) why we need to be pushing implementations to
> tcp.

absolutely, or even better, [D]TLS.

problem is, in reality, UDP is not going away overnight.  we're far
more likely to get vendors to add a mechanism in that doesn't cost too
much in terms of implementation than expect everyone stops using UDP.

> Secondly, if you want to do this for UDP, you could instead use anonymous
> authentication. Then, the existing nonce takes on the role of a cookie. I
> suspect that anonymous authentication is not widely implemented, but neither
> is a new protocol. That said, I think the right answer still is 'use tcp'.

I did consider that.  The problem then is that it requires end-to-end
support, where as this is hop-by-hop, and as such means it doesn't
have HERFP problems or require every endpoint to upgrade - only the
hops that talk to public servers.   Via cookies also work over
multiple hops, where as anonymous would require a request-per-hop,
each request getting one hop further.  This could be very expensive in
both network load and call setup time.

> Finally, the media portion of this attack, as you point out, is far more
> disruptive. That attack does not require spoofing even; just listing the IP
> address/port of the target in the SDP of the INVITE. We were calling this
> the 'voice hammer' attack, originally documented here:
>
> http://tools.ietf.org/html/draft-rosenberg-mmusic-rtp-denialofservice-00

excellent.  i'd not come across that before.  i'll have a read tonight.

> and described in Section 18.5.1 of ICE. The via cookie mechanism you
> propose, SIP-over-TCP, or anonymous authentication, none of them fix this
> attack.

indeed - i wanted to solve the signalling side before i moved onto the
(albeit more serious) media problem

> AFAIK, ICE is the only remedy.

Yes, i came to the same conclusion; it's a shame that an ICE lite
implementation wouldn't itself fix it. (could it? :-))

There is also the problem of targeting a dialog at the victim (think
SUBSCRIBE with Contact of the victim), which is the 3rd issue.

 ~ Theo

From mary.barnes@nortel.com  Sat Mar 14 07:43:59 2009
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Thread-Topic: [Sip] Issue 3 - two tags (was RE:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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References: <899F1518-6872-442E-AD3E-D3F343A605DC@ntt-at.com><9ae56b1e0903121425j760fc6aeyf181bca65f803fe4@mail.gmail.com><0D5F89FAC29E2C41B98A6A762007F5D001A281EF@GBNTHT12009MSX.gb002.siemens.net><9ae56b1e0903130733k135db506lf3a134e1406d44d0@mail.gmail.com><0D5F89FAC29E2C41B98A6A762007F5D001A28210@GBNTHT12009MSX.gb002.siemens.net><9ae56b1e0903130751i5c9718abwbd44a6371130d258@mail.gmail.com><0D5F89FAC29E2C41B98A6A762007F5D001A2832E@GBNTHT12009MSX.gb002.siemens.net><1ECE0EB50388174790F9694F77522CCF1CDCAC9E@zrc2hxm0.corp.nortel.com><9ae56b1e0903131417q75a0f630u70df2929324cef7d@mail.gmail.com><1ECE0EB50388174790F9694F77522CCF1CDCB21A@zrc2hxm0.corp.nortel.com> <9ae56b1e0903140054l467fcd8eu50676ea31c914734@mail.gmail.com>
From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Hans Erik van Elburg" <ietf.hanserik@gmail.com>, <sip@ietf.org>
Subject: Re: [Sip] Issue 3 - two tags (was RE:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Correct, you don't know anything about the URI that overwrites the
request-URI.   BUT, you know the type of URI that was overwritten (with
respect to how it is used at the incoming Proxy). You know that the
request URI that was overwritten was one of the types we've been
discussing - i.e., a URI that can be used for a service Lookup in a
Registrar, whether that URI is mapped, configured, etc. or whether that
URI is just for a proxy hop - i.e., it's not a URI for which the proxy
is responsible. And, you are using that tag against the URI that is
being overwritten.
=20
Mary.=20

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Hans Erik van Elburg
Sent: Saturday, March 14, 2009 2:55 AM
To: sip@ietf.org List
Subject: Re: [Sip] Issue 3 - two tags (was
RE:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


I just realised that there is an additional problem that makes it even
harder to do any distinction.

A UA may register as a contact an AOR of another user or resource.
Effectively this causes the request to be forwarded to the other user by
overwriting the Request-URI.

So the fact that the Request-URI was overwritten due to a location
lookup, doesn't tell you a bit about the nature of the rewrite.

/Hans Erik van Elburg



On Fri, Mar 13, 2009 at 10:53 PM, Mary Barnes <mary.barnes@nortel.com>
wrote:


	Responses below [MB].
=09
	________________________________
=09

	From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
=09
	Sent: Friday, March 13, 2009 4:17 PM
=09
	To: Barnes, Mary (RICH2:AR00)
=09
	Cc: Elwell, John; sip@ietf.org
=09
	Subject: Re: [Sip] Issue 3 - two tags (was RE:
	I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
=09
=09
=09
	I guess you mean that you agree with John that such rule can not
be
	reliably construed.
=09
	[MB] Which rule?  I fully agree we can discern the 3 cases as I
	explained below. Per 16.5 (Determining Request Targets) in RFC
3261,
	these are unique and recognized situations. The reason I refer
to loose
	routing is that it's my understanding it is far more commonly
used and
	RFC 3261 doesn't even describe strict routing - it focuses on
loose
	routing and strict routing support is for 2543 backwards
compatibility
	as I understood. But, actually it makes no difference to
History-Info
	processing.  The important thing really, is that you MUST tag
the old
	entry as you are doing the processing per 16.5 as that
information is
	lost by the time you get around to adding the next History-Info
header
	in section 16.6 ( Request Forwarding). I think one thing that
might be
	being missed in this thread is that I have been very careful to
rely
	explicitly on what is described in 16.5 - that a must read to
understand
	what I am saying - again there is a lack of clarity in that
section and
	we can suggest clarifications once we all can finally agree
appropriate
	terms. If you'd like me to pull the explicit text that supports
	everything I've been saying, I can do so, but it's somewhat a
waste of
	time. [/MB]
=09

	I dont think I have seen convincing proof that such is the case.
=09
	[MB] And, I'm not saying that you can't construct such a rule -
indeed I
	am saying that you certainly should be able to per RFC 3261
section
	16.5.  Can you point out what text below that suggests I am
saying that?
	[/MB]
=09

	Are you also saying that we may only consider loose routing
being used?
=09
	[MB] Nope. [/MB]
=09

	Further for "configured mappings" for which we never saw a
proper
	definition. But assuming that this is a proxy that can be
configured to
	do a Request-URI  rewrite based on some configured setting.
=09
	[MB] The concept of "configured mappings" is described in RFC
3261 -
	perhaps not with the level of granularity or specificity that
some might
	desire, but it's there - it's initially described as an abstract
	location service (1st Paragraph section 16.5).  And, then later
it
	specifies that a proxy can determine "targets" (3261 term not
mine) just
	about any ole way it wants, so I don't know that we need to
define a new
	term for "configured mappings" per se - there is no limitation
as to the
	mechanism one can use to determine the request-URI for the
outgoing
	request (once it is determined that the Proxy is ALLOWED  to
overwrite
	the request-URI - the right domain, etc.),  This is a
fundamental SIP
	concept that many products use to give a (human) user
flexibility in
	configuring where an incoming call might terminate. So, I see no
issue
	at all is specifying two tags to capture this rewrite as it's
already in
	section 16.5 that this is information that a Proxy should be
able to
	derive. [/MB]
=09

	In that case there are two variants of such mapping:
	1. A mapping that changes the identity of the target to be
reached.
	(This is what call forwarding or diversion does normally.)
	2. A mapping that only changes  the next hop route taken towards
the
	target identity that has not been changed.
=09
	For the UAS it only matters that the home domain does the
tagging
	consistently and correctly if it wants to rely on such
information.
=09
	[MB] I don't know that I agree it's the home domain - it's a
proxy that
	has responsibility for the domain in the request-URI that is
able to do
	the service lookup or using any other sort of mapping to
determine the
	new URI for the request-URI in the outgoing request.  I don't
know that
	we want to restrict that to the "home" domain. If we restrict it
to the
	home domain, then that implies that we are only tagging the
entries that
	are determined based on service lookup in a Registrar (per RFC
3261's
	definition of a "home domain").  [/MB]
=09

	Otherwise we would have to start adding information about the
domain
	that added a specific hi-entry, if one wants to be sure who
added the
	particular hi-entry.
=09
	[MB] I'm not sure why this would be needed and I wasn't
espousing that
	it would - we know by definition that the domain that mucked
with a URI
	is the domain associated with the URI that was overwritten/lost,
etc.
	[/MB]
=09

	/Hans Erik van Elburg
=09
=09
=09
		---remainder of theread snipped by Mary. --------
=09



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To: "Hans Erik van Elburg" <ietf.hanserik@gmail.com>
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thread-topic: [Sip] Issue 3 - two tags (was RE:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Subject: Re: [Sip] Issue 3 - two tags (was RE:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Yeah, I tought about that too. Not sure how to handle it.

On Mar 14, 2009, at 0:55, "Hans Erik van Elburg" <ietf.hanserik@gmail.com 
 > wrote:

> I just realised that there is an additional problem that makes it  
> even harder to do any distinction.
>
> A UA may register as a contact an AOR of another user or resource.  
> Effectively this causes the request to be forwarded to the other  
> user by overwriting the Request-URI.
>
> So the fact that the Request-URI was overwritten due to a location  
> lookup, doesn't tell you a bit about the nature of the rewrite.
>
> /Hans Erik van Elburg
>
>
> On Fri, Mar 13, 2009 at 10:53 PM, Mary Barnes  
> <mary.barnes@nortel.com> wrote:
> Responses below [MB].
>
> ________________________________
>
> From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
> Sent: Friday, March 13, 2009 4:17 PM
> To: Barnes, Mary (RICH2:AR00)
> Cc: Elwell, John; sip@ietf.org
> Subject: Re: [Sip] Issue 3 - two tags (was RE:
> I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>
>
> I guess you mean that you agree with John that such rule can not be
> reliably construed.
> [MB] Which rule?  I fully agree we can discern the 3 cases as I
> explained below. Per 16.5 (Determining Request Targets) in RFC 3261,
> these are unique and recognized situations. The reason I refer to  
> loose
> routing is that it's my understanding it is far more commonly used and
> RFC 3261 doesn't even describe strict routing - it focuses on loose
> routing and strict routing support is for 2543 backwards compatibility
> as I understood. But, actually it makes no difference to History-Info
> processing.  The important thing really, is that you MUST tag the old
> entry as you are doing the processing per 16.5 as that information is
> lost by the time you get around to adding the next History-Info header
> in section 16.6 ( Request Forwarding). I think one thing that might be
> being missed in this thread is that I have been very careful to rely
> explicitly on what is described in 16.5 - that a must read to  
> understand
> what I am saying - again there is a lack of clarity in that section  
> and
> we can suggest clarifications once we all can finally agree  
> appropriate
> terms. If you'd like me to pull the explicit text that supports
> everything I've been saying, I can do so, but it's somewhat a waste of
> time. [/MB]
>
> I dont think I have seen convincing proof that such is the case.
> [MB] And, I'm not saying that you can't construct such a rule -  
> indeed I
> am saying that you certainly should be able to per RFC 3261 section
> 16.5.  Can you point out what text below that suggests I am saying  
> that?
> [/MB]
>
> Are you also saying that we may only consider loose routing being  
> used?
> [MB] Nope. [/MB]
>
> Further for "configured mappings" for which we never saw a proper
> definition. But assuming that this is a proxy that can be configured  
> to
> do a Request-URI  rewrite based on some configured setting.
> [MB] The concept of "configured mappings" is described in RFC 3261 -
> perhaps not with the level of granularity or specificity that some  
> might
> desire, but it's there - it's initially described as an abstract
> location service (1st Paragraph section 16.5).  And, then later it
> specifies that a proxy can determine "targets" (3261 term not mine)  
> just
> about any ole way it wants, so I don't know that we need to define a  
> new
> term for "configured mappings" per se - there is no limitation as to  
> the
> mechanism one can use to determine the request-URI for the outgoing
> request (once it is determined that the Proxy is ALLOWED  to overwrite
> the request-URI - the right domain, etc.),  This is a fundamental SIP
> concept that many products use to give a (human) user flexibility in
> configuring where an incoming call might terminate. So, I see no issue
> at all is specifying two tags to capture this rewrite as it's  
> already in
> section 16.5 that this is information that a Proxy should be able to
> derive. [/MB]
>
> In that case there are two variants of such mapping:
> 1. A mapping that changes the identity of the target to be reached.
> (This is what call forwarding or diversion does normally.)
> 2. A mapping that only changes  the next hop route taken towards the
> target identity that has not been changed.
>
> For the UAS it only matters that the home domain does the tagging
> consistently and correctly if it wants to rely on such information.
> [MB] I don't know that I agree it's the home domain - it's a proxy  
> that
> has responsibility for the domain in the request-URI that is able to  
> do
> the service lookup or using any other sort of mapping to determine the
> new URI for the request-URI in the outgoing request.  I don't know  
> that
> we want to restrict that to the "home" domain. If we restrict it to  
> the
> home domain, then that implies that we are only tagging the entries  
> that
> are determined based on service lookup in a Registrar (per RFC 3261's
> definition of a "home domain").  [/MB]
>
> Otherwise we would have to start adding information about the domain
> that added a specific hi-entry, if one wants to be sure who added the
> particular hi-entry.
> [MB] I'm not sure why this would be needed and I wasn't espousing that
> it would - we know by definition that the domain that mucked with a  
> URI
> is the domain associated with the URI that was overwritten/lost, etc.
> [/MB]
>
> /Hans Erik van Elburg
>
>
>
> On Fri, Mar 13, 2009 at 7:43 PM, Mary Barnes <mary.barnes@nortel.com>
> wrote:
>
>
>        Hi John,
>
>        Thanks for all your feedback thus far on this topic. I have a
> couple
>        comments below [MB].
>
>        Mary.
>
>
>        -----Original Message-----
>        From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> Behalf Of
>
>        Elwell, John
>        Sent: Friday, March 13, 2009 11:31 AM
>        To: Hans Erik van Elburg
>
>        Cc: sip@ietf.org
>        Subject: Re: [Sip] Issue 3 - two tags (was RE:
>        I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>
>
>
>
>        > -----Original Message-----
>        > From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
>        > Sent: 13 March 2009 14:52
>        > To: Elwell, John
>        > Cc: sip@ietf.org
>        > Subject: Re: [Sip] Issue 3 - two tags (was RE:
>        > I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
>        >
>        > It knows when it rewrites the Request-URI due to a location
> lookup -->
>
>        > tag "aor".
>        > It knows (is configured to) when it rewrites the Request-URI
> with a
>        > value that leads the request to target with a different
> identity then
>        > the one that has been replaced --> tag new entry "istarget".
>        > It knows when it rewrites the Request-URI with a value and
> this leads
>        > to the first History-Info entry being recorded --> tag
> "istarget".
>        [JRE] It does not know that this is the first target, because
> upstream
>        nodes may have retargeted without inserting H-I. In any case,
> the To
>        header field URI is the first target, so we don't need that
> anyway.
>
>        [MB] John is correct. There is no rule about tagging the first
>        History-Info entry. It may be tagged if the next node changes
> the
>        request-URI via a service lookup or some other manner such as
>        configuration.[/MB]
>
>
>        I can accept that a proxy might know, having done a LS look-up,
> whether
>        the new URI is a registered contact or some other URI, and on
> that basis
>        it could choose whether or not to insert a single tag. I  
> thought
> that
>        was the basis on which we agreed to go ahead with this updated
> to RFC
>        4424, rather than following the loose route draft. It is not
> clear to me
>        that a proxy would have enough information to make a 3-way
> choice.
>
>        [MB] My understanding is that there are just two ways in which
> the
>        request-URI changes if you are using loose routing. It's either
> changed
>        due to a lookup in a location server, whose information was
> populated by
>        the UA during registration, OR it is changed due to
> configuration data
>        (either system, user, etc.). In both cases, obviously, the  
> proxy
> that is
>        doing this is responsible for the domain of the request-URI in
> the
>        incoming request. If it were not the case, then all the proxy
> can do is
>        forward the outgoing request to the next hop. So, I think that
> may be
>        where the concept of a 3rd choice comes in. But, as far as
> tagging, we
>        (as I understand it) at least were discussing the potential of
> two tags
>        only - i.e., further differentiating the cases that are
> currently all
>        captured by the "retarget" as defined in 4244bis, which is the
> same
>        mechanism that is specified in the target-uri document EXCEPT
> that
>        document was only capturing the service lookups, which we
> believe is
>        incorrect. [/MB]
>
>
>        John
>
>
>        >
> ---remainder of theread snipped by Mary. --------
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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<html><body bgcolor="#FFFFFF"><div>Yeah, I tought about that too. Not sure how to handle it.<br><br>On Mar 14, 2009, at 0:55, "Hans Erik van Elburg" &lt;<a href="mailto:ietf.hanserik@gmail.com">ietf.hanserik@gmail.com</a>> wrote:<br><br></div><div></div><blockquote type="cite"><div>I just realised that there is an additional problem that makes it even harder to do any distinction.<br><br>A UA may register as a contact an AOR of another user or resource. Effectively this causes the request to be forwarded to the other user by overwriting the Request-URI.<br>
<br>So the fact that the Request-URI was overwritten due to a location lookup, doesn't tell you a bit about the nature of the rewrite.<br><br clear="all">/Hans Erik van Elburg<br>
<br><br><div class="gmail_quote">On Fri, Mar 13, 2009 at 10:53 PM, Mary Barnes <span dir="ltr">&lt;<a href="mailto:mary.barnes@nortel.com"><a href="mailto:mary.barnes@nortel.com">mary.barnes@nortel.com</a></a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Responses below [MB].<br>
<br>
________________________________<br>
<div class="im"><br>
From: Hans Erik van Elburg [mailto:<a href="mailto:ietf.hanserik@gmail.com"><a href="mailto:ietf.hanserik@gmail.com">ietf.hanserik@gmail.com</a></a>]<br>
</div>Sent: Friday, March 13, 2009 4:17 PM<br>
<div class="im">To: Barnes, Mary (RICH2:AR00)<br>
</div>Cc: Elwell, John; <a href="mailto:sip@ietf.org"><a href="mailto:sip@ietf.org">sip@ietf.org</a></a><br>
<div class="im">Subject: Re: [Sip] Issue 3 - two tags (was RE:<br>
I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br>
<br>
<br>
</div><div class="im">I guess you mean that you agree with John that such rule can not be<br>
reliably construed.<br>
</div>[MB] Which rule? &nbsp;I fully agree we can discern the 3 cases as I<br>
explained below. Per 16.5 (Determining Request Targets) in RFC 3261,<br>
these are unique and recognized situations. The reason I refer to loose<br>
routing is that it's my understanding it is far more commonly used and<br>
RFC 3261 doesn't even describe strict routing - it focuses on loose<br>
routing and strict routing support is for 2543 backwards compatibility<br>
as I understood. But, actually it makes no difference to History-Info<br>
processing. &nbsp;The important thing really, is that you MUST tag the old<br>
entry as you are doing the processing per 16.5 as that information is<br>
lost by the time you get around to adding the next History-Info header<br>
in section 16.6 ( Request Forwarding). I think one thing that might be<br>
being missed in this thread is that I have been very careful to rely<br>
explicitly on what is described in 16.5 - that a must read to understand<br>
what I am saying - again there is a lack of clarity in that section and<br>
we can suggest clarifications once we all can finally agree appropriate<br>
terms. If you'd like me to pull the explicit text that supports<br>
everything I've been saying, I can do so, but it's somewhat a waste of<br>
time. [/MB]<br>
<div class="im"><br>
I dont think I have seen convincing proof that such is the case.<br>
</div>[MB] And, I'm not saying that you can't construct such a rule - indeed I<br>
am saying that you certainly should be able to per RFC 3261 section<br>
16.5. &nbsp;Can you point out what text below that suggests I am saying that?<br>
[/MB]<br>
<div class="im"><br>
Are you also saying that we may only consider loose routing being used?<br>
</div>[MB] Nope. [/MB]<br>
<div class="im"><br>
Further for "configured mappings" for which we never saw a proper<br>
definition. But assuming that this is a proxy that can be configured to<br>
do a Request-URI &nbsp;rewrite based on some configured setting.<br>
</div>[MB] The concept of "configured mappings" is described in RFC 3261 -<br>
perhaps not with the level of granularity or specificity that some might<br>
desire, but it's there - it's initially described as an abstract<br>
location service (1st Paragraph section 16.5). &nbsp;And, then later it<br>
specifies that a proxy can determine "targets" (3261 term not mine) just<br>
about any ole way it wants, so I don't know that we need to define a new<br>
term for "configured mappings" per se - there is no limitation as to the<br>
mechanism one can use to determine the request-URI for the outgoing<br>
request (once it is determined that the Proxy is ALLOWED &nbsp;to overwrite<br>
the request-URI - the right domain, etc.), &nbsp;This is a fundamental SIP<br>
concept that many products use to give a (human) user flexibility in<br>
configuring where an incoming call might terminate. So, I see no issue<br>
at all is specifying two tags to capture this rewrite as it's already in<br>
section 16.5 that this is information that a Proxy should be able to<br>
derive. [/MB]<br>
<div class="im"><br>
In that case there are two variants of such mapping:<br>
1. A mapping that changes the identity of the target to be reached.<br>
(This is what call forwarding or diversion does normally.)<br>
2. A mapping that only changes &nbsp;the next hop route taken towards the<br>
target identity that has not been changed.<br>
<br>
For the UAS it only matters that the home domain does the tagging<br>
consistently and correctly if it wants to rely on such information.<br>
</div>[MB] I don't know that I agree it's the home domain - it's a proxy that<br>
has responsibility for the domain in the request-URI that is able to do<br>
the service lookup or using any other sort of mapping to determine the<br>
new URI for the request-URI in the outgoing request. &nbsp;I don't know that<br>
we want to restrict that to the "home" domain. If we restrict it to the<br>
home domain, then that implies that we are only tagging the entries that<br>
are determined based on service lookup in a Registrar (per RFC 3261's<br>
definition of a "home domain"). &nbsp;[/MB]<br>
<div class="im"><br>
Otherwise we would have to start adding information about the domain<br>
that added a specific hi-entry, if one wants to be sure who added the<br>
particular hi-entry.<br>
</div>[MB] I'm not sure why this would be needed and I wasn't espousing that<br>
it would - we know by definition that the domain that mucked with a URI<br>
is the domain associated with the URI that was overwritten/lost, etc.<br>
[/MB]<br>
<div><div></div><div class="h5"><br>
/Hans Erik van Elburg<br>
<br>
<br>
<br>
On Fri, Mar 13, 2009 at 7:43 PM, Mary Barnes &lt;<a href="mailto:mary.barnes@nortel.com"><a href="mailto:mary.barnes@nortel.com">mary.barnes@nortel.com</a></a>><br>
wrote:<br>
<br>
<br>
 &nbsp; &nbsp; &nbsp; &nbsp;Hi John,<br>
<br>
 &nbsp; &nbsp; &nbsp; &nbsp;Thanks for all your feedback thus far on this topic. I have a<br>
couple<br>
 &nbsp; &nbsp; &nbsp; &nbsp;comments below [MB].<br>
<br>
 &nbsp; &nbsp; &nbsp; &nbsp;Mary.<br>
<br>
<br>
 &nbsp; &nbsp; &nbsp; &nbsp;-----Original Message-----<br>
 &nbsp; &nbsp; &nbsp; &nbsp;From: <a href="mailto:sip-bounces@ietf.org"><a href="mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a></a> [mailto:<a href="mailto:sip-bounces@ietf.org"><a href="mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a></a>] On<br>
Behalf Of<br>
<br>
 &nbsp; &nbsp; &nbsp; &nbsp;Elwell, John<br>
 &nbsp; &nbsp; &nbsp; &nbsp;Sent: Friday, March 13, 2009 11:31 AM<br>
 &nbsp; &nbsp; &nbsp; &nbsp;To: Hans Erik van Elburg<br>
<br>
 &nbsp; &nbsp; &nbsp; &nbsp;Cc: <a href="mailto:sip@ietf.org"><a href="mailto:sip@ietf.org">sip@ietf.org</a></a><br>
 &nbsp; &nbsp; &nbsp; &nbsp;Subject: Re: [Sip] Issue 3 - two tags (was RE:<br>
 &nbsp; &nbsp; &nbsp; &nbsp;I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br>
<br>
<br>
<br>
<br>
 &nbsp; &nbsp; &nbsp; &nbsp;> -----Original Message-----<br>
 &nbsp; &nbsp; &nbsp; &nbsp;> From: Hans Erik van Elburg [mailto:<a href="mailto:ietf.hanserik@gmail.com"><a href="mailto:ietf.hanserik@gmail.com">ietf.hanserik@gmail.com</a></a>]<br>
 &nbsp; &nbsp; &nbsp; &nbsp;> Sent: 13 March 2009 14:52<br>
 &nbsp; &nbsp; &nbsp; &nbsp;> To: Elwell, John<br>
 &nbsp; &nbsp; &nbsp; &nbsp;> Cc: <a href="mailto:sip@ietf.org"><a href="mailto:sip@ietf.org">sip@ietf.org</a></a><br>
 &nbsp; &nbsp; &nbsp; &nbsp;> Subject: Re: [Sip] Issue 3 - two tags (was RE:<br>
 &nbsp; &nbsp; &nbsp; &nbsp;> I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt<br>
 &nbsp; &nbsp; &nbsp; &nbsp;><br>
 &nbsp; &nbsp; &nbsp; &nbsp;> It knows when it rewrites the Request-URI due to a location<br>
lookup --><br>
<br>
 &nbsp; &nbsp; &nbsp; &nbsp;> tag "aor".<br>
 &nbsp; &nbsp; &nbsp; &nbsp;> It knows (is configured to) when it rewrites the Request-URI<br>
with a<br>
 &nbsp; &nbsp; &nbsp; &nbsp;> value that leads the request to target with a different<br>
identity then<br>
 &nbsp; &nbsp; &nbsp; &nbsp;> the one that has been replaced --> tag new entry "istarget".<br>
 &nbsp; &nbsp; &nbsp; &nbsp;> It knows when it rewrites the Request-URI with a value and<br>
this leads<br>
 &nbsp; &nbsp; &nbsp; &nbsp;> to the first History-Info entry being recorded --> tag<br>
"istarget".<br>
 &nbsp; &nbsp; &nbsp; &nbsp;[JRE] It does not know that this is the first target, because<br>
upstream<br>
 &nbsp; &nbsp; &nbsp; &nbsp;nodes may have retargeted without inserting H-I. In any case,<br>
the To<br>
 &nbsp; &nbsp; &nbsp; &nbsp;header field URI is the first target, so we don't need that<br>
anyway.<br>
<br>
 &nbsp; &nbsp; &nbsp; &nbsp;[MB] John is correct. There is no rule about tagging the first<br>
 &nbsp; &nbsp; &nbsp; &nbsp;History-Info entry. It may be tagged if the next node changes<br>
the<br>
 &nbsp; &nbsp; &nbsp; &nbsp;request-URI via a service lookup or some other manner such as<br>
 &nbsp; &nbsp; &nbsp; &nbsp;configuration.[/MB]<br>
<br>
<br>
 &nbsp; &nbsp; &nbsp; &nbsp;I can accept that a proxy might know, having done a LS look-up,<br>
whether<br>
 &nbsp; &nbsp; &nbsp; &nbsp;the new URI is a registered contact or some other URI, and on<br>
that basis<br>
 &nbsp; &nbsp; &nbsp; &nbsp;it could choose whether or not to insert a single tag. I thought<br>
that<br>
 &nbsp; &nbsp; &nbsp; &nbsp;was the basis on which we agreed to go ahead with this updated<br>
to RFC<br>
 &nbsp; &nbsp; &nbsp; &nbsp;4424, rather than following the loose route draft. It is not<br>
clear to me<br>
 &nbsp; &nbsp; &nbsp; &nbsp;that a proxy would have enough information to make a 3-way<br>
choice.<br>
<br>
 &nbsp; &nbsp; &nbsp; &nbsp;[MB] My understanding is that there are just two ways in which<br>
the<br>
 &nbsp; &nbsp; &nbsp; &nbsp;request-URI changes if you are using loose routing. It's either<br>
changed<br>
 &nbsp; &nbsp; &nbsp; &nbsp;due to a lookup in a location server, whose information was<br>
populated by<br>
 &nbsp; &nbsp; &nbsp; &nbsp;the UA during registration, OR it is changed due to<br>
configuration data<br>
 &nbsp; &nbsp; &nbsp; &nbsp;(either system, user, etc.). In both cases, obviously, the proxy<br>
that is<br>
 &nbsp; &nbsp; &nbsp; &nbsp;doing this is responsible for the domain of the request-URI in<br>
the<br>
 &nbsp; &nbsp; &nbsp; &nbsp;incoming request. If it were not the case, then all the proxy<br>
can do is<br>
 &nbsp; &nbsp; &nbsp; &nbsp;forward the outgoing request to the next hop. So, I think that<br>
may be<br>
 &nbsp; &nbsp; &nbsp; &nbsp;where the concept of a 3rd choice comes in. But, as far as<br>
tagging, we<br>
 &nbsp; &nbsp; &nbsp; &nbsp;(as I understand it) at least were discussing the potential of<br>
two tags<br>
 &nbsp; &nbsp; &nbsp; &nbsp;only - i.e., further differentiating the cases that are<br>
currently all<br>
 &nbsp; &nbsp; &nbsp; &nbsp;captured by the "retarget" as defined in 4244bis, which is the<br>
same<br>
 &nbsp; &nbsp; &nbsp; &nbsp;mechanism that is specified in the target-uri document EXCEPT<br>
that<br>
 &nbsp; &nbsp; &nbsp; &nbsp;document was only capturing the service lookups, which we<br>
believe is<br>
 &nbsp; &nbsp; &nbsp; &nbsp;incorrect. [/MB]<br>
<br>
<br>
 &nbsp; &nbsp; &nbsp; &nbsp;John<br>
<br>
<br>
 &nbsp; &nbsp; &nbsp; &nbsp;><br>
</div></div>---remainder of theread snipped by Mary. --------<br>
</blockquote></div><br>
</div></blockquote><blockquote type="cite"><div><span>_______________________________________________</span><br><span>Sip mailing list &nbsp;<a href="https://www.ietf.org/mailman/listinfo/sip"><a href="https://www.ietf.org/mailman/listinfo/sip">https://www.ietf.org/mailman/listinfo/sip</a></a></span><br><span>This list is for NEW development of the core SIP Protocol</span><br><span>Use <a href="mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.columbia.edu</a> for questions on current sip</span><br><span>Use <a href="mailto:sipping@ietf.org">sipping@ietf.org</a> for new developments on the application of sip</span></div></blockquote></body></html>
--Apple-Mail-1-1038211793--

From christer.holmberg@ericsson.com  Sat Mar 14 16:54:28 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Mary Barnes" <mary.barnes@nortel.com>, "Hans Erik van Elburg" <ietf.hanserik@gmail.com>, <sip@ietf.org>
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Subject: Re: [Sip] Issue 3 - two tags (wasRE:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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When someone is calling the AOR, it will be replaced with the registered
contact. Does it matter is someone else registered the contact/AOR? The
call is not for that someone.

Regards,

Christer

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Mary Barnes
Sent: Saturday, March 14, 2009 4:47 PM
To: Hans Erik van Elburg; sip@ietf.org
Subject: Re: [Sip] Issue 3 - two tags
(wasRE:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt

Correct, you don't know anything about the URI that overwrites the
request-URI.   BUT, you know the type of URI that was overwritten (with
respect to how it is used at the incoming Proxy). You know that the
request URI that was overwritten was one of the types we've been
discussing - i.e., a URI that can be used for a service Lookup in a
Registrar, whether that URI is mapped, configured, etc. or whether that
URI is just for a proxy hop - i.e., it's not a URI for which the proxy
is responsible. And, you are using that tag against the URI that is
being overwritten.
=20
Mary.=20

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Hans Erik van Elburg
Sent: Saturday, March 14, 2009 2:55 AM
To: sip@ietf.org List
Subject: Re: [Sip] Issue 3 - two tags (was
RE:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt


I just realised that there is an additional problem that makes it even
harder to do any distinction.

A UA may register as a contact an AOR of another user or resource.
Effectively this causes the request to be forwarded to the other user by
overwriting the Request-URI.

So the fact that the Request-URI was overwritten due to a location
lookup, doesn't tell you a bit about the nature of the rewrite.

/Hans Erik van Elburg



On Fri, Mar 13, 2009 at 10:53 PM, Mary Barnes <mary.barnes@nortel.com>
wrote:


	Responses below [MB].
=09
	________________________________
=09

	From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
=09
	Sent: Friday, March 13, 2009 4:17 PM
=09
	To: Barnes, Mary (RICH2:AR00)
=09
	Cc: Elwell, John; sip@ietf.org
=09
	Subject: Re: [Sip] Issue 3 - two tags (was RE:
	I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
=09
=09
=09
	I guess you mean that you agree with John that such rule can not
be
	reliably construed.
=09
	[MB] Which rule?  I fully agree we can discern the 3 cases as I
	explained below. Per 16.5 (Determining Request Targets) in RFC
3261,
	these are unique and recognized situations. The reason I refer
to loose
	routing is that it's my understanding it is far more commonly
used and
	RFC 3261 doesn't even describe strict routing - it focuses on
loose
	routing and strict routing support is for 2543 backwards
compatibility
	as I understood. But, actually it makes no difference to
History-Info
	processing.  The important thing really, is that you MUST tag
the old
	entry as you are doing the processing per 16.5 as that
information is
	lost by the time you get around to adding the next History-Info
header
	in section 16.6 ( Request Forwarding). I think one thing that
might be
	being missed in this thread is that I have been very careful to
rely
	explicitly on what is described in 16.5 - that a must read to
understand
	what I am saying - again there is a lack of clarity in that
section and
	we can suggest clarifications once we all can finally agree
appropriate
	terms. If you'd like me to pull the explicit text that supports
	everything I've been saying, I can do so, but it's somewhat a
waste of
	time. [/MB]
=09

	I dont think I have seen convincing proof that such is the case.
=09
	[MB] And, I'm not saying that you can't construct such a rule -
indeed I
	am saying that you certainly should be able to per RFC 3261
section
	16.5.  Can you point out what text below that suggests I am
saying that?
	[/MB]
=09

	Are you also saying that we may only consider loose routing
being used?
=09
	[MB] Nope. [/MB]
=09

	Further for "configured mappings" for which we never saw a
proper
	definition. But assuming that this is a proxy that can be
configured to
	do a Request-URI  rewrite based on some configured setting.
=09
	[MB] The concept of "configured mappings" is described in RFC
3261 -
	perhaps not with the level of granularity or specificity that
some might
	desire, but it's there - it's initially described as an abstract
	location service (1st Paragraph section 16.5).  And, then later
it
	specifies that a proxy can determine "targets" (3261 term not
mine) just
	about any ole way it wants, so I don't know that we need to
define a new
	term for "configured mappings" per se - there is no limitation
as to the
	mechanism one can use to determine the request-URI for the
outgoing
	request (once it is determined that the Proxy is ALLOWED  to
overwrite
	the request-URI - the right domain, etc.),  This is a
fundamental SIP
	concept that many products use to give a (human) user
flexibility in
	configuring where an incoming call might terminate. So, I see no
issue
	at all is specifying two tags to capture this rewrite as it's
already in
	section 16.5 that this is information that a Proxy should be
able to
	derive. [/MB]
=09

	In that case there are two variants of such mapping:
	1. A mapping that changes the identity of the target to be
reached.
	(This is what call forwarding or diversion does normally.)
	2. A mapping that only changes  the next hop route taken towards
the
	target identity that has not been changed.
=09
	For the UAS it only matters that the home domain does the
tagging
	consistently and correctly if it wants to rely on such
information.
=09
	[MB] I don't know that I agree it's the home domain - it's a
proxy that
	has responsibility for the domain in the request-URI that is
able to do
	the service lookup or using any other sort of mapping to
determine the
	new URI for the request-URI in the outgoing request.  I don't
know that
	we want to restrict that to the "home" domain. If we restrict it
to the
	home domain, then that implies that we are only tagging the
entries that
	are determined based on service lookup in a Registrar (per RFC
3261's
	definition of a "home domain").  [/MB]
=09

	Otherwise we would have to start adding information about the
domain
	that added a specific hi-entry, if one wants to be sure who
added the
	particular hi-entry.
=09
	[MB] I'm not sure why this would be needed and I wasn't
espousing that
	it would - we know by definition that the domain that mucked
with a URI
	is the domain associated with the URI that was overwritten/lost,
etc.
	[/MB]
=09

	/Hans Erik van Elburg
=09
=09
=09
		---remainder of theread snipped by Mary. --------
=09


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Cc: sip@ietf.org
Subject: Re: [Sip] Issue 3 - two tags (wasRE:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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The point is that the registered contact could be an AOR of another user.

/Hans Erik

Christer Holmberg wrote:
> When someone is calling the AOR, it will be replaced with the registered
> contact. Does it matter is someone else registered the contact/AOR? The
> call is not for that someone.
>
> Regards,
>
> Christer
>
>   

From christer.holmberg@ericsson.com  Sun Mar 15 13:29:26 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Hans Erik van Elburg" <ietf.hanserik@gmail.com>
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Subject: Re: [Sip] Issue 3 - two tags (wasRE:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Hi,=20

>The point is that the registered contact could be an AOR of another
user.

Ok, now I understand. Yes, that is an interesting case, eventhough I
guess it's really not the correct usage of the registration procedure.

Regards,

Christer



/Hans Erik

Christer Holmberg wrote:
> When someone is calling the AOR, it will be replaced with the=20
> registered contact. Does it matter is someone else registered the=20
> contact/AOR? The call is not for that someone.
>
> Regards,
>
> Christer
>
>  =20

From dean.willis@softarmor.com  Sun Mar 15 18:03:19 2009
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Subject: Re: [Sip] Issue 3 - two tags	(wasRE:I-DACTION:draft-rosenberg-sip-target-uri-delivery-01.txt
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Christer Holmberg wrote:
> Hi, 
> 
>> The point is that the registered contact could be an AOR of another
> user.
> 
> Ok, now I understand. Yes, that is an interesting case, eventhough I
> guess it's really not the correct usage of the registration procedure.
> 

Any contact could be an AOR, not a final destination, in that any UA 
could also decide to become a proxy and send the call somewhere else. 
Such a determination might be made on a call-by-call basis.

In such case, two possibilities apply:

1) the transformation is a reroute, in that the identity of the target 
is not being changed, just the location at which that target is being 
sought.


2) The transformation is a retarget, in that the identity of the target 
is being sought after the transformation is different than the identity 
being sought before the transformation.

Of course, it is possible for a proxy making such a transformation to 
not know the identity of the target. In such a case, it is impossible 
for the proxy to inform the UAC of the transformation and thereby avoid 
an unanticipated respondent scenario. This is the world we live in for 
ALL proxy operations today, and it has its drawbacks.

--
Dean


From rco@iptel.org  Mon Mar 16 02:45:46 2009
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Jonathan Rosenberg wrote:
> Raphael and others,
>
> Thanks for putting this together.
>
> Considering for a moment figure 1, this attack is possible only if UAs 
> accept incoming requests from any place, and not just their proxy. In 
> practice, this is seldom allowed. Indeed, if it is allowed, there are 
> worse attacks than this which can be launched (free phone calls, spam 
> calling, etc.). Using the SIP recommended TLS between UA and proxy 
> also mitigates that.
>
That's right, and also the reason we called this the basic attack. Using 
TLS would be indeed a good choice, unfortunately, only few providers do 
offer it to their customers (but of course, only for the small panel I 
know of, and those I use).

> So really figure 2 is the interesting one. However, this attack 
> assumes that Alice has credentials on multiple systems. Again, in 
> practice, this is extremely uncommon. Certainly none of the existing 
> deployed enterprise or service provider consumer deployments are of 
> this nature.
I understand that this might not be a common case, as the common case is 
more focused on a vonage (or alike) customer, receiving a 
ready-configured phone that just has to be pluged into the residential 
gateway. However, a lot of SIP freaks like us do have a lot of 
configured accounts on the same phone, and not only for test purposes. 
For my part, my home phone is connected with 3 accounts: one default 
(PSTN), my iptel.org account, and another provider that own one of our 
PSTN numbers (only inbound).

I have seen one year ago an application for a residential gateway 
implementing a SIP least cost router. It was available as an unofficial 
upgrade for a very popular type of resential gateway. In the case, the 
configuration was at its minimum: all the required config was downloaded 
by the least cost router application. All you had to do was to create 
some accounts at some PSTN provider, and configure them into the box, 
which would do the rest.

I can believe that this kind of application could become more popular, 
just like they were in the PSTN.

>
>
> As such, I dont think this attack is likely in practice. However, in 
> theory it is possible. The essence of the attack is that the victim is 
> providing credentials to an unauthenticated server (since the attacker 
> is acting like a server, asking for credentials). In that way, as 
> others have pointed out, it is similar to baiting attacks that have 
> been previously documented. With SIP it is most easily remedied by a 
> rule which says, 'don't pass credentials for domain X to a server that 
> is not domain X'.
Which means that you exclude any relays in between. I think it also 
implies reverse DNS lookups, right?

> Server identity can be verified by normal server-only auth between a 
> client and its server, but even that is not needed. 
Right, mutual authentication seems to be the best way.

> A client will know which domain its proxy is representing, and once 
> connected, it only provides credentials for that domain.
>
What do you mean by "connected"? And why should a UA only provide 
credentials for one domain only?
Is that about TCP or SCTP connections? and providing only credentials 
for a particular domain on a particular connection?

-Raphael.

From drage@alcatel-lucent.com  Mon Mar 16 04:33:21 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: "sip@ietf.org" <sip@ietf.org>
Date: Mon, 16 Mar 2009 12:33:58 +0100
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Cc: "draft-ietf-sip-location-conveyance@tools.ietf.org" <draft-ietf-sip-location-conveyance@tools.ietf.org>
Subject: [Sip] WGLC for draft-ietf-sip-location-conveyance
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(As WG chair)

This is to initiate a WGLC for comments on draft-ietf-sip-location-conveyan=
ce

http://tools.ietf.org/id/draft-ietf-sip-location-conveyance-13.txt

as proposed standard.

This is a one week WGLC and therefore please supply your comments by close =
of Monday 23rd March 2009.

While we acknowledge this may interrupt your reading for the forthcoming IE=
TF meeting, please add this to your reading list and comment.

We will take steps to see that people have actually read this and are eithe=
r OK or not, so if you do so read it, please send an email to the chairs in=
dicating this, even if you have no comments. If necessary we will go for a =
further period of WGLC after the meeting, but to be honest, we'd rather avo=
id this and get this somewhat aged document out of the door.

Comments obviously to the list and editors, and it would help if you give t=
he proposed correction (or even some hint of it) along with the change.

regards

Keith=

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Raphael Coeffic wrote:

>> As such, I dont think this attack is likely in practice. However, in 
>> theory it is possible. The essence of the attack is that the victim is 
>> providing credentials to an unauthenticated server (since the attacker 
>> is acting like a server, asking for credentials). In that way, as 
>> others have pointed out, it is similar to baiting attacks that have 
>> been previously documented. With SIP it is most easily remedied by a 
>> rule which says, 'don't pass credentials for domain X to a server that 
>> is not domain X'.
> Which means that you exclude any relays in between. I think it also 
> implies reverse DNS lookups, right?

No - no reverse DNS.
And I am excluding cases where, you are connected (meaning, the domain 
you registered to) to one domain and making a call through that domain 
reach another domain which requires your credentials. I think that is 
very unlikely.

> 
>> Server identity can be verified by normal server-only auth between a 
>> client and its server, but even that is not needed. 
> Right, mutual authentication seems to be the best way.
> 
>> A client will know which domain its proxy is representing, and once 
>> connected, it only provides credentials for that domain.
>>
> What do you mean by "connected"? And why should a UA only provide 
> credentials for one domain only?

I'm saying, when a phone registers or makes a call, it does so by 
connecting to a SIP server, through a domain name or IP config or 
whatever. THat server will have an associated credential. For any 
request sent to that server, it should never provide a credential except 
the one associated with it.

Your attack is possible only when the client sends credentials for a 
domain, different than the one it is currently registered or placed the 
call through in the first place.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
Cisco Fellow                                   Iselin, NJ 08830
Cisco, Voice Technology Group
jdrosen@cisco.com
http://www.jdrosen.net                         PHONE: (408) 902-3084
http://www.cisco.com

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Subject: [Sip] New rev of Location Conveyance (to -13)
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SIP

I've revved Location Conveyance, which the chairs just announced a 1 
week WGLC for  it.

I would appreciate some reviews, even just giving this doc a thumbs 
up (to the list of course).

http://tools.ietf.org/id/draft-ietf-sip-location-conveyance-13.txt

The changes are slight, and only centered on one topic, which is the 
text surrounding the Geolocation header's ABNF wrt the placement of 
the "routing-allowed" header parameter. I made the text consistent with the

The diff is here

http://tools.ietf.org/wg/sip/draft-ietf-sip-location-conveyance/draft-ietf-sip-location-conveyance-13-from-12.diff.html

Thanks

James


From lists@ohlmeier.org  Mon Mar 16 15:00:19 2009
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>>> Server identity can be verified by normal server-only auth between a
>>> client and its server, but even that is not needed.
>> Right, mutual authentication seems to be the best way.
>>
>>> A client will know which domain its proxy is representing, and once
>>> connected, it only provides credentials for that domain.
>>>
>> What do you mean by "connected"? And why should a UA only provide
>> credentials for one domain only?
>
> I'm saying, when a phone registers or makes a call, it does so by
> connecting to a SIP server, through a domain name or IP config or
> whatever. THat server will have an associated credential. For any
> request sent to that server, it should never provide a credential except
> the one associated with it.
>
> Your attack is possible only when the client sends credentials for a
> domain, different than the one it is currently registered or placed the
> call through in the first place.

And the SIP server which relays the INVITE from the attacker to the 
victim has to drop (?) the challenge if it receives such from the 
attacker (at least if the realm is identical to the one used by the 
server itself).

This is basically the same description which I provided already earlier.

The only open "issue" is then relaying of 407/401 replies from 
downstream elements by the proxy. I agree with Jonathan that I also do 
not believe that this actually a real world scenario. But mutual 
authentication in the challenge might help to solve this issue if people 
insist of keeping this scenario.

Regards
   Nils

From marianne.mohali@orange-ftgroup.com  Mon Mar 16 15:11:10 2009
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Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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Hi Mary and Fran=E7ois,

I have a suggestion and few comments on the document.

** To make the History-Info header usefull for applications and services =
interactions, it could be interesting to be able to include in the =
hi-target-to-uri IARI and ICSC values as described in TS 24.229 (3GPP) =
wich ar coded as URN. An example of an ICSI for a 3GPP defined IMS =
communication service is: urn:urn-xxx:3gpp-service.ims.icsi.mmtel
Do you think, this kind of enhancement for Histroy-Info header could be =
added in the document ?

** Section 4.1: About Retarget (hi-target), it is indicated "A retarget =
parameter is not added for a hi-entry when it is first added in a =
History-
Info header, but rather is added when the retargeting actually occurs". =
It is indicated when the Retarget indication should be added but it =
could be 	good to add an information on where the parameter should be =
added (associated to the retargeting hi-targeted-to-uri). This is just =
to be clear on 	the History-Info header structure.=20
I have the same comment for the Reason header escaped in the =
History-Info. About that, there is an inconsistency between Reason =
header associated to 	the retargeting URI and cause-param parameter =
(RFC4458) associated to the retargeted-to URI. This should be corrected =
in RFC4458 but I don't know if 	it is possible.

** Section 4.5.1 and following: in the call flows there is a "retarget" =
parmeter inserted in the second INVITE by Proxy1 when sending the =
request to Proxy2. In my understanding, it should only be added (in the =
same place) by Proxy2 when retargeting the request to Bob. Did I miss =
something ?

** Privacy: I also think that it should be recommended in the document =
to use the Privacy header with the value "history" escaped in the =
hi-entries rather than directly in the INVITE request. =20


Best Regards,
Marianne

From dean.willis@softarmor.com  Mon Mar 16 15:56:22 2009
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Subject: [Sip] Publication requested for draft-ietf-sip-record-route-fix-06
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We have requested publication of this WG effort as a BCP. I'd like to  
thank Nils for volunteering to serve as the draft shepherd, and for  
preparing most of the writeup below.

--
Dean Willis, SIP WG co-chair.


Writeup follows:


The SIP working group would like to request publication of draft-ietf-
sip-record-route-fix-06. This is a product of the SIP working group and
is intended for Best Current Practice status.

Nils Ohlmeier <nils@ohlmeier.com> will serve as the draft shepherd,
supported by SIP working group chair Dean Willis.

A write-up, prepared by Nils, is attached.

[This template version is dated September 17, 2008.]

(for draft-ietf-sip-record-route-fix-06)

     (1.a) Who is the Document Shepherd for this document? Has the
           Document Shepherd personally reviewed this version of the
           document and, in particular, does he or she believe this
           version is ready for forwarding to the IESG for publication?

The Document Shepherd for this document is Nils Ohlmeier. I have  read
this version of the draft and believe it is ready for forwarding to
the IESG for publication.

     (1.b) Has the document had adequate review both from key WG members
           and from key non-WG members? Does the Document Shepherd have
           any concerns about the depth or breadth of the reviews that
           have been performed?

This document has had in-depth reviews from working group members with
no concerns.

     (1.c) Does the Document Shepherd have concerns that the document
           needs more review from a particular or broader perspective,
           e.g., security, operational complexity, someone familiar with
           AAA, internationalization or XML?

As the document describes a best current practice which is widely
deployed and the document got several reviews there are no concerns.

     (1.d) Does the Document Shepherd have any specific concerns or
           issues with this document that the Responsible Area Director
           and/or the IESG should be aware of? For example, perhaps he
           or she is uncomfortable with certain parts of the document,  
or
           has concerns whether there really is a need for it. In any
           event, if the WG has discussed those issues and has indicated
           that it still wishes to advance the document, detail those
           concerns here. Has an IPR disclosure related to this document
           been filed? If so, please include a reference to the
           disclosure and summarize the WG discussion and conclusion on
           this issue.

I believe the described technique should be documented and I have no
concerns with the document.
There have been no IPR disclosures for this document.

     (1.e) How solid is the WG consensus behind this document? Does it
           represent the strong concurrence of a few individuals, with
           others being silent, or does the WG as a whole understand and
           agree with it?

The WG consensus appears to be strong.

     (1.f) Has anyone threatened an appeal or otherwise indicated  
extreme
           discontent? If so, please summarise the areas of conflict in
           separate email messages to the Responsible Area Director. (It
           should be in a separate email because this questionnaire is
           entered into the ID Tracker.)

No one has expressed any discontent with this document.

     (1.g) Has the Document Shepherd personally verified that the
           document satisfies all ID nits? (See
           http://www.ietf.org/ID-Checklist.html and
           http://tools.ietf.org/tools/idnits/). Boilerplate checks are
           not enough; this check needs to be thorough. Has the document
           met all formal review criteria it needs to, such as the MIB
           Doctor, media type and URI type reviews?

The document shepherd has personally verified that this version of the
document satisfies all of the ID nits, bar a warning about legal
boilerplate about pre-RFC5378 work.

     (1.h) Has the document split its references into normative and
           informative? Are there normative references to documents that
           are not ready for advancement or are otherwise in an unclear
           state? If such normative references exist, what is the
           strategy for their completion? Are there normative references
           that are downward references, as described in [RFC3967]? If
           so, list these downward references to support the Area
           Director in the Last Call procedure for them [RFC3967].

The document has been split into normative and informative references.
All normative references are stable.

     (1.i) Has the Document Shepherd verified that the document IANA
           consideration section exists and is consistent with the body
           of the document? If the document specifies protocol
           extensions, are reservations requested in appropriate IANA
           registries? Are the IANA registries clearly identified? If
           the document creates a new registry, does it define the
           proposed initial contents of the registry and an allocation
           procedure for future registrations? Does it suggest a
           reasonable name for the new registry? See [RFC5226]. If the
           document describes an Expert Review process has Shepherd
           conferred with the Responsible Area Director so that the IESG
           can appoint the needed Expert during the IESG Evaluation?

This document has an IANA considerations section with no actions
for the IANA, which matches the body of the document.

     (1.j) Has the Document Shepherd verified that sections of the
           document that are written in a formal language, such as XML
           code, BNF rules, MIB definitions, etc., validate correctly in
           an automated checker?

SIP messages in examples have been manually checked and appear to be
valid.

     (1.k) The IESG approval announcement includes a Document
           Announcement Write-Up. Please provide such a Document
           Announcement Write-Up? Recent examples can be found in the
           "Action" announcements for approved documents. The approval
           announcement contains the following sections:

           Technical Summary
              Relevant content can frequently be found in the abstract
              and/or introduction of the document. If not, this may be
              an indication that there are deficiencies in the abstract
              or introduction.


Multi-homed SIP proxies (having two or more interfaces) raise special
requirements when handling SIP requests that have different ingress
and egress interfaces.  One technique that has been widely deployed is
to insert a single Record-Route header describing the egress interface.
When another request in the  same dialog then traverses the proxy in a
reverse direction, the proxy must edit the headers to reflect the new
egress interface. This document formalizes another technique that has
also been widely deployed and found to be simpler and more effective.
Using this technique, the proxy inserts two record-route headers, one
for the ingress interface and one for the egress interface. This keeps  
the
route symmetric, such that it can be traversed in the reverse direction
without requiring examination and editing by the proxy, thereby
reducing system complexity and processing load in the proxy.


           Working Group Summary
              Was there anything in WG process that is worth noting? For
              example, was there controversy about particular points or
              were there decisions where the consensus was particularly
              rough?

The biggest discussion around this document was which status this  
document
should get. The working group decided to target for Best Current  
Practice
and against a normative update of RFC3261. Work on this document began  
in
March 2007 and the WGLC was held in July 2008.

           Document Quality
              Are there existing implementations of the protocol? Have a
              significant number of vendors indicated their plan to
              implement the specification? Are there any reviewers that
              merit special mention as having done a thorough review,
              e.g., one that resulted in important changes or a
              conclusion that the document had no substantive issues? If
              there was a MIB Doctor, Media Type or other expert review,
              what was its course (briefly)? In the case of a Media Type
              review, on what date was the request posted?

This document specifically addresses an area of SIP that has been an
interoperability problem in the past. The SIPit interoperability
events have seen many problems in the area of Record-Route handling
in multi homed scenarios (including traversal between different sub
networks and changes of transport or IP protocol). The described
technique is widely deployed in several SIP proxies already.

This document has been reviewed by many participants over the lifetime
of the document, by the following members of the WG:

   - Francois Audet
   - Jonathan Rosenberg
   - Cullen Jennings
   - Paul Kyzivat
   - Robert Sparks
   - Nils Ohlmeier

From drage@alcatel-lucent.com  Mon Mar 16 16:54:18 2009
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To: "marianne.mohali@orange-ftgroup.com" <marianne.mohali@orange-ftgroup.com>,  "mary.barnes@nortel.com" <mary.barnes@nortel.com>, "audet@nortel.com" <audet@nortel.com>
Date: Tue, 17 Mar 2009 00:54:55 +0100
Thread-Topic: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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Cc: "sip@ietf.org" <sip@ietf.org>
Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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You wrote:

> ** To make the History-Info header usefull for applications=20
> and services interactions, it could be interesting to be able=20
> to include in the hi-target-to-uri IARI and ICSC values as=20
> described in TS 24.229 (3GPP) wich ar coded as URN. An=20
> example of an ICSI for a 3GPP defined IMS communication=20
> service is: urn:urn-xxx:3gpp-service.ims.icsi.mmtel
> Do you think, this kind of enhancement for Histroy-Info=20
> header could be added in the document ?

Question: What have these values to do with the history of a retargetting?

Keith=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of marianne.mohali@orange-ftgroup.com
> Sent: Monday, March 16, 2009 10:11 PM
> To: mary.barnes@nortel.com; audet@nortel.com
> Cc: sip@ietf.org
> Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
>=20
>=20
> Hi Mary and Fran=E7ois,
>=20
> I have a suggestion and few comments on the document.
>=20
> ** To make the History-Info header usefull for applications=20
> and services interactions, it could be interesting to be able=20
> to include in the hi-target-to-uri IARI and ICSC values as=20
> described in TS 24.229 (3GPP) wich ar coded as URN. An=20
> example of an ICSI for a 3GPP defined IMS communication=20
> service is: urn:urn-xxx:3gpp-service.ims.icsi.mmtel
> Do you think, this kind of enhancement for Histroy-Info=20
> header could be added in the document ?
>=20
> ** Section 4.1: About Retarget (hi-target), it is indicated=20
> "A retarget parameter is not added for a hi-entry when it is=20
> first added in a History-
> Info header, but rather is added when the retargeting=20
> actually occurs". It is indicated when the Retarget=20
> indication should be added but it could be 	good to add an=20
> information on where the parameter should be added=20
> (associated to the retargeting hi-targeted-to-uri). This is=20
> just to be clear on 	the History-Info header structure.=20
> I have the same comment for the Reason header escaped in the=20
> History-Info. About that, there is an inconsistency between=20
> Reason header associated to 	the retargeting URI and=20
> cause-param parameter (RFC4458) associated to the=20
> retargeted-to URI. This should be corrected in RFC4458 but I=20
> don't know if 	it is possible.
>=20
> ** Section 4.5.1 and following: in the call flows there is a=20
> "retarget" parmeter inserted in the second INVITE by Proxy1=20
> when sending the request to Proxy2. In my understanding, it=20
> should only be added (in the same place) by Proxy2 when=20
> retargeting the request to Bob. Did I miss something ?
>=20
> ** Privacy: I also think that it should be recommended in the=20
> document to use the Privacy header with the value "history"=20
> escaped in the hi-entries rather than directly in the INVITE=20
> request. =20
>=20
>=20
> Best Regards,
> Marianne
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
> =

From rco@iptel.org  Tue Mar 17 02:48:16 2009
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Nils, Jonathan,

You may be aware of it or not, however, we have already implemented this 
scenario at iptel.org. This is because iptel.org does not have any own 
PSTN gateway. In this case, the user can configure a PSTN gateway he 
whishes to use when such a number is dialed. We implemented like this to 
cope with all the low-end phones that do not provide multiple accounts, 
or do not offer a convenient way for the user to provide routing rules.

Even cisco and snom phones (at least those we have) are not able to 
select the outbound account on their own, based on the R-URI. On the 
other hand, this is very easy to offer at the proxy level, provided that 
my proxy does relay challenges and responses to the UA (in this case, 
the realm is different).

So, as a conclusion, I do not think we are talking about theory, but 
about real existing applications. Of course, you could also store your 
other PSTN credentials at the proxy, and make it deal with the 
authentication for you. However, I'm quite sure you wouldn't be very 
confident about that, would you?

Regards,
Raphael.

Nils Ohlmeier wrote:
>>>> Server identity can be verified by normal server-only auth between a
>>>> client and its server, but even that is not needed.
>>> Right, mutual authentication seems to be the best way.
>>>
>>>> A client will know which domain its proxy is representing, and once
>>>> connected, it only provides credentials for that domain.
>>>>
>>> What do you mean by "connected"? And why should a UA only provide
>>> credentials for one domain only?
>>
>> I'm saying, when a phone registers or makes a call, it does so by
>> connecting to a SIP server, through a domain name or IP config or
>> whatever. THat server will have an associated credential. For any
>> request sent to that server, it should never provide a credential except
>> the one associated with it.
>>
>> Your attack is possible only when the client sends credentials for a
>> domain, different than the one it is currently registered or placed the
>> call through in the first place.
>
> And the SIP server which relays the INVITE from the attacker to the 
> victim has to drop (?) the challenge if it receives such from the 
> attacker (at least if the realm is identical to the one used by the 
> server itself).
>
> This is basically the same description which I provided already earlier.
>
> The only open "issue" is then relaying of 407/401 replies from 
> downstream elements by the proxy. I agree with Jonathan that I also do 
> not believe that this actually a real world scenario. But mutual 
> authentication in the challenge might help to solve this issue if 
> people insist of keeping this scenario.
>
> Regards
>   Nils


From victor.pascual.avila@gmail.com  Tue Mar 17 03:16:54 2009
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From: =?UTF-8?Q?Victor_Pascual_=C3=81vila?= <victor.pascual.avila@gmail.com>
To: Raphael Coeffic <rco@iptel.org>
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Subject: Re: [Sip] draft-state-sip-relay-attack-00
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On Tue, Mar 17, 2009 at 10:48 AM, Raphael Coeffic <rco@iptel.org> wrote:
> You may be aware of it or not, however, we have already implemented this
> scenario at iptel.org. This is because iptel.org does not have any own PS=
TN
> gateway. In this case, the user can configure a PSTN gateway he whishes t=
o
> use when such a number is dialed. We implemented like this to cope with a=
ll
> the low-end phones that do not provide multiple accounts, or do not offer=
 a
> convenient way for the user to provide routing rules.

I agree-- some terminating domains may require authentication. Indeed,
this is defined in rfc3665 as a BCP. In Section 3.3 (Session with
Multiple Proxy Authentication), Proxy 1 is relaying challenges and
responses for different (sub)domains.

> So, as a conclusion, I do not think we are talking about theory, but abou=
t
> real existing applications. Of course, you could also store your other PS=
TN
> credentials at the proxy, and make it deal with the authentication for yo=
u.
> However, I'm quite sure you wouldn't be very confident about that, would
> you?

Once again, IMHO this should be as much end-to-end as possible. Some
folks are still using SIP as a protocol for the (open) Internet.

Cheers,
--=20
Victor Pascual =C3=81vila

From marianne.mohali@orange-ftgroup.com  Thu Mar 19 02:52:51 2009
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From: <marianne.mohali@orange-ftgroup.com>
To: <drage@alcatel-lucent.com>
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Cc: sip@ietf.org, audet@nortel.com, mary.barnes@nortel.com
Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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Hi Keith,

Let's just talk about IARI. My idea was just to suggest a trail to =
improve History-Info. Before to create new things (headers or =
parameters), we should look what is available and reusable. If it is not =
conceivable to use IARI values in a History-Info header allowing a proxy =
server to indicate what has been done in term of service, could you =
please give a reason.=20

Best Regards,
Marianne=20

-----Message d'origine-----
De : DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]=20
Envoy=E9 : mardi 17 mars 2009 00:55
=C0 : MOHALI Marianne RD-CORE-ISS; mary.barnes@nortel.com; =
audet@nortel.com
Cc : sip@ietf.org
Objet : RE: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt

You wrote:

> ** To make the History-Info header usefull for applications and=20
> services interactions, it could be interesting to be able to include=20
> in the hi-target-to-uri IARI and ICSC values as described in TS 24.229 =

> (3GPP) wich ar coded as URN. An example of an ICSI for a 3GPP defined=20
> IMS communication service is: urn:urn-xxx:3gpp-service.ims.icsi.mmtel
> Do you think, this kind of enhancement for Histroy-Info header could=20
> be added in the document ?

Question: What have these values to do with the history of a =
retargetting?

Keith=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of=20
> marianne.mohali@orange-ftgroup.com
> Sent: Monday, March 16, 2009 10:11 PM
> To: mary.barnes@nortel.com; audet@nortel.com
> Cc: sip@ietf.org
> Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
>=20
>=20
> Hi Mary and Fran=E7ois,
>=20
> I have a suggestion and few comments on the document.
>=20
> ** To make the History-Info header usefull for applications and=20
> services interactions, it could be interesting to be able to include=20
> in the hi-target-to-uri IARI and ICSC values as described in TS 24.229 =

> (3GPP) wich ar coded as URN. An example of an ICSI for a 3GPP defined=20
> IMS communication service is: urn:urn-xxx:3gpp-service.ims.icsi.mmtel
> Do you think, this kind of enhancement for Histroy-Info header could=20
> be added in the document ?
>=20
> ** Section 4.1: About Retarget (hi-target), it is indicated "A=20
> retarget parameter is not added for a hi-entry when it is first added=20
> in a History- Info header, but rather is added when the retargeting=20
> actually occurs". It is indicated when the Retarget
> indication should be added but it could be 	good to add an=20
> information on where the parameter should be added (associated to the=20
> retargeting hi-targeted-to-uri). This is
> just to be clear on 	the History-Info header structure.=20
> I have the same comment for the Reason header escaped in the=20
> History-Info. About that, there is an inconsistency between
> Reason header associated to 	the retargeting URI and=20
> cause-param parameter (RFC4458) associated to the retargeted-to URI.=20
> This should be corrected in RFC4458 but I
> don't know if 	it is possible.
>=20
> ** Section 4.5.1 and following: in the call flows there is a=20
> "retarget" parmeter inserted in the second INVITE by Proxy1 when=20
> sending the request to Proxy2. In my understanding, it should only be=20
> added (in the same place) by Proxy2 when retargeting the request to=20
> Bob. Did I miss something ?
>=20
> ** Privacy: I also think that it should be recommended in the document =

> to use the Privacy header with the value "history"
> escaped in the hi-entries rather than directly in the INVITE request.
>=20
>=20
> Best Regards,
> Marianne
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> sipping@ietf.org for new developments on the application of sip
>=20

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Date: Thu, 19 Mar 2009 11:51:11 +0100
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From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: mary.barnes@nortel.com
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Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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Hi Mary,

Other discussion on the list triggered the following question when looking
at how Request-URI's are recorded in 4244.

Assume: A UAS that receiving the History-Info in a request. The UAS needs
the original parameters that where present on a particular recorded
Request-URI.

Q1: Does the current 4244 text ensure that the Request-URI parameters are
recorded?
Q2: Assuming they are recorded, how to distinguish parameters that where
originally present and those that where added in the History-Info recording
process?
Q3: What was the design objective for not having reason headers etc be
recorded as hi-param? I.e. why is the recorded URI modified?

Best Regards,
/Hans Erik van Elburg

--00504502d2f61682570465769454
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Hi Mary,<br><br>Other discussion on the list triggered the following questi=
on when looking at how Request-URI&#39;s are recorded in 4244.<br><br>Assum=
e: A UAS that receiving the History-Info in a request. The UAS needs the or=
iginal parameters that where present on a particular recorded Request-URI.<=
br>
<br>Q1: Does the current 4244 text ensure that the Request-URI parameters a=
re recorded?<br>Q2: Assuming they are recorded, how to distinguish paramete=
rs that where originally present and those that where added in the History-=
Info recording process?<br>
Q3: What was the design objective for not having reason headers etc be reco=
rded as hi-param? I.e. why is the recorded URI modified?<br><br>Best Regard=
s,<br clear=3D"all">/Hans Erik van Elburg<br>
<br><br>

--00504502d2f61682570465769454--

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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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Hi,

Hans Erik sent this text already earlier, but I would like to re-send it =
as a separate thread. It is related to the definitions which I think we =
should agree on.

Regards,

Christer




Current target

The current target of an initial request for a dialog or standalone
request is the name or address to which the request is targeted, i.e.
either the initial target inserted in the Request-URI by the UAC that
originates the request, or when a retarget occurred, the target
provided in that retarget operation. Reroute and translation
operations never change the current target.


This defintion only makes sense when the following definitions are used:

Name:

A name is a moniker for an entity which refers to it in a way which =
reveals nothing about where it is in a network. In SIP, tel URI which =
doesn't represent the location of the entity is a name.=20

Address:

An address is an identifier for an entity which describes it by its =
location on the network. In SIP, the SIP URI itself is a form of address =
because the host part of the URI, the only mandatory part of the URI =
besides the scheme itself, indicates the location of a SIP server that =
can be used to handle the request.

Route:

Finally, a route is a sequence of SIP entities (including the UA =
itself!) which are traversed in order to forward a request to an address =
or name.

Retarget (other term might be needed, as this is highly confusing):

A Request-URI rewrite operation that changes the target identity of the =
request.

Reroute (other term might be needed):

A Request-URI rewrite operation that does not change the current target =
of the request, but determines the route/next hop taken to reach the =
target-identity.=20


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
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charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
6.5.7654.12">
<TITLE>DEFINITION: Target, Address, Name, Retarget, Reroute</TITLE>
</HEAD>
<BODY>
<!-- Converted from text/rtf format -->
<BR>

<P><FONT SIZE=3D2 FACE=3D"Arial">Hi,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Hans Erik sent this text already =
earlier, but I would like to re-send it as a separate thread. It is =
related to the definitions which I think we should agree on.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Regards,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Christer</FONT>
</P>
<BR>
<BR>
<BR>

<P><B><FONT FACE=3D"Arial">Current target</FONT></B><BR>
<BR>
<FONT FACE=3D"Arial">The current target of an initial request for a =
dialog or standalone<BR>
request is the name or address to which the request is targeted, =
i.e.<BR>
either the initial target inserted in the Request-URI by the UAC =
that<BR>
originates the request, or when a retarget occurred, the target<BR>
provided in that retarget operation. Reroute and translation<BR>
operations never change the current target.<BR>
<BR>
<BR>
This defintion only makes sense when the following definitions are =
used:<BR>
<BR>
</FONT><B><FONT FACE=3D"Arial">Name:</FONT></B><BR>
<BR>
<FONT FACE=3D"Arial">A name is a moniker for an entity which refers to =
it in a way which reveals nothing about where it is in a network. In =
SIP, tel URI which doesn't represent the location of the entity is a =
name.<BR>
<BR>
</FONT><B><FONT FACE=3D"Arial">Address:<BR>
</FONT></B><BR>
<FONT FACE=3D"Arial">An address is an identifier for an entity which =
describes it by its location on the network. In SIP, the SIP URI itself =
is a form of address because the host part of the URI, the only =
mandatory part of the URI besides the scheme itself, indicates the =
location of a SIP server that can be used to handle the request.<BR>
<BR>
</FONT><B><FONT FACE=3D"Arial">Route:<BR>
</FONT></B><BR>
<FONT FACE=3D"Arial">Finally, a route is a sequence of SIP entities =
(including the UA itself!) which are traversed in order to forward a =
request to an address or name.<BR>
<BR>
</FONT><B><FONT FACE=3D"Arial">Retarget (other term might be needed, as =
this is highly confusing):<BR>
</FONT></B><BR>
<FONT FACE=3D"Arial">A Request-URI rewrite operation that changes the =
target identity of the request.<BR>
<BR>
</FONT><B><FONT FACE=3D"Arial">Reroute (other term might be needed):<BR>
</FONT></B><BR>
<FONT FACE=3D"Arial">A Request-URI rewrite operation that does not =
change the current target of the request, but determines the route/next =
hop taken to reach the target-identity.=A0</FONT></P>

</BODY>
</HTML>
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From drage@alcatel-lucent.com  Thu Mar 19 15:46:58 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "sip@ietf.org" <sip@ietf.org>
Date: Thu, 19 Mar 2009 23:47:39 +0100
Thread-Topic: DEFINITION: Target, Address, Name, Retarget, Reroute
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Subject: Re: [Sip] DEFINITION: Target, Address, Name, Retarget, Reroute
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Your definition of name implies that if a name contains any information at =
all that could be addressing information, then it cannot be a name. If I re=
ceive a greeting card that says "Greetings to the occupants of no. 10", tha=
t to me is using "occupants of no. 10" as a name, even though it may also i=
n other circumstances parts of it may form part of an address.
=20
So I would suggest something more along the lines of:

"A name is a moniker for an entity which refers to it in a way which is not=
 expected to reveal anything about where it is in a network. In SIP, tel UR=
I which doesn't represent the location of the entity is a name."

Also tel URIs still contain a domain name, so does that no make them potent=
ially addressing information at least in this respect, in the same way as y=
ou have described for the SIP URI in respect of addresses.

regards

Keith



________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Chri=
ster Holmberg
	Sent: Thursday, March 19, 2009 2:34 PM
	To: sip@ietf.org
	Subject: [Sip] DEFINITION: Target, Address, Name, Retarget, Reroute
=09
=09
=09
=09

	Hi,=20

	Hans Erik sent this text already earlier, but I would like to re-send it a=
s a separate thread. It is related to the definitions which I think we shou=
ld agree on.

	Regards,=20

	Christer=20




	Current target
=09
	The current target of an initial request for a dialog or standalone
	request is the name or address to which the request is targeted, i.e.
	either the initial target inserted in the Request-URI by the UAC that
	originates the request, or when a retarget occurred, the target
	provided in that retarget operation. Reroute and translation
	operations never change the current target.
=09
=09
	This defintion only makes sense when the following definitions are used:
=09
	Name:
=09
	A name is a moniker for an entity which refers to it in a way which reveal=
s nothing about where it is in a network. In SIP, tel URI which doesn't rep=
resent the location of the entity is a name.
=09
	Address:
=09
	An address is an identifier for an entity which describes it by its locati=
on on the network. In SIP, the SIP URI itself is a form of address because =
the host part of the URI, the only mandatory part of the URI besides the sc=
heme itself, indicates the location of a SIP server that can be used to han=
dle the request.
=09
	Route:
=09
	Finally, a route is a sequence of SIP entities (including the UA itself!) =
which are traversed in order to forward a request to an address or name.
=09
	Retarget (other term might be needed, as this is highly confusing):
=09
	A Request-URI rewrite operation that changes the target identity of the re=
quest.
=09
	Reroute (other term might be needed):
=09
	A Request-URI rewrite operation that does not change the current target of=
 the request, but determines the route/next hop taken to reach the target-i=
dentity.=20


From drage@alcatel-lucent.com  Thu Mar 19 15:46:58 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: "marianne.mohali@orange-ftgroup.com" <marianne.mohali@orange-ftgroup.com>
Date: Thu, 19 Mar 2009 23:47:40 +0100
Thread-Topic: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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Cc: "sip@ietf.org" <sip@ietf.org>, "audet@nortel.com" <audet@nortel.com>, "mary.barnes@nortel.com" <mary.barnes@nortel.com>
Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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IARI is end to end (either user terminal or application) and is not changed=
 by intermediate call functionality.=20

Therefore any intermediate application that changes this is essentially mak=
ing a new call that is not the original call. Think of something like a con=
ference bridge.

If you start recording IARI in History-Info, then that would suggest you re=
cord all the other headers and bodies that are end to end information.

regards

Keith

> -----Original Message-----
> From: marianne.mohali@orange-ftgroup.com=20
> [mailto:marianne.mohali@orange-ftgroup.com]=20
> Sent: Thursday, March 19, 2009 9:54 AM
> To: DRAGE, Keith (Keith)
> Cc: sip@ietf.org; mary.barnes@nortel.com; audet@nortel.com
> Subject: RE: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
>=20
> Hi Keith,
>=20
> Let's just talk about IARI. My idea was just to suggest a=20
> trail to improve History-Info. Before to create new things=20
> (headers or parameters), we should look what is available and=20
> reusable. If it is not conceivable to use IARI values in a=20
> History-Info header allowing a proxy server to indicate what=20
> has been done in term of service, could you please give a reason.=20
>=20
> Best Regards,
> Marianne=20
>=20
> -----Message d'origine-----
> De : DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]=20
> Envoy=E9 : mardi 17 mars 2009 00:55 =C0 : MOHALI Marianne=20
> RD-CORE-ISS; mary.barnes@nortel.com; audet@nortel.com Cc :=20
> sip@ietf.org Objet : RE: [Sip] I-D=20
> Action:draft-barnes-sip-rfc4244bis-00.txt
>=20
> You wrote:
>=20
> > ** To make the History-Info header usefull for applications and=20
> > services interactions, it could be interesting to be able=20
> to include=20
> > in the hi-target-to-uri IARI and ICSC values as described=20
> in TS 24.229
> > (3GPP) wich ar coded as URN. An example of an ICSI for a=20
> 3GPP defined=20
> > IMS communication service is:=20
> urn:urn-xxx:3gpp-service.ims.icsi.mmtel
> > Do you think, this kind of enhancement for Histroy-Info=20
> header could=20
> > be added in the document ?
>=20
> Question: What have these values to do with the history of a=20
> retargetting?
>=20
> Keith=20
>=20
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of=20
> > marianne.mohali@orange-ftgroup.com
> > Sent: Monday, March 16, 2009 10:11 PM
> > To: mary.barnes@nortel.com; audet@nortel.com
> > Cc: sip@ietf.org
> > Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
> >=20
> >=20
> > Hi Mary and Fran=E7ois,
> >=20
> > I have a suggestion and few comments on the document.
> >=20
> > ** To make the History-Info header usefull for applications and=20
> > services interactions, it could be interesting to be able=20
> to include=20
> > in the hi-target-to-uri IARI and ICSC values as described=20
> in TS 24.229
> > (3GPP) wich ar coded as URN. An example of an ICSI for a=20
> 3GPP defined=20
> > IMS communication service is:=20
> urn:urn-xxx:3gpp-service.ims.icsi.mmtel
> > Do you think, this kind of enhancement for Histroy-Info=20
> header could=20
> > be added in the document ?
> >=20
> > ** Section 4.1: About Retarget (hi-target), it is indicated "A=20
> > retarget parameter is not added for a hi-entry when it is=20
> first added=20
> > in a History- Info header, but rather is added when the retargeting=20
> > actually occurs". It is indicated when the Retarget
> > indication should be added but it could be 	good to add an=20
> > information on where the parameter should be added=20
> (associated to the=20
> > retargeting hi-targeted-to-uri). This is
> > just to be clear on 	the History-Info header structure.=20
> > I have the same comment for the Reason header escaped in the=20
> > History-Info. About that, there is an inconsistency between
> > Reason header associated to 	the retargeting URI and=20
> > cause-param parameter (RFC4458) associated to the=20
> retargeted-to URI.=20
> > This should be corrected in RFC4458 but I
> > don't know if 	it is possible.
> >=20
> > ** Section 4.5.1 and following: in the call flows there is a=20
> > "retarget" parmeter inserted in the second INVITE by Proxy1 when=20
> > sending the request to Proxy2. In my understanding, it=20
> should only be=20
> > added (in the same place) by Proxy2 when retargeting the request to=20
> > Bob. Did I miss something ?
> >=20
> > ** Privacy: I also think that it should be recommended in=20
> the document=20
> > to use the Privacy header with the value "history"
> > escaped in the hi-entries rather than directly in the=20
> INVITE request.
> >=20
> >=20
> > Best Regards,
> > Marianne
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
> =

From mary.barnes@nortel.com  Thu Mar 19 22:33:30 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Mary Barnes" <mary.barnes@nortel.com>, "Hans Erik van Elburg" <ietf.hanserik@gmail.com>, <sip@ietf.org>
Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
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Resending to include SIP WG mailing list.


-----Original Message-----
From: Barnes, Mary (RICH2:AR00)
Sent: Fri 3/20/2009 12:32 AM
To: Hans Erik van Elburg
Subject: RE: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
=20
Hi Hans Erick,

Responses below [MB].

Thanks,
Mary.


-----Original Message-----
From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
Sent: Thu 3/19/2009 5:51 AM
To: Barnes, Mary (RICH2:AR00)
Cc: sip@ietf.org
Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt
=20
Hi Mary,

Other discussion on the list triggered the following question when =
looking
at how Request-URI's are recorded in 4244.

Assume: A UAS that receiving the History-Info in a request. The UAS =
needs
the original parameters that where present on a particular recorded
Request-URI.

Q1: Does the current 4244 text ensure that the Request-URI parameters =
are
recorded?
[MB] The current text doesn't specifically discuss Request-URI =
parameters, but per the general rule that information is not removed =
from messages, the parameters should be left intact. I can add something =
to that affect to the next version to clarify that point. [/MB]
Q2: Assuming they are recorded, how to distinguish parameters that where
originally present and those that where added in the History-Info =
recording
process?
[MB] History-Info doesn't add any new Request-URI parameters.  The =
Reason header is escaped in the Request-URI, but it's not something that =
would be part of any SIP Request and would only appear in a Request-URI =
that has been captured in the History-Info header. We could also add a =
statement clarifying that to RFC 4244.=20
Q3: What was the design objective for not having reason headers etc be
recorded as hi-param? I.e. why is the recorded URI modified?
[MB] That was just a matter of reusing an existing header versus =
defining a parameter that had the exact same information. This was an =
issue discussed at IETF-55 in Nov 22 (Issue 2, chart 5):
http://www.ietf.org/proceedings/02nov/slides/sipping-2/index.html
You can see in those charts (and older versions of the drafts that you =
may or may not be able to dig up) that the Reason was originally an =
hi-param. The escaping also saves a few bytes per hi-entry.   The =
recorded URI is modified because the intent was to capture "why" that =
URI was retargeted. You can also look at those charts above =
(specifically chart 4 that has the ABNF) and see that we were originally =
capturing two entries at each node - old and new. But, again, to make =
the header more compact we simplified to just record the new and thus =
have the "old" already captured by the time the request is received by =
the next node. The Reason header is only added (to the "old" hi-entry) =
in the case that the retargeted URI is in the domain for which the =
entity doing the retargeting is responsible. [/MB]

Best Regards,
/Hans Erik van Elburg



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<TITLE>RE: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt</TITLE>
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<P><FONT SIZE=3D2>Resending to include SIP WG mailing list.<BR>
<BR>
<BR>
-----Original Message-----<BR>
From: Barnes, Mary (RICH2:AR00)<BR>
Sent: Fri 3/20/2009 12:32 AM<BR>
To: Hans Erik van Elburg<BR>
Subject: RE: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt<BR>
<BR>
Hi Hans Erick,<BR>
<BR>
Responses below [MB].<BR>
<BR>
Thanks,<BR>
Mary.<BR>
<BR>
<BR>
-----Original Message-----<BR>
From: Hans Erik van Elburg [<A =
HREF=3D"mailto:ietf.hanserik@gmail.com">mailto:ietf.hanserik@gmail.com</A=
>]<BR>
Sent: Thu 3/19/2009 5:51 AM<BR>
To: Barnes, Mary (RICH2:AR00)<BR>
Cc: sip@ietf.org<BR>
Subject: Re: [Sip] I-D Action:draft-barnes-sip-rfc4244bis-00.txt<BR>
<BR>
Hi Mary,<BR>
<BR>
Other discussion on the list triggered the following question when =
looking<BR>
at how Request-URI's are recorded in 4244.<BR>
<BR>
Assume: A UAS that receiving the History-Info in a request. The UAS =
needs<BR>
the original parameters that where present on a particular recorded<BR>
Request-URI.<BR>
<BR>
Q1: Does the current 4244 text ensure that the Request-URI parameters =
are<BR>
recorded?<BR>
[MB] The current text doesn't specifically discuss Request-URI =
parameters, but per the general rule that information is not removed =
from messages, the parameters should be left intact. I can add something =
to that affect to the next version to clarify that point. [/MB]<BR>
Q2: Assuming they are recorded, how to distinguish parameters that =
where<BR>
originally present and those that where added in the History-Info =
recording<BR>
process?<BR>
[MB] History-Info doesn't add any new Request-URI parameters.&nbsp; The =
Reason header is escaped in the Request-URI, but it's not something that =
would be part of any SIP Request and would only appear in a Request-URI =
that has been captured in the History-Info header. We could also add a =
statement clarifying that to RFC 4244.<BR>
Q3: What was the design objective for not having reason headers etc =
be<BR>
recorded as hi-param? I.e. why is the recorded URI modified?<BR>
[MB] That was just a matter of reusing an existing header versus =
defining a parameter that had the exact same information. This was an =
issue discussed at IETF-55 in Nov 22 (Issue 2, chart 5):<BR>
<A =
HREF=3D"http://www.ietf.org/proceedings/02nov/slides/sipping-2/index.html=
">http://www.ietf.org/proceedings/02nov/slides/sipping-2/index.html</A><B=
R>
You can see in those charts (and older versions of the drafts that you =
may or may not be able to dig up) that the Reason was originally an =
hi-param. The escaping also saves a few bytes per hi-entry.&nbsp;&nbsp; =
The recorded URI is modified because the intent was to capture =
&quot;why&quot; that URI was retargeted. You can also look at those =
charts above (specifically chart 4 that has the ABNF) and see that we =
were originally capturing two entries at each node - old and new. But, =
again, to make the header more compact we simplified to just record the =
new and thus have the &quot;old&quot; already captured by the time the =
request is received by the next node. The Reason header is only added =
(to the &quot;old&quot; hi-entry) in the case that the retargeted URI is =
in the domain for which the entity doing the retargeting is responsible. =
[/MB]<BR>
<BR>
Best Regards,<BR>
/Hans Erik van Elburg<BR>
<BR>
<BR>
</FONT>
</P>

</BODY>
</HTML>
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From gonzalo.camarillo@ericsson.com  Fri Mar 20 03:47:02 2009
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Date: Fri, 20 Mar 2009 12:47:27 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [Sip] Policy to register media feature tags
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Folks,

as you know, RFC 3840 specifies how to indicate UA's capabilities using
media feature tags. Section 12.1 of RFC 3840 creates the SIP Media
Feature Tag Registration Tree. Media feature tags applicable to SIP are
registered in this tree.

http://www.ietf.org/rfc/rfc3840.txt

RFC 2506 defines the global tree to register feature tags. Section 3.1.2
of RFC 2506 says: "A registration may be proposed for the global tree by
anyone who has the need to allow for communication on a particular
capability or preference".

http://www.ietf.org/rfc/rfc2506.txt

The currently registered feature tags in both the SIP and the global
trees can be found at:

http://www.iana.org/assignments/media-feature-tags

As you can see, the only three feature tags registered in the global
tree come from 3GPP.

The issue we are facing is that the SIP and the global trees have
different registration policies. The SIP registry requires an RFC
defining the feature tag while the global tree only requires an expert
review.

Now, 3GPP would like to register two new feature tags:  g.3gpp.icsi-ref,
g.3gpp.iari-ref

This is the description of how these feature tags will be used:

"Each value of the Application Reference media feature-tag indicates the
software applications supported by the agent. The values for this tag
equal the IMS communication Service Identifier (ICSI) and IMS
Application Reference Identifier (IARI) values supported by
the agent. The Application Reference media feature tag is
defined to fulfil the requirements for forking to an appropriate UE when
multiple UEs are registered and dispatch to an appropriate application
within the UE based upon the IMS communication Service Identifier
(ICSI) and IMS Application Reference Identifier (IARI) values as
stated in 3GPP TS 23.228.  Multiple tag-values can be included in the
Application Reference media feature-tag."


The expert reviewer indicated that this tags will not contain simple
features but rather more complex services, or feature sets, or
application logic (pick your favorite term). Therefore, he did not find
it appropriate to register them in the global tree. He suggested that
RFC 2506 is amended so that it covers this type of more complex features.

However, before going down that path, I would like to get comments from
the SIP community. The underlying issue is that we need to decide which
policy we want to apply to these types of registrations in SIP.

1) We decide that an RFC is always needed. Then, we can just use the SIP
registry. We would probably need to clarify or to stress somewhere that
the global tree is not appropriate for SIP tags.

2) We decide that an expert review is good enough. Then we would
probably need to amend RFC 2506 or RFC 3840 to reflect this.

Comments are welcome.

Thanks,

Gonzalo


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Cc: sip <sip@ietf.org>
Subject: Re: [Sip] Policy to register media feature tags
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Some issues just never go away. :-(

This seems to be a resurection of issues that first surfaced during the 
initial discussions of extended presence status many years ago. And it 
also harks back to discussions last year or the year before about 
communication service identifiers and application service identifiers.

The problem is one of openness. Specifically, do you have to be "part of 
the club" in order to properly interoperate with a device that uses 
these identifiers to identify its preferences or capabilities.

On one side were those who wanted capabilities and preferences to be 
stated in terms of feature tags that are well known and orthogonal. On 
the other side were those who prefer to associate arbitrary names to 
collections of features and then negotiate on the basis of the names of 
those collections.

Using the latter approach its possible that interoperation will fail 
because the parties don't share a common name for a collection of 
features even though they both possess the necessary features to 
interoperate.

I don't have a problem with independent groups defining new feature 
tags, as long as they are primitive and orthogonal to existing ones. But 
I do have a problem with defining feature tags that identify collections 
of features, especially when the mapping from the identifier to the 
collection of features is not public.

The bottom line is that I think this sort of thing needs to be thrashed 
out within the sip community. So I think the RFC process is the right 
process.

	Thanks,
	Paul

Gonzalo Camarillo wrote:
> Folks,
> 
> as you know, RFC 3840 specifies how to indicate UA's capabilities using
> media feature tags. Section 12.1 of RFC 3840 creates the SIP Media
> Feature Tag Registration Tree. Media feature tags applicable to SIP are
> registered in this tree.
> 
> http://www.ietf.org/rfc/rfc3840.txt
> 
> RFC 2506 defines the global tree to register feature tags. Section 3.1.2
> of RFC 2506 says: "A registration may be proposed for the global tree by
> anyone who has the need to allow for communication on a particular
> capability or preference".
> 
> http://www.ietf.org/rfc/rfc2506.txt
> 
> The currently registered feature tags in both the SIP and the global
> trees can be found at:
> 
> http://www.iana.org/assignments/media-feature-tags
> 
> As you can see, the only three feature tags registered in the global
> tree come from 3GPP.
> 
> The issue we are facing is that the SIP and the global trees have
> different registration policies. The SIP registry requires an RFC
> defining the feature tag while the global tree only requires an expert
> review.
> 
> Now, 3GPP would like to register two new feature tags:  g.3gpp.icsi-ref,
> g.3gpp.iari-ref
> 
> This is the description of how these feature tags will be used:
> 
> "Each value of the Application Reference media feature-tag indicates the
> software applications supported by the agent. The values for this tag
> equal the IMS communication Service Identifier (ICSI) and IMS
> Application Reference Identifier (IARI) values supported by
> the agent. The Application Reference media feature tag is
> defined to fulfil the requirements for forking to an appropriate UE when
> multiple UEs are registered and dispatch to an appropriate application
> within the UE based upon the IMS communication Service Identifier
> (ICSI) and IMS Application Reference Identifier (IARI) values as
> stated in 3GPP TS 23.228.  Multiple tag-values can be included in the
> Application Reference media feature-tag."
> 
> 
> The expert reviewer indicated that this tags will not contain simple
> features but rather more complex services, or feature sets, or
> application logic (pick your favorite term). Therefore, he did not find
> it appropriate to register them in the global tree. He suggested that
> RFC 2506 is amended so that it covers this type of more complex features.
> 
> However, before going down that path, I would like to get comments from
> the SIP community. The underlying issue is that we need to decide which
> policy we want to apply to these types of registrations in SIP.
> 
> 1) We decide that an RFC is always needed. Then, we can just use the SIP
> registry. We would probably need to clarify or to stress somewhere that
> the global tree is not appropriate for SIP tags.
> 
> 2) We decide that an expert review is good enough. Then we would
> probably need to amend RFC 2506 or RFC 3840 to reflect this.
> 
> Comments are welcome.
> 
> Thanks,
> 
> Gonzalo
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

From pkyzivat@cisco.com  Fri Mar 20 06:46:12 2009
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From: Paul Kyzivat <pkyzivat@cisco.com>
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Cc: "sip@ietf.org" <sip@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] DEFINITION: Target, Address, Name, Retarget, Reroute
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DRAGE, Keith (Keith) wrote:
> Your definition of name implies that if a name contains any information at all that could be addressing information, then it cannot be a name. If I receive a greeting card that says "Greetings to the occupants of no. 10", that to me is using "occupants of no. 10" as a name, even though it may also in other circumstances parts of it may form part of an address.
>  
> So I would suggest something more along the lines of:
> 
> "A name is a moniker for an entity which refers to it in a way which is not expected to reveal anything about where it is in a network. In SIP, tel URI which doesn't represent the location of the entity is a name."
> 
> Also tel URIs still contain a domain name, so does that no make them potentially addressing information at least in this respect, in the same way as you have described for the SIP URI in respect of addresses.

Keith, I agreed with you up to this point.
But tel URIs do not contain domain names.
A sip URI, with user=phone contains the essence of a tel uri in its user 
part, and also contains a domain name, but that is still a sip URI, not 
a tel URI.

(A minor exception to the above is a tel uri with a phone-context 
containing a domain name.)

	Thanks,
	Paul

> regards
> 
> Keith
> 
> 
> 
> ________________________________
> 
> 	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Christer Holmberg
> 	Sent: Thursday, March 19, 2009 2:34 PM
> 	To: sip@ietf.org
> 	Subject: [Sip] DEFINITION: Target, Address, Name, Retarget, Reroute
> 	
> 	
> 	
> 	
> 
> 	Hi, 
> 
> 	Hans Erik sent this text already earlier, but I would like to re-send it as a separate thread. It is related to the definitions which I think we should agree on.
> 
> 	Regards, 
> 
> 	Christer 
> 
> 
> 
> 
> 	Current target
> 	
> 	The current target of an initial request for a dialog or standalone
> 	request is the name or address to which the request is targeted, i.e.
> 	either the initial target inserted in the Request-URI by the UAC that
> 	originates the request, or when a retarget occurred, the target
> 	provided in that retarget operation. Reroute and translation
> 	operations never change the current target.
> 	
> 	
> 	This defintion only makes sense when the following definitions are used:
> 	
> 	Name:
> 	
> 	A name is a moniker for an entity which refers to it in a way which reveals nothing about where it is in a network. In SIP, tel URI which doesn't represent the location of the entity is a name.
> 	
> 	Address:
> 	
> 	An address is an identifier for an entity which describes it by its location on the network. In SIP, the SIP URI itself is a form of address because the host part of the URI, the only mandatory part of the URI besides the scheme itself, indicates the location of a SIP server that can be used to handle the request.
> 	
> 	Route:
> 	
> 	Finally, a route is a sequence of SIP entities (including the UA itself!) which are traversed in order to forward a request to an address or name.
> 	
> 	Retarget (other term might be needed, as this is highly confusing):
> 	
> 	A Request-URI rewrite operation that changes the target identity of the request.
> 	
> 	Reroute (other term might be needed):
> 	
> 	A Request-URI rewrite operation that does not change the current target of the request, but determines the route/next hop taken to reach the target-identity. 
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

From dworley@nortel.com  Fri Mar 20 11:46:02 2009
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From: "Dale Worley" <dworley@nortel.com>
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Subject: Re: [Sip] RFC3265bis
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On Tue, 2009-03-03 at 16:56 -0600, Adam Roach wrote:
> After 6 years of testing at SIPits and other experience in the field, it 
> feels like time to fix some of the known bugs in RFC 3265. I've put 
> together a 3265bis that addresses most of the bugs that have been 
> reported to date (although some are called out as known issues):
> 
> http://www.ietf.org/internet-drafts/draft-roach-sipcore-rfc3265bis-00.txt

That looks good.  I've only skimmed it yet, but it cleans up a lot of
the things we've seen problems with.

Dale



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Subject: [Sip] draft-ietf-sip-199-06
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Between Minneapolis and now, this document has become a lot more 
confusing, which I think bodes pretty ill for its hopes of being deployed.

For example, in section 5:

   If the received initial request contains an 199 option tag, the UAS
   SHOULD NOT send a 199 response for a dialog on which it intends to
   send a final response...

This reads as if sending a 199 response somehow absolves a UAS of 
sending a final response. Certainly this is not the intention of the 
draft (at least, not according to the text in section 7). Section 5 
needs to be re-phrased to indicate what is actually intended here.

In fact, on that topic, it's radically unclear to me how UASes fit into 
the entire scheme at all -- my understanding is that the 199 response is 
only sent by forking proxies. Given that a UAS is inherently an 
endpoint, the entirety of section 5 seems nonsensical. Either I'm 
missing something (in which case it needs some pretty explicit 
motivating text), or section 5 needs to be removed completely.

In section 6:

   When a forking proxy receives a non-2xx final response that it
   recognizes as terminating one or more early dialogs, if the proxy
   does not intend to forward the final response immediately (in
   accordance with rules for a forking proxy) and the UAC has indicated
   support for the 199 response code, the proxy SHOULD generate and send
   a 199 response upstream for each early dialog terminated on the
   downstream side by the non-2xx final response, except for any early
   dialog for which the proxy has previously received and forwarded a
   199 response.

What? On my best run at reading that, I made it to about the 81st word 
before the first predicate clause fell out of my head. If it's really as 
complicated as the structure of this sentence makes it seem, we need a 
flowchart or some other simplifying mechanism.


/a


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Subject: Re: [Sip] Policy to register media feature tags
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Responses inline...

/Hans Erik

Paul Kyzivat wrote:
> On one side were those who wanted capabilities and preferences to be 
> stated in terms of feature tags that are well known and orthogonal. On 
> the other side were those who prefer to associate arbitrary names to 
> collections of features and then negotiate on the basis of the names 
> of those collections.
>
> Using the latter approach its possible that interoperation will fail 
> because the parties don't share a common name for a collection of 
> features even though they both possess the necessary features to 
> interoperate.
That is a feature. This interoperability "problem" only occurs when the 
originating party includes the header field parameters "require" and 
"explicit" in the Accept-Contact header field containing the feature 
tag. In this case the failure of the call when not all of the receiving  
UA's support  the feature is intended!
> I don't have a problem with independent groups defining new feature 
> tags, as long as they are primitive and orthogonal to existing ones. 
> But I do have a problem with defining feature tags that identify 
> collections of features, especially when the mapping from the 
> identifier to the collection of features is not public.
Well who is going to determine what can be called a feature and what not.
I also like to see where this principle is documented to apply to the 
global tree.
>
> The bottom line is that I think this sort of thing needs to be 
> thrashed out within the sip community. So I think the RFC process is 
> the right process.
I think that bending the rules because some SIP people don't like on how 
other organisations are applying SIP in their solutions will alieanate 
those SIP users from that same SIP community. So I seriously hope we 
will not go down that path.

/Hans Erik

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Subject: Re: [Sip] Fwd: I-D Action:draft-sparks-sip-3261-interop-statement-00.txt
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Thanks, Robert.

Before commenting on the specific list of things, my meta-question is 
what exactly we plan to do with this list. I think we have agreed at 
previous meetings that we were not currently planning on taking SIP to 
draft. Do you think that has changed?

The document does say:

>  While not required by process until
>    advancing to Draft, such a report would provide useful information to
>    guide where we focus our energy maintaing the standard at Proposed.

its not clear to me, what this means. Will we take this list, run tests 
at sipit, and based on that define updates to 3261 to clarify? Does this 
imply that there are tests we're NOT running which we should be, that 
this draft helps us sort out?

Thanks,
Jonathan R.

Robert Sparks wrote:
> Also split out of steps-to-draft
> 
> Begin forwarded message:
> 
>> *From: *Internet-Drafts@ietf.org <mailto:Internet-Drafts@ietf.org>
>> *Date: *March 2, 2009 8:30:02 AM CST
>> *To: *i-d-announce@ietf.org <mailto:i-d-announce@ietf.org>
>> *Subject: **I-D Action:draft-sparks-sip-3261-interop-statement-00.txt *
>> *Reply-To: *internet-drafts@ietf.org <mailto:internet-drafts@ietf.org>
>>
>> A New Internet-Draft is available from the on-line Internet-Drafts 
>> directories.
>>
>> Title           : RFC3261 Interop Statement
>> Author(s)       : R. Sparks
>> Filename        : draft-sparks-sip-3261-interop-statement-00.txt
>> Pages           : 9
>> Date            : 2009-03-02
>>
>> This document captures an outline of the interoperability statements
>> that will be collected to construct an interoperability report for
>> RFC 3261.  The outline is stil under review and should not be treated
>> as complete, but will drive data collection at upcoming
>> interoperability events.
>>
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-sparks-sip-3261-interop-statement-00.txt
>>
>> Internet-Drafts are also available by anonymous FTP at:
>> ftp://ftp.ietf.org/internet-drafts/
>>
>> Below is the data which will enable a MIME compliant mail reader
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>> Internet-Draft.
> 
> ------------------------------------------------------------------------
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> 
> 
> ------------------------------------------------------------------------
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> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
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> Use sipping@ietf.org for new developments on the application of sip

-- 
Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
Cisco Fellow                                   Iselin, NJ 08830
Cisco, Voice Technology Group
jdrosen@cisco.com
http://www.jdrosen.net                         PHONE: (408) 902-3084
http://www.cisco.com

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Subject: Re: [Sip] Fwd: I-D Action:draft-sparks-sip-invfix-03.txt
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Thanks, Robert. Comments below:

> It modifies state transitions in the INVITE server state
>    machine to absorb retransmissions of the INVITE request after
>    encountering a unrecoverable transport error when sending a response.

this little tidbit is not mentioned in section 3, whereas the other 
parts of this section (section 4) are mentioned. I think a paragraph 
describing this error is appropriate.

> Implementations strictly conformant to [RFC3261] will process
>    retransmitted initial INVITE requests as new requests.  Proxies may
>    forward them to different locations than the original.  Proxies may
>    also be used as anonymizing forwarders of bulk traffic.

should mention the consequences of not addressing this transport-error 
issue in my comment above.

> 7.1.  UAS Impacts
> 
>    To allow a UAS to recognize retransmissions of an INVITE as
>    retransmissions instead of new requests, a new state, "Accepted", is
>    added to the INVITE server transaction state machine. 

this change is not JUST for a UAS - its also for proxies implementing 
the server transaction machine, right? Is there a reason this is worded 
for UAS?

> Timer L reflects the amount of time the TU will wait to
>    receive an ACK for the 2xx it is emitting before considering the
>    transaction failed.

but an ACK-200 is not part of this transaction...


also, I think its worth at least noting that this change does not 100% 
solve the problem. If the UAC and any server transaction between it, and 
the UAS, use a different value of T1, there can still be a problem. In 
particular if the value of T1 at the UAC is larger than the value at any 
of the downstream server transactions, the server transaction may be 
destroyed while the UAC remains in the calling state.

> UAC Impacts
> 
>    In order to correctly distinguish retransmissions of 2xx responses
>    from stray 2xx responses, the INVITE client state machine is modified
>    to not transition immediately to "Terminated" on receipt of a 2xx
>    response.  

as above, this is not specific to UAC and covers proxy/b2b client 
transactions too, no?

> Proxy Considerations
> 
>    A direct consequence of the change to the UAC state machine is that a
>    transaction-stateful proxy will not foward any stray INVITE
>    responses.

It is not a consequence of the new state machines - but it is a change 
in behavior we are making.


Meta-comment: A drawback of this change is that proxies and b2bua will 
now hold transaction state for 64*t1 seconds after the transaction is 
done, even for calls that set up immediately. In some implementations 
this may be a non-trivial increase in memory usage, and there may be 
other impacts (for example if an implementation had a non-optimized 
transaction matching mechanism which only worked well with a small 
number of transaction machines e.g., a linear search). I think this 
implication needs to be called out more strongly. You mention it as a 
security vulnerability, but its not just a security issue.

-Jonathan R.

Robert Sparks wrote:
> This got a fairly big touch due to better handling of transport error 
> conditions. Please review this carefully.
> 
> RjS
> 
> Begin forwarded message:
> 
>> *From: *Internet-Drafts@ietf.org <mailto:Internet-Drafts@ietf.org>
>> *Date: *March 2, 2009 8:45:01 AM CST
>> *To: *i-d-announce@ietf.org <mailto:i-d-announce@ietf.org>
>> *Subject: **I-D Action:draft-sparks-sip-invfix-03.txt *
>> *Reply-To: *internet-drafts@ietf.org <mailto:internet-drafts@ietf.org>
>>
>> A New Internet-Draft is available from the on-line Internet-Drafts 
>> directories.
>>
>> Title           : Correct transaction handling for 200 responses to 
>> Session Initiation Protocol INVITE requests
>> Author(s)       : R. Sparks
>> Filename        : draft-sparks-sip-invfix-03.txt
>> Pages           : 20
>> Date            : 2009-03-02
>>
>> This document normatively updates RFC 3261, the Session Initiation
>> Protocol (SIP), to address an error in the specified handling of
>> success (200 class) responses to INVITE requests.  Elements following
>> RFC 3261 exactly will misidentify retransmissions of the request as a
>> new, unassociated, request.  The correction involves modifying the
>> INVITE transaction state machines.  The correction also changes the
>> way responses that cannot be matched to an existing transaction are
>> handled to address a security risk.
>>
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-sparks-sip-invfix-03.txt
>>
>> Internet-Drafts are also available by anonymous FTP at:
>> ftp://ftp.ietf.org/internet-drafts/
>>
>> Below is the data which will enable a MIME compliant mail reader
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> 
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> 
> 
> ------------------------------------------------------------------------
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> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

-- 
Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
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http://www.jdrosen.net                         PHONE: (408) 902-3084
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My main comment at this time, is that this bis document doesn't seem 
much different from the original. Section 10 calls out a bunch of things 
that I would put in the 'minor' category - certainly not worth a 
wholesale bis document if this is all we do.

I think it'd be worth having a high level discussion on the changes we'd 
like to make to this document, and assess whether there are enough to 
warrant a bis. For example, many have complained about the complexity of 
HI; are there changes meant to help that? So far, frankly, the few 
comments on this draft have been suggestions for additions for the most 
part, and NOT simplifications. Is this meant as an omnibus for a bunch 
of extensions to HI?

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
Cisco Fellow                                   Iselin, NJ 08830
Cisco, Voice Technology Group
jdrosen@cisco.com
http://www.jdrosen.net                         PHONE: (408) 902-3084
http://www.cisco.com

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Subject: [Sip] comments on draft-kaplan-sip-secure-call-id-00.txt
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Few comments.

While this is not a bad idea, its not clear what the point of it is, 
given draft-kaplan-sip-session-id-01.txt suggests anyway a different 
header field for dialog identification through b2bua. Indeed the 
introduction in this document still talks about dialog correlation. Is 
this meant as a way to fix up existing call-id using extensions in a 
backwards compatible way, while session-id is for future things?

I do apologize I did not follow much of the discussion at the end of 
last year if this was discussed then.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
Cisco Fellow                                   Iselin, NJ 08830
Cisco, Voice Technology Group
jdrosen@cisco.com
http://www.jdrosen.net                         PHONE: (408) 902-3084
http://www.cisco.com

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Subject: [Sip] comments on draft-kaplan-sip-session-id-01
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I think we all agree that it would be nice if there was an identifier 
that worked through b2bua, which we could use to identify dialogs.

However, I think the problem is far more complicated than this document 
talks about. The focus is on making it work through SBC type of boxes 
which aren't doing a lot of fancy call control, but are rewriting 
call-id and doing security things. There, I think the problem is easy.

However, there are lots of cases where more complicated dialog 
manipulation mucks this up in a really horrible way. In particular, 
enterprise features like transfer, park/pickup, and so on, are often 
implemented by means of re-INVITEs on dialogs all anchored in the 
IP-PBX. For example, consider the following transfer implementation:

A calls B. A calls C. Both calls are routed through an IP-PBX, which 
acts as a b2bua, and so we end up with something which looks like this:

  Please view in a fixed-width font such as Courier.







                     +------------+       S1
                S1   |            | ------------    B
             --------|            |
      A         S2   |   server   |
             --------|            |       S2
                     |            | -------------   C
                     |            |
                     +------------+

Where S1 and S2 are the two session IDs. So far, so good. Now, A 
transfers the call to B. A sends a REFER on dialog S1. Like it or not, 
due to frequent REFER/transfer interop issues, this is sometimes 
implemented as a re-INVITE from the server, connecting B and C's media 
streams together, and then disconnecting A, so that you end up with:

  Please view in a fixed-width font such
               as Courier.







         +------------+       S1
         |            | ------------    B
         |            |
         |   server   |
         |            |       S2
         |            | -------------   C
         |            |
         +------------+

Now B and C are talking to each other. What is the session ID for this 
dialog? B and C were both originally the UAS on different dialogs with 
different session IDs!

The problem happens more generally for any 3pcc operation, and the 
meta-issue is how to come up with a clear notion of who 'owns' the 
sessionID creation and destruction in cases where calls are transferred, 
moved, conferenced, parked and picked up all over the place.

As such, a true comprehensive solution to this problem is much more 
complicated than just 'tell the SBCs to stop mucking with this header'. 
It would need to account for a wider range of 3pcc and other dialog 
manipulations which occur in the wild. Or, we could try to scope this in 
a more limited way, focusing on the basic A-calls-B case and solve just 
the SBC problem.

Which is it?

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
Cisco Fellow                                   Iselin, NJ 08830
Cisco, Voice Technology Group
jdrosen@cisco.com
http://www.jdrosen.net                         PHONE: (408) 902-3084
http://www.cisco.com

From rjsparks@nostrum.com  Sat Mar 21 08:34:30 2009
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Subject: Re: [Sip] Fwd: I-D Action:draft-sparks-sip-3261-interop-statement-00.txt
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On Mar 21, 2009, at 7:28 AM, Jonathan Rosenberg wrote:

> Thanks, Robert.
>
> Before commenting on the specific list of things, my meta-question  
> is what exactly we plan to do with this list. I think we have agreed  
> at previous meetings that we were not currently planning on taking  
> SIP to draft. Do you think that has changed?
>
> The document does say:
>
>> While not required by process until
>>   advancing to Draft, such a report would provide useful  
>> information to
>>   guide where we focus our energy maintaing the standard at Proposed.
>
> its not clear to me, what this means. Will we take this list, run  
> tests at sipit, and based on that define updates to 3261 to clarify?  
> Does this imply that there are tests we're NOT running which we  
> should be, that this draft helps us sort out?

Yes - what I'm currently planning to do is start to collect interop  
statements on this list, evolve the list over time, and use that body  
of information to further inform what's implemented, what's giving  
people trouble, and what's not being used.

>
>
> Thanks,
> Jonathan R.
>
> Robert Sparks wrote:
>> Also split out of steps-to-draft
>> Begin forwarded message:
>>> *From: *Internet-Drafts@ietf.org <mailto:Internet-Drafts@ietf.org>
>>> *Date: *March 2, 2009 8:30:02 AM CST
>>> *To: *i-d-announce@ietf.org <mailto:i-d-announce@ietf.org>
>>> *Subject: **I-D Action:draft-sparks-sip-3261-interop- 
>>> statement-00.txt *
>>> *Reply-To: *internet-drafts@ietf.org <mailto:internet-drafts@ietf.org 
>>> >
>>>
>>> A New Internet-Draft is available from the on-line Internet-Drafts  
>>> directories.
>>>
>>> Title           : RFC3261 Interop Statement
>>> Author(s)       : R. Sparks
>>> Filename        : draft-sparks-sip-3261-interop-statement-00.txt
>>> Pages           : 9
>>> Date            : 2009-03-02
>>>
>>> This document captures an outline of the interoperability statements
>>> that will be collected to construct an interoperability report for
>>> RFC 3261.  The outline is stil under review and should not be  
>>> treated
>>> as complete, but will drive data collection at upcoming
>>> interoperability events.
>>>
>>> A URL for this Internet-Draft is:
>>> http://www.ietf.org/internet-drafts/draft-sparks-sip-3261-interop-statement-00.txt
>>>
>>> Internet-Drafts are also available by anonymous FTP at:
>>> ftp://ftp.ietf.org/internet-drafts/
>>>
>>> Below is the data which will enable a MIME compliant mail reader
>>> implementation to automatically retrieve the ASCII version of the
>>> Internet-Draft.
>> ------------------------------------------------------------------------
>>> _______________________________________________
>>> I-D-Announce mailing list
>>> I-D-Announce@ietf.org
>>> https://www.ietf.org/mailman/listinfo/i-d-announce
>>> Internet-Draft directories: http://www.ietf.org/shadow.html
>>> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>> ------------------------------------------------------------------------
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>
> -- 
> Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
> Cisco Fellow                                   Iselin, NJ 08830
> Cisco, Voice Technology Group
> jdrosen@cisco.com
> http://www.jdrosen.net                         PHONE: (408) 902-3084
> http://www.cisco.com


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On Mar 21, 2009, at 7:59 AM, Jonathan Rosenberg wrote:

> Thanks, Robert. Comments below:
>
>> It modifies state transitions in the INVITE server state
>>   machine to absorb retransmissions of the INVITE request after
>>   encountering a unrecoverable transport error when sending a  
>> response.
>
> this little tidbit is not mentioned in section 3, whereas the other  
> parts of this section (section 4) are mentioned. I think a paragraph  
> describing this error is appropriate.

Edit-itis. I will add such a paragraph.
>
>
>> Implementations strictly conformant to [RFC3261] will process
>>   retransmitted initial INVITE requests as new requests.  Proxies may
>>   forward them to different locations than the original.  Proxies may
>>   also be used as anonymizing forwarders of bulk traffic.
>
> should mention the consequences of not addressing this transport- 
> error issue in my comment above.

ok

>
>
>> 7.1.  UAS Impacts
>>   To allow a UAS to recognize retransmissions of an INVITE as
>>   retransmissions instead of new requests, a new state, "Accepted",  
>> is
>>   added to the INVITE server transaction state machine.
>
> this change is not JUST for a UAS - its also for proxies  
> implementing the server transaction machine, right? Is there a  
> reason this is worded for UAS?

Hmm. I'm going to have to go read this through fairly carefully again.  
The change is to the client and server transaction state machines. It  
may be that the
prose started to erroneously use UAC and UAS as synonyms for those  
state machines. Let me look through it and propose correction.

>
>
>> Timer L reflects the amount of time the TU will wait to
>>   receive an ACK for the 2xx it is emitting before considering the
>>   transaction failed.
>
> but an ACK-200 is not part of this transaction...
Right, the TU is tying those two transactions together. This is the  
period of time that the mentioned TU can be handing down 200s for  
retransmission.
Given your question, it looks like I should change this motivation  
sentence to say that instead?
>
>
>
> also, I think its worth at least noting that this change does not  
> 100% solve the problem. If the UAC and any server transaction  
> between it, and the UAS, use a different value of T1, there can  
> still be a problem. In particular if the value of T1 at the UAC is  
> larger than the value at any of the downstream server transactions,  
> the server transaction may be destroyed while the UAC remains in the  
> calling state.

I agree (and think there are other places where we should more  
strongly point out the pitfalls of changing T1 only in one part of a  
system).
Could you propose text for this instance though?

>
>
>> UAC Impacts
>>   In order to correctly distinguish retransmissions of 2xx responses
>>   from stray 2xx responses, the INVITE client state machine is  
>> modified
>>   to not transition immediately to "Terminated" on receipt of a 2xx
>>   response.
>
> as above, this is not specific to UAC and covers proxy/b2b client  
> transactions too, no?
>
>> Proxy Considerations
>>   A direct consequence of the change to the UAC state machine is  
>> that a
>>   transaction-stateful proxy will not foward any stray INVITE
>>   responses.
>
> It is not a consequence of the new state machines - but it is a  
> change in behavior we are making.
I understand your objection, and think this may be another aspect of  
unintentional conflation of UAS/UAC with server and client state  
machines. When I propose a correction as mentioned above, I'll capture  
this.
>
>
>
> Meta-comment: A drawback of this change is that proxies and b2bua  
> will now hold transaction state for 64*t1 seconds after the  
> transaction is done, even for calls that set up immediately. In some  
> implementations this may be a non-trivial increase in memory usage,  
> and there may be other impacts (for example if an implementation had  
> a non-optimized transaction matching mechanism which only worked  
> well with a small number of transaction machines e.g., a linear  
> search). I think this implication needs to be called out more  
> strongly. You mention it as a security vulnerability, but its not  
> just a security issue.
Where in the document would you think such a highlight would have the  
best impact?
>
>
> -Jonathan R.
>
> Robert Sparks wrote:
>> This got a fairly big touch due to better handling of transport  
>> error conditions. Please review this carefully.
>> RjS
>> Begin forwarded message:
>>> *From: *Internet-Drafts@ietf.org <mailto:Internet-Drafts@ietf.org>
>>> *Date: *March 2, 2009 8:45:01 AM CST
>>> *To: *i-d-announce@ietf.org <mailto:i-d-announce@ietf.org>
>>> *Subject: **I-D Action:draft-sparks-sip-invfix-03.txt *
>>> *Reply-To: *internet-drafts@ietf.org <mailto:internet-drafts@ietf.org 
>>> >
>>>
>>> A New Internet-Draft is available from the on-line Internet-Drafts  
>>> directories.
>>>
>>> Title           : Correct transaction handling for 200 responses  
>>> to Session Initiation Protocol INVITE requests
>>> Author(s)       : R. Sparks
>>> Filename        : draft-sparks-sip-invfix-03.txt
>>> Pages           : 20
>>> Date            : 2009-03-02
>>>
>>> This document normatively updates RFC 3261, the Session Initiation
>>> Protocol (SIP), to address an error in the specified handling of
>>> success (200 class) responses to INVITE requests.  Elements  
>>> following
>>> RFC 3261 exactly will misidentify retransmissions of the request  
>>> as a
>>> new, unassociated, request.  The correction involves modifying the
>>> INVITE transaction state machines.  The correction also changes the
>>> way responses that cannot be matched to an existing transaction are
>>> handled to address a security risk.
>>>
>>> A URL for this Internet-Draft is:
>>> http://www.ietf.org/internet-drafts/draft-sparks-sip-invfix-03.txt
>>>
>>> Internet-Drafts are also available by anonymous FTP at:
>>> ftp://ftp.ietf.org/internet-drafts/
>>>
>>> Below is the data which will enable a MIME compliant mail reader
>>> implementation to automatically retrieve the ASCII version of the
>>> Internet-Draft.
>> ------------------------------------------------------------------------
>>> _______________________________________________
>>> I-D-Announce mailing list
>>> I-D-Announce@ietf.org
>>> https://www.ietf.org/mailman/listinfo/i-d-announce
>>> Internet-Draft directories: http://www.ietf.org/shadow.html
>>> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>> ------------------------------------------------------------------------
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>
> -- 
> Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
> Cisco Fellow                                   Iselin, NJ 08830
> Cisco, Voice Technology Group
> jdrosen@cisco.com
> http://www.jdrosen.net                         PHONE: (408) 902-3084
> http://www.cisco.com


From HKaplan@acmepacket.com  Sat Mar 21 15:50:13 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Jonathan Rosenberg <jdrosen@cisco.com>, IETF SIP List <sip@ietf.org>
Date: Sat, 21 Mar 2009 18:50:56 -0400
Thread-Topic: [Sip] comments on draft-kaplan-sip-secure-call-id-00.txt
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Hey Jonathan,
Thanks for reviewing the draft!  This draft is just about making UA's send =
actual Call-ID values which do not reveal sensitive information, so that mi=
ddleboxes are not made to change them purely for security purposes.

The other draft, session-id, was changed - rev 02 is the latest - and that =
rev is only about troubleshooting not dialog matching. This secure-call-id =
draft is the one and only thing I propose for dialog matching purposes (and=
 it is the actual Call-ID header field value).

-hadriel

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Jonathan Rosenberg
> Sent: Saturday, March 21, 2009 11:18 AM
> To: IETF SIP List
> Subject: [Sip] comments on draft-kaplan-sip-secure-call-id-00.txt
>=20
> Few comments.
>=20
> While this is not a bad idea, its not clear what the point of it is,
> given draft-kaplan-sip-session-id-01.txt suggests anyway a different
> header field for dialog identification through b2bua. Indeed the
> introduction in this document still talks about dialog correlation. Is
> this meant as a way to fix up existing call-id using extensions in a
> backwards compatible way, while session-id is for future things?
>=20
> I do apologize I did not follow much of the discussion at the end of
> last year if this was discussed then.
>=20
> -Jonathan R.
> --
> Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
> Cisco Fellow                                   Iselin, NJ 08830
> Cisco, Voice Technology Group
> jdrosen@cisco.com
> http://www.jdrosen.net                         PHONE: (408) 902-3084
> http://www.cisco.com
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

From HKaplan@acmepacket.com  Sat Mar 21 16:03:49 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Jonathan Rosenberg <jdrosen@cisco.com>, IETF SIP List <sip@ietf.org>
Date: Sat, 21 Mar 2009 19:04:32 -0400
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Hi Jonathan,
Yeah there was some email discussion about the various cases whereby the se=
ssion-id wouldn't match anymore.  There are many, no doubt.  My point back =
then was that's ok: this is fixing cases that are broken, and not breaking =
cases that are working, so it's improving the situation even if not with 10=
0% coverage.

But I got the sense several folks didn't like the idea of not fixing 100% o=
f the cases, so I updated this draft a couple weeks ago to rev-02, and remo=
ved all the dialog-matching stuff and just made session-id for troubleshoot=
ing purposes.  I submitted secure-call-id to handle the Call-ID changing is=
sues separately, in a different way.

-hadriel

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Jonathan Rosenberg
> Sent: Saturday, March 21, 2009 11:29 AM
> To: IETF SIP List
> Subject: [Sip] comments on draft-kaplan-sip-session-id-01
>=20
> I think we all agree that it would be nice if there was an identifier
> that worked through b2bua, which we could use to identify dialogs.
>=20
> However, I think the problem is far more complicated than this document
> talks about. The focus is on making it work through SBC type of boxes
> which aren't doing a lot of fancy call control, but are rewriting
> call-id and doing security things. There, I think the problem is easy.
>=20
> However, there are lots of cases where more complicated dialog
> manipulation mucks this up in a really horrible way. In particular,
> enterprise features like transfer, park/pickup, and so on, are often
> implemented by means of re-INVITEs on dialogs all anchored in the
> IP-PBX. For example, consider the following transfer implementation:
>=20
> A calls B. A calls C. Both calls are routed through an IP-PBX, which
> acts as a b2bua, and so we end up with something which looks like this:
>=20
>   Please view in a fixed-width font such as Courier.
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>                      +------------+       S1
>                 S1   |            | ------------    B
>              --------|            |
>       A         S2   |   server   |
>              --------|            |       S2
>                      |            | -------------   C
>                      |            |
>                      +------------+
>=20
> Where S1 and S2 are the two session IDs. So far, so good. Now, A
> transfers the call to B. A sends a REFER on dialog S1. Like it or not,
> due to frequent REFER/transfer interop issues, this is sometimes
> implemented as a re-INVITE from the server, connecting B and C's media
> streams together, and then disconnecting A, so that you end up with:
>=20
>   Please view in a fixed-width font such
>                as Courier.
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>          +------------+       S1
>          |            | ------------    B
>          |            |
>          |   server   |
>          |            |       S2
>          |            | -------------   C
>          |            |
>          +------------+
>=20
> Now B and C are talking to each other. What is the session ID for this
> dialog? B and C were both originally the UAS on different dialogs with
> different session IDs!
>=20
> The problem happens more generally for any 3pcc operation, and the
> meta-issue is how to come up with a clear notion of who 'owns' the
> sessionID creation and destruction in cases where calls are transferred,
> moved, conferenced, parked and picked up all over the place.
>=20
> As such, a true comprehensive solution to this problem is much more
> complicated than just 'tell the SBCs to stop mucking with this header'.
> It would need to account for a wider range of 3pcc and other dialog
> manipulations which occur in the wild. Or, we could try to scope this in
> a more limited way, focusing on the basic A-calls-B case and solve just
> the SBC problem.
>=20
> Which is it?
>=20
> -Jonathan R.
>=20
> --
> Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
> Cisco Fellow                                   Iselin, NJ 08830
> Cisco, Voice Technology Group
> jdrosen@cisco.com
> http://www.jdrosen.net                         PHONE: (408) 902-3084
> http://www.cisco.com
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

From john.elwell@siemens.com  Sun Mar 22 08:40:29 2009
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From: "Elwell, John" <john.elwell@siemens.com>
To: Adam Roach <adam@nostrum.com>, sip@ietf.org
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Adam,

Thanks for producing this draft, which should set a good example for
revisions of the other SIPCORE RFCs.

I found little to comment on, apart from the following:

In 4.5.1:
"Notifiers MUST implement the GRUU extension defined in
   [I-D.ietf-sip-gruu], and MUST use a GRUU as their local target."

1. Is this just for a SUBSCRIBE-initiated dialog or for any dialog?

2. Should this be package-dependent? For example, there may be some
event packages for which there is no need for subscribing to a resource
on the same device.

3. What if the registrar doesn't support GRUU, and therefore the
notifier is unable to obtain a GRUU?

And by the way, in appendix C I didn't see a mention of changes relating
to the use of GRUU.

John

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From: "Francois Audet" <audet@nortel.com>
To: Jonathan Rosenberg <jdrosen@cisco.com>, SIP IETF <sip@ietf.org>, Mary BARNES <Barnes@zcars04e.nortel.com>, "B601:EXCH]"@zcars04e.nortel.com
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Subject: Re: [Sip] comments on draft-barnes-sip-rfc4244bis-00.txt
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Jonathan, you are correct, we have not done too many changes so far.
In part because we are waiting on target-URI, in part because we didn't have
enough time. And mostly, because we'd like to get agreement on how to
proceed before. I'd like to discuss in the meeting.

The problems with HI today, IMHO, are the following:

- In order to make it work, today, people make "assumptions" about HI to
  be able to use it. Specifically, it is assumed that the first
  retargeting is listed, and so are the 2 last ones. This is then
  used by implementations as part of service logic and/or mapping to
  PSTN information (Original Called Number, Redirecting Number, Last
  Called Number). Unfortunately, HI (the spec) makes no such garantee:
  current usage however does. Furthermore, it is assumed that there are
  no recorded "routing" entries before the first one or between the last
  retarget and the final contact. One "simplification" that I would like
  to see is rules that makes the outcome predictable. First, I would like
  to mandate that at a minimum those entries (first, last two) are NEVER
  removed by proxies, and second, that retargets proper are labeled as
  such (as per target-URI discussion). That would greatly simplify
  History-Info.

- There is a bug in ABNF. Corrected in current draft.

- There is a gratuitous requirement that TLS be used on each hop for
  HI. I'd like to revisit that. Nobody enforces that, and I see no reason
  why HI would be treated differently than other similar headers (Via,
  Record-Route).

- The examples are all wrong... They are all using Strict Routing. The
  chances of having an implementation using HI and Strict Routing in real
  life are close to zero. It makes the reading more complex, and makes
  HI look like some form of Record-Routing.

- There have been some editorial comments on text being repetitive and
  wordy, and some proposals for simplification.

- Finally, and importantly, the idea would be to integrate all the
  changes from target-URI, including the mid-path retarget (as per point
  one above), and the last-Contact replacement.



On Mar21 2009 08:07 , "Jonathan Rosenberg" <jdrosen@cisco.com> wrote:

> My main comment at this time, is that this bis document doesn't seem
> much different from the original. Section 10 calls out a bunch of things
> that I would put in the 'minor' category - certainly not worth a
> wholesale bis document if this is all we do.
> 
> I think it'd be worth having a high level discussion on the changes we'd
> like to make to this document, and assess whether there are enough to
> warrant a bis. For example, many have complained about the complexity of
> HI; are there changes meant to help that? So far, frankly, the few
> comments on this draft have been suggestions for additions for the most
> part, and NOT simplifications. Is this meant as an omnibus for a bunch
> of extensions to HI?
> 
> Thanks,
> Jonathan R.


From christer.holmberg@ericsson.com  Sun Mar 22 09:48:46 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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Hi,=20

>Between Minneapolis and now, this document has become a lot more
confusing, which I think=20
>bodes pretty ill for its hopes of being deployed.

I can tell you it will be deployed, but let's focus on your comments :)

>For example, in section 5:
>
>   If the received initial request contains an 199 option tag, the UAS
>   SHOULD NOT send a 199 response for a dialog on which it intends to
>   send a final response...
>
>This reads as if sending a 199 response somehow absolves a UAS of
sending a final response.=20
>Certainly this is not the intention of the draft (at least, not
according to the text in=20
>section 7). Section 5 needs to be re-phrased to indicate what is
actually intended here.

I undrstand what you mean. The text can be read in a way that a UA could
use 199 instead of the final response. That is certainly not the
intention. The idea was to cover B2BUAs, which similar to forking
proxies may not send final response for all dialogs.

>In fact, on that topic, it's radically unclear to me how UASes fit into
the entire scheme at=20
>all -- my understanding is that the 199 response is only sent by
forking proxies.

No, a UAS (both a UA and B2BUA) can also send it. This was discussed
before, and there were no objections to allowing the UAS to send it.

I do agree that end terminals normally may not send it, but you may have
network entities (applicatoin servers, B2BUAs generating multiple
dialogs etc) which will.

>In section 6:
>
>   When a forking proxy receives a non-2xx final response that it
>   recognizes as terminating one or more early dialogs, if the proxy
>   does not intend to forward the final response immediately (in
>   accordance with rules for a forking proxy) and the UAC has indicated
>   support for the 199 response code, the proxy SHOULD generate and
send
>   a 199 response upstream for each early dialog terminated on the
>   downstream side by the non-2xx final response, except for any early
>   dialog for which the proxy has previously received and forwarded a
>   199 response.
>
>What? On my best run at reading that, I made it to about the 81st word
before the first=20
>predicate clause fell out of my head. If it's really as complicated as
the structure of this=20
>sentence makes it seem, we need a flowchart or some other simplifying
mechanism.

I don't understand what is unclear, eventhough the wording and style may
have some 3GPP influences :) Sure, there may be 81 words, but I think it
describes what the proxy does in a rather straight forward way.

I also don't see how a flowchart would make this text more clear,
because the actions depends on what has/has not happened before. But, if
you have a proposal on how to make the text more simple I am happy to
incorporate that.

Regards,

Christer

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In the hopes of making the discussion as short as possible in SIPPING:

This is what we have consensus on the list:
1. Change all occurrences of "dialog mumble" with "dialog parameters"

2. Fix Table 1 to remove archaic references to 2543

3. Clarify language for overlapping INFOs and the "nil" package



Open issues:
We all agree on what the text says for the INVITE, UPDATE, and ACK  
methods, as well as the 101-299 (stuff counts) and 300-699 (something  
is broken, so it does not matter) response codes.

It is probably sensible to exclude 100 from Recv-Info processing.   
Work for you?

What about PRACK/PRACK 2xx?  My vote is they count like everything  
else.  What about you?

Likewise, what about REFER/REFER 2xx?  My vote here is INFO packages  
would be nonsensical during a refer, but perhaps you have a use case?   
If so, they work like everything else.

The current text is pretty adamant that a missing Recv-Info means you  
do not (want to) understand Info Packages.  If you start with a Recv- 
Info and you later drop the Recv-Info header, all bets are off.  Are  
you happy with that?  Since we do have Recv-Info: nil, we do have the  
opportunity to say we only change the Info Package set if a Recv-Info  
header is present.  What do you think?

Responses to the list now will hopefully make the discussion go swiftly.

Likewise, if I have missed something, please bring it up now, NOT at  
the meeting.  Recall this draft has finished WGLC.

See you in SF, if you will be in SF; on Jabber if you will be remote;  
or via e-mail if you will be asynchronous.
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From john.elwell@siemens.com  Sun Mar 22 16:20:02 2009
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Date: Sun, 22 Mar 2009 23:20:44 +0000
From: "Elwell, John" <john.elwell@siemens.com>
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To: Francois Audet <audet@nortel.com>, Jonathan Rosenberg <jdrosen@cisco.com>,  SIP IETF <sip@ietf.org>, Mary BARNES <Barnes@zcars04e.nortel.com>, "B601:EXCH]"@zcars04e.nortel.com
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Subject: Re: [Sip] comments on draft-barnes-sip-rfc4244bis-00.txt
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Part of the complexity is in the indexing mechanism. Do we really need
this?

John=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Francois Audet
> Sent: 22 March 2009 16:11
> To: Jonathan Rosenberg; SIP IETF; Mary BARNES;=20
> "B601:EXCH]"@zcars04e.nortel.com
> Subject: Re: [Sip] comments on draft-barnes-sip-rfc4244bis-00.txt
>=20
> Jonathan, you are correct, we have not done too many changes so far.
> In part because we are waiting on target-URI, in part because=20
> we didn't have
> enough time. And mostly, because we'd like to get agreement on how to
> proceed before. I'd like to discuss in the meeting.
>=20
> The problems with HI today, IMHO, are the following:
>=20
> - In order to make it work, today, people make "assumptions"=20
> about HI to
>   be able to use it. Specifically, it is assumed that the first
>   retargeting is listed, and so are the 2 last ones. This is then
>   used by implementations as part of service logic and/or mapping to
>   PSTN information (Original Called Number, Redirecting Number, Last
>   Called Number). Unfortunately, HI (the spec) makes no such garantee:
>   current usage however does. Furthermore, it is assumed that=20
> there are
>   no recorded "routing" entries before the first one or=20
> between the last
>   retarget and the final contact. One "simplification" that I=20
> would like
>   to see is rules that makes the outcome predictable. First,=20
> I would like
>   to mandate that at a minimum those entries (first, last=20
> two) are NEVER
>   removed by proxies, and second, that retargets proper are labeled as
>   such (as per target-URI discussion). That would greatly simplify
>   History-Info.
>=20
> - There is a bug in ABNF. Corrected in current draft.
>=20
> - There is a gratuitous requirement that TLS be used on each hop for
>   HI. I'd like to revisit that. Nobody enforces that, and I=20
> see no reason
>   why HI would be treated differently than other similar headers (Via,
>   Record-Route).
>=20
> - The examples are all wrong... They are all using Strict Routing. The
>   chances of having an implementation using HI and Strict=20
> Routing in real
>   life are close to zero. It makes the reading more complex, and makes
>   HI look like some form of Record-Routing.
>=20
> - There have been some editorial comments on text being repetitive and
>   wordy, and some proposals for simplification.
>=20
> - Finally, and importantly, the idea would be to integrate all the
>   changes from target-URI, including the mid-path retarget=20
> (as per point
>   one above), and the last-Contact replacement.
>=20
>=20
>=20
> On Mar21 2009 08:07 , "Jonathan Rosenberg" <jdrosen@cisco.com> wrote:
>=20
> > My main comment at this time, is that this bis document doesn't seem
> > much different from the original. Section 10 calls out a=20
> bunch of things
> > that I would put in the 'minor' category - certainly not worth a
> > wholesale bis document if this is all we do.
> >=20
> > I think it'd be worth having a high level discussion on the=20
> changes we'd
> > like to make to this document, and assess whether there are=20
> enough to
> > warrant a bis. For example, many have complained about the=20
> complexity of
> > HI; are there changes meant to help that? So far, frankly, the few
> > comments on this draft have been suggestions for additions=20
> for the most
> > part, and NOT simplifications. Is this meant as an omnibus=20
> for a bunch
> > of extensions to HI?
> >=20
> > Thanks,
> > Jonathan R.
>=20
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

From adam@nostrum.com  Sun Mar 22 16:23:21 2009
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Cc: sip@ietf.org
Subject: Re: [Sip] Comment on draft-roach-sipcore-rfc3265bis-00
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Elwell, John wrote:
> I found little to comment on, apart from the following:
>
> In 4.5.1:
> "Notifiers MUST implement the GRUU extension defined in
>    [I-D.ietf-sip-gruu], and MUST use a GRUU as their local target."
>
> 1. Is this just for a SUBSCRIBE-initiated dialog or for any dialog?
>   

The intention is that any UA capable of sending a NOTIFY message in any 
context will use a GRUU for all dialogs. This mechanism is to be used 
instead of dialog re-use. The most common case of dialog re-use right 
now is sending a SUBSCRIBE down an existing dialog that was initially 
established by an INVITE. So, for current use cases, using GRUU in 
INVITE dialogs is the most important thing. For future-proofing, we need 
to require GRUU for all dialogs.

> 2. Should this be package-dependent? For example, there may be some
> event packages for which there is no need for subscribing to a resource
> on the same device.
>   

I think it's likely to be too difficult to guess, ahead of time, which 
packages will never need to target the same device as another dialog. 
It's far too easy to envision a circumstance in which we say, "I can't 
think of any reason someone would need to target a specific device for 
package foo," only to discover a really clever usage several years later 
that we have accidentally precluded.

> 3. What if the registrar doesn't support GRUU, and therefore the
> notifier is unable to obtain a GRUU?
>   

Note, first of all, that this constraint is placed on notifiers, and not 
on subscribers. For the vast majority of notifiers, a self-made GRUU 
(which requires no registrar support) is appropriate.

However, your point is taken, and we should discuss circumstances in 
which registering clients that send NOTIFY messages request a GRUU from 
their registrar, yet receive none. Because of the true interoperability 
nightmare that has resulted from dialog sharing, my suggestion would be 
to disable notifier functionality on those endpoints under such 
circumstances (perhaps with an implementation option to fall back to 
3265 behavior for the truly masochistic).

> And by the way, in appendix C I didn't see a mention of changes relating
> to the use of GRUU.
>   

Sorry about that. My intention was to include that as part of C.1.21, 
but I left it out.

/a

From john.elwell@siemens.com  Sun Mar 22 16:29:30 2009
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From: "Elwell, John" <john.elwell@siemens.com>
In-reply-to: A <CA9998CD4A020D418654FCDEF4E707DF0B16805F@esealmw113.eemea.ericsson.se>
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=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Christer Holmberg
> Sent: 22 March 2009 16:50
> To: Adam Roach; SIP WG
> Subject: Re: [Sip] draft-ietf-sip-199-06
>=20
>=20
> Hi,=20
>=20
> >Between Minneapolis and now, this document has become a lot more
> confusing, which I think=20
> >bodes pretty ill for its hopes of being deployed.
>=20
> I can tell you it will be deployed, but let's focus on your=20
> comments :)
>=20
> >For example, in section 5:
> >
> >   If the received initial request contains an 199 option=20
> tag, the UAS
> >   SHOULD NOT send a 199 response for a dialog on which it intends to
> >   send a final response...
> >
> >This reads as if sending a 199 response somehow absolves a UAS of
> sending a final response.=20
> >Certainly this is not the intention of the draft (at least, not
> according to the text in=20
> >section 7). Section 5 needs to be re-phrased to indicate what is
> actually intended here.
>=20
> I undrstand what you mean. The text can be read in a way that=20
> a UA could
> use 199 instead of the final response. That is certainly not the
> intention. The idea was to cover B2BUAs, which similar to forking
> proxies may not send final response for all dialogs.
>=20
> >In fact, on that topic, it's radically unclear to me how=20
> UASes fit into
> the entire scheme at=20
> >all -- my understanding is that the 199 response is only sent by
> forking proxies.
>=20
> No, a UAS (both a UA and B2BUA) can also send it. This was discussed
> before, and there were no objections to allowing the UAS to send it.
>=20
> I do agree that end terminals normally may not send it, but=20
> you may have
> network entities (applicatoin servers, B2BUAs generating multiple
> dialogs etc) which will.
[JRE] The reason we need to cover B2BAUs is that a B2BUA may behave like
a proxy and withhold the sending of a final response until it receives
responses from other branches it has created. So can we not have the
same rule for any server (proxy or UAS), telling it what to do when a
final response is withheld? This might make it simpler. There would not
then be any need for a normal UAS to send 199.

>=20
> >In section 6:
> >
> >   When a forking proxy receives a non-2xx final response that it
> >   recognizes as terminating one or more early dialogs, if the proxy
> >   does not intend to forward the final response immediately (in
> >   accordance with rules for a forking proxy) and the UAC=20
> has indicated
> >   support for the 199 response code, the proxy SHOULD generate and
> send
> >   a 199 response upstream for each early dialog terminated on the
> >   downstream side by the non-2xx final response, except for=20
> any early
> >   dialog for which the proxy has previously received and forwarded a
> >   199 response.
> >
> >What? On my best run at reading that, I made it to about the=20
> 81st word
> before the first=20
> >predicate clause fell out of my head. If it's really as=20
> complicated as
> the structure of this=20
> >sentence makes it seem, we need a flowchart or some other simplifying
> mechanism.
>=20
> I don't understand what is unclear, eventhough the wording=20
> and style may
> have some 3GPP influences :) Sure, there may be 81 words, but=20
> I think it
> describes what the proxy does in a rather straight forward way.
>=20
> I also don't see how a flowchart would make this text more clear,
> because the actions depends on what has/has not happened=20
> before. But, if
> you have a proposal on how to make the text more simple I am happy to
> incorporate that.
[JRE] I agree with Adam that it is unclear. When trying to write down a
normative requirement that is dependent on multiple conditions, it often
helps to list the conditions in a bulleted list, e.g.:
If
- foo1
- and foo2
- and not foo3
then do bar.

John
>=20
> Regards,
>=20
> Christer
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

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From christer.holmberg@ericsson.com  Sun Mar 22 23:47:36 2009
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Thread-Topic: [Sip] draft-ietf-sip-199-06
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens.com>, "Adam Roach" <adam@nostrum.com>, "SIP WG" <sip@ietf.org>
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Subject: Re: [Sip] draft-ietf-sip-199-06
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Hi John,

>>For example, in section 5:
>>
>>   If the received initial request contains an 199 option tag, the UAS
>>   SHOULD NOT send a 199 response for a dialog on which it intends to
>>   send a final response...
>>
>>This reads as if sending a 199 response somehow absolves a UAS of =
sending a final response.=20
>>Certainly this is not the intention of the draft (at least, not =
according to the text in=20
>>section 7). Section 5 needs to be re-phrased to indicate what is =
actually intended here.
>=20
>I undrstand what you mean. The text can be read in a way that a UA =
could
>use 199 instead of the final response. That is certainly not the
>intention. The idea was to cover B2BUAs, which similar to forking
>proxies may not send final response for all dialogs.
>=20
>>In fact, on that topic, it's radically unclear to me how UASes fit =
into the entire scheme at=20
>>all -- my understanding is that the 199 response is only sent by =
forking proxies.
>=20
>No, a UAS (both a UA and B2BUA) can also send it. This was discussed
>before, and there were no objections to allowing the UAS to send it.
>=20
>I do agree that end terminals normally may not send it, but you may =
have
>network entities (applicatoin servers, B2BUAs generating multiple
>dialogs etc) which will.
>[JRE] The reason we need to cover B2BAUs is that a B2BUA may behave =
like
>a proxy and withhold the sending of a final response until it receives
>responses from other branches it has created. So can we not have the
>same rule for any server (proxy or UAS), telling it what to do when a
>final response is withheld? This might make it simpler. There would not
>then be any need for a normal UAS to send 199.

You can have network entities which act as "normal" UAS, which also want =
to send 199. They may e.g. have been configured to do so, since the =
forking proxy does not support 199. Etc.

And, the text already says that a UAS should not send a 199, unless it =
has been configured to do so. But, I don't think we should forbid it =
from doing so.


>>In section 6:
>>
>>   When a forking proxy receives a non-2xx final response that it
>>   recognizes as terminating one or more early dialogs, if the proxy
>>   does not intend to forward the final response immediately (in
>>   accordance with rules for a forking proxy) and the UAC has =
indicated
>>   support for the 199 response code, the proxy SHOULD generate and =
send
>>   a 199 response upstream for each early dialog terminated on the
>>   downstream side by the non-2xx final response, except for any early
>>   dialog for which the proxy has previously received and forwarded a
>>   199 response.
>>
>>What? On my best run at reading that, I made it to about the 81st word
>>before the first predicate clause fell out of my head. If it's really =
as=20
>>complicated as the structure of this sentence makes it seem, we need a =
flowchart or some other=20
>>simplifying mechanism.
>=20
>I don't understand what is unclear, eventhough the wording and style =
may
>have some 3GPP influences :) Sure, there may be 81 words, but I think =
it
>describes what the proxy does in a rather straight forward way.
>=20
>I also don't see how a flowchart would make this text more clear,
>because the actions depends on what has/has not happened=20
>before. But, if you have a proposal on how to make the text more simple =
I am happy to
>incorporate that.
>[JRE] I agree with Adam that it is unclear. When trying to write down a
>normative requirement that is dependent on multiple conditions, it =
often
>helps to list the conditions in a bulleted list, e.g.:
>If
>- foo1
>- and foo2
>- and not foo3
>then do bar.

I can try to do that, if people think it's more clear.

Regards,

Christer

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Date: Mon, 23 Mar 2009 09:51:02 +0100
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From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: SIP WG <sip@ietf.org>, Adam Roach <adam@nostrum.com>, "Elwell, John" <john.elwell@siemens.com>
Subject: Re: [Sip] draft-ietf-sip-199-06
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--0016364d2933c0ce6d0465c55db3
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Yes you should, people do think that ;-)

/Hans Erik van Elburg

On Mon, Mar 23, 2009 at 7:48 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> I can try to do that, if people think it's more clear.

--0016364d2933c0ce6d0465c55db3
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Yes you should, people do think that ;-)<br><br clear=3D"all">/Hans Erik va=
n Elburg<br>
<br><div class=3D"gmail_quote">On Mon, Mar 23, 2009 at 7:48 AM, Christer Ho=
lmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.c=
om">christer.holmberg@ericsson.com</a>&gt;</span> wrote:<br><blockquote cla=
ss=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, 204, 204); marg=
in: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
I can try to do that, if people think it&#39;s more clear.</blockquote></di=
v><br>

--0016364d2933c0ce6d0465c55db3--

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Subject: Re: [Sip] draft-ietf-sip-199-06
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Some text , as requested.

Christer Holmberg wrote:
...
> 
>> In section 6:
>>
>>   When a forking proxy receives a non-2xx final response that it
>>   recognizes as terminating one or more early dialogs, if the proxy
>>   does not intend to forward the final response immediately (in
>>   accordance with rules for a forking proxy) and the UAC has indicated
>>   support for the 199 response code, the proxy SHOULD generate and
> send
>>   a 199 response upstream for each early dialog terminated on the
>>   downstream side by the non-2xx final response, except for any early
>>   dialog for which the proxy has previously received and forwarded a
>>   199 response.
>>
...

My suggestion:

"When a forking proxy receives a non-2xx final response that it
  recognizes as terminating one or more early dialogs, it SHOULD generate
  and send a 199 response upstream for each such early dialog satisfying the
  following conditions:
  -- the proxy does not intend to forward the final response immediately
     (in accordance with rules for a forking proxy);
  -- the UAC has indicated support for the 199 response code;
  -- the proxy has not already received and forwarded a 199 response for
     that early dialogue."

From victor.pascual.avila@gmail.com  Mon Mar 23 06:17:09 2009
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On Mon, Mar 23, 2009 at 6:39 AM, Dan Wing <dwing@cisco.com> wrote:
> http://www.networkworld.com/news/2009/032309-skype-sip.html?hpg1=3Dbn

http://ecommmedia.com/blog/2009/03/mark-spencer-transcript.html
--=20
Victor Pascual =C3=81vila

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Subject: Re: [Sip] SIP INFO
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> We all agree on what the text says for the INVITE, UPDATE,=20
> and ACK methods, as well as the 101-299 (stuff counts) and=20
> 300-699 (something is broken, so it does not matter)=20
> response codes.

In case it matters, ACK can't be authenticated and introduces more race con=
ditions.

RFC 3261 did not define Supported and Allow to be optional within an ACK.  =
I mention it in case there were similar race condition and authentication c=
oncerns associated with using ACK to indicate/update what is supported and =
allowed.


> It is probably sensible to exclude 100 from Recv-Info processing.
> Work for you?

Yes.


> What about PRACK/PRACK 2xx?  My vote is they count like everything
> else.  What about you?

Because of race conditions, I prefer the header not be allowed.  However my=
 concerns would be partially alleviated if not required to include Recv-Inf=
o to avoid disabling support.

It has been awhile since I read RFC 3262 in depth.  If UAS sends INVITE 2xx=
 while awaiting PRACK, should it return PRACK 481 or PRACK 200 upon receipt=
 of PRACK?  If acceptable to return PRACK 200 during the race condition, sh=
ould the PRACK's Recv-Info be ignored if received before or after ACK?  If =
it returns PRACK 200 and UAS receives PRACK 200 after INVITE 200, should th=
e UAC ignore PRACK 200's Recv-Info?


> Likewise, what about REFER/REFER 2xx?  My vote here is INFO packages
> would be nonsensical during a refer, but perhaps you have a use case?
> If so, they work like everything else.

I have no preference, especially if not required to include Recv-Info to av=
oid disabling support.


> The current text is pretty adamant that a missing Recv-Info means you
> do not (want to) understand Info Packages.  If you start with a Recv-
> Info and you later drop the Recv-Info header, all bets are off.  Are
> you happy with that? =20

No.  It requires uselessly adding the header everywhere.  It increases race=
 condition complications.

It shouldn't be more mandatory everywhere than Supported and Allow.  I'm no=
t opposed to making it mandatory within INVITE, re-INVITE, and their 2xx; i=
t should at least be optional within UPDATE.  Concerning other locations, I=
'm not opposed to it being optional although it introduces more potential f=
or race conditions.


> Since we do have Recv-Info: nil, we do have the opportunity to=20
> say we only change the Info Package set if a Recv-Info header=20
> is present.  What do you think?

I like the suggested modification.

If retarget to a device not supporting the RFC, the modification potentiall=
y raises the need for a Supported option-tag unless Recv-Info remains manda=
tory (to avoid disabling) within a few locations.  However B2BUAs can still=
 work around the issue if the working group doesn't want to define an optio=
n-tag for the RFC.


> Likewise, if I have missed something, please bring it up now, NOT at
> the meeting.  Recall this draft has finished WGLC.

What was the outcome of INFO 489 response MUST trigger release of call disc=
ussion?

Thanks,
Brett


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Date: Mon, 23 Mar 2009 09:26:23 -0400
From: Paul Kyzivat <pkyzivat@cisco.com>
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Cc: sip <sip@ietf.org>
Subject: Re: [Sip] Policy to register media feature tags
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Inline...

Hans Erik van Elburg wrote:
> Responses inline...
> 
> /Hans Erik
> 
> Paul Kyzivat wrote:
>> On one side were those who wanted capabilities and preferences to be 
>> stated in terms of feature tags that are well known and orthogonal. On 
>> the other side were those who prefer to associate arbitrary names to 
>> collections of features and then negotiate on the basis of the names 
>> of those collections.
>>
>> Using the latter approach its possible that interoperation will fail 
>> because the parties don't share a common name for a collection of 
>> features even though they both possess the necessary features to 
>> interoperate.
> That is a feature.

*what* is a feature?

> This interoperability "problem" only occurs when the 
> originating party includes the header field parameters "require" and 
> "explicit" in the Accept-Contact header field containing the feature 
> tag. In this case the failure of the call when not all of the receiving  
> UA's support  the feature is intended!

Lets get tangible about this. (This is retracing old ground, but people 
sometimes have short memories.)

Suppose Alice calls Bob. Alice has the latest and greatest 3gpp phone 
with the capability to make audio/video calls, and would prefer to do so 
if possible. Bob has an open source softphone running on his PC. It is 
fully capable of audio/video operation using a wide source of codecs. He 
also has a plain vanilla audio phone registered to the same AOR.

Alice's phone will send an invite with offer containing audio and video 
media. It can also include callerprefs to bias the selection of device 
in favor of one that will support audio and video. It could do so a 
variety of different ways:
- indicate preference for audio and video
- indicate preference for the 3gpp "videophone" feature.
- indicate preference for audio, video, *and* 3gpp "videophone"

Assuming none of these *require* these features, the first and the last 
will both bias towards the softphone and result in an audio/video call. 
The middle will probably cause both of bob's phones to ring and 
audio/video will result if he happens to answer the softphone.

The situation is worse if Alice *requires* both audio and video, and as 
a result *requires* the 3gpp videophone feature.

The problem is one of orthogonality. the features should all be 
orthogonal to one another. If "3gpp videophone" encompasses "audio, 
video, and the 3gpp videophone look and feel", then it isn't orthogonal 
to the existing audio and video feature tags. If it only represents a 
preference for whatever extra goodies are provided by 3gpp over and 
above audio and video, then it may be a reasonable feature tag.

(Note that I have no knowledge of how 3gpp intends to name an use 
features. I'm just exploring the possibilities.)

>> I don't have a problem with independent groups defining new feature 
>> tags, as long as they are primitive and orthogonal to existing ones. 
>> But I do have a problem with defining feature tags that identify 
>> collections of features, especially when the mapping from the 
>> identifier to the collection of features is not public.
> Well who is going to determine what can be called a feature and what not.
> I also like to see where this principle is documented to apply to the 
> global tree.

IMO it was probably a mistake to use the existing feature tag mechanism 
for callee caps in the first place. While there are some simmilarities, 
the intentions are quite different. But we are where we are.

I don't know if we can impose additional rules on the global tree or 
not. I expect that imposing an orthogonality constraint might not be so 
controversial, and so *might* be possible. But the challenge may be in 
*defining* orthogonality in an application-independent way. It would be 
easier to do that in the context of sip, though still not trivial.

>> The bottom line is that I think this sort of thing needs to be 
>> thrashed out within the sip community. So I think the RFC process is 
>> the right process.
> I think that bending the rules because some SIP people don't like on how 
> other organisations are applying SIP in their solutions will alieanate 
> those SIP users from that same SIP community. So I seriously hope we 
> will not go down that path.

Lets see what others say.

IMO its important that those other communities don't use this mechanism 
to erect more walls around their garden.

	Thanks,
	Paul

> /Hans Erik
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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Subject: Re: [Sip] SIP INFO
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On Mar 23, 2009, at 6:19 AM, Brett Tate wrote:
> What was the outcome of INFO 489 response MUST trigger release of =20
> call discussion?

Whoops - that was slide 2...

=95Get a 489, can all still be fine?
	=95Option 1: Tear down dialog
		Something is hopelessly broken, give up

	=95Option 2: ignore it
		=96We say INFO does not change dialog state,
                   and a 489 tearing down dialog changes state
		=96But, handling 489 differently for INFO makes INFO =
=93special=94
		=96But what if I said =93package X is OK=94 in an =
INVITE, but there is
                   glare between you sending an INFO from package X =20
and my
                  dropping X in my ACK?


Thoughts?=

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Subject: Re: [Sip] Comment on draft-roach-sipcore-rfc3265bis-00
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Adam Roach wrote:

> Because of the true interoperability 
> nightmare that has resulted from dialog sharing, my suggestion would be 
> to disable notifier functionality on those endpoints under such 
> circumstances (perhaps with an implementation option to fall back to 
> 3265 behavior for the truly masochistic).

Can you say more about the "true interoperability nightmare that has 
resulted from dialog sharing"?

I understand that there undoubtedly are interop issues due to some UAs 
not supporting dialog sharing. There are also interop issues due to the 
UAs not supporting an event package, or not supporting GRUU, or 
whatever. Is one more of a nightmare than another?

Dialog sharing has been a legal possibility since 3261 was published. So 
it shouldn't be a surprise to anybody. If somebody didn't implement 
support, well, what can you say?

It was used primarily because contacts in dialogs often don't have the 
gruu property, and GRUU hasn't been widely implemented. Has that 
situation changed? Will 3265bis somehow result in widespread 
implementation of GRUU by registrars?

	Thanks,
	Paul

From brett@broadsoft.com  Mon Mar 23 06:57:06 2009
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From: Brett Tate <brett@broadsoft.com>
To: Eric Burger <eburger@standardstrack.com>, SIP IETF <SIP@ietf.org>
Date: Mon, 23 Mar 2009 06:57:55 -0700
Thread-Topic: [Sip] SIP INFO
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Because of race/glare conditions, I prefer INFO 489 not MUST or SHOULD trig=
ger the release of the call.  Concerning calls, I see little difference bet=
ween impacts of 489 and 405/501/420.  Thus I prefer option 2.

Option 2 appears reasonably consistent with notes 3 and 12 within RFC 5057 =
section 5.1.

> -----Original Message-----
> From: Eric Burger [mailto:eburger@standardstrack.com]
> Sent: Monday, March 23, 2009 9:29 AM
> To: Brett Tate
> Cc: SIP IETF
> Subject: Re: [Sip] SIP INFO
>=20
> On Mar 23, 2009, at 6:19 AM, Brett Tate wrote:
> > What was the outcome of INFO 489 response MUST trigger release of
> > call discussion?
>=20
> Whoops - that was slide 2...
>=20
> *Get a 489, can all still be fine?
> 	*Option 1: Tear down dialog
> 		Something is hopelessly broken, give up
>=20
> 	*Option 2: ignore it
> 		-We say INFO does not change dialog state,
>                    and a 489 tearing down dialog changes state
> 		-But, handling 489 differently for INFO makes INFO "special"
> 		-But what if I said "package X is OK" in an INVITE, but there
> is
>                    glare between you sending an INFO from package X
> and my
>                   dropping X in my ACK?
>=20
>=20
> Thoughts?

From john.elwell@siemens.com  Mon Mar 23 07:11:26 2009
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From: "Elwell, John" <john.elwell@siemens.com>
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To: Tom Taylor <tom.taylor@rogers.com>, Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: SIP WG <sip@ietf.org>, Adam Roach <adam@nostrum.com>
Subject: Re: [Sip] draft-ietf-sip-199-06
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=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Tom Taylor
> Sent: 23 March 2009 12:35
> To: Christer Holmberg
> Cc: SIP WG; Adam Roach
> Subject: Re: [Sip] draft-ietf-sip-199-06
>=20
> Some text , as requested.
>=20
> Christer Holmberg wrote:
> ...
> >=20
> >> In section 6:
> >>
> >>   When a forking proxy receives a non-2xx final response that it
> >>   recognizes as terminating one or more early dialogs, if the proxy
> >>   does not intend to forward the final response immediately (in
> >>   accordance with rules for a forking proxy) and the UAC=20
> has indicated
> >>   support for the 199 response code, the proxy SHOULD generate and
> > send
> >>   a 199 response upstream for each early dialog terminated on the
> >>   downstream side by the non-2xx final response, except=20
> for any early
> >>   dialog for which the proxy has previously received and=20
> forwarded a
> >>   199 response.
> >>
> ...
>=20
> My suggestion:
>=20
> "When a forking proxy receives a non-2xx final response that it
>   recognizes as terminating one or more early dialogs, it=20
> SHOULD generate
>   and send a 199 response upstream for each such early dialog=20
> satisfying the
>   following conditions:
>   -- the proxy does not intend to forward the final response=20
> immediately
>      (in accordance with rules for a forking proxy);
>   -- the UAC has indicated support for the 199 response code;
>   -- the proxy has not already received and forwarded a 199=20
> response for
>      that early dialogue."
[JRE] Make it clear that all 3 conditions must be true, e.g., by saying
"satisfying all of the following conditions".

John

> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

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Subject: Re: [Sip] SIP INFO
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Eric Burger wrote:

> Likewise, what about REFER/REFER 2xx?  My vote here is INFO packages 
> would be nonsensical during a refer, but perhaps you have a use case?  
> If so, they work like everything else.

REFER initiates subscription establishment, if not already within a 
dialog. But INFO belongs with INVITE dialog-usages, not subscriptions. 
So the INFO negotiation makes no sense in out-of-dialog REFERs.

If the REFER is within an existing dialog that has an INVITE dialog 
usage, it is still (IIUC) technically not within the invite dialog 
usage. So it shouldn't be used for negotiating INFO packages.

	Thanks,
	Paul

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Dan,

Yes, I was amused that Skype would announce this on the first day of  
IETF 74 (although I somehow wonder if Skype even knew).  Two other  
posts on the topic that went into more detail (one of them is  
admittedly mine):

http://www.disruptivetelephony.com/2009/03/skype-tears-down-more-walls-with-skype-for-sip.html
http://voiceontheweb.biz/2009/03/business-long-distance-death-by-1000000-pbxs/

The big missing piece to me is that it can't do SIP -> Skype name.   
This new "Skype For SIP" service will support:

- Skype Clients --> PBX
- SkypeIn/Online numbers (PSTN) --> PBX
- PBX --> SkypeOut (PSTN)

So for outbound Skype becomes a "SIP service provider" for PSTN  
connectivity.  But you still can't connect from your IP-PBX via SIP to  
a random Skype user, which could be interesting on a number of  
levels.  Skype says they are open to exploring this during the beta  
period, though.  The Skype folks I've spoken to also say they see this  
"Skype For SIP" with voice as only a first step...  they say they do  
want to expand that interconnectivity to presence, video, etc.  We'll  
see.

Dan

On Mar 23, 2009, at 1:39 AM, Dan Wing wrote:

> http://www.networkworld.com/news/2009/032309-skype-sip.html?hpg1=bn



-- 
Dan York, Director of Conversations
Voxeo Corporation   http://www.voxeo.com  dyork@voxeo.com
Phone: +1-407-455-5859    Skype: danyork
Blogs: http://blogs.voxeo.com  http://www.disruptivetelephony.com

Build powerful voice applications quickly
Find out how at http://www.voxeo.com/free








From brett@broadsoft.com  Mon Mar 23 08:17:58 2009
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From: Brett Tate <brett@broadsoft.com>
To: SIP IETF <SIP@ietf.org>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Mon, 23 Mar 2009 08:18:47 -0700
Thread-Topic: Is REFER within dialog part of INVITE or subscription usage? (was RE: [Sip] SIP INFO)
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Subject: [Sip] Is REFER within dialog part of INVITE or subscription usage? (was RE: SIP INFO)
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> If the REFER is within an existing dialog that has an INVITE=20
> dialog usage, it is still (IIUC) technically not within the=20
> invite dialog usage. So it shouldn't be used for negotiating=20
> INFO packages.

Paul: Thanks for raising the issues.

Everyone: Is REFER within dialog part of INVITE usage, SUBSCRIPTION usage, =
or neither?

I'm asking mainly because of REFER 481 impacts; however the answer also has=
 potential implications concerning draft-ietf-sip-info-events.  RFC 5057 ap=
pears to imply that it is part of the subscription usage; however I'm not p=
ositive if my interpretation and/or RFC 5057 is correct.

If REFER isn't considered part of the INVITE usage, how should REFER 481 be=
 interpreted?  More specifically, should receiver of REFER 481 send BYE?

Thanks,
Brett


From christer.holmberg@ericsson.com  Mon Mar 23 09:24:27 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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Hi,

Thanks for the text, Tom!

Regarding documenting that all 3 conditions must be fullfilled, and way
is to write ";and" after each condition.

 "When a forking proxy receives a non-2xx final response that it
  recognizes as terminating one or more early dialogs, it SHOULD
generate
  and send a 199 response upstream for each such early dialog satisfying
the
  following conditions:
  -- the proxy does not intend to forward the final response immediately
     (in accordance with rules for a forking proxy);and
  -- the UAC has indicated support for the 199 response code;and
  -- the proxy has not already received and forwarded a 199 response for
     that early dialogue."

Reegards,

Christer

-----Original Message-----
From: Elwell, John [mailto:john.elwell@siemens.com]=20
Sent: Monday, March 23, 2009 4:12 PM
To: Tom Taylor; Christer Holmberg
Cc: SIP WG; Adam Roach
Subject: RE: [Sip] draft-ietf-sip-199-06

=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of=20
> Tom Taylor
> Sent: 23 March 2009 12:35
> To: Christer Holmberg
> Cc: SIP WG; Adam Roach
> Subject: Re: [Sip] draft-ietf-sip-199-06
>=20
> Some text , as requested.
>=20
> Christer Holmberg wrote:
> ...
> >=20
> >> In section 6:
> >>
> >>   When a forking proxy receives a non-2xx final response that it
> >>   recognizes as terminating one or more early dialogs, if the proxy
> >>   does not intend to forward the final response immediately (in
> >>   accordance with rules for a forking proxy) and the UAC
> has indicated
> >>   support for the 199 response code, the proxy SHOULD generate and
> > send
> >>   a 199 response upstream for each early dialog terminated on the
> >>   downstream side by the non-2xx final response, except
> for any early
> >>   dialog for which the proxy has previously received and
> forwarded a
> >>   199 response.
> >>
> ...
>=20
> My suggestion:
>=20
> "When a forking proxy receives a non-2xx final response that it
>   recognizes as terminating one or more early dialogs, it SHOULD=20
> generate
>   and send a 199 response upstream for each such early dialog=20
> satisfying the
>   following conditions:
>   -- the proxy does not intend to forward the final response=20
> immediately
>      (in accordance with rules for a forking proxy);
>   -- the UAC has indicated support for the 199 response code;
>   -- the proxy has not already received and forwarded a 199 response=20
> for
>      that early dialogue."
[JRE] Make it clear that all 3 conditions must be true, e.g., by saying
"satisfying all of the following conditions".

John

> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> sipping@ietf.org for new developments on the application of sip
>=20

From dean.willis@softarmor.com  Mon Mar 23 09:30:31 2009
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Subject: [Sip] skype for sip announced
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Normally, I wouldn't mention product announcements here, but I thought 
this was quite interesting:

http://blog.wired.com/business/2009/03/skype-means-bus.html

--
dean

From L.Liess@telekom.de  Mon Mar 23 10:21:18 2009
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Jonathan,

The Session-ID it's already on the way to be implemented because it's
needed quite badly now. Most SIP services use B2BUAs.  Their operation
guys have big problems to correlate the call legs and find what's wrong
when customers call because their SIP ATA-box doesn't work. I can guess
your comment but there isn't anything we can do about it. The B2BUAs are
there to stay.=20
The alternative to the Session-ID is every SIP serrvice implementing ist
own proprietary stuff. We can not want this happen.

We currently use the IP-address and ports in SDP for correlation in SIP
and MGCP/MeGaCo. This covers now about 90% of the traffic, but this will
change soon and we need something else.  =20

I agree with you that there is a clear need to solve the whole problem.
But I don't think we have a solution for it within the next eight weeks.


So let's get the Session-ID agreed as soon as possible so people can use
it. On the other side let's work on the more general problem. In my
opinion, the current version of the context-ID requirements draft is
intended to go into this direction, or did I miss something?=20

Laura=20



> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Jonathan Rosenberg
> Sent: Saturday, March 21, 2009 4:29 Pm=20
> To: IETF SIP List
> Subject: [Sip] comments on draft-kaplan-sip-session-id-01
>=20
> I think we all agree that it would be nice if there was an identifier=20
> that worked through b2bua, which we could use to identify dialogs.
>=20
> However, I think the problem is far more complicated than=20
> this document=20
> talks about. The focus is on making it work through SBC type of boxes=20
> which aren't doing a lot of fancy call control, but are rewriting=20
> call-id and doing security things. There, I think the problem is easy.
>=20
> However, there are lots of cases where more complicated dialog=20
> manipulation mucks this up in a really horrible way. In particular,=20
> enterprise features like transfer, park/pickup, and so on, are often=20
> implemented by means of re-INVITEs on dialogs all anchored in the=20
> IP-PBX. For example, consider the following transfer implementation:
>=20
> A calls B. A calls C. Both calls are routed through an IP-PBX, which=20
> acts as a b2bua, and so we end up with something which looks=20
> like this:
>=20
>   Please view in a fixed-width font such as Courier.
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>                      +------------+       S1
>                 S1   |            | ------------    B
>              --------|            |
>       A         S2   |   server   |
>              --------|            |       S2
>                      |            | -------------   C
>                      |            |
>                      +------------+
>=20
> Where S1 and S2 are the two session IDs. So far, so good. Now, A=20
> transfers the call to B. A sends a REFER on dialog S1. Like=20
> it or not,=20
> due to frequent REFER/transfer interop issues, this is sometimes=20
> implemented as a re-INVITE from the server, connecting B and=20
> C's media=20
> streams together, and then disconnecting A, so that you end up with:
>=20
>   Please view in a fixed-width font such
>                as Courier.
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>          +------------+       S1
>          |            | ------------    B
>          |            |
>          |   server   |
>          |            |       S2
>          |            | -------------   C
>          |            |
>          +------------+
>=20
> Now B and C are talking to each other. What is the session ID=20
> for this=20
> dialog? B and C were both originally the UAS on different=20
> dialogs with=20
> different session IDs!
>=20
> The problem happens more generally for any 3pcc operation, and the=20
> meta-issue is how to come up with a clear notion of who 'owns' the=20
> sessionID creation and destruction in cases where calls are=20
> transferred,=20
> moved, conferenced, parked and picked up all over the place.
>=20
> As such, a true comprehensive solution to this problem is much more=20
> complicated than just 'tell the SBCs to stop mucking with=20
> this header'.=20
> It would need to account for a wider range of 3pcc and other dialog=20
> manipulations which occur in the wild. Or, we could try to=20
> scope this in=20
> a more limited way, focusing on the basic A-calls-B case and=20
> solve just=20
> the SBC problem.
>=20
> Which is it?
>=20
> -Jonathan R.
>=20
> --=20
> Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
> Cisco Fellow                                   Iselin, NJ 08830
> Cisco, Voice Technology Group
> jdrosen@cisco.com
> http://www.jdrosen.net                         PHONE: (408) 902-3084
> http://www.cisco.com
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

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Date: Mon, 23 Mar 2009 18:49:33 +0100
Message-ID: <9ae56b1e0903231049j53ad700cpc9b13e2d890551ba@mail.gmail.com>
From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
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Cc: sip <sip@ietf.org>
Subject: Re: [Sip] Policy to register media feature tags
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--00504502d175a127d20465cce33e
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There is no rule stating that requiring "audio" and "video" will imply that
you also require one of the new tags. In that sense these tags are
completely orthogonal.

If the caller adds "require" header field parameter then it is a consequence
of the callerpreferences processing that some calls could fail that would
otherwise have succeeded. If the caller added this, then this must have been
its intention. That is the *feature* that RFC3841 provides to the caller.
The 3gpp specification does not say that one has to use "require" header
field parameter, it even warns that this might hamper interoperability.

So I think your following sentence was a fair analysis of how the newly
proposed tags are actually used/should be judged: "If it only represents a
preference for whatever extra goodies are provided by 3gpp over and above
audio and video, then it may be a reasonable feature tag."

/Hans Erik van Elburg


On Mon, Mar 23, 2009 at 2:26 PM, Paul Kyzivat <pkyzivat@cisco.com> wrote:

> Inline...
>
> Hans Erik van Elburg wrote:
>
>> Responses inline...
>>
>> /Hans Erik
>>
>> Paul Kyzivat wrote:
>>
>>> On one side were those who wanted capabilities and preferences to be
>>> stated in terms of feature tags that are well known and orthogonal. On the
>>> other side were those who prefer to associate arbitrary names to collections
>>> of features and then negotiate on the basis of the names of those
>>> collections.
>>>
>>> Using the latter approach its possible that interoperation will fail
>>> because the parties don't share a common name for a collection of features
>>> even though they both possess the necessary features to interoperate.
>>>
>> That is a feature.
>>
>
> *what* is a feature?
>
>  This interoperability "problem" only occurs when the originating party
>> includes the header field parameters "require" and "explicit" in the
>> Accept-Contact header field containing the feature tag. In this case the
>> failure of the call when not all of the receiving  UA's support  the feature
>> is intended!
>>
>
> Lets get tangible about this. (This is retracing old ground, but people
> sometimes have short memories.)
>
> Suppose Alice calls Bob. Alice has the latest and greatest 3gpp phone with
> the capability to make audio/video calls, and would prefer to do so if
> possible. Bob has an open source softphone running on his PC. It is fully
> capable of audio/video operation using a wide source of codecs. He also has
> a plain vanilla audio phone registered to the same AOR.
>
> Alice's phone will send an invite with offer containing audio and video
> media. It can also include callerprefs to bias the selection of device in
> favor of one that will support audio and video. It could do so a variety of
> different ways:
> - indicate preference for audio and video
> - indicate preference for the 3gpp "videophone" feature.
> - indicate preference for audio, video, *and* 3gpp "videophone"
>
> Assuming none of these *require* these features, the first and the last
> will both bias towards the softphone and result in an audio/video call. The
> middle will probably cause both of bob's phones to ring and audio/video will
> result if he happens to answer the softphone.
>
> The situation is worse if Alice *requires* both audio and video, and as a
> result *requires* the 3gpp videophone feature.
>
> The problem is one of orthogonality. the features should all be orthogonal
> to one another. If "3gpp videophone" encompasses "audio, video, and the 3gpp
> videophone look and feel", then it isn't orthogonal to the existing audio
> and video feature tags. If it only represents a preference for whatever
> extra goodies are provided by 3gpp over and above audio and video, then it
> may be a reasonable feature tag.
>
> (Note that I have no knowledge of how 3gpp intends to name an use features.
> I'm just exploring the possibilities.)
>
>  I don't have a problem with independent groups defining new feature tags,
>>> as long as they are primitive and orthogonal to existing ones. But I do have
>>> a problem with defining feature tags that identify collections of features,
>>> especially when the mapping from the identifier to the collection of
>>> features is not public.
>>>
>> Well who is going to determine what can be called a feature and what not.
>> I also like to see where this principle is documented to apply to the
>> global tree.
>>
>
> IMO it was probably a mistake to use the existing feature tag mechanism for
> callee caps in the first place. While there are some simmilarities, the
> intentions are quite different. But we are where we are.
>
> I don't know if we can impose additional rules on the global tree or not. I
> expect that imposing an orthogonality constraint might not be so
> controversial, and so *might* be possible. But the challenge may be in
> *defining* orthogonality in an application-independent way. It would be
> easier to do that in the context of sip, though still not trivial.
>
>  The bottom line is that I think this sort of thing needs to be thrashed
>>> out within the sip community. So I think the RFC process is the right
>>> process.
>>>
>> I think that bending the rules because some SIP people don't like on how
>> other organisations are applying SIP in their solutions will alieanate those
>> SIP users from that same SIP community. So I seriously hope we will not go
>> down that path.
>>
>
> Lets see what others say.
>
> IMO its important that those other communities don't use this mechanism to
> erect more walls around their garden.
>
>        Thanks,
>        Paul
>
>  /Hans Erik
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
>>

--00504502d175a127d20465cce33e
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

There is no rule stating that requiring &quot;audio&quot; and &quot;video&q=
uot; will imply that you also require one of the new tags. In that sense th=
ese tags are completely orthogonal.<br><br>If the caller adds &quot;require=
&quot; header field parameter then it is a consequence of the callerprefere=
nces processing that some calls could fail that would otherwise have succee=
ded. If the caller added this, then this must have been its intention. That=
 is the *feature* that RFC3841 provides to the caller. The 3gpp specificati=
on does not say that one has to use &quot;require&quot; header field parame=
ter, it even warns that this might hamper interoperability.<br>
<br>So I think your following sentence was a fair analysis of how the newly=
 proposed tags are actually used/should be judged: &quot;If it only represe=
nts a preference for whatever extra goodies are
provided by 3gpp over and above audio and video, then it may be a
reasonable feature tag.&quot;<br><br clear=3D"all">/Hans Erik van Elburg<br=
>
<br><br><div class=3D"gmail_quote">On Mon, Mar 23, 2009 at 2:26 PM, Paul Ky=
zivat <span dir=3D"ltr">&lt;<a href=3D"mailto:pkyzivat@cisco.com">pkyzivat@=
cisco.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; p=
adding-left: 1ex;">
Inline...<div class=3D"im"><br>
<br>
Hans Erik van Elburg wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Responses inline...<br>
<br>
/Hans Erik<br>
<br>
Paul Kyzivat wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
On one side were those who wanted capabilities and preferences to be stated=
 in terms of feature tags that are well known and orthogonal. On the other =
side were those who prefer to associate arbitrary names to collections of f=
eatures and then negotiate on the basis of the names of those collections.<=
br>

<br>
Using the latter approach its possible that interoperation will fail becaus=
e the parties don&#39;t share a common name for a collection of features ev=
en though they both possess the necessary features to interoperate.<br>

</blockquote>
That is a feature.<br>
</blockquote>
<br></div>
*what* is a feature?<div class=3D"im"><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
This interoperability &quot;problem&quot; only occurs when the originating =
party includes the header field parameters &quot;require&quot; and &quot;ex=
plicit&quot; in the Accept-Contact header field containing the feature tag.=
 In this case the failure of the call when not all of the receiving =A0UA&#=
39;s support =A0the feature is intended!<br>

</blockquote>
<br></div>
Lets get tangible about this. (This is retracing old ground, but people som=
etimes have short memories.)<br>
<br>
Suppose Alice calls Bob. Alice has the latest and greatest 3gpp phone with =
the capability to make audio/video calls, and would prefer to do so if poss=
ible. Bob has an open source softphone running on his PC. It is fully capab=
le of audio/video operation using a wide source of codecs. He also has a pl=
ain vanilla audio phone registered to the same AOR.<br>

<br>
Alice&#39;s phone will send an invite with offer containing audio and video=
 media. It can also include callerprefs to bias the selection of device in =
favor of one that will support audio and video. It could do so a variety of=
 different ways:<br>

- indicate preference for audio and video<br>
- indicate preference for the 3gpp &quot;videophone&quot; feature.<br>
- indicate preference for audio, video, *and* 3gpp &quot;videophone&quot;<b=
r>
<br>
Assuming none of these *require* these features, the first and the last wil=
l both bias towards the softphone and result in an audio/video call. The mi=
ddle will probably cause both of bob&#39;s phones to ring and audio/video w=
ill result if he happens to answer the softphone.<br>

<br>
The situation is worse if Alice *requires* both audio and video, and as a r=
esult *requires* the 3gpp videophone feature.<br>
<br>
The problem is one of orthogonality. the features should all be orthogonal =
to one another. If &quot;3gpp videophone&quot; encompasses &quot;audio, vid=
eo, and the 3gpp videophone look and feel&quot;, then it isn&#39;t orthogon=
al to the existing audio and video feature tags. If it only represents a pr=
eference for whatever extra goodies are provided by 3gpp over and above aud=
io and video, then it may be a reasonable feature tag.<br>

<br>
(Note that I have no knowledge of how 3gpp intends to name an use features.=
 I&#39;m just exploring the possibilities.)<div class=3D"im"><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><blockquote class=
=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, 204, 204); margin=
: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">

I don&#39;t have a problem with independent groups defining new feature tag=
s, as long as they are primitive and orthogonal to existing ones. But I do =
have a problem with defining feature tags that identify collections of feat=
ures, especially when the mapping from the identifier to the collection of =
features is not public.<br>

</blockquote>
Well who is going to determine what can be called a feature and what not.<b=
r>
I also like to see where this principle is documented to apply to the globa=
l tree.<br>
</blockquote>
<br></div>
IMO it was probably a mistake to use the existing feature tag mechanism for=
 callee caps in the first place. While there are some simmilarities, the in=
tentions are quite different. But we are where we are.<br>
<br>
I don&#39;t know if we can impose additional rules on the global tree or no=
t. I expect that imposing an orthogonality constraint might not be so contr=
oversial, and so *might* be possible. But the challenge may be in *defining=
* orthogonality in an application-independent way. It would be easier to do=
 that in the context of sip, though still not trivial.<div class=3D"im">
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><blockquote class=
=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, 204, 204); margin=
: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">

The bottom line is that I think this sort of thing needs to be thrashed out=
 within the sip community. So I think the RFC process is the right process.=
<br>
</blockquote>
I think that bending the rules because some SIP people don&#39;t like on ho=
w other organisations are applying SIP in their solutions will alieanate th=
ose SIP users from that same SIP community. So I seriously hope we will not=
 go down that path.<br>

</blockquote>
<br></div>
Lets see what others say.<br>
<br>
IMO its important that those other communities don&#39;t use this mechanism=
 to erect more walls around their garden.<br>
<br>
 =A0 =A0 =A0 =A0Thanks,<br>
 =A0 =A0 =A0 =A0Paul<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
/Hans Erik<div class=3D"im"><br>
_______________________________________________<br>
Sip mailing list =A0<a href=3D"https://www.ietf.org/mailman/listinfo/sip" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
This list is for NEW development of the core SIP Protocol<br>
Use <a href=3D"mailto:sip-implementors@cs.columbia.edu" target=3D"_blank">s=
ip-implementors@cs.columbia.edu</a> for questions on current sip<br>
Use <a href=3D"mailto:sipping@ietf.org" target=3D"_blank">sipping@ietf.org<=
/a> for new developments on the application of sip<br>
<br>
</div></blockquote>
</blockquote></div><br>

--00504502d175a127d20465cce33e--

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Cc: sip <sip@ietf.org>
Subject: Re: [Sip] Policy to register media feature tags
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Hans Erik van Elburg wrote:
> There is no rule stating that requiring "audio" and "video" will imply 
> that you also require one of the new tags. In that sense these tags are 
> completely orthogonal.

Its good to hear that. While I had no specifics, I had hints to the 
contrary. I guess those were wrong.

> If the caller adds "require" header field parameter then it is a 
> consequence of the callerpreferences processing that some calls could 
> fail that would otherwise have succeeded. If the caller added this, then 
> this must have been its intention. That is the *feature* that RFC3841 
> provides to the caller. The 3gpp specification does not say that one has 
> to use "require" header field parameter, it even warns that this might 
> hamper interoperability.
> 
> So I think your following sentence was a fair analysis of how the newly 
> proposed tags are actually used/should be judged: "If it only represents 
> a preference for whatever extra goodies are provided by 3gpp over and 
> above audio and video, then it may be a reasonable feature tag."

If that is the intent, then I am reassured.

Presumably the end user isn't actually selecting callerprefs. Rather the 
end user is just pushing a button on the phone to initiate a call, and 
some application on the phone is deciding what callerprefs to use.

So in the end it will be a question of whether the app constructs the 
call to maximize interop, or not.

	Thanks,
	Paul

> /Hans Erik van Elburg
> 
> 
> On Mon, Mar 23, 2009 at 2:26 PM, Paul Kyzivat <pkyzivat@cisco.com 
> <mailto:pkyzivat@cisco.com>> wrote:
> 
>     Inline...
> 
> 
>     Hans Erik van Elburg wrote:
> 
>         Responses inline...
> 
>         /Hans Erik
> 
>         Paul Kyzivat wrote:
> 
>             On one side were those who wanted capabilities and
>             preferences to be stated in terms of feature tags that are
>             well known and orthogonal. On the other side were those who
>             prefer to associate arbitrary names to collections of
>             features and then negotiate on the basis of the names of
>             those collections.
> 
>             Using the latter approach its possible that interoperation
>             will fail because the parties don't share a common name for
>             a collection of features even though they both possess the
>             necessary features to interoperate.
> 
>         That is a feature.
> 
> 
>     *what* is a feature?
> 
> 
>         This interoperability "problem" only occurs when the originating
>         party includes the header field parameters "require" and
>         "explicit" in the Accept-Contact header field containing the
>         feature tag. In this case the failure of the call when not all
>         of the receiving  UA's support  the feature is intended!
> 
> 
>     Lets get tangible about this. (This is retracing old ground, but
>     people sometimes have short memories.)
> 
>     Suppose Alice calls Bob. Alice has the latest and greatest 3gpp
>     phone with the capability to make audio/video calls, and would
>     prefer to do so if possible. Bob has an open source softphone
>     running on his PC. It is fully capable of audio/video operation
>     using a wide source of codecs. He also has a plain vanilla audio
>     phone registered to the same AOR.
> 
>     Alice's phone will send an invite with offer containing audio and
>     video media. It can also include callerprefs to bias the selection
>     of device in favor of one that will support audio and video. It
>     could do so a variety of different ways:
>     - indicate preference for audio and video
>     - indicate preference for the 3gpp "videophone" feature.
>     - indicate preference for audio, video, *and* 3gpp "videophone"
> 
>     Assuming none of these *require* these features, the first and the
>     last will both bias towards the softphone and result in an
>     audio/video call. The middle will probably cause both of bob's
>     phones to ring and audio/video will result if he happens to answer
>     the softphone.
> 
>     The situation is worse if Alice *requires* both audio and video, and
>     as a result *requires* the 3gpp videophone feature.
> 
>     The problem is one of orthogonality. the features should all be
>     orthogonal to one another. If "3gpp videophone" encompasses "audio,
>     video, and the 3gpp videophone look and feel", then it isn't
>     orthogonal to the existing audio and video feature tags. If it only
>     represents a preference for whatever extra goodies are provided by
>     3gpp over and above audio and video, then it may be a reasonable
>     feature tag.
> 
>     (Note that I have no knowledge of how 3gpp intends to name an use
>     features. I'm just exploring the possibilities.)
> 
> 
>             I don't have a problem with independent groups defining new
>             feature tags, as long as they are primitive and orthogonal
>             to existing ones. But I do have a problem with defining
>             feature tags that identify collections of features,
>             especially when the mapping from the identifier to the
>             collection of features is not public.
> 
>         Well who is going to determine what can be called a feature and
>         what not.
>         I also like to see where this principle is documented to apply
>         to the global tree.
> 
> 
>     IMO it was probably a mistake to use the existing feature tag
>     mechanism for callee caps in the first place. While there are some
>     simmilarities, the intentions are quite different. But we are where
>     we are.
> 
>     I don't know if we can impose additional rules on the global tree or
>     not. I expect that imposing an orthogonality constraint might not be
>     so controversial, and so *might* be possible. But the challenge may
>     be in *defining* orthogonality in an application-independent way. It
>     would be easier to do that in the context of sip, though still not
>     trivial.
> 
> 
>             The bottom line is that I think this sort of thing needs to
>             be thrashed out within the sip community. So I think the RFC
>             process is the right process.
> 
>         I think that bending the rules because some SIP people don't
>         like on how other organisations are applying SIP in their
>         solutions will alieanate those SIP users from that same SIP
>         community. So I seriously hope we will not go down that path.
> 
> 
>     Lets see what others say.
> 
>     IMO its important that those other communities don't use this
>     mechanism to erect more walls around their garden.
> 
>            Thanks,
>            Paul
> 
>         /Hans Erik
> 
>         _______________________________________________
>         Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>         This list is for NEW development of the core SIP Protocol
>         Use sip-implementors@cs.columbia.edu
>         <mailto:sip-implementors@cs.columbia.edu> for questions on
>         current sip
>         Use sipping@ietf.org <mailto:sipping@ietf.org> for new
>         developments on the application of sip
> 
> 

From Adam.Uzelac@globalcrossing.com  Mon Mar 23 13:59:35 2009
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From: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
To: Dan York <dyork@voxeo.com>, Dan Wing <dwing@cisco.com>
Date: Mon, 23 Mar 2009 17:00:21 -0400
Thread-Topic: [Sip] Skype for SIP
Thread-Index: AcmrxJVaA4dZHjfzRja43DiHFeNcCwANOFPg
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Cc: "sip@ietf.org" <sip@ietf.org>
Subject: Re: [Sip] Skype for SIP
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Skype for SIP =3D SKIP?

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Dan York
> Sent: Monday, March 23, 2009 7:34 AM
> To: Dan Wing
> Cc: sip@ietf.org
> Subject: Re: [Sip] Skype for SIP
>=20
> Dan,
>=20
> Yes, I was amused that Skype would announce this on the first day of
> IETF 74 (although I somehow wonder if Skype even knew).  Two other
> posts on the topic that went into more detail (one of them is
> admittedly mine):
>=20
> http://www.disruptivetelephony.com/2009/03/skype-tears-down-more-walls-
> with-skype-for-sip.html
> http://voiceontheweb.biz/2009/03/business-long-distance-death-by-
> 1000000-pbxs/
>=20
> The big missing piece to me is that it can't do SIP -> Skype name.
> This new "Skype For SIP" service will support:
>=20
> - Skype Clients --> PBX
> - SkypeIn/Online numbers (PSTN) --> PBX
> - PBX --> SkypeOut (PSTN)
>=20
> So for outbound Skype becomes a "SIP service provider" for PSTN
> connectivity.  But you still can't connect from your IP-PBX via SIP to
> a random Skype user, which could be interesting on a number of
> levels.  Skype says they are open to exploring this during the beta
> period, though.  The Skype folks I've spoken to also say they see this
> "Skype For SIP" with voice as only a first step...  they say they do
> want to expand that interconnectivity to presence, video, etc.  We'll
> see.
>=20
> Dan
>=20
> On Mar 23, 2009, at 1:39 AM, Dan Wing wrote:
>=20
> > http://www.networkworld.com/news/2009/032309-skype-sip.html?hpg1=3Dbn
>=20
>=20
>=20
> --
> Dan York, Director of Conversations
> Voxeo Corporation   http://www.voxeo.com  dyork@voxeo.com
> Phone: +1-407-455-5859    Skype: danyork
> Blogs: http://blogs.voxeo.com  http://www.disruptivetelephony.com
>=20
> Build powerful voice applications quickly
> Find out how at http://www.voxeo.com/free
>=20
>=20
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

From adam@nostrum.com  Mon Mar 23 15:44:32 2009
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Cc: sip@ietf.org, "Elwell, John" <john.elwell@siemens.com>
Subject: Re: [Sip] Comment on draft-roach-sipcore-rfc3265bis-00
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Paul Kyzivat wrote:
>
>
> Adam Roach wrote:
>
>> Because of the true interoperability nightmare that has resulted from 
>> dialog sharing, my suggestion would be to disable notifier 
>> functionality on those endpoints under such circumstances (perhaps 
>> with an implementation option to fall back to 3265 behavior for the 
>> truly masochistic).
>
> Can you say more about the "true interoperability nightmare that has 
> resulted from dialog sharing"?
>
> I understand that there undoubtedly are interop issues due to some UAs 
> not supporting dialog sharing. There are also interop issues due to 
> the UAs not supporting an event package, or not supporting GRUU, or 
> whatever. Is one more of a nightmare than another?

It's not lack of support that worries me -- it's mismatched views of 
dialog and usage state between two endpoints who both think they've 
implemented dialog sharing. The problem is not that things negotiate 
down to a lesser set of capabilities -- it's that things act broken in a 
way that exposes this mixed-up view of whether a dialog (and its 
associated session/subscription) is active.

> Dialog sharing has been a legal possibility since 3261 was published. 
> So it shouldn't be a surprise to anybody. If somebody didn't implement 
> support, well, what can you say?

As one of the key perpetrators of that particular aspect of 3261, I'm 
comfortable in saying that we added it in a slapdash fashion at the 11th 
hour, without sufficient analysis (and I mean this in a self-deprecating 
way, without intending to cast aspersions on any of 3261's editors). 
Interactions between usages in a dialog were ridiculously 
underspecified. Robert spent a long time trying to rationalize this 
behavior after the fact, with the result being RFC 5057. And RFC 5057 
really helps a lot for the minority of implementors who actually know it 
exists. However, the issue is complicated enough that it takes 26 pages 
to untangle it, and the rules are a tangle of exceptions. Go read 5057 
again, and tell me that people are more likely to get that right than 
they are to get confused.

> It was used primarily because contacts in dialogs often don't have the 
> gruu property, and GRUU hasn't been widely implemented. Has that 
> situation changed? Will 3265bis somehow result in widespread 
> implementation of GRUU by registrars? 

3265bis is less mature than GRUU (in that one is a -00 individual draft, 
and the other is in the RFC editor queue). I think it's reasonable to 
rely on predecessor technologies.


/a

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Cc: SIP WG <sip@ietf.org>, "Elwell, John" <john.elwell@siemens.com>
Subject: Re: [Sip] draft-ietf-sip-199-06
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Christer Holmberg wrote:
> You can have network entities which act as "normal" UAS, which also 
> want to send 199. They may e.g. have been configured to do so, since 
> the forking proxy does not support 199. Etc.
> And, the text already says that a UAS should not send a 199, unless it has been configured to do so. But, I don't think we should forbid it from doing so.
>   


No, but you need to motivate why they might do so. Explain that they're 
sending it because they think there's an upstream forking proxy that 
won't. Explain that they might be a B2BUA that has created more than one 
early dialog with the caller, and wants to clean some of them up. It 
makes little sense to say that UAs can do this without some kind of 
explanation about why they would do so.

/a

From pkyzivat@cisco.com  Mon Mar 23 16:24:35 2009
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Cc: sip@ietf.org, "Elwell, John" <john.elwell@siemens.com>
Subject: Re: [Sip] Comment on draft-roach-sipcore-rfc3265bis-00
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Adam,

Don't get me wrong. I think this is a problem any way you turn.

I was pretty heavily involved in gruu. It certainly intended to solve a 
problem, but it too ended up much more complex than anybody hoped it 
would - to the point that I suspect there is a lot of reluctance to 
implement it, and a fair amount of potential to implement it wrong. On 
top of that, I suspect many feel they haven't needed it so far, and can 
get away without it, because the magic of SBCs eliminates the need.

Also, I know of significant deployed uses of dialog sharing, that 
probably won't go away any time soon even after deprecation.

If we had it to do over, I would not include dialog sharing in 3261. But 
despite the potential problems with it, I think that horse is out of the 
barn and will not be put back in, especially since that barn is on fire.

IMO the more likely thing is to encourage everyone who can to put a ;gr 
on their contacts, and to make a best practice of using out-of-dialog 
subscriptions/refers, etc. whenever the target address has ;gr. But if 
it doesn't, then I think one should go ahead and use dialog sharing. Of 
course one is free to refrain from carrying out some feature rather than 
initiating it via dialog sharing, but if you are on the receiving end, 
then I think support is still expected.

	Thanks,
	Paul

Adam Roach wrote:
> Paul Kyzivat wrote:
>>
>>
>> Adam Roach wrote:
>>
>>> Because of the true interoperability nightmare that has resulted from 
>>> dialog sharing, my suggestion would be to disable notifier 
>>> functionality on those endpoints under such circumstances (perhaps 
>>> with an implementation option to fall back to 3265 behavior for the 
>>> truly masochistic).
>>
>> Can you say more about the "true interoperability nightmare that has 
>> resulted from dialog sharing"?
>>
>> I understand that there undoubtedly are interop issues due to some UAs 
>> not supporting dialog sharing. There are also interop issues due to 
>> the UAs not supporting an event package, or not supporting GRUU, or 
>> whatever. Is one more of a nightmare than another?
> 
> It's not lack of support that worries me -- it's mismatched views of 
> dialog and usage state between two endpoints who both think they've 
> implemented dialog sharing. The problem is not that things negotiate 
> down to a lesser set of capabilities -- it's that things act broken in a 
> way that exposes this mixed-up view of whether a dialog (and its 
> associated session/subscription) is active.
> 
>> Dialog sharing has been a legal possibility since 3261 was published. 
>> So it shouldn't be a surprise to anybody. If somebody didn't implement 
>> support, well, what can you say?
> 
> As one of the key perpetrators of that particular aspect of 3261, I'm 
> comfortable in saying that we added it in a slapdash fashion at the 11th 
> hour, without sufficient analysis (and I mean this in a self-deprecating 
> way, without intending to cast aspersions on any of 3261's editors). 
> Interactions between usages in a dialog were ridiculously 
> underspecified. Robert spent a long time trying to rationalize this 
> behavior after the fact, with the result being RFC 5057. And RFC 5057 
> really helps a lot for the minority of implementors who actually know it 
> exists. However, the issue is complicated enough that it takes 26 pages 
> to untangle it, and the rules are a tangle of exceptions. Go read 5057 
> again, and tell me that people are more likely to get that right than 
> they are to get confused.
> 
>> It was used primarily because contacts in dialogs often don't have the 
>> gruu property, and GRUU hasn't been widely implemented. Has that 
>> situation changed? Will 3265bis somehow result in widespread 
>> implementation of GRUU by registrars? 
> 
> 3265bis is less mature than GRUU (in that one is a -00 individual draft, 
> and the other is in the RFC editor queue). I think it's reasonable to 
> rely on predecessor technologies.
> 
> 
> /a
> 

From christer.holmberg@ericsson.com  Mon Mar 23 16:30:49 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Adam Roach" <adam@nostrum.com>
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Hi,=20

>>You can have network entities which act as "normal" UAS, which also=20
>>want to send 199. They may e.g. have been configured to do so, since=20
>>the forking proxy does not support 199. Etc.
>>And, the text already says that a UAS should not send a 199, unless it
has been configured to do so. But, I don't think=20
>>we should forbid it from doing so.
>  =20
>
>No, but you need to motivate why they might do so. Explain that they're
sending it because they think there's an upstream=20
>forking proxy that won't. Explain that they might be a B2BUA that has
created more than one early dialog with the caller,=20
>and wants to clean some of them up. It makes little sense to say that
UAs can do this without some kind of explanation=20
>about why they would do so.

The text does talk about the case when the UAS knows, based on
configuration, that the proxy does not support 199, but I agree that
some text about UAS/B2BUAs creating more than one dialog can be added.

Regards,

Christer

From adam@nostrum.com  Mon Mar 23 16:42:08 2009
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Subject: Re: [Sip] Comment on draft-roach-sipcore-rfc3265bis-00
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On 3/23/09 4:25 PM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:

> I was pretty heavily involved in gruu. It certainly intended to solve a
> problem, but it too ended up much more complex than anybody hoped it
> would - to the point that I suspect there is a lot of reluctance to
> implement it, and a fair amount of potential to implement it wrong.

It definitely spent a lot of its life being quite complicated. But I just
recently re-read the version actually in the queue, and it seems much
simpler and much more implementable than many of the drafts that preceded
it.

/a



From dworley@nortel.com  Mon Mar 23 16:50:16 2009
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Subject: Re: [Sip] Comment on draft-roach-sipcore-rfc3265bis-00
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It seems that with a few adjustments we get to the point that the -bis-
is quite reasonable and straightforward regarding "subscribe dialog
re-use" issues.

Clearly, we have to moderate 4.5.1 to "Notifiers MUST implement GRUU and
SHOULD use a GRUU as their contact URI for all dialogs (including
non-subscription dialogs)."  The reason for the SHOULD is that it means
what we must say, "use a GRUU unless there is a very good reason that
you can't", by which we mean, "unless you can't get a GRUU" -- there
being no other reason that a notifier can't use a GRUU fairly easily.

4.5.1 already requires a subscriber to use a separate dialog if the
destination of the subscription is a GRUU.  Actually, that sentence
doesn't contain MUST, so it should be rephrased "If so, the subscriber
MUST create the subscription on a new dialog, using the GRUU as the
request URI for the new SUBSCRIBE."  (Leaving unspecified the
possibility that the subscriber could add Route headers to cope with
mandatory outbound proxies and other odd circumstances.)

In 4.5.2, it seems to me to be unwise to allow the notifier to respond
403 to subscriptions within existing dialogs, as that is a significant
constriction of the current requirements.  In the future, when -bis- has
been implemented and subscribers can reasonably depend on all notifiers
to implement it, then we can safely allow notifiers to be more
particular.  At least, we need to narrow the statement so that notifiers
can only reject shared-dialog SUBSCRIBEs with 403 if the notifier always
uses a GRUU as its contact URI.

In regard to the long list following "To implement dialog sharing", is
any of this material new?  As far as I can tell, it is a summary of
3265/5057.  If so, it should be explicitly labeled as such, and refer
the reader to those RFCs as the normative text.  If not, the
relationship of this list with those RFCs needs to be made explicit.

Dale



From dworley@nortel.com  Mon Mar 23 17:06:34 2009
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Subject: Re: [Sip] Is REFER within dialog part of INVITE or subscription	usage? (was RE: SIP INFO)
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On Mon, 2009-03-23 at 08:18 -0700, Brett Tate wrote:
> Everyone: Is REFER within dialog part of INVITE usage, SUBSCRIPTION
> usage, or neither?
> 
> I'm asking mainly because of REFER 481 impacts; however the answer
> also has potential implications concerning draft-ietf-sip-info-events.
> RFC 5057 appears to imply that it is part of the subscription usage;
> however I'm not positive if my interpretation and/or RFC 5057 is
> correct.
> 
> If REFER isn't considered part of the INVITE usage, how should REFER
> 481 be interpreted?  More specifically, should receiver of REFER 481
> send BYE?

I would argue that the REFER is part of the INVITE usage, because REFER
in that context is intimately related to the INVITE usage -- it is
intended to transfer the INVITE usage/dialog transferred to a different
destination.

And it seems quite safe that if the REFER receives a 481 response, the
INVITE usage must not be functional any more, since the far end of that
usage has denied that it can act on the REFER.

Now it's possible that the REFER, if it is successful, is the first
transaction of the subscription usage.  But I don't think that has any
paradoxical consequences, as by hypothesis, the REFER succeeded.

Dale



From adam@nostrum.com  Mon Mar 23 17:31:23 2009
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Subject: Re: [Sip] Comment on draft-roach-sipcore-rfc3265bis-00
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Dale Worley wrote:
> It seems that with a few adjustments we get to the point that the -bis-
> is quite reasonable and straightforward regarding "subscribe dialog
> re-use" issues.
>
> Clearly, we have to moderate 4.5.1 to "Notifiers MUST implement GRUU and
> SHOULD use a GRUU as their contact URI for all dialogs (including
> non-subscription dialogs)."  The reason for the SHOULD is that it means
> what we must say, "use a GRUU unless there is a very good reason that
> you can't", by which we mean, "unless you can't get a GRUU" -- there
> being no other reason that a notifier can't use a GRUU fairly easily.
>   

That seems a reasonable approach, with the caveat that I'd call out the 
circumstances surrounding the SHOULD explicitly. That is, I would phrase 
it like "...MUST use GRUU as their contact URI for all dialogs unless 
they cannot obtain a GRUU. Note that the only circumstances in which a 
notifier cannot obtain a GRUU is when they don't meet the criteria for 
creating self-made GRUUs and their registrar does not support GRUU."

> 4.5.1 already requires a subscriber to use a separate dialog if the
> destination of the subscription is a GRUU.  Actually, that sentence
> doesn't contain MUST, so it should be rephrased "If so, the subscriber
> MUST create the subscription on a new dialog, using the GRUU as the
> request URI for the new SUBSCRIBE."  (Leaving unspecified the
> possibility that the subscriber could add Route headers to cope with
> mandatory outbound proxies and other odd circumstances.)
>   

I'm happy to rephrase this to be use RFC 2119 language.

> In 4.5.2, it seems to me to be unwise to allow the notifier to respond
> 403 to subscriptions within existing dialogs, as that is a significant
> constriction of the current requirements.

In what way? Implementors are still allowed to accept such incoming 
connections, but one of the key changes I think is necessary in this 
revision is to free implementors from the morass of coding for dialog 
sharing -- because MOST PEOPLE GET IT WRONG. I see little value with 
maintaining backwards "compatibility" with a feature that, by and large, 
wasn't compatible between multiple implementations in the first place.

> In regard to the long list following "To implement dialog sharing", is
> any of this material new?  As far as I can tell, it is a summary of
> 3265/5057.  If so, it should be explicitly labeled as such, and refer
> the reader to those RFCs as the normative text.  If not, the
> relationship of this list with those RFCs needs to be made explicit.
>   

The list you refer to is copied from various parts of the original text 
(i.e., 3265). The intention is for this document to obsolete 3265, so 
referring to 3265 for procedures would make no sense.

/a

From rjsparks@nostrum.com  Mon Mar 23 18:08:39 2009
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Cc: SIP IETF <SIP@ietf.org>, Paul Kyzivat <pkyzivat@cisco.com>, Brett Tate <brett@broadsoft.com>
Subject: Re: [Sip] Is REFER within dialog part of INVITE or subscription	usage? (was RE: SIP INFO)
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 From a usage point of view, it is far more useful to model REFER as  
if it were a SUBSCRIBE request.
It creates a subscribe usage if it succeeds.
The impact it has on the invite usage if it fails is the same as if a  
FOO request failed (which depends on the failure).

RjS


On Mar 23, 2009, at 5:07 PM, Dale Worley wrote:

> On Mon, 2009-03-23 at 08:18 -0700, Brett Tate wrote:
>> Everyone: Is REFER within dialog part of INVITE usage, SUBSCRIPTION
>> usage, or neither?
>>
>> I'm asking mainly because of REFER 481 impacts; however the answer
>> also has potential implications concerning draft-ietf-sip-info- 
>> events.
>> RFC 5057 appears to imply that it is part of the subscription usage;
>> however I'm not positive if my interpretation and/or RFC 5057 is
>> correct.
>>
>> If REFER isn't considered part of the INVITE usage, how should REFER
>> 481 be interpreted?  More specifically, should receiver of REFER 481
>> send BYE?
>
> I would argue that the REFER is part of the INVITE usage, because  
> REFER
> in that context is intimately related to the INVITE usage -- it is
> intended to transfer the INVITE usage/dialog transferred to a  
> different
> destination.
>
> And it seems quite safe that if the REFER receives a 481 response, the
> INVITE usage must not be functional any more, since the far end of  
> that
> usage has denied that it can act on the REFER.
>
> Now it's possible that the REFER, if it is successful, is the first
> transaction of the subscription usage.  But I don't think that has any
> paradoxical consequences, as by hypothesis, the REFER succeeded.
>
> Dale
>
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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Cc: SIP <sip@ietf.org>
Subject: Re: [Sip] Comment on draft-roach-sipcore-rfc3265bis-00
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Adam Roach wrote:

>> In 4.5.2, it seems to me to be unwise to allow the notifier to respond
>> 403 to subscriptions within existing dialogs, as that is a significant
>> constriction of the current requirements.
> 
> In what way? Implementors are still allowed to accept such incoming 
> connections, but one of the key changes I think is necessary in this 
> revision is to free implementors from the morass of coding for dialog 
> sharing -- because MOST PEOPLE GET IT WRONG. I see little value with 
> maintaining backwards "compatibility" with a feature that, by and large, 
> wasn't compatible between multiple implementations in the first place.

I'd be interested in hearing in what ways "most people" get it wrong.
Getting the common cases of dialog sharing right isn't so difficult.
I expect that "most people" get corner cases wrong in lots of things.

This only frees implementors if:
- they are capable of creating their own gruus without registration
- or they ONLY work with registrars that support GRUU.
- and they only interact with other UAs that support 3265bis.

Notably, any phone that *might* be attached to a registrar that doesn't 
support GRUU will need to support dialog sharing. And any UA that might 
establish a dialog with somebody that doesn't implement 3265bis should 
be prepared to have the other end initiate dialog sharing.

I guess the question is whether the bis should indicate what should be 
done to maintain backward compatibility with 3265, or if that should 
simply be left for implementors to figure out for themselves. I can see 
the merit in leaving the old cruft out so that it *eventually* goes away.

	Thanks,
	Paul

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Cc: SIP IETF <SIP@ietf.org>, Brett Tate <brett@broadsoft.com>
Subject: Re: [Sip] Is REFER within dialog part of INVITE or subscription	usage? (was RE: SIP INFO)
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Dale Worley wrote:
> On Mon, 2009-03-23 at 08:18 -0700, Brett Tate wrote:
>> Everyone: Is REFER within dialog part of INVITE usage, SUBSCRIPTION
>> usage, or neither?
>>
>> I'm asking mainly because of REFER 481 impacts; however the answer
>> also has potential implications concerning draft-ietf-sip-info-events.
>> RFC 5057 appears to imply that it is part of the subscription usage;
>> however I'm not positive if my interpretation and/or RFC 5057 is
>> correct.
>>
>> If REFER isn't considered part of the INVITE usage, how should REFER
>> 481 be interpreted?  More specifically, should receiver of REFER 481
>> send BYE?
> 
> I would argue that the REFER is part of the INVITE usage, because REFER
> in that context is intimately related to the INVITE usage -- it is
> intended to transfer the INVITE usage/dialog transferred to a different
> destination.

Well, I guess there is that implied association. (Part of the "do what 
*I* had in mind" approach that REFER has.) Since this was never spelled 
out, it could also be simply "while this isn't part of an INVITE usage, 
if there happens to be an INVITE usage, please refer that one."

I guess the question is: would it be valid to send a REFER within an 
existing dialog that had no INVITE usage? If it were valid, presumably 
it would have the same meaning as sending one outside a dialog, except 
that the resulting subscription would share the dialog. I expect that is 
a usage it would be difficult to find in the wild.

Ah, but now I want to reconsider. Remember that REFER can take a method 
name. So, rather than referring an INVITE, I could be referring MESSAGE, 
OPTIONS, BYE, or maybe (????) SUBSCRIBE. In some of those cases, the 
association to the INVITE usage is probably meaningless. (Of course its 
meaningful for BYE.)

> And it seems quite safe that if the REFER receives a 481 response, the
> INVITE usage must not be functional any more, since the far end of that
> usage has denied that it can act on the REFER.

IIRC, 5057 decided that each dialog usage must get its own 481 before it 
goes away. I don't think there are any cases where a single 481 response 
can make multiple usages go away.

So the important question is whether the REFER is part of the INVITE 
usage and so its 481 takes down the INVITE usage or not. That gets 
especially interesting if  the dialog already had both an INVITE usage 
and a SUBSCRIBE usage. Does it make sense that the refer take down the 
INVITE usage and leave up the SUBSCRIBE usage?

> Now it's possible that the REFER, if it is successful, is the first
> transaction of the subscription usage.  But I don't think that has any
> paradoxical consequences, as by hypothesis, the REFER succeeded.

Yeah, its already so weird, why not a little more weird?

Also, I think it has recently been clarified that the 200 of a SUBSCRIBE 
should not be considered to create a dialog - that its the NOTIFY that 
creates the dialog. If so, then I suppose the same should be true of REFER.

Bottom line: I can live with it either way, but think it should be 
documented one way or the other.

	Thanks,
	Paul

From dworley@nortel.com  Mon Mar 23 18:36:42 2009
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On Mon, 2009-03-23 at 17:32 -0700, Adam Roach wrote:
> self-made GRUUs

What is a "self-made GRUU"?

> > In 4.5.2, it seems to me to be unwise to allow the notifier to respond
> > 403 to subscriptions within existing dialogs, as that is a significant
> > constriction of the current requirements.
> 
> In what way? Implementors are still allowed to accept such incoming 
> connections, but one of the key changes I think is necessary in this 
> revision is to free implementors from the morass of coding for dialog 
> sharing -- because MOST PEOPLE GET IT WRONG. I see little value with 
> maintaining backwards "compatibility" with a feature that, by and large, 
> wasn't compatible between multiple implementations in the first place.

That's a good point, although I fear that the real effect will be to
free implementors from *fixing their implementations* if something is
incorrect.

At the least, I think that you *must* limit the ability to send 403 to
devices that *are* using GRUUs as their contact.  I can *easily* see an
implementor implementing the "403" part of -bis- and deciding to omit
the "GRUU contact" part.

> > In regard to the long list following "To implement dialog sharing", is
> > any of this material new?  As far as I can tell, it is a summary of
> > 3265/5057.  If so, it should be explicitly labeled as such, and refer
> > the reader to those RFCs as the normative text.  If not, the
> > relationship of this list with those RFCs needs to be made explicit. 
> 
> The list you refer to is copied from various parts of the original text 
> (i.e., 3265). The intention is for this document to obsolete 3265, so 
> referring to 3265 for procedures would make no sense.

So you're saying that this list is *complete*?

Dale



From drage@alcatel-lucent.com  Mon Mar 23 18:37:13 2009
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(As WG chair)

This is my current understanding of the status of the SIP deliverables

regards

Keith




Documents published since IETF #73 (4 documents)
------------------------------------------------

draft-ietf-sip-answermode-07 (Proposed standard)
Published as RFC 5373
Andrew Allen, Dean Willis

draft-ietf-sip-hitchhikers-guide-05 (Informational)
Published as RFC 5411
Jonathan Rosenberg

draft-ietf-sip-rph-new-namespaces-04 (Proposed standard)
Published as RFC 5478
James Polk

draft-ietf-sip-fork-loop-fix-08 (Proposed standard)
Published as RFC 5393
Robert Sparks, Scott Lawrence, Alan Hawrylyshen, Brian Campen

Documents in RFC editor's queue (6 documents)
---------------------------------------------

draft-ietf-sip-gruu-15 (Proposed standard)
Keith Drage is document shepherd
Currently in state: In RFC Editor's Queue - MISSREF (waiting for draft-ietf=
-sip-outbound).

draft-munakata-sip-privacy-guideline (Informational)
Individual submission to RFC editor. Not from SIP WG.
Currently in state: In RFC Editor Queue in AUTH48

draft-ietf-sip-ice-option-tag-02 (Proposed standard)
Keith Drage is PROTO document shepherd
Currently in state: In RFC Editor Queue

draft-ietf-sip-sips-09 (Proposed standard)
Dean Willis is PROTO document shepherd
Currently in state: In RFC Editor's Queue - MISSREF (waiting for draft-ietf=
-sip-outbound).

draft-ietf-sip-media-security-requirements-09 (Informational)
Dean Willis is PROTO document shepherd
Currently in state: In RFC Editor Queue in AUTH48

draft-ietf-sip-dtls-srtp-framework-07 (Proposed standard)
Dean Willis is PROTO document shepherd
Currently in state: In RFC Editor Queue

Documents with IESG (7 documents)
----------------------------------

draft-ietf-sip-certs-07 (Proposed standard)
Dean Willis is PROTO document shepherd
Publication requested 1st February 2008.
Jon Peterson is responsible AD.
Currently in state: Waiting for AD Go-Ahead

draft-ietf-sip-xcapevent-04 (Proposed standard)
Dean Willis is document shepherd
OMA liaison - interaction with draft-ietf-simple-xcap-diff
Publication requested 2nd July 2008
Currently in state: IESG Evaluation:: Revised ID Needed

draft-ietf-sip-outbound-16 (Proposed standard)
Publication requested 28th July 2008
Dean Willis is PROTO document shepherd
Currently in state: Publication requested

draft-ietf-sip-connect-reuse-13
Candidate: Proposed standard
Publication requested 24th October 2008
Keith Drage is PROTO document shepherd
Currently in state: AD Evaluation:: AD Followup

draft-ietf-sip-session-policy-framework-05
Candidate: Proposed standard
Publication requested 1st November 2008
Keith Drage is PROTO document shepherd
Currently in state: IESG Evaluation:: Revised ID Needed

draft-ietf-sip-record-route-fix-06
Candidate: Best Current Practice
Publication requested 16th March 2009
Dean Willis is PROTO document shepherd
Currently in state: Publication requested

draft-ietf-sip-body-handling-06
Candidate: Proposed standard
Publication requested 11th March 2009
Currently in state: Publication requested

Documents returned to the WG from IESG (1 document)
---------------------------------------------------

draft-ietf-sip-e2m-sec-06 (Proposed standard)
Dean Willis is PROTO document shepherd=20
IESG last call completed 14th May 2007. No last call comments. Problems wit=
h proposed standard status
See draft-draft-rai-sip-header-process for possible process changes to allo=
w progressing as experimental
Current status: Dead

Documents finished in the WG, awaiting submission to IESG (3 documents)
-------------------------------------------------------------------------

draft-ietf-sip-domain-certs-02
Candidate: Informational ? Proposed standard
Milestone for WGLC November 2007
WGLC initiated 22nd February 2008 to complete 7th March 2008
Keith Drage is PROTO document shepherd
Milestone for submission to IESG September 2008

draft-ietf-sip-eku-03
Candidate: Proposed standard
Milestone for WGLC November 2007
WGLC initiated 22nd February 2008 to complete 7th March 2008
Keith Drage is PROTO document shepherd
Milestone for submission to IESG September 2008

draft-ietf-sip-subnot-etags-03
Agreed as WG item (subject to AD) at IETF#67
Candidate: proposed standard
September 2007 -- Extension for use of etags in conditional notification to=
 WGLC
WGLC initiated 4th September 2007 to complete 18th September 2007
Dean Willis is PROTO document shepherd
December 2007 -- Extension for use in etags in conditional notification to =
IESG (PS)

Documents in working group last call (5 documents)
--------------------------------------------------

draft-ietf-sip-location-conveyance-13
Candidate: Proposed standard
Milestone for submission to IESG is December 2008
WGLC initiated 4th March 2007 to complete 2nd April 2007. WGLC initiated 16=
th March 2009 to complete 23rd March 2009
Keith Drage is PROTO document shepherd

draft-ietf-sip-saml-06
Candidate: Experimental
Milestone for submission to IESG is December 2008
WGLC initiated 4th November 2008 to complete 18th November 2008
Keith Drage is PROTO document shepherd

draft-ietf-sip-199-06
Candidate: Proposed standard
WGLC initiated 4th November 2008 to complete 18th November 2008
Keith Drage is PROTO document shepherd
Milestone for submission to IESG Nov 2008

draft-ietf-sip-ua-privacy-06
Candidate: Informational
Milestone for submission to IESG April 2009
Keith Drage is PROTO document shepherd
WGLC initiated 17th February 2009 to complete 2nd March 2009 on -03 version

draft-ietf-sip-info-events-03
Candidate: Proposed standard
WGLC initiated 16th February 2009 to complete 4th March 2009 on -03 version
Milestone for submission to IESG is April 2009
Keith Drage is PROTO document shepherd

Documents still in discussion in working group (2 documents)
------------------------------------------------------------

draft-ietf-sip-sec-flows-01
Candidate: Informational
Milestone for WGLC is March 2009
Milestone for submission to IESG is July 2009
To a certain extent dependent on resolution of ideas in (and therefore wait=
ing on them):
draft-ietf-sip-sips-02 =20
draft-gurbani-sip-domain-certs-04=20
Dean Willis is PROTO document shepherd
Draft expired and removed from drafts directory=20

draft-rosenberg-sip-target-uri-delivery-01=20
Agreed as WG item at IETF#67
Candidate: proposed standard
December 2008: Delivering request-URI and parameters to UAS via proxy to WG=
LC
February 2009: Delivering request-URI and parameters to UAS via proxy to IE=
SG (PS)
Dean Willis is PROTO document shepherd

Other WG work
--------------

Essential corrections
Milestone: 1st batch to IESG in March 2009
Robert Sparks to be lead editor
Process in draft-drage-sip-essential-correction-03
Will roll in RFC 3420 and draft-ietf-sip-fork-loop-fix in first published v=
ersion of RFC 3261 update
See: http://www.softarmor.com/mediawiki/index.php/Essential_Corrections_Tra=
cking
Current documents under consideration:
draft-ietf-sip-ipv6-abnf-fix-02 (WGLC complete)
draft-sparks-sip-invfix-03 (WGLC complete but revised)
Keith Drage is PROTO document shepherd

Identify requirements for test matrix to move SIP to Draft Standard
Milestone March 2009
Robert Sparks coordinating input
Draft-sparks-sip-steps-to-draft-00 contains a proposal
Now updated to:
draft-sparks-sip-3261-interop-statement-00=20
draft-sparks-sip-3261-norm-ref-status-00 =

From adam@nostrum.com  Mon Mar 23 18:44:39 2009
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On 3/23/09 6:36 PM, "Dale Worley" <dworley@nortel.com> wrote:

> On Mon, 2009-03-23 at 17:32 -0700, Adam Roach wrote:
>> self-made GRUUs
> 
> What is a "self-made GRUU"?
 

See http://tools.ietf.org/html/draft-ietf-sip-gruu-15#section-4.3

 
> At the least, I think that you *must* limit the ability to send 403 to
> devices that *are* using GRUUs as their contact.  I can *easily* see an
> implementor implementing the "403" part of -bis- and deciding to omit
> the "GRUU contact" part.

Sounds reasonable. Of course, you can send a 403 to any request at any time
according to local policy, but I see the potential need to warn people away
from sending it for this particular reason under these particular
circumstances.
 
>> The list you refer to is copied from various parts of the original text
>> (i.e., 3265). The intention is for this document to obsolete 3265, so
>> referring to 3265 for procedures would make no sense.
> 
> So you're saying that this list is *complete*?

What I'm claiming right now is that it is a complete list of every sentence
and clause in 3265 relating to the topic. A *complete* list would include
the stuff in 5057 -- but I'm hesitant to suck in 20+ pages of information
that is already specified elsewhere.

/a



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Subject: [Sip] Slides for SIP and SIPPING sessions
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(As SIP WG chair)

Please make sure you submit your slides well before the start of the sessio=
ns of SIP or SIPPING in which you are presenting. We would suggest deadline=
s of:

Tuesday 13:00 session - submit by Tuesday 9:00
Wednesday 13:00 session - submit by Wednesday 9:00
Friday 9:00 session - submit by Thursday 15:00

These are the latest time for submission - there is nothing wrong with send=
ing them to us early.

To avoid confusion, you may want to send them to both sets of chairs, given=
 the combined nature of the agenda this time round.

regards

Keith=

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Date: Tue, 24 Mar 2009 04:00:04 +0030
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From: aayush bhatnagar <abhatnagar192006@gmail.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
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Cc: SIP IETF <SIP@ietf.org>, Brett Tate <brett@broadsoft.com>
Subject: Re: [Sip] Is REFER within dialog part of INVITE or subscription usage? (was RE: SIP INFO)
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REFER if placed within the scope of the existing dialog usage of
INVITE will reuse its dialog usage, and create an implicit
subscription for the refer event package at the recipient. However, if
the recipient chooses to reject the in dialog REFER request, the
communication established by the INVITE should not be affected. Such a
rejection scenario implies that the call transfer failed..however the
existing active call should not be torn down.

REFER if sent outside the scope of any exisiting dialog will behave as
a dialog creating request and will establish a subscription at the
recipient as described above. This subscription is only used to notify
the status of the sip session referral to the originator and is
seperate from any other subscribe usage that may have existed at the
recipient.

As far as accessing the referred resource is concerned...using the
method parameter, it depends upon the business logic of the recipient
of the REFER request. You may send a REFER to a conferencing server
with a method INVITE to invite a user to join a conference. A method
parameter with the value BYE would imply removal from the conference
by the server and so on.

I dont think you can include a SUBSCRIBE in the method parameter of
the REFER request. You have no way of telling the event package usage
of the subscription to the recipient, as the event header cannot be
included in the REFER request. Moreover, you can never be sure whether
the subscription usage is even applicable to the recipient or whether
the recipient even supports it.

aayush

On 3/24/09, Paul Kyzivat <pkyzivat@cisco.com> wrote:
>
>
> Dale Worley wrote:
>> On Mon, 2009-03-23 at 08:18 -0700, Brett Tate wrote:
>>> Everyone: Is REFER within dialog part of INVITE usage, SUBSCRIPTION
>>> usage, or neither?
>>>
>>> I'm asking mainly because of REFER 481 impacts; however the answer
>>> also has potential implications concerning draft-ietf-sip-info-events.
>>> RFC 5057 appears to imply that it is part of the subscription usage;
>>> however I'm not positive if my interpretation and/or RFC 5057 is
>>> correct.
>>>
>>> If REFER isn't considered part of the INVITE usage, how should REFER
>>> 481 be interpreted?  More specifically, should receiver of REFER 481
>>> send BYE?
>>
>> I would argue that the REFER is part of the INVITE usage, because REFER
>> in that context is intimately related to the INVITE usage -- it is
>> intended to transfer the INVITE usage/dialog transferred to a different
>> destination.
>
> Well, I guess there is that implied association. (Part of the "do what
> *I* had in mind" approach that REFER has.) Since this was never spelled
> out, it could also be simply "while this isn't part of an INVITE usage,
> if there happens to be an INVITE usage, please refer that one."
>
> I guess the question is: would it be valid to send a REFER within an
> existing dialog that had no INVITE usage? If it were valid, presumably
> it would have the same meaning as sending one outside a dialog, except
> that the resulting subscription would share the dialog. I expect that is
> a usage it would be difficult to find in the wild.
>
> Ah, but now I want to reconsider. Remember that REFER can take a method
> name. So, rather than referring an INVITE, I could be referring MESSAGE,
> OPTIONS, BYE, or maybe (????) SUBSCRIBE. In some of those cases, the
> association to the INVITE usage is probably meaningless. (Of course its
> meaningful for BYE.)
>
>> And it seems quite safe that if the REFER receives a 481 response, the
>> INVITE usage must not be functional any more, since the far end of that
>> usage has denied that it can act on the REFER.
>
> IIRC, 5057 decided that each dialog usage must get its own 481 before it
> goes away. I don't think there are any cases where a single 481 response
> can make multiple usages go away.
>
> So the important question is whether the REFER is part of the INVITE
> usage and so its 481 takes down the INVITE usage or not. That gets
> especially interesting if  the dialog already had both an INVITE usage
> and a SUBSCRIBE usage. Does it make sense that the refer take down the
> INVITE usage and leave up the SUBSCRIBE usage?
>
>> Now it's possible that the REFER, if it is successful, is the first
>> transaction of the subscription usage.  But I don't think that has any
>> paradoxical consequences, as by hypothesis, the REFER succeeded.
>
> Yeah, its already so weird, why not a little more weird?
>
> Also, I think it has recently been clarified that the 200 of a SUBSCRIBE
> should not be considered to create a dialog - that its the NOTIFY that
> creates the dialog. If so, then I suppose the same should be true of REFER.
>
> Bottom line: I can live with it either way, but think it should be
> documented one way or the other.
>
> 	Thanks,
> 	Paul
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

From jiri@iptel.org  Mon Mar 23 21:53:04 2009
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Cc: SIP <sip@ietf.org>
Subject: Re: [Sip] Diagnostic idea
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Dean Willis wrote:
> 
> On Feb 23, 2009, at 3:37 PM, Dale Worley wrote:
> 
>> I've been considering the following idea for a diagnostic tool, and I'd
>> like to get some feedback on it.
>>
>> The goal is to be able to trace the progress of a SIP request through
>> the network, including seeing the forking structure.  We first need to
>> pick a provisional response codes.  It appears that "170" is not
>> currently used.  This response code is also used as an option-tag for
>> this feature.  The processing is that whenever a SIP element receives a
>> request that contains "Supported: 170", the element will immediately (in
>> addition to anything else that it would do with the request) send a 170
>> response upstream, containing the request as its body (media type
>> message/sipfrag).
> 
> 
> Your idea has merit. I believe Jiri Kuthan was working on something like 
> this a few years ago.

the problem and solution has been quite the same as later appeared with the
sip-frag based draft. (to reiterate: it is a problem for upstream 
troubleshooter
to figure out what's going on downstream in a proxy chain, when it does not
even know where the problem could have occured)

In our proxy implementation we do don't reveal the whole SIP message in 
error replies,
just very few items that are critical to locating the trouble. In 
particular, the
current request  URI, via-count, and upstream IP (from the 
error-producing hop's perspective).
I don't  think these pieces would be ever causing message size concerns 
in the field.

This implementation works indeeed well in the field and is useful. A 
standard would be
better to allow monitoring equipment to provide better troubleshooting
capabilities. Nevertheless, I think that troubleshooting has been 
historically of rather
marginal interest here, so I'm not sure if there is really an action to be
made.


-jiri

for historical records: 
http://tools.ietf.org/id/draft-kuthan-sipping-diag-00.txt
also: a great deal of troubleshooting information can be achieved using
traceroute-like tool, sipsak. It takes no standard changes for its basic
operation. Alas, for most qualified assertions more diagnostic information
would be useful, or at least Warning header-field which would reveal who
declined a request. (maybe a topic for a BCP? and religious discussions
about benefits of being able to troubleshoot a network versus benefits
of being able to hide it? ;-)


> 
> I've also previously suggested a similar provisional response to inform 
> the UAC about retargeting being done by a proxy. In addition to 
> debugging, this helps avoid the unanticipated respondent problem.
> 
> -- 
> Dean
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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Cc: SIP <sip@ietf.org>
Subject: Re: [Sip] Diagnostic idea
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Robert Sparks wrote:
> Won't this run into exactly the same response-size obstacle that the 
> previous diagnostics work tripped up on?

IMO, the most compelling piece of information is what one needs to learn
about where an error occured in the SIP chain at all. at least turning
on Warning HF. Ideally revealing current request URI too. This amount
of information helps a lot but does not break message size constraints
in the field.

> It might help to spend some time trying to find some other pattern than 
> trying to use responses for moving the information you're trying to 
> capture around.
> Source-routed requests in the opposite direction? (using content 
> indirection?) Subscribe/Notify? 

I'm a bit worried that SUB-NOT is another piece of complexity which is
not necessarily helpful when a simple transaction fails. (chance is
high SUB-NOT will fail too)

-jiri

> Maybe something else...
> 
> RjS
> 
> On Feb 23, 2009, at 3:37 PM, Dale Worley wrote:
> 
>> I've been considering the following idea for a diagnostic tool, and I'd
>> like to get some feedback on it.
>>
>> The goal is to be able to trace the progress of a SIP request through
>> the network, including seeing the forking structure.  We first need to
>> pick a provisional response codes.  It appears that "170" is not
>> currently used.  This response code is also used as an option-tag for
>> this feature.  The processing is that whenever a SIP element receives a
>> request that contains "Supported: 170", the element will immediately (in
>> addition to anything else that it would do with the request) send a 170
>> response upstream, containing the request as its body (media type
>> message/sipfrag).
>>
>> Dale
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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The proxy only needs to use the top Via that it inserted.

The proxy needs to keep a list of early dialogs associated with each client=
 transacton. As part of that early dialog state, it needs to have a flag to=
 indicate whether or not a 199 was received and forwarded by the proxy so t=
hat no more than one 199 response is sent for a given dialog. When a 1xx re=
sponse that creates an early dialog is received, the proxy adds the early d=
ialog to the list (if it has not already). When the non-2xx final response =
is received, it can generate 199 responses for each early dialog in the lis=
t which has not already had a 199 sent.

________________________________________
From: Christer Holmberg [christer.holmberg@ericsson.com]
Sent: Wednesday, March 11, 2009 9:20 AM
To: Bob Penfield
Cc: sip@ietf.org
Subject: RE: [Sip] I-D Action:draft-ietf-sip-199-06.txt

Hi Bob,

>>>In fact, if the proxy is keeping track of early dialogs, under what
>>>circumstances would it not be "able to identify additional early
>>>dialogs which are terminated by the same non-2xx final response"?
>>
>>Just because the proxy keeps track of early dialogs, it doesn't
>>necessarily know that an error response on early dialog X also
>>terminates early dialog Y. In order to know that, the forking proxy
>>needs to know that early dialogs X and Y come from the same forking
>>proxy further down the path.
>>
>
>Of course it can. If the proxy forwarded the request to Z,
>and Z forked to X and Y, the provisional responses from X and
>Y that created the early dialogs would have the same
>branch-id. The error response from Z would have that same
>branch-id. The proxy would know that all early dialogs
>created by a 1xx with that branch-id would have terminated
>(i.e. both X and Y).

But, the branch-id is part of the Via, which is removed by each entity
when forwarding the response. So, the forking proxy is never going to
get the branch-id inserted by Z, is it?


>>>Section 8 (199 Early Dialog Terminated) seems out of place.
>>>Shouldn't this be in the introduction? or in an "Overview Of
>>>Operation" between sections 3 and 4?
>>
>>I have tried to use other similar drafts as "template".
>>
>>
>>>The text in Section 9 (Usage with SDP offer/answer) should be in
>>>section 5 (User Agent Server behavior).
>>
>>I think it is good to have a separate chapter. Of course, I could
refer
>>to chapter 9 from chapter 5.
>
>My point is that all too often behavior requirements (MUST,
>SHOULD, MAY, etc.) get buried is separate sections and folks
>can miss them. If all the requirements are in the applicable
>"behavior" section, they are easy to find and developers are less
likely
>to miss them. If you want it to be a separate section, make
>it 5.1. All UAS requirements should be in section 5 or a
>subsection thereof.
>
>>>The statement in section 10 recommending that the UAS send
>>>199 unreliably should be in section 5.
>>>
>>>The statement in section 10 forbidding proxies from sending
>>>199 reliably should be in section 6 (Proxy behavior).
>>
>>Same as for offer/answer.
>>
>>Since the 199 procedures related to offer/answer and reliable
responses
>>are special, I think it is good to have specific chapters for it.
>>
>The requirements/restrictions for reliable provisional
>responses are very important. They could easily be missed by
>having them in a separate chapter in the back of the
>document. Again, all UAS behavior requirements should be in
>section 5 (or sub-section) and all proxy behavior
>requirements should be in chapter 6 (or sub-section).

I'll think of something. But, as you ok with the text as such? It's only
the location of the text which you have an issue with?


Regards,

Christer=

From christer.holmberg@ericsson.com  Tue Mar 24 00:10:01 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Bob Penfield" <BPenfield@acmepacket.com>
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Hi,

=20
>The proxy only needs to use the top Via that it inserted.
>
>The proxy needs to keep a list of early dialogs associated with each =
client transacton. As part of that=20
>early dialog state, it needs to have a flag to indicate whether or not =
a 199 was received and forwarded by=20
>the proxy so that no more than one 199 response is sent for a given =
dialog. When a 1xx response that creates=20
>an early dialog is received, the proxy adds the early dialog to the =
list (if it has not already). When the=20
>non-2xx final response is received, it can generate 199 responses for =
each early dialog in the list which=20
>has not already had a 199 sent.

Yes, but the issue is how the proxy knows whether it has sent a 199 for =
the early dialog indicated in the error response.

The issue is whether the proxy can detect that a final response is =
generated by another forking proxy, and therefor terminate more than one =
early dialog.

Maybe the Via mechanism you describe works... I need to draw some =
figures :)

Regards,

Christer






_______________________________________
From: Christer Holmberg [christer.holmberg@ericsson.com]
Sent: Wednesday, March 11, 2009 9:20 AM
To: Bob Penfield
Cc: sip@ietf.org
Subject: RE: [Sip] I-D Action:draft-ietf-sip-199-06.txt

Hi Bob,

>>>In fact, if the proxy is keeping track of early dialogs, under what
>>>circumstances would it not be "able to identify additional early
>>>dialogs which are terminated by the same non-2xx final response"?
>>
>>Just because the proxy keeps track of early dialogs, it doesn't
>>necessarily know that an error response on early dialog X also
>>terminates early dialog Y. In order to know that, the forking proxy
>>needs to know that early dialogs X and Y come from the same forking
>>proxy further down the path.
>>
>
>Of course it can. If the proxy forwarded the request to Z,
>and Z forked to X and Y, the provisional responses from X and
>Y that created the early dialogs would have the same
>branch-id. The error response from Z would have that same
>branch-id. The proxy would know that all early dialogs
>created by a 1xx with that branch-id would have terminated
>(i.e. both X and Y).

But, the branch-id is part of the Via, which is removed by each entity
when forwarding the response. So, the forking proxy is never going to
get the branch-id inserted by Z, is it?


>>>Section 8 (199 Early Dialog Terminated) seems out of place.
>>>Shouldn't this be in the introduction? or in an "Overview Of
>>>Operation" between sections 3 and 4?
>>
>>I have tried to use other similar drafts as "template".
>>
>>
>>>The text in Section 9 (Usage with SDP offer/answer) should be in
>>>section 5 (User Agent Server behavior).
>>
>>I think it is good to have a separate chapter. Of course, I could
refer
>>to chapter 9 from chapter 5.
>
>My point is that all too often behavior requirements (MUST,
>SHOULD, MAY, etc.) get buried is separate sections and folks
>can miss them. If all the requirements are in the applicable
>"behavior" section, they are easy to find and developers are less
likely
>to miss them. If you want it to be a separate section, make
>it 5.1. All UAS requirements should be in section 5 or a
>subsection thereof.
>
>>>The statement in section 10 recommending that the UAS send
>>>199 unreliably should be in section 5.
>>>
>>>The statement in section 10 forbidding proxies from sending
>>>199 reliably should be in section 6 (Proxy behavior).
>>
>>Same as for offer/answer.
>>
>>Since the 199 procedures related to offer/answer and reliable
responses
>>are special, I think it is good to have specific chapters for it.
>>
>The requirements/restrictions for reliable provisional
>responses are very important. They could easily be missed by
>having them in a separate chapter in the back of the
>document. Again, all UAS behavior requirements should be in
>section 5 (or sub-section) and all proxy behavior
>requirements should be in chapter 6 (or sub-section).

I'll think of something. But, as you ok with the text as such? It's only
the location of the text which you have an issue with?


Regards,

Christer


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From: "Doken, Serhad" <sdoken@qualcomm.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, Dale Worley <dworley@nortel.com>
Date: Tue, 24 Mar 2009 00:17:09 -0700
Thread-Topic: [Sip] Is REFER within dialog part of INVITE or subscription usage? (was RE: SIP INFO)
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Cc: SIP IETF <SIP@ietf.org>, Brett Tate <brett@broadsoft.com>
Subject: Re: [Sip] Is REFER within dialog part of INVITE or subscription	usage? (was RE: SIP INFO)
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Paul Kyzivat
> Sent: Monday, March 23, 2009 6:33 PM
> To: Dale Worley
> Cc: SIP IETF; Brett Tate
> Subject: Re: [Sip] Is REFER within dialog part of INVITE or
> subscription usage? (was RE: SIP INFO)
>=20
>=20
>=20
> Dale Worley wrote:
> > On Mon, 2009-03-23 at 08:18 -0700, Brett Tate wrote:
> >> Everyone: Is REFER within dialog part of INVITE usage, SUBSCRIPTION
> >> usage, or neither?
> >>
> >> I'm asking mainly because of REFER 481 impacts; however the answer
> >> also has potential implications concerning draft-ietf-sip-info-
> events.
> >> RFC 5057 appears to imply that it is part of the subscription usage;
> >> however I'm not positive if my interpretation and/or RFC 5057 is
> >> correct.
> >>
> >> If REFER isn't considered part of the INVITE usage, how should REFER
> >> 481 be interpreted?  More specifically, should receiver of REFER 481
> >> send BYE?
> >
> > I would argue that the REFER is part of the INVITE usage, because
> REFER
> > in that context is intimately related to the INVITE usage -- it is
> > intended to transfer the INVITE usage/dialog transferred to a
> different
> > destination.
>=20
> Well, I guess there is that implied association. (Part of the "do what
> *I* had in mind" approach that REFER has.) Since this was never spelled
> out, it could also be simply "while this isn't part of an INVITE usage,
> if there happens to be an INVITE usage, please refer that one."
>=20
> I guess the question is: would it be valid to send a REFER within an
> existing dialog that had no INVITE usage? If it were valid, presumably

Yes, I think so.


> it would have the same meaning as sending one outside a dialog, except
> that the resulting subscription would share the dialog. I expect that
> is
> a usage it would be difficult to find in the wild.

Consider a third party control application that forces an endpoint to regis=
ter :

REFER sip:phone.doken.com SIP/2.0
Refer-To: <sip:serhad@doken.com;method=3DREGISTER>

that triggers a REG to the registrar as :

REGISTER sip:serhad@doken.com SIP/2.0

>=20
> Ah, but now I want to reconsider. Remember that REFER can take a method
> name. So, rather than referring an INVITE, I could be referring
> MESSAGE,
> OPTIONS, BYE, or maybe (????) SUBSCRIBE. In some of those cases, the
> association to the INVITE usage is probably meaningless. (Of course its
> meaningful for BYE.)
>=20
> > And it seems quite safe that if the REFER receives a 481 response,
> the
> > INVITE usage must not be functional any more, since the far end of
> that
> > usage has denied that it can act on the REFER.
>=20
> IIRC, 5057 decided that each dialog usage must get its own 481 before
> it
> goes away. I don't think there are any cases where a single 481
> response
> can make multiple usages go away.
>=20
> So the important question is whether the REFER is part of the INVITE
> usage and so its 481 takes down the INVITE usage or not. That gets
> especially interesting if  the dialog already had both an INVITE usage
> and a SUBSCRIBE usage. Does it make sense that the refer take down the
> INVITE usage and leave up the SUBSCRIBE usage?
>=20
> > Now it's possible that the REFER, if it is successful, is the first
> > transaction of the subscription usage.  But I don't think that has
> any
> > paradoxical consequences, as by hypothesis, the REFER succeeded.
>=20
> Yeah, its already so weird, why not a little more weird?
>=20
> Also, I think it has recently been clarified that the 200 of a
> SUBSCRIBE
> should not be considered to create a dialog - that its the NOTIFY that
> creates the dialog. If so, then I suppose the same should be true of
> REFER.
>=20
> Bottom line: I can live with it either way, but think it should be
> documented one way or the other.
>=20
> 	Thanks,
> 	Paul
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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From: "Doken, Serhad" <sdoken@qualcomm.com>
To: aayush bhatnagar <abhatnagar192006@gmail.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Tue, 24 Mar 2009 00:31:47 -0700
Thread-Topic: [Sip] Is REFER within dialog part of INVITE or subscription usage? (was RE: SIP INFO)
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Cc: SIP IETF <SIP@ietf.org>, Brett Tate <brett@broadsoft.com>
Subject: Re: [Sip] Is REFER within dialog part of INVITE or subscription	usage? (was RE: SIP INFO)
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> aayush bhatnagar
> Sent: Monday, March 23, 2009 8:30 PM
> To: Paul Kyzivat
> Cc: SIP IETF; Brett Tate
> Subject: Re: [Sip] Is REFER within dialog part of INVITE or
> subscription usage? (was RE: SIP INFO)
>=20
> REFER if placed within the scope of the existing dialog usage of
> INVITE will reuse its dialog usage, and create an implicit
> subscription for the refer event package at the recipient. However, if
> the recipient chooses to reject the in dialog REFER request, the
> communication established by the INVITE should not be affected. Such a
> rejection scenario implies that the call transfer failed..however the
> existing active call should not be torn down.
>=20
> REFER if sent outside the scope of any exisiting dialog will behave as
> a dialog creating request and will establish a subscription at the
> recipient as described above. This subscription is only used to notify
> the status of the sip session referral to the originator and is
> seperate from any other subscribe usage that may have existed at the
> recipient.

One can send a out of dialog, norefersub REFER and no dialog is established=
 at the recipient. Only an implicit subscription is created and a NTFY may =
be sent or not sent at all.

>=20
> As far as accessing the referred resource is concerned...using the
> method parameter, it depends upon the business logic of the recipient
> of the REFER request. You may send a REFER to a conferencing server
> with a method INVITE to invite a user to join a conference. A method
> parameter with the value BYE would imply removal from the conference
> by the server and so on.
>=20
> I dont think you can include a SUBSCRIBE in the method parameter of
> the REFER request. You have no way of telling the event package usage
> of the subscription to the recipient, as the event header cannot be

It should be possible to do Refer-To: sip:serhad@doken.com;method=3DSUBSCRI=
BE?Event=3Dpresence asking REFER receiver to send me a SUBSCRIBE in order t=
o find out about my presence.

> included in the REFER request. Moreover, you can never be sure whether
> the subscription usage is even applicable to the recipient or whether
> the recipient even supports it.
>=20
> aayush
>=20
> On 3/24/09, Paul Kyzivat <pkyzivat@cisco.com> wrote:
> >
> >
> > Dale Worley wrote:
> >> On Mon, 2009-03-23 at 08:18 -0700, Brett Tate wrote:
> >>> Everyone: Is REFER within dialog part of INVITE usage, SUBSCRIPTION
> >>> usage, or neither?
> >>>
> >>> I'm asking mainly because of REFER 481 impacts; however the answer
> >>> also has potential implications concerning draft-ietf-sip-info-
> events.
> >>> RFC 5057 appears to imply that it is part of the subscription
> usage;
> >>> however I'm not positive if my interpretation and/or RFC 5057 is
> >>> correct.
> >>>
> >>> If REFER isn't considered part of the INVITE usage, how should
> REFER
> >>> 481 be interpreted?  More specifically, should receiver of REFER
> 481
> >>> send BYE?
> >>
> >> I would argue that the REFER is part of the INVITE usage, because
> REFER
> >> in that context is intimately related to the INVITE usage -- it is
> >> intended to transfer the INVITE usage/dialog transferred to a
> different
> >> destination.
> >
> > Well, I guess there is that implied association. (Part of the "do
> what
> > *I* had in mind" approach that REFER has.) Since this was never
> spelled
> > out, it could also be simply "while this isn't part of an INVITE
> usage,
> > if there happens to be an INVITE usage, please refer that one."
> >
> > I guess the question is: would it be valid to send a REFER within an
> > existing dialog that had no INVITE usage? If it were valid,
> presumably
> > it would have the same meaning as sending one outside a dialog,
> except
> > that the resulting subscription would share the dialog. I expect that
> is
> > a usage it would be difficult to find in the wild.
> >
> > Ah, but now I want to reconsider. Remember that REFER can take a
> method
> > name. So, rather than referring an INVITE, I could be referring
> MESSAGE,
> > OPTIONS, BYE, or maybe (????) SUBSCRIBE. In some of those cases, the
> > association to the INVITE usage is probably meaningless. (Of course
> its
> > meaningful for BYE.)
> >
> >> And it seems quite safe that if the REFER receives a 481 response,
> the
> >> INVITE usage must not be functional any more, since the far end of
> that
> >> usage has denied that it can act on the REFER.
> >
> > IIRC, 5057 decided that each dialog usage must get its own 481 before
> it
> > goes away. I don't think there are any cases where a single 481
> response
> > can make multiple usages go away.
> >
> > So the important question is whether the REFER is part of the INVITE
> > usage and so its 481 takes down the INVITE usage or not. That gets
> > especially interesting if  the dialog already had both an INVITE
> usage
> > and a SUBSCRIBE usage. Does it make sense that the refer take down
> the
> > INVITE usage and leave up the SUBSCRIBE usage?
> >
> >> Now it's possible that the REFER, if it is successful, is the first
> >> transaction of the subscription usage.  But I don't think that has
> any
> >> paradoxical consequences, as by hypothesis, the REFER succeeded.
> >
> > Yeah, its already so weird, why not a little more weird?
> >
> > Also, I think it has recently been clarified that the 200 of a
> SUBSCRIBE
> > should not be considered to create a dialog - that its the NOTIFY
> that
> > creates the dialog. If so, then I suppose the same should be true of
> REFER.
> >
> > Bottom line: I can live with it either way, but think it should be
> > documented one way or the other.
> >
> > 	Thanks,
> > 	Paul
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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From: "Doken, Serhad" <sdoken@qualcomm.com>
To: Robert Sparks <rjsparks@nostrum.com>, SIP List <sip@ietf.org>
Date: Tue, 24 Mar 2009 00:39:04 -0700
Thread-Topic: [Sip] Fwd: I-D Action:draft-sparks-sip-3261-interop-statement-00.txt
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Couple items for the list that popped to my mind (based on my share of inte=
rop issues I've seen) :


 *   Demonstrate correct recognition of when to retry requests and when to =
give up (401/423/403/480/500/503 etc use cases)
 *   Demonstrate correct implementations of retrying requests after the spe=
cified time internal when servers are busy/overloaded etc

Serhad

________________________________
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Rober=
t Sparks
Sent: Monday, March 02, 2009 12:19 PM
To: SIP List
Subject: [Sip] Fwd: I-D Action:draft-sparks-sip-3261-interop-statement-00.t=
xt

Also split out of steps-to-draft

Begin forwarded message:


From: Internet-Drafts@ietf.org<mailto:Internet-Drafts@ietf.org>
Date: March 2, 2009 8:30:02 AM CST
To: i-d-announce@ietf.org<mailto:i-d-announce@ietf.org>
Subject: I-D Action:draft-sparks-sip-3261-interop-statement-00.txt
Reply-To: internet-drafts@ietf.org<mailto:internet-drafts@ietf.org>

A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.

          Title           : RFC3261 Interop Statement
          Author(s)       : R. Sparks
          Filename        : draft-sparks-sip-3261-interop-statement-00.txt
          Pages           : 9
          Date            : 2009-03-02

This document captures an outline of the interoperability statements
that will be collected to construct an interoperability report for
RFC 3261.  The outline is stil under review and should not be treated
as complete, but will drive data collection at upcoming
interoperability events.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-sparks-sip-3261-interop-statement=
-00.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

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</head>

<body lang=3DEN-US link=3Dblue vlink=3Dblue>

<div class=3DSection1>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif";
color:navy'>Couple items for the list that popped to my mind (based on my s=
hare
of interop issues I&#8217;ve seen) :<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif";
color:navy'><o:p>&nbsp;</o:p></span></p>

<ul style=3D'margin-top:0in' type=3Ddisc>
 <li class=3DMsoNormal style=3D'color:navy;mso-list:l0 level1 lfo3'><span
     style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Demonstrat=
e
     correct recognition of when to retry requests and when to give up
     (401/423/403/480/500/503 etc use cases)<o:p></o:p></span></li>
 <li class=3DMsoNormal style=3D'color:navy;mso-list:l0 level1 lfo3'><span
     style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Demonstrat=
e
     correct implementations of retrying requests after the specified time
     internal when servers are busy/overloaded etc<o:p></o:p></span></li>
</ul>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif";
color:navy'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif";
color:navy'>Serhad<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif";
color:navy'><o:p>&nbsp;</o:p></span></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</div>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma=
","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <b>On Behalf Of </b>Robe=
rt
Sparks<br>
<b>Sent:</b> Monday, March 02, 2009 12:19 PM<br>
<b>To:</b> SIP List<br>
<b>Subject:</b> [Sip] Fwd: I-D
Action:draft-sparks-sip-3261-interop-statement-00.txt</span><o:p></o:p></p>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal>Also split out of steps-to-draft<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal>Begin forwarded message:<o:p></o:p></p>

</div>

<p class=3DMsoNormal><br>
<br>
<o:p></o:p></p>

<div>

<div>

<p class=3DMsoNormal><b><span style=3D'font-size:9.0pt;font-family:"Helveti=
ca","sans-serif";
color:black'>From: </span></b><span style=3D'font-size:9.0pt;font-family:"H=
elvetica","sans-serif"'><a
href=3D"mailto:Internet-Drafts@ietf.org">Internet-Drafts@ietf.org</a></span=
><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><b><span style=3D'font-size:9.0pt;font-family:"Helveti=
ca","sans-serif";
color:black'>Date: </span></b><span style=3D'font-size:9.0pt;font-family:"H=
elvetica","sans-serif"'>March
2, 2009 8:30:02 AM CST</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><b><span style=3D'font-size:9.0pt;font-family:"Helveti=
ca","sans-serif";
color:black'>To: </span></b><span style=3D'font-size:9.0pt;font-family:"Hel=
vetica","sans-serif"'><a
href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a></span><o:p>=
</o:p></p>

</div>

<div>

<p class=3DMsoNormal><b><span style=3D'font-size:9.0pt;font-family:"Helveti=
ca","sans-serif";
color:black'>Subject: </span></b><b><span style=3D'font-size:9.0pt;font-fam=
ily:
"Helvetica","sans-serif"'>I-D
Action:draft-sparks-sip-3261-interop-statement-00.txt<span
class=3Dapple-converted-space>&nbsp;</span></span></b><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><b><span style=3D'font-size:9.0pt;font-family:"Helveti=
ca","sans-serif";
color:black'>Reply-To: </span></b><span style=3D'font-size:9.0pt;font-famil=
y:
"Helvetica","sans-serif"'><a href=3D"mailto:internet-drafts@ietf.org">inter=
net-drafts@ietf.org</a></span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</div>

<div>

<p class=3DMsoNormal>A New Internet-Draft is available from the on-line
Internet-Drafts directories.<br>
<br>
<span class=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; </span>Title
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: RFC3261 Inter=
op
Statement<br>
<span class=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; </span>Author(s)
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: R. Sparks<br>
<span class=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; </span>Filename
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;:
draft-sparks-sip-3261-interop-statement-00.txt<br>
<span class=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; </span>Pages
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: 9<br>
<span class=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; </span>Date
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: 2009-03=
-02<br>
<br>
This document captures an outline of the interoperability statements<br>
that will be collected to construct an interoperability report for<br>
RFC 3261. &nbsp;The outline is stil under review and should not be treated<=
br>
as complete, but will drive data collection at upcoming<br>
interoperability events.<br>
<br>
A URL for this Internet-Draft is:<br>
<a
href=3D"http://www.ietf.org/internet-drafts/draft-sparks-sip-3261-interop-s=
tatement-00.txt">http://www.ietf.org/internet-drafts/draft-sparks-sip-3261-=
interop-statement-00.txt</a><br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/">ftp://ftp.ietf.org/internet=
-drafts/</a><br>
<br>
Below is the data which will enable a MIME compliant mail reader<br>
implementation to automatically retrieve the ASCII version of the<br>
Internet-Draft.<o:p></o:p></p>

</div>

</div>

</div>

</div>

</body>

</html>

--_000_ED88AAAE8B3D764B9FD8558DE1775B69139274F554NASANEXMB09na_--

From abhatnagar192006@gmail.com  Tue Mar 24 01:18:12 2009
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In-Reply-To: <ED88AAAE8B3D764B9FD8558DE1775B69139274F553@NASANEXMB09.na.qualcomm.com>
References: <24ACC38D-8B7D-46E6-B767-73A33F6C9266@standardstrack.com> <49C79CFD.1010408@cisco.com> <9B2A061A1137254BBE4F4B2CD843646A10BFF48DF0@mbx02.citservers.local> <1237853234.16613.41.camel@victoria-pingtel-com.us.nortel.com> <49C83831.7010102@cisco.com> <abcc59880903232030r413b6856q7c77cf68df7ca564@mail.gmail.com> <ED88AAAE8B3D764B9FD8558DE1775B69139274F553@NASANEXMB09.na.qualcomm.com>
Date: Tue, 24 Mar 2009 13:49:00 +0530
Message-ID: <abcc59880903240119o54625b36mf2884b57db92b350@mail.gmail.com>
From: aayush bhatnagar <abhatnagar192006@gmail.com>
To: "Doken, Serhad" <sdoken@qualcomm.com>
Content-Type: multipart/alternative; boundary=001636e0a64a01a6980465d90980
Cc: SIP IETF <SIP@ietf.org>, Paul Kyzivat <pkyzivat@cisco.com>, Brett Tate <brett@broadsoft.com>
Subject: Re: [Sip] Is REFER within dialog part of INVITE or subscription usage? (was RE: SIP INFO)
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--001636e0a64a01a6980465d90980
Content-Type: text/plain; charset=ISO-8859-1
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>One can send a out of dialog, norefersub REFER and no dialog is established
at the recipient. >Only an implicit subscription is created and a NTFY may
be sent or not sent at all.
you are right.

>It should be possible to do Refer-To: sip:serhad@doken.com<sip%3Aserhad@doken.com>;method=SUBSCRIBE?>Event=presence
asking REFER receiver to send me a SUBSCRIBE in order to find out about >my
presence.

In case the receiver is a server in the network, it must implement the role
of a PNA. If its an endpoint, it will subscribe only after approval of the
human user, else respond back with 400. I am not sure how many presence
clients support forced subscriptions like this.

On Tue, Mar 24, 2009 at 1:01 PM, Doken, Serhad <sdoken@qualcomm.com> wrote:

>
>
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> > aayush bhatnagar
> > Sent: Monday, March 23, 2009 8:30 PM
> > To: Paul Kyzivat
> > Cc: SIP IETF; Brett Tate
> > Subject: Re: [Sip] Is REFER within dialog part of INVITE or
> > subscription usage? (was RE: SIP INFO)
> >
> > REFER if placed within the scope of the existing dialog usage of
> > INVITE will reuse its dialog usage, and create an implicit
> > subscription for the refer event package at the recipient. However, if
> > the recipient chooses to reject the in dialog REFER request, the
> > communication established by the INVITE should not be affected. Such a
> > rejection scenario implies that the call transfer failed..however the
> > existing active call should not be torn down.
> >
> > REFER if sent outside the scope of any exisiting dialog will behave as
> > a dialog creating request and will establish a subscription at the
> > recipient as described above. This subscription is only used to notify
> > the status of the sip session referral to the originator and is
> > seperate from any other subscribe usage that may have existed at the
> > recipient.
>
> One can send a out of dialog, norefersub REFER and no dialog is established
> at the recipient. Only an implicit subscription is created and a NTFY may be
> sent or not sent at all.
>
> >
> > As far as accessing the referred resource is concerned...using the
> > method parameter, it depends upon the business logic of the recipient
> > of the REFER request. You may send a REFER to a conferencing server
> > with a method INVITE to invite a user to join a conference. A method
> > parameter with the value BYE would imply removal from the conference
> > by the server and so on.
> >
> > I dont think you can include a SUBSCRIBE in the method parameter of
> > the REFER request. You have no way of telling the event package usage
> > of the subscription to the recipient, as the event header cannot be
>
> It should be possible to do Refer-To: sip:serhad@doken.com<sip%3Aserhad@doken.com>;method=SUBSCRIBE?Event=presence
> asking REFER receiver to send me a SUBSCRIBE in order to find out about my
> presence.
>
> > included in the REFER request. Moreover, you can never be sure whether
> > the subscription usage is even applicable to the recipient or whether
> > the recipient even supports it.
> >
> > aayush
> >
> > On 3/24/09, Paul Kyzivat <pkyzivat@cisco.com> wrote:
> > >
> > >
> > > Dale Worley wrote:
> > >> On Mon, 2009-03-23 at 08:18 -0700, Brett Tate wrote:
> > >>> Everyone: Is REFER within dialog part of INVITE usage, SUBSCRIPTION
> > >>> usage, or neither?
> > >>>
> > >>> I'm asking mainly because of REFER 481 impacts; however the answer
> > >>> also has potential implications concerning draft-ietf-sip-info-
> > events.
> > >>> RFC 5057 appears to imply that it is part of the subscription
> > usage;
> > >>> however I'm not positive if my interpretation and/or RFC 5057 is
> > >>> correct.
> > >>>
> > >>> If REFER isn't considered part of the INVITE usage, how should
> > REFER
> > >>> 481 be interpreted?  More specifically, should receiver of REFER
> > 481
> > >>> send BYE?
> > >>
> > >> I would argue that the REFER is part of the INVITE usage, because
> > REFER
> > >> in that context is intimately related to the INVITE usage -- it is
> > >> intended to transfer the INVITE usage/dialog transferred to a
> > different
> > >> destination.
> > >
> > > Well, I guess there is that implied association. (Part of the "do
> > what
> > > *I* had in mind" approach that REFER has.) Since this was never
> > spelled
> > > out, it could also be simply "while this isn't part of an INVITE
> > usage,
> > > if there happens to be an INVITE usage, please refer that one."
> > >
> > > I guess the question is: would it be valid to send a REFER within an
> > > existing dialog that had no INVITE usage? If it were valid,
> > presumably
> > > it would have the same meaning as sending one outside a dialog,
> > except
> > > that the resulting subscription would share the dialog. I expect that
> > is
> > > a usage it would be difficult to find in the wild.
> > >
> > > Ah, but now I want to reconsider. Remember that REFER can take a
> > method
> > > name. So, rather than referring an INVITE, I could be referring
> > MESSAGE,
> > > OPTIONS, BYE, or maybe (????) SUBSCRIBE. In some of those cases, the
> > > association to the INVITE usage is probably meaningless. (Of course
> > its
> > > meaningful for BYE.)
> > >
> > >> And it seems quite safe that if the REFER receives a 481 response,
> > the
> > >> INVITE usage must not be functional any more, since the far end of
> > that
> > >> usage has denied that it can act on the REFER.
> > >
> > > IIRC, 5057 decided that each dialog usage must get its own 481 before
> > it
> > > goes away. I don't think there are any cases where a single 481
> > response
> > > can make multiple usages go away.
> > >
> > > So the important question is whether the REFER is part of the INVITE
> > > usage and so its 481 takes down the INVITE usage or not. That gets
> > > especially interesting if  the dialog already had both an INVITE
> > usage
> > > and a SUBSCRIBE usage. Does it make sense that the refer take down
> > the
> > > INVITE usage and leave up the SUBSCRIBE usage?
> > >
> > >> Now it's possible that the REFER, if it is successful, is the first
> > >> transaction of the subscription usage.  But I don't think that has
> > any
> > >> paradoxical consequences, as by hypothesis, the REFER succeeded.
> > >
> > > Yeah, its already so weird, why not a little more weird?
> > >
> > > Also, I think it has recently been clarified that the 200 of a
> > SUBSCRIBE
> > > should not be considered to create a dialog - that its the NOTIFY
> > that
> > > creates the dialog. If so, then I suppose the same should be true of
> > REFER.
> > >
> > > Bottom line: I can live with it either way, but think it should be
> > > documented one way or the other.
> > >
> > >     Thanks,
> > >     Paul
> > > _______________________________________________
> > > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the application of sip
> > >
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
>

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&gt;One can send a out of dialog, norefersub REFER and no dialog is establi=
shed at the recipient. &gt;Only an implicit subscription is created and a N=
TFY may be sent or not sent at all.<div><br></div><div>you are right.=A0</d=
iv>
<div><br></div><div>&gt;It should be possible to do Refer-To:=A0<a href=3D"=
mailto:sip%3Aserhad@doken.com">sip:serhad@doken.com</a>;method=3DSUBSCRIBE?=
&gt;Event=3Dpresence asking REFER receiver to send me a SUBSCRIBE in order =
to find out about &gt;my presence.</div>
<div><br></div><div>In case the receiver is a server in the network, it mus=
t implement the role of a PNA. If its an endpoint, it will subscribe only a=
fter approval of the human user, else respond back with 400. I am not sure =
how many presence clients support forced subscriptions like this.=A0<br>
<br><div class=3D"gmail_quote">On Tue, Mar 24, 2009 at 1:01 PM, Doken, Serh=
ad <span dir=3D"ltr">&lt;<a href=3D"mailto:sdoken@qualcomm.com">sdoken@qual=
comm.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div class=3D"im"><br>
<br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a>=
 [mailto:<a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a>] =
On Behalf Of<br>
</div><div class=3D"im">&gt; aayush bhatnagar<br>
&gt; Sent: Monday, March 23, 2009 8:30 PM<br>
&gt; To: Paul Kyzivat<br>
&gt; Cc: SIP IETF; Brett Tate<br>
&gt; Subject: Re: [Sip] Is REFER within dialog part of INVITE or<br>
&gt; subscription usage? (was RE: SIP INFO)<br>
&gt;<br>
</div><div class=3D"im">&gt; REFER if placed within the scope of the existi=
ng dialog usage of<br>
&gt; INVITE will reuse its dialog usage, and create an implicit<br>
&gt; subscription for the refer event package at the recipient. However, if=
<br>
&gt; the recipient chooses to reject the in dialog REFER request, the<br>
&gt; communication established by the INVITE should not be affected. Such a=
<br>
&gt; rejection scenario implies that the call transfer failed..however the<=
br>
&gt; existing active call should not be torn down.<br>
&gt;<br>
&gt; REFER if sent outside the scope of any exisiting dialog will behave as=
<br>
&gt; a dialog creating request and will establish a subscription at the<br>
&gt; recipient as described above. This subscription is only used to notify=
<br>
&gt; the status of the sip session referral to the originator and is<br>
&gt; seperate from any other subscribe usage that may have existed at the<b=
r>
&gt; recipient.<br>
<br>
</div>One can send a out of dialog, norefersub REFER and no dialog is estab=
lished at the recipient. Only an implicit subscription is created and a NTF=
Y may be sent or not sent at all.<br>
<div class=3D"im"><br>
&gt;<br>
&gt; As far as accessing the referred resource is concerned...using the<br>
&gt; method parameter, it depends upon the business logic of the recipient<=
br>
&gt; of the REFER request. You may send a REFER to a conferencing server<br=
>
&gt; with a method INVITE to invite a user to join a conference. A method<b=
r>
&gt; parameter with the value BYE would imply removal from the conference<b=
r>
&gt; by the server and so on.<br>
&gt;<br>
&gt; I dont think you can include a SUBSCRIBE in the method parameter of<br=
>
&gt; the REFER request. You have no way of telling the event package usage<=
br>
&gt; of the subscription to the recipient, as the event header cannot be<br=
>
<br>
</div>It should be possible to do Refer-To: <a href=3D"mailto:sip%3Aserhad@=
doken.com">sip:serhad@doken.com</a>;method=3DSUBSCRIBE?Event=3Dpresence ask=
ing REFER receiver to send me a SUBSCRIBE in order to find out about my pre=
sence.<br>

<div><div></div><div class=3D"h5"><br>
&gt; included in the REFER request. Moreover, you can never be sure whether=
<br>
&gt; the subscription usage is even applicable to the recipient or whether<=
br>
&gt; the recipient even supports it.<br>
&gt;<br>
&gt; aayush<br>
&gt;<br>
&gt; On 3/24/09, Paul Kyzivat &lt;<a href=3D"mailto:pkyzivat@cisco.com">pky=
zivat@cisco.com</a>&gt; wrote:<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; Dale Worley wrote:<br>
&gt; &gt;&gt; On Mon, 2009-03-23 at 08:18 -0700, Brett Tate wrote:<br>
&gt; &gt;&gt;&gt; Everyone: Is REFER within dialog part of INVITE usage, SU=
BSCRIPTION<br>
&gt; &gt;&gt;&gt; usage, or neither?<br>
&gt; &gt;&gt;&gt;<br>
&gt; &gt;&gt;&gt; I&#39;m asking mainly because of REFER 481 impacts; howev=
er the answer<br>
&gt; &gt;&gt;&gt; also has potential implications concerning draft-ietf-sip=
-info-<br>
&gt; events.<br>
&gt; &gt;&gt;&gt; RFC 5057 appears to imply that it is part of the subscrip=
tion<br>
&gt; usage;<br>
&gt; &gt;&gt;&gt; however I&#39;m not positive if my interpretation and/or =
RFC 5057 is<br>
&gt; &gt;&gt;&gt; correct.<br>
&gt; &gt;&gt;&gt;<br>
&gt; &gt;&gt;&gt; If REFER isn&#39;t considered part of the INVITE usage, h=
ow should<br>
&gt; REFER<br>
&gt; &gt;&gt;&gt; 481 be interpreted? =A0More specifically, should receiver=
 of REFER<br>
&gt; 481<br>
&gt; &gt;&gt;&gt; send BYE?<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; I would argue that the REFER is part of the INVITE usage, bec=
ause<br>
&gt; REFER<br>
&gt; &gt;&gt; in that context is intimately related to the INVITE usage -- =
it is<br>
&gt; &gt;&gt; intended to transfer the INVITE usage/dialog transferred to a=
<br>
&gt; different<br>
&gt; &gt;&gt; destination.<br>
&gt; &gt;<br>
&gt; &gt; Well, I guess there is that implied association. (Part of the &qu=
ot;do<br>
&gt; what<br>
&gt; &gt; *I* had in mind&quot; approach that REFER has.) Since this was ne=
ver<br>
&gt; spelled<br>
&gt; &gt; out, it could also be simply &quot;while this isn&#39;t part of a=
n INVITE<br>
&gt; usage,<br>
&gt; &gt; if there happens to be an INVITE usage, please refer that one.&qu=
ot;<br>
&gt; &gt;<br>
&gt; &gt; I guess the question is: would it be valid to send a REFER within=
 an<br>
&gt; &gt; existing dialog that had no INVITE usage? If it were valid,<br>
&gt; presumably<br>
&gt; &gt; it would have the same meaning as sending one outside a dialog,<b=
r>
&gt; except<br>
&gt; &gt; that the resulting subscription would share the dialog. I expect =
that<br>
&gt; is<br>
&gt; &gt; a usage it would be difficult to find in the wild.<br>
&gt; &gt;<br>
&gt; &gt; Ah, but now I want to reconsider. Remember that REFER can take a<=
br>
&gt; method<br>
&gt; &gt; name. So, rather than referring an INVITE, I could be referring<b=
r>
&gt; MESSAGE,<br>
&gt; &gt; OPTIONS, BYE, or maybe (????) SUBSCRIBE. In some of those cases, =
the<br>
&gt; &gt; association to the INVITE usage is probably meaningless. (Of cour=
se<br>
&gt; its<br>
&gt; &gt; meaningful for BYE.)<br>
&gt; &gt;<br>
&gt; &gt;&gt; And it seems quite safe that if the REFER receives a 481 resp=
onse,<br>
&gt; the<br>
&gt; &gt;&gt; INVITE usage must not be functional any more, since the far e=
nd of<br>
&gt; that<br>
&gt; &gt;&gt; usage has denied that it can act on the REFER.<br>
&gt; &gt;<br>
&gt; &gt; IIRC, 5057 decided that each dialog usage must get its own 481 be=
fore<br>
&gt; it<br>
&gt; &gt; goes away. I don&#39;t think there are any cases where a single 4=
81<br>
&gt; response<br>
&gt; &gt; can make multiple usages go away.<br>
&gt; &gt;<br>
&gt; &gt; So the important question is whether the REFER is part of the INV=
ITE<br>
&gt; &gt; usage and so its 481 takes down the INVITE usage or not. That get=
s<br>
&gt; &gt; especially interesting if =A0the dialog already had both an INVIT=
E<br>
&gt; usage<br>
&gt; &gt; and a SUBSCRIBE usage. Does it make sense that the refer take dow=
n<br>
&gt; the<br>
&gt; &gt; INVITE usage and leave up the SUBSCRIBE usage?<br>
&gt; &gt;<br>
&gt; &gt;&gt; Now it&#39;s possible that the REFER, if it is successful, is=
 the first<br>
&gt; &gt;&gt; transaction of the subscription usage. =A0But I don&#39;t thi=
nk that has<br>
&gt; any<br>
&gt; &gt;&gt; paradoxical consequences, as by hypothesis, the REFER succeed=
ed.<br>
&gt; &gt;<br>
&gt; &gt; Yeah, its already so weird, why not a little more weird?<br>
&gt; &gt;<br>
&gt; &gt; Also, I think it has recently been clarified that the 200 of a<br=
>
&gt; SUBSCRIBE<br>
&gt; &gt; should not be considered to create a dialog - that its the NOTIFY=
<br>
&gt; that<br>
&gt; &gt; creates the dialog. If so, then I suppose the same should be true=
 of<br>
&gt; REFER.<br>
&gt; &gt;<br>
&gt; &gt; Bottom line: I can live with it either way, but think it should b=
e<br>
&gt; &gt; documented one way or the other.<br>
&gt; &gt;<br>
&gt; &gt; =A0 =A0 Thanks,<br>
&gt; &gt; =A0 =A0 Paul<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; Sip mailing list =A0<a href=3D"https://www.ietf.org/mailman/listi=
nfo/sip" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br=
>
&gt; &gt; This list is for NEW development of the core SIP Protocol<br>
&gt; &gt; Use <a href=3D"mailto:sip-implementors@cs.columbia.edu">sip-imple=
mentors@cs.columbia.edu</a> for questions on current sip<br>
&gt; &gt; Use <a href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a> for =
new developments on the application of sip<br>
&gt; &gt;<br>
&gt; _______________________________________________<br>
&gt; Sip mailing list =A0<a href=3D"https://www.ietf.org/mailman/listinfo/s=
ip" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
&gt; This list is for NEW development of the core SIP Protocol<br>
&gt; Use <a href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implemento=
rs@cs.columbia.edu</a> for questions on current sip<br>
&gt; Use <a href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a> for new d=
evelopments on the application of sip<br>
</div></div></blockquote></div><br></div>

--001636e0a64a01a6980465d90980--

From HKaplan@acmepacket.com  Tue Mar 24 02:05:35 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Jiri Kuthan <jiri@iptel.org>, Dean Willis <dean.willis@softarmor.com>
Date: Tue, 24 Mar 2009 05:06:25 -0400
Thread-Topic: [Sip] Diagnostic idea
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Jir=
i
> Kuthan
> Sent: Tuesday, March 24, 2009 12:54 AM
>=20
> Nevertheless, I think that troubleshooting has been
> historically of rather
> marginal interest here, so I'm not sure if there is really an action to b=
e
> made.

I think dealing with the operational aspects of SIP networks, especially wi=
th troubleshooting them, is a topic whose time has come.  There are at leas=
t 3 recent drafts introducing mechanisms to aid in troubleshooting SIP.  Pe=
rsonally I think it's one of the next hurdles in growing SIP's adoption.

>From a DISPATCH draft-routing-to-WG perspective, one could argue we actuall=
y need a SipOps WG.

-hadriel

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Adam Roach <adam@nostrum.com>, "Elwell, John" <john.elwell@siemens.com>
Date: Tue, 24 Mar 2009 09:35:30 -0400
Thread-Topic: [Sip] Comment on draft-roach-sipcore-rfc3265bis-00
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Ada=
m
> Roach
> Sent: Sunday, March 22, 2009 7:24 PM
>=20
> The intention is that any UA capable of sending a NOTIFY message in any
> context will use a GRUU for all dialogs. This mechanism is to be used
> instead of dialog re-use. The most common case of dialog re-use right
> now is sending a SUBSCRIBE down an existing dialog that was initially
> established by an INVITE. So, for current use cases, using GRUU in
> INVITE dialogs is the most important thing. For future-proofing, we need
> to require GRUU for all dialogs.
=20
If the most common use-case for dialog-reuse of SUB/NOT is within INVITE di=
alogs, then what you're really saying is all INVITE-receiving or INVITE-cre=
ating UA's which also support receiving SUBSCRIBE must use GRUU's for the *=
INVITE* dialog to begin with.  That way the SUBSCRIBE can be sent to the sa=
me far-end UA, right?

Do you really think it's realistic to make such a requirement, to require c=
hanges in INVITE-dialogs, for an update to rfc3265 Sub/Not that ostensibly =
is to fix found issues and provide better interop? =20

GRUU is not what one would call a wildly successful mechanism, yet, and we =
don't yet know what all of its deployment issues will be using it in INVITE=
s.  And the most popular implementation we've seen using gruu doesn't even =
use the right param name (it uses an older gruu draft's param name).

-hadriel

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Cc: sip <sip@ietf.org>, Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
Subject: Re: [Sip] Policy to register media feature tags
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Atle Monrad wrote:
> Paul 
> 
> For you information, the tags that 3gpp have specified and registered
> are described in publicly available 3GPP specifications which are
> referenced from the IANA registration. The same apply for the three tags
> that are in the pipeline. The reason for this is to secure openness and
> operability for vendors that want to support certain 3gpp-specific
> capabilities.
> 
> 3GPP would like to get all tags registered by IANA in order to secure
> that the tags are public and thereby the reference to the specifications
> where this particular 3GPP-functionality is specified is publicly
> available. 

I agree that is a good thing.

> To communicate these tags e-2-e is needed to support certain
> 3gpp-capabilities, and I do not really see the linkage towards you
> statement "On the other side were those who prefer to associate
> arbitrary names to collections of features and then negotiate on the
> basis of the names of those collections."

I will take that statement back as being based only on old impressions, 
not on real facts.

BUT, I still think there is reason to have *some* process to determine 
when something is a new independent feature, orthogonal to existing 
features, vs. one that turns out to overlap or conflict with existing 
features.

For instance, I went and looked up the 3gpp tags that have already been 
registered. I found the following in the registration:

   g.3gpp.smsip:
   This feature-tag indicates that the device is capable of accepting
   SMS messages via SIP as specified by ETSI TS 124 341.

Then I went and looked at ETSI TS 124 341, and found:

> 5.3.3.4 Delivering a short message in a SIP MESSAGE request
> 
> If a short message is received from the SMS-GMSC, the IP-SM-GW shall 
> extract the IMSI of the SM-over-IP receiver from the received message. 
> Then the IP-SM-GW shall obtain a corresponding public user identity and 
> deliver a SIP MESSAGE request with the following information:
> 
> a) the Request-URI, which shall contain a public user identity of the 
> SM-over-IP receiver;
> 
> b) the Accept-Contact header, which shall contain a "+g.3gpp.smsip" 
> parameter and the "explicit" and "require" tags according to RFC 3841 
> [17];
> 
> Editor's Note: It is FFS whether the media feature tag value will be 
> hard coded or it will be an operator specific value distributed as part 
> of the service provisioning process.
> 
> c) the Request-Disposition header which shall contain the "no-fork" 
> directive;
> 
> d) the Content-Type header which shall contain 
> "application/vnd.3gpp.sms"; and
> 
> e) the body of the request which shall contain an RP-DATA message as 
> defined in 3GPP TS 24.011 [8], including the SMS headers and the SMS 
> user information encoded as specified in 3GPP TS 23.040 [3].

 From a practical perspective, an SMS is pretty much equivalent to a 
MESSAGE message. Normally it contains *text*, and could be delivered in 
a basic MESSAGE with a body type of text/plain. In that form it could be 
received by anything that can handle a basic MESSAGE message.

Yet the above turns the SMS into a *different* feature, that will *only* 
be delivered to UAs that support that unique, but non-orthogonal, feature.

Even if SMS's aren't always just text, there are models to handle that 
more openly. For instance, multipart/alternative can be used to 
represent the message as text for those who only understand text, and as 
something richer for those who are capable of understanding that.

My point is not to harp on this feature in particular, but only to use 
it as an example. If there were some discussion about new feature tags, 
of whether they are unique or could be factored some other way in order 
to coexist better with existing features, then the potential for interop 
would be improved.

	Thanks,
	Paul


> Hope this clarifies
> 
> Regards
> /atle
> ______
> 
> Atle Monrad
> Standardisation
> 
> ST-Ericsson
> Product Group Mobile Platforms
>  
> Televeien 1
> N-4820, Grimstad
> Norway
> www.stericsson.com
> 
>  
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Paul Kyzivat
> Sent: 20. mars 2009 14:42
> To: Gonzalo Camarillo
> Cc: sip
> Subject: Re: [Sip] Policy to register media feature tags
> 
> Some issues just never go away. :-(
> 
> This seems to be a resurection of issues that first surfaced during the
> initial discussions of extended presence status many years ago. And it
> also harks back to discussions last year or the year before about
> communication service identifiers and application service identifiers.
> 
> The problem is one of openness. Specifically, do you have to be "part of
> the club" in order to properly interoperate with a device that uses
> these identifiers to identify its preferences or capabilities.
> 
> On one side were those who wanted capabilities and preferences to be
> stated in terms of feature tags that are well known and orthogonal. On
> the other side were those who prefer to associate arbitrary names to
> collections of features and then negotiate on the basis of the names of
> those collections.
> 
> Using the latter approach its possible that interoperation will fail
> because the parties don't share a common name for a collection of
> features even though they both possess the necessary features to
> interoperate.
> 
> I don't have a problem with independent groups defining new feature
> tags, as long as they are primitive and orthogonal to existing ones. But
> I do have a problem with defining feature tags that identify collections
> of features, especially when the mapping from the identifier to the
> collection of features is not public.
> 
> The bottom line is that I think this sort of thing needs to be thrashed
> out within the sip community. So I think the RFC process is the right
> process.
> 
> 	Thanks,
> 	Paul
> 
> Gonzalo Camarillo wrote:
>> Folks,
>>
>> as you know, RFC 3840 specifies how to indicate UA's capabilities 
>> using media feature tags. Section 12.1 of RFC 3840 creates the SIP 
>> Media Feature Tag Registration Tree. Media feature tags applicable to 
>> SIP are registered in this tree.
>>
>> http://www.ietf.org/rfc/rfc3840.txt
>>
>> RFC 2506 defines the global tree to register feature tags. Section 
>> 3.1.2 of RFC 2506 says: "A registration may be proposed for the global
> 
>> tree by anyone who has the need to allow for communication on a 
>> particular capability or preference".
>>
>> http://www.ietf.org/rfc/rfc2506.txt
>>
>> The currently registered feature tags in both the SIP and the global 
>> trees can be found at:
>>
>> http://www.iana.org/assignments/media-feature-tags
>>
>> As you can see, the only three feature tags registered in the global 
>> tree come from 3GPP.
>>
>> The issue we are facing is that the SIP and the global trees have 
>> different registration policies. The SIP registry requires an RFC 
>> defining the feature tag while the global tree only requires an expert
> 
>> review.
>>
>> Now, 3GPP would like to register two new feature tags:  
>> g.3gpp.icsi-ref, g.3gpp.iari-ref
>>
>> This is the description of how these feature tags will be used:
>>
>> "Each value of the Application Reference media feature-tag indicates 
>> the software applications supported by the agent. The values for this 
>> tag equal the IMS communication Service Identifier (ICSI) and IMS 
>> Application Reference Identifier (IARI) values supported by the agent.
> 
>> The Application Reference media feature tag is defined to fulfil the 
>> requirements for forking to an appropriate UE when multiple UEs are 
>> registered and dispatch to an appropriate application within the UE 
>> based upon the IMS communication Service Identifier
>> (ICSI) and IMS Application Reference Identifier (IARI) values as 
>> stated in 3GPP TS 23.228.  Multiple tag-values can be included in the 
>> Application Reference media feature-tag."
>>
>>
>> The expert reviewer indicated that this tags will not contain simple 
>> features but rather more complex services, or feature sets, or 
>> application logic (pick your favorite term). Therefore, he did not 
>> find it appropriate to register them in the global tree. He suggested 
>> that RFC 2506 is amended so that it covers this type of more complex
> features.
>> However, before going down that path, I would like to get comments 
>> from the SIP community. The underlying issue is that we need to decide
> 
>> which policy we want to apply to these types of registrations in SIP.
>>
>> 1) We decide that an RFC is always needed. Then, we can just use the 
>> SIP registry. We would probably need to clarify or to stress somewhere
> 
>> that the global tree is not appropriate for SIP tags.
>>
>> 2) We decide that an expert review is good enough. Then we would 
>> probably need to amend RFC 2506 or RFC 3840 to reflect this.
>>
>> Comments are welcome.
>>
>> Thanks,
>>
>> Gonzalo
>>
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use 
>> sip-implementors@cs.columbia.edu for questions on current sip Use 
>> sipping@ietf.org for new developments on the application of sip
>>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use
> sip-implementors@cs.columbia.edu for questions on current sip Use
> sipping@ietf.org for new developments on the application of sip
> 

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Cc: SIP IETF <SIP@ietf.org>, Brett Tate <brett@broadsoft.com>
Subject: Re: [Sip] Is REFER within dialog part of INVITE or subscription usage? (was RE: SIP INFO)
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What you say below is stated as *fact*. But the reason we are having 
this discussion is because its not entirely clear what the proper 
interpretation should be. So I will take your statements as being your 
*opinion* on the subject. Your opinion would be of more value if you 
gave the justifications for it.

More inline.


aayush bhatnagar wrote:
> REFER if placed within the scope of the existing dialog usage of
> INVITE will reuse its dialog usage, and create an implicit
> subscription for the refer event package at the recipient. However, if
> the recipient chooses to reject the in dialog REFER request, the
> communication established by the INVITE should not be affected. Such a
> rejection scenario implies that the call transfer failed..however the
> existing active call should not be torn down.

> REFER if sent outside the scope of any exisiting dialog will behave as
> a dialog creating request and will establish a subscription at the
> recipient as described above. This subscription is only used to notify
> the status of the sip session referral to the originator and is
> seperate from any other subscribe usage that may have existed at the
> recipient.
> 
> As far as accessing the referred resource is concerned...using the
> method parameter, it depends upon the business logic of the recipient
> of the REFER request. You may send a REFER to a conferencing server
> with a method INVITE to invite a user to join a conference. A method
> parameter with the value BYE would imply removal from the conference
> by the server and so on.

It has been observed before, that REFER is very weak in defining clearly 
what actions a particular REFER should elicit. Because it is unclear, 
your are right that some discretion on the part of the recipient is 
required. However that's a bad thing if the sender and receiver of the 
REFER don't agree on what the action should be.

If I send a REFER to a conference focus, I probably expect it send an 
invite to the URI I supply and include the resulting session as another 
participant in the conference. If the focus decided instead that I just 
wanted it play a message to the recipient and then hang up, then I would 
be displeased.

But that is a digression. My point in bringing up the method was that 
the method potentially may have nothing to do with the INVITE usage even 
when the REFER is sent within a dialog that has an INVITE usage. That is 
an argument for saying that the REFER is not part of the INVITE usage.

> I dont think you can include a SUBSCRIBE in the method parameter of
> the REFER request. You have no way of telling the event package usage
> of the subscription to the recipient, as the event header cannot be
> included in the REFER request. Moreover, you can never be sure whether
> the subscription usage is even applicable to the recipient or whether
> the recipient even supports it.

There was already a reply to this.

I think it is certainly possible to see how you could provide enough 
info to enable the recipient to generate a valid SUBSCRIBE request. It 
is indeed harder to imagine how you could expect the recipient to 
extract any value from the resulting subscription. But that is part of 
the magic of REFER that remains unresolved.

	Thanks,
	Paul

> aayush
> 
> On 3/24/09, Paul Kyzivat <pkyzivat@cisco.com> wrote:
>>
>> Dale Worley wrote:
>>> On Mon, 2009-03-23 at 08:18 -0700, Brett Tate wrote:
>>>> Everyone: Is REFER within dialog part of INVITE usage, SUBSCRIPTION
>>>> usage, or neither?
>>>>
>>>> I'm asking mainly because of REFER 481 impacts; however the answer
>>>> also has potential implications concerning draft-ietf-sip-info-events.
>>>> RFC 5057 appears to imply that it is part of the subscription usage;
>>>> however I'm not positive if my interpretation and/or RFC 5057 is
>>>> correct.
>>>>
>>>> If REFER isn't considered part of the INVITE usage, how should REFER
>>>> 481 be interpreted?  More specifically, should receiver of REFER 481
>>>> send BYE?
>>> I would argue that the REFER is part of the INVITE usage, because REFER
>>> in that context is intimately related to the INVITE usage -- it is
>>> intended to transfer the INVITE usage/dialog transferred to a different
>>> destination.
>> Well, I guess there is that implied association. (Part of the "do what
>> *I* had in mind" approach that REFER has.) Since this was never spelled
>> out, it could also be simply "while this isn't part of an INVITE usage,
>> if there happens to be an INVITE usage, please refer that one."
>>
>> I guess the question is: would it be valid to send a REFER within an
>> existing dialog that had no INVITE usage? If it were valid, presumably
>> it would have the same meaning as sending one outside a dialog, except
>> that the resulting subscription would share the dialog. I expect that is
>> a usage it would be difficult to find in the wild.
>>
>> Ah, but now I want to reconsider. Remember that REFER can take a method
>> name. So, rather than referring an INVITE, I could be referring MESSAGE,
>> OPTIONS, BYE, or maybe (????) SUBSCRIBE. In some of those cases, the
>> association to the INVITE usage is probably meaningless. (Of course its
>> meaningful for BYE.)
>>
>>> And it seems quite safe that if the REFER receives a 481 response, the
>>> INVITE usage must not be functional any more, since the far end of that
>>> usage has denied that it can act on the REFER.
>> IIRC, 5057 decided that each dialog usage must get its own 481 before it
>> goes away. I don't think there are any cases where a single 481 response
>> can make multiple usages go away.
>>
>> So the important question is whether the REFER is part of the INVITE
>> usage and so its 481 takes down the INVITE usage or not. That gets
>> especially interesting if  the dialog already had both an INVITE usage
>> and a SUBSCRIBE usage. Does it make sense that the refer take down the
>> INVITE usage and leave up the SUBSCRIBE usage?
>>
>>> Now it's possible that the REFER, if it is successful, is the first
>>> transaction of the subscription usage.  But I don't think that has any
>>> paradoxical consequences, as by hypothesis, the REFER succeeded.
>> Yeah, its already so weird, why not a little more weird?
>>
>> Also, I think it has recently been clarified that the 200 of a SUBSCRIBE
>> should not be considered to create a dialog - that its the NOTIFY that
>> creates the dialog. If so, then I suppose the same should be true of REFER.
>>
>> Bottom line: I can live with it either way, but think it should be
>> documented one way or the other.
>>
>> 	Thanks,
>> 	Paul
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
> 

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Cc: SIP IETF <SIP@ietf.org>, Brett Tate <brett@broadsoft.com>
Subject: Re: [Sip] Is REFER within dialog part of INVITE or	subscription	usage? (was RE: SIP INFO)
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Doken, Serhad wrote:

>> I guess the question is: would it be valid to send a REFER within an
>> existing dialog that had no INVITE usage? If it were valid, presumably
> 
> Yes, I think so.
> 
> 
>> it would have the same meaning as sending one outside a dialog, except
>> that the resulting subscription would share the dialog. I expect that
>> is
>> a usage it would be difficult to find in the wild.
> 
> Consider a third party control application that forces an endpoint to register :
> 
> REFER sip:phone.doken.com SIP/2.0
> Refer-To: <sip:serhad@doken.com;method=REGISTER>
> 
> that triggers a REG to the registrar as :
> 
> REGISTER sip:serhad@doken.com SIP/2.0

You are suggesting that first this 3pcc application sends a SUBSCRIBE to 
this UA, establishing a dialog. Then it sends a REFER within the dialog, 
asking the UA REGISTER? That then would presumably result in a 
subscription to the refer event package sharing the dialog with the 
prior subscription. I guess this is *possible*, though I don't know 
*why* it would be done. What would the initial SUBSCRIBE be to, and how 
would it be established before the UA has registered?

Are you aware of such usage in the wild?

(Its incidental to this discussion, but REGISTER presents unique 
challenges because of the way addressing works for it. Contrary to your 
example above, we would typically expect that the To: would contain the 
full URI, while the R-URI would only be sip:doken.com. This would 
require more special case processing.)

	Thanks,
	Paul

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From: Bob Penfield <BPenfield@acmepacket.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Date: Tue, 24 Mar 2009 12:35:35 -0400
Thread-Topic: [Sip] I-D Action:draft-ietf-sip-199-06.txt - early dialog state in proxy
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Subject: Re: [Sip] I-D Action:draft-ietf-sip-199-06.txt - early dialog state in proxy
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=20
>
>Yes, but the issue is how the proxy knows whether it has sent a 199 for th=
e early=20
>dialog indicated in the error response.

Again, the proxy needs to have a list of early dialogs and within that stat=
e data needs to remember whether or not a 199 has been sent/forwarded to th=
e UAC.

>
>The issue is whether the proxy can detect that a final response is generat=
ed by=20
>another forking proxy, and therefor terminate more than one early dialog.

It does not matter whether the final response came from a forking proxy or =
not. When a final response is received on a client transaction, the proxy h=
as to assume that all downstream forks that the request created have been t=
erminated. RFC 3261 requires that the proxies and UASes send only one non-2=
xx final response, so any downstream proxy that forked the request would no=
t send back a non-2xx final response until all of its forks completed.

>
>Maybe the Via mechanism you describe works... I need to draw some figures =
:)
>

From christer.holmberg@ericsson.com  Tue Mar 24 09:40:14 2009
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Subject: Re: [Sip] I-D Action:draft-ietf-sip-199-06.txt - early dialog state in proxy
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Hi,=20

>>Yes, but the issue is how the proxy knows whether it has sent a 199
for the early dialog indicated in the error=20
>>response.
>
>Again, the proxy needs to have a list of early dialogs and within that
state data needs to remember whether or not a 199=20
>has been sent/forwarded to the UAC.

Again, I agree with you :)

>>The issue is whether the proxy can detect that a final response is=20
>>generated by another forking proxy, and therefor terminate more than
one early dialog.
>
>It does not matter whether the final response came from a forking proxy
or not. When a final response is received on a=20
>client transaction, the proxy has to assume that all downstream forks
that the request created have been terminated. RFC=20
>3261 requires that the proxies and UASes send only one non-2xx final
response, so any downstream proxy that forked the=20
>request would not send back a non-2xx final response until all of its
forks completed.

Yes, and the issue was HOW the proxy knows WHICH downstream forks are
terminated by the single final response. Your proposal is to look at the
Via branch, and I think that may work.

So, I think we're on the same page :)

Regards,

Christer


From BPenfield@acmepacket.com  Tue Mar 24 09:44:54 2009
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From: Bob Penfield <BPenfield@acmepacket.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, Adam Roach <adam@nostrum.com>, "Elwell, John" <john.elwell@siemens.com>
Date: Tue, 24 Mar 2009 12:45:40 -0400
Thread-Topic: [Sip] Comment on draft-roach-sipcore-rfc3265bis-00
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=20

>GRUU is not what one would call a wildly successful mechanism, yet, and we=
 don't yet
>know what all of its deployment issues will be using it in INVITEs.  And t=
he most
>popular implementation we've seen using gruu doesn't even use the right pa=
ram name
>(it uses an older gruu draft's param name).

I think its way too early to make any judgment on the success of GRUU since=
 it is not even an RFC yet. Many vendors have made the smart decision to no=
t implement things in their core products until there is an RFC. When folks=
 start implementing it, INVITE will likely be the first place they do. Main=
ly for call transfer use cases as well as subscriptions related to the INVI=
TE usage.

_______________________________________________
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Subject: Re: [Sip] Comment on draft-roach-sipcore-rfc3265bis-00
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Hadriel Kaplan wrote:
> If the most common use-case for dialog-reuse of SUB/NOT is within INVITE dialogs, then what you're really saying is all INVITE-receiving or INVITE-creating UA's which also support receiving SUBSCRIBE must use GRUU's for the *INVITE* dialog to begin with.  That way the SUBSCRIBE can be sent to the same far-end UA, right?
>   

That's the important part, yes.

> Do you really think it's realistic to make such a requirement, to require changes in INVITE-dialogs, for an update to rfc3265 Sub/Not that ostensibly is to fix found issues and provide better interop?

Yep. We knew, when we put dialog sharing in, that it was a horrible 
hack. I can probably even dig up the fossil record where we explicitly 
agreed (prior to 3265's publication) to go forward with what we 
acknowledged it to be a horrible hack, and further agreed to try to come 
up with a better way to do this in the future. I concede that consensus 
reached eight years ago can be overturned if we have new technical 
information, but this really *has* been the plan all along.

In fact, getting GRUU in the editor's queue was one of the key triggers 
that led me to believe that we finally have the tools necessary to issue 
a bis, since dialog sharing was a known (and necessary) flaw at the time 
of publication.

> GRUU is not what one would call a wildly successful mechanism, yet

Of course it's not. It's not an RFC yet. After the whole multi-year TURN 
whipsaw experience, I'm stunned anyone bothers to implement anything 
before it has an RFC number assigned any more. But, suprisingly, they 
do: more than one in every eight implementations at SIPit 22 implemented 
GRUU.

And, while I'm flattered that you think that the publication and/or the 
implementation of 3265bis will somehow outrun a document that's 
effectively at the finish line already, I think you are being somewhat 
over-optimistic in your evaluation. By the time 3265bis is an RFC, 
implementations will be further along.

/a

From HKaplan@acmepacket.com  Tue Mar 24 10:54:04 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Adam Roach <adam@nostrum.com>
Date: Tue, 24 Mar 2009 13:54:41 -0400
Thread-Topic: [Sip] Comment on draft-roach-sipcore-rfc3265bis-00
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> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com]
> Sent: Tuesday, March 24, 2009 12:53 PM
> To: Hadriel Kaplan
>
> And, while I'm flattered that you think that the publication and/or the
> implementation of 3265bis will somehow outrun a document that's
> effectively at the finish line already, I think you are being somewhat
> over-optimistic in your evaluation. By the time 3265bis is an RFC,
> implementations will be further along.

My concern was more that it has nothing to do with implementations - it has=
 to do with use.  I find many new drafts getting early implementation at Si=
pit's yet not finding them in actual use or even GA code.  Just because we =
publish an RFC does not mean a heck of a lot as to whether the market will =
use it.  :(

-hadriel

From Martin.Thomson@andrew.com  Tue Mar 24 13:00:13 2009
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Subject: [Sip] UA Config Ad Hoc
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The UA Config (SIPPING)  Ad Hoc will be in Franciscan D from 7:45am -  
9am on Wednesday, 3/25.  Grab breakfast first - IETF breakfast is near  
the room.
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Subject: [Sip] SIP-CLF Ad Hoc
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The SIP-CLF (common log format) SIPPING ad hoc will be in Yosemite A  
from 3pm - 4pm on Wednesday 3/25.

Jabber will be on the SIPPING channel.
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From eburger@standardstrack.com  Tue Mar 24 13:47:39 2009
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From: Eric Burger <eburger@standardstrack.com>
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Subject: Re: [Sip] UA Config Ad Hoc
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Jabber will be on the SIPPING channel.

On Mar 24, 2009, at 1:47 PM, Eric Burger wrote:

> The UA Config (SIPPING)  Ad Hoc will be in Franciscan D from 7:45am  
> - 9am on Wednesday, 3/25.  Grab breakfast first - IETF breakfast is  
> near the room.

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Subject: Re: [Sip] FW: [Geopriv] draft-ietf-sip-location-conveyance-13: WGLC comments
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At 03:01 PM 3/24/2009, Thomson, Martin wrote:
>As far as I am aware, the WGLC on this document attracted no 
>response on the SIP WG list.  I can't entirely attribute this to 
>ambivalence, but would be concerned if anyone got the impression 
>that there were no issues on this document.
>
>This is a document that has had more changes recently

The only change that is recent is addressing the "routing-allowed" 
flag, which had very little new next.

the biggest change in the last 4 years was 2 years ago, surrounding 
the errors section (which Hannes now says he wants removed, but the 
rest of the folks at that bar bof (2 years ago) didn't agree with him 
then, and it would leave this spec underspecified... so that's not a 
good option IMO).

I haven't heard any really substantial change requests in the last 2 
years - other than this "routing-allowed" flag issue, which I have 
addressed in the current doc.

>and it is still: (a) difficult to read, far too long; and (b) 
>generating little or no list discussion.

a lot of (b) can be attributed to this fact; the ID has existed since 
2003, and perhaps folks are tired of reading the doc that has to deal 
with Geopriv's inconsistencies over the years.


>This document needs to be finished.  However, I don't believe that 
>it is fit for publication.

I don't see comments to address here from you, so I cannot comment on 
those comments, and I certainly cannot address them either if I don't 
know what they are...


>I've forwarded comments from Hannes, which are very similar to those 
>I made many moons ago.  Unfortunately, these comments were sent to 
>the wrong list.
>
>So... if you want to get overly procedural, you might interpret the 
>response as representing consensus for ambivalence.  That would be a mistake.
>
>Cheers,
>Martin
>
>-----Original Message-----
>From: geopriv-bounces@ietf.org [mailto:geopriv-bounces@ietf.org] On 
>Behalf Of Hannes Tschofenig
>Sent: Saturday, 21 March 2009 12:05 PM
>To: 'GEOPRIV'
>Subject: [Geopriv] draft-ietf-sip-location-conveyance-13: WGLC comments
>
>I read through
>http://tools.ietf.org/html/draft-ietf-sip-location-conveyance-13.
>I have concerning the current version of the document.
>
>Here are my high-level comments. Detailed comments on paper available for
>the authors.
>
>* The draft comes up with new terminology even though we have these terms
>already defined in other documents. This is bad (tm).
>Cite terms (a common and preferred procedure) or copy the text exactly (and
>quote it) rather than defining your own. Useless and as we saw in recent
>discussions is rather harmful.
>
>* The document repeats stuff over and over again. Gives the reader the
>impression that this is rocket science. Could be significantly shorter.
>
>* The draft has to avoid the term "using protocol" at all costs. As we know,
>this term is very confusing.
>
>Example:    Other protocols used in the Location URI MUST be reviewed
>against
>    the RFC 3693 criteria for a Using Protocol.
>
>In human readable language this means: You have to consider security and
>privacy with new extensions.
>
>* Don't understand why this document does not allow a HELD URI to be
>conveyed. I commented this already several times and got ignored.
>
>See Section 4.5. Having an IANA registry appears to be totally crazy to me.
>Even saying that the protocols are mandatory to implement is also not a good
>idea. I really wonder why this is useful given that the recipient then just
>cannot fetch location when it receives something that it does not
>understand. What is the big deal with that?
>
>* Is "routing-allowed" policy really useful and, if so, does it need to be
>in the SIP header rather than in the PIDF-LO where all the other policies
>are?
>
>The problem is, again (also mentioned several times), with the lack of
>description of the recipient is for a PIDF-LO with respect to the
>'retransmission-allowed' policy. For example, in a location application
>where the message is directed towards a Service URN the intended behavior is
>that there is some form of location based routing involved.
>
>* This document is about the transport of a PIDF-LO and not about deployment
>considerations.
>
>Example:
>"
>    Adding a new locationValue to an in-transit request SHOULD NOT
>    occur for at least two reasons,
>
>    #1 - SIP Servers are not the best Sighters, as defined by [RFC3693],
>         of geographically where a UAC can be; meaning the location
>         information is not necessarily the greatest.  There MAY be
>         exceptions, but this SHOULD be the rule of thumb.
>
>    #2 - without appropriate caution to the fact that Location
>         Recipients might not understand how to process more than one
>         location, given this document's limited guidance as to what a
>         Location Recipient should do when receiving more than one
>         location  (i.e., currently no priority instructions are given
>         for which locationValue to use if there are more than one).  A
>         Location Recipient can easily be confused by too much location
>         information, producing undesirable results.  The <tuple id>
>         element in the PIDF-LO XML indicates whose location is
>         contained in the PIDF-LO.
>"
>
>This example also shows inappropriate usage of RFC2119 language which should
>be used for interoperability rather than philosophical stuff.
>
>We know who wants some of this functionality and what the exact use case it.
>
>* The location-based error codes (100, 200, 300, 400, 500 on page 14ff) are
>extremly fuzzy. They clearly lack implementation experience (as the text
>continuesly repeats by saying "for future work") and I believe they should
>be removed and incorporated in a future version when more implementation
>experience is available.
>
>
>* " S/MIME can be used for just signing, and not encrypting, a
>    PIDF-LO message body to ensure the integrity of the PIDF-LO is
>    maintained."
>
>   I am not sure what this means. S/MIME cannot only be used for signing.
>
>"
>    Use of self-signed certificates is inappropriate for use in
>    protecting a PIDF, as the sender does not have a secure identity of
>    the recipient.
>"
>
>  This sentence does not make a lot of sense to me.
>
>* This document does not need to repeat the text from other documents.
>Example:
>    "
>    When using LbyR, the LbyR MUST NOT contain any
>    user identifying information. For example, use something
>    unidentifiable like
>    "
>
>   Already in the LbyR requirements document. Reference it -- no need to
>write new text.
>
>    Other examples can be found with the description how
>http://www.ietf.org/internet-drafts/draft-ietf-geopriv-loc-filters-03.txt
>works.
>
>* "
>If the Location Generator wishes to control whether any location
>    included in the SIP request or added along the signaling path of
>    this request can be viewed for routing decisions, the Location
>    Generator adds a Geolocation header value including the
>    "routing-allowed=no" parameter.
>"
>
>The Rule Maker controls, not the LG.
>
>* Remote useless statements. Example:
>   "
>    Unless stated otherwise by future standards-track publication(s), a
>    LbyR URI only has meaning within the Geolocation header field and
>    MUST NOT appear in any other SIP header field.
>   "
>
>* Avoid author bias.
>
>"Proxies inserting location for location-based routing are
>    unable to meet this requirement, and such use is NOT RECOMMENDED."
>
>The document defines functionality to allow the proxy to add location and we
>know who is going to use this.
>
>* Avoid inappropriate treatment of subjects that are discussed somewhere
>else.
>
>"
>    One facet within this extension is such that locations can be placed
>    on a remote server, accessible with the possession of a URI.  The
>    concept of an LbyR URI has its own security considerations.  It is
>    tempting to assume that the dereference would have authentication,
>    authorization and other security mechanisms that limit the access to
>    information.  Unfortunately, this might not be true.  The access
>    network the UAC is connected to can be the source of location
>    reference, and it might not have any credentialing mechanism
>    suitable for controlling access to location.  Consider,
>    specifically, a nomadic user connected to an access network in a
>    hotel.  The UAC has no way to provide a credential acceptable to
>    the hotel Location Server (LS) to any of its intended Location
>    Recipients.  The recipient of a reference does not know if a
>    reference has appropriate authorization policies or not.  The LS
>    should provide location to any requestor.
>"
>
>The LbyR can have policies attached to it, if needed. We have worked on
>solutions to get this to work.
>If in a particular scenario policies cannot be uploaded to, for example, a
>LIS then put the reference somewhere else where you have the policies.
>
>* S/MIME example in the appendix isn't useful as it is currently put in
>there. Everybody knows that S/MIME allows you to encrypt a MIME body. So, if
>you want to put it in there then encrypt the stuff and attach the keys so
>that other folks can decrypt it.
>
>Ciao
>Hannes
>
>_______________________________________________
>Geopriv mailing list
>Geopriv@ietf.org
>https://www.ietf.org/mailman/listinfo/geopriv
>
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>------------------------------------------------------------------------------------------------
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On Mon, 2009-03-23 at 18:45 -0700, Adam Roach wrote:
> What I'm claiming right now is that it is a complete list of every sentence
> and clause in 3265 relating to the topic. A *complete* list would include
> the stuff in 5057 -- but I'm hesitant to suck in 20+ pages of information
> that is already specified elsewhere.

It wasn't clear to me that it included all the relevant parts of 3265,
but since this document is intended to obsolete 3265, its text can't
assert its own relationship to 3265.  Of course, you can refer to 5057,
because it won't be obsoleted.

Dale



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Subject: Re: [Sip] Comment on draft-roach-sipcore-rfc3265bis-00
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On Tue, 2009-03-24 at 09:52 -0700, Adam Roach wrote:
> > GRUU is not what one would call a wildly successful mechanism, yet
> 
> Of course it's not. It's not an RFC yet. After the whole multi-year TURN 
> whipsaw experience, I'm stunned anyone bothers to implement anything 
> before it has an RFC number assigned any more. But, suprisingly, they 
> do: more than one in every eight implementations at SIPit 22 implemented 
> GRUU.

I'm not so concerned about getting GRUU implemented.  It's known that
GRUU is necessary to get many call-control mechanisms to work, so
everyone assumes they'll have to implement it eventually.

Dale



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Subject: Re: [Sip] Is REFER within dialog part of INVITE or	subscription	usage? (was RE: SIP INFO)
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On Mon, 2009-03-23 at 21:32 -0400, Paul Kyzivat wrote:
> > I would argue that the REFER is part of the INVITE usage, because REFER
> > in that context is intimately related to the INVITE usage -- it is
> > intended to transfer the INVITE usage/dialog transferred to a different
> > destination.
> 
> Well, I guess there is that implied association. (Part of the "do what 
> *I* had in mind" approach that REFER has.) Since this was never spelled 
> out, it could also be simply "while this isn't part of an INVITE usage, 
> if there happens to be an INVITE usage, please refer that one."
> 
> I guess the question is: would it be valid to send a REFER within an 
> existing dialog that had no INVITE usage? If it were valid, presumably 
> it would have the same meaning as sending one outside a dialog, except 
> that the resulting subscription would share the dialog. I expect that is 
> a usage it would be difficult to find in the wild.

That's a nice pathological case.  But within the context of the current
5 possible meanings of REFER, it seems to be most parallel to the
out-of-dialog REFER.

> Ah, but now I want to reconsider. Remember that REFER can take a method 
> name. So, rather than referring an INVITE, I could be referring MESSAGE, 
> OPTIONS, BYE, or maybe (????) SUBSCRIBE. In some of those cases, the 
> association to the INVITE usage is probably meaningless. (Of course its 
> meaningful for BYE.)

A fascinating possibility!  But I'm not sure that we've ever defined how
a REFER within a simple INVITE dialog, specifying a non-INVITE method
should act.  So there's no analog to follow.

> > And it seems quite safe that if the REFER receives a 481 response, the
> > INVITE usage must not be functional any more, since the far end of that
> > usage has denied that it can act on the REFER.
> 
> IIRC, 5057 decided that each dialog usage must get its own 481 before it 
> goes away. I don't think there are any cases where a single 481 response 
> can make multiple usages go away.
> 
> So the important question is whether the REFER is part of the INVITE 
> usage and so its 481 takes down the INVITE usage or not. That gets 
> especially interesting if  the dialog already had both an INVITE usage 
> and a SUBSCRIBE usage. Does it make sense that the refer take down the 
> INVITE usage and leave up the SUBSCRIBE usage?

I think we're a victim of sloppy terminology here.  If I send a REFER,
and the far end responds 481, then clearly the far end knows of no
INVITE usage, or otherwise it would not have responded 481.  So I should
delete my belief that there is an INVITE usage.  Whether we consider the
REFER in question to be "part of" the INVITE usage or not doesn't change
anything -- the 481 is evidence that the far end has no knowledge of the
usage.

However, we do have this fine paradox:  If we assume that a REFER within
an established dialog that has no INVITE usage is to be treated as an
out-of-dialog REFER, then *no* REFER will receive a 481 response!
Indeed, if I send a REFER thinking that there is an active INVITE usage,
but the far end believes the INVITE usage has ended, will cause the far
end to do something (initiate an independent dialog) that is
considerably different from what I expected (transfer the current INVITE
usage).

Dale



From mary.barnes@nortel.com  Tue Mar 24 17:40:55 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, <sip@ietf.org>, "sipping" <sipping@ietf.org>
Cc: sip-chairs@tools.ietf.org, sipping-chairs@tools.ietf.org
Subject: Re: [Sip] [Sipping] Slides for SIP and SIPPING sessions
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A reminder that we need all the charts for tomorrow's SIP WG session no =
later than 9am so that we can get them all uploaded and on a single =
computer for presenting. So, far we've gotten one presentation.=20

If you could please send the charts to both sets of chairs, that would =
be helpful.

Thanks,
Mary.=20


-----Original Message-----
From: sipping-bounces@ietf.org on behalf of DRAGE, Keith (Keith)
Sent: Mon 3/23/2009 9:00 PM
To: sip@ietf.org; sipping
Cc: sip-chairs@tools.ietf.org; sipping-chairs@tools.ietf.org
Subject: [Sipping] Slides for SIP and SIPPING sessions
=20
(As SIP WG chair)

Please make sure you submit your slides well before the start of the =
sessions of SIP or SIPPING in which you are presenting. We would suggest =
deadlines of:

Tuesday 13:00 session - submit by Tuesday 9:00
Wednesday 13:00 session - submit by Wednesday 9:00
Friday 9:00 session - submit by Thursday 15:00

These are the latest time for submission - there is nothing wrong with =
sending them to us early.

To avoid confusion, you may want to send them to both sets of chairs, =
given the combined nature of the agenda this time round.

regards

Keith
_______________________________________________
Sipping mailing list  https://www.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP


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<P><FONT SIZE=3D2>A reminder that we need all the charts for tomorrow's =
SIP WG session no later than 9am so that we can get them all uploaded =
and on a single computer for presenting. So, far we've gotten one =
presentation.<BR>
<BR>
If you could please send the charts to both sets of chairs, that would =
be helpful.<BR>
<BR>
Thanks,<BR>
Mary.<BR>
<BR>
<BR>
-----Original Message-----<BR>
From: sipping-bounces@ietf.org on behalf of DRAGE, Keith (Keith)<BR>
Sent: Mon 3/23/2009 9:00 PM<BR>
To: sip@ietf.org; sipping<BR>
Cc: sip-chairs@tools.ietf.org; sipping-chairs@tools.ietf.org<BR>
Subject: [Sipping] Slides for SIP and SIPPING sessions<BR>
<BR>
(As SIP WG chair)<BR>
<BR>
Please make sure you submit your slides well before the start of the =
sessions of SIP or SIPPING in which you are presenting. We would suggest =
deadlines of:<BR>
<BR>
Tuesday 13:00 session - submit by Tuesday 9:00<BR>
Wednesday 13:00 session - submit by Wednesday 9:00<BR>
Friday 9:00 session - submit by Thursday 15:00<BR>
<BR>
These are the latest time for submission - there is nothing wrong with =
sending them to us early.<BR>
<BR>
To avoid confusion, you may want to send them to both sets of chairs, =
given the combined nature of the agenda this time round.<BR>
<BR>
regards<BR>
<BR>
Keith<BR>
_______________________________________________<BR>
Sipping mailing list&nbsp; <A =
HREF=3D"https://www.ietf.org/mailman/listinfo/sipping">https://www.ietf.o=
rg/mailman/listinfo/sipping</A><BR>
This list is for NEW development of the application of SIP<BR>
Use sip-implementors@cs.columbia.edu for questions on current sip<BR>
Use sip@ietf.org for new developments of core SIP<BR>
<BR>
</FONT>
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From spencer@wonderhamster.org  Tue Mar 24 17:58:18 2009
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From: "Spencer Dawkins" <spencer@wonderhamster.org>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>, "Jiri Kuthan" <jiri@iptel.org>, "Dean Willis" <dean.willis@softarmor.com>
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Subject: Re: [Sip] Diagnostic idea
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>> Nevertheless, I think that troubleshooting has been
>> historically of rather
>> marginal interest here, so I'm not sure if there is really an action to 
>> be
>> made.
>
> I think dealing with the operational aspects of SIP networks, especially 
> with troubleshooting them, is a topic whose time has come.  There are at 
> least 3 recent drafts introducing mechanisms to aid in troubleshooting 
> SIP.  Personally I think it's one of the next hurdles in growing SIP's 
> adoption.
>
> From a DISPATCH draft-routing-to-WG perspective, one could argue we 
> actually need a SipOps WG.

Hadriel,

I really agree with this. I'm assuming that "operational aspects of SIP 
networks" is

(1) big enough, and

(2) obvious enough

to justify requesting a traditional BOF (not sliding through DISPATCH). So, 
a couple of questions:

(1) is that OK, based on the current view of DISPATCH? or should we be 
requesting DISPATCH time in Stockholm? and

(2) do we have enough people with enough clue to propose a SIPOps BOF?

Thanks,

Spencer 



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Date: Wed, 25 Mar 2009 02:58:10 +0030
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Subject: Re: [Sip] Is REFER within dialog part of INVITE or subscription usage? (was RE: SIP INFO)
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If you send an in dialog request (REFER in our case), then you cannot
decouple its semantics with the existing dialog usage. So saying that
even if i send a REFER within an existing dialog, and still decouple
it from the existing dialog usage will not be possible.
If there were such a requirement, then REFER should have been sent as
a standalone dialog creating request outside the INVITE usage.

The recipient of the in dialog REFER will treat it like it should any
in dialog request. It has no way to decide if this request potentially
had nothing to do with the existing dialog usage.

i agree with you when you say that an indialog refer receiving a 481
would mean that the INVITE dialog has been torn down.

However, a refer dialog usage can also terminate, if the UAC sends a
481 in response to the NOTIFY sent by the UAS (Suppose the transaction
did not match). This scenario may happen irrespective of whether the
REFER was within the invite usage or created its own distinct dialog.
If this happens when REFER was sent within an Invite dialog, then the
UAC will be mistaken to clean up its INVITE dialog state.

I  did not understand one part in the paradox above when you said...
"A REFER within an established dialog having no invite usage to be
considered as an out of dialog REFER"
Do you mean to imply a REFER in some other Subscribe event package
dialog for example?

aayush.

On 3/25/09, Dale Worley <dworley@nortel.com> wrote:
> On Mon, 2009-03-23 at 21:32 -0400, Paul Kyzivat wrote:
>> > I would argue that the REFER is part of the INVITE usage, because REFER
>> > in that context is intimately related to the INVITE usage -- it is
>> > intended to transfer the INVITE usage/dialog transferred to a different
>> > destination.
>>
>> Well, I guess there is that implied association. (Part of the "do what
>> *I* had in mind" approach that REFER has.) Since this was never spelled
>> out, it could also be simply "while this isn't part of an INVITE usage,
>> if there happens to be an INVITE usage, please refer that one."
>>
>> I guess the question is: would it be valid to send a REFER within an
>> existing dialog that had no INVITE usage? If it were valid, presumably
>> it would have the same meaning as sending one outside a dialog, except
>> that the resulting subscription would share the dialog. I expect that is
>> a usage it would be difficult to find in the wild.
>
> That's a nice pathological case.  But within the context of the current
> 5 possible meanings of REFER, it seems to be most parallel to the
> out-of-dialog REFER.
>
>> Ah, but now I want to reconsider. Remember that REFER can take a method
>> name. So, rather than referring an INVITE, I could be referring MESSAGE,
>> OPTIONS, BYE, or maybe (????) SUBSCRIBE. In some of those cases, the
>> association to the INVITE usage is probably meaningless. (Of course its
>> meaningful for BYE.)
>
> A fascinating possibility!  But I'm not sure that we've ever defined how
> a REFER within a simple INVITE dialog, specifying a non-INVITE method
> should act.  So there's no analog to follow.
>
>> > And it seems quite safe that if the REFER receives a 481 response, the
>> > INVITE usage must not be functional any more, since the far end of that
>> > usage has denied that it can act on the REFER.
>>
>> IIRC, 5057 decided that each dialog usage must get its own 481 before it
>> goes away. I don't think there are any cases where a single 481 response
>> can make multiple usages go away.
>>
>> So the important question is whether the REFER is part of the INVITE
>> usage and so its 481 takes down the INVITE usage or not. That gets
>> especially interesting if  the dialog already had both an INVITE usage
>> and a SUBSCRIBE usage. Does it make sense that the refer take down the
>> INVITE usage and leave up the SUBSCRIBE usage?
>
> I think we're a victim of sloppy terminology here.  If I send a REFER,
> and the far end responds 481, then clearly the far end knows of no
> INVITE usage, or otherwise it would not have responded 481.  So I should
> delete my belief that there is an INVITE usage.  Whether we consider the
> REFER in question to be "part of" the INVITE usage or not doesn't change
> anything -- the 481 is evidence that the far end has no knowledge of the
> usage.
>
> However, we do have this fine paradox:  If we assume that a REFER within
> an established dialog that has no INVITE usage is to be treated as an
> out-of-dialog REFER, then *no* REFER will receive a 481 response!
> Indeed, if I send a REFER thinking that there is an active INVITE usage,
> but the far end believes the INVITE usage has ended, will cause the far
> end to do something (initiate an independent dialog) that is
> considerably different from what I expected (transfer the current INVITE
> usage).
>
> Dale
>
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

From dworley@nortel.com  Tue Mar 24 22:04:05 2009
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I'm not sure how much benefit comes from this idea.

On the one hand, it contains no mandate of the form "If the call-id is
of format xxxxx, then SBCs WILL NOT change it."  So there's SBC behavior
in particular that an outside agent can depend on, even if it knows that
all UAs in the system follow the I-D.  My guess is that SBCs will munge
the call-ids anyway, rather than trying to implement an algorithm to
determine compliance -- on a UAC-by-UAC basis!

The other side of this is that it makes life harder for anyone trying to
diagnose problems.  In sipX, we currently generate pseudo-random
call-ids.  Actually, the format is "s-XXXXXXXXXXX-NNN", where XXXXXXXX
is pseudo-random and NNN is a sequence number.  That way you can
mentally say "dialog 7" and "dialog 45", but still get uniqueness.  But
we've got a work item to append the host DNS name to the format so that
we can easily distinguish "dialog 45 from sipx1.example.com" and "dialog
45 from sipx2.example.com".

Dale



From dworley@nortel.com  Tue Mar 24 22:09:08 2009
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On Fri, 2009-03-06 at 12:59 -0500, Eric Burger wrote:
> Wouldn't SIP-CLF be a better way to go?

Not really, because log files slice the universe of data the wrong way.
A log file tells everything that happens at a particular element, almost
all of which is irrelevant regarding a particular transaction.

Transaction tracing is to extract the data relevant to a particular
transaction, across all elements that handle that transaction.  In many
cases, the person doing the trace is not and should not be authorized to
see the log of the elements the transaction traverses -- but they should
be authorized to see what happened to that one transaction.

Dale



From dworley@nortel.com  Tue Mar 24 22:17:53 2009
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On Fri, 2009-03-06 at 12:52 -0500, Hadriel Kaplan wrote:
> I appreciate that doing troubleshooting of SIP in SIP is very
> appealing, but I'm worried about it.  There are some real security
> issues with this, even if we don't want to admit it.  The reality is
> that if this trace mechanism is actually adopted by any sizeable
> measure, then middleboxes will be forced to munge the messages more
> than they already do, to prevent it from being used/work.  In other
> words it's a self-defeating mechanism.

Yes, there should be a mechanism so that a transit network can make
itself appear to be a wire, to be one hop, with regard to the trace
mechanism.  But there is no reason that the routing on the *far side* of
a transit network should be hidden from the UAC -- at least, from the
point of view of the transit network, the far-end network might have
policies.

(I'm envisioning a world in which the UAC's end may be involved in
complex service provisioning in which there may be more than one transit
network and more than one element that provides complex call routing.
Though a transit network should be able to hide its internal structure,
the UAC has the right to know what element the transit network delivered
the request to, since that is what the UAC is paying the transit network
to do.)

> Today you can do incrementing Max-Forwards traceroute, similar to the
> ICMP traceroute TTL trick, and even that's gotten some folks worried
> enough to make us munge the Max-Forwards value (despite the obvious
> dangers in doing so).  But anyway why isn't that type of trace enough
> for your needs?

I envision sending 20 INVITEs to the same request-URI, with gradually
increasing Max-Forwards values.  Any UAS that can be reached in less
than the maximum number of hops will get hammered.

And, as you say, SBCs are already trying to prevent that.

> Also, a couple specific questions/comments regarding this draft:
> 
> 1) You use the term "final response" a couple times as being sent in
> the 170's body.  How would a final response be sent in a 170 response,
> since the 170 would be before a final response?

Clearly, the 170 would have to be formulated after the final response is
formulated.  But I don't see any particular prohibition on that, and
there isn't any real restriction in the order that responses are sent.

> 2) You state that a UAC should not create an early-dialog for the 170,
> but for any B2BUA in the path which did not know about this new trace
> mechanism, wouldn't it create early dialogs on the b2bua since it
> would treat it as a 183?

I suppose it would.  Which would be somewhat resource-inefficient.  But
I don't expect the fraction of requests that are traced to be over 0.1%,
so it shouldn't be a problem in practice.

Dale



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On Sat, 2009-03-21 at 11:28 -0400, Jonathan Rosenberg wrote:
> Now B and C are talking to each other. What is the session ID for this 
> dialog? B and C were both originally the UAS on different dialogs with 
> different session IDs!

The essential problem (which also shows up in
draft-loreto-sipping-context-id-requirements) is that you need to
express a correlation between two dialogs, but the correlation is
discovered after both of the dialogs have been initiated.

If I may be immodest, the References header could a solution:  State
that two dialogs are related to each other by (effectively) naming both
call-id's together, rather than by assigning them a common identifier
when they are created.

In regard to correlating the two legs of a call through an SBC, see
section 5.2 of draft-worley-references-02.  The SBC can put a References
header in a message going out on one leg that contains the call-id of
the dialog on the other leg, or vice-versa.  If the SBC wants to be
paranoid, it can create a pseudo-random pseudo-Call-Id and put it in a
References header in two messages, one going out each leg.  That's not
elegant, but it's effective and safe -- and doesn't depend on the UAs.

As far as I can tell, the References header handles all of the use cases
people have stated.

Dale



From HKaplan@acmepacket.com  Tue Mar 24 23:42:30 2009
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To: Dale Worley <dworley@nortel.com>
Date: Wed, 25 Mar 2009 02:43:22 -0400
Thread-Topic: [Sip] I-D Inaction: draft-kaplan-sip-secure-call-id-00.txt
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> -----Original Message-----
> From: Dale Worley [mailto:dworley@nortel.com]
> Sent: Tuesday, March 24, 2009 11:59 PM
>=20
> I'm not sure how much benefit comes from this idea.
> On the one hand, it contains no mandate of the form "If the call-id is
> of format xxxxx, then SBCs WILL NOT change it." =20

Huh, you're right.  I should add in a MUST NOT change it for security purpo=
ses.  Although realistically Call-ID's are changed by many B2BUA's other th=
an SBC's for reasons other than security.  This draft is only trying to rem=
ove the incentive for changing them for security reasons.  There is no way =
to stop other B2BUA's from changing them for other reasons. (which is why I=
 have the session-id draft)


> So there's SBC behavior
> in particular that an outside agent can depend on, even if it knows that
> all UAs in the system follow the I-D.  My guess is that SBCs will munge
> the call-ids anyway, rather than trying to implement an algorithm to
> determine compliance -- on a UAC-by-UAC basis!

Sure, SBC's may not follow this draft.  So?  UA's and other devices may not=
 follow any given draft/RFC.  But right now there's actually a reasonably d=
efendable rationale for why the Call-ID's should be changed.  I'm trying to=
 get rid of that rationale.  The owners of the SBC's will then be able to t=
ell their vendors "go do this draft", if they find they need SIP extensions=
 to work which rely on the Call-ID not changing.  The market will sort out =
if that's valuable or not.

=20
> The other side of this is that it makes life harder for anyone trying to
> diagnose problems.  In sipX, we currently generate pseudo-random
> call-ids.  Actually, the format is "s-XXXXXXXXXXX-NNN", where XXXXXXXX
> is pseudo-random and NNN is a sequence number.  That way you can
> mentally say "dialog 7" and "dialog 45", but still get uniqueness.  But
> we've got a work item to append the host DNS name to the format so that
> we can easily distinguish "dialog 45 from sipx1.example.com" and "dialog
> 45 from sipx2.example.com".

If you do that, SBC's will continue to change the Call-ID from your devices=
.  There's always a trade-off between revealing enough information to troub=
leshoot/diagnose, and revealing information that can lead to abuse.

-hadriel

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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Dal=
e
> Worley
> Sent: Wednesday, March 25, 2009 12:26 AM
>=20
> The essential problem (which also shows up in
> draft-loreto-sipping-context-id-requirements) is that you need to
> express a correlation between two dialogs, but the correlation is
> discovered after both of the dialogs have been initiated.

Yeah but I'm not trying to solve that scenario. Session-ID is only trying t=
o create a dialog identifier for a single "end-end dialog" that survives B2=
BUA's, for the purposes of troubleshooting that dialog.  It's NOT trying to=
 correlate all subsequent dialogs, across all space/time, which may have so=
me relationship to the original one.
 =20
For example, take Jonathan's example of two-calls get joined together with =
a REFER processed by a PBX (I think that's one we've actually discussed bef=
ore, but maybe not).  There is dialog A and dialog B, and at some point hal=
f of A and half of B are joined together with re-INVITEs.  Personally I act=
ually think that's a brand new session.  Because had the REFER actually gon=
e end-to-end, that's exactly what would have happened: dialog C would be cr=
eated.  Just because they happen to be re-using dialog identifiers (call-id=
/tags) from the original A and B dialogs, is essentially just a nuance of S=
IP protocol mechanics.  From the user's perspective they got transferred, f=
rom the admins/provider's perspective they got transferred, and from an arc=
hitecture perspective they got transferred (i.e., the real "ends" of the di=
alog have now changed).

I don't debate that from a troubleshooting perspective it would be valuable=
 to know that the new C dialog was created from the original A and B dialog=
s, but I'm not sure that needs to be known on the wire.  I mean minimally t=
he PBX logs will show it.  But I can see the benefit for both.

=20
> If I may be immodest, the References header could a solution:  State
> that two dialogs are related to each other by (effectively) naming both
> call-id's together, rather than by assigning them a common identifier
> when they are created.

Yeah I think it adds value to have the References header.  I just don't thi=
nk it should be referencing Call-ID's, or else we'll have to replace that t=
oo.  It should be referencing Session-ID's or secure-call-id's.  :)

-hadriel

From HKaplan@acmepacket.com  Wed Mar 25 00:29:05 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dale Worley <dworley@nortel.com>
Date: Wed, 25 Mar 2009 03:29:58 -0400
Thread-Topic: [Sip] SIP Tracing Facility
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> -----Original Message-----
> From: Dale Worley [mailto:dworley@nortel.com]
> Sent: Wednesday, March 25, 2009 12:13 AM
>=20
> (I'm envisioning a world in which the UAC's end may be involved in
> complex service provisioning in which there may be more than one transit
> network and more than one element that provides complex call routing.
> Though a transit network should be able to hide its internal structure,
> the UAC has the right to know what element the transit network delivered
> the request to, since that is what the UAC is paying the transit network
> to do.)

IMO the UAC has no right to know any such thing.  It has no rights - it's n=
ot a human.  The calling human has a right to know the far-end called human=
 it got delivered to.  But they don't care nor have a right to know the *ho=
st* element it got delivered to, or host elements it forked to but did not =
succeed at, etc.  It's none of their business, frankly.

And if we're thinking the far-end terminating Enterprise or provider will w=
ant to tell some originating Enterprise or provider what's going on inside =
their network, well... once again we all live in different worlds. :)

-hadriel=20


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Subject: [Sip] [Technical Errata Reported] RFC3261 (1742)
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The following errata report has been submitted for RFC3261,
"SIP: Session Initiation Protocol".

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=3261&eid=1742

--------------------------------------
Type: Technical
Reported by: Pankaj Jain <jainp1979@gmail.com>

Section: 17.1.1.2

Original Text
-------------
Timer D reflects the amount of time that the server transaction can
   remain in the "Completed" state when unreliable transports are used.


Corrected Text
--------------
Timer D reflects the amount of time that the client transaction can
   remain in the "Completed" state when unreliable transports are used.


Notes
-----
server transaction => client transaction

Instructions:
-------------
This errata is currently posted as "Reported". If necessary, please
use "Reply All" to discuss whether it should be verified or
rejected. When a decision is reached, the verifying party (IESG)
can log in to change the status and edit the report, if necessary. 

--------------------------------------
RFC3261 (draft-ietf-sip-rfc2543bis-09)
--------------------------------------
Title               : SIP: Session Initiation Protocol
Publication Date    : June 2002
Author(s)           : J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, E. Schooler
Category            : PROPOSED STANDARD
Source              : Session Initiation Protocol
Area                : Real-time Applications and Infrastructure
Stream              : IETF
Verifying Party     : IESG

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Cc: SIP IETF <SIP@ietf.org>, Brett Tate <brett@broadsoft.com>
Subject: Re: [Sip] Is REFER within dialog part of INVITE or	subscription	usage? (was RE: SIP INFO)
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Dale Worley wrote:

>>> And it seems quite safe that if the REFER receives a 481 response, the
>>> INVITE usage must not be functional any more, since the far end of that
>>> usage has denied that it can act on the REFER.
>> IIRC, 5057 decided that each dialog usage must get its own 481 before it 
>> goes away. I don't think there are any cases where a single 481 response 
>> can make multiple usages go away.
>>
>> So the important question is whether the REFER is part of the INVITE 
>> usage and so its 481 takes down the INVITE usage or not. That gets 
>> especially interesting if  the dialog already had both an INVITE usage 
>> and a SUBSCRIBE usage. Does it make sense that the refer take down the 
>> INVITE usage and leave up the SUBSCRIBE usage?
> 
> I think we're a victim of sloppy terminology here.  If I send a REFER,
> and the far end responds 481, then clearly the far end knows of no
> INVITE usage, or otherwise it would not have responded 481. 

This is not "clear".

There are (at least) two alternatives, depending on whether the 
recipient believes the REFER to be part of an INVITE usage or not.

If it believes that it must be part of an INVITE usage (in the sense 
that BYE is known to be part of an INVITE usage), then the 481 response 
can mean that there is no INVITE usage, even though there is an existing 
matching dialog without an invite usage.

If the recipient believes that the REFER is *not* part of any usage, 
then there can still be a 481 response, indicating that that there is no 
matching dialog.

This is why its important to agree whether the REFER is to be considered 
part of an INVITE usage.

> So I should
> delete my belief that there is an INVITE usage.  Whether we consider the
> REFER in question to be "part of" the INVITE usage or not doesn't change
> anything -- the 481 is evidence that the far end has no knowledge of the
> usage.
> 
> However, we do have this fine paradox:  If we assume that a REFER within
> an established dialog that has no INVITE usage is to be treated as an
> out-of-dialog REFER, then *no* REFER will receive a 481 response!

I disagree here.

If the message is sent with dialog id info (notably, a To-tag in 
addition to a From-tag and Call-ID) then it should (or at least may) get 
a 481 response if there is no such matching dialog at the recipient. 
This is true even if the message is then to be treated as an 
out-of-dialog message.

> Indeed, if I send a REFER thinking that there is an active INVITE usage,
> but the far end believes the INVITE usage has ended, will cause the far
> end to do something (initiate an independent dialog) that is
> considerably different from what I expected (transfer the current INVITE
> usage).

That is why dialog matching should still be done. I have no business 
inventing a To-tag for you. It has to be *your* To-tag. If you don't 
recognize it then you send me a 481.

	Thanks,
	Paul

From jiri@iptel.org  Wed Mar 25 07:03:45 2009
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Subject: Re: [Sip] draft-state-sip-relay-attack-00
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Jonathan Rosenberg wrote:
> Rafael and others,
> 
> Thanks for putting this together.
> 
> Considering for a moment figure 1, this attack is possible only if UAs 
> accept incoming requests from any place, and not just their proxy. In 
> practice, this is seldom allowed. 

Hi Jonathan,
I see two things with this argument: first being a practical, the
other normative. on contrary to what you have observed, in the SIP
deployments I'm aware of the SIP UA accept SIP from everywhere.
That I consider a poor practice, but it seems to be the case.
(perhaps a differnet market segment experience?)
It has been also utilized in the DE attack last year -- attacker
sent SIP packets directly to UAs.
www.ipcom.at/fileadmin/public/2008-10-22_Analysis_of_a_VoIP_Attack.pdf
I should be in position to do a live measurement of a quite
SIP-UA-wise heterogenous SIP service, should we need more accurate
field intelligence.

The normative argument is that a SIP UA must be prepared to receive
traffic for a dialog setup by an RFC3261 proxy, which has no
obligation to record-route, and the traffic can therefore come
from anywhere. Unless we standardize that the UAs
are behind NATs, deprecate record-routing as optional or more
seriously document BCPs (don't talk with strangers), a SIP-compliant
UA is an open invitation for unsolicited traffic.

I'm under impression that folks in this threads feel easy about
the problem because they have all sorts of workarounds -- these
however do not rely on any sort of normative reference and seem
thus rather brittle.

Possibly addressing some of these practices in a BCP could
help to sort out some of that. Actually some ITSPs I know have been
privately approaching their UA vendors and felt they have never
made progress because there was no reference for such behaviour.

-jiri

> Indeed, if it is allowed, there are
> worse attacks than this which can be launched (free phone calls, spam 
> calling, etc.). Using the SIP recommended TLS between UA and proxy also 
> mitigates that.
> 
> So really figure 2 is the interesting one. However, this attack assumes 
> that Alice has credentials on multiple systems. Again, in practice, this 
> is extremely uncommon. Certainly none of the existing deployed 
> enterprise or service provider consumer deployments are of this nature.
> 
> As such, I dont think this attack is likely in practice. However, in 
> theory it is possible. The essence of the attack is that the victim is 
> providing credentials to an unauthenticated server (since the attacker 
> is acting like a server, asking for credentials). In that way, as others 
> have pointed out, it is similar to baiting attacks that have been 
> previously documented. With SIP it is most easily remedied by a rule 
> which says, 'don't pass credentials for domain X to a server that is not 
> domain X'. Server identity can be verified by normal server-only auth 
> between a client and its server, but even that is not needed. A client 
> will know which domain its proxy is representing, and once connected, it 
> only provides credentials for that domain.
> 
> -Jonathan R.
> 
> 
> Raphael Coeffic wrote:
>> Hello,
>>
>> a new internet draft has been published concerning the relay attack on 
>> digest authentication and SIP. The attack itself has been first 
>> disclosed 2 years ago by the maydnes team from the french INRIA. Until 
>> now, no document has been pushlished that documents the attack and 
>> provides guidance to SIP operators or handset manufacturers.
>>
>> http://tools.ietf.org/html/draft-state-sip-relay-attack-00
>>
>> The appropriate mitigations of problem resolutions are still not 100% 
>> clear. We hope that this draft can help start a discussion on how to 
>> best resolve this problem.
>>
>>
>> Regards,
>>
>> Raphael Coeffic.
>> (on behalf of all the authors of this draft)
>>
>> --------------------------------------------------------------------------------------------------- 
>>
>>
>> Filename:       draft-state-sip-relay-attack
>> Version:       00
>> Staging URL:       
>> http://www3.ietf.org/proceedings/staging/draft-state-sip-relay-attack-00.txt 
>>
>> Title:           SIP digest authentication relay attack
>> Creation_date:       2009-03-02
>> WG ID:           Indvidual Submission
>> Number_of_pages: 18
>> Abstract:
>> The Session Initiation Protocol (SIP [RFC3261]) provides a mechanism
>> for creating, modifying, and terminating sessions with one or more
>> participants.  This document describes a vulnerability of SIP
>> combined with HTTP Digest Access Authentication [RFC2617] through
>> which an attacker can leverage the victim's credentials to send
>> authenticated requests on his behalf.  This attack is different from
>> the man-in-the-middle (MITM) attack and does not require any
>> eavesdropping, DNS or IP spoofing.
>>
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
> 

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Cc: SIP IETF <SIP@ietf.org>, Brett Tate <brett@broadsoft.com>
Subject: Re: [Sip] Is REFER within dialog part of INVITE or subscription	usage? (was RE: SIP INFO)
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aayush bhatnagar wrote:
> If you send an in dialog request (REFER in our case), then you cannot
> decouple its semantics with the existing dialog usage.

You say "*the* dialog usage" as if there can be only one. There could be 
more than one. It might be coupled to:
- the *only* existing usage
- to *one* of several existing usages
- to *none* of the one or more existing usages

For instance, I could have a dialog with just an INVITE usage (the 
common case) and within that dialog send a REFER of a MESSAGE. That 
MESSAGE really has nothing to do with the INVITE usage.

> So saying that
> even if i send a REFER within an existing dialog, and still decouple
> it from the existing dialog usage will not be possible.
> If there were such a requirement, then REFER should have been sent as
> a standalone dialog creating request outside the INVITE usage.

One can argue that it *should* have been. But that isn't the point of 
this discussion. In many cases using the same dialog as been an 
expedient to ensure reaching the same UA when you aren't sure the remote 
target url for the dialog is in fact routable.

> The recipient of the in dialog REFER will treat it like it should any
> in dialog request. It has no way to decide if this request potentially
> had nothing to do with the existing dialog usage.

You are being sloppy in your use of "dialog" and "dialog usage", as if 
you are not familiar with RFC 5057. If you aren't, please read it.

> i agree with you when you say that an indialog refer receiving a 481
> would mean that the INVITE dialog has been torn down.

See my reply to Dale.

> However, a refer dialog usage can also terminate, if the UAC sends a
> 481 in response to the NOTIFY sent by the UAS (Suppose the transaction
> did not match).

You need to spell out more details of the case. If it were the *first* 
NOTIFY, resulting from an in-dialog REFER, then that response would be 
indicative of a screwup somewhere.

However, if there had been a prior NOTIFY on the refer subscription that 
indicated that the subscription was over, and then somehow *another* 
NOTIFY was sent, the 481 would be appropriate, indicating that the 
refer-usage was no longer present.

> This scenario may happen irrespective of whether the
> REFER was within the invite usage or created its own distinct dialog.
> If this happens when REFER was sent within an Invite dialog, then the
> UAC will be mistaken to clean up its INVITE dialog state.

Go read 5057. The usages are to be handled separately. So this should 
*not* provoke the UAC to clean up its INVITE dialog usage.

> I  did not understand one part in the paradox above when you said...
> "A REFER within an established dialog having no invite usage to be
> considered as an out of dialog REFER"
> Do you mean to imply a REFER in some other Subscribe event package
> dialog for example?

Yes, that is what I meant. I don't have any meaningful cases in mind, 
but it is a theoretical possibility.

	Paul

> aayush.
> 
> On 3/25/09, Dale Worley <dworley@nortel.com> wrote:
>> On Mon, 2009-03-23 at 21:32 -0400, Paul Kyzivat wrote:
>>>> I would argue that the REFER is part of the INVITE usage, because REFER
>>>> in that context is intimately related to the INVITE usage -- it is
>>>> intended to transfer the INVITE usage/dialog transferred to a different
>>>> destination.
>>> Well, I guess there is that implied association. (Part of the "do what
>>> *I* had in mind" approach that REFER has.) Since this was never spelled
>>> out, it could also be simply "while this isn't part of an INVITE usage,
>>> if there happens to be an INVITE usage, please refer that one."
>>>
>>> I guess the question is: would it be valid to send a REFER within an
>>> existing dialog that had no INVITE usage? If it were valid, presumably
>>> it would have the same meaning as sending one outside a dialog, except
>>> that the resulting subscription would share the dialog. I expect that is
>>> a usage it would be difficult to find in the wild.
>> That's a nice pathological case.  But within the context of the current
>> 5 possible meanings of REFER, it seems to be most parallel to the
>> out-of-dialog REFER.
>>
>>> Ah, but now I want to reconsider. Remember that REFER can take a method
>>> name. So, rather than referring an INVITE, I could be referring MESSAGE,
>>> OPTIONS, BYE, or maybe (????) SUBSCRIBE. In some of those cases, the
>>> association to the INVITE usage is probably meaningless. (Of course its
>>> meaningful for BYE.)
>> A fascinating possibility!  But I'm not sure that we've ever defined how
>> a REFER within a simple INVITE dialog, specifying a non-INVITE method
>> should act.  So there's no analog to follow.
>>
>>>> And it seems quite safe that if the REFER receives a 481 response, the
>>>> INVITE usage must not be functional any more, since the far end of that
>>>> usage has denied that it can act on the REFER.
>>> IIRC, 5057 decided that each dialog usage must get its own 481 before it
>>> goes away. I don't think there are any cases where a single 481 response
>>> can make multiple usages go away.
>>>
>>> So the important question is whether the REFER is part of the INVITE
>>> usage and so its 481 takes down the INVITE usage or not. That gets
>>> especially interesting if  the dialog already had both an INVITE usage
>>> and a SUBSCRIBE usage. Does it make sense that the refer take down the
>>> INVITE usage and leave up the SUBSCRIBE usage?
>> I think we're a victim of sloppy terminology here.  If I send a REFER,
>> and the far end responds 481, then clearly the far end knows of no
>> INVITE usage, or otherwise it would not have responded 481.  So I should
>> delete my belief that there is an INVITE usage.  Whether we consider the
>> REFER in question to be "part of" the INVITE usage or not doesn't change
>> anything -- the 481 is evidence that the far end has no knowledge of the
>> usage.
>>
>> However, we do have this fine paradox:  If we assume that a REFER within
>> an established dialog that has no INVITE usage is to be treated as an
>> out-of-dialog REFER, then *no* REFER will receive a 481 response!
>> Indeed, if I send a REFER thinking that there is an active INVITE usage,
>> but the far end believes the INVITE usage has ended, will cause the far
>> end to do something (initiate an independent dialog) that is
>> considerably different from what I expected (transfer the current INVITE
>> usage).
>>
>> Dale
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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Cc: SIP <sip@ietf.org>
Subject: Re: [Sip] SIP Tracing Facility
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Hadriel Kaplan wrote:
> 
>> -----Original Message-----
>> From: Dale Worley [mailto:dworley@nortel.com]
>> Sent: Wednesday, March 25, 2009 12:13 AM
>>
>> (I'm envisioning a world in which the UAC's end may be involved in
>> complex service provisioning in which there may be more than one transit
>> network and more than one element that provides complex call routing.
>> Though a transit network should be able to hide its internal structure,
>> the UAC has the right to know what element the transit network delivered
>> the request to, since that is what the UAC is paying the transit network
>> to do.)
> 
> IMO the UAC has no right to know any such thing.  It has no rights - it's not a human.  The calling human has a right to know the far-end called human it got delivered to.  But they don't care nor have a right to know the *host* element it got delivered to, or host elements it forked to but did not succeed at, etc.  It's none of their business, frankly.
> 
> And if we're thinking the far-end terminating Enterprise or provider will want to tell some originating Enterprise or provider what's going on inside their network, well... once again we all live in different worlds. :)

I think Dale's point is that it should not be the business of the 
transit network to prevent the UAC from learning things about the path 
after it leaves the transit network.

If the UAS network doesn't want that divulged, then fine. But if it is 
willing to have it divulged, then that should be fine too.

A really good example of that is when the UAC and UAS networks are 
branches of the same organization, that use a sip service provider for 
their interconnect.

	Thanks,
	Paul

From atle.monrad@stericsson.com  Tue Mar 24 02:56:53 2009
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Hello=20

To use one registration policy for all kind of feature tags may be a
"one size fits all" policy that in my view can lead to unnecessary
bottlenecks and delays. Personally I find it useful how RFC 2506
describe various mechanisms for registration of feature tags, and think
the current approach where a feature tag can be specified within one
organization and subsequently registered by IANA works well and is a
useful approach. At least it worked well for the first tags...=20

Similar ergistration is also performed for other parameters from IANA,
and 3GPP has also "registration procedures" with other organizations as
GSMA, TISPAN and OMA for various other parameters as well. In some cases
the registration is made by 3GPP and in some cases the registration is
made by other organisations. Who act as registrar depends on who "owns"
the basic functionality of the parameter. Technical discussions are
often held to find a solution, but when the technical discussions are
held and a solution is chosen, the registration process is done to
register and not to re-open the discussion.=20

3GPP has three feature tags described in 3GPP specifications and
registered in the global tree and three more feature tags described in
3GPP specifications waiting for similar registrations by IANA. It might
be useful to distinguish between the possible need for changing the
process for registration of future tags and performing registration of
tags that are in operation only missing the formal registration. I would
assume that the current process is in operation until a new registration
policy is available and agreed.


Regards
Atle Monrad=20
3GPP CT1 Chairman
______

Atle Monrad
Standardisation

ST-Ericsson
Product Group Mobile Platforms
=20
Televeien 1
N-4820, Grimstad
Norway
www.stericsson.com

=20
Office: +47 37 29 30 40
Mobile: +47 45 41 06 65
Fax: +47 37 04 23 70
Email: atle.monrad@stericsson.com

=09
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-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Gonzalo Camarillo
Sent: 20. mars 2009 11:47
To: sip
Subject: [Sip] Policy to register media feature tags

Folks,

as you know, RFC 3840 specifies how to indicate UA's capabilities using
media feature tags. Section 12.1 of RFC 3840 creates the SIP Media
Feature Tag Registration Tree. Media feature tags applicable to SIP are
registered in this tree.

http://www.ietf.org/rfc/rfc3840.txt

RFC 2506 defines the global tree to register feature tags. Section 3.1.2
of RFC 2506 says: "A registration may be proposed for the global tree by
anyone who has the need to allow for communication on a particular
capability or preference".

http://www.ietf.org/rfc/rfc2506.txt

The currently registered feature tags in both the SIP and the global
trees can be found at:

http://www.iana.org/assignments/media-feature-tags

As you can see, the only three feature tags registered in the global
tree come from 3GPP.

The issue we are facing is that the SIP and the global trees have
different registration policies. The SIP registry requires an RFC
defining the feature tag while the global tree only requires an expert
review.

Now, 3GPP would like to register two new feature tags:  g.3gpp.icsi-ref,
g.3gpp.iari-ref

This is the description of how these feature tags will be used:

"Each value of the Application Reference media feature-tag indicates the
software applications supported by the agent. The values for this tag
equal the IMS communication Service Identifier (ICSI) and IMS
Application Reference Identifier (IARI) values supported by the agent.
The Application Reference media feature tag is defined to fulfil the
requirements for forking to an appropriate UE when multiple UEs are
registered and dispatch to an appropriate application within the UE
based upon the IMS communication Service Identifier
(ICSI) and IMS Application Reference Identifier (IARI) values as stated
in 3GPP TS 23.228.  Multiple tag-values can be included in the
Application Reference media feature-tag."


The expert reviewer indicated that this tags will not contain simple
features but rather more complex services, or feature sets, or
application logic (pick your favorite term). Therefore, he did not find
it appropriate to register them in the global tree. He suggested that
RFC 2506 is amended so that it covers this type of more complex
features.

However, before going down that path, I would like to get comments from
the SIP community. The underlying issue is that we need to decide which
policy we want to apply to these types of registrations in SIP.

1) We decide that an RFC is always needed. Then, we can just use the SIP
registry. We would probably need to clarify or to stress somewhere that
the global tree is not appropriate for SIP tags.

2) We decide that an expert review is good enough. Then we would
probably need to amend RFC 2506 or RFC 3840 to reflect this.

Comments are welcome.

Thanks,

Gonzalo

_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol Use
sip-implementors@cs.columbia.edu for questions on current sip Use
sipping@ietf.org for new developments on the application of sip

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From: "Atle Monrad" <atle.monrad@stericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>, "Gonzalo Camarillo" <gonzalo.camarillo@ericsson.com>
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Paul=20

For you information, the tags that 3gpp have specified and registered
are described in publicly available 3GPP specifications which are
referenced from the IANA registration. The same apply for the three tags
that are in the pipeline. The reason for this is to secure openness and
operability for vendors that want to support certain 3gpp-specific
capabilities.

3GPP would like to get all tags registered by IANA in order to secure
that the tags are public and thereby the reference to the specifications
where this particular 3GPP-functionality is specified is publicly
available.=20

To communicate these tags e-2-e is needed to support certain
3gpp-capabilities, and I do not really see the linkage towards you
statement "On the other side were those who prefer to associate
arbitrary names to collections of features and then negotiate on the
basis of the names of those collections."

Hope this clarifies

Regards
/atle
______

Atle Monrad
Standardisation

ST-Ericsson
Product Group Mobile Platforms
=20
Televeien 1
N-4820, Grimstad
Norway
www.stericsson.com

=20
Office: +47 37 29 30 40
Mobile: +47 45 41 06 65
Fax: +47 37 04 23 70
Email: atle.monrad@stericsson.com

=09
This communication is confidential and intended solely for the
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-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Paul Kyzivat
Sent: 20. mars 2009 14:42
To: Gonzalo Camarillo
Cc: sip
Subject: Re: [Sip] Policy to register media feature tags

Some issues just never go away. :-(

This seems to be a resurection of issues that first surfaced during the
initial discussions of extended presence status many years ago. And it
also harks back to discussions last year or the year before about
communication service identifiers and application service identifiers.

The problem is one of openness. Specifically, do you have to be "part of
the club" in order to properly interoperate with a device that uses
these identifiers to identify its preferences or capabilities.

On one side were those who wanted capabilities and preferences to be
stated in terms of feature tags that are well known and orthogonal. On
the other side were those who prefer to associate arbitrary names to
collections of features and then negotiate on the basis of the names of
those collections.

Using the latter approach its possible that interoperation will fail
because the parties don't share a common name for a collection of
features even though they both possess the necessary features to
interoperate.

I don't have a problem with independent groups defining new feature
tags, as long as they are primitive and orthogonal to existing ones. But
I do have a problem with defining feature tags that identify collections
of features, especially when the mapping from the identifier to the
collection of features is not public.

The bottom line is that I think this sort of thing needs to be thrashed
out within the sip community. So I think the RFC process is the right
process.

	Thanks,
	Paul

Gonzalo Camarillo wrote:
> Folks,
>=20
> as you know, RFC 3840 specifies how to indicate UA's capabilities=20
> using media feature tags. Section 12.1 of RFC 3840 creates the SIP=20
> Media Feature Tag Registration Tree. Media feature tags applicable to=20
> SIP are registered in this tree.
>=20
> http://www.ietf.org/rfc/rfc3840.txt
>=20
> RFC 2506 defines the global tree to register feature tags. Section=20
> 3.1.2 of RFC 2506 says: "A registration may be proposed for the global

> tree by anyone who has the need to allow for communication on a=20
> particular capability or preference".
>=20
> http://www.ietf.org/rfc/rfc2506.txt
>=20
> The currently registered feature tags in both the SIP and the global=20
> trees can be found at:
>=20
> http://www.iana.org/assignments/media-feature-tags
>=20
> As you can see, the only three feature tags registered in the global=20
> tree come from 3GPP.
>=20
> The issue we are facing is that the SIP and the global trees have=20
> different registration policies. The SIP registry requires an RFC=20
> defining the feature tag while the global tree only requires an expert

> review.
>=20
> Now, 3GPP would like to register two new feature tags: =20
> g.3gpp.icsi-ref, g.3gpp.iari-ref
>=20
> This is the description of how these feature tags will be used:
>=20
> "Each value of the Application Reference media feature-tag indicates=20
> the software applications supported by the agent. The values for this=20
> tag equal the IMS communication Service Identifier (ICSI) and IMS=20
> Application Reference Identifier (IARI) values supported by the agent.

> The Application Reference media feature tag is defined to fulfil the=20
> requirements for forking to an appropriate UE when multiple UEs are=20
> registered and dispatch to an appropriate application within the UE=20
> based upon the IMS communication Service Identifier
> (ICSI) and IMS Application Reference Identifier (IARI) values as=20
> stated in 3GPP TS 23.228.  Multiple tag-values can be included in the=20
> Application Reference media feature-tag."
>=20
>=20
> The expert reviewer indicated that this tags will not contain simple=20
> features but rather more complex services, or feature sets, or=20
> application logic (pick your favorite term). Therefore, he did not=20
> find it appropriate to register them in the global tree. He suggested=20
> that RFC 2506 is amended so that it covers this type of more complex
features.
>=20
> However, before going down that path, I would like to get comments=20
> from the SIP community. The underlying issue is that we need to decide

> which policy we want to apply to these types of registrations in SIP.
>=20
> 1) We decide that an RFC is always needed. Then, we can just use the=20
> SIP registry. We would probably need to clarify or to stress somewhere

> that the global tree is not appropriate for SIP tags.
>=20
> 2) We decide that an expert review is good enough. Then we would=20
> probably need to amend RFC 2506 or RFC 3840 to reflect this.
>=20
> Comments are welcome.
>=20
> Thanks,
>=20
> Gonzalo
>=20
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> sipping@ietf.org for new developments on the application of sip
>=20
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol Use
sip-implementors@cs.columbia.edu for questions on current sip Use
sipping@ietf.org for new developments on the application of sip

From vkg@alcatel-lucent.com  Wed Mar 25 08:37:24 2009
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From: Vijay Gurbani <vkg@alcatel-lucent.com>
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Subject: Re: [Sip] SIP Tracing Facility
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Dale Worley wrote:
> Not really, because log files slice the universe of data the wrong way.

Not as much wrong as different; i.e., log files slice the universe of
data a different way.

> A log file tells everything that happens at a particular element, almost
> all of which is irrelevant regarding a particular transaction.

Depends.  If that element is misbehaving -- either through flawed source
code or because of external influences -- the state of a transaction
preserved in a log file from the point of that element is of extreme
interest for analysis.

However, as you note, log files do not help in dynamic tracing
of the transaction across a set of elements.

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
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From abhatnagar192006@gmail.com  Wed Mar 25 09:21:22 2009
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Date: Wed, 25 Mar 2009 21:52:10 +0530
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Subject: Re: [Sip] Is REFER within dialog part of INVITE or subscription usage? (was RE: SIP INFO)
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>Go read 5057. The usages are to be handled separately. So this should *not*
provoke the UAC >to clean up its INVITE dialog usage.
I have read the relevant RFC. From 5057:

-Existing implementations that destroy all other usages in the dialog will
continue to function as -they do now, except that peers following the
recommendation will attempt to do things with the -other usages and this
element will return 481s for each of them until they are all gone.

I think it is more feasible to destroy the entire dialog, than keeping on to
'hung' state until all the dialog usages are all gone as suggested above. I
am familiar with implementations that clean up dialog state when they get a
481-dialog does not exist response, even if there were other 'active' usages
of that dialog.

The reason being, that if an implementation cannot find a matching dialog as
characterized by the local tag, remote tag, call-id, local and remote
sequence nos etc and has rejected the request by a 481, there is no point in
keeping the dialog state at the UAC.

As for example, if you receive a 481 response for a REFER sent within the
INVITE dialog, there is no point in keeping the REFER usage alive. You
probably sent the REFER to transfer a call, which the server complains that
the call itself does not exist (481).

Moreover from 5057:

-Processing multiple usages correctly is not completely understood.
-What is understood is difficult to implement and is very likely to
-lead to interoperability problems. The best way to avoid the trouble
-that comes with such complexity is to avoid it altogether.
-When designing new applications or features that use SIP dialogs, do
-not require endpoints to construct multiple usages to participate in
-the application or use the feature.

This is very true. By defining multiple dialog usages, you are essentially
extending the dialog state (as defined by 3261) by adding more attributes to
it (such as subscription duration etc). I cant think of a practical use
case, where a subscription that shared the invite dialog usage still needed
to be preserved, even when the original INVITE dialog was no longer active.

As for example a subscription to the 'conference' event package sent within
the scope of an INVITE dialog. If i leave the conference by sending a BYE,
and my subscription's duration is still valid for an hour...will i still
keep on getting redundant SIP notifications for changes to the membership to
that conference ? In the sense of 5057, the subscribe usage will be
preserved until it expires, or until the UAC explicitly un-subscribes, or
the conference server has to send me a final NOTIFY with my
subsription-state = terminated.

It is not important that 5057 is allowing us many theoretical possibilities
by presenting multiple dialog usages.

What is more important is what extra do we get by reusing a dialog, and how
do implementations leverage upon such a possibility and create meaningful
applications around it ?

I would like to share some examples:

1. As for example an application that can practically leverage upon 5057 may
send a SUBSCRIBE to the 'dialog' event package (rfc 4235) that reuses the
INVITE dialog can be used for creating SIP call tracing applications.

2. Other Implementations may include automatic callback applications by this
mechanism (ring back when my sip phone is idle) etc.

3. Yet another application may send me by pre-paid account balance after i
disconnect my ongoing dialog, as it is subscribed to my dialog state.

In all these example applications, the endpoint receiving the SUBSCRIBE will
need to send a NOTIFY that re-uses its existing INVITE dialog (NOTIFY acting
as a target refresh in the sense of 5057).

Best Regards

aayush


On Wed, Mar 25, 2009 at 7:36 PM, Paul Kyzivat <pkyzivat@cisco.com> wrote:

>
>
> aayush bhatnagar wrote:
>
>> If you send an in dialog request (REFER in our case), then you cannot
>> decouple its semantics with the existing dialog usage.
>>
>
> You say "*the* dialog usage" as if there can be only one. There could be
> more than one. It might be coupled to:
> - the *only* existing usage
> - to *one* of several existing usages
> - to *none* of the one or more existing usages
>
> For instance, I could have a dialog with just an INVITE usage (the common
> case) and within that dialog send a REFER of a MESSAGE. That MESSAGE really
> has nothing to do with the INVITE usage.
>
>  So saying that
>> even if i send a REFER within an existing dialog, and still decouple
>> it from the existing dialog usage will not be possible.
>> If there were such a requirement, then REFER should have been sent as
>> a standalone dialog creating request outside the INVITE usage.
>>
>
> One can argue that it *should* have been. But that isn't the point of this
> discussion. In many cases using the same dialog as been an expedient to
> ensure reaching the same UA when you aren't sure the remote target url for
> the dialog is in fact routable.
>
>  The recipient of the in dialog REFER will treat it like it should any
>> in dialog request. It has no way to decide if this request potentially
>> had nothing to do with the existing dialog usage.
>>
>
> You are being sloppy in your use of "dialog" and "dialog usage", as if you
> are not familiar with RFC 5057. If you aren't, please read it.
>
>  i agree with you when you say that an indialog refer receiving a 481
>> would mean that the INVITE dialog has been torn down.
>>
>
> See my reply to Dale.
>
>  However, a refer dialog usage can also terminate, if the UAC sends a
>> 481 in response to the NOTIFY sent by the UAS (Suppose the transaction
>> did not match).
>>
>
> You need to spell out more details of the case. If it were the *first*
> NOTIFY, resulting from an in-dialog REFER, then that response would be
> indicative of a screwup somewhere.
>
> However, if there had been a prior NOTIFY on the refer subscription that
> indicated that the subscription was over, and then somehow *another* NOTIFY
> was sent, the 481 would be appropriate, indicating that the refer-usage was
> no longer present.
>
>  This scenario may happen irrespective of whether the
>> REFER was within the invite usage or created its own distinct dialog.
>> If this happens when REFER was sent within an Invite dialog, then the
>> UAC will be mistaken to clean up its INVITE dialog state.
>>
>
> Go read 5057. The usages are to be handled separately. So this should *not*
> provoke the UAC to clean up its INVITE dialog usage.
>
>  I  did not understand one part in the paradox above when you said...
>> "A REFER within an established dialog having no invite usage to be
>> considered as an out of dialog REFER"
>> Do you mean to imply a REFER in some other Subscribe event package
>> dialog for example?
>>
>
> Yes, that is what I meant. I don't have any meaningful cases in mind, but
> it is a theoretical possibility.
>
>        Paul
>
>
>  aayush.
>>
>> On 3/25/09, Dale Worley <dworley@nortel.com> wrote:
>>
>>> On Mon, 2009-03-23 at 21:32 -0400, Paul Kyzivat wrote:
>>>
>>>> I would argue that the REFER is part of the INVITE usage, because REFER
>>>>> in that context is intimately related to the INVITE usage -- it is
>>>>> intended to transfer the INVITE usage/dialog transferred to a different
>>>>> destination.
>>>>>
>>>> Well, I guess there is that implied association. (Part of the "do what
>>>> *I* had in mind" approach that REFER has.) Since this was never spelled
>>>> out, it could also be simply "while this isn't part of an INVITE usage,
>>>> if there happens to be an INVITE usage, please refer that one."
>>>>
>>>> I guess the question is: would it be valid to send a REFER within an
>>>> existing dialog that had no INVITE usage? If it were valid, presumably
>>>> it would have the same meaning as sending one outside a dialog, except
>>>> that the resulting subscription would share the dialog. I expect that is
>>>> a usage it would be difficult to find in the wild.
>>>>
>>> That's a nice pathological case.  But within the context of the current
>>> 5 possible meanings of REFER, it seems to be most parallel to the
>>> out-of-dialog REFER.
>>>
>>>  Ah, but now I want to reconsider. Remember that REFER can take a method
>>>> name. So, rather than referring an INVITE, I could be referring MESSAGE,
>>>> OPTIONS, BYE, or maybe (????) SUBSCRIBE. In some of those cases, the
>>>> association to the INVITE usage is probably meaningless. (Of course its
>>>> meaningful for BYE.)
>>>>
>>> A fascinating possibility!  But I'm not sure that we've ever defined how
>>> a REFER within a simple INVITE dialog, specifying a non-INVITE method
>>> should act.  So there's no analog to follow.
>>>
>>>  And it seems quite safe that if the REFER receives a 481 response, the
>>>>> INVITE usage must not be functional any more, since the far end of that
>>>>> usage has denied that it can act on the REFER.
>>>>>
>>>> IIRC, 5057 decided that each dialog usage must get its own 481 before it
>>>> goes away. I don't think there are any cases where a single 481 response
>>>> can make multiple usages go away.
>>>>
>>>> So the important question is whether the REFER is part of the INVITE
>>>> usage and so its 481 takes down the INVITE usage or not. That gets
>>>> especially interesting if  the dialog already had both an INVITE usage
>>>> and a SUBSCRIBE usage. Does it make sense that the refer take down the
>>>> INVITE usage and leave up the SUBSCRIBE usage?
>>>>
>>> I think we're a victim of sloppy terminology here.  If I send a REFER,
>>> and the far end responds 481, then clearly the far end knows of no
>>> INVITE usage, or otherwise it would not have responded 481.  So I should
>>> delete my belief that there is an INVITE usage.  Whether we consider the
>>> REFER in question to be "part of" the INVITE usage or not doesn't change
>>> anything -- the 481 is evidence that the far end has no knowledge of the
>>> usage.
>>>
>>> However, we do have this fine paradox:  If we assume that a REFER within
>>> an established dialog that has no INVITE usage is to be treated as an
>>> out-of-dialog REFER, then *no* REFER will receive a 481 response!
>>> Indeed, if I send a REFER thinking that there is an active INVITE usage,
>>> but the far end believes the INVITE usage has ended, will cause the far
>>> end to do something (initiate an independent dialog) that is
>>> considerably different from what I expected (transfer the current INVITE
>>> usage).
>>>
>>> Dale
>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
>>>
>>>  _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
>>

--000e0cd241fcd4cdd90465f3e678
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div><span class=3D"Apple-style-span" style=3D"font-style: italic;">&gt;Go =
read 5057. The usages are to be handled separately. So this should *not* pr=
ovoke the UAC &gt;to clean up its INVITE dialog usage.</span><div><br></div=
>
<div>I have read the relevant RFC. From 5057:</div><div><br></div><div><spa=
n class=3D"Apple-style-span" style=3D"font-style: italic;">-Existing implem=
entations that destroy all other usages in the dialog will continue to func=
tion as -they do now, except that peers following the recommendation will a=
ttempt to do things with the -other usages and this element will return 481=
s for each of them until they are all gone.</span></div>
<div><br></div><div>I think it is more feasible to destroy the entire dialo=
g, than keeping on to &#39;hung&#39; state until all the dialog usages are =
all gone as suggested above. I am familiar with implementations that clean =
up dialog state when they get a 481-dialog does not exist response, even if=
 there were other &#39;active&#39; usages of that dialog.=A0</div>
<div><br></div><div>The reason being, that if an implementation cannot find=
 a matching dialog as characterized by the local tag, remote tag, call-id, =
local and remote sequence nos etc and has rejected the request by a 481, th=
ere is no point in keeping the dialog state at the UAC.</div>
<div><br></div><div>As for example, if you receive a 481 response for a REF=
ER sent within the INVITE dialog, there is no point in keeping the REFER us=
age alive. You probably sent the REFER to transfer a call, which the server=
 complains that the call itself does not exist (481).=A0</div>
<div><br></div><div>Moreover from 5057:</div><div><br></div><div>-Processin=
g multiple usages correctly is not completely understood.<br></div>-What is=
 understood is difficult to implement and is very likely to<br>-lead to int=
eroperability problems. The best way to avoid the trouble<br>
-that comes with such complexity is to avoid it altogether.<br>-When design=
ing new applications or features that use SIP dialogs, do<br>-not require e=
ndpoints to construct multiple usages to participate in<br>-the application=
 or use the feature.<br>
</div><div><br></div><div>This is very true. By defining multiple dialog us=
ages, you are essentially extending the dialog state (as defined by 3261) b=
y adding more attributes to it (such as subscription duration etc). I cant =
think of a practical use case, where a subscription that shared the invite =
dialog usage still needed to be preserved, even when the original INVITE di=
alog was no longer active.</div>
<div><br></div><div>As for example a subscription to the &#39;conference&#3=
9; event package sent within the scope of an INVITE dialog. If i leave the =
conference by sending a BYE, and my subscription&#39;s duration is still va=
lid for an hour...will i still keep on getting redundant SIP notifications =
for changes to the membership to that conference ? In the sense of 5057, th=
e subscribe usage will be preserved until it expires, or until the UAC expl=
icitly un-subscribes, or the conference server has to send me a final NOTIF=
Y with my subsription-state =3D terminated.=A0</div>
<div><br></div><div>It is not important that 5057 is allowing us many theor=
etical possibilities by presenting multiple dialog usages.=A0</div><div><br=
></div><div>What is more important is what extra do we get by reusing a dia=
log, and how do implementations leverage upon such a possibility and create=
 meaningful applications around it ?</div>
<div><br></div><div>I would like to share some examples:</div><div><br></di=
v><div>1. As for example an application that can <span class=3D"Apple-style=
-span" style=3D"font-weight: bold;">practically leverage</span> upon 5057 m=
ay send a SUBSCRIBE to the &#39;dialog&#39; event package (rfc 4235) that r=
euses the INVITE dialog can be used for creating SIP call tracing applicati=
ons.=A0</div>
<div><br></div><div>2. Other Implementations may include automatic callback=
 applications by this mechanism (ring back when my sip phone is idle) etc.<=
/div><div><br></div><div>3. Yet another application may send me by pre-paid=
 account balance after i disconnect my ongoing dialog, as it is subscribed =
to my dialog state.</div>
<div><br></div><div>In all these example applications, the endpoint receivi=
ng the SUBSCRIBE will need to send a NOTIFY that re-uses its existing INVIT=
E dialog (NOTIFY acting as a target refresh in the sense of 5057).=A0</div>
<div><br></div><div>Best Regards</div><div><br></div><div>aayush</div><div>=
<br></div><div><div><div><br><div class=3D"gmail_quote">On Wed, Mar 25, 200=
9 at 7:36 PM, Paul Kyzivat <span dir=3D"ltr">&lt;<a href=3D"mailto:pkyzivat=
@cisco.com">pkyzivat@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;"><div class=3D"im"><br>
<br>
aayush bhatnagar wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
If you send an in dialog request (REFER in our case), then you cannot<br>
decouple its semantics with the existing dialog usage.<br>
</blockquote>
<br></div>
You say &quot;*the* dialog usage&quot; as if there can be only one. There c=
ould be more than one. It might be coupled to:<br>
- the *only* existing usage<br>
- to *one* of several existing usages<br>
- to *none* of the one or more existing usages<br>
<br>
For instance, I could have a dialog with just an INVITE usage (the common c=
ase) and within that dialog send a REFER of a MESSAGE. That MESSAGE really =
has nothing to do with the INVITE usage.<div class=3D"im"><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
So saying that<br>
even if i send a REFER within an existing dialog, and still decouple<br>
it from the existing dialog usage will not be possible.<br>
If there were such a requirement, then REFER should have been sent as<br>
a standalone dialog creating request outside the INVITE usage.<br>
</blockquote>
<br></div>
One can argue that it *should* have been. But that isn&#39;t the point of t=
his discussion. In many cases using the same dialog as been an expedient to=
 ensure reaching the same UA when you aren&#39;t sure the remote target url=
 for the dialog is in fact routable.<div class=3D"im">
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
The recipient of the in dialog REFER will treat it like it should any<br>
in dialog request. It has no way to decide if this request potentially<br>
had nothing to do with the existing dialog usage.<br>
</blockquote>
<br></div>
You are being sloppy in your use of &quot;dialog&quot; and &quot;dialog usa=
ge&quot;, as if you are not familiar with RFC 5057. If you aren&#39;t, plea=
se read it.<div class=3D"im"><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
i agree with you when you say that an indialog refer receiving a 481<br>
would mean that the INVITE dialog has been torn down.<br>
</blockquote>
<br></div>
See my reply to Dale.<div class=3D"im"><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
However, a refer dialog usage can also terminate, if the UAC sends a<br>
481 in response to the NOTIFY sent by the UAS (Suppose the transaction<br>
did not match).<br>
</blockquote>
<br></div>
You need to spell out more details of the case. If it were the *first* NOTI=
FY, resulting from an in-dialog REFER, then that response would be indicati=
ve of a screwup somewhere.<br>
<br>
However, if there had been a prior NOTIFY on the refer subscription that in=
dicated that the subscription was over, and then somehow *another* NOTIFY w=
as sent, the 481 would be appropriate, indicating that the refer-usage was =
no longer present.<div class=3D"im">
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
This scenario may happen irrespective of whether the<br>
REFER was within the invite usage or created its own distinct dialog.<br>
If this happens when REFER was sent within an Invite dialog, then the<br>
UAC will be mistaken to clean up its INVITE dialog state.<br>
</blockquote>
<br></div>
Go read 5057. The usages are to be handled separately. So this should *not*=
 provoke the UAC to clean up its INVITE dialog usage.<div class=3D"im"><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
I =A0did not understand one part in the paradox above when you said...<br>
&quot;A REFER within an established dialog having no invite usage to be<br>
considered as an out of dialog REFER&quot;<br>
Do you mean to imply a REFER in some other Subscribe event package<br>
dialog for example?<br>
</blockquote>
<br></div>
Yes, that is what I meant. I don&#39;t have any meaningful cases in mind, b=
ut it is a theoretical possibility.<br><font color=3D"#888888">
<br>
 =A0 =A0 =A0 =A0Paul</font><div><div></div><div class=3D"h5"><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
aayush.<br>
<br>
On 3/25/09, Dale Worley &lt;<a href=3D"mailto:dworley@nortel.com" target=3D=
"_blank">dworley@nortel.com</a>&gt; wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On Mon, 2009-03-23 at 21:32 -0400, Paul Kyzivat wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I would argue that the REFER is part of the INVITE usage, because REFER<br>
in that context is intimately related to the INVITE usage -- it is<br>
intended to transfer the INVITE usage/dialog transferred to a different<br>
destination.<br>
</blockquote>
Well, I guess there is that implied association. (Part of the &quot;do what=
<br>
*I* had in mind&quot; approach that REFER has.) Since this was never spelle=
d<br>
out, it could also be simply &quot;while this isn&#39;t part of an INVITE u=
sage,<br>
if there happens to be an INVITE usage, please refer that one.&quot;<br>
<br>
I guess the question is: would it be valid to send a REFER within an<br>
existing dialog that had no INVITE usage? If it were valid, presumably<br>
it would have the same meaning as sending one outside a dialog, except<br>
that the resulting subscription would share the dialog. I expect that is<br=
>
a usage it would be difficult to find in the wild.<br>
</blockquote>
That&#39;s a nice pathological case. =A0But within the context of the curre=
nt<br>
5 possible meanings of REFER, it seems to be most parallel to the<br>
out-of-dialog REFER.<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Ah, but now I want to reconsider. Remember that REFER can take a method<br>
name. So, rather than referring an INVITE, I could be referring MESSAGE,<br=
>
OPTIONS, BYE, or maybe (????) SUBSCRIBE. In some of those cases, the<br>
association to the INVITE usage is probably meaningless. (Of course its<br>
meaningful for BYE.)<br>
</blockquote>
A fascinating possibility! =A0But I&#39;m not sure that we&#39;ve ever defi=
ned how<br>
a REFER within a simple INVITE dialog, specifying a non-INVITE method<br>
should act. =A0So there&#39;s no analog to follow.<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
And it seems quite safe that if the REFER receives a 481 response, the<br>
INVITE usage must not be functional any more, since the far end of that<br>
usage has denied that it can act on the REFER.<br>
</blockquote>
IIRC, 5057 decided that each dialog usage must get its own 481 before it<br=
>
goes away. I don&#39;t think there are any cases where a single 481 respons=
e<br>
can make multiple usages go away.<br>
<br>
So the important question is whether the REFER is part of the INVITE<br>
usage and so its 481 takes down the INVITE usage or not. That gets<br>
especially interesting if =A0the dialog already had both an INVITE usage<br=
>
and a SUBSCRIBE usage. Does it make sense that the refer take down the<br>
INVITE usage and leave up the SUBSCRIBE usage?<br>
</blockquote>
I think we&#39;re a victim of sloppy terminology here. =A0If I send a REFER=
,<br>
and the far end responds 481, then clearly the far end knows of no<br>
INVITE usage, or otherwise it would not have responded 481. =A0So I should<=
br>
delete my belief that there is an INVITE usage. =A0Whether we consider the<=
br>
REFER in question to be &quot;part of&quot; the INVITE usage or not doesn&#=
39;t change<br>
anything -- the 481 is evidence that the far end has no knowledge of the<br=
>
usage.<br>
<br>
However, we do have this fine paradox: =A0If we assume that a REFER within<=
br>
an established dialog that has no INVITE usage is to be treated as an<br>
out-of-dialog REFER, then *no* REFER will receive a 481 response!<br>
Indeed, if I send a REFER thinking that there is an active INVITE usage,<br=
>
but the far end believes the INVITE usage has ended, will cause the far<br>
end to do something (initiate an independent dialog) that is<br>
considerably different from what I expected (transfer the current INVITE<br=
>
usage).<br>
<br>
Dale<br>
<br>
<br>
_______________________________________________<br>
Sip mailing list =A0<a href=3D"https://www.ietf.org/mailman/listinfo/sip" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
This list is for NEW development of the core SIP Protocol<br>
Use <a href=3D"mailto:sip-implementors@cs.columbia.edu" target=3D"_blank">s=
ip-implementors@cs.columbia.edu</a> for questions on current sip<br>
Use <a href=3D"mailto:sipping@ietf.org" target=3D"_blank">sipping@ietf.org<=
/a> for new developments on the application of sip<br>
<br>
</blockquote>
_______________________________________________<br>
Sip mailing list =A0<a href=3D"https://www.ietf.org/mailman/listinfo/sip" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
This list is for NEW development of the core SIP Protocol<br>
Use <a href=3D"mailto:sip-implementors@cs.columbia.edu" target=3D"_blank">s=
ip-implementors@cs.columbia.edu</a> for questions on current sip<br>
Use <a href=3D"mailto:sipping@ietf.org" target=3D"_blank">sipping@ietf.org<=
/a> for new developments on the application of sip<br>
<br>
</blockquote>
</div></div></blockquote></div><br></div></div></div>

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On Wed, Mar 25, 2009 at 12:58 AM, Spencer Dawkins
<spencer@wonderhamster.org> wrote:
> (2) do we have enough people with enough clue to propose a SIPOps BOF?

I was going to suggest exactly this, I just hadn't worked out the
process for doing it yet :)

I think it would be a great, great idea and would love to see more
focus on operations in the same was an dns, ipv6 etc does, or even
something less formal like the IEPG on sunday morning?

recent statistics show SIP has a huge penetration on the *public*
internet - way more than i think any of us did.

 ~ Theo

From L.Liess@telekom.de  Wed Mar 25 09:44:54 2009
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Subject: Re: [Sip] Diagnostic idea
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It's a great idea.=20

Laura

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Theo Zourzouvillys
> Sent: Wednesday, March 25, 2009 5:29 PM
> To: Spencer Dawkins
> Cc: SIP; Dean Willis
> Subject: Re: [Sip] Diagnostic idea
>=20
> On Wed, Mar 25, 2009 at 12:58 AM, Spencer Dawkins
> <spencer@wonderhamster.org> wrote:
> > (2) do we have enough people with enough clue to propose a=20
> SIPOps BOF?
>=20
> I was going to suggest exactly this, I just hadn't worked out the
> process for doing it yet :)
>=20
> I think it would be a great, great idea and would love to see more
> focus on operations in the same was an dns, ipv6 etc does, or even
> something less formal like the IEPG on sunday morning?
>=20
> recent statistics show SIP has a huge penetration on the *public*
> internet - way more than i think any of us did.
>=20
>  ~ Theo
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

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On Wed, Mar 25, 2009 at 5:45 PM,  <L.Liess@telekom.de> wrote:
>
> It's a great idea.

+1
--=20
Victor Pascual =C3=81vila

From jiri@iptel.org  Wed Mar 25 09:48:36 2009
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Theo Zourzouvillys wrote:
> On Wed, Mar 25, 2009 at 12:58 AM, Spencer Dawkins
> <spencer@wonderhamster.org> wrote:
>> (2) do we have enough people with enough clue to propose a SIPOps BOF?
> 
> I was going to suggest exactly this, I just hadn't worked out the
> process for doing it yet :)
> 
> I think it would be a great, great idea and would love to see more
> focus on operations in the same was an dns, ipv6 etc does, or even
> something less formal like the IEPG on sunday morning?

I think too, there is a large array of topics which are underrepresented
and have large impact on the field. It seems too that the same important
these are, the same unattractive they seem to many who prefer to focus
their creative juices on building new stuff :-)

anyhow, to be more specific: I think diagnostics (such as "proxy chain
troublehooting",disclosing SIP routing decisions to monitoring equipment,
...), security BCPs (see the thread on how to mititage relay attacks,
how to sort out all the identity mechanisms we have,security policies,..),
some interop stuff (Hadriel's HOWTO for session id crossing over 
middleboxes...),
global service discovery, and certainly more.


> 
> recent statistics show SIP has a huge penetration on the *public*
> internet - way more than i think any of us did.

Some hard publicly disclosed numbers?

Thanks!

-jiri

> 
>  ~ Theo
> 

From theo@crazygreek.co.uk  Wed Mar 25 10:01:48 2009
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On Wed, Mar 25, 2009 at 4:49 PM, Jiri Kuthan <jiri@iptel.org> wrote:

> Some hard publicly disclosed numbers?

wait until my presentation in SIP after lunch :-)

lots of people have expressed interest in the stats - i've started
doign regular scans fngerprinting, etc and will provide the full
details publically over the next few weeks as well as ongoing graphs
etc.

i just need to find some time to put them up and provide web interface
to the database, and have obviously been rather busy the last few
weeks ;)

 ~ Theo

From dworley@nortel.com  Wed Mar 25 11:23:57 2009
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Subject: Re: [Sip] SIP Tracing Facility
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On Wed, 2009-03-25 at 10:38 -0500, Vijay Gurbani wrote:
> Dale Worley wrote:
> > Not really, because log files slice the universe of data the wrong way.
> 
> Not as much wrong as different; i.e., log files slice the universe of
> data a different way.
> 
> > A log file tells everything that happens at a particular element, almost
> > all of which is irrelevant regarding a particular transaction.
> 
> Depends.  If that element is misbehaving -- either through flawed source
> code or because of external influences -- the state of a transaction
> preserved in a log file from the point of that element is of extreme
> interest for analysis.
> 
> However, as you note, log files do not help in dynamic tracing
> of the transaction across a set of elements.

As we both understand, the two mechanisms slice the universe of data in
different ways, which are most useful for attacking different problems.

What makes the problem more complex is that different elements have
different owners, and one can reasonably say that the UAC-owner can look
at the action of a proxy in regard to one of the UAC's requests, but not
have the right to examine the proxy's full log.

Conversely, taking the intersection of the two questions, a log file
format should record at least enough information about a transaction as
you would want in a trace facility.

Dale



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From: "Dale Worley" <dworley@nortel.com>
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Subject: Re: [Sip] SIP Tracing Facility
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On Wed, 2009-03-25 at 03:29 -0400, Hadriel Kaplan wrote:
> IMO the UAC has no right to know any such thing.  It has no rights -
> it's not a human.  The calling human has a right to know the far-end
> called human it got delivered to.  But they don't care nor have a
> right to know the *host* element it got delivered to, or host elements
> it forked to but did not succeed at, etc.  It's none of their
> business, frankly.
> 
> And if we're thinking the far-end terminating Enterprise or provider
> will want to tell some originating Enterprise or provider what's going
> on inside their network, well... once again we all live in different
> worlds. :)

I'm sure we live in different worlds.

But there are considerable situations where the UAC's owner, or the
enterprise or service provider supervising the UAC has rights to know
quite a bit.  The obvious case is when the owner has contracted with the
transit network to provide particular intelligent routing services --
they're going to want to be able to verify that the transit network is
performing as contracted.

To construct a more interesting case, one that will be a reasonable
business case within a year or two:

- a UAC within an enterprise network
- a transit network from the enterprise to a provider of intelligent
call routing
- the provider of intelligent call routing
- another transit network from the provider to a destination enterprise
- a UAS within the destination enterprise network

Each of the last 4 operations may want to hide the details of its
network, but the originator of the call should have the right (assuming
that it's contracting for these services) to monitor the four
connections between these networks.

Dale



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Subject: [Sip] Reminder: SIPit24 registration closes April 30
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SIPit 24 registration closes April 30. Please register now if you
haven't done so already.

Be sure to make your lodging reservations early.

Links to more information, including the registration site are below.

Thanks,

RjS

On Mar 3, 2009, at 8:51 AM, Robert Sparks wrote:

> Registration for SIPit 24 is open!
> SIPit 24 will be held May 18-22, 2009 in Akihabara, Tokyo, Japan
> hosted by JPNIC and NICT.
> Additional information is available at http://www.sipit.net and http://www.nic.ad.jp/en/sipit24/

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<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">SIPit 24 registration closes April 30. Please register now if you &nbsp;<br>haven't done so already.<br><br>Be sure to make your lodging reservations early.<br><br>Links to more information, including the registration site are below.<br><br>Thanks,<br><br>RjS<br><br>On Mar 3, 2009, at 8:51 AM, Robert Sparks wrote:<br><br><blockquote type="cite">Registration for SIPit 24 is open!<br></blockquote><blockquote type="cite">SIPit 24 will be held May 18-22, 2009 in Akihabara, Tokyo, Japan &nbsp;<br></blockquote><blockquote type="cite">hosted by JPNIC and NICT.<br></blockquote><blockquote type="cite">Additional information is available at&nbsp;<a href="http://www.sipit.net/">http://www.sipit.net</a>&nbsp;and&nbsp;<a href="http://www.nic.ad.jp/en/sipit24/">http://www.nic.ad.jp/en/sipit24/</a></blockquote></body></html>
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Cc: SIP IETF <SIP@ietf.org>, Brett Tate <brett@broadsoft.com>
Subject: Re: [Sip] Is REFER within dialog part of INVITE or subscription usage? (was RE: SIP INFO)
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aayush bhatnagar wrote:
>>Go read 5057. The usages are to be handled separately. So this should 
> *not* provoke the UAC >to clean up its INVITE dialog usage.
> 
> I have read the relevant RFC. From 5057:

OK, sorry. It just sounded like you might not have.

> -Existing implementations that destroy all other usages in the dialog 
> will continue to function as -they do now, except that peers following 
> the recommendation will attempt to do things with the -other usages and 
> this element will return 481s for each of them until they are all gone.
> 
> I think it is more feasible to destroy the entire dialog, than keeping 
> on to 'hung' state until all the dialog usages are all gone as suggested 
> above. I am familiar with implementations that clean up dialog state 
> when they get a 481-dialog does not exist response, even if there were 
> other 'active' usages of that dialog. 

I'm not surprised. Its often "more feasible" to do the wrong thing than 
the right thing. But that doesn't make it right. While its not written 
in black and white in 3261, there is enough info there to infer that you 
need to treat them separately.

> The reason being, that if an implementation cannot find a matching 
> dialog as characterized by the local tag, remote tag, call-id, local and 
> remote sequence nos etc and has rejected the request by a 481, there is 
> no point in keeping the dialog state at the UAC.

This is all hashed out in 5057. The problem is that the 481 response is 
*ambiguous*. The way to maximize interop is to treat the 481 as meaning 
the least serious thing that it could mean - that only the usage has 
gone away. If it turns out that the whole dialog went away, then you 
will eventually find that out anyway. Of course it doesn't hurt to treat 
it as meaning all the usages went away if you would not continue all of 
them anyway. But if the other usage(s) continue to be of value then it 
you are hurting yourself by assuming they have gone away when they might 
not have.

> As for example, if you receive a 481 response for a REFER sent within 
> the INVITE dialog, there is no point in keeping the REFER usage alive. 
> You probably sent the REFER to transfer a call, which the server 
> complains that the call itself does not exist (481). 

That's not the case of relevance. If you got a 481 for a REFER, the 
refer of course failed, and so there is no refer usage to keep alive.

We are still, in this thread, trying to decide what a 481 to a REFER 
means. I think it won't be ambiguous - that it can *only* mean that the 
whole dialog is gone, since it is never associated with a preexisting usage.

Even so, you would be well advised to probe any existing usages with 
in-usage messages to verify if they are alive, even though most likely 
they are not.

> Moreover from 5057:
> 
> -Processing multiple usages correctly is not completely understood.
> -What is understood is difficult to implement and is very likely to
> -lead to interoperability problems. The best way to avoid the trouble
> -that comes with such complexity is to avoid it altogether.
> -When designing new applications or features that use SIP dialogs, do
> -not require endpoints to construct multiple usages to participate in
> -the application or use the feature.
> 
> This is very true. By defining multiple dialog usages, you are 
> essentially extending the dialog state (as defined by 3261) by adding 
> more attributes to it (such as subscription duration etc). I cant think 
> of a practical use case, where a subscription that shared the invite 
> dialog usage still needed to be preserved, even when the original INVITE 
> dialog was no longer active.

We aren't *adding more attributes to it*. It is what it is and has been. 
It has been implicitly legal in spite of the problems. And it *is* used. 
There are even reasonable use cases for independent termination of the 
usages. For instance, a refer usage sharing a dialog with an invite 
usage may meaningfully survive after the invite usage terminates.

> As for example a subscription to the 'conference' event package sent 
> within the scope of an INVITE dialog. If i leave the conference by 
> sending a BYE, and my subscription's duration is still valid for an 
> hour...will i still keep on getting redundant SIP notifications for 
> changes to the membership to that conference ?


> In the sense of 5057, the 
> subscribe usage will be preserved until it expires, or until the UAC 
> explicitly un-subscribes, or the conference server has to send me a 
> final NOTIFY with my subsription-state = terminated. 

IMO that is exactly as it *should* be. If the subscriber doesn't want to 
maintain the subscription, it can simply wipe out the state of the 
subscription. Then when another NOTIFY is received, a 481 will be sent 
because there is no state, forcing the end of the usage.

But *assuming* that the subscription *should* go away when the invite 
ends is, IMO, wrong. There may be valid reasons to retain the subscription.

> It is not important that 5057 is allowing us many theoretical 
> possibilities by presenting multiple dialog usages. 
> 
> What is more important is what extra do we get by reusing a dialog, and 
> how do implementations leverage upon such a possibility and create 
> meaningful applications around it ?
> 
> I would like to share some examples:
> 
> 1. As for example an application that can practically leverage upon 5057 
> may send a SUBSCRIBE to the 'dialog' event package (rfc 4235) that 
> reuses the INVITE dialog can be used for creating SIP call tracing 
> applications. 
> 
> 2. Other Implementations may include automatic callback applications by 
> this mechanism (ring back when my sip phone is idle) etc.
> 
> 3. Yet another application may send me by pre-paid account balance after 
> i disconnect my ongoing dialog, as it is subscribed to my dialog state.
> 
> In all these example applications, the endpoint receiving the SUBSCRIBE 
> will need to send a NOTIFY that re-uses its existing INVITE dialog 
> (NOTIFY acting as a target refresh in the sense of 5057). 

Sounds like you are in favor of dialog reuse.

	Paul

> Best Regards
> 
> aayush
> 
> 
> On Wed, Mar 25, 2009 at 7:36 PM, Paul Kyzivat <pkyzivat@cisco.com 
> <mailto:pkyzivat@cisco.com>> wrote:
> 
> 
> 
>     aayush bhatnagar wrote:
> 
>         If you send an in dialog request (REFER in our case), then you
>         cannot
>         decouple its semantics with the existing dialog usage.
> 
> 
>     You say "*the* dialog usage" as if there can be only one. There
>     could be more than one. It might be coupled to:
>     - the *only* existing usage
>     - to *one* of several existing usages
>     - to *none* of the one or more existing usages
> 
>     For instance, I could have a dialog with just an INVITE usage (the
>     common case) and within that dialog send a REFER of a MESSAGE. That
>     MESSAGE really has nothing to do with the INVITE usage.
> 
> 
>         So saying that
>         even if i send a REFER within an existing dialog, and still decouple
>         it from the existing dialog usage will not be possible.
>         If there were such a requirement, then REFER should have been
>         sent as
>         a standalone dialog creating request outside the INVITE usage.
> 
> 
>     One can argue that it *should* have been. But that isn't the point
>     of this discussion. In many cases using the same dialog as been an
>     expedient to ensure reaching the same UA when you aren't sure the
>     remote target url for the dialog is in fact routable.
> 
> 
>         The recipient of the in dialog REFER will treat it like it
>         should any
>         in dialog request. It has no way to decide if this request
>         potentially
>         had nothing to do with the existing dialog usage.
> 
> 
>     You are being sloppy in your use of "dialog" and "dialog usage", as
>     if you are not familiar with RFC 5057. If you aren't, please read it.
> 
> 
>         i agree with you when you say that an indialog refer receiving a 481
>         would mean that the INVITE dialog has been torn down.
> 
> 
>     See my reply to Dale.
> 
> 
>         However, a refer dialog usage can also terminate, if the UAC sends a
>         481 in response to the NOTIFY sent by the UAS (Suppose the
>         transaction
>         did not match).
> 
> 
>     You need to spell out more details of the case. If it were the
>     *first* NOTIFY, resulting from an in-dialog REFER, then that
>     response would be indicative of a screwup somewhere.
> 
>     However, if there had been a prior NOTIFY on the refer subscription
>     that indicated that the subscription was over, and then somehow
>     *another* NOTIFY was sent, the 481 would be appropriate, indicating
>     that the refer-usage was no longer present.
> 
> 
>         This scenario may happen irrespective of whether the
>         REFER was within the invite usage or created its own distinct
>         dialog.
>         If this happens when REFER was sent within an Invite dialog,
>         then the
>         UAC will be mistaken to clean up its INVITE dialog state.
> 
> 
>     Go read 5057. The usages are to be handled separately. So this
>     should *not* provoke the UAC to clean up its INVITE dialog usage.
> 
> 
>         I  did not understand one part in the paradox above when you said...
>         "A REFER within an established dialog having no invite usage to be
>         considered as an out of dialog REFER"
>         Do you mean to imply a REFER in some other Subscribe event package
>         dialog for example?
> 
> 
>     Yes, that is what I meant. I don't have any meaningful cases in
>     mind, but it is a theoretical possibility.
> 
>            Paul
> 
> 
>         aayush.
> 
>         On 3/25/09, Dale Worley <dworley@nortel.com
>         <mailto:dworley@nortel.com>> wrote:
> 
>             On Mon, 2009-03-23 at 21:32 -0400, Paul Kyzivat wrote:
> 
>                     I would argue that the REFER is part of the INVITE
>                     usage, because REFER
>                     in that context is intimately related to the INVITE
>                     usage -- it is
>                     intended to transfer the INVITE usage/dialog
>                     transferred to a different
>                     destination.
> 
>                 Well, I guess there is that implied association. (Part
>                 of the "do what
>                 *I* had in mind" approach that REFER has.) Since this
>                 was never spelled
>                 out, it could also be simply "while this isn't part of
>                 an INVITE usage,
>                 if there happens to be an INVITE usage, please refer
>                 that one."
> 
>                 I guess the question is: would it be valid to send a
>                 REFER within an
>                 existing dialog that had no INVITE usage? If it were
>                 valid, presumably
>                 it would have the same meaning as sending one outside a
>                 dialog, except
>                 that the resulting subscription would share the dialog.
>                 I expect that is
>                 a usage it would be difficult to find in the wild.
> 
>             That's a nice pathological case.  But within the context of
>             the current
>             5 possible meanings of REFER, it seems to be most parallel
>             to the
>             out-of-dialog REFER.
> 
>                 Ah, but now I want to reconsider. Remember that REFER
>                 can take a method
>                 name. So, rather than referring an INVITE, I could be
>                 referring MESSAGE,
>                 OPTIONS, BYE, or maybe (????) SUBSCRIBE. In some of
>                 those cases, the
>                 association to the INVITE usage is probably meaningless.
>                 (Of course its
>                 meaningful for BYE.)
> 
>             A fascinating possibility!  But I'm not sure that we've ever
>             defined how
>             a REFER within a simple INVITE dialog, specifying a
>             non-INVITE method
>             should act.  So there's no analog to follow.
> 
>                     And it seems quite safe that if the REFER receives a
>                     481 response, the
>                     INVITE usage must not be functional any more, since
>                     the far end of that
>                     usage has denied that it can act on the REFER.
> 
>                 IIRC, 5057 decided that each dialog usage must get its
>                 own 481 before it
>                 goes away. I don't think there are any cases where a
>                 single 481 response
>                 can make multiple usages go away.
> 
>                 So the important question is whether the REFER is part
>                 of the INVITE
>                 usage and so its 481 takes down the INVITE usage or not.
>                 That gets
>                 especially interesting if  the dialog already had both
>                 an INVITE usage
>                 and a SUBSCRIBE usage. Does it make sense that the refer
>                 take down the
>                 INVITE usage and leave up the SUBSCRIBE usage?
> 
>             I think we're a victim of sloppy terminology here.  If I
>             send a REFER,
>             and the far end responds 481, then clearly the far end knows
>             of no
>             INVITE usage, or otherwise it would not have responded 481.
>              So I should
>             delete my belief that there is an INVITE usage.  Whether we
>             consider the
>             REFER in question to be "part of" the INVITE usage or not
>             doesn't change
>             anything -- the 481 is evidence that the far end has no
>             knowledge of the
>             usage.
> 
>             However, we do have this fine paradox:  If we assume that a
>             REFER within
>             an established dialog that has no INVITE usage is to be
>             treated as an
>             out-of-dialog REFER, then *no* REFER will receive a 481
>             response!
>             Indeed, if I send a REFER thinking that there is an active
>             INVITE usage,
>             but the far end believes the INVITE usage has ended, will
>             cause the far
>             end to do something (initiate an independent dialog) that is
>             considerably different from what I expected (transfer the
>             current INVITE
>             usage).
> 
>             Dale
> 
> 
>             _______________________________________________
>             Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>             This list is for NEW development of the core SIP Protocol
>             Use sip-implementors@cs.columbia.edu
>             <mailto:sip-implementors@cs.columbia.edu> for questions on
>             current sip
>             Use sipping@ietf.org <mailto:sipping@ietf.org> for new
>             developments on the application of sip
> 
>         _______________________________________________
>         Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>         This list is for NEW development of the core SIP Protocol
>         Use sip-implementors@cs.columbia.edu
>         <mailto:sip-implementors@cs.columbia.edu> for questions on
>         current sip
>         Use sipping@ietf.org <mailto:sipping@ietf.org> for new
>         developments on the application of sip
> 
> 

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Subject: [Sip] The problem with draft-kaplan-sip-secure-call-id-00
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I didnt get a chance to say this at the mike.

You *cannot* change the BNF for the call-id. If you did, you might end 
up with the following problem:

A new UA gets built, compliant to this update. It is one of these 
'aggressive parsing' UAs. It rejects any message which is not compliant 
to the BNF. An older, normal RFC-3261 compliant UA sends a call-id which 
includes the @-sign.

According to the BNF of this updated 3261, and based on the strictness 
of this aggressive-parsing UAs, it rejects that request as non-compliant.

As such, you need to keep the BNF, but change the normative language 
such that an endpoint MUST NOT include the [@ host] but MUST be prepared 
to receive it.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
Cisco Fellow                                   Iselin, NJ 08830
Cisco, Voice Technology Group
jdrosen@cisco.com
http://www.jdrosen.net                         PHONE: (408) 902-3084
http://www.cisco.com

From dworley@nortel.com  Wed Mar 25 13:59:03 2009
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Subject: Re: [Sip] comments on draft-kaplan-sip-session-id-01
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On Wed, 2009-03-25 at 03:10 -0400, Hadriel Kaplan wrote:
> > If I may be immodest, the References header could a solution:  State
> > that two dialogs are related to each other by (effectively) naming both
> > call-id's together, rather than by assigning them a common identifier
> > when they are created.
> 
> Yeah I think it adds value to have the References header.  I just
> don't think it should be referencing Call-ID's, or else we'll have to
> replace that too.  It should be referencing Session-ID's or
> secure-call-id's.  :)

Certainly in security-sensitive situations, referencing Call-IDs, or at
least, ones created by UAs, would be undesirable.  But down in section
5.2 it describes how to have the B2BUA generate "linking" identifiers:

   A possible solution to this problem is for the B2BUA to create a
   "phantom" Call-Id that is suitably random, and use it in References
   headers sent in both the initial request and response.  By the
   transitivity property[Section 2], the dialogs on both sides of the
   B2BUA are declared to be related, even if the referenced dialog
   contains no messages.

   For two chained B2BUAs with no forking, this would give a message
   flow like this:

       UA 1            B2BUA 1         B2BUA 2         UA 2

            --->
            INVITE 123@aa.example.com
            Call-Id: qwerty@aa

                            --->
                            INVITE 123@transit.example.com
                            Call-Id: asdfgh@transit
                            References: QAZWSXEDCRFV;rel=serial

                                            --->
                                            INVITE 123@bb.example.com
                                            Call-Id: zxcvbn@bb
                                            References: QAZWSXEDCRFV
                                                 ;rel=serial

                                            <---
                                            SIP/2.0 200 OK
                                            Call-Id: zxcvbn@bb

                            <---
                            SIP/2.0 200 OK
                            Call-Id: asdfgh@transit
                            References: QAZWSXEDCRFV;rel=serial

            <---
            SIP/2.0 200 OK
            Call-Id: qwerty@aa
            References: QAZWSXEDCRFV;rel=serial

Dale



From HKaplan@acmepacket.com  Wed Mar 25 14:04:32 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Jonathan Rosenberg <jdrosen@cisco.com>, IETF SIP List <sip@ietf.org>
Date: Wed, 25 Mar 2009 17:05:24 -0400
Thread-Topic: [Sip] The problem with draft-kaplan-sip-secure-call-id-00
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Ooh, good point.  Fair enough, I'll change that.

-hadriel

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Jonathan Rosenberg
> Sent: Wednesday, March 25, 2009 4:51 PM
> To: IETF SIP List
> Subject: [Sip] The problem with draft-kaplan-sip-secure-call-id-00
>=20
> I didnt get a chance to say this at the mike.
>=20
> You *cannot* change the BNF for the call-id. If you did, you might end
> up with the following problem:
>=20
> A new UA gets built, compliant to this update. It is one of these
> 'aggressive parsing' UAs. It rejects any message which is not compliant
> to the BNF. An older, normal RFC-3261 compliant UA sends a call-id which
> includes the @-sign.
>=20
> According to the BNF of this updated 3261, and based on the strictness
> of this aggressive-parsing UAs, it rejects that request as non-compliant.
>=20
> As such, you need to keep the BNF, but change the normative language
> such that an endpoint MUST NOT include the [@ host] but MUST be prepared
> to receive it.
>=20
> -Jonathan R.
>=20
> --
> Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
> Cisco Fellow                                   Iselin, NJ 08830
> Cisco, Voice Technology Group
> jdrosen@cisco.com
> http://www.jdrosen.net                         PHONE: (408) 902-3084
> http://www.cisco.com
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

From drage@alcatel-lucent.com  Wed Mar 25 14:19:27 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: "sip@ietf.org" <sip@ietf.org>
Date: Wed, 25 Mar 2009 22:20:16 +0100
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Subject: [Sip] draft-kaplan-sip-secure-call-id-00
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On: http://tools.ietf.org/id/draft-kaplan-sip-secure-call-id-00.txt=09

I believe the essential question at the moment is: "Do people see this as s=
omething that solves problems or not?".

However can I point out the process implications.

The draft as currently provided updates RFC 3261 in my view. That essential=
ly makes it SIP-CORE if the WG / charter changes go ahead.

The draft appears to claim that it corrects a problem in RFC 3261? Is it an=
 essential correction? If so it should possible form part of our essential =
corrections process, rather than ultimately generating a second draft. Doin=
g so would certainly make some of the documentation implications in the sli=
des as currently stated easier. So, is it an essential correction or a SIP =
extension? I would claim there are no other standards track SIP documents.

regards

Keith


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From: "Spencer Dawkins" <spencer@wonderhamster.org>
To: "Jonathan Rosenberg" <jdrosen@cisco.com>, "IETF SIP List" <sip@ietf.org>
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Subject: Re: [Sip] The problem with draft-kaplan-sip-secure-call-id-00
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I agree with Jonathan - this is the right way to go.

Spencer


>I didnt get a chance to say this at the mike.
> 
> You *cannot* change the BNF for the call-id. If you did, you might end 
> up with the following problem:
> 
> A new UA gets built, compliant to this update. It is one of these 
> 'aggressive parsing' UAs. It rejects any message which is not compliant 
> to the BNF. An older, normal RFC-3261 compliant UA sends a call-id which 
> includes the @-sign.
> 
> According to the BNF of this updated 3261, and based on the strictness 
> of this aggressive-parsing UAs, it rejects that request as non-compliant.
> 
> As such, you need to keep the BNF, but change the normative language 
> such that an endpoint MUST NOT include the [@ host] but MUST be prepared 
> to receive it.
> 
> -Jonathan R.


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From: Venkatesh <vvenkatar@gmail.com>
To: Jonathan Rosenberg <jdrosen@cisco.com>
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Cc: IETF SIP List <sip@ietf.org>
Subject: Re: [Sip] The problem with draft-kaplan-sip-secure-call-id-00
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Hello:

I am not sure that this alone will fix the problem in totality. If I were to
use the reverse analogy i.e. what if a UA that conforms to this
specification offers the request to a "aggressive parsing" UA that is only
compliant with RFC 3261?
If we are not planning on changing the BNF, I am wondering if this draft
should simply say that the string after the "@" sign MUST NOT include a
domain name, host address, MAC....? Or if this is ambiguous, simply say that
the UA MUST generate *another* cryptographic identifier to be placed after
the @ sign?

Thanks,
Venkatesh

On Wed, Mar 25, 2009 at 1:51 PM, Jonathan Rosenberg <jdrosen@cisco.com>wrote:

> I didnt get a chance to say this at the mike.
>
> You *cannot* change the BNF for the call-id. If you did, you might end up
> with the following problem:
>
> A new UA gets built, compliant to this update. It is one of these
> 'aggressive parsing' UAs. It rejects any message which is not compliant to
> the BNF. An older, normal RFC-3261 compliant UA sends a call-id which
> includes the @-sign.
>
> According to the BNF of this updated 3261, and based on the strictness of
> this aggressive-parsing UAs, it rejects that request as non-compliant.
>
> As such, you need to keep the BNF, but change the normative language such
> that an endpoint MUST NOT include the [@ host] but MUST be prepared to
> receive it.
>
> -Jonathan R.
>
> --
> Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
> Cisco Fellow                                   Iselin, NJ 08830
> Cisco, Voice Technology Group
> jdrosen@cisco.com
> http://www.jdrosen.net                         PHONE: (408) 902-3084
> http://www.cisco.com
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

--0016369205b1217c050465f844ab
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<div>Hello:</div><div><br></div>I am not sure that this alone will fix the =
problem in totality. If I were to use the reverse analogy i.e. what if a UA=
 that conforms to this specification offers the request to a &quot;aggressi=
ve parsing&quot; UA that is only compliant with RFC 3261?<div>
<br></div><div>If we are not planning on changing the BNF, I am wondering i=
f this draft should simply say that the string after the &quot;@&quot; sign=
 MUST NOT include a domain name, host address, MAC....? Or if this is=A0amb=
iguous, simply say that the UA MUST generate *another* cryptographic identi=
fier to be placed after the @ sign?</div>
<div><br></div><div>Thanks,</div><div>Venkatesh<br><br><div class=3D"gmail_=
quote">On Wed, Mar 25, 2009 at 1:51 PM, Jonathan Rosenberg <span dir=3D"ltr=
">&lt;<a href=3D"mailto:jdrosen@cisco.com">jdrosen@cisco.com</a>&gt;</span>=
 wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">I didnt get a chance to say this at the mik=
e.<br>
<br>
You *cannot* change the BNF for the call-id. If you did, you might end up w=
ith the following problem:<br>
<br>
A new UA gets built, compliant to this update. It is one of these &#39;aggr=
essive parsing&#39; UAs. It rejects any message which is not compliant to t=
he BNF. An older, normal RFC-3261 compliant UA sends a call-id which includ=
es the @-sign.<br>

<br>
According to the BNF of this updated 3261, and based on the strictness of t=
his aggressive-parsing UAs, it rejects that request as non-compliant.<br>
<br>
As such, you need to keep the BNF, but change the normative language such t=
hat an endpoint MUST NOT include the [@ host] but MUST be prepared to recei=
ve it.<br>
<br>
-Jonathan R.<br>
<br>
-- <br>
Jonathan D. Rosenberg, Ph.D. =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 111 Wood A=
venue South<br>
Cisco Fellow =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 Iselin, NJ 08830<br>
Cisco, Voice Technology Group<br>
<a href=3D"mailto:jdrosen@cisco.com" target=3D"_blank">jdrosen@cisco.com</a=
><br>
<a href=3D"http://www.jdrosen.net" target=3D"_blank">http://www.jdrosen.net=
</a> =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 PHONE: (408) 902-3084<=
br>
<a href=3D"http://www.cisco.com" target=3D"_blank">http://www.cisco.com</a>=
<br>
_______________________________________________<br>
Sip mailing list =A0<a href=3D"https://www.ietf.org/mailman/listinfo/sip" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
This list is for NEW development of the core SIP Protocol<br>
Use <a href=3D"mailto:sip-implementors@cs.columbia.edu" target=3D"_blank">s=
ip-implementors@cs.columbia.edu</a> for questions on current sip<br>
Use <a href=3D"mailto:sipping@ietf.org" target=3D"_blank">sipping@ietf.org<=
/a> for new developments on the application of sip<br>
</blockquote></div><br></div>

--0016369205b1217c050465f844ab--

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Date: Wed, 25 Mar 2009 22:40:13 +0100
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From: =?UTF-8?Q?Victor_Pascual_=C3=81vila?= <victor.pascual.avila@gmail.com>
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Subject: Re: [Sip] The problem with draft-kaplan-sip-secure-call-id-00
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On Wed, Mar 25, 2009 at 10:34 PM, Venkatesh <vvenkatar@gmail.com> wrote:
(snip)
> If we are not planning on changing the BNF, I am wondering if this draft
> should simply say that the string after the "@" sign MUST NOT include a
> domain name, host address, MAC....? Or if this is=C2=A0ambiguous, simply =
say that
> the UA MUST generate *another* cryptographic identifier to be placed afte=
r
> the @ sign?

This would be fair enough.
--=20
Victor Pascual =C3=81vila

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Date: Wed, 25 Mar 2009 14:40:57 -0700
Message-ID: <4ff4e7370903251440l6e57f4ecp99a4f7bafcb0c873@mail.gmail.com>
From: Venkatesh <vvenkatar@gmail.com>
To: Dale Worley <dworley@nortel.com>
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Cc: IETF SIP List <sip@ietf.org>
Subject: Re: [Sip] comments on draft-kaplan-sip-session-id-01
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A crazy thought occurred to me while I was reading this draft. If the intent
is for debugging and such, I am wondering if there is any value in
considering the re-use of the P-Charging-Vector header for this purpose?
After all, this is supposed to be unique, required to be carried across
Proxies, B2BUA's if one is present, reflected in CDR's if one is generated
by an entity?? Agree that this header had specific intents and cannot span
across "trust" boundaries, but I am not sure if a SP cares when the request
'leaves' their trust boundary? It seems to satisfy the requirement for
debugging? Flame away :-).
Venkatesh

On Wed, Mar 25, 2009 at 1:59 PM, Dale Worley <dworley@nortel.com> wrote:

> On Wed, 2009-03-25 at 03:10 -0400, Hadriel Kaplan wrote:
> > > If I may be immodest, the References header could a solution:  State
> > > that two dialogs are related to each other by (effectively) naming both
> > > call-id's together, rather than by assigning them a common identifier
> > > when they are created.
> >
> > Yeah I think it adds value to have the References header.  I just
> > don't think it should be referencing Call-ID's, or else we'll have to
> > replace that too.  It should be referencing Session-ID's or
> > secure-call-id's.  :)
>
> Certainly in security-sensitive situations, referencing Call-IDs, or at
> least, ones created by UAs, would be undesirable.  But down in section
> 5.2 it describes how to have the B2BUA generate "linking" identifiers:
>
>   A possible solution to this problem is for the B2BUA to create a
>   "phantom" Call-Id that is suitably random, and use it in References
>   headers sent in both the initial request and response.  By the
>   transitivity property[Section 2], the dialogs on both sides of the
>   B2BUA are declared to be related, even if the referenced dialog
>   contains no messages.
>
>   For two chained B2BUAs with no forking, this would give a message
>   flow like this:
>
>       UA 1            B2BUA 1         B2BUA 2         UA 2
>
>            --->
>            INVITE 123@aa.example.com
>            Call-Id: qwerty@aa
>
>                            --->
>                            INVITE 123@transit.example.com
>                            Call-Id: asdfgh@transit
>                            References: QAZWSXEDCRFV;rel=serial
>
>                                            --->
>                                            INVITE 123@bb.example.com
>                                            Call-Id: zxcvbn@bb
>                                            References: QAZWSXEDCRFV
>                                                 ;rel=serial
>
>                                            <---
>                                            SIP/2.0 200 OK
>                                            Call-Id: zxcvbn@bb
>
>                            <---
>                            SIP/2.0 200 OK
>                            Call-Id: asdfgh@transit
>                            References: QAZWSXEDCRFV;rel=serial
>
>            <---
>            SIP/2.0 200 OK
>            Call-Id: qwerty@aa
>            References: QAZWSXEDCRFV;rel=serial
>
> Dale
>
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

--0016e6dd966fdf34950465f85aff
Content-Type: text/html; charset=ISO-8859-1
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A crazy thought occurred to me while I was reading this draft. If the inten=
t is for debugging and such, I am wondering if there is any value in consid=
ering the re-use of the P-Charging-Vector header for this purpose? After al=
l, this is supposed to be unique, required to be carried across Proxies, B2=
BUA&#39;s if one is present, reflected in CDR&#39;s if one is generated by =
an entity?? Agree that this header had specific intents and cannot span acr=
oss &quot;trust&quot; boundaries, but I am not sure if a SP cares when the =
request &#39;leaves&#39; their trust boundary? It seems to satisfy the requ=
irement for debugging? Flame away :-).<div>
<br></div><div>Venkatesh<br><br><div class=3D"gmail_quote">On Wed, Mar 25, =
2009 at 1:59 PM, Dale Worley <span dir=3D"ltr">&lt;<a href=3D"mailto:dworle=
y@nortel.com">dworley@nortel.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex;">
On Wed, 2009-03-25 at 03:10 -0400, Hadriel Kaplan wrote:<br>
&gt; &gt; If I may be immodest, the References header could a solution: =A0=
State<br>
&gt; &gt; that two dialogs are related to each other by (effectively) namin=
g both<br>
&gt; &gt; call-id&#39;s together, rather than by assigning them a common id=
entifier<br>
&gt; &gt; when they are created.<br>
&gt;<br>
&gt; Yeah I think it adds value to have the References header. =A0I just<br=
>
&gt; don&#39;t think it should be referencing Call-ID&#39;s, or else we&#39=
;ll have to<br>
&gt; replace that too. =A0It should be referencing Session-ID&#39;s or<br>
&gt; secure-call-id&#39;s. =A0:)<br>
<br>
Certainly in security-sensitive situations, referencing Call-IDs, or at<br>
least, ones created by UAs, would be undesirable. =A0But down in section<br=
>
5.2 it describes how to have the B2BUA generate &quot;linking&quot; identif=
iers:<br>
<br>
 =A0 A possible solution to this problem is for the B2BUA to create a<br>
 =A0 &quot;phantom&quot; Call-Id that is suitably random, and use it in Ref=
erences<br>
 =A0 headers sent in both the initial request and response. =A0By the<br>
 =A0 transitivity property[Section 2], the dialogs on both sides of the<br>
 =A0 B2BUA are declared to be related, even if the referenced dialog<br>
 =A0 contains no messages.<br>
<br>
 =A0 For two chained B2BUAs with no forking, this would give a message<br>
 =A0 flow like this:<br>
<br>
 =A0 =A0 =A0 UA 1 =A0 =A0 =A0 =A0 =A0 =A0B2BUA 1 =A0 =A0 =A0 =A0 B2BUA 2 =
=A0 =A0 =A0 =A0 UA 2<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0---&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0INVITE <a href=3D"mailto:123@aa.example.com">123@aa=
.example.com</a><br>
 =A0 =A0 =A0 =A0 =A0 =A0Call-Id: qwerty@aa<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0---&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0INVITE <a href=3D"m=
ailto:123@transit.example.com">123@transit.example.com</a><br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Call-Id: asdfgh@tra=
nsit<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0References: QAZWSXE=
DCRFV;rel=3Dserial<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0---&gt;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0INVITE <a href=3D"mailto:123@bb.example.com">123@bb.example.=
com</a><br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0Call-Id: zxcvbn@bb<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0References: QAZWSXEDCRFV<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 ;rel=3Dserial<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0&lt;---<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0SIP/2.0 200 OK<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0Call-Id: zxcvbn@bb<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&lt;---<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0SIP/2.0 200 OK<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Call-Id: asdfgh@tra=
nsit<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0References: QAZWSXE=
DCRFV;rel=3Dserial<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0&lt;---<br>
 =A0 =A0 =A0 =A0 =A0 =A0SIP/2.0 200 OK<br>
 =A0 =A0 =A0 =A0 =A0 =A0Call-Id: qwerty@aa<br>
 =A0 =A0 =A0 =A0 =A0 =A0References: QAZWSXEDCRFV;rel=3Dserial<br>
<br>
Dale<br>
<br>
<br>
_______________________________________________<br>
Sip mailing list =A0<a href=3D"https://www.ietf.org/mailman/listinfo/sip" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
This list is for NEW development of the core SIP Protocol<br>
Use <a href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs=
.columbia.edu</a> for questions on current sip<br>
Use <a href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a> for new develo=
pments on the application of sip<br>
</blockquote></div><br></div>

--0016e6dd966fdf34950465f85aff--

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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: Venkatesh <vvenkatar@gmail.com>, Dale Worley <dworley@nortel.com>
Date: Wed, 25 Mar 2009 23:32:00 +0100
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Subject: Re: [Sip] comments on draft-kaplan-sip-session-id-01
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This only works if essentially your insertion and deletion requirements of =
the header are the same as those required by your accounting system.

Keith

________________________________
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Venka=
tesh
Sent: Wednesday, March 25, 2009 9:41 PM
To: Dale Worley
Cc: IETF SIP List
Subject: Re: [Sip] comments on draft-kaplan-sip-session-id-01

A crazy thought occurred to me while I was reading this draft. If the inten=
t is for debugging and such, I am wondering if there is any value in consid=
ering the re-use of the P-Charging-Vector header for this purpose? After al=
l, this is supposed to be unique, required to be carried across Proxies, B2=
BUA's if one is present, reflected in CDR's if one is generated by an entit=
y?? Agree that this header had specific intents and cannot span across "tru=
st" boundaries, but I am not sure if a SP cares when the request 'leaves' t=
heir trust boundary? It seems to satisfy the requirement for debugging? Fla=
me away :-).

Venkatesh

On Wed, Mar 25, 2009 at 1:59 PM, Dale Worley <dworley@nortel.com<mailto:dwo=
rley@nortel.com>> wrote:
On Wed, 2009-03-25 at 03:10 -0400, Hadriel Kaplan wrote:
> > If I may be immodest, the References header could a solution:  State
> > that two dialogs are related to each other by (effectively) naming both
> > call-id's together, rather than by assigning them a common identifier
> > when they are created.
>
> Yeah I think it adds value to have the References header.  I just
> don't think it should be referencing Call-ID's, or else we'll have to
> replace that too.  It should be referencing Session-ID's or
> secure-call-id's.  :)

Certainly in security-sensitive situations, referencing Call-IDs, or at
least, ones created by UAs, would be undesirable.  But down in section
5.2 it describes how to have the B2BUA generate "linking" identifiers:

  A possible solution to this problem is for the B2BUA to create a
  "phantom" Call-Id that is suitably random, and use it in References
  headers sent in both the initial request and response.  By the
  transitivity property[Section 2], the dialogs on both sides of the
  B2BUA are declared to be related, even if the referenced dialog
  contains no messages.

  For two chained B2BUAs with no forking, this would give a message
  flow like this:

      UA 1            B2BUA 1         B2BUA 2         UA 2

           --->
           INVITE 123@aa.example.com<mailto:123@aa.example.com>
           Call-Id: qwerty@aa

                           --->
                           INVITE 123@transit.example.com<mailto:123@transi=
t.example.com>
                           Call-Id: asdfgh@transit
                           References: QAZWSXEDCRFV;rel=3Dserial

                                           --->
                                           INVITE 123@bb.example.com<mailto=
:123@bb.example.com>
                                           Call-Id: zxcvbn@bb
                                           References: QAZWSXEDCRFV
                                                ;rel=3Dserial

                                           <---
                                           SIP/2.0 200 OK
                                           Call-Id: zxcvbn@bb

                           <---
                           SIP/2.0 200 OK
                           Call-Id: asdfgh@transit
                           References: QAZWSXEDCRFV;rel=3Dserial

           <---
           SIP/2.0 200 OK
           Call-Id: qwerty@aa
           References: QAZWSXEDCRFV;rel=3Dserial

Dale


_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu<mailto:sip-implementors@cs.columbia.ed=
u> for questions on current sip
Use sipping@ietf.org<mailto:sipping@ietf.org> for new developments on the a=
pplication of sip


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>
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<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D396462922-25032009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>This only works if essentially your insertion and =
deletion=20
requirements of the header are the same as those required by your accountin=
g=20
system.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D396462922-25032009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D396462922-25032009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Keith</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px soli=
d; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Venkatesh<BR><B>Sent:</=
B>=20
  Wednesday, March 25, 2009 9:41 PM<BR><B>To:</B> Dale Worley<BR><B>Cc:</B>=
 IETF=20
  SIP List<BR><B>Subject:</B> Re: [Sip] comments on=20
  draft-kaplan-sip-session-id-01<BR></FONT><BR></DIV>
  <DIV></DIV>A crazy thought occurred to me while I was reading this draft.=
 If=20
  the intent is for debugging and such, I am wondering if there is any valu=
e in=20
  considering the re-use of the P-Charging-Vector header for this purpose? =
After=20
  all, this is supposed to be unique, required to be carried across Proxies=
,=20
  B2BUA's if one is present, reflected in CDR's if one is generated by an=20
  entity?? Agree that this header had specific intents and cannot span acro=
ss=20
  "trust" boundaries, but I am not sure if a SP cares when the request 'lea=
ves'=20
  their trust boundary? It seems to satisfy the requirement for debugging? =
Flame=20
  away :-).
  <DIV><BR></DIV>
  <DIV>Venkatesh<BR><BR>
  <DIV class=3Dgmail_quote>On Wed, Mar 25, 2009 at 1:59 PM, Dale Worley <SP=
AN=20
  dir=3Dltr>&lt;<A=20
  href=3D"mailto:dworley@nortel.com">dworley@nortel.com</A>&gt;</SPAN> wrot=
e:<BR>
  <BLOCKQUOTE class=3Dgmail_quote=20
  style=3D"PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc =
1px solid">On=20
    Wed, 2009-03-25 at 03:10 -0400, Hadriel Kaplan wrote:<BR>&gt; &gt; If I=
 may=20
    be immodest, the References header could a solution: &nbsp;State<BR>&gt=
;=20
    &gt; that two dialogs are related to each other by (effectively) naming=
=20
    both<BR>&gt; &gt; call-id's together, rather than by assigning them a c=
ommon=20
    identifier<BR>&gt; &gt; when they are created.<BR>&gt;<BR>&gt; Yeah I t=
hink=20
    it adds value to have the References header. &nbsp;I just<BR>&gt; don't=
=20
    think it should be referencing Call-ID's, or else we'll have to<BR>&gt;=
=20
    replace that too. &nbsp;It should be referencing Session-ID's or<BR>&gt=
;=20
    secure-call-id's. &nbsp;:)<BR><BR>Certainly in security-sensitive=20
    situations, referencing Call-IDs, or at<BR>least, ones created by UAs, =
would=20
    be undesirable. &nbsp;But down in section<BR>5.2 it describes how to ha=
ve=20
    the B2BUA generate "linking" identifiers:<BR><BR>&nbsp; A possible solu=
tion=20
    to this problem is for the B2BUA to create a<BR>&nbsp; "phantom" Call-I=
d=20
    that is suitably random, and use it in References<BR>&nbsp; headers sen=
t in=20
    both the initial request and response. &nbsp;By the<BR>&nbsp; transitiv=
ity=20
    property[Section 2], the dialogs on both sides of the<BR>&nbsp; B2BUA a=
re=20
    declared to be related, even if the referenced dialog<BR>&nbsp; contain=
s no=20
    messages.<BR><BR>&nbsp; For two chained B2BUAs with no forking, this wo=
uld=20
    give a message<BR>&nbsp; flow like this:<BR><BR>&nbsp; &nbsp; &nbsp; UA=
 1=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;B2BUA 1 &nbsp; &nbsp; &nbsp; &=
nbsp;=20
    B2BUA 2 &nbsp; &nbsp; &nbsp; &nbsp; UA 2<BR><BR>&nbsp; &nbsp; &nbsp; &n=
bsp;=20
    &nbsp; &nbsp;---&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;INVITE=
 <A=20
    href=3D"mailto:123@aa.example.com">123@aa.example.com</A><BR>&nbsp; &nb=
sp;=20
    &nbsp; &nbsp; &nbsp; &nbsp;Call-Id: qwerty@aa<BR><BR>&nbsp; &nbsp; &nbs=
p;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;---&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;INVITE <A=20
    href=3D"mailto:123@transit.example.com">123@transit.example.com</A><BR>=
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &=
nbsp;=20
    &nbsp; &nbsp;Call-Id: asdfgh@transit<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nb=
sp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Reference=
s:=20
    QAZWSXEDCRFV;rel=3Dserial<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nb=
sp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &=
nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;---&gt;<BR>&nbsp; &nbsp; &nbsp; &nbsp=
;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &=
nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;INVITE <A=20
    href=3D"mailto:123@bb.example.com">123@bb.example.com</A><BR>&nbsp; &nb=
sp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &=
nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Call-Id:=
=20
    zxcvbn@bb<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &=
nbsp;=20
    &nbsp; &nbsp;References: QAZWSXEDCRFV<BR>&nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &=
nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    ;rel=3Dserial<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &=
nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &=
nbsp;=20
    &nbsp; &nbsp; &nbsp;&lt;---<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &=
nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;SIP/2.0 200 OK<BR>&nbsp; &nbsp; &nbsp=
;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &=
nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Call-Id:=20
    zxcvbn@bb<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&lt;---<BR>&nbsp; &nbsp; &nbsp=
;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;SIP/2.0 200 OK<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Call-Id:=20
    asdfgh@transit<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;References:=20
    QAZWSXEDCRFV;rel=3Dserial<BR><BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=20
    &nbsp;&lt;---<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;SIP/2.0 200=20
    OK<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Call-Id: qwerty@aa<BR>&n=
bsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;References:=20
    QAZWSXEDCRFV;rel=3Dserial<BR><BR>Dale<BR><BR><BR>______________________=
_________________________<BR>Sip=20
    mailing list &nbsp;<A href=3D"https://www.ietf.org/mailman/listinfo/sip=
"=20
    target=3D_blank>https://www.ietf.org/mailman/listinfo/sip</A><BR>This l=
ist is=20
    for NEW development of the core SIP Protocol<BR>Use <A=20
    href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.co=
lumbia.edu</A>=20
    for questions on current sip<BR>Use <A=20
    href=3D"mailto:sipping@ietf.org">sipping@ietf.org</A> for new developme=
nts on=20
    the application of=20
sip<BR></BLOCKQUOTE></DIV><BR></DIV></BLOCKQUOTE></BODY></HTML>

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From: "Mary Barnes" <mary.barnes@nortel.com>
To: <sip@ietf.org>, <sipping@ietf.org>
Subject: [Sip] FW: [Sipping] Slides for SIP and SIPPING sessions
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=20
A reminder that we need all the charts for Friday's SIPPING WG session =
no later than 1500 tomorrow (Thursday, March 26) so that we can get them =
all uploaded and on a single computer for presenting. So, far we've =
gotten one presentation.=20

If you could please send the charts to both sets of chairs, that would =
be helpful, since this is another mixed session covering SIP and SIPPING =
chartered/mature drafts.

Thanks,
Mary.=20


-----Original Message-----
From: sipping-bounces@ietf.org on behalf of DRAGE, Keith (Keith)
Sent: Mon 3/23/2009 9:00 PM
To: sip@ietf.org; sipping
Cc: sip-chairs@tools.ietf.org; sipping-chairs@tools.ietf.org
Subject: [Sipping] Slides for SIP and SIPPING sessions
=20
(As SIP WG chair)

Please make sure you submit your slides well before the start of the =
sessions of SIP or SIPPING in which you are presenting. We would suggest =
deadlines of:

Tuesday 13:00 session - submit by Tuesday 9:00
Wednesday 13:00 session - submit by Wednesday 9:00
Friday 9:00 session - submit by Thursday 15:00

These are the latest time for submission - there is nothing wrong with =
sending them to us early.

To avoid confusion, you may want to send them to both sets of chairs, =
given the combined nature of the agenda this time round.

regards

Keith
_______________________________________________
Sipping mailing list  https://www.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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<P><FONT SIZE=3D2><BR>
A reminder that we need all the charts for Friday's SIPPING WG session =
no later than 1500 tomorrow (Thursday, March 26) so that we can get them =
all uploaded and on a single computer for presenting. So, far we've =
gotten one presentation.<BR>
<BR>
If you could please send the charts to both sets of chairs, that would =
be helpful, since this is another mixed session covering SIP and SIPPING =
chartered/mature drafts.<BR>
<BR>
Thanks,<BR>
Mary.<BR>
<BR>
<BR>
-----Original Message-----<BR>
From: sipping-bounces@ietf.org on behalf of DRAGE, Keith (Keith)<BR>
Sent: Mon 3/23/2009 9:00 PM<BR>
To: sip@ietf.org; sipping<BR>
Cc: sip-chairs@tools.ietf.org; sipping-chairs@tools.ietf.org<BR>
Subject: [Sipping] Slides for SIP and SIPPING sessions<BR>
<BR>
(As SIP WG chair)<BR>
<BR>
Please make sure you submit your slides well before the start of the =
sessions of SIP or SIPPING in which you are presenting. We would suggest =
deadlines of:<BR>
<BR>
Tuesday 13:00 session - submit by Tuesday 9:00<BR>
Wednesday 13:00 session - submit by Wednesday 9:00<BR>
Friday 9:00 session - submit by Thursday 15:00<BR>
<BR>
These are the latest time for submission - there is nothing wrong with =
sending them to us early.<BR>
<BR>
To avoid confusion, you may want to send them to both sets of chairs, =
given the combined nature of the agenda this time round.<BR>
<BR>
regards<BR>
<BR>
Keith<BR>
_______________________________________________<BR>
Sipping mailing list&nbsp; <A =
HREF=3D"https://www.ietf.org/mailman/listinfo/sipping">https://www.ietf.o=
rg/mailman/listinfo/sipping</A><BR>
This list is for NEW development of the application of SIP<BR>
Use sip-implementors@cs.columbia.edu for questions on current sip<BR>
Use sip@ietf.org for new developments of core SIP<BR>
<BR>
<BR>
</FONT>
</P>

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From abhatnagar192006@gmail.com  Wed Mar 25 18:51:19 2009
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Date: Thu, 26 Mar 2009 02:22:10 +0030
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From: aayush bhatnagar <abhatnagar192006@gmail.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
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Cc: SIP IETF <SIP@ietf.org>, Brett Tate <brett@broadsoft.com>
Subject: Re: [Sip] Is REFER within dialog part of INVITE or subscription usage? (was RE: SIP INFO)
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I am in favor of dialog reuse as long as i can achieve something by it
practically. Coming from the IMS world, i am always on the lookout for
creating interesting sip apps around new concepts.

This RFC, i think also imposes certain functionality on the sip client
(clients being in a very sorry state in IMS).

aayush

On 3/25/09, Paul Kyzivat <pkyzivat@cisco.com> wrote:
>
>
> aayush bhatnagar wrote:
>>>Go read 5057. The usages are to be handled separately. So this should
>> *not* provoke the UAC >to clean up its INVITE dialog usage.
>>
>> I have read the relevant RFC. From 5057:
>
> OK, sorry. It just sounded like you might not have.
>
>> -Existing implementations that destroy all other usages in the dialog
>> will continue to function as -they do now, except that peers following
>> the recommendation will attempt to do things with the -other usages and
>> this element will return 481s for each of them until they are all gone.
>>
>> I think it is more feasible to destroy the entire dialog, than keeping
>> on to 'hung' state until all the dialog usages are all gone as suggested
>> above. I am familiar with implementations that clean up dialog state
>> when they get a 481-dialog does not exist response, even if there were
>> other 'active' usages of that dialog.
>
> I'm not surprised. Its often "more feasible" to do the wrong thing than
> the right thing. But that doesn't make it right. While its not written
> in black and white in 3261, there is enough info there to infer that you
> need to treat them separately.
>
>> The reason being, that if an implementation cannot find a matching
>> dialog as characterized by the local tag, remote tag, call-id, local and
>> remote sequence nos etc and has rejected the request by a 481, there is
>> no point in keeping the dialog state at the UAC.
>
> This is all hashed out in 5057. The problem is that the 481 response is
> *ambiguous*. The way to maximize interop is to treat the 481 as meaning
> the least serious thing that it could mean - that only the usage has
> gone away. If it turns out that the whole dialog went away, then you
> will eventually find that out anyway. Of course it doesn't hurt to treat
> it as meaning all the usages went away if you would not continue all of
> them anyway. But if the other usage(s) continue to be of value then it
> you are hurting yourself by assuming they have gone away when they might
> not have.
>
>> As for example, if you receive a 481 response for a REFER sent within
>> the INVITE dialog, there is no point in keeping the REFER usage alive.
>> You probably sent the REFER to transfer a call, which the server
>> complains that the call itself does not exist (481).
>
> That's not the case of relevance. If you got a 481 for a REFER, the
> refer of course failed, and so there is no refer usage to keep alive.
>
> We are still, in this thread, trying to decide what a 481 to a REFER
> means. I think it won't be ambiguous - that it can *only* mean that the
> whole dialog is gone, since it is never associated with a preexisting usage.
>
> Even so, you would be well advised to probe any existing usages with
> in-usage messages to verify if they are alive, even though most likely
> they are not.
>
>> Moreover from 5057:
>>
>> -Processing multiple usages correctly is not completely understood.
>> -What is understood is difficult to implement and is very likely to
>> -lead to interoperability problems. The best way to avoid the trouble
>> -that comes with such complexity is to avoid it altogether.
>> -When designing new applications or features that use SIP dialogs, do
>> -not require endpoints to construct multiple usages to participate in
>> -the application or use the feature.
>>
>> This is very true. By defining multiple dialog usages, you are
>> essentially extending the dialog state (as defined by 3261) by adding
>> more attributes to it (such as subscription duration etc). I cant think
>> of a practical use case, where a subscription that shared the invite
>> dialog usage still needed to be preserved, even when the original INVITE
>> dialog was no longer active.
>
> We aren't *adding more attributes to it*. It is what it is and has been.
> It has been implicitly legal in spite of the problems. And it *is* used.
> There are even reasonable use cases for independent termination of the
> usages. For instance, a refer usage sharing a dialog with an invite
> usage may meaningfully survive after the invite usage terminates.
>
>> As for example a subscription to the 'conference' event package sent
>> within the scope of an INVITE dialog. If i leave the conference by
>> sending a BYE, and my subscription's duration is still valid for an
>> hour...will i still keep on getting redundant SIP notifications for
>> changes to the membership to that conference ?
>
>
>> In the sense of 5057, the
>> subscribe usage will be preserved until it expires, or until the UAC
>> explicitly un-subscribes, or the conference server has to send me a
>> final NOTIFY with my subsription-state = terminated.
>
> IMO that is exactly as it *should* be. If the subscriber doesn't want to
> maintain the subscription, it can simply wipe out the state of the
> subscription. Then when another NOTIFY is received, a 481 will be sent
> because there is no state, forcing the end of the usage.
>
> But *assuming* that the subscription *should* go away when the invite
> ends is, IMO, wrong. There may be valid reasons to retain the subscription.
>
>> It is not important that 5057 is allowing us many theoretical
>> possibilities by presenting multiple dialog usages.
>>
>> What is more important is what extra do we get by reusing a dialog, and
>> how do implementations leverage upon such a possibility and create
>> meaningful applications around it ?
>>
>> I would like to share some examples:
>>
>> 1. As for example an application that can practically leverage upon 5057
>> may send a SUBSCRIBE to the 'dialog' event package (rfc 4235) that
>> reuses the INVITE dialog can be used for creating SIP call tracing
>> applications.
>>
>> 2. Other Implementations may include automatic callback applications by
>> this mechanism (ring back when my sip phone is idle) etc.
>>
>> 3. Yet another application may send me by pre-paid account balance after
>> i disconnect my ongoing dialog, as it is subscribed to my dialog state.
>>
>> In all these example applications, the endpoint receiving the SUBSCRIBE
>> will need to send a NOTIFY that re-uses its existing INVITE dialog
>> (NOTIFY acting as a target refresh in the sense of 5057).
>
> Sounds like you are in favor of dialog reuse.
>
> 	Paul
>
>> Best Regards
>>
>> aayush
>>
>>
>> On Wed, Mar 25, 2009 at 7:36 PM, Paul Kyzivat <pkyzivat@cisco.com
>> <mailto:pkyzivat@cisco.com>> wrote:
>>
>>
>>
>>     aayush bhatnagar wrote:
>>
>>         If you send an in dialog request (REFER in our case), then you
>>         cannot
>>         decouple its semantics with the existing dialog usage.
>>
>>
>>     You say "*the* dialog usage" as if there can be only one. There
>>     could be more than one. It might be coupled to:
>>     - the *only* existing usage
>>     - to *one* of several existing usages
>>     - to *none* of the one or more existing usages
>>
>>     For instance, I could have a dialog with just an INVITE usage (the
>>     common case) and within that dialog send a REFER of a MESSAGE. That
>>     MESSAGE really has nothing to do with the INVITE usage.
>>
>>
>>         So saying that
>>         even if i send a REFER within an existing dialog, and still
>> decouple
>>         it from the existing dialog usage will not be possible.
>>         If there were such a requirement, then REFER should have been
>>         sent as
>>         a standalone dialog creating request outside the INVITE usage.
>>
>>
>>     One can argue that it *should* have been. But that isn't the point
>>     of this discussion. In many cases using the same dialog as been an
>>     expedient to ensure reaching the same UA when you aren't sure the
>>     remote target url for the dialog is in fact routable.
>>
>>
>>         The recipient of the in dialog REFER will treat it like it
>>         should any
>>         in dialog request. It has no way to decide if this request
>>         potentially
>>         had nothing to do with the existing dialog usage.
>>
>>
>>     You are being sloppy in your use of "dialog" and "dialog usage", as
>>     if you are not familiar with RFC 5057. If you aren't, please read it.
>>
>>
>>         i agree with you when you say that an indialog refer receiving a
>> 481
>>         would mean that the INVITE dialog has been torn down.
>>
>>
>>     See my reply to Dale.
>>
>>
>>         However, a refer dialog usage can also terminate, if the UAC sends
>> a
>>         481 in response to the NOTIFY sent by the UAS (Suppose the
>>         transaction
>>         did not match).
>>
>>
>>     You need to spell out more details of the case. If it were the
>>     *first* NOTIFY, resulting from an in-dialog REFER, then that
>>     response would be indicative of a screwup somewhere.
>>
>>     However, if there had been a prior NOTIFY on the refer subscription
>>     that indicated that the subscription was over, and then somehow
>>     *another* NOTIFY was sent, the 481 would be appropriate, indicating
>>     that the refer-usage was no longer present.
>>
>>
>>         This scenario may happen irrespective of whether the
>>         REFER was within the invite usage or created its own distinct
>>         dialog.
>>         If this happens when REFER was sent within an Invite dialog,
>>         then the
>>         UAC will be mistaken to clean up its INVITE dialog state.
>>
>>
>>     Go read 5057. The usages are to be handled separately. So this
>>     should *not* provoke the UAC to clean up its INVITE dialog usage.
>>
>>
>>         I  did not understand one part in the paradox above when you
>> said...
>>         "A REFER within an established dialog having no invite usage to be
>>         considered as an out of dialog REFER"
>>         Do you mean to imply a REFER in some other Subscribe event package
>>         dialog for example?
>>
>>
>>     Yes, that is what I meant. I don't have any meaningful cases in
>>     mind, but it is a theoretical possibility.
>>
>>            Paul
>>
>>
>>         aayush.
>>
>>         On 3/25/09, Dale Worley <dworley@nortel.com
>>         <mailto:dworley@nortel.com>> wrote:
>>
>>             On Mon, 2009-03-23 at 21:32 -0400, Paul Kyzivat wrote:
>>
>>                     I would argue that the REFER is part of the INVITE
>>                     usage, because REFER
>>                     in that context is intimately related to the INVITE
>>                     usage -- it is
>>                     intended to transfer the INVITE usage/dialog
>>                     transferred to a different
>>                     destination.
>>
>>                 Well, I guess there is that implied association. (Part
>>                 of the "do what
>>                 *I* had in mind" approach that REFER has.) Since this
>>                 was never spelled
>>                 out, it could also be simply "while this isn't part of
>>                 an INVITE usage,
>>                 if there happens to be an INVITE usage, please refer
>>                 that one."
>>
>>                 I guess the question is: would it be valid to send a
>>                 REFER within an
>>                 existing dialog that had no INVITE usage? If it were
>>                 valid, presumably
>>                 it would have the same meaning as sending one outside a
>>                 dialog, except
>>                 that the resulting subscription would share the dialog.
>>                 I expect that is
>>                 a usage it would be difficult to find in the wild.
>>
>>             That's a nice pathological case.  But within the context of
>>             the current
>>             5 possible meanings of REFER, it seems to be most parallel
>>             to the
>>             out-of-dialog REFER.
>>
>>                 Ah, but now I want to reconsider. Remember that REFER
>>                 can take a method
>>                 name. So, rather than referring an INVITE, I could be
>>                 referring MESSAGE,
>>                 OPTIONS, BYE, or maybe (????) SUBSCRIBE. In some of
>>                 those cases, the
>>                 association to the INVITE usage is probably meaningless.
>>                 (Of course its
>>                 meaningful for BYE.)
>>
>>             A fascinating possibility!  But I'm not sure that we've ever
>>             defined how
>>             a REFER within a simple INVITE dialog, specifying a
>>             non-INVITE method
>>             should act.  So there's no analog to follow.
>>
>>                     And it seems quite safe that if the REFER receives a
>>                     481 response, the
>>                     INVITE usage must not be functional any more, since
>>                     the far end of that
>>                     usage has denied that it can act on the REFER.
>>
>>                 IIRC, 5057 decided that each dialog usage must get its
>>                 own 481 before it
>>                 goes away. I don't think there are any cases where a
>>                 single 481 response
>>                 can make multiple usages go away.
>>
>>                 So the important question is whether the REFER is part
>>                 of the INVITE
>>                 usage and so its 481 takes down the INVITE usage or not.
>>                 That gets
>>                 especially interesting if  the dialog already had both
>>                 an INVITE usage
>>                 and a SUBSCRIBE usage. Does it make sense that the refer
>>                 take down the
>>                 INVITE usage and leave up the SUBSCRIBE usage?
>>
>>             I think we're a victim of sloppy terminology here.  If I
>>             send a REFER,
>>             and the far end responds 481, then clearly the far end knows
>>             of no
>>             INVITE usage, or otherwise it would not have responded 481.
>>              So I should
>>             delete my belief that there is an INVITE usage.  Whether we
>>             consider the
>>             REFER in question to be "part of" the INVITE usage or not
>>             doesn't change
>>             anything -- the 481 is evidence that the far end has no
>>             knowledge of the
>>             usage.
>>
>>             However, we do have this fine paradox:  If we assume that a
>>             REFER within
>>             an established dialog that has no INVITE usage is to be
>>             treated as an
>>             out-of-dialog REFER, then *no* REFER will receive a 481
>>             response!
>>             Indeed, if I send a REFER thinking that there is an active
>>             INVITE usage,
>>             but the far end believes the INVITE usage has ended, will
>>             cause the far
>>             end to do something (initiate an independent dialog) that is
>>             considerably different from what I expected (transfer the
>>             current INVITE
>>             usage).
>>
>>             Dale
>>
>>
>>             _______________________________________________
>>             Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>             This list is for NEW development of the core SIP Protocol
>>             Use sip-implementors@cs.columbia.edu
>>             <mailto:sip-implementors@cs.columbia.edu> for questions on
>>             current sip
>>             Use sipping@ietf.org <mailto:sipping@ietf.org> for new
>>             developments on the application of sip
>>
>>         _______________________________________________
>>         Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>         This list is for NEW development of the core SIP Protocol
>>         Use sip-implementors@cs.columbia.edu
>>         <mailto:sip-implementors@cs.columbia.edu> for questions on
>>         current sip
>>         Use sipping@ietf.org <mailto:sipping@ietf.org> for new
>>         developments on the application of sip
>>
>>
>

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To: Venkatesh <vvenkatar@gmail.com>, Jonathan Rosenberg <jdrosen@cisco.com>
Date: Wed, 25 Mar 2009 22:39:41 -0400
Thread-Topic: [Sip] The problem with draft-kaplan-sip-secure-call-id-00
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RFC 3261 made the @ and after it optional in ABNF.  So a strict parser to 3=
261 must allow it.  In fact we've seen plenty of Call-ID's without it, from=
 some very common equipment.
We just can't change the BNF because a strict parser parsing to this new dr=
aft would not accept 3261 call-id's.
-hadriel


________________________________
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Venka=
tesh
Sent: Wednesday, March 25, 2009 5:35 PM
To: Jonathan Rosenberg
Cc: IETF SIP List
Subject: Re: [Sip] The problem with draft-kaplan-sip-secure-call-id-00

Hello:

I am not sure that this alone will fix the problem in totality. If I were t=
o use the reverse analogy i.e. what if a UA that conforms to this specifica=
tion offers the request to a "aggressive parsing" UA that is only compliant=
 with RFC 3261?

If we are not planning on changing the BNF, I am wondering if this draft sh=
ould simply say that the string after the "@" sign MUST NOT include a domai=
n name, host address, MAC....? Or if this is ambiguous, simply say that the=
 UA MUST generate *another* cryptographic identifier to be placed after the=
 @ sign?

Thanks,
Venkatesh
On Wed, Mar 25, 2009 at 1:51 PM, Jonathan Rosenberg <jdrosen@cisco.com<mail=
to:jdrosen@cisco.com>> wrote:
I didnt get a chance to say this at the mike.

You *cannot* change the BNF for the call-id. If you did, you might end up w=
ith the following problem:

A new UA gets built, compliant to this update. It is one of these 'aggressi=
ve parsing' UAs. It rejects any message which is not compliant to the BNF. =
An older, normal RFC-3261 compliant UA sends a call-id which includes the @=
-sign.

According to the BNF of this updated 3261, and based on the strictness of t=
his aggressive-parsing UAs, it rejects that request as non-compliant.

As such, you need to keep the BNF, but change the normative language such t=
hat an endpoint MUST NOT include the [@ host] but MUST be prepared to recei=
ve it.

-Jonathan R.

--
Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
Cisco Fellow                                   Iselin, NJ 08830
Cisco, Voice Technology Group
jdrosen@cisco.com<mailto:jdrosen@cisco.com>
http://www.jdrosen.net                         PHONE: (408) 902-3084
http://www.cisco.com
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu<mailto:sip-implementors@cs.columbia.ed=
u> for questions on current sip
Use sipping@ietf.org<mailto:sipping@ietf.org> for new developments on the a=
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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>RFC 3261 made the @ and after it optio=
nal
in ABNF. &nbsp;So a strict parser to 3261 must allow it. &nbsp;In fact we&#=
8217;ve seen plenty
of Call-ID&#8217;s without it, from some very common equipment. <o:p></o:p>=
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>We just can&#8217;t change the BNF bec=
ause
a strict parser parsing to this new draft would not accept 3261 call-id&#82=
17;s.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>-hadriel<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>
sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <b><span style=3D'font-w=
eight:
bold'>On Behalf Of </span></b>Venkatesh<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March 25, 2=
009
5:35 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Jonathan Rosenberg<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <st1:PersonName w:st=3D"=
on">IETF
 SIP List</st1:PersonName><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [Sip] The probl=
em with
draft-kaplan-sip-secure-call-id-00</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>Hello:<o:p></o:p></span></font></p>

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<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>I am not sure that this alone will fix the problem in totality. If =
I
were to use the reverse analogy i.e. what if a UA that conforms to this
specification offers the request to a &quot;aggressive parsing&quot; UA tha=
t is
only compliant with RFC 3261?<o:p></o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>If we are not planning on changing the BNF, I am wondering if this
draft should simply say that the string after the &quot;@&quot; sign MUST N=
OT
include a domain name, host address, MAC....? Or if this is&nbsp;ambiguous,
simply say that the UA MUST generate *another* cryptographic identifier to =
be
placed after the @ sign?<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>Thanks,<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>Venkatesh<o:p></o=
:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>On Wed, Mar 25, 2009 at 1:51 PM, Jonathan Rosenberg &lt;<a
href=3D"mailto:jdrosen@cisco.com">jdrosen@cisco.com</a>&gt; wrote:<o:p></o:=
p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>I didnt get a chance to say this at the mike.<br>
<br>
You *cannot* change the BNF for the call-id. If you did, you might end up w=
ith
the following problem:<br>
<br>
A new UA gets built, compliant to this update. It is one of these 'aggressi=
ve
parsing' UAs. It rejects any message which is not compliant to the BNF. An
older, normal RFC-3261 compliant UA sends a call-id which includes the @-si=
gn.<br>
<br>
According to the BNF of this updated 3261, and based on the strictness of t=
his
aggressive-parsing UAs, it rejects that request as non-compliant.<br>
<br>
As such, you need to keep the BNF, but change the normative language such t=
hat
an endpoint MUST NOT include the [@ host] but MUST be prepared to receive i=
t.<br>
<br>
-Jonathan R.<br>
<br>
-- <br>
Jonathan D. Rosenberg, Ph.D. &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p;
&nbsp; &nbsp; <st1:Street w:st=3D"on"><st1:address w:st=3D"on">111 Wood Ave=
nue
  South</st1:address></st1:Street><br>
Cisco Fellow &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; <st1:place w:st=3D"=
on"><st1:City
 w:st=3D"on">Iselin</st1:City>, <st1:State w:st=3D"on">NJ</st1:State> <st1:=
PostalCode
 w:st=3D"on">08830</st1:PostalCode></st1:place><br>
Cisco, Voice Technology Group<br>
<a href=3D"mailto:jdrosen@cisco.com" target=3D"_blank">jdrosen@cisco.com</a=
><br>
<a href=3D"http://www.jdrosen.net" target=3D"_blank">http://www.jdrosen.net=
</a>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; PHONE: (408) 902-3084<br>
<a href=3D"http://www.cisco.com" target=3D"_blank">http://www.cisco.com</a>=
<br>
_______________________________________________<br>
Sip mailing list &nbsp;<a href=3D"https://www.ietf.org/mailman/listinfo/sip=
"
target=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
This list is for NEW development of the core SIP Protocol<br>
Use <a href=3D"mailto:sip-implementors@cs.columbia.edu" target=3D"_blank">s=
ip-implementors@cs.columbia.edu</a>
for questions on current sip<br>
Use <a href=3D"mailto:sipping@ietf.org" target=3D"_blank">sipping@ietf.org<=
/a> for
new developments on the application of sip<o:p></o:p></span></font></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Venkatesh <vvenkatar@gmail.com>, Dale Worley <dworley@nortel.com>
Date: Wed, 25 Mar 2009 22:44:04 -0400
Thread-Topic: [Sip] comments on draft-kaplan-sip-session-id-01
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Yeah JDR mentioned the same thing.  Unfortunately that header value is crea=
ted by middleboxes and removed by middleboxes.
-hadriel


________________________________
From: Venkatesh [mailto:vvenkatar@gmail.com]
Sent: Wednesday, March 25, 2009 5:41 PM
To: Dale Worley
Cc: Hadriel Kaplan; IETF SIP List
Subject: Re: [Sip] comments on draft-kaplan-sip-session-id-01

A crazy thought occurred to me while I was reading this draft. If the inten=
t is for debugging and such, I am wondering if there is any value in consid=
ering the re-use of the P-Charging-Vector header for this purpose? After al=
l, this is supposed to be unique, required to be carried across Proxies, B2=
BUA's if one is present, reflected in CDR's if one is generated by an entit=
y?? Agree that this header had specific intents and cannot span across "tru=
st" boundaries, but I am not sure if a SP cares when the request 'leaves' t=
heir trust boundary? It seems to satisfy the requirement for debugging? Fla=
me away :-).

Venkatesh
On Wed, Mar 25, 2009 at 1:59 PM, Dale Worley <dworley@nortel.com<mailto:dwo=
rley@nortel.com>> wrote:
On Wed, 2009-03-25 at 03:10 -0400, Hadriel Kaplan wrote:
> > If I may be immodest, the References header could a solution:  State
> > that two dialogs are related to each other by (effectively) naming both
> > call-id's together, rather than by assigning them a common identifier
> > when they are created.
>
> Yeah I think it adds value to have the References header.  I just
> don't think it should be referencing Call-ID's, or else we'll have to
> replace that too.  It should be referencing Session-ID's or
> secure-call-id's.  :)

Certainly in security-sensitive situations, referencing Call-IDs, or at
least, ones created by UAs, would be undesirable.  But down in section
5.2 it describes how to have the B2BUA generate "linking" identifiers:

  A possible solution to this problem is for the B2BUA to create a
  "phantom" Call-Id that is suitably random, and use it in References
  headers sent in both the initial request and response.  By the
  transitivity property[Section 2], the dialogs on both sides of the
  B2BUA are declared to be related, even if the referenced dialog
  contains no messages.

  For two chained B2BUAs with no forking, this would give a message
  flow like this:

      UA 1            B2BUA 1         B2BUA 2         UA 2

           --->
           INVITE 123@aa.example.com<mailto:123@aa.example.com>
           Call-Id: qwerty@aa

                           --->
                           INVITE 123@transit.example.com<mailto:123@transi=
t.example.com>
                           Call-Id: asdfgh@transit
                           References: QAZWSXEDCRFV;rel=3Dserial

                                           --->
                                           INVITE 123@bb.example.com<mailto=
:123@bb.example.com>
                                           Call-Id: zxcvbn@bb
                                           References: QAZWSXEDCRFV
                                                ;rel=3Dserial

                                           <---
                                           SIP/2.0 200 OK
                                           Call-Id: zxcvbn@bb

                           <---
                           SIP/2.0 200 OK
                           Call-Id: asdfgh@transit
                           References: QAZWSXEDCRFV;rel=3Dserial

           <---
           SIP/2.0 200 OK
           Call-Id: qwerty@aa
           References: QAZWSXEDCRFV;rel=3Dserial

Dale


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu<mailto:sip-implementors@cs.columbia.ed=
u> for questions on current sip
Use sipping@ietf.org<mailto:sipping@ietf.org> for new developments on the a=
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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Yeah JDR mentioned the same thing. &nb=
sp;Unfortunately
that header value is created by middleboxes and removed by middleboxes.<o:p=
></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>-hadriel<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Venkates=
h
[mailto:vvenkatar@gmail.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March 25, 2=
009
5:41 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> <st1:PersonName w:st=3D"=
on">Dale
 Worley</st1:PersonName><br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> Hadriel Kaplan; <st1:Per=
sonName
w:st=3D"on">IETF SIP List</st1:PersonName><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [Sip] comments =
on
draft-kaplan-sip-session-id-01</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>A crazy thought occurred to me while I was reading this draft. If t=
he
intent is for debugging and such, I am wondering if there is any value in c=
onsidering
the re-use of the P-Charging-Vector header for this purpose? After all, thi=
s is
supposed to be unique, required to be carried across Proxies, B2BUA's if on=
e is
present, reflected in CDR's if one is generated by an entity?? Agree that t=
his
header had specific intents and cannot span across &quot;trust&quot;
boundaries, but I am not sure if a SP cares when the request 'leaves' their
trust boundary? It seems to satisfy the requirement for debugging? Flame aw=
ay
:-).<o:p></o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>Venkatesh<o:p></o=
:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>On Wed, Mar 25, 2009 at 1:59 PM, <st1:PersonName w:st=3D"on">Dale W=
orley</st1:PersonName>
&lt;<a href=3D"mailto:dworley@nortel.com">dworley@nortel.com</a>&gt; wrote:=
<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>On Wed, 2009-03-25 at 03:10 -0400, Hadriel Kaplan wrote:<br>
&gt; &gt; If I may be immodest, the References header could a solution:
&nbsp;State<br>
&gt; &gt; that two dialogs are related to each other by (effectively) namin=
g
both<br>
&gt; &gt; call-id's together, rather than by assigning them a common identi=
fier<br>
&gt; &gt; when they are created.<br>
&gt;<br>
&gt; Yeah I think it adds value to have the References header. &nbsp;I just=
<br>
&gt; don't think it should be referencing Call-ID's, or else we'll have to<=
br>
&gt; replace that too. &nbsp;It should be referencing Session-ID's or<br>
&gt; secure-call-id's. &nbsp;:)<br>
<br>
Certainly in security-sensitive situations, referencing Call-IDs, or at<br>
least, ones created by UAs, would be undesirable. &nbsp;But down in section=
<br>
5.2 it describes how to have the B2BUA generate &quot;linking&quot;
identifiers:<br>
<br>
&nbsp; A possible solution to this problem is for the B2BUA to create a<br>
&nbsp; &quot;phantom&quot; Call-Id that is suitably random, and use it in
References<br>
&nbsp; headers sent in both the initial request and response. &nbsp;By the<=
br>
&nbsp; transitivity property[Section 2], the dialogs on both sides of the<b=
r>
&nbsp; B2BUA are declared to be related, even if the referenced dialog<br>
&nbsp; contains no messages.<br>
<br>
&nbsp; For two chained B2BUAs with no forking, this would give a message<br=
>
&nbsp; flow like this:<br>
<br>
&nbsp; &nbsp; &nbsp; UA 1 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;B2BUA 1
&nbsp; &nbsp; &nbsp; &nbsp; B2BUA 2 &nbsp; &nbsp; &nbsp; &nbsp; UA 2<br>
<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;---&gt;<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;INVITE <a
href=3D"mailto:123@aa.example.com">123@aa.example.com</a><br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Call-Id: qwerty@aa<br>
<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; &nbsp; &nbsp;---&gt;<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; &nbsp; &nbsp;INVITE <a href=3D"mailto:123@transit.example.com">123@t=
ransit.example.com</a><br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; &nbsp; &nbsp;Call-Id: asdfgh@transit<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; &nbsp; &nbsp;References: QAZWSXEDCRFV;rel=3Dserial<br>
<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp;---&gt;<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp;INVITE <a href=3D"mailto:123@bb.example.com">123@bb.example.com</a><b=
r>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp;Call-Id: zxcvbn@bb<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp;References: QAZWSXEDCRFV<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; &nbsp; ;rel=3Dserial<br>
<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp;&lt;---<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp;SIP/2.0 200 OK<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp;Call-Id: zxcvbn@bb<br>
<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; &nbsp; &nbsp;&lt;---<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; &nbsp; &nbsp;SIP/2.0 200 OK<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; &nbsp; &nbsp;Call-Id: asdfgh@transit<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; &nbsp; &nbsp;References: QAZWSXEDCRFV;rel=3Dserial<br>
<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&lt;---<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;SIP/2.0 200 OK<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Call-Id: qwerty@aa<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;References: QAZWSXEDCRFV;rel=3Dser=
ial<br>
<br>
Dale<br>
<br>
<br>
_______________________________________________<br>
Sip mailing list &nbsp;<a href=3D"https://www.ietf.org/mailman/listinfo/sip=
"
target=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
This list is for NEW development of the core SIP Protocol<br>
Use <a href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs=
.columbia.edu</a>
for questions on current sip<br>
Use <a href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a> for new develo=
pments
on the application of sip<o:p></o:p></span></font></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

</div>

</div>

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From chris@ns-technologies.com  Thu Mar 26 07:36:25 2009
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Hi,

A '00' Draft has been submitted to the IETF archives that can be 
retrieved from the following link -

http://www.ietf.org/internet-drafts/draft-boulton-sip-endpoint-view-00.txt.  
The draft discusses some problems encountered when using certain SIP 
mechanisms in an application sequenced environment.
All feedback welcomed.

Thanks.

Chris.


From christer.holmberg@ericsson.com  Thu Mar 26 10:40:32 2009
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Hi,

I've submitted a new version (-04) of the keep draft.

The requirement text is now alligned with the text shown on the slides
during the SFO SIP WG presentation.

There are no other changes in the document from the previous version.

Regards,

Christer

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<P><FONT SIZE=3D2 FACE=3D"Arial">Hi,</FONT>
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<P><FONT SIZE=3D2 FACE=3D"Arial">I've submitted a new version (-04) of =
the keep draft.</FONT>
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<P><FONT SIZE=3D2 FACE=3D"Arial">The requirement text is now alligned =
with the text shown on the slides during the SFO SIP WG =
presentation.</FONT>
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<P><FONT SIZE=3D2 FACE=3D"Arial">There are no other changes in the =
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<P><FONT SIZE=3D2 FACE=3D"Arial">Regards,</FONT>
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<P><FONT SIZE=3D2 FACE=3D"Arial">Christer</FONT>
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Hadriel,
=20
My company operates a SIP service with B2BUAs. We are very interested in
the progress of this draftt because without it trubleshooting is in many
cases not possible. Every SIP service provider which uses B2BUAs runs
sooner or later into this problem. From my point of view, the current
version is quite mature and coud become soon a WG item.=20
=20
Laura Liess
Deutsche Telekom
=20
 =20

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<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
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<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D288420717-26032009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D288420717-26032009>My company operates a SIP service with =
B2BUAs.&nbsp;We=20
are very interested in the progress of this draftt because without it=20
trubleshooting is in many cases not possible. Every SIP service provider =
which=20
uses B2BUAs runs sooner or later into this problem. From&nbsp;my point =
of view,=20
the current version&nbsp;is quite mature&nbsp;and coud become soon a WG=20
item.&nbsp;</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D288420717-26032009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D288420717-26032009>Laura Liess</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D288420717-26032009>Deutsche Telekom</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D288420717-26032009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
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Thread-Topic: [Sip] draft-kaplan-sip-session-id-02
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References: <49C507BB.9010804@cisco.com><1237955188.22543.30.camel@victoria-pingtel-com.us.nortel.com><E6C2E8958BA59A4FB960963D475F7AC314FD69308D@mail><1238014787.22565.12.camel@victoria-pingtel-com.us.nortel.com><4ff4e7370903251440l6e57f4ecp99a4f7bafcb0c873@mail.gmail.com><E6C2E8958BA59A4FB960963D475F7AC314FEBFE9E5@mail> <40FB0FFB97588246A1BEFB05759DC8A002CA6CBD@S4DE9JSAANI.ost.t-com.de>
From: "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>
To: <L.Liess@telekom.de>, <HKaplan@acmepacket.com>
Cc: sip@ietf.org
Subject: Re: [Sip] draft-kaplan-sip-session-id-02
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We support this draft as well

=20

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
L.Liess@telekom.de
Sent: Thursday, March 26, 2009 1:53 PM
To: HKaplan@acmepacket.com
Cc: sip@ietf.org
Subject: Re: [Sip] draft-kaplan-sip-session-id-02

=20

Hadriel,

=20

My company operates a SIP service with B2BUAs. We are very interested in
the progress of this draftt because without it trubleshooting is in many
cases not possible. Every SIP service provider which uses B2BUAs runs
sooner or later into this problem. From my point of view, the current
version is quite mature and coud become soon a WG item.=20

=20

Laura Liess

Deutsche Telekom

=20

 =20


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<p class=3DMsoNormal><span =
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color:#1F497D'>We support this draft as well<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

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0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
sip-bounces@ietf.org
[mailto:sip-bounces@ietf.org] <b>On Behalf Of </b>L.Liess@telekom.de<br>
<b>Sent:</b> Thursday, March 26, 2009 1:53 PM<br>
<b>To:</b> HKaplan@acmepacket.com<br>
<b>Cc:</b> sip@ietf.org<br>
<b>Subject:</b> Re: [Sip] =
draft-kaplan-sip-session-id-02<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Hadriel,</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>My company operates a SIP service with B2BUAs.&nbsp;We are =
very
interested in the progress of this draftt because without it =
trubleshooting is
in many cases not possible. Every SIP service provider which uses B2BUAs =
runs
sooner or later into this problem. From&nbsp;my point of view, the =
current
version&nbsp;is quite mature&nbsp;and coud become soon a WG =
item.&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Laura Liess</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Deutsche Telekom</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>&nbsp;&nbsp;</span><o:p></o:p></p>

</div>

</body>

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From vvenkatar@gmail.com  Thu Mar 26 14:31:55 2009
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References: <49C507BB.9010804@cisco.com> <1237955188.22543.30.camel@victoria-pingtel-com.us.nortel.com> <E6C2E8958BA59A4FB960963D475F7AC314FD69308D@mail> <1238014787.22565.12.camel@victoria-pingtel-com.us.nortel.com> <4ff4e7370903251440l6e57f4ecp99a4f7bafcb0c873@mail.gmail.com> <28B7C3AA2A7ABA4A841F11217ABE78D6757BE70C@FRMRSSXCHMBSB3.dc-m.alcatel-lucent.com>
Date: Thu, 26 Mar 2009 14:32:46 -0700
Message-ID: <4ff4e7370903261432o24afc450xd20bfabf2e0d94b1@mail.gmail.com>
From: Venkatesh <vvenkatar@gmail.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
Content-Type: multipart/alternative; boundary=001636765bbd7126ec04660c5b64
Cc: IETF SIP List <sip@ietf.org>
Subject: Re: [Sip] comments on draft-kaplan-sip-session-id-01
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Thought per the 3GPP spec, this header got added at the first ingress point
into the network and removed at the last egress point from the network? So,
essentially you can trace the call using this header within your network?
Venkatesh

On Wed, Mar 25, 2009 at 3:32 PM, DRAGE, Keith (Keith) <
drage@alcatel-lucent.com> wrote:

>  This only works if essentially your insertion and deletion requirements
> of the header are the same as those required by your accounting system.
>
> Keith
>
>  ------------------------------
> *From:* sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] *On Behalf Of *
> Venkatesh
> *Sent:* Wednesday, March 25, 2009 9:41 PM
> *To:* Dale Worley
> *Cc:* IETF SIP List
> *Subject:* Re: [Sip] comments on draft-kaplan-sip-session-id-01
>
> A crazy thought occurred to me while I was reading this draft. If the
> intent is for debugging and such, I am wondering if there is any value in
> considering the re-use of the P-Charging-Vector header for this purpose?
> After all, this is supposed to be unique, required to be carried across
> Proxies, B2BUA's if one is present, reflected in CDR's if one is generated
> by an entity?? Agree that this header had specific intents and cannot span
> across "trust" boundaries, but I am not sure if a SP cares when the request
> 'leaves' their trust boundary? It seems to satisfy the requirement for
> debugging? Flame away :-).
> Venkatesh
>
> On Wed, Mar 25, 2009 at 1:59 PM, Dale Worley <dworley@nortel.com> wrote:
>
>> On Wed, 2009-03-25 at 03:10 -0400, Hadriel Kaplan wrote:
>> > > If I may be immodest, the References header could a solution:  State
>> > > that two dialogs are related to each other by (effectively) naming
>> both
>> > > call-id's together, rather than by assigning them a common identifier
>> > > when they are created.
>> >
>> > Yeah I think it adds value to have the References header.  I just
>> > don't think it should be referencing Call-ID's, or else we'll have to
>> > replace that too.  It should be referencing Session-ID's or
>> > secure-call-id's.  :)
>>
>> Certainly in security-sensitive situations, referencing Call-IDs, or at
>> least, ones created by UAs, would be undesirable.  But down in section
>> 5.2 it describes how to have the B2BUA generate "linking" identifiers:
>>
>>   A possible solution to this problem is for the B2BUA to create a
>>   "phantom" Call-Id that is suitably random, and use it in References
>>   headers sent in both the initial request and response.  By the
>>   transitivity property[Section 2], the dialogs on both sides of the
>>   B2BUA are declared to be related, even if the referenced dialog
>>   contains no messages.
>>
>>   For two chained B2BUAs with no forking, this would give a message
>>   flow like this:
>>
>>       UA 1            B2BUA 1         B2BUA 2         UA 2
>>
>>            --->
>>            INVITE 123@aa.example.com
>>            Call-Id: qwerty@aa
>>
>>                            --->
>>                            INVITE 123@transit.example.com
>>                            Call-Id: asdfgh@transit
>>                            References: QAZWSXEDCRFV;rel=serial
>>
>>                                            --->
>>                                            INVITE 123@bb.example.com
>>                                            Call-Id: zxcvbn@bb
>>                                            References: QAZWSXEDCRFV
>>                                                 ;rel=serial
>>
>>                                            <---
>>                                            SIP/2.0 200 OK
>>                                            Call-Id: zxcvbn@bb
>>
>>                            <---
>>                            SIP/2.0 200 OK
>>                            Call-Id: asdfgh@transit
>>                            References: QAZWSXEDCRFV;rel=serial
>>
>>            <---
>>            SIP/2.0 200 OK
>>            Call-Id: qwerty@aa
>>            References: QAZWSXEDCRFV;rel=serial
>>
>> Dale
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
>
>

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Thought per the 3GPP spec, this header got added at the first ingress point=
 into the network and removed at the last egress point from the network? So=
, essentially you can trace the call using this header within your network?=
<div>
<br></div><div>Venkatesh<br><br><div class=3D"gmail_quote">On Wed, Mar 25, =
2009 at 3:32 PM, DRAGE, Keith (Keith) <span dir=3D"ltr">&lt;<a href=3D"mail=
to:drage@alcatel-lucent.com">drage@alcatel-lucent.com</a>&gt;</span> wrote:=
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">



<div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
" size=3D"2">This only works if essentially your insertion and deletion=20
requirements of the header are the same as those required by your accountin=
g=20
system.</font></span></div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
" size=3D"2"></font></span>=A0</div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
" size=3D"2">Keith</font></span></div><br>
<blockquote dir=3D"ltr" style=3D"padding-left:5px;margin-left:5px;border-le=
ft:#0000ff 2px solid;margin-right:0px">
  <div lang=3D"en-us" dir=3D"ltr" align=3D"left">
  <hr>
  <font face=3D"Tahoma" size=3D"2"><b>From:</b> <a href=3D"mailto:sip-bounc=
es@ietf.org" target=3D"_blank">sip-bounces@ietf.org</a>=20
  [mailto:<a href=3D"mailto:sip-bounces@ietf.org" target=3D"_blank">sip-bou=
nces@ietf.org</a>] <b>On Behalf Of </b>Venkatesh<br><b>Sent:</b>=20
  Wednesday, March 25, 2009 9:41 PM<br><b>To:</b> Dale Worley<br><b>Cc:</b>=
 IETF=20
  SIP List<br><b>Subject:</b> Re: [Sip] comments on=20
  draft-kaplan-sip-session-id-01<br></font><br></div><div><div></div><div c=
lass=3D"h5">
  <div></div>A crazy thought occurred to me while I was reading this draft.=
 If=20
  the intent is for debugging and such, I am wondering if there is any valu=
e in=20
  considering the re-use of the P-Charging-Vector header for this purpose? =
After=20
  all, this is supposed to be unique, required to be carried across Proxies=
,=20
  B2BUA&#39;s if one is present, reflected in CDR&#39;s if one is generated=
 by an=20
  entity?? Agree that this header had specific intents and cannot span acro=
ss=20
  &quot;trust&quot; boundaries, but I am not sure if a SP cares when the re=
quest &#39;leaves&#39;=20
  their trust boundary? It seems to satisfy the requirement for debugging? =
Flame=20
  away :-).
  <div><br></div>
  <div>Venkatesh<br><br>
  <div class=3D"gmail_quote">On Wed, Mar 25, 2009 at 1:59 PM, Dale Worley <=
span dir=3D"ltr">&lt;<a href=3D"mailto:dworley@nortel.com" target=3D"_blank=
">dworley@nortel.com</a>&gt;</span> wrote:<br>
  <blockquote class=3D"gmail_quote" style=3D"padding-left:1ex;margin:0px 0p=
x 0px 0.8ex;border-left:#ccc 1px solid">On=20
    Wed, 2009-03-25 at 03:10 -0400, Hadriel Kaplan wrote:<br>&gt; &gt; If I=
 may=20
    be immodest, the References header could a solution: =A0State<br>&gt;=
=20
    &gt; that two dialogs are related to each other by (effectively) naming=
=20
    both<br>&gt; &gt; call-id&#39;s together, rather than by assigning them=
 a common=20
    identifier<br>&gt; &gt; when they are created.<br>&gt;<br>&gt; Yeah I t=
hink=20
    it adds value to have the References header. =A0I just<br>&gt; don&#39;=
t=20
    think it should be referencing Call-ID&#39;s, or else we&#39;ll have to=
<br>&gt;=20
    replace that too. =A0It should be referencing Session-ID&#39;s or<br>&g=
t;=20
    secure-call-id&#39;s. =A0:)<br><br>Certainly in security-sensitive=20
    situations, referencing Call-IDs, or at<br>least, ones created by UAs, =
would=20
    be undesirable. =A0But down in section<br>5.2 it describes how to have=
=20
    the B2BUA generate &quot;linking&quot; identifiers:<br><br>=A0 A possib=
le solution=20
    to this problem is for the B2BUA to create a<br>=A0 &quot;phantom&quot;=
 Call-Id=20
    that is suitably random, and use it in References<br>=A0 headers sent i=
n=20
    both the initial request and response. =A0By the<br>=A0 transitivity=20
    property[Section 2], the dialogs on both sides of the<br>=A0 B2BUA are=
=20
    declared to be related, even if the referenced dialog<br>=A0 contains n=
o=20
    messages.<br><br>=A0 For two chained B2BUAs with no forking, this would=
=20
    give a message<br>=A0 flow like this:<br><br>=A0 =A0 =A0 UA 1=20
    =A0 =A0 =A0 =A0 =A0 =A0B2BUA 1 =A0 =A0 =A0 =A0=20
    B2BUA 2 =A0 =A0 =A0 =A0 UA 2<br><br>=A0 =A0 =A0 =A0=20
    =A0 =A0---&gt;<br>=A0 =A0 =A0 =A0 =A0 =A0INVITE <a href=3D"mailto:123@a=
a.example.com" target=3D"_blank">123@aa.example.com</a><br>=A0 =A0=20
    =A0 =A0 =A0 =A0Call-Id: qwerty@aa<br><br>=A0 =A0 =A0=20
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
    =A0---&gt;<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
    =A0 =A0 =A0 =A0 =A0 =A0INVITE <a href=3D"mailto:123@transit.example.com=
" target=3D"_blank">123@transit.example.com</a><br>=A0=20
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
    =A0 =A0Call-Id: asdfgh@transit<br>=A0 =A0 =A0 =A0 =A0=20
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0References:=20
    QAZWSXEDCRFV;rel=3Dserial<br><br>=A0 =A0 =A0 =A0 =A0 =A0=20
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
    =A0 =A0 =A0 =A0 =A0---&gt;<br>=A0 =A0 =A0 =A0=20
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
    =A0 =A0 =A0 =A0 =A0 =A0 =A0INVITE <a href=3D"mailto:123@bb.example.com"=
 target=3D"_blank">123@bb.example.com</a><br>=A0 =A0=20
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Call-Id:=20
    zxcvbn@bb<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
    =A0 =A0References: QAZWSXEDCRFV<br>=A0 =A0 =A0 =A0 =A0=20
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
    ;rel=3Dserial<br><br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
    =A0 =A0 =A0&lt;---<br>=A0 =A0 =A0 =A0 =A0 =A0=20
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
    =A0 =A0 =A0 =A0 =A0SIP/2.0 200 OK<br>=A0 =A0 =A0=20
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Call-Id:=20
    zxcvbn@bb<br><br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
    =A0 =A0 =A0 =A0 =A0 =A0&lt;---<br>=A0 =A0 =A0=20
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
    =A0SIP/2.0 200 OK<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0=20
    =A0 =A0 =A0 =A0 =A0 =A0 =A0Call-Id:=20
    asdfgh@transit<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=20
    =A0 =A0 =A0 =A0 =A0 =A0References:=20
    QAZWSXEDCRFV;rel=3Dserial<br><br>=A0 =A0 =A0 =A0 =A0=20
    =A0&lt;---<br>=A0 =A0 =A0 =A0 =A0 =A0SIP/2.0 200=20
    OK<br>=A0 =A0 =A0 =A0 =A0 =A0Call-Id: qwerty@aa<br>=A0=20
    =A0 =A0 =A0 =A0 =A0References:=20
    QAZWSXEDCRFV;rel=3Dserial<br><br>Dale<br><br><br>______________________=
_________________________<br>Sip=20
    mailing list =A0<a href=3D"https://www.ietf.org/mailman/listinfo/sip" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>This list=
 is=20
    for NEW development of the core SIP Protocol<br>Use <a href=3D"mailto:s=
ip-implementors@cs.columbia.edu" target=3D"_blank">sip-implementors@cs.colu=
mbia.edu</a>=20
    for questions on current sip<br>Use <a href=3D"mailto:sipping@ietf.org"=
 target=3D"_blank">sipping@ietf.org</a> for new developments on=20
    the application of=20
sip<br></blockquote></div><br></div></div></div></blockquote></div>
</blockquote></div><br></div>

--001636765bbd7126ec04660c5b64--

From jiri@iptel.org  Thu Mar 26 15:34:04 2009
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Date: Thu, 26 Mar 2009 23:34:28 +0100
From: Jiri Kuthan <jiri@iptel.org>
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Cc: IETF SIP List <sip@ietf.org>
Subject: Re: [Sip] The problem with draft-kaplan-sip-secure-call-id-00
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First of all I think it is definitely worth trying.

Not being able to correlate dialogs through B2BUA is a big
operational problem. I'm not sure that this will get really
adopted (who knows what today's obfuscators choose to obfuscate
tomorrow...). However in hope for solving this problem, presence of
incentives for adoption and absence  of any known harm it is
definitely worth doing.

Another idea how to accommodate receivers that are silly not
to be liberal is using a fake IP address (popular 0.0.0.0).
While not aesthetic at all, it could accommodate such receivers
(under assumption they assign no syntactical meaning to it)
and signal to B2BUA there is no disclosure of IP addresses.

OTOH it could be time too to abandon 2543 legacy. Supporting
strict routing and other ancient stuff like IP in callid does
not seem worth the trouble to me.

-jiri

Hadriel Kaplan wrote:
> Ooh, good point.  Fair enough, I'll change that.
> 
> -hadriel
> 
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
>> Jonathan Rosenberg
>> Sent: Wednesday, March 25, 2009 4:51 PM
>> To: IETF SIP List
>> Subject: [Sip] The problem with draft-kaplan-sip-secure-call-id-00
>>
>> I didnt get a chance to say this at the mike.
>>
>> You *cannot* change the BNF for the call-id. If you did, you might end
>> up with the following problem:
>>
>> A new UA gets built, compliant to this update. It is one of these
>> 'aggressive parsing' UAs. It rejects any message which is not compliant
>> to the BNF. An older, normal RFC-3261 compliant UA sends a call-id which
>> includes the @-sign.
>>
>> According to the BNF of this updated 3261, and based on the strictness
>> of this aggressive-parsing UAs, it rejects that request as non-compliant.
>>
>> As such, you need to keep the BNF, but change the normative language
>> such that an endpoint MUST NOT include the [@ host] but MUST be prepared
>> to receive it.
>>
>> -Jonathan R.
>>
>> --
>> Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
>> Cisco Fellow                                   Iselin, NJ 08830
>> Cisco, Voice Technology Group
>> jdrosen@cisco.com
>> http://www.jdrosen.net                         PHONE: (408) 902-3084
>> http://www.cisco.com
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

From mary.barnes@nortel.com  Thu Mar 26 16:16:08 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>, <sip@ietf.org>
Subject: Re: [Sip] New version of draft-holmberg-sip-keep
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Hi folks,

Christer has updated the document based on the action item from =
Wednesday's SIP session (chaired by SIPPING chairs and modeled as the =
proposed DISPATCH WG).  If folks could please review the document and =
raise any concerns on the document now, that would be appreciated. We =
ask specifically for an answer to the question:  Do you support the =
document being adopted as a WG document in RAI (WG tbd)?  A simple "yes" =
is fine, however, if you don't think we should adopt this document, =
could you please provide the reason.

For your convenience, here's the link to the document:
http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt
And, the diff:
http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-holm=
berg-sip-keep-03.txt&url2=3Dhttp://tools.ietf.org/id/draft-holmberg-sip-k=
eep-04.txt
tinyurl: http://tinyurl.com/dfyvxz

Regards,
Mary
SIPPING WG co-chair


-----Original Message-----
From: sip-bounces@ietf.org on behalf of Christer Holmberg
Sent: Thu 3/26/2009 12:41 PM
To: sip@ietf.org
Subject: [Sip] New version of draft-holmberg-sip-keep
=20

Hi,

I've submitted a new version (-04) of the keep draft.

The requirement text is now alligned with the text shown on the slides
during the SFO SIP WG presentation.

There are no other changes in the document from the previous version.

Regards,

Christer


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<P><FONT SIZE=3D2>Hi folks,<BR>
<BR>
Christer has updated the document based on the action item from =
Wednesday's SIP session (chaired by SIPPING chairs and modeled as the =
proposed DISPATCH WG).&nbsp; If folks could please review the document =
and raise any concerns on the document now, that would be appreciated. =
We ask specifically for an answer to the question:&nbsp; Do you support =
the document being adopted as a WG document in RAI (WG tbd)?&nbsp; A =
simple &quot;yes&quot; is fine, however, if you don't think we should =
adopt this document, could you please provide the reason.<BR>
<BR>
For your convenience, here's the link to the document:<BR>
<A =
HREF=3D"http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt">http://t=
ools.ietf.org/id/draft-holmberg-sip-keep-04.txt</A><BR>
And, the diff:<BR>
<A =
HREF=3D"http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/dr=
aft-holmberg-sip-keep-03.txt&url2=3Dhttp://tools.ietf.org/id/draft-holmbe=
rg-sip-keep-04.txt">http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ie=
tf.org/id/draft-holmberg-sip-keep-03.txt&url2=3Dhttp://tools.ietf.org/id/=
draft-holmberg-sip-keep-04.txt</A><BR>
tinyurl: <A =
HREF=3D"http://tinyurl.com/dfyvxz">http://tinyurl.com/dfyvxz</A><BR>
<BR>
Regards,<BR>
Mary<BR>
SIPPING WG co-chair<BR>
<BR>
<BR>
-----Original Message-----<BR>
From: sip-bounces@ietf.org on behalf of Christer Holmberg<BR>
Sent: Thu 3/26/2009 12:41 PM<BR>
To: sip@ietf.org<BR>
Subject: [Sip] New version of draft-holmberg-sip-keep<BR>
<BR>
<BR>
Hi,<BR>
<BR>
I've submitted a new version (-04) of the keep draft.<BR>
<BR>
The requirement text is now alligned with the text shown on the =
slides<BR>
during the SFO SIP WG presentation.<BR>
<BR>
There are no other changes in the document from the previous =
version.<BR>
<BR>
Regards,<BR>
<BR>
Christer<BR>
<BR>
</FONT>
</P>

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From: "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>
To: "Mary Barnes" <mary.barnes@nortel.com>, "Christer Holmberg" <christer.holmberg@ericsson.com>, <sip@ietf.org>
Subject: Re: [Sip] New version of draft-holmberg-sip-keep
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Yes, we support this draft as a WG document.

=20

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Mary Barnes
Sent: Thursday, March 26, 2009 7:17 PM
To: Christer Holmberg; sip@ietf.org
Subject: Re: [Sip] New version of draft-holmberg-sip-keep

=20

Hi folks,

Christer has updated the document based on the action item from
Wednesday's SIP session (chaired by SIPPING chairs and modeled as the
proposed DISPATCH WG).  If folks could please review the document and
raise any concerns on the document now, that would be appreciated. We
ask specifically for an answer to the question:  Do you support the
document being adopted as a WG document in RAI (WG tbd)?  A simple "yes"
is fine, however, if you don't think we should adopt this document,
could you please provide the reason.

For your convenience, here's the link to the document:
http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt
And, the diff:
http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-holm=
b
erg-sip-keep-03.txt&url2=3Dhttp://tools.ietf.org/id/draft-holmberg-sip-ke=
e
p-04.txt
<http://tools.ietf.org/rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-holm=
b
erg-sip-keep-03.txt&url2=3Dhttp://tools.ietf.org/id/draft-holmberg-sip-ke=
e
p-04.txt>=20
tinyurl: http://tinyurl.com/dfyvxz

Regards,
Mary
SIPPING WG co-chair


-----Original Message-----
From: sip-bounces@ietf.org on behalf of Christer Holmberg
Sent: Thu 3/26/2009 12:41 PM
To: sip@ietf.org
Subject: [Sip] New version of draft-holmberg-sip-keep


Hi,

I've submitted a new version (-04) of the keep draft.

The requirement text is now alligned with the text shown on the slides
during the SFO SIP WG presentation.

There are no other changes in the document from the previous version.

Regards,

Christer


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<title>RE: [Sip] New version of draft-holmberg-sip-keep</title>
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<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Yes, we support this draft as a WG =
document.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
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0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <b>On Behalf Of =
</b>Mary
Barnes<br>
<b>Sent:</b> Thursday, March 26, 2009 7:17 PM<br>
<b>To:</b> Christer Holmberg; sip@ietf.org<br>
<b>Subject:</b> Re: [Sip] New version of =
draft-holmberg-sip-keep<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p style=3D'margin-bottom:12.0pt'><span style=3D'font-size:10.0pt'>Hi =
folks,<br>
<br>
Christer has updated the document based on the action item from =
Wednesday's SIP
session (chaired by SIPPING chairs and modeled as the proposed DISPATCH
WG).&nbsp; If folks could please review the document and raise any =
concerns on
the document now, that would be appreciated. We ask specifically for an =
answer
to the question:&nbsp; Do you support the document being adopted as a WG =
document
in RAI (WG tbd)?&nbsp; A simple &quot;yes&quot; is fine, however, if you =
don't
think we should adopt this document, could you please provide the =
reason.<br>
<br>
For your convenience, here's the link to the document:<br>
<a =
href=3D"http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt">http://t=
ools.ietf.org/id/draft-holmberg-sip-keep-04.txt</a><br>
And, the diff:<br>
<a
href=3D"http://tools.ietf.org/rfcdiff?url1=3Dhttp://tools.ietf.org/id/dra=
ft-holmberg-sip-keep-03.txt&amp;url2=3Dhttp://tools.ietf.org/id/draft-hol=
mberg-sip-keep-04.txt">http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools=
.ietf.org/id/draft-holmberg-sip-keep-03.txt&amp;url2=3Dhttp://tools.ietf.=
org/id/draft-holmberg-sip-keep-04.txt</a><br>
tinyurl: <a =
href=3D"http://tinyurl.com/dfyvxz">http://tinyurl.com/dfyvxz</a><br>
<br>
Regards,<br>
Mary<br>
SIPPING WG co-chair<br>
<br>
<br>
-----Original Message-----<br>
From: sip-bounces@ietf.org on behalf of Christer Holmberg<br>
Sent: Thu 3/26/2009 12:41 PM<br>
To: sip@ietf.org<br>
Subject: [Sip] New version of draft-holmberg-sip-keep<br>
<br>
<br>
Hi,<br>
<br>
I've submitted a new version (-04) of the keep draft.<br>
<br>
The requirement text is now alligned with the text shown on the =
slides<br>
during the SFO SIP WG presentation.<br>
<br>
There are no other changes in the document from the previous =
version.<br>
<br>
Regards,<br>
<br>
Christer</span><o:p></o:p></p>

</div>

</body>

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To: Mary Barnes <mary.barnes@nortel.com>, Christer Holmberg <christer.holmberg@ericsson.com>, "sip@ietf.org" <sip@ietf.org>
Date: Thu, 26 Mar 2009 20:10:25 -0400
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"yes"


________________________________
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Mary =
Barnes
Sent: Thursday, March 26, 2009 7:17 PM
To: Christer Holmberg; sip@ietf.org
Subject: Re: [Sip] New version of draft-holmberg-sip-keep


Hi folks,

Christer has updated the document based on the action item from Wednesday's=
 SIP session (chaired by SIPPING chairs and modeled as the proposed DISPATC=
H WG).  If folks could please review the document and raise any concerns on=
 the document now, that would be appreciated. We ask specifically for an an=
swer to the question:  Do you support the document being adopted as a WG do=
cument in RAI (WG tbd)?  A simple "yes" is fine, however, if you don't thin=
k we should adopt this document, could you please provide the reason.

For your convenience, here's the link to the document:
http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt
And, the diff:
http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-holmbe=
rg-sip-keep-03.txt&url2=3Dhttp://tools.ietf.org/id/draft-holmberg-sip-keep-=
04.txt<http://tools.ietf.org/rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-=
holmberg-sip-keep-03.txt&url2=3Dhttp://tools.ietf.org/id/draft-holmberg-sip=
-keep-04.txt>
tinyurl: http://tinyurl.com/dfyvxz

Regards,
Mary
SIPPING WG co-chair


-----Original Message-----
From: sip-bounces@ietf.org on behalf of Christer Holmberg
Sent: Thu 3/26/2009 12:41 PM
To: sip@ietf.org
Subject: [Sip] New version of draft-holmberg-sip-keep


Hi,

I've submitted a new version (-04) of the keep draft.

The requirement text is now alligned with the text shown on the slides
during the SFO SIP WG presentation.

There are no other changes in the document from the previous version.

Regards,

Christer

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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
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10.0pt;font-family:Arial;color:navy'>&#8220;yes&#8221;<o:p></o:p></span></f=
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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
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=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>
sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <b><span style=3D'font-w=
eight:
bold'>On Behalf Of </span></b>Mary Barnes<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, March 26, 20=
09
7:17 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Christer Holmberg;
sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [Sip] New versi=
on of
draft-holmberg-sip-keep</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p style=3D'margin-bottom:12.0pt'><font size=3D2 face=3D"Times New Roman"><=
span
style=3D'font-size:10.0pt'>Hi folks,<br>
<br>
Christer has updated the document based on the action item from Wednesday's=
 SIP
session (chaired by SIPPING chairs and modeled as the proposed DISPATCH
WG).&nbsp; If folks could please review the document and raise any concerns=
 on
the document now, that would be appreciated. We ask specifically for an ans=
wer
to the question:&nbsp; Do you support the document being adopted as a WG
document in RAI (WG tbd)?&nbsp; A simple &quot;yes&quot; is fine, however, =
if
you don't think we should adopt this document, could you please provide the
reason.<br>
<br>
For your convenience, here's the link to the document:<br>
<a href=3D"http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt">http://=
tools.ietf.org/id/draft-holmberg-sip-keep-04.txt</a><br>
And, the diff:<br>
<a
href=3D"http://tools.ietf.org/rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft=
-holmberg-sip-keep-03.txt&amp;url2=3Dhttp://tools.ietf.org/id/draft-holmber=
g-sip-keep-04.txt">http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.=
org/id/draft-holmberg-sip-keep-03.txt&amp;url2=3Dhttp://tools.ietf.org/id/d=
raft-holmberg-sip-keep-04.txt</a><br>
tinyurl: <a href=3D"http://tinyurl.com/dfyvxz">http://tinyurl.com/dfyvxz</a=
><br>
<br>
Regards,<br>
Mary<br>
SIPPING WG co-chair<br>
<br>
<br>
-----Original Message-----<br>
From: sip-bounces@ietf.org on behalf of Christer Holmberg<br>
Sent: Thu 3/26/2009 12:41 PM<br>
To: sip@ietf.org<br>
Subject: [Sip] New version of draft-holmberg-sip-keep<br>
<br>
<br>
Hi,<br>
<br>
I've submitted a new version (-04) of the keep draft.<br>
<br>
The requirement text is now alligned with the text shown on the slides<br>
during the SFO SIP WG presentation.<br>
<br>
There are no other changes in the document from the previous version.<br>
<br>
Regards,<br>
<br>
Christer</span></font><o:p></o:p></p>

</div>

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From theo@crazygreek.co.uk  Thu Mar 26 17:35:33 2009
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On Thu, Mar 26, 2009 at 11:16 PM, Mary Barnes <mary.barnes@nortel.com> wrote:

> Do you support the document being adopted as a WG document in RAI (WG tbd)?

yes

 ~ Theo

Sent from: San Francisco California United States.

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Cc: sip@ietf.org, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] New version of draft-holmberg-sip-keep
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"yes"

Mary Barnes wrote:
>
> Hi folks,
>
> Christer has updated the document based on the action item from 
> Wednesday's SIP session (chaired by SIPPING chairs and modeled as the 
> proposed DISPATCH WG).  If folks could please review the document and 
> raise any concerns on the document now, that would be appreciated. We 
> ask specifically for an answer to the question:  Do you support the 
> document being adopted as a WG document in RAI (WG tbd)?  A simple 
> "yes" is fine, however, if you don't think we should adopt this 
> document, could you please provide the reason.
>
> For your convenience, here's the link to the document:
> http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt
> And, the diff:
> http://tools.ietf.org//rfcdiff?url1=http://tools.ietf.org/id/draft-holmberg-sip-keep-03.txt&url2=http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt 
> <http://tools.ietf.org//rfcdiff?url1=http://tools.ietf.org/id/draft-holmberg-sip-keep-03.txt&url2=http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt>
> tinyurl: http://tinyurl.com/dfyvxz
>
> Regards,
> Mary
> SIPPING WG co-chair
>
>
> -----Original Message-----
> From: sip-bounces@ietf.org on behalf of Christer Holmberg
> Sent: Thu 3/26/2009 12:41 PM
> To: sip@ietf.org
> Subject: [Sip] New version of draft-holmberg-sip-keep
>
>
> Hi,
>
> I've submitted a new version (-04) of the keep draft.
>
> The requirement text is now alligned with the text shown on the slides
> during the SFO SIP WG presentation.
>
> There are no other changes in the document from the previous version.
>
> Regards,
>
> Christer
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

From BPenfield@acmepacket.com  Thu Mar 26 17:57:11 2009
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To: Mary Barnes <mary.barnes@nortel.com>, Christer Holmberg <christer.holmberg@ericsson.com>, "sip@ietf.org" <sip@ietf.org>
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>Do you support the document being adopted as a WG document in RAI (WG tbd)=
? =20

YES!

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Mary =
Barnes
Sent: Thursday, March 26, 2009 7:17 PM
To: Christer Holmberg; sip@ietf.org
Subject: Re: [Sip] New version of draft-holmberg-sip-keep



Hi folks,

Christer has updated the document based on the action item from Wednesday's=
 SIP session (chaired by SIPPING chairs and modeled as the proposed DISPATC=
H WG).  If folks could please review the document and raise any concerns on=
 the document now, that would be appreciated. We ask specifically for an an=
swer to the question:  Do you support the document being adopted as a WG do=
cument in RAI (WG tbd)?  A simple "yes" is fine, however, if you don't thin=
k we should adopt this document, could you please provide the reason.

For your convenience, here's the link to the document:
http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt
And, the diff:
http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-holmbe=
rg-sip-keep-03.txt&url2=3Dhttp://tools.ietf.org/id/draft-holmberg-sip-keep-=
04.txt
tinyurl: http://tinyurl.com/dfyvxz

Regards,
Mary
SIPPING WG co-chair


-----Original Message-----
From: sip-bounces@ietf.org on behalf of Christer Holmberg
Sent: Thu 3/26/2009 12:41 PM
To: sip@ietf.org
Subject: [Sip] New version of draft-holmberg-sip-keep


Hi,

I've submitted a new version (-04) of the keep draft.

The requirement text is now alligned with the text shown on the slides
during the SFO SIP WG presentation.

There are no other changes in the document from the previous version.

Regards,

Christer




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From: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
To: Theo Zourzouvillys <theo@crazygreek.co.uk>, Mary Barnes <mary.barnes@nortel.com>
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+1

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Theo Zourzouvillys
> Sent: Thursday, March 26, 2009 5:34 PM
> To: Mary Barnes
> Cc: sip@ietf.org; Christer Holmberg
> Subject: Re: [Sip] New version of draft-holmberg-sip-keep
>=20
> On Thu, Mar 26, 2009 at 11:16 PM, Mary Barnes <mary.barnes@nortel.com>
> wrote:
>=20
> > Do you support the document being adopted as a WG document in RAI (WG
> tbd)?
>=20
> yes
>=20
>  ~ Theo
>=20
> Sent from: San Francisco California United States.
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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From: "Bharrat, Shaun" <SBharrat@sonusnet.com>
To: "Mary Barnes" <mary.barnes@nortel.com>, "Christer Holmberg" <christer.holmberg@ericsson.com>, <sip@ietf.org>
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yes

-----Original Message-----
From: sip-bounces@ietf.org on behalf of Mary Barnes
Sent: Thu 3/26/2009 7:16 PM
To: Christer Holmberg; sip@ietf.org
Subject: Re: [Sip] New version of draft-holmberg-sip-keep
=20
Hi folks,

Christer has updated the document based on the action item from =
Wednesday's SIP session (chaired by SIPPING chairs and modeled as the =
proposed DISPATCH WG).  If folks could please review the document and =
raise any concerns on the document now, that would be appreciated. We =
ask specifically for an answer to the question:  Do you support the =
document being adopted as a WG document in RAI (WG tbd)?  A simple "yes" =
is fine, however, if you don't think we should adopt this document, =
could you please provide the reason.

For your convenience, here's the link to the document:
http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt
And, the diff:
http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-holm=
berg-sip-keep-03.txt&url2=3Dhttp://tools.ietf.org/id/draft-holmberg-sip-k=
eep-04.txt
tinyurl: http://tinyurl.com/dfyvxz

Regards,
Mary
SIPPING WG co-chair


-----Original Message-----
From: sip-bounces@ietf.org on behalf of Christer Holmberg
Sent: Thu 3/26/2009 12:41 PM
To: sip@ietf.org
Subject: [Sip] New version of draft-holmberg-sip-keep
=20

Hi,

I've submitted a new version (-04) of the keep draft.

The requirement text is now alligned with the text shown on the slides
during the SFO SIP WG presentation.

There are no other changes in the document from the previous version.

Regards,

Christer



From drage@alcatel-lucent.com  Thu Mar 26 21:52:13 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: Mary Barnes <mary.barnes@nortel.com>, Christer Holmberg <christer.holmberg@ericsson.com>, "sip@ietf.org" <sip@ietf.org>
Date: Fri, 27 Mar 2009 05:53:03 +0100
Thread-Topic: New version of draft-holmberg-sip-keep
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References: <CA9998CD4A020D418654FCDEF4E707DF0B168098@esealmw113.eemea.ericsson.se> <1ECE0EB50388174790F9694F77522CCF1BF5FDAD@zrc2hxm0.corp.nortel.com>
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Subject: Re: [Sip] New version of draft-holmberg-sip-keep
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--_000_28B7C3AA2A7ABA4A841F11217ABE78D6757BE91EFRMRSSXCHMBSB3d_
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As individual

Support

Keith

________________________________
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Mary =
Barnes
Sent: Thursday, March 26, 2009 11:17 PM
To: Christer Holmberg; sip@ietf.org
Subject: Re: [Sip] New version of draft-holmberg-sip-keep


Hi folks,

Christer has updated the document based on the action item from Wednesday's=
 SIP session (chaired by SIPPING chairs and modeled as the proposed DISPATC=
H WG).  If folks could please review the document and raise any concerns on=
 the document now, that would be appreciated. We ask specifically for an an=
swer to the question:  Do you support the document being adopted as a WG do=
cument in RAI (WG tbd)?  A simple "yes" is fine, however, if you don't thin=
k we should adopt this document, could you please provide the reason.

For your convenience, here's the link to the document:
http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt
And, the diff:
http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-holmbe=
rg-sip-keep-03.txt&url2=3Dhttp://tools.ietf.org/id/draft-holmberg-sip-keep-=
04.txt
tinyurl: http://tinyurl.com/dfyvxz

Regards,
Mary
SIPPING WG co-chair


-----Original Message-----
From: sip-bounces@ietf.org on behalf of Christer Holmberg
Sent: Thu 3/26/2009 12:41 PM
To: sip@ietf.org
Subject: [Sip] New version of draft-holmberg-sip-keep


Hi,

I've submitted a new version (-04) of the keep draft.

The requirement text is now alligned with the text shown on the slides
during the SFO SIP WG presentation.

There are no other changes in the document from the previous version.

Regards,

Christer


--_000_28B7C3AA2A7ABA4A841F11217ABE78D6757BE91EFRMRSSXCHMBSB3d_
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>RE: [Sip] New version of draft-holmberg-sip-keep</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Diso-8859-1"=
>
<META content=3D"MSHTML 6.00.2900.3492" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D062415204-27032009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>As individual</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D062415204-27032009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D062415204-27032009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Support</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D062415204-27032009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D062415204-27032009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Keith</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px soli=
d; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Mary Barnes<BR><B>Sent:=
</B>=20
  Thursday, March 26, 2009 11:17 PM<BR><B>To:</B> Christer Holmberg;=20
  sip@ietf.org<BR><B>Subject:</B> Re: [Sip] New version of=20
  draft-holmberg-sip-keep<BR></FONT><BR></DIV>
  <DIV></DIV><!-- Converted from text/plain format -->
  <P><FONT size=3D2>Hi folks,<BR><BR>Christer has updated the document base=
d on=20
  the action item from Wednesday's SIP session (chaired by SIPPING chairs a=
nd=20
  modeled as the proposed DISPATCH WG).&nbsp; If folks could please review =
the=20
  document and raise any concerns on the document now, that would be=20
  appreciated. We ask specifically for an answer to the question:&nbsp; Do =
you=20
  support the document being adopted as a WG document in RAI (WG tbd)?&nbsp=
; A=20
  simple "yes" is fine, however, if you don't think we should adopt this=20
  document, could you please provide the reason.<BR><BR>For your convenienc=
e,=20
  here's the link to the document:<BR><A=20
  href=3D"http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt">http://t=
ools.ietf.org/id/draft-holmberg-sip-keep-04.txt</A><BR>And,=20
  the diff:<BR><A=20
  href=3D"http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/dr=
aft-holmberg-sip-keep-03.txt&amp;url2=3Dhttp://tools.ietf.org/id/draft-holm=
berg-sip-keep-04.txt">http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ie=
tf.org/id/draft-holmberg-sip-keep-03.txt&amp;url2=3Dhttp://tools.ietf.org/i=
d/draft-holmberg-sip-keep-04.txt</A><BR>tinyurl:=20
  <A=20
  href=3D"http://tinyurl.com/dfyvxz">http://tinyurl.com/dfyvxz</A><BR><BR>R=
egards,<BR>Mary<BR>SIPPING=20
  WG co-chair<BR><BR><BR>-----Original Message-----<BR>From:=20
  sip-bounces@ietf.org on behalf of Christer Holmberg<BR>Sent: Thu 3/26/200=
9=20
  12:41 PM<BR>To: sip@ietf.org<BR>Subject: [Sip] New version of=20
  draft-holmberg-sip-keep<BR><BR><BR>Hi,<BR><BR>I've submitted a new versio=
n=20
  (-04) of the keep draft.<BR><BR>The requirement text is now alligned with=
 the=20
  text shown on the slides<BR>during the SFO SIP WG presentation.<BR><BR>Th=
ere=20
  are no other changes in the document from the previous=20
  version.<BR><BR>Regards,<BR><BR>Christer<BR><BR></FONT></P></BLOCKQUOTE><=
/BODY></HTML>

--_000_28B7C3AA2A7ABA4A841F11217ABE78D6757BE91EFRMRSSXCHMBSB3d_--

From R.Jesske@telekom.de  Thu Mar 26 21:52:27 2009
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From: <R.Jesske@telekom.de>
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X-OriginalArrivalTime: 27 Mar 2009 04:53:14.0330 (UTC) FILETIME=[EFCDB3A0:01C9AE97]
Subject: Re: [Sip] New version of draft-holmberg-sip-keep
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"Yes"
We too.
=20
Roland

________________________________

Von: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] Im Auftrag von =
DOLLY, MARTIN C, ATTLABS
Gesendet: Freitag, 27. M=E4rz 2009 00:36
An: Mary Barnes; Christer Holmberg; sip@ietf.org
Betreff: Re: [Sip] New version of draft-holmberg-sip-keep



Yes, we support this draft as a WG document.

=20

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
Mary Barnes
Sent: Thursday, March 26, 2009 7:17 PM
To: Christer Holmberg; sip@ietf.org
Subject: Re: [Sip] New version of draft-holmberg-sip-keep

=20

Hi folks,

Christer has updated the document based on the action item from =
Wednesday's SIP session (chaired by SIPPING chairs and modeled as the =
proposed DISPATCH WG).  If folks could please review the document and =
raise any concerns on the document now, that would be appreciated. We =
ask specifically for an answer to the question:  Do you support the =
document being adopted as a WG document in RAI (WG tbd)?  A simple "yes" =
is fine, however, if you don't think we should adopt this document, =
could you please provide the reason.

For your convenience, here's the link to the document:
http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt
And, the diff:
http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-holm=
berg-sip-keep-03.txt&url2=3Dhttp://tools.ietf.org/id/draft-holmberg-sip-k=
eep-04.txt =
<http://tools.ietf.org/rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-holm=
berg-sip-keep-03.txt&url2=3Dhttp://tools.ietf.org/id/draft-holmberg-sip-k=
eep-04.txt>=20
tinyurl: http://tinyurl.com/dfyvxz

Regards,
Mary
SIPPING WG co-chair


-----Original Message-----
From: sip-bounces@ietf.org on behalf of Christer Holmberg
Sent: Thu 3/26/2009 12:41 PM
To: sip@ietf.org
Subject: [Sip] New version of draft-holmberg-sip-keep


Hi,

I've submitted a new version (-04) of the keep draft.

The requirement text is now alligned with the text shown on the slides
during the SFO SIP WG presentation.

There are no other changes in the document from the previous version.

Regards,

Christer


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<DIV dir=3Dltr align=3Dleft><SPAN class=3D015485204-27032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>"Yes"</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D015485204-27032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>We too.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D015485204-27032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D015485204-27032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Roland</FONT></SPAN></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Dde dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>Von:</B> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>Im Auftrag von </B>DOLLY, MARTIN C,=20
ATTLABS<BR><B>Gesendet:</B> Freitag, 27. M&#xE4;rz 2009 =
00:36<BR><B>An:</B> Mary=20
Barnes; Christer Holmberg; sip@ietf.org<BR><B>Betreff:</B> Re: [Sip] New =
version=20
of draft-holmberg-sip-keep<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=3DSection1>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'">Yes,=20
we support this draft as a WG document.<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'"><o:p>&nbsp;</o:p></SPAN></P>
<DIV>
<DIV=20
style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: =
#b5c4df 1pt solid; PADDING-LEFT: 0in; PADDING-BOTTOM: 0in; BORDER-LEFT: =
medium none; PADDING-TOP: 3pt; BORDER-BOTTOM: medium none">
<P class=3DMsoNormal><B><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
'Tahoma','sans-serif'">From:</SPAN></B><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'">=20
sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <B>On Behalf Of =
</B>Mary=20
Barnes<BR><B>Sent:</B> Thursday, March 26, 2009 7:17 PM<BR><B>To:</B> =
Christer=20
Holmberg; sip@ietf.org<BR><B>Subject:</B> Re: [Sip] New version of=20
draft-holmberg-sip-keep<o:p></o:p></SPAN></P></DIV></DIV>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<P style=3D"MARGIN-BOTTOM: 12pt"><SPAN style=3D"FONT-SIZE: 10pt">Hi=20
folks,<BR><BR>Christer has updated the document based on the action item =
from=20
Wednesday's SIP session (chaired by SIPPING chairs and modeled as the =
proposed=20
DISPATCH WG).&nbsp; If folks could please review the document and raise =
any=20
concerns on the document now, that would be appreciated. We ask =
specifically for=20
an answer to the question:&nbsp; Do you support the document being =
adopted as a=20
WG document in RAI (WG tbd)?&nbsp; A simple "yes" is fine, however, if =
you don't=20
think we should adopt this document, could you please provide the=20
reason.<BR><BR>For your convenience, here's the link to the =
document:<BR><A=20
href=3D"http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt">http://t=
ools.ietf.org/id/draft-holmberg-sip-keep-04.txt</A><BR>And,=20
the diff:<BR><A=20
href=3D"http://tools.ietf.org/rfcdiff?url1=3Dhttp://tools.ietf.org/id/dra=
ft-holmberg-sip-keep-03.txt&amp;url2=3Dhttp://tools.ietf.org/id/draft-hol=
mberg-sip-keep-04.txt">http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools=
.ietf.org/id/draft-holmberg-sip-keep-03.txt&amp;url2=3Dhttp://tools.ietf.=
org/id/draft-holmberg-sip-keep-04.txt</A><BR>tinyurl:=20
<A=20
href=3D"http://tinyurl.com/dfyvxz">http://tinyurl.com/dfyvxz</A><BR><BR>R=
egards,<BR>Mary<BR>SIPPING=20
WG co-chair<BR><BR><BR>-----Original Message-----<BR>From: =
sip-bounces@ietf.org=20
on behalf of Christer Holmberg<BR>Sent: Thu 3/26/2009 12:41 PM<BR>To:=20
sip@ietf.org<BR>Subject: [Sip] New version of=20
draft-holmberg-sip-keep<BR><BR><BR>Hi,<BR><BR>I've submitted a new =
version (-04)=20
of the keep draft.<BR><BR>The requirement text is now alligned with the =
text=20
shown on the slides<BR>during the SFO SIP WG presentation.<BR><BR>There =
are no=20
other changes in the document from the previous=20
version.<BR><BR>Regards,<BR><BR>Christer</SPAN><o:p></o:p></P></DIV></BOD=
Y></HTML>

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From shida@ntt-at.com  Thu Mar 26 22:12:23 2009
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Cc: "sip@ietf.org List" <sip@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] New version of draft-holmberg-sip-keep
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Yes I support this work too.

On 26-Mar-09, at 4:16 PM, Mary Barnes wrote:

> Hi folks,
>
> Christer has updated the document based on the action item from  
> Wednesday's SIP session (chaired by SIPPING chairs and modeled as  
> the proposed DISPATCH WG).  If folks could please review the  
> document and raise any concerns on the document now, that would be  
> appreciated. We ask specifically for an answer to the question:  Do  
> you support the document being adopted as a WG document in RAI (WG  
> tbd)?  A simple "yes" is fine, however, if you don't think we should  
> adopt this document, could you please provide the reason.
>
> For your convenience, here's the link to the document:
> http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt
> And, the diff:
> http://tools.ietf.org//rfcdiff?url1=http://tools.ietf.org/id/draft-holmberg-sip-keep-03.txt&url2=http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt
> tinyurl: http://tinyurl.com/dfyvxz
>
> Regards,
> Mary
> SIPPING WG co-chair
>
>
> -----Original Message-----
> From: sip-bounces@ietf.org on behalf of Christer Holmberg
> Sent: Thu 3/26/2009 12:41 PM
> To: sip@ietf.org
> Subject: [Sip] New version of draft-holmberg-sip-keep
>
>
> Hi,
>
> I've submitted a new version (-04) of the keep draft.
>
> The requirement text is now alligned with the text shown on the slides
> during the SFO SIP WG presentation.
>
> There are no other changes in the document from the previous version.
>
> Regards,
>
> Christer
>
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><div><br></div><div>Yes I =
support this work too.</div><br><div><div>On 26-Mar-09, at 4:16 PM, Mary =
Barnes wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"> <div> <!-- Converted from text/plain format --><p><font =
size=3D"2">Hi folks,<br> <br> Christer has updated the document based on =
the action item from Wednesday's SIP session (chaired by SIPPING chairs =
and modeled as the proposed DISPATCH WG).&nbsp; If folks could please =
review the document and raise any concerns on the document now, that =
would be appreciated. We ask specifically for an answer to the =
question:&nbsp; Do you support the document being adopted as a WG =
document in RAI (WG tbd)?&nbsp; A simple "yes" is fine, however, if you =
don't think we should adopt this document, could you please provide the =
reason.<br> <br> For your convenience, here's the link to the =
document:<br> <a =
href=3D"http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt">http://to=
ols.ietf.org/id/draft-holmberg-sip-keep-04.txt</a><br> And, the =
diff:<br> <a =
href=3D"http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/dra=
ft-holmberg-sip-keep-03.txt&amp;url2=3Dhttp://tools.ietf.org/id/draft-holm=
berg-sip-keep-04.txt">http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.i=
etf.org/id/draft-holmberg-sip-keep-03.txt&amp;url2=3Dhttp://tools.ietf.org=
/id/draft-holmberg-sip-keep-04.txt</a><br> tinyurl: <a =
href=3D"http://tinyurl.com/dfyvxz">http://tinyurl.com/dfyvxz</a><br> =
<br> Regards,<br> Mary<br> SIPPING WG co-chair<br> <br> <br> =
-----Original Message-----<br> From: <a =
href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a> on behalf =
of Christer Holmberg<br> Sent: Thu 3/26/2009 12:41 PM<br> To: <a =
href=3D"mailto:sip@ietf.org">sip@ietf.org</a><br> Subject: [Sip] New =
version of draft-holmberg-sip-keep<br> <br> <br> Hi,<br> <br> I've =
submitted a new version (-04) of the keep draft.<br> <br> The =
requirement text is now alligned with the text shown on the slides<br> =
during the SFO SIP WG presentation.<br> <br> There are no other changes =
in the document from the previous version.<br> <br> Regards,<br> <br> =
Christer<br> <br> </font> </p> </div> =
_______________________________________________<br>Sip mailing list =
&nbsp;<a =
href=3D"https://www.ietf.org/mailman/listinfo/sip">https://www.ietf.org/ma=
ilman/listinfo/sip</a><br>This list is for NEW development of the core =
SIP Protocol<br>Use <a =
href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.colum=
bia.edu</a> for questions on current sip<br>Use <a =
href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a> for new =
developments on the application of =
sip</blockquote></div><br></body></html>=

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To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Fri, 27 Mar 2009 00:13:47 -0700
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Subject: Re: [Sip] Is REFER within dialog part of INVITE or	subscription	usage? (was RE: SIP INFO)
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> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Tuesday, March 24, 2009 7:43 AM
> To: Doken, Serhad
> Cc: Dale Worley; SIP IETF; Brett Tate
> Subject: Re: [Sip] Is REFER within dialog part of INVITE or
> subscription usage? (was RE: SIP INFO)
>=20
>=20
>=20
> Doken, Serhad wrote:
>=20
> >> I guess the question is: would it be valid to send a REFER within an
> >> existing dialog that had no INVITE usage? If it were valid,
> presumably
> >
> > Yes, I think so.
> >
> >
> >> it would have the same meaning as sending one outside a dialog,
> except
> >> that the resulting subscription would share the dialog. I expect
> that
> >> is
> >> a usage it would be difficult to find in the wild.
> >
> > Consider a third party control application that forces an endpoint to
> register :
> >
> > REFER sip:phone.doken.com SIP/2.0
> > Refer-To: <sip:serhad@doken.com;method=3DREGISTER>
> >
> > that triggers a REG to the registrar as :
> >
> > REGISTER sip:serhad@doken.com SIP/2.0
>=20
> You are suggesting that first this 3pcc application sends a SUBSCRIBE
> to
> this UA, establishing a dialog. Then it sends a REFER within the
> dialog,
> asking the UA REGISTER? That then would presumably result in a
> subscription to the refer event package sharing the dialog with the
> prior subscription. I guess this is *possible*, though I don't know
> *why* it would be done. What would the initial SUBSCRIBE be to, and how
> would it be established before the UA has registered?

3rd party contact center desktop monitoring application is used by the supe=
rvisor to see agent's activities. Thru a CTI <->SIP translator, it SUBs to =
the agent's presence. Once the customer service agent comes in the morning =
and turns his phone's status to Online(Open), app notices this and forces i=
t to Register. At the end of the day when agent leaves, does the opposite a=
nd de-registers it. Thus, it is made sure that they only get billed for the=
 time that agent was registered(online) by the service provider.

>=20
> Are you aware of such usage in the wild?

I have seen somewhat similar examples. If you are offering enough stimulus/=
TARP money, wild frontiers will get closer.

It's mostly used for Xfer however as you alluded to, REFER is implicit or a=
mbiguous that it is open to be used as the Swiss knife if I wanted the othe=
r side to do something for me. As long as the sender and receiver know/unde=
rstand what is meant or the context(especially if they are from the same ve=
ndor), everything is fine, if not misinterpretations may happen.

>=20
> (Its incidental to this discussion, but REGISTER presents unique
> challenges because of the way addressing works for it. Contrary to your
> example above, we would typically expect that the To: would contain the
> full URI, while the R-URI would only be sip:doken.com. This would

You are right.

Thanks,
Serhad

> require more special case processing.)
>=20
> 	Thanks,
> 	Paul

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From: "Doken, Serhad" <sdoken@qualcomm.com>
To: aayush bhatnagar <abhatnagar192006@gmail.com>
Date: Fri, 27 Mar 2009 00:13:34 -0700
Thread-Topic: [Sip] Is REFER within dialog part of INVITE or subscription usage? (was RE: SIP INFO)
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Cc: SIP IETF <SIP@ietf.org>, Paul Kyzivat <pkyzivat@cisco.com>, Brett Tate <brett@broadsoft.com>
Subject: Re: [Sip] Is REFER within dialog part of INVITE or subscription usage? (was RE: SIP INFO)
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Perhaps presence was not the best example there given what is supported tod=
ay though that should not limit the imagination since new services/devices =
pop up every day. Point is  this can be enabled by the basic Sip building b=
locks.

Event=3Dkpml will do for a use case that an endpoint tells the server that =
it wants the key presses to be monitored.

I was making the case that SUB can be included in method parm of REFER.

Serhad

From: aayush bhatnagar [mailto:abhatnagar192006@gmail.com]
Sent: Tuesday, March 24, 2009 1:19 AM
To: Doken, Serhad
Cc: Paul Kyzivat; SIP IETF; Brett Tate
Subject: Re: [Sip] Is REFER within dialog part of INVITE or subscription us=
age? (was RE: SIP INFO)

>One can send a out of dialog, norefersub REFER and no dialog is establishe=
d at the recipient. >Only an implicit subscription is created and a NTFY ma=
y be sent or not sent at all.

you are right.

>It should be possible to do Refer-To: sip:serhad@doken.com<mailto:sip%3Ase=
rhad@doken.com>;method=3DSUBSCRIBE?>Event=3Dpresence asking REFER receiver =
to send me a SUBSCRIBE in order to find out about >my presence.

In case the receiver is a server in the network, it must implement the role=
 of a PNA. If its an endpoint, it will subscribe only after approval of the=
 human user, else respond back with 400. I am not sure how many presence cl=
ients support forced subscriptions like this.
On Tue, Mar 24, 2009 at 1:01 PM, Doken, Serhad <sdoken@qualcomm.com<mailto:=
sdoken@qualcomm.com>> wrote:


> -----Original Message-----
> From: sip-bounces@ietf.org<mailto:sip-bounces@ietf.org> [mailto:sip-bounc=
es@ietf.org<mailto:sip-bounces@ietf.org>] On Behalf Of
> aayush bhatnagar
> Sent: Monday, March 23, 2009 8:30 PM
> To: Paul Kyzivat
> Cc: SIP IETF; Brett Tate
> Subject: Re: [Sip] Is REFER within dialog part of INVITE or
> subscription usage? (was RE: SIP INFO)
>
> REFER if placed within the scope of the existing dialog usage of
> INVITE will reuse its dialog usage, and create an implicit
> subscription for the refer event package at the recipient. However, if
> the recipient chooses to reject the in dialog REFER request, the
> communication established by the INVITE should not be affected. Such a
> rejection scenario implies that the call transfer failed..however the
> existing active call should not be torn down.
>
> REFER if sent outside the scope of any exisiting dialog will behave as
> a dialog creating request and will establish a subscription at the
> recipient as described above. This subscription is only used to notify
> the status of the sip session referral to the originator and is
> seperate from any other subscribe usage that may have existed at the
> recipient.
One can send a out of dialog, norefersub REFER and no dialog is established=
 at the recipient. Only an implicit subscription is created and a NTFY may =
be sent or not sent at all.

>
> As far as accessing the referred resource is concerned...using the
> method parameter, it depends upon the business logic of the recipient
> of the REFER request. You may send a REFER to a conferencing server
> with a method INVITE to invite a user to join a conference. A method
> parameter with the value BYE would imply removal from the conference
> by the server and so on.
>
> I dont think you can include a SUBSCRIBE in the method parameter of
> the REFER request. You have no way of telling the event package usage
> of the subscription to the recipient, as the event header cannot be
It should be possible to do Refer-To: sip:serhad@doken.com<mailto:sip%3Aser=
had@doken.com>;method=3DSUBSCRIBE?Event=3Dpresence asking REFER receiver to=
 send me a SUBSCRIBE in order to find out about my presence.

> included in the REFER request. Moreover, you can never be sure whether
> the subscription usage is even applicable to the recipient or whether
> the recipient even supports it.
>
> aayush
>
> On 3/24/09, Paul Kyzivat <pkyzivat@cisco.com<mailto:pkyzivat@cisco.com>> =
wrote:
> >
> >
> > Dale Worley wrote:
> >> On Mon, 2009-03-23 at 08:18 -0700, Brett Tate wrote:
> >>> Everyone: Is REFER within dialog part of INVITE usage, SUBSCRIPTION
> >>> usage, or neither?
> >>>
> >>> I'm asking mainly because of REFER 481 impacts; however the answer
> >>> also has potential implications concerning draft-ietf-sip-info-
> events.
> >>> RFC 5057 appears to imply that it is part of the subscription
> usage;
> >>> however I'm not positive if my interpretation and/or RFC 5057 is
> >>> correct.
> >>>
> >>> If REFER isn't considered part of the INVITE usage, how should
> REFER
> >>> 481 be interpreted?  More specifically, should receiver of REFER
> 481
> >>> send BYE?
> >>
> >> I would argue that the REFER is part of the INVITE usage, because
> REFER
> >> in that context is intimately related to the INVITE usage -- it is
> >> intended to transfer the INVITE usage/dialog transferred to a
> different
> >> destination.
> >
> > Well, I guess there is that implied association. (Part of the "do
> what
> > *I* had in mind" approach that REFER has.) Since this was never
> spelled
> > out, it could also be simply "while this isn't part of an INVITE
> usage,
> > if there happens to be an INVITE usage, please refer that one."
> >
> > I guess the question is: would it be valid to send a REFER within an
> > existing dialog that had no INVITE usage? If it were valid,
> presumably
> > it would have the same meaning as sending one outside a dialog,
> except
> > that the resulting subscription would share the dialog. I expect that
> is
> > a usage it would be difficult to find in the wild.
> >
> > Ah, but now I want to reconsider. Remember that REFER can take a
> method
> > name. So, rather than referring an INVITE, I could be referring
> MESSAGE,
> > OPTIONS, BYE, or maybe (????) SUBSCRIBE. In some of those cases, the
> > association to the INVITE usage is probably meaningless. (Of course
> its
> > meaningful for BYE.)
> >
> >> And it seems quite safe that if the REFER receives a 481 response,
> the
> >> INVITE usage must not be functional any more, since the far end of
> that
> >> usage has denied that it can act on the REFER.
> >
> > IIRC, 5057 decided that each dialog usage must get its own 481 before
> it
> > goes away. I don't think there are any cases where a single 481
> response
> > can make multiple usages go away.
> >
> > So the important question is whether the REFER is part of the INVITE
> > usage and so its 481 takes down the INVITE usage or not. That gets
> > especially interesting if  the dialog already had both an INVITE
> usage
> > and a SUBSCRIBE usage. Does it make sense that the refer take down
> the
> > INVITE usage and leave up the SUBSCRIBE usage?
> >
> >> Now it's possible that the REFER, if it is successful, is the first
> >> transaction of the subscription usage.  But I don't think that has
> any
> >> paradoxical consequences, as by hypothesis, the REFER succeeded.
> >
> > Yeah, its already so weird, why not a little more weird?
> >
> > Also, I think it has recently been clarified that the 200 of a
> SUBSCRIBE
> > should not be considered to create a dialog - that its the NOTIFY
> that
> > creates the dialog. If so, then I suppose the same should be true of
> REFER.
> >
> > Bottom line: I can live with it either way, but think it should be
> > documented one way or the other.
> >
> >     Thanks,
> >     Paul
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu<mailto:sip-implementors@cs.columbi=
a.edu> for questions on current sip
> > Use sipping@ietf.org<mailto:sipping@ietf.org> for new developments on t=
he application of sip
> >
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu<mailto:sip-implementors@cs.columbia.=
edu> for questions on current sip
> Use sipping@ietf.org<mailto:sipping@ietf.org> for new developments on the=
 application of sip


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<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>Perhaps presence was not the best example there given what i=
s
supported today though that should not limit the imagination since new
services/devices pop up every day. Point is &nbsp;this can be enabled by th=
e
basic Sip building blocks.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>Event=3Dkpml will do for a use case that an endpoint tells t=
he
server that it wants the key presses to be monitored. <o:p></o:p></span></p=
>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>I was making the case that SUB can be included in method par=
m of
REFER.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>Serhad<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in'>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma=
","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> aayush bhatna=
gar
[mailto:abhatnagar192006@gmail.com] <br>
<b>Sent:</b> Tuesday, March 24, 2009 1:19 AM<br>
<b>To:</b> Doken, Serhad<br>
<b>Cc:</b> Paul Kyzivat; SIP IETF; Brett Tate<br>
<b>Subject:</b> Re: [Sip] Is REFER within dialog part of INVITE or subscrip=
tion
usage? (was RE: SIP INFO)<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>&gt;One can send a out of dialog, norefersub REFER and=
 no
dialog is established at the recipient. &gt;Only an implicit subscription i=
s
created and a NTFY may be sent or not sent at all.<o:p></o:p></p>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>you are right.&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>&gt;It should be possible to do Refer-To:&nbsp;<a
href=3D"mailto:sip%3Aserhad@doken.com">sip:serhad@doken.com</a>;method=3DSU=
BSCRIBE?&gt;Event=3Dpresence
asking REFER receiver to send me a SUBSCRIBE in order to find out about &gt=
;my
presence.<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'>In case the receiver is=
 a
server in the network, it must implement the role of a PNA. If its an endpo=
int,
it will subscribe only after approval of the human user, else respond back =
with
400. I am not sure how many presence clients support forced subscriptions l=
ike
this.&nbsp;<o:p></o:p></p>

<div>

<p class=3DMsoNormal>On Tue, Mar 24, 2009 at 1:01 PM, Doken, Serhad &lt;<a
href=3D"mailto:sdoken@qualcomm.com">sdoken@qualcomm.com</a>&gt; wrote:<o:p>=
</o:p></p>

<div>

<p class=3DMsoNormal><br>
<br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a>
[mailto:<a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a>] O=
n
Behalf Of<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&gt; aayush bhatnagar<br>
&gt; Sent: Monday, March 23, 2009 8:30 PM<br>
&gt; To: Paul Kyzivat<br>
&gt; Cc: SIP IETF; Brett Tate<br>
&gt; Subject: Re: [Sip] Is REFER within dialog part of INVITE or<br>
&gt; subscription usage? (was RE: SIP INFO)<br>
&gt;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'>&gt; REFER if placed wi=
thin the
scope of the existing dialog usage of<br>
&gt; INVITE will reuse its dialog usage, and create an implicit<br>
&gt; subscription for the refer event package at the recipient. However, if=
<br>
&gt; the recipient chooses to reject the in dialog REFER request, the<br>
&gt; communication established by the INVITE should not be affected. Such a=
<br>
&gt; rejection scenario implies that the call transfer failed..however the<=
br>
&gt; existing active call should not be torn down.<br>
&gt;<br>
&gt; REFER if sent outside the scope of any exisiting dialog will behave as=
<br>
&gt; a dialog creating request and will establish a subscription at the<br>
&gt; recipient as described above. This subscription is only used to notify=
<br>
&gt; the status of the sip session referral to the originator and is<br>
&gt; seperate from any other subscribe usage that may have existed at the<b=
r>
&gt; recipient.<o:p></o:p></p>

</div>

<p class=3DMsoNormal>One can send a out of dialog, norefersub REFER and no =
dialog
is established at the recipient. Only an implicit subscription is created a=
nd a
NTFY may be sent or not sent at all.<o:p></o:p></p>

<div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><br>
&gt;<br>
&gt; As far as accessing the referred resource is concerned...using the<br>
&gt; method parameter, it depends upon the business logic of the recipient<=
br>
&gt; of the REFER request. You may send a REFER to a conferencing server<br=
>
&gt; with a method INVITE to invite a user to join a conference. A method<b=
r>
&gt; parameter with the value BYE would imply removal from the conference<b=
r>
&gt; by the server and so on.<br>
&gt;<br>
&gt; I dont think you can include a SUBSCRIBE in the method parameter of<br=
>
&gt; the REFER request. You have no way of telling the event package usage<=
br>
&gt; of the subscription to the recipient, as the event header cannot be<o:=
p></o:p></p>

</div>

<p class=3DMsoNormal>It should be possible to do Refer-To: <a
href=3D"mailto:sip%3Aserhad@doken.com">sip:serhad@doken.com</a>;method=3DSU=
BSCRIBE?Event=3Dpresence
asking REFER receiver to send me a SUBSCRIBE in order to find out about my
presence.<o:p></o:p></p>

<div>

<div>

<p class=3DMsoNormal><br>
&gt; included in the REFER request. Moreover, you can never be sure whether=
<br>
&gt; the subscription usage is even applicable to the recipient or whether<=
br>
&gt; the recipient even supports it.<br>
&gt;<br>
&gt; aayush<br>
&gt;<br>
&gt; On 3/24/09, Paul Kyzivat &lt;<a href=3D"mailto:pkyzivat@cisco.com">pky=
zivat@cisco.com</a>&gt;
wrote:<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; Dale Worley wrote:<br>
&gt; &gt;&gt; On Mon, 2009-03-23 at 08:18 -0700, Brett Tate wrote:<br>
&gt; &gt;&gt;&gt; Everyone: Is REFER within dialog part of INVITE usage,
SUBSCRIPTION<br>
&gt; &gt;&gt;&gt; usage, or neither?<br>
&gt; &gt;&gt;&gt;<br>
&gt; &gt;&gt;&gt; I'm asking mainly because of REFER 481 impacts; however t=
he
answer<br>
&gt; &gt;&gt;&gt; also has potential implications concerning
draft-ietf-sip-info-<br>
&gt; events.<br>
&gt; &gt;&gt;&gt; RFC 5057 appears to imply that it is part of the subscrip=
tion<br>
&gt; usage;<br>
&gt; &gt;&gt;&gt; however I'm not positive if my interpretation and/or RFC =
5057
is<br>
&gt; &gt;&gt;&gt; correct.<br>
&gt; &gt;&gt;&gt;<br>
&gt; &gt;&gt;&gt; If REFER isn't considered part of the INVITE usage, how
should<br>
&gt; REFER<br>
&gt; &gt;&gt;&gt; 481 be interpreted? &nbsp;More specifically, should recei=
ver
of REFER<br>
&gt; 481<br>
&gt; &gt;&gt;&gt; send BYE?<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; I would argue that the REFER is part of the INVITE usage, bec=
ause<br>
&gt; REFER<br>
&gt; &gt;&gt; in that context is intimately related to the INVITE usage -- =
it
is<br>
&gt; &gt;&gt; intended to transfer the INVITE usage/dialog transferred to a=
<br>
&gt; different<br>
&gt; &gt;&gt; destination.<br>
&gt; &gt;<br>
&gt; &gt; Well, I guess there is that implied association. (Part of the
&quot;do<br>
&gt; what<br>
&gt; &gt; *I* had in mind&quot; approach that REFER has.) Since this was ne=
ver<br>
&gt; spelled<br>
&gt; &gt; out, it could also be simply &quot;while this isn't part of an IN=
VITE<br>
&gt; usage,<br>
&gt; &gt; if there happens to be an INVITE usage, please refer that one.&qu=
ot;<br>
&gt; &gt;<br>
&gt; &gt; I guess the question is: would it be valid to send a REFER within=
 an<br>
&gt; &gt; existing dialog that had no INVITE usage? If it were valid,<br>
&gt; presumably<br>
&gt; &gt; it would have the same meaning as sending one outside a dialog,<b=
r>
&gt; except<br>
&gt; &gt; that the resulting subscription would share the dialog. I expect =
that<br>
&gt; is<br>
&gt; &gt; a usage it would be difficult to find in the wild.<br>
&gt; &gt;<br>
&gt; &gt; Ah, but now I want to reconsider. Remember that REFER can take a<=
br>
&gt; method<br>
&gt; &gt; name. So, rather than referring an INVITE, I could be referring<b=
r>
&gt; MESSAGE,<br>
&gt; &gt; OPTIONS, BYE, or maybe (????) SUBSCRIBE. In some of those cases, =
the<br>
&gt; &gt; association to the INVITE usage is probably meaningless. (Of cour=
se<br>
&gt; its<br>
&gt; &gt; meaningful for BYE.)<br>
&gt; &gt;<br>
&gt; &gt;&gt; And it seems quite safe that if the REFER receives a 481
response,<br>
&gt; the<br>
&gt; &gt;&gt; INVITE usage must not be functional any more, since the far e=
nd
of<br>
&gt; that<br>
&gt; &gt;&gt; usage has denied that it can act on the REFER.<br>
&gt; &gt;<br>
&gt; &gt; IIRC, 5057 decided that each dialog usage must get its own 481 be=
fore<br>
&gt; it<br>
&gt; &gt; goes away. I don't think there are any cases where a single 481<b=
r>
&gt; response<br>
&gt; &gt; can make multiple usages go away.<br>
&gt; &gt;<br>
&gt; &gt; So the important question is whether the REFER is part of the INV=
ITE<br>
&gt; &gt; usage and so its 481 takes down the INVITE usage or not. That get=
s<br>
&gt; &gt; especially interesting if &nbsp;the dialog already had both an IN=
VITE<br>
&gt; usage<br>
&gt; &gt; and a SUBSCRIBE usage. Does it make sense that the refer take dow=
n<br>
&gt; the<br>
&gt; &gt; INVITE usage and leave up the SUBSCRIBE usage?<br>
&gt; &gt;<br>
&gt; &gt;&gt; Now it's possible that the REFER, if it is successful, is the
first<br>
&gt; &gt;&gt; transaction of the subscription usage. &nbsp;But I don't thin=
k
that has<br>
&gt; any<br>
&gt; &gt;&gt; paradoxical consequences, as by hypothesis, the REFER succeed=
ed.<br>
&gt; &gt;<br>
&gt; &gt; Yeah, its already so weird, why not a little more weird?<br>
&gt; &gt;<br>
&gt; &gt; Also, I think it has recently been clarified that the 200 of a<br=
>
&gt; SUBSCRIBE<br>
&gt; &gt; should not be considered to create a dialog - that its the NOTIFY=
<br>
&gt; that<br>
&gt; &gt; creates the dialog. If so, then I suppose the same should be true=
 of<br>
&gt; REFER.<br>
&gt; &gt;<br>
&gt; &gt; Bottom line: I can live with it either way, but think it should b=
e<br>
&gt; &gt; documented one way or the other.<br>
&gt; &gt;<br>
&gt; &gt; &nbsp; &nbsp; Thanks,<br>
&gt; &gt; &nbsp; &nbsp; Paul<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; Sip mailing list &nbsp;<a
href=3D"https://www.ietf.org/mailman/listinfo/sip" target=3D"_blank">https:=
//www.ietf.org/mailman/listinfo/sip</a><br>
&gt; &gt; This list is for NEW development of the core SIP Protocol<br>
&gt; &gt; Use <a href=3D"mailto:sip-implementors@cs.columbia.edu">sip-imple=
mentors@cs.columbia.edu</a>
for questions on current sip<br>
&gt; &gt; Use <a href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a> for =
new
developments on the application of sip<br>
&gt; &gt;<br>
&gt; _______________________________________________<br>
&gt; Sip mailing list &nbsp;<a href=3D"https://www.ietf.org/mailman/listinf=
o/sip"
target=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
&gt; This list is for NEW development of the core SIP Protocol<br>
&gt; Use <a href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implemento=
rs@cs.columbia.edu</a>
for questions on current sip<br>
&gt; Use <a href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a> for new
developments on the application of sip<o:p></o:p></p>

</div>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</div>

</div>

</body>

</html>

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To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, Mary Barnes <mary.barnes@nortel.com>, Christer Holmberg <christer.holmberg@ericsson.com>, "sip@ietf.org" <sip@ietf.org>
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++

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of DRAGE=
, Keith (Keith)
Sent: Thursday, March 26, 2009 9:53 PM
To: Mary Barnes; Christer Holmberg; sip@ietf.org
Subject: Re: [Sip] New version of draft-holmberg-sip-keep

As individual

Support

Keith

________________________________
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Mary =
Barnes
Sent: Thursday, March 26, 2009 11:17 PM
To: Christer Holmberg; sip@ietf.org
Subject: Re: [Sip] New version of draft-holmberg-sip-keep

Hi folks,

Christer has updated the document based on the action item from Wednesday's=
 SIP session (chaired by SIPPING chairs and modeled as the proposed DISPATC=
H WG).  If folks could please review the document and raise any concerns on=
 the document now, that would be appreciated. We ask specifically for an an=
swer to the question:  Do you support the document being adopted as a WG do=
cument in RAI (WG tbd)?  A simple "yes" is fine, however, if you don't thin=
k we should adopt this document, could you please provide the reason.

For your convenience, here's the link to the document:
http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt
And, the diff:
http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-holmbe=
rg-sip-keep-03.txt&url2=3Dhttp://tools.ietf.org/id/draft-holmberg-sip-keep-=
04.txt<http://tools.ietf.org/rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-=
holmberg-sip-keep-03.txt&url2=3Dhttp://tools.ietf.org/id/draft-holmberg-sip=
-keep-04.txt>
tinyurl: http://tinyurl.com/dfyvxz

Regards,
Mary
SIPPING WG co-chair


-----Original Message-----
From: sip-bounces@ietf.org on behalf of Christer Holmberg
Sent: Thu 3/26/2009 12:41 PM
To: sip@ietf.org
Subject: [Sip] New version of draft-holmberg-sip-keep


Hi,

I've submitted a new version (-04) of the keep draft.

The requirement text is now alligned with the text shown on the slides
during the SFO SIP WG presentation.

There are no other changes in the document from the previous version.

Regards,

Christer

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<title>RE: [Sip] New version of draft-holmberg-sip-keep</title>
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<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma=
","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <b>On Behalf Of </b>DRAG=
E,
Keith (Keith)<br>
<b>Sent:</b> Thursday, March 26, 2009 9:53 PM<br>
<b>To:</b> Mary Barnes; Christer Holmberg; sip@ietf.org<br>
<b>Subject:</b> Re: [Sip] New version of draft-holmberg-sip-keep<o:p></o:p>=
</span></p>

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<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
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color:blue'>As individual</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif";
color:blue'>Support</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif";
color:blue'>Keith</span><o:p></o:p></p>

<blockquote style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0=
in 0in 4.0pt;
margin-left:3.75pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'>

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<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><span style=3D'font-=
size:10.0pt;
font-family:"Tahoma","sans-serif"'>From:</span></b><span style=3D'font-size=
:10.0pt;
font-family:"Tahoma","sans-serif"'> sip-bounces@ietf.org
[mailto:sip-bounces@ietf.org] <b>On Behalf Of </b>Mary Barnes<br>
<b>Sent:</b> Thursday, March 26, 2009 11:17 PM<br>
<b>To:</b> Christer Holmberg; sip@ietf.org<br>
<b>Subject:</b> Re: [Sip] New version of draft-holmberg-sip-keep</span><o:p=
></o:p></p>

<p style=3D'margin-bottom:12.0pt'><span style=3D'font-size:10.0pt'>Hi folks=
,<br>
<br>
Christer has updated the document based on the action item from Wednesday's=
 SIP
session (chaired by SIPPING chairs and modeled as the proposed DISPATCH
WG).&nbsp; If folks could please review the document and raise any concerns=
 on
the document now, that would be appreciated. We ask specifically for an ans=
wer
to the question:&nbsp; Do you support the document being adopted as a WG
document in RAI (WG tbd)?&nbsp; A simple &quot;yes&quot; is fine, however, =
if
you don't think we should adopt this document, could you please provide the
reason.<br>
<br>
For your convenience, here's the link to the document:<br>
<a href=3D"http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt">http://=
tools.ietf.org/id/draft-holmberg-sip-keep-04.txt</a><br>
And, the diff:<br>
<a
href=3D"http://tools.ietf.org/rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft=
-holmberg-sip-keep-03.txt&amp;url2=3Dhttp://tools.ietf.org/id/draft-holmber=
g-sip-keep-04.txt">http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.=
org/id/draft-holmberg-sip-keep-03.txt&amp;url2=3Dhttp://tools.ietf.org/id/d=
raft-holmberg-sip-keep-04.txt</a><br>
tinyurl: <a href=3D"http://tinyurl.com/dfyvxz">http://tinyurl.com/dfyvxz</a=
><br>
<br>
Regards,<br>
Mary<br>
SIPPING WG co-chair<br>
<br>
<br>
-----Original Message-----<br>
From: sip-bounces@ietf.org on behalf of Christer Holmberg<br>
Sent: Thu 3/26/2009 12:41 PM<br>
To: sip@ietf.org<br>
Subject: [Sip] New version of draft-holmberg-sip-keep<br>
<br>
<br>
Hi,<br>
<br>
I've submitted a new version (-04) of the keep draft.<br>
<br>
The requirement text is now alligned with the text shown on the slides<br>
during the SFO SIP WG presentation.<br>
<br>
There are no other changes in the document from the previous version.<br>
<br>
Regards,<br>
<br>
Christer</span><o:p></o:p></p>

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From marianne.mohali@orange-ftgroup.com  Fri Mar 27 00:14:09 2009
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Subject: Re: [Sip] New version of draft-holmberg-sip-keep
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Yes

________________________________

De : sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] De la part de =
Shida Schubert
Envoy=E9 : jeudi 26 mars 2009 22:13
=C0 : Mary Barnes
Cc : sip@ietf.org List; Christer Holmberg
Objet : Re: [Sip] New version of draft-holmberg-sip-keep



Yes I support this work too.

On 26-Mar-09, at 4:16 PM, Mary Barnes wrote:


	Hi folks,
=09
	Christer has updated the document based on the action item from =
Wednesday's SIP session (chaired by SIPPING chairs and modeled as the =
proposed DISPATCH WG).  If folks could please review the document and =
raise any concerns on the document now, that would be appreciated. We =
ask specifically for an answer to the question:  Do you support the =
document being adopted as a WG document in RAI (WG tbd)?  A simple "yes" =
is fine, however, if you don't think we should adopt this document, =
could you please provide the reason.
=09
	For your convenience, here's the link to the document:
	http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt
	And, the diff:
	=
http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-holm=
berg-sip-keep-03.txt&url2=3Dhttp://tools.ietf.org/id/draft-holmberg-sip-k=
eep-04.txt
	tinyurl: http://tinyurl.com/dfyvxz
=09
	Regards,
	Mary
	SIPPING WG co-chair
=09
=09
	-----Original Message-----
	From: sip-bounces@ietf.org on behalf of Christer Holmberg
	Sent: Thu 3/26/2009 12:41 PM
	To: sip@ietf.org
	Subject: [Sip] New version of draft-holmberg-sip-keep
=09
=09
	Hi,
=09
	I've submitted a new version (-04) of the keep draft.
=09
	The requirement text is now alligned with the text shown on the slides
	during the SFO SIP WG presentation.
=09
	There are no other changes in the document from the previous version.
=09
	Regards,
=09
	Christer
=09
=09

	_______________________________________________
	Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
	This list is for NEW development of the core SIP Protocol
	Use sip-implementors@cs.columbia.edu for questions on current sip
	Use sipping@ietf.org for new developments on the application of sip



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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2900.3492" name=3DGENERATOR></HEAD>
<BODY=20
style=3D"WORD-WRAP: break-word; webkit-nbsp-mode: space; =
webkit-line-break: after-white-space">
<DIV dir=3Dltr align=3Dleft><SPAN class=3D423231407-27032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Yes</FONT></SPAN></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Dfr dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>De&nbsp;:</B> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>De la part de</B> Shida=20
Schubert<BR><B>Envoy=E9&nbsp;:</B> jeudi 26 mars 2009 =
22:13<BR><B>=C0&nbsp;:</B>=20
Mary Barnes<BR><B>Cc&nbsp;:</B> sip@ietf.org List; Christer=20
Holmberg<BR><B>Objet&nbsp;:</B> Re: [Sip] New version of=20
draft-holmberg-sip-keep<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><BR></DIV>
<DIV>Yes I support this work too.</DIV><BR>
<DIV>
<DIV>On 26-Mar-09, at 4:16 PM, Mary Barnes wrote:</DIV><BR=20
class=3DApple-interchange-newline>
<BLOCKQUOTE type=3D"cite">
  <DIV><!-- Converted from text/plain format -->
  <P><FONT size=3D2>Hi folks,<BR><BR>Christer has updated the document =
based on=20
  the action item from Wednesday's SIP session (chaired by SIPPING =
chairs and=20
  modeled as the proposed DISPATCH WG).&nbsp; If folks could please =
review the=20
  document and raise any concerns on the document now, that would be=20
  appreciated. We ask specifically for an answer to the question:&nbsp; =
Do you=20
  support the document being adopted as a WG document in RAI (WG =
tbd)?&nbsp; A=20
  simple "yes" is fine, however, if you don't think we should adopt this =

  document, could you please provide the reason.<BR><BR>For your =
convenience,=20
  here's the link to the document:<BR><A=20
  =
href=3D"http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt">http://t=
ools.ietf.org/id/draft-holmberg-sip-keep-04.txt</A><BR>And,=20
  the diff:<BR><A=20
  =
href=3D"http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/dr=
aft-holmberg-sip-keep-03.txt&amp;url2=3Dhttp://tools.ietf.org/id/draft-ho=
lmberg-sip-keep-04.txt">http://tools.ietf.org//rfcdiff?url1=3Dhttp://tool=
s.ietf.org/id/draft-holmberg-sip-keep-03.txt&amp;url2=3Dhttp://tools.ietf=
.org/id/draft-holmberg-sip-keep-04.txt</A><BR>tinyurl:=20
  <A=20
  =
href=3D"http://tinyurl.com/dfyvxz">http://tinyurl.com/dfyvxz</A><BR><BR>R=
egards,<BR>Mary<BR>SIPPING=20
  WG co-chair<BR><BR><BR>-----Original Message-----<BR>From: <A=20
  href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</A> on =
behalf of=20
  Christer Holmberg<BR>Sent: Thu 3/26/2009 12:41 PM<BR>To: <A=20
  href=3D"mailto:sip@ietf.org">sip@ietf.org</A><BR>Subject: [Sip] New =
version of=20
  draft-holmberg-sip-keep<BR><BR><BR>Hi,<BR><BR>I've submitted a new =
version=20
  (-04) of the keep draft.<BR><BR>The requirement text is now alligned =
with the=20
  text shown on the slides<BR>during the SFO SIP WG =
presentation.<BR><BR>There=20
  are no other changes in the document from the previous=20
  =
version.<BR><BR>Regards,<BR><BR>Christer<BR><BR></FONT></P></DIV>________=
_______________________________________<BR>Sip=20
  mailing list &nbsp;<A=20
  =
href=3D"https://www.ietf.org/mailman/listinfo/sip">https://www.ietf.org/m=
ailman/listinfo/sip</A><BR>This=20
  list is for NEW development of the core SIP Protocol<BR>Use <A=20
  =
href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.colu=
mbia.edu</A>=20
  for questions on current sip<BR>Use <A=20
  href=3D"mailto:sipping@ietf.org">sipping@ietf.org</A> for new =
developments on=20
  the application of sip</BLOCKQUOTE></DIV><BR></BODY></HTML>

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Cc: SIP IETF <SIP@ietf.org>, Brett Tate <brett@broadsoft.com>
Subject: Re: [Sip] Is REFER within dialog part of INVITE or	subscription	usage? (was RE: SIP INFO)
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Doken, Serhad wrote:
> 
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Tuesday, March 24, 2009 7:43 AM
>> To: Doken, Serhad
>> Cc: Dale Worley; SIP IETF; Brett Tate
>> Subject: Re: [Sip] Is REFER within dialog part of INVITE or
>> subscription usage? (was RE: SIP INFO)
>>
>>
>>
>> Doken, Serhad wrote:
>>
>>>> I guess the question is: would it be valid to send a REFER within an
>>>> existing dialog that had no INVITE usage? If it were valid,
>> presumably
>>> Yes, I think so.
>>>
>>>
>>>> it would have the same meaning as sending one outside a dialog,
>> except
>>>> that the resulting subscription would share the dialog. I expect
>> that
>>>> is
>>>> a usage it would be difficult to find in the wild.
>>> Consider a third party control application that forces an endpoint to
>> register :
>>> REFER sip:phone.doken.com SIP/2.0
>>> Refer-To: <sip:serhad@doken.com;method=REGISTER>
>>>
>>> that triggers a REG to the registrar as :
>>>
>>> REGISTER sip:serhad@doken.com SIP/2.0
>> You are suggesting that first this 3pcc application sends a SUBSCRIBE
>> to
>> this UA, establishing a dialog. Then it sends a REFER within the
>> dialog,
>> asking the UA REGISTER? That then would presumably result in a
>> subscription to the refer event package sharing the dialog with the
>> prior subscription. I guess this is *possible*, though I don't know
>> *why* it would be done. What would the initial SUBSCRIBE be to, and how
>> would it be established before the UA has registered?
> 
> 3rd party contact center desktop monitoring application is used by the supervisor to see agent's activities. Thru a CTI <->SIP translator, it SUBs to the agent's presence. Once the customer service agent comes in the morning and turns his phone's status to Online(Open), app notices this and forces it to Register. At the end of the day when agent leaves, does the opposite and de-registers it. Thus, it is made sure that they only get billed for the time that agent was registered(online) by the service provider.

I find this less than compelling. How does the monitoring application 
discover and address the UA to establish the subscription? If that was 
possible, then the UA must have a globally routable URI that is known to 
the application. And that could be used to send an out-of-dialog REFER.

	Thanks,
	Paul

>> Are you aware of such usage in the wild?
> 
> I have seen somewhat similar examples. If you are offering enough stimulus/TARP money, wild frontiers will get closer.
> 
> It's mostly used for Xfer however as you alluded to, REFER is implicit or ambiguous that it is open to be used as the Swiss knife if I wanted the other side to do something for me. As long as the sender and receiver know/understand what is meant or the context(especially if they are from the same vendor), everything is fine, if not misinterpretations may happen.
> 
>> (Its incidental to this discussion, but REGISTER presents unique
>> challenges because of the way addressing works for it. Contrary to your
>> example above, we would typically expect that the To: would contain the
>> full URI, while the R-URI would only be sip:doken.com. This would
> 
> You are right.
> 
> Thanks,
> Serhad
> 
>> require more special case processing.)
>>
>> 	Thanks,
>> 	Paul
> 

From ietf.hanserik@gmail.com  Fri Mar 27 05:32:46 2009
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References: <24ACC38D-8B7D-46E6-B767-73A33F6C9266@standardstrack.com> <49C79CFD.1010408@cisco.com> <9B2A061A1137254BBE4F4B2CD843646A10BFF48DF0@mbx02.citservers.local> <1237853234.16613.41.camel@victoria-pingtel-com.us.nortel.com> <49C83831.7010102@cisco.com> <ED88AAAE8B3D764B9FD8558DE1775B69139274F552@NASANEXMB09.na.qualcomm.com> <49C8F18D.1040800@cisco.com> <ED88AAAE8B3D764B9FD8558DE1775B69139274F821@NASANEXMB09.na.qualcomm.com> <49CCC1A1.9040501@cisco.com>
Date: Fri, 27 Mar 2009 13:33:36 +0100
Message-ID: <9ae56b1e0903270533tf8cbb4dh44fe66ef605be34c@mail.gmail.com>
From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Content-Type: multipart/alternative; boundary=0015174c37701b880f046618f111
Cc: SIP IETF <SIP@ietf.org>, Brett Tate <brett@broadsoft.com>
Subject: Re: [Sip] Is REFER within dialog part of INVITE or subscription usage? (was RE: SIP INFO)
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--0015174c37701b880f046618f111
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That, AND why would you not use the OPEN line button to just REGISTER? And
make the application be notified that the registration/deregistration
happened. You could for example route the registrations through the
application hosting proxy to the registrar. Or have like in IMS the
possibility for the registrar to notify an application server upon
registration of a particular user. Much simpler.

/Hans Erik van Elburg

On Fri, Mar 27, 2009 at 1:08 PM, Paul Kyzivat <pkyzivat@cisco.com> wrote:

> You are suggesting that first this 3pcc application sends a SUBSCRIBE
>>> to
>>> this UA, establishing a dialog. Then it sends a REFER within the
>>> dialog,
>>> asking the UA REGISTER? That then would presumably result in a
>>> subscription to the refer event package sharing the dialog with the
>>> prior subscription. I guess this is *possible*, though I don't know
>>> *why* it would be done. What would the initial SUBSCRIBE be to, and how
>>> would it be established before the UA has registered?
>>>
>>
>> 3rd party contact center desktop monitoring application is used by the
>> supervisor to see agent's activities. Thru a CTI <->SIP translator, it SUBs
>> to the agent's presence. Once the customer service agent comes in the
>> morning and turns his phone's status to Online(Open), app notices this and
>> forces it to Register. At the end of the day when agent leaves, does the
>> opposite and de-registers it. Thus, it is made sure that they only get
>> billed for the time that agent was registered(online) by the service
>> provider.
>>
>
> I find this less than compelling. How does the monitoring application
> discover and address the UA to establish the subscription? If that was
> possible, then the UA must have a globally routable URI that is known to the
> application. And that could be used to send an out-of-dialog REFER.
>
>        Thanks,
>        Paul

--0015174c37701b880f046618f111
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

That, AND why would you not use the OPEN line button to just REGISTER? And =
make the application be notified that the registration/deregistration happe=
ned. You could for example route the registrations through the application =
hosting proxy to the registrar. Or have like in IMS the possibility for the=
 registrar to notify an application server upon registration of a particula=
r user. Much simpler.<br>
<br clear=3D"all">/Hans Erik van Elburg<br>
<br><div class=3D"gmail_quote">On Fri, Mar 27, 2009 at 1:08 PM, Paul Kyziva=
t <span dir=3D"ltr">&lt;<a href=3D"mailto:pkyzivat@cisco.com">pkyzivat@cisc=
o.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"b=
order-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; paddin=
g-left: 1ex;">
<div><div class=3D"im"><blockquote class=3D"gmail_quote" style=3D"border-le=
ft: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: =
1ex;"><blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb=
(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
You are suggesting that first this 3pcc application sends a SUBSCRIBE<br>
to<br>
this UA, establishing a dialog. Then it sends a REFER within the<br>
dialog,<br>
asking the UA REGISTER? That then would presumably result in a<br>
subscription to the refer event package sharing the dialog with the<br>
prior subscription. I guess this is *possible*, though I don&#39;t know<br>
*why* it would be done. What would the initial SUBSCRIBE be to, and how<br>
would it be established before the UA has registered?<br>
</blockquote>
<br>
3rd party contact center desktop monitoring application is used by the supe=
rvisor to see agent&#39;s activities. Thru a CTI &lt;-&gt;SIP translator, i=
t SUBs to the agent&#39;s presence. Once the customer service agent comes i=
n the morning and turns his phone&#39;s status to Online(Open), app notices=
 this and forces it to Register. At the end of the day when agent leaves, d=
oes the opposite and de-registers it. Thus, it is made sure that they only =
get billed for the time that agent was registered(online) by the service pr=
ovider.<br>

</blockquote>
<br></div></div>
I find this less than compelling. How does the monitoring application disco=
ver and address the UA to establish the subscription? If that was possible,=
 then the UA must have a globally routable URI that is known to the applica=
tion. And that could be used to send an out-of-dialog REFER.<br>

<br>
 =A0 =A0 =A0 =A0Thanks,<br><font color=3D"#888888">
 =A0 =A0 =A0 =A0Paul</font></blockquote></div><br>

--0015174c37701b880f046618f111--

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Thread-Topic: [Sip] New version of draft-holmberg-sip-keep
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Subject: Re: [Sip] New version of draft-holmberg-sip-keep
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Yes

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Mary Barnes
Sent: 26 March 2009 23:17
To: Christer Holmberg; sip@ietf.org
Subject: Re: [Sip] New version of draft-holmberg-sip-keep



Hi folks,

Christer has updated the document based on the action item from
Wednesday's SIP session (chaired by SIPPING chairs and modeled as the
proposed DISPATCH WG).  If folks could please review the document and
raise any concerns on the document now, that would be appreciated. We
ask specifically for an answer to the question:  Do you support the
document being adopted as a WG document in RAI (WG tbd)?  A simple "yes"
is fine, however, if you don't think we should adopt this document,
could you please provide the reason.

For your convenience, here's the link to the document:
http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt
And, the diff:
http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-holm=
b
erg-sip-keep-03.txt&url2=3Dhttp://tools.ietf.org/id/draft-holmberg-sip-ke=
e
p-04.txt
tinyurl: http://tinyurl.com/dfyvxz

Regards,
Mary
SIPPING WG co-chair


-----Original Message-----
From: sip-bounces@ietf.org on behalf of Christer Holmberg
Sent: Thu 3/26/2009 12:41 PM
To: sip@ietf.org
Subject: [Sip] New version of draft-holmberg-sip-keep


Hi,

I've submitted a new version (-04) of the keep draft.

The requirement text is now alligned with the text shown on the slides
during the SFO SIP WG presentation.

There are no other changes in the document from the previous version.

Regards,

Christer




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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>RE: [Sip] New version of =
draft-holmberg-sip-keep</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.3492" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D515171416-27032009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Yes</FONT></SPAN></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Mary =
Barnes<BR><B>Sent:</B> 26=20
March 2009 23:17<BR><B>To:</B> Christer Holmberg;=20
sip@ietf.org<BR><B>Subject:</B> Re: [Sip] New version of=20
draft-holmberg-sip-keep<BR></FONT><BR></DIV>
<DIV></DIV><!-- Converted from text/plain format -->
<P><FONT size=3D2>Hi folks,<BR><BR>Christer has updated the document =
based on the=20
action item from Wednesday's SIP session (chaired by SIPPING chairs and =
modeled=20
as the proposed DISPATCH WG).&nbsp; If folks could please review the =
document=20
and raise any concerns on the document now, that would be appreciated. =
We ask=20
specifically for an answer to the question:&nbsp; Do you support the =
document=20
being adopted as a WG document in RAI (WG tbd)?&nbsp; A simple "yes" is =
fine,=20
however, if you don't think we should adopt this document, could you =
please=20
provide the reason.<BR><BR>For your convenience, here's the link to the=20
document:<BR><A=20
href=3D"http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt">http://t=
ools.ietf.org/id/draft-holmberg-sip-keep-04.txt</A><BR>And,=20
the diff:<BR><A=20
href=3D"http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/dr=
aft-holmberg-sip-keep-03.txt&amp;url2=3Dhttp://tools.ietf.org/id/draft-ho=
lmberg-sip-keep-04.txt">http://tools.ietf.org//rfcdiff?url1=3Dhttp://tool=
s.ietf.org/id/draft-holmberg-sip-keep-03.txt&amp;url2=3Dhttp://tools.ietf=
.org/id/draft-holmberg-sip-keep-04.txt</A><BR>tinyurl:=20
<A=20
href=3D"http://tinyurl.com/dfyvxz">http://tinyurl.com/dfyvxz</A><BR><BR>R=
egards,<BR>Mary<BR>SIPPING=20
WG co-chair<BR><BR><BR>-----Original Message-----<BR>From: =
sip-bounces@ietf.org=20
on behalf of Christer Holmberg<BR>Sent: Thu 3/26/2009 12:41 PM<BR>To:=20
sip@ietf.org<BR>Subject: [Sip] New version of=20
draft-holmberg-sip-keep<BR><BR><BR>Hi,<BR><BR>I've submitted a new =
version (-04)=20
of the keep draft.<BR><BR>The requirement text is now alligned with the =
text=20
shown on the slides<BR>during the SFO SIP WG presentation.<BR><BR>There =
are no=20
other changes in the document from the previous=20
version.<BR><BR>Regards,<BR><BR>Christer<BR><BR></FONT></P></BODY></HTML>=


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From: "Francois Audet" <audet@nortel.com>
To: "Mary Barnes" <mary.barnes@nortel.com>, "Christer Holmberg" <christer.holmberg@ericsson.com>, <sip@ietf.org>
Subject: Re: [Sip] New version of draft-holmberg-sip-keep
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YES.


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
Barnes, Mary (RICH2:AR00)
	Sent: Thursday, March 26, 2009 16:17
	To: Christer Holmberg; sip@ietf.org
	Subject: Re: [Sip] New version of draft-holmberg-sip-keep
=09
=09

	Hi folks,
=09
	Christer has updated the document based on the action item from =
Wednesday's SIP session (chaired by SIPPING chairs and modeled as the =
proposed DISPATCH WG).  If folks could please review the document and =
raise any concerns on the document now, that would be appreciated. We =
ask specifically for an answer to the question:  Do you support the =
document being adopted as a WG document in RAI (WG tbd)?  A simple "yes" =
is fine, however, if you don't think we should adopt this document, =
could you please provide the reason.
=09
	For your convenience, here's the link to the document:
	http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt
	And, the diff:
	=
http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-holm=
berg-sip-keep-03.txt&url2=3Dhttp://tools.ietf.org/id/draft-holmberg-sip-k=
eep-04.txt
	tinyurl: http://tinyurl.com/dfyvxz
=09
	Regards,
	Mary
	SIPPING WG co-chair
=09
=09
	-----Original Message-----
	From: sip-bounces@ietf.org on behalf of Christer Holmberg
	Sent: Thu 3/26/2009 12:41 PM
	To: sip@ietf.org
	Subject: [Sip] New version of draft-holmberg-sip-keep
=09
=09
	Hi,
=09
	I've submitted a new version (-04) of the keep draft.
=09
	The requirement text is now alligned with the text shown on the slides
	during the SFO SIP WG presentation.
=09
	There are no other changes in the document from the previous version.
=09
	Regards,
=09
	Christer
=09
=09


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>RE: [Sip] New version of =
draft-holmberg-sip-keep</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2900.3492" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D187142716-27032009><FONT face=3DArial color=3D#800000 =

size=3D2>YES.</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Barnes, Mary=20
  (RICH2:AR00)<BR><B>Sent:</B> Thursday, March 26, 2009 =
16:17<BR><B>To:</B>=20
  Christer Holmberg; sip@ietf.org<BR><B>Subject:</B> Re: [Sip] New =
version of=20
  draft-holmberg-sip-keep<BR></FONT><BR></DIV>
  <DIV></DIV><!-- Converted from text/plain format -->
  <P><FONT size=3D2>Hi folks,<BR><BR>Christer has updated the document =
based on=20
  the action item from Wednesday's SIP session (chaired by SIPPING =
chairs and=20
  modeled as the proposed DISPATCH WG).&nbsp; If folks could please =
review the=20
  document and raise any concerns on the document now, that would be=20
  appreciated. We ask specifically for an answer to the question:&nbsp; =
Do you=20
  support the document being adopted as a WG document in RAI (WG =
tbd)?&nbsp; A=20
  simple "yes" is fine, however, if you don't think we should adopt this =

  document, could you please provide the reason.<BR><BR>For your =
convenience,=20
  here's the link to the document:<BR><A=20
  =
href=3D"http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt">http://t=
ools.ietf.org/id/draft-holmberg-sip-keep-04.txt</A><BR>And,=20
  the diff:<BR><A=20
  =
href=3D"http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/dr=
aft-holmberg-sip-keep-03.txt&amp;url2=3Dhttp://tools.ietf.org/id/draft-ho=
lmberg-sip-keep-04.txt">http://tools.ietf.org//rfcdiff?url1=3Dhttp://tool=
s.ietf.org/id/draft-holmberg-sip-keep-03.txt&amp;url2=3Dhttp://tools.ietf=
.org/id/draft-holmberg-sip-keep-04.txt</A><BR>tinyurl:=20
  <A=20
  =
href=3D"http://tinyurl.com/dfyvxz">http://tinyurl.com/dfyvxz</A><BR><BR>R=
egards,<BR>Mary<BR>SIPPING=20
  WG co-chair<BR><BR><BR>-----Original Message-----<BR>From:=20
  sip-bounces@ietf.org on behalf of Christer Holmberg<BR>Sent: Thu =
3/26/2009=20
  12:41 PM<BR>To: sip@ietf.org<BR>Subject: [Sip] New version of=20
  draft-holmberg-sip-keep<BR><BR><BR>Hi,<BR><BR>I've submitted a new =
version=20
  (-04) of the keep draft.<BR><BR>The requirement text is now alligned =
with the=20
  text shown on the slides<BR>during the SFO SIP WG =
presentation.<BR><BR>There=20
  are no other changes in the document from the previous=20
  =
version.<BR><BR>Regards,<BR><BR>Christer<BR><BR></FONT></P></BLOCKQUOTE><=
/BODY></HTML>

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Cc: sip@ietf.org, Mary Barnes <mary.barnes@nortel.com>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] New version of draft-holmberg-sip-keep
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Yes

>
>     ------------------------------------------------------------------------
>     *From:* sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] *On
>     Behalf Of *Barnes, Mary (RICH2:AR00)
>     *Sent:* Thursday, March 26, 2009 16:17
>     *To:* Christer Holmberg; sip@ietf.org
>     *Subject:* Re: [Sip] New version of draft-holmberg-sip-keep
>
>     Hi folks,
>
>     Christer has updated the document based on the action item from
>     Wednesday's SIP session (chaired by SIPPING chairs and modeled as
>     the proposed DISPATCH WG).  If folks could please review the
>     document and raise any concerns on the document now, that would be
>     appreciated. We ask specifically for an answer to the question: 
>     Do you support the document being adopted as a WG document in RAI
>     (WG tbd)?  A simple "yes" is fine, however, if you don't think we
>     should adopt this document, could you please provide the reason.
>
>     For your convenience, here's the link to the document:
>     http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt
>     And, the diff:
>     http://tools.ietf.org//rfcdiff?url1=http://tools.ietf.org/id/draft-holmberg-sip-keep-03.txt&url2=http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt
>     <http://tools.ietf.org//rfcdiff?url1=http://tools.ietf.org/id/draft-holmberg-sip-keep-03.txt&url2=http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt>
>     tinyurl: http://tinyurl.com/dfyvxz
>
>     Regards,
>     Mary
>     SIPPING WG co-chair
>
>
>     -----Original Message-----
>     From: sip-bounces@ietf.org on behalf of Christer Holmberg
>     Sent: Thu 3/26/2009 12:41 PM
>     To: sip@ietf.org
>     Subject: [Sip] New version of draft-holmberg-sip-keep
>
>
>     Hi,
>
>     I've submitted a new version (-04) of the keep draft.
>
>     The requirement text is now alligned with the text shown on the slides
>     during the SFO SIP WG presentation.
>
>     There are no other changes in the document from the previous version.
>
>     Regards,
>
>     Christer
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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Thread-Topic: [Sip] The problem with draft-kaplan-sip-secure-call-id-00
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Subject: Re: [Sip] The problem with draft-kaplan-sip-secure-call-id-00
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Jiri, =20
I agree with you on the importance of the problem, but I don't agree
with you that we can solve the problem by changing one of the protocol
basics at this time, with millions of existing boxes, network boxes as
well as end devices. Especially the end devices are critical. Most of
them send today the IP-address in the call-ID. Customers often do not
accept the upgrades. Some customers also have their own boxes. It will
take at least 10 years until we can can rely on the secure call-id for
dialog correlation. This is far too late. We also do not realy know
which problems may occur if we change the protocol basics.=20

I support the session-ID troubleshooting. It works fine and it works
now. For dialog matching we should work on a more general framework,
e.g. context-id or something else.

We can try secure call-id as a long term improvement of the protocol.
But we have to make sure that we do not cause problems for the existing
boxes and services which are up and running.      =20

 Laura


=20
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Jiri Kuthan
> Sent: Thursday, March 26, 2009 11:34 PM
> To: Hadriel Kaplan
> Cc: IETF SIP List
> Subject: Re: [Sip] The problem with draft-kaplan-sip-secure-call-id-00
>=20
> First of all I think it is definitely worth trying.
>=20
> Not being able to correlate dialogs through B2BUA is a big
> operational problem. I'm not sure that this will get really
> adopted (who knows what today's obfuscators choose to obfuscate
> tomorrow...). However in hope for solving this problem, presence of
> incentives for adoption and absence  of any known harm it is
> definitely worth doing.
>=20
> Another idea how to accommodate receivers that are silly not
> to be liberal is using a fake IP address (popular 0.0.0.0).
> While not aesthetic at all, it could accommodate such receivers
> (under assumption they assign no syntactical meaning to it)
> and signal to B2BUA there is no disclosure of IP addresses.
>=20
> OTOH it could be time too to abandon 2543 legacy. Supporting
> strict routing and other ancient stuff like IP in callid does
> not seem worth the trouble to me.
>=20
> -jiri
>=20
> Hadriel Kaplan wrote:
> > Ooh, good point.  Fair enough, I'll change that.
> >=20
> > -hadriel
> >=20
> >> -----Original Message-----
> >> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]=20
> On Behalf Of
> >> Jonathan Rosenberg
> >> Sent: Wednesday, March 25, 2009 4:51 PM
> >> To: IETF SIP List
> >> Subject: [Sip] The problem with draft-kaplan-sip-secure-call-id-00
> >>
> >> I didnt get a chance to say this at the mike.
> >>
> >> You *cannot* change the BNF for the call-id. If you did,=20
> you might end
> >> up with the following problem:
> >>
> >> A new UA gets built, compliant to this update. It is one of these
> >> 'aggressive parsing' UAs. It rejects any message which is=20
> not compliant
> >> to the BNF. An older, normal RFC-3261 compliant UA sends a=20
> call-id which
> >> includes the @-sign.
> >>
> >> According to the BNF of this updated 3261, and based on=20
> the strictness
> >> of this aggressive-parsing UAs, it rejects that request as=20
> non-compliant.
> >>
> >> As such, you need to keep the BNF, but change the=20
> normative language
> >> such that an endpoint MUST NOT include the [@ host] but=20
> MUST be prepared
> >> to receive it.
> >>
> >> -Jonathan R.
> >>
> >> --
> >> Jonathan D. Rosenberg, Ph.D.                   111 Wood=20
> Avenue South
> >> Cisco Fellow                                   Iselin, NJ 08830
> >> Cisco, Voice Technology Group
> >> jdrosen@cisco.com
> >> http://www.jdrosen.net                         PHONE:=20
> (408) 902-3084
> >> http://www.cisco.com
> >> _______________________________________________
> >> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol
> >> Use sip-implementors@cs.columbia.edu for questions on current sip
> >> Use sipping@ietf.org for new developments on the application of sip
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >=20
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

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Cc: sip@ietf.org, christer.holmberg@ericsson.com
Subject: Re: [Sip] New version of draft-holmberg-sip-keep
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yes

=20

________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Mary Barnes
	Sent: 26 March 2009 23:17
	To: Christer Holmberg; sip@ietf.org
	Subject: Re: [Sip] New version of draft-holmberg-sip-keep
=09
=09

	Hi folks,
=09
	Christer has updated the document based on the action item from
Wednesday's SIP session (chaired by SIPPING chairs and modeled as the
proposed DISPATCH WG).  If folks could please review the document and
raise any concerns on the document now, that would be appreciated. We
ask specifically for an answer to the question:  Do you support the
document being adopted as a WG document in RAI (WG tbd)?  A simple "yes"
is fine, however, if you don't think we should adopt this document,
could you please provide the reason.
=09
	For your convenience, here's the link to the document:
	http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt
	And, the diff:
=09
http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/draft-holm=
b
erg-sip-keep-03.txt&url2=3Dhttp://tools.ietf.org/id/draft-holmberg-sip-ke=
e
p-04.txt
	tinyurl: http://tinyurl.com/dfyvxz
=09
	Regards,
	Mary
	SIPPING WG co-chair
=09
=09
	-----Original Message-----
	From: sip-bounces@ietf.org on behalf of Christer Holmberg
	Sent: Thu 3/26/2009 12:41 PM
	To: sip@ietf.org
	Subject: [Sip] New version of draft-holmberg-sip-keep
=09
=09
	Hi,
=09
	I've submitted a new version (-04) of the keep draft.
=09
	The requirement text is now alligned with the text shown on the
slides
	during the SFO SIP WG presentation.
=09
	There are no other changes in the document from the previous
version.
=09
	Regards,
=09
	Christer
=09
=09


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>RE: [Sip] New version of =
draft-holmberg-sip-keep</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.3492" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D058154617-27032009>yes</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT><BR>&nbsp;</DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Mary =
Barnes<BR><B>Sent:</B>=20
  26 March 2009 23:17<BR><B>To:</B> Christer Holmberg;=20
  sip@ietf.org<BR><B>Subject:</B> Re: [Sip] New version of=20
  draft-holmberg-sip-keep<BR></FONT><BR></DIV>
  <DIV></DIV><!-- Converted from text/plain format -->
  <P><FONT size=3D2>Hi folks,<BR><BR>Christer has updated the document =
based on=20
  the action item from Wednesday's SIP session (chaired by SIPPING =
chairs and=20
  modeled as the proposed DISPATCH WG).&nbsp; If folks could please =
review the=20
  document and raise any concerns on the document now, that would be=20
  appreciated. We ask specifically for an answer to the question:&nbsp; =
Do you=20
  support the document being adopted as a WG document in RAI (WG =
tbd)?&nbsp; A=20
  simple "yes" is fine, however, if you don't think we should adopt this =

  document, could you please provide the reason.<BR><BR>For your =
convenience,=20
  here's the link to the document:<BR><A=20
  =
href=3D"http://tools.ietf.org/id/draft-holmberg-sip-keep-04.txt">http://t=
ools.ietf.org/id/draft-holmberg-sip-keep-04.txt</A><BR>And,=20
  the diff:<BR><A=20
  =
href=3D"http://tools.ietf.org//rfcdiff?url1=3Dhttp://tools.ietf.org/id/dr=
aft-holmberg-sip-keep-03.txt&amp;url2=3Dhttp://tools.ietf.org/id/draft-ho=
lmberg-sip-keep-04.txt">http://tools.ietf.org//rfcdiff?url1=3Dhttp://tool=
s.ietf.org/id/draft-holmberg-sip-keep-03.txt&amp;url2=3Dhttp://tools.ietf=
.org/id/draft-holmberg-sip-keep-04.txt</A><BR>tinyurl:=20
  <A=20
  =
href=3D"http://tinyurl.com/dfyvxz">http://tinyurl.com/dfyvxz</A><BR><BR>R=
egards,<BR>Mary<BR>SIPPING=20
  WG co-chair<BR><BR><BR>-----Original Message-----<BR>From:=20
  sip-bounces@ietf.org on behalf of Christer Holmberg<BR>Sent: Thu =
3/26/2009=20
  12:41 PM<BR>To: sip@ietf.org<BR>Subject: [Sip] New version of=20
  draft-holmberg-sip-keep<BR><BR><BR>Hi,<BR><BR>I've submitted a new =
version=20
  (-04) of the keep draft.<BR><BR>The requirement text is now alligned =
with the=20
  text shown on the slides<BR>during the SFO SIP WG =
presentation.<BR><BR>There=20
  are no other changes in the document from the previous=20
  =
version.<BR><BR>Regards,<BR><BR>Christer<BR><BR></FONT></P></BLOCKQUOTE><=
/BODY></HTML>

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From adam@nostrum.com  Fri Mar 27 13:19:06 2009
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Cc: SIP <sip@ietf.org>
Subject: Re: [Sip] I-D Inaction: draft-kaplan-sip-secure-call-id-00.txt
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Hadriel Kaplan wrote:
> Sure, SBC's may not follow this draft. So? UA's and other devices may 
> not follow any given draft/RFC. But right now there's actually a 
> reasonably defendable rationale for why the Call-ID's should be 
> changed. I'm trying to get rid of that rationale. The owners of the 
> SBC's will then be able to tell their vendors "go do this draft", if 
> they find they need SIP extensions to work which rely on the Call-ID 
> not changing. The market will sort out if that's valuable or not.

Which is why an informational draft explaining the problem is useful, 
while a normative document... not quite so much.

Tell UACs: "If you include machine names in your Call-IDs, B2BUAs will 
have reason to change them." There's no MUST or MUST NOT here, just an 
explanation of rationale.

Tell B2BUAs: "If a Call-ID doesn't contain an @ sign, you'll do everyone 
a favor by not changing them, because it allows proper correlation of 
related transactions an dialogs." Again, no MUST; no MUST NOT. Just 
cause and effect.

Then see if people want to play nice. The market will sort out if that's 
valuable or not.

/a

From drage@alcatel-lucent.com  Fri Mar 27 18:53:39 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: "sip@ietf.org" <sip@ietf.org>
Date: Sat, 28 Mar 2009 02:54:24 +0100
Thread-Topic: Completion of draft-ietf-sip-ua-privacy-06
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Subject: [Sip] Completion of draft-ietf-sip-ua-privacy-06
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(As WG chair)

This document can be found at:

http://www.ietf.org/internet-drafts/draft-ietf-sip-ua-privacy-06.txt

I am assuming we are pretty much done with this document and therefore will=
 proceed with writing up the publication request.

If anyone thinks otherwise, now would be a very appropriate time to make th=
at position known to the list.

Expect some editorials as a result of me writing this up.

regards

Keith=

From HKaplan@acmepacket.com  Fri Mar 27 22:18:30 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Adam Roach <adam@nostrum.com>
Date: Sat, 28 Mar 2009 01:19:24 -0400
Thread-Topic: [Sip] I-D Inaction: draft-kaplan-sip-secure-call-id-00.txt
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Yeah I've been thinking about what you said the other day about this, and I=
 think I understand your point, but I still think MUST's and an update to 3=
261 would be good, for the following reasons:

1) We really should want to change 3261 to say MUST NOT put an ip-address/h=
ostname in the call-id.  Really in 20/20 hindsight I don't think it should =
have had one to begin with.

2) It's easier for customers to force vendors to do something if there's a =
MUST statement, standards-track RFC, vs. an informational recommendation/sh=
ould.

3) Endpoint UAC vendors should do it *before* getting deployed and being to=
ld by customers to change after-wards.  Specifically I'm thinking about ven=
dors who don't attend IETF and aren't that up-to-speed on all the nuances a=
nd such.  An update to 3261 with normative mandatory language is clearer, I=
 think.

-hadriel


> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com]
> Sent: Friday, March 27, 2009 4:20 PM
>=20
> Which is why an informational draft explaining the problem is useful,
> while a normative document... not quite so much.
>=20
> Tell UACs: "If you include machine names in your Call-IDs, B2BUAs will
> have reason to change them." There's no MUST or MUST NOT here, just an
> explanation of rationale.
>=20
> Tell B2BUAs: "If a Call-ID doesn't contain an @ sign, you'll do everyone
> a favor by not changing them, because it allows proper correlation of
> related transactions an dialogs." Again, no MUST; no MUST NOT. Just
> cause and effect.
>=20
> Then see if people want to play nice. The market will sort out if that's
> valuable or not.
>=20
> /a

From HKaplan@acmepacket.com  Fri Mar 27 22:24:09 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: SIP <sip@ietf.org>
Date: Sat, 28 Mar 2009 01:25:03 -0400
Thread-Topic: B2BUA's changing call-id's: draft-kaplan-sip-secure-call-id-00.txt
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Subject: [Sip] B2BUA's changing call-id's: draft-kaplan-sip-secure-call-id-00.txt
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Howdy,
In talking with some folks about this draft afterwards, I think some of the=
 consternation around the draft is due to the language around mandating B2B=
UA's not to change the Call-ID.  I thought it was clear that I only meant f=
or the purposes of security/privacy, not for other purposes - in other word=
s, if you have some reason you need to change the Call-ID other than privac=
y, then the must-not-change doesn't apply.

I think to alleviate the concern I'll just remove the language around that,=
 and change it to more of just a discussion for that topic - with no normat=
ive requirements for B2BUA's, other than that if they do change the Call-ID=
, they should act as a UAC and create one that doesn't contain an ip/host p=
ortion.

Would people prefer that?

-hadriel

From mdolly@att.com  Sat Mar 28 09:27:42 2009
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Thread-Topic: [Sip] I-D Inaction: draft-kaplan-sip-secure-call-id-00.txt
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From: "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>, "Adam Roach" <adam@nostrum.com>
Cc: SIP <sip@ietf.org>
Subject: Re: [Sip] I-D Inaction: draft-kaplan-sip-secure-call-id-00.txt
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I agree. Lack there of, only means the IETF does not want an
interoperable solution

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Hadriel Kaplan
Sent: Saturday, March 28, 2009 1:19 AM
To: Adam Roach
Cc: SIP
Subject: Re: [Sip] I-D Inaction: draft-kaplan-sip-secure-call-id-00.txt


Yeah I've been thinking about what you said the other day about this,
and I think I understand your point, but I still think MUST's and an
update to 3261 would be good, for the following reasons:

1) We really should want to change 3261 to say MUST NOT put an
ip-address/hostname in the call-id.  Really in 20/20 hindsight I don't
think it should have had one to begin with.

2) It's easier for customers to force vendors to do something if there's
a MUST statement, standards-track RFC, vs. an informational
recommendation/should.

3) Endpoint UAC vendors should do it *before* getting deployed and being
told by customers to change after-wards.  Specifically I'm thinking
about vendors who don't attend IETF and aren't that up-to-speed on all
the nuances and such.  An update to 3261 with normative mandatory
language is clearer, I think.

-hadriel


> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com]
> Sent: Friday, March 27, 2009 4:20 PM
>=20
> Which is why an informational draft explaining the problem is useful,
> while a normative document... not quite so much.
>=20
> Tell UACs: "If you include machine names in your Call-IDs, B2BUAs will
> have reason to change them." There's no MUST or MUST NOT here, just an
> explanation of rationale.
>=20
> Tell B2BUAs: "If a Call-ID doesn't contain an @ sign, you'll do
everyone
> a favor by not changing them, because it allows proper correlation of
> related transactions an dialogs." Again, no MUST; no MUST NOT. Just
> cause and effect.
>=20
> Then see if people want to play nice. The market will sort out if
that's
> valuable or not.
>=20
> /a
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

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From: "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>
To: <hgs@cs.columbia.edu>, <acooper@cdt.org>, <sip@ietf.org>
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From jiri@iptel.org  Sat Mar 28 14:02:37 2009
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Date: Sat, 28 Mar 2009 21:02:29 +0100
From: Jiri Kuthan <jiri@iptel.org>
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Cc: Francois Audet <audet@nortel.com>
Subject: [Sip] francois' comments and why RFC4474 not used in the field
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As we were running out of time in the meeting, I wanted to
follow up on an argument which Francois has made: RFC 4474
works if the identity provider and verifiers are "next to
each other", thus we have no problem with this RFC.

/* hope I didn't misinterpret, let me know should that be
the case */

I'm worried this is only a wishful thinking. While perfectly
logical, still even in such constrained setups some bizzar
ALGs do in my experience appear in the middle, change SDP
and make thus the identity worthless.

I think the bottom line is the protocol is not really end-to-end.
It makes quite some assumptions about the quality of underlying
network (particularly about not changing SDPs) as opposed to
trying to deal with lack of it. In fact, SDP payloads are
changed at quite a number of devices: proxy servers, B2BUAs,
SBCs, you name it. Assuming there is nothing in the way
which won't change SDP is just NOT ROBUST.

Other way to look at it is it is function overloading. You ask
for identity and get message integrity -- consequently when
the latter fails, the former, which you really initially wanted,
fails too. ekr's argument is certainly right that a message
of certain origin but uncertain content is close to worthless.
OTOH, the particular piece, IP address and port numbers in SDP
are certainly not to be covered by a message integrity checks.
In fact, for accepting calls based on who is calling, as
a minimalistic application example, one does  not need
message integrity.

In conclusion, I do not think that RFC4474 can be of any help
(other than in controlled environments which beats its purpose)
as long as IP/port in SDP are message-integrity-protected.

-jiri


From jiri@iptel.org  Sat Mar 28 17:55:25 2009
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From: Jiri Kuthan <jiri@iptel.org>
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References: <49CA9943.1020808@cisco.com><E6C2E8958BA59A4FB960963D475F7AC314FD693ECB@mail> <49CC02F4.6070507@iptel.org> <40FB0FFB97588246A1BEFB05759DC8A002CA7355@S4DE9JSAANI.ost.t-com.de>
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Subject: Re: [Sip] The problem with draft-kaplan-sip-secure-call-id-00
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Interestingly I made a measurement which supports your argument.
In short, a vast majority of SIP traffic going through iptel.org
has IP address in it. Details of the measurement follow bellow
for those interested in details (including links to colorful
non-ASCII graphs ;-)). Key characteristics of the  iptel.org
SIP service is high variance in type of UAs using it.

As for the backwards-compatibility argument, I guess that it will
take same long time to get UAs do session-id, so IMO an improvement
is only possible under an assumption that network-asserted session-id
is easier to introduce than UA-asserted, right? That appears
meaningful to me.

-jiri (hoping for better life for unfortunate ones, who have chosen
to operate callid-mangling B2BUAs and RFC-2543 clients without upgrade
strategy  ;-))

Measurement Details
--------------------
a) types of UA
for a pie-chart see: http://www.iptel.org/~jiri/etc/ua-breakdown.png
(picture to big to be attached, sorry)
It is saying that there are 162 diffent types of User Agents for
1131 users. Apart from the most popular UA types seen in the graph,
the greatest variety occurs in the "other devices" chunk representing
150+ User Agent Types.
In summary I think the selection of UAs is quite representative.

b) callid stats
I pulled about 18 minutes of SIP traffic at about 0.5 Mbps (including
layer-2 headers; resulting in about 65e3 packets totaling to 35e6
bytes of SIP data, avg SIP packet size 540 bytes) and postprocessed
it as follows:

# IP addres in grep -v is an availabaility checker -- to be excluded
ngrep -d any -W byline  port 5060| grep -i Call-ID|grep -v 
"213.192.59.93"|sort -u >/tmp/callid2
# number of callids captured in the sample: 4345
tlb1:~# wc /tmp/callid2
   4345   8733 184947 /tmp/callid2
# number of @IP callids: 3695 (85%)
tlb1:~# egrep "@[0-9]{1,3}\.[0-9]{1,3}\.[0-9]{1,3}\.[0-9]{1,3}" 
/tmp/callid2|wc
3695    7433  152577
# number of callids w/o @: 470 (11%)
tlb1:~# egrep -v "@" /tmp/callid2|wc
     470     940   23053
# numver of callids with @ and some non-nummerical stuff after it 
(domain name typically)
tlb1:~# egrep  "@[^0-9]" /tmp/callid2|wc
     149     298    8220
# number of "natted callids"
tlb1:~# egrep  "@(10|192\.168|172\.(16|17|18|19|2[0-9]|30|31))\." 
/tmp/callid2|wc
    1092    2184   47063


c) measurement mistakes: a and b have been measured independently, it
is possible but unlikely that the statements in b related to a different
set of UAs featuring  a different variance. This likelihood of such
circumstances to occur doesn't seem very high to me. b) has been measured
on an off-the shelf PC attached to the same Ethernet swtich at iptel.org's
load-balancer --it is possible that such a PC could have missed some
IP packets. If so, I assume it has done so on a fair basis which does not
impact the CallId assertion.

L.Liess@telekom.de wrote:
> Jiri,  
> I agree with you on the im portance of the problem, but I don't agree
> with you that we can solve the problem by changing one of the protocol
> basics at this time, with millions of existing boxes, network boxes as
> well as end devices. Especially the end devices are critical. Most of
> them send today the IP-address in the call-ID. Customers often do not
> accept the upgrades. Some customers also have their own boxes. It will
> take at least 10 years until we can can rely on the secure call-id for
> dialog correlation. This is far too late. We also do not realy know
> which problems may occur if we change the protocol basics. 
> 
> I support the session-ID troubleshooting. It works fine and it works
> now. For dialog matching we should work on a more general framework,
> e.g. context-id or something else.
> 
> We can try secure call-id as a long term improvement of the protocol.
> But we have to make sure that we do not cause problems for the existing
> boxes and services which are up and running.       
> 
>  Laura
> 
> 
>  
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
>> Behalf Of Jiri Kuthan
>> Sent: Thursday, March 26, 2009 11:34 PM
>> To: Hadriel Kaplan
>> Cc: IETF SIP List
>> Subject: Re: [Sip] The problem with draft-kaplan-sip-secure-call-id-00
>>
>> First of all I think it is definitely worth trying.
>>
>> Not being able to correlate dialogs through B2BUA is a big
>> operational problem. I'm not sure that this will get really
>> adopted (who knows what today's obfuscators choose to obfuscate
>> tomorrow...). However in hope for solving this problem, presence of
>> incentives for adoption and absence  of any known harm it is
>> definitely worth doing.
>>
>> Another idea how to accommodate receivers that are silly not
>> to be liberal is using a fake IP address (popular 0.0.0.0).
>> While not aesthetic at all, it could accommodate such receivers
>> (under assumption they assign no syntactical meaning to it)
>> and signal to B2BUA there is no disclosure of IP addresses.
>>
>> OTOH it could be time too to abandon 2543 legacy. Supporting
>> strict routing and other ancient stuff like IP in callid does
>> not seem worth the trouble to me.
>>
>> -jiri
>>
>> Hadriel Kaplan wrote:
>>> Ooh, good point.  Fair enough, I'll change that.
>>>
>>> -hadriel
>>>
>>>> -----Original Message-----
>>>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] 
>> On Behalf Of
>>>> Jonathan Rosenberg
>>>> Sent: Wednesday, March 25, 2009 4:51 PM
>>>> To: IETF SIP List
>>>> Subject: [Sip] The problem with draft-kaplan-sip-secure-call-id-00
>>>>
>>>> I didnt get a chance to say this at the mike.
>>>>
>>>> You *cannot* change the BNF for the call-id. If you did, 
>> you might end
>>>> up with the following problem:
>>>>
>>>> A new UA gets built, compliant to this update. It is one of these
>>>> 'aggressive parsing' UAs. It rejects any message which is 
>> not compliant
>>>> to the BNF. An older, normal RFC-3261 compliant UA sends a 
>> call-id which
>>>> includes the @-sign.
>>>>
>>>> According to the BNF of this updated 3261, and based on 
>> the strictness
>>>> of this aggressive-parsing UAs, it rejects that request as 
>> non-compliant.
>>>> As such, you need to keep the BNF, but change the 
>> normative language
>>>> such that an endpoint MUST NOT include the [@ host] but 
>> MUST be prepared
>>>> to receive it.
>>>>
>>>> -Jonathan R.
>>>>
>>>> --
>>>> Jonathan D. Rosenberg, Ph.D.                   111 Wood 
>> Avenue South
>>>> Cisco Fellow                                   Iselin, NJ 08830
>>>> Cisco, Voice Technology Group
>>>> jdrosen@cisco.com
>>>> http://www.jdrosen.net                         PHONE: 
>> (408) 902-3084
>>>> http://www.cisco.com
>>>> _______________________________________________
>>>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol
>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>> Use sipping@ietf.org for new developments on the application of sip
>>> _______________________________________________
>>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
>>>
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
> 

From AUDET@nortel.com  Sat Mar 28 21:32:29 2009
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Thread-Topic: francois' comments and why RFC4474 not used in the field
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References: <49CE8255.4000103@iptel.org>
From: "Francois Audet" <audet@nortel.com>
To: "Jiri Kuthan" <jiri@iptel.org>, <sip@ietf.org>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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I think our views are not that appart.

See belows. =20

> -----Original Message-----
> From: Jiri Kuthan [mailto:jiri@iptel.org]=20
> Sent: Saturday, March 28, 2009 13:02
> To: sip@ietf.org
> Cc: Audet, Francois (SC100:3055)
> Subject: francois' comments and why RFC4474 not used in the field
>=20
> As we were running out of time in the meeting, I wanted to=20
> follow up on an argument which Francois has made: RFC 4474=20
> works if the identity provider and verifiers are "next to=20
> each other", thus we have no problem with this RFC.
>=20
> /* hope I didn't misinterpret, let me know should that be the case */

First part is currect, but I didn't say "thus we have no problem with
this RFC."

As you say, it only works if the identity provider and the verifier are
"next to each other". A federation of Enterprises for example. Or =
between
service providers.

However, one of the most predominant model today doesn't work like this.
A domain is often relying on a third party, a "service provider".

So it's really not "end-to-end". That was the point I was trying to =
make.

RFC 4474 appears to work fine for end-to-end identity: unfortunately,
many deployment models (perhaps the majority) and in fact, not =
end-to-end,
but mediated. In a typical scenario, each end would have it's own
"service provider", so 2 intermediaries.

> I'm worried this is only a wishful thinking. While perfectly=20
> logical, still even in such constrained setups some bizzar=20
> ALGs do in my experience appear in the middle, change SDP and=20
> make thus the identity worthless.

Yes, I agree 4474 doesn't work if there are intermidiaries.

> I think the bottom line is the protocol is not really end-to-end.

EXACTLY!


> It makes quite some assumptions about the quality of=20
> underlying network (particularly about not changing SDPs) as=20
> opposed to trying to deal with lack of it. In fact, SDP=20
> payloads are changed at quite a number of devices: proxy=20
> servers, B2BUAs, SBCs, you name it. Assuming there is nothing=20
> in the way which won't change SDP is just NOT ROBUST.
>=20
> Other way to look at it is it is function overloading. You=20
> ask for identity and get message integrity -- consequently=20
> when the latter fails, the former, which you really initially=20
> wanted, fails too. ekr's argument is certainly right that a=20
> message of certain origin but uncertain content is close to worthless.
> OTOH, the particular piece, IP address and port numbers in=20
> SDP are certainly not to be covered by a message integrity checks.
> In fact, for accepting calls based on who is calling, as a=20
> minimalistic application example, one does  not need message=20
> integrity.
>=20
> In conclusion, I do not think that RFC4474 can be of any help=20
> (other than in controlled environments which beats its=20
> purpose) as long as IP/port in SDP are message-integrity-protected.

Here is what I think:

RFC 4474 can work in the following situations:

1 - Enterprise-to-Enterprise: i.e., the network as a dumb pipe. The =
enterprises
have their own SBCs modifying SIP/SDP (or they can use ICE).

2 - Subscriber on Service provider A to Subscriber on service provider =
B:=20
i.e., the "domain" for the first subscriber is actually that of the =
subscriber.
This can work for "residential users" and small enterprises.

3 - Any permutation of the above.

In all those cases, there is effectively end-to-end identity.

RFC 4474 does NOT work if it's not end-to-end. For example, if=20
the "Enterprise" has it's own domain, but it relies on it's=20
service provider to perform a number of functions. NAT traversal,
topology hiding, etc.

So, what can we do?  I see 2 possibilities:

1 - Make it end-to-end. This means tackling one by one the problem =
areas.
    Use TURN/ICE instead for relying on SDP payload manipulation. Use
    Call-ID and Contact headers that don't reveal topology, etc.

2 - Create a non end-to-end Identity assertion mechanism that is =
different
    from RFC 4474 and more secure than P-Asserted-Identity. I have a=20
    feeling that it won't be trivial. We may have to define the=20
    trust relationship between the ends and the intermediaries.=20

What I was trying to say in the meeting is that for "end-to-end
identity", 4474 appears to work. But if the problem we are trying to
solve is NOT end-to-end identity, but strong mediated identity, then,
let's just say so. It's just a DIFFERENT problem to solve.


From dwing@cisco.com  Sat Mar 28 22:30:57 2009
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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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> However, one of the most predominant model today doesn't work 
> like this.
> A domain is often relying on a third party, a "service provider".

I observed Hadriel's presentation from the very back of the
room.  My interpretation is that we lack consensus that there
are intermediate service providers.

Until consensus on that point is reached everything else is
for naught.

-d


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On Sat, Mar 28, 2009 at 6:25 AM, Hadriel Kaplan <HKaplan@acmepacket.com> wrote:

> Would people prefer that?

yup.

Also:

   [OPEN ISSUE: should we make this a fixed format, defined
   randomization mechanism to avoid detecting the vendor type from the
   Call-ID format?]

How about defining it to be a v5 uuid (rfc4122), which:

 - is well defined (including generating) in RFC 4122
 - has many libraries for generating and parsing
 - a v5 uuid can be detected (by looking at bits 4-7 == 0101) and thus
knows if it's secure
 - is "almost" unique across a four dimensional space-time continuum -
certainly a _very_ low chance of any collision, even when we're
communicating with alpha centauri over SIP/2.0.

 ~ Theo

Sent from: Bicester Oxfordshire United Kingdom.

From theo@crazygreek.co.uk  Sun Mar 29 04:41:31 2009
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On Thu, Mar 26, 2009 at 7:06 PM, DOLLY, MARTIN C, ATTLABS
<mdolly@att.com> wrote:
> We support this draft as well

I also support this draft.

 ~ Theo

Sent from: Bicester Oxfordshire United Kingdom.

From theo@crazygreek.co.uk  Sun Mar 29 04:58:23 2009
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Subject: Re: [Sip] The problem with draft-kaplan-sip-secure-call-id-00
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On Sun, Mar 29, 2009 at 1:56 AM, Jiri Kuthan <jiri@iptel.org> wrote:

> Interestingly I made a measurement which supports your argument.
> In short, a vast majority of SIP traffic going through iptel.org
> has IP address in it.

Yup, i have come to the same conclusion with our customers, too.

 ~ Theo

From jiri@iptel.org  Sun Mar 29 06:37:11 2009
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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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Francois Audet wrote:
> I think our views are not that appart.
> 
> See belows.  
> 
>> -----Original Message-----
>> From: Jiri Kuthan [mailto:jiri@iptel.org] 
>> Sent: Saturday, March 28, 2009 13:02
>> To: sip@ietf.org
>> Cc: Audet, Francois (SC100:3055)
>> Subject: francois' comments and why RFC4474 not used in the field
>>
>> As we were running out of time in the meeting, I wanted to 
>> follow up on an argument which Francois has made: RFC 4474 
>> works if the identity provider and verifiers are "next to 
>> each other", thus we have no problem with this RFC.
>>
>> /* hope I didn't misinterpret, let me know should that be the case */
> 
> First part is currect, but I didn't say "thus we have no problem with
> this RFC."
> 
> As you say, it only works if the identity provider and the verifier are
> "next to each other". A federation of Enterprises for example. Or between
> service providers.
> 
> However, one of the most predominant model today doesn't work like this.
> A domain is often relying on a third party, a "service provider".
> 
> So it's really not "end-to-end". That was the point I was trying to make.
> 
> RFC 4474 appears to work fine for end-to-end identity: unfortunately,
> many deployment models (perhaps the majority) and in fact, not end-to-end,
> but mediated. In a typical scenario, each end would have it's own
> "service provider", so 2 intermediaries.
> 
>> I'm worried this is only a wishful thinking. While perfectly 
>> logical, still even in such constrained setups some bizzar 
>> ALGs do in my experience appear in the middle, change SDP and 
>> make thus the identity worthless.
> 
> Yes, I agree 4474 doesn't work if there are intermidiaries.
> 
>> I think the bottom line is the protocol is not really end-to-end.
> 
> EXACTLY!
> 
> 
>> It makes quite some assumptions about the quality of 
>> underlying network (particularly about not changing SDPs) as 
>> opposed to trying to deal with lack of it. In fact, SDP 
>> payloads are changed at quite a number of devices: proxy 
>> servers, B2BUAs, SBCs, you name it. Assuming there is nothing 
>> in the way which won't change SDP is just NOT ROBUST.
>>
>> Other way to look at it is it is function overloading. You 
>> ask for identity and get message integrity -- consequently 
>> when the latter fails, the former, which you really initially 
>> wanted, fails too. ekr's argument is certainly right that a 
>> message of certain origin but uncertain content is close to worthless.
>> OTOH, the particular piece, IP address and port numbers in 
>> SDP are certainly not to be covered by a message integrity checks.
>> In fact, for accepting calls based on who is calling, as a 
>> minimalistic application example, one does  not need message 
>> integrity.
>>
>> In conclusion, I do not think that RFC4474 can be of any help 
>> (other than in controlled environments which beats its 
>> purpose) as long as IP/port in SDP are message-integrity-protected.

Thanks for the clarifications. I think we are on the same page then.

> 
> Here is what I think:
> 
> RFC 4474 can work in the following situations:
> 
> 1 - Enterprise-to-Enterprise: i.e., the network as a dumb pipe. The enterprises
> have their own SBCs modifying SIP/SDP (or they can use ICE).
> 
> 2 - Subscriber on Service provider A to Subscriber on service provider B: 
> i.e., the "domain" for the first subscriber is actually that of the subscriber.
> This can work for "residential users" and small enterprises.
> 
> 3 - Any permutation of the above.
> 
> In all those cases, there is effectively end-to-end
> 
> RFC 4474 does NOT work if it's not end-to-end. For example, if 
> the "Enterprise" has it's own domain, but it relies on it's 
> service provider to perform a number of functions. NAT traversal,
> topology hiding, etc.
> 
> So, what can we do?  I see 2 possibilities:
> 
> 1 - Make it end-to-end. This means tackling one by one the problem areas.
>     Use TURN/ICE instead for relying on SDP payload manipulation. Use
>     Call-ID and Contact headers that don't reveal topology, etc.
> 
> 2 - Create a non end-to-end Identity assertion mechanism that is different
>     from RFC 4474 and more secure than P-Asserted-Identity. I have a 
>     feeling that it won't be trivial. We may have to define the 
>     trust relationship between the ends and the intermediaries. 
> 
> What I was trying to say in the meeting is that for "end-to-end
> identity", 4474 appears to work. But if the problem we are trying to
> solve is NOT end-to-end identity, but strong mediated identity, then,
> let's just say so. It's just a DIFFERENT problem to solve.

Well, I'm afraid that 1 is too complicated to work. Bringing TURN/ICE to
work is already quite an enterprise, and as much success I wish to it,
I would be careful and avoid dependency on it (which is generally
a good protocol design practice anyhow).

2 is then probably the safer direction to me. The point is really that
RFC4474 is not having as broad applicability as we have hoped, and
leaving SIP without Identity is going to be real-time irritating,
compared to convenience of Email's off-line irritation.

Some of the RFC 4474 applicability limitations could be lifted by
reducing its integrity protection scope. I'm not sure how popular
will it be then (some have had different opinions on adoptability
of things relying on CA) but it  seems worth trying.

I would not dare to restrain ourselves to fixing RFC4474 though.
I'm very much in favor of testing other alternatives. IETF has
had a very successful tradition of testing several approaches.
A more ellaborated email on the topic was BTW recently written
by Dave Crocker: 
http://www.ietf.org/mail-archive/web/mmox/current/msg01233.html

Obviously DERIVE (or perhaps non-SIP based reverse verifications)
appears a feasible alternative to me (because  easy to deploy and
offering incentives to do so), but given the interest it is quite
certain there will be more different alternatives.


-jiri

From jiri@iptel.org  Sun Mar 29 06:52:30 2009
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Cc: sip@ietf.org, 'Francois Audet' <audet@nortel.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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Dan Wing wrote:
>> However, one of the most predominant model today doesn't work 
>> like this.
>> A domain is often relying on a third party, a "service provider".
> 
> I observed Hadriel's presentation from the very back of the
> room.  My interpretation is that we lack consensus that there
> are intermediate service providers.
> 
> Until consensus on that point is reached everything else is
> for naught.

I'm worried that "intermediate service providers" is a sort of
scenario which is underplaying the RFC4474 flaw.

The key problem is that RFC4474 is not robust. It ties likely-to-fail
integrity to identity, resulting in identity broken. Integrity is
likely to fail, because regardless how close the assertion party
is to the verification party, some sort of SDP-changer will be
there. Maybe "intermediate service provider" or an ALG. One can
argue that ALG has nothing to do there (quite rightfully so)
but relying on such not to be there is just way too brittle.

So I don't think the problem is we are missing a definition of
intermediate service provider. I think we are missing identity
vehicles that have robustness to overcome all sorts of middleboxes.

-jiri


> 
> -d
> 

From dean.willis@softarmor.com  Sun Mar 29 20:52:57 2009
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Subject: Re: [Sip] New version of draft-holmberg-sip-keep
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On Mar 26, 2009, at 6:16 PM, Mary Barnes wrote:

> Hi folks,
>
> Christer has updated the document based on the action item from  
> Wednesday's SIP session (chaired by SIPPING chairs and modeled as  
> the proposed DISPATCH WG).  If folks could please review the  
> document and raise any concerns on the document now, that would be  
> appreciated. We ask specifically for an answer to the question:  Do  
> you support the document being adopted as a WG document in RAI (WG  
> tbd)?  A simple "yes" is fine, however, if you don't think we should  
> adopt this document, could you please provide the reason.
>
>
>

So, given the outpouring of support, should we pull back the draft- 
ietf-sip-outbound and re-write it to refer to the keepalive mechanism  
of this draft instead of containing similar but not identical text?

--
Dean

From dean.willis@softarmor.com  Sun Mar 29 20:54:51 2009
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On Mar 28, 2009, at 12:19 AM, Hadriel Kaplan wrote:

>
> Yeah I've been thinking about what you said the other day about  
> this, and I think I understand your point, but I still think MUST's  
> and an update to 3261 would be good, for the following reasons:
>
> 1) We really should want to change 3261 to say MUST NOT put an ip- 
> address/hostname in the call-id.  Really in 20/20 hindsight I don't  
> think it should have had one to begin with.
>
> 2) It's easier for customers to force vendors to do something if  
> there's a MUST statement, standards-track RFC, vs. an informational  
> recommendation/should.
>
> 3) Endpoint UAC vendors should do it *before* getting deployed and  
> being told by customers to change after-wards.  Specifically I'm  
> thinking about vendors who don't attend IETF and aren't that up-to- 
> speed on all the nuances and such.  An update to 3261 with normative  
> mandatory language is clearer, I think.
>

I find myself in agreement with all three arguments here.

--
Dean


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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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On Mar 28, 2009, at 3:02 PM, Jiri Kuthan wrote:

>
>
> I'm worried this is only a wishful thinking. While perfectly
> logical, still even in such constrained setups some bizzar
> ALGs do in my experience appear in the middle, change SDP
> and make thus the identity worthless.


If they change the SDP,  they've changed the identity assertion. They  
need to re-assert. There might reasonably be a need to have a stack of  
assertions with diffs, so one could go back and see the original SDP,  
with its original key fingerprint and original verifiable signature,  
but just changing stuff and pretending the associated identity  
assertion hasn't changed is fundamentally bogus.

The MAIN reason, it seems to me, to want to change the SDP (and  
especially the key fingerprint) undetectably is to enable intercept  
(lawful or otherwise) without detectability. That just won't do.

Perhaps we only need are two different identity assertions; one on the  
original SDP with fingerprint, and one on whatever has resulted post- 
modification by the MITM attackers. Intermediate manipulations by yet  
more MITN attackers might well be worthless.

Let's put it another way: One of the very important things that RFC  
4474 does is protect the DTLS/SRTP key fingerprint. Arguably, its  
protection of the other stuff in the SDP, such as the port numbers and  
IP addresses used for media, is just a side effect. Of course,  
changing the codec (transcoding) is going to break DTLS/SRTP quite  
horribly, so the codec information might as well be protected by  
Identity.

Would we be willing to accept a model where the fingerprint is  
preserved across SBCs but the IP address/port number information is  
not? I'm not saying that would be easy to produce; it would seem to  
require a very complex modification to RFC 4474's handling of SDP, or  
complementary fundamental changes to RFC 4474 and the DTLS draft  
family. But would it get us past this this stone wall?

If not, why not? And you're going to have to convince me that you're  
not just building a phone-tapper protocol here . . .

--
Dean


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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Dean Willis
> Sent: Sunday, March 29, 2009 9:09 PM
> To: Jiri Kuthan; Jon Peterson
> Cc: SIP List; Francois Audet
> Subject: Re: [Sip] francois' comments and why RFC4474 not 
> used in the field
> 
> 
> On Mar 28, 2009, at 3:02 PM, Jiri Kuthan wrote:
> 
> >
> >
> > I'm worried this is only a wishful thinking. While perfectly
> > logical, still even in such constrained setups some bizzar
> > ALGs do in my experience appear in the middle, change SDP
> > and make thus the identity worthless.
> 
> 
> If they change the SDP,  they've changed the identity 
> assertion. They  
> need to re-assert. There might reasonably be a need to have a 
> stack of  
> assertions with diffs, so one could go back and see the 
> original SDP,  
> with its original key fingerprint and original verifiable signature,  
> but just changing stuff and pretending the associated identity  
> assertion hasn't changed is fundamentally bogus.
> 
> The MAIN reason, it seems to me, to want to change the SDP (and  
> especially the key fingerprint) undetectably is to enable intercept  
> (lawful or otherwise) without detectability. That just won't do.
> 
> Perhaps we only need are two different identity assertions; 
> one on the  
> original SDP with fingerprint, and one on whatever has resulted post- 
> modification by the MITM attackers. Intermediate 
> manipulations by yet  
> more MITN attackers might well be worthless.
> 
> Let's put it another way: One of the very important things that RFC  
> 4474 does is protect the DTLS/SRTP key fingerprint. Arguably, its  
> protection of the other stuff in the SDP, such as the port 
> numbers and  
> IP addresses used for media, is just a side effect. Of course,  
> changing the codec (transcoding) is going to break DTLS/SRTP quite  
> horribly, so the codec information might as well be protected by  
> Identity.
> 
> Would we be willing to accept a model where the fingerprint is  
> preserved across SBCs but the IP address/port number information is  
> not? I'm not saying that would be easy to produce; it would seem to  
> require a very complex modification to RFC 4474's handling of 
> SDP, or  
> complementary fundamental changes to RFC 4474 and the DTLS draft  
> family. But would it get us past this this stone wall?

It would not get us past this stone wall, because this very idea
was already proposed and already rejected,
http://tools.ietf.org/html/draft-wing-sip-identity-media-02#section-4.2

The reasons for the rejection, as I recall, were along the lines of "we 
don't want to require SRTP to provide identity".  That argument is 
countered with the ICE technique in Section 4.3 of the same draft; 
we could change that from ICE to something else that does not 
overload ICE semantics of course.  Other reasons included "but SBCs 
might [destroy | modify | manipulate]" the messages.  The counter 
argument to that is that the operators of SBCs have an incentive 
to allow identity through (namely, so that people will accept 
incoming SIP calls), very much like the long distance phone 
companies in the 1980's added echo cancellation tone detectors 
(because it was something needed by the end users).  Other arguments
questioned how non-Invite requests would get protection (also
answered in that same draft).

It is not a lack of solutions in this space.  It is a lack of 
participation by interested parties and an unwillingness to admit
that voice networks are using SBCs -- as if SBCs operated by service
providers are as real as the tooth fairy or Santa Claus.

> If not, why not? And you're going to have to convince me that you're  
> not just building a phone-tapper protocol here . . .

Strong identity makes it *harder* to build a phone-tapping system.  This
hop-by-hop signing thing that you are describing above, and Jon described at
the microphone, is transitive trust with extra CPU horsepower and something
you could fight better in a lawsuit when you learned that a service provider
lied.  I don't see a service provider getting very interested in the
additional capex, opex, and business risk of doing such signing, especially
compared to the three proposals on the table which require the service
provider do _no_ signing and no stronger attestation than today (same business
risk as today).

-d


From HKaplan@acmepacket.com  Sun Mar 29 22:50:02 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dean Willis <dean.willis@softarmor.com>, Jiri Kuthan <jiri@iptel.org>, Jon Peterson <jon.peterson@neustar.biz>
Date: Mon, 30 Mar 2009 01:50:56 -0400
Thread-Topic: [Sip] francois' comments and why RFC4474 not used in the field
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Cc: SIP List <sip@ietf.org>, Francois Audet <audet@nortel.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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What is it in the SDP that you feel is integral to the identity of the send=
er of the request?  If I send an offer-less Invite and sign it with rfc4474=
 does it not confirm my identity, because there was no SDP?  Clearly it wou=
ld indicate that the signed sender did not send SDP, so it prevents a MITM =
from inserting one, but does it actually let a MITM change the Identity of =
the sender of the request if it inserts one?

You say below that one of the important properties is that it protects the =
dtls-srtp fingerprint, presumably because that allows it to tie the identit=
y of the sender of the SIP request to the identity of the media.  Every dra=
ft proposal thus far to do rfc4474 differently has included the ability to =
sign that fingerprint, specifically for that purpose.  The question is whet=
her the IP-Addr/port of the media is itself integral to the identity of the=
 sender, or not.

-hadriel


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Dea=
n
> Willis
> Sent: Monday, March 30, 2009 12:09 AM
>=20
> On Mar 28, 2009, at 3:02 PM, Jiri Kuthan wrote:
>=20
> > I'm worried this is only a wishful thinking. While perfectly
> > logical, still even in such constrained setups some bizzar
> > ALGs do in my experience appear in the middle, change SDP
> > and make thus the identity worthless.
>=20
>=20
> If they change the SDP,  they've changed the identity assertion. They
> need to re-assert. There might reasonably be a need to have a stack of
> assertions with diffs, so one could go back and see the original SDP,
> with its original key fingerprint and original verifiable signature,
> but just changing stuff and pretending the associated identity
> assertion hasn't changed is fundamentally bogus.
>=20
> The MAIN reason, it seems to me, to want to change the SDP (and
> especially the key fingerprint) undetectably is to enable intercept
> (lawful or otherwise) without detectability. That just won't do.
>=20
> Perhaps we only need are two different identity assertions; one on the
> original SDP with fingerprint, and one on whatever has resulted post-
> modification by the MITM attackers. Intermediate manipulations by yet
> more MITN attackers might well be worthless.
>=20
> Let's put it another way: One of the very important things that RFC
> 4474 does is protect the DTLS/SRTP key fingerprint. Arguably, its
> protection of the other stuff in the SDP, such as the port numbers and
> IP addresses used for media, is just a side effect. Of course,
> changing the codec (transcoding) is going to break DTLS/SRTP quite
> horribly, so the codec information might as well be protected by
> Identity.
>=20
> Would we be willing to accept a model where the fingerprint is
> preserved across SBCs but the IP address/port number information is
> not? I'm not saying that would be easy to produce; it would seem to
> require a very complex modification to RFC 4474's handling of SDP, or
> complementary fundamental changes to RFC 4474 and the DTLS draft
> family. But would it get us past this this stone wall?
>=20
> If not, why not? And you're going to have to convince me that you're
> not just building a phone-tapper protocol here . . .
>=20
> --
> Dean
>=20
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

From HKaplan@acmepacket.com  Sun Mar 29 22:59:17 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dean Willis <dean.willis@softarmor.com>, Jiri Kuthan <jiri@iptel.org>, Jon Peterson <jon.peterson@neustar.biz>
Date: Mon, 30 Mar 2009 02:00:11 -0400
Thread-Topic: [Sip] francois' comments and why RFC4474 not used in the field
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Cc: SIP List <sip@ietf.org>, Francois Audet <audet@nortel.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Dea=
n
> Willis
> Sent: Monday, March 30, 2009 12:09 AM
>=20
> The MAIN reason, it seems to me, to want to change the SDP (and
> especially the key fingerprint) undetectably is to enable intercept
> (lawful or otherwise) without detectability. That just won't do.

No, that's not at all the main reason.  RFC4474 is already not end-to-end. =
 It's signed by a middlebox in the originating domain, and verified by a mi=
ddlebox in the terminating domain.  There is ample opportunity to change th=
e SDP at either of those domains to perform lawful intercept.=20

The main reason to change SDP is to steer the media, for numerous reasons.

-hadriel

From christer.holmberg@ericsson.com  Mon Mar 30 07:18:53 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Dean Willis" <dean.willis@softarmor.com>, "Mary Barnes" <mary.barnes@nortel.com>
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Hi,

Personally I have nothing against moving the keep-alive mechanism to
draft-keep, and I guess most people agree it is how it should have been
done.

However, we have previously agreed not to do it, so I hope we will not
spend lots of time to discuss it again.

Regards,

Christer


-----Original Message-----
From: Dean Willis [mailto:dean.willis@softarmor.com]=20
Sent: Monday, March 30, 2009 4:54 AM
To: Mary Barnes
Cc: Christer Holmberg; sip@ietf.org
Subject: Re: [Sip] New version of draft-holmberg-sip-keep


On Mar 26, 2009, at 6:16 PM, Mary Barnes wrote:

> Hi folks,
>
> Christer has updated the document based on the action item from=20
> Wednesday's SIP session (chaired by SIPPING chairs and modeled as the=20
> proposed DISPATCH WG).  If folks could please review the document and=20
> raise any concerns on the document now, that would be appreciated. We=20
> ask specifically for an answer to the question:  Do you support the=20
> document being adopted as a WG document in RAI (WG tbd)?  A simple=20
> "yes" is fine, however, if you don't think we should adopt this=20
> document, could you please provide the reason.
>
>
>

So, given the outpouring of support, should we pull back the draft-
ietf-sip-outbound and re-write it to refer to the keepalive mechanism of
this draft instead of containing similar but not identical text?

--
Dean

From AUDET@nortel.com  Mon Mar 30 08:32:22 2009
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From: "Francois Audet" <audet@nortel.com>
To: "Jiri Kuthan" <jiri@iptel.org>
Cc: sip@ietf.org
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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=20

> -----Original Message-----
> From: Jiri Kuthan [mailto:jiri@iptel.org]=20
> Sent: Sunday, March 29, 2009 06:38
> To: Audet, Francois (SC100:3055)
> Cc: sip@ietf.org
> Subject: Re: francois' comments and why RFC4474 not used in the field
>=20
> > So, what can we do?  I see 2 possibilities:
> >=20
> > 1 - Make it end-to-end. This means tackling one by one the=20
> problem areas.
> >     Use TURN/ICE instead for relying on SDP payload=20
> manipulation. Use
> >     Call-ID and Contact headers that don't reveal topology, etc.
> >=20
> > 2 - Create a non end-to-end Identity assertion mechanism=20
> that is different
> >     from RFC 4474 and more secure than P-Asserted-Identity.=20
> I have a=20
> >     feeling that it won't be trivial. We may have to define the=20
> >     trust relationship between the ends and the intermediaries.=20
> >=20
> > What I was trying to say in the meeting is that for "end-to-end=20
> > identity", 4474 appears to work. But if the problem we are=20
> trying to=20
> > solve is NOT end-to-end identity, but strong mediated=20
> identity, then,=20
> > let's just say so. It's just a DIFFERENT problem to solve.
>=20
> Well, I'm afraid that 1 is too complicated to work. Bringing=20
> TURN/ICE to work is already quite an enterprise, and as much=20
> success I wish to it, I would be careful and avoid dependency=20
> on it (which is generally a good protocol design practice anyhow).
>=20
> 2 is then probably the safer direction to me. The point is really that
> RFC4474 is not having as broad applicability as we have=20
> hoped, and leaving SIP without Identity is going to be=20
> real-time irritating, compared to convenience of Email's=20
> off-line irritation.

1 is definitively difficult to implement. However, I'm quite afraid that
2 is actually worst...
=20
> Some of the RFC 4474 applicability limitations could be=20
> lifted by reducing its integrity protection scope. I'm not=20
> sure how popular will it be then (some have had different=20
> opinions on adoptability of things relying on CA) but it =20
> seems worth trying.

My fear is that by doing so, we will not do something that
will be substantially better than P-Asserted-Identity.

But like you said, it may be worth trying.

From AUDET@nortel.com  Mon Mar 30 08:47:04 2009
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From: "Francois Audet" <audet@nortel.com>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>, "Dean Willis" <dean.willis@softarmor.com>, "Jiri Kuthan" <jiri@iptel.org>,  "Jon Peterson" <jon.peterson@neustar.biz>
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=20
> No, that's not at all the main reason.  RFC4474 is already=20
> not end-to-end.  It's signed by a middlebox in the=20
> originating domain, and verified by a middlebox in the=20
> terminating domain.  There is ample opportunity to change the=20
> SDP at either of those domains to perform lawful intercept.=20

End-to-end means from end domain to end domain. So 4474 is
end-to-end.

Again, if enterprise A wants to talk to enterprise B, without
some service provider spying or redirecting media for their
own potentially nefarious purposes, then 4474 is a perfect
way to ensure that you are talking to who you think you are
talking to.

The fact that these enterprise may or may not be using their
own SBCs to deal with NAT traversal and topology hiding is
not relevant.

> The main reason to change SDP is to steer the media, for=20
> numerous reasons.

It should be up to the Enterprise to decide if it is willing
to deal with a man-in-the-middle. If it does not, then it
can use 4474.

If it does, then it's not end-to-end security. This is probably
acceptable to many enterprises, provided that they can trust
their service provider.

So again, perhaps what we need is a non end-to-end secure identity.
Perhaps something that requires a broker/service provider.

From pkyzivat@cisco.com  Mon Mar 30 09:53:55 2009
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Cc: SIP <sip@ietf.org>, Adam Roach <adam@nostrum.com>
Subject: Re: [Sip] I-D Inaction: draft-kaplan-sip-secure-call-id-00.txt
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Hadriel,

I understand your logic. BUT are we agreed that we should be making 
non-backward-compatible normative changes to 3261 (or any of the key 
RFCs) for reasons other than correcting out and out errors and 
ambiguities? Seems like that is the start of SIP/3.0.

	Paul

Hadriel Kaplan wrote:
> Yeah I've been thinking about what you said the other day about this, and I think I understand your point, but I still think MUST's and an update to 3261 would be good, for the following reasons:
> 
> 1) We really should want to change 3261 to say MUST NOT put an ip-address/hostname in the call-id.  Really in 20/20 hindsight I don't think it should have had one to begin with.
> 
> 2) It's easier for customers to force vendors to do something if there's a MUST statement, standards-track RFC, vs. an informational recommendation/should.
> 
> 3) Endpoint UAC vendors should do it *before* getting deployed and being told by customers to change after-wards.  Specifically I'm thinking about vendors who don't attend IETF and aren't that up-to-speed on all the nuances and such.  An update to 3261 with normative mandatory language is clearer, I think.
> 
> -hadriel
> 
> 
>> -----Original Message-----
>> From: Adam Roach [mailto:adam@nostrum.com]
>> Sent: Friday, March 27, 2009 4:20 PM
>>
>> Which is why an informational draft explaining the problem is useful,
>> while a normative document... not quite so much.
>>
>> Tell UACs: "If you include machine names in your Call-IDs, B2BUAs will
>> have reason to change them." There's no MUST or MUST NOT here, just an
>> explanation of rationale.
>>
>> Tell B2BUAs: "If a Call-ID doesn't contain an @ sign, you'll do everyone
>> a favor by not changing them, because it allows proper correlation of
>> related transactions an dialogs." Again, no MUST; no MUST NOT. Just
>> cause and effect.
>>
>> Then see if people want to play nice. The market will sort out if that's
>> valuable or not.
>>
>> /a
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

From HKaplan@acmepacket.com  Mon Mar 30 10:01:41 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Mon, 30 Mar 2009 13:02:37 -0400
Thread-Topic: [Sip] I-D Inaction: draft-kaplan-sip-secure-call-id-00.txt
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Cc: SIP <sip@ietf.org>, Adam Roach <adam@nostrum.com>
Subject: Re: [Sip] I-D Inaction: draft-kaplan-sip-secure-call-id-00.txt
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What's non-backward-compatible about it?  The text would say a UAC should g=
enerate a call-id without the IP/host (which is completely in-line with 326=
1), and that a UAS must continue to accept a call-id with ip/host.

I'll remove the ABNF change piece since people are worried that may affect =
backwards compatibility, due to a future strict-compliance parser checking =
for this update's ABNF and rejecting legacy 3261 ABNF.  So this update woul=
d only affect the normative text and not the ABNF.

-hadriel

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Monday, March 30, 2009 12:55 PM
> To: Hadriel Kaplan
> Cc: Adam Roach; SIP
> Subject: Re: [Sip] I-D Inaction: draft-kaplan-sip-secure-call-id-00.txt
>=20
> Hadriel,
>=20
> I understand your logic. BUT are we agreed that we should be making
> non-backward-compatible normative changes to 3261 (or any of the key
> RFCs) for reasons other than correcting out and out errors and
> ambiguities? Seems like that is the start of SIP/3.0.
>=20
> 	Paul
>=20
> Hadriel Kaplan wrote:
> > Yeah I've been thinking about what you said the other day about this,
> and I think I understand your point, but I still think MUST's and an
> update to 3261 would be good, for the following reasons:
> >
> > 1) We really should want to change 3261 to say MUST NOT put an ip-
> address/hostname in the call-id.  Really in 20/20 hindsight I don't think
> it should have had one to begin with.
> >
> > 2) It's easier for customers to force vendors to do something if there'=
s
> a MUST statement, standards-track RFC, vs. an informational
> recommendation/should.
> >
> > 3) Endpoint UAC vendors should do it *before* getting deployed and bein=
g
> told by customers to change after-wards.  Specifically I'm thinking about
> vendors who don't attend IETF and aren't that up-to-speed on all the
> nuances and such.  An update to 3261 with normative mandatory language is
> clearer, I think.
> >
> > -hadriel
> >
> >
> >> -----Original Message-----
> >> From: Adam Roach [mailto:adam@nostrum.com]
> >> Sent: Friday, March 27, 2009 4:20 PM
> >>
> >> Which is why an informational draft explaining the problem is useful,
> >> while a normative document... not quite so much.
> >>
> >> Tell UACs: "If you include machine names in your Call-IDs, B2BUAs will
> >> have reason to change them." There's no MUST or MUST NOT here, just an
> >> explanation of rationale.
> >>
> >> Tell B2BUAs: "If a Call-ID doesn't contain an @ sign, you'll do
> everyone
> >> a favor by not changing them, because it allows proper correlation of
> >> related transactions an dialogs." Again, no MUST; no MUST NOT. Just
> >> cause and effect.
> >>
> >> Then see if people want to play nice. The market will sort out if
> that's
> >> valuable or not.
> >>
> >> /a
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >

From AUDET@nortel.com  Mon Mar 30 10:09:16 2009
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From: "Francois Audet" <audet@nortel.com>
To: "Dean Willis" <dean.willis@softarmor.com>, "Mary Barnes" <mary.barnes@nortel.com>
Cc: sip@ietf.org, Cullen Jennings <fluffy@cisco.com>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] New version of draft-holmberg-sip-keep
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Dean,=20

There is no reason to do so.

There is no "similar but not identical text" in draft-keep.
Draft-keep only defines a parameter that allows the usage
of the keep-alive defined in draft-ietf-sip-outbound, outside of
the context of the registration procedures
of draft-ietf-sip-outbound.=20

Currently, the two drafts are carefully choregraphed to
be consistent. I don't want to undo this at the last minute.

Note that some other drafts in the queue are dependent on=20
draft-ietf-outbound (e.g., draft-ietf-sip-sips). I really
don't want us to undo our work for no good reason.

I would recommend moving draft-keep swiftly to WG document.
I believe it should rapidly move into last call.

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Dean Willis
> Sent: Sunday, March 29, 2009 20:54
> To: Barnes, Mary (RICH2:AR00)
> Cc: sip@ietf.org; Christer Holmberg
> Subject: Re: [Sip] New version of draft-holmberg-sip-keep
>=20
>=20
> On Mar 26, 2009, at 6:16 PM, Mary Barnes wrote:
>=20
> > Hi folks,
> >
> > Christer has updated the document based on the action item from=20
> > Wednesday's SIP session (chaired by SIPPING chairs and=20
> modeled as the=20
> > proposed DISPATCH WG).  If folks could please review the=20
> document and=20
> > raise any concerns on the document now, that would be=20
> appreciated. We=20
> > ask specifically for an answer to the question:  Do you support the=20
> > document being adopted as a WG document in RAI (WG tbd)?  A simple=20
> > "yes" is fine, however, if you don't think we should adopt this=20
> > document, could you please provide the reason.
> >
> >
> >
>=20
> So, given the outpouring of support, should we pull back the=20
> draft- ietf-sip-outbound and re-write it to refer to the=20
> keepalive mechanism of this draft instead of containing=20
> similar but not identical text?
>=20
> --
> Dean
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

From victor.pascual.avila@gmail.com  Mon Mar 30 10:17:22 2009
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From: =?UTF-8?Q?Victor_Pascual_=C3=81vila?= <victor.pascual.avila@gmail.com>
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Subject: Re: [Sip] The problem with draft-kaplan-sip-secure-call-id-00
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On Sun, Mar 29, 2009 at 2:56 AM, Jiri Kuthan <jiri@iptel.org> wrote:
(snip)
> As for the backwards-compatibility argument, I guess that it will
> take same long time to get UAs do session-id, so IMO an improvement
> is only possible under an assumption that network-asserted session-id
> is easier to introduce than UA-asserted, right? That appears
> meaningful to me.

I'm still worried about out-of-dialog messages referencing to Call-ID
values. It sounds like "you break it, you fix it".
--=20
Victor Pascual =C3=81vila

From dean.willis@softarmor.com  Mon Mar 30 11:17:37 2009
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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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On Mar 30, 2009, at 1:00 AM, Hadriel Kaplan wrote:

>
>
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf  
>> Of Dean
>> Willis
>> Sent: Monday, March 30, 2009 12:09 AM
>>
>> The MAIN reason, it seems to me, to want to change the SDP (and
>> especially the key fingerprint) undetectably is to enable intercept
>> (lawful or otherwise) without detectability. That just won't do.
>
> No, that's not at all the main reason.  RFC4474 is already not end- 
> to-end.  It's signed by a middlebox in the originating domain, and  
> verified by a middlebox in the terminating domain.  There is ample  
> opportunity to change the SDP at either of those domains to perform  
> lawful intercept.

RFC 4474 can be used end to end. It all depends on which cert is used  
to sign the identity, where that signing occurs, and where it gets  
checked. For stupid legacy phones, it can be signed and checked at  
middleboxes operating under some level of transitive trust. For  
smarter devices, it happens in the endpoint.

Signing and checking at the endpoint, coupled with the DTLS  
fingerprint check. enables end-to-end verification of media integrity  
and privacy. Once you start ripping these signatures out in the middle  
of the network and replacing them, we have lost this absolutely  
critical property. If transit providers even think it MIGHT be  
reasonable to rip out the signatures, then we have lost the key  
functionality. Removing signatures MUST BE FORBIDDEN. Adding new ones  
might be ok.

I might be willing to compromise on the mechanism (I've already  
proposed several potential approaches to doing so), but I'm unwilling  
to compromise on the fundamental requirement for this end-to-end  
applicability.

--
Dean

From jiri@iptel.org  Mon Mar 30 11:24:40 2009
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Cc: SIP List <sip@ietf.org>, Francois Audet <audet@nortel.com>, Jon Peterson <jon.peterson@neustar.biz>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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Dean Willis wrote:
> 
> On Mar 28, 2009, at 3:02 PM, Jiri Kuthan wrote:
> 
>>
>>
>> I'm worried this is only a wishful thinking. While perfectly
>> logical, still even in such constrained setups some bizzar
>> ALGs do in my experience appear in the middle, change SDP
>> and make thus the identity worthless.
> 
> 
> If they change the SDP,  they've changed the identity assertion. They 
> need to re-assert. There might reasonably be a need to have a stack of 
> assertions with diffs, so one could go back and see the original SDP, 
> with its original key fingerprint and original verifiable signature, but 
> just changing stuff and pretending the associated identity assertion 
> hasn't changed is fundamentally bogus.

Hi Dean,

My argument has been that including SDP's IP address and port numbers
in message integrity check is not an identity assertion, but a protocol
shortcoming (certainly well-meant, but making things hard to deploy).
I really think that identity and integrity of SDP are two different things.

Or do you think that these elements somehow belong to identity assertion?

> 
> The MAIN reason, it seems to me, to want to change the SDP (and 
> especially the key fingerprint) undetectably is to enable intercept 
> (lawful or otherwise) without detectability. That just won't do.

In the field, the desire to change SDP I have witnessed is due to NATs.
We don't have TURN & ICE in the field and I'm quite worried that very
few care today. In fact, SDP rewriting has been in use for so long, that
I consider TURN's success chances very marginal. Other I have witnessed
is transcoding.

To make it clear: I'm not opposed to all sorts of integrity protection.
I'm just worried that if we try to solve identity in a single package
with integrity of ever changing SDP, we will unlikely succeed.

> 
> Perhaps we only need are two different identity assertions; one on the 
> original SDP with fingerprint, and one on whatever has resulted 
> post-modification by the MITM attackers. Intermediate manipulations by 
> yet more MITN attackers might well be worthless.
> 
> Let's put it another way: One of the very important things that RFC 4474 
> does is protect the DTLS/SRTP key fingerprint. Arguably, its protection 
> of the other stuff in the SDP, such as the port numbers and IP addresses 
> used for media, is just a side effect. Of course, changing the codec 
> (transcoding) is going to break DTLS/SRTP quite horribly, so the codec 
> information might as well be protected by Identity.
> 
> Would we be willing to accept a model where the fingerprint is preserved 
> across SBCs but the IP address/port number information is not? I'm not 
> saying that would be easy to produce; it would seem to require a very 
> complex modification to RFC 4474's handling of SDP, or complementary 
> fundamental changes to RFC 4474 and the DTLS draft family. But would it 
> get us past this this stone wall?

Logically yes, integrity of some pieces that are known not to be useful
to be rewritten would be ok.

The practicability is of course another question, on which many may have
different opinions. Should mine be asked, I would be in favor of using
confidentiality protocols that do not have dependencies in general,
and dependencies on protocols with unenforceable identity in particular:
zRTP.

> 
> If not, why not? And you're going to have to convince me that you're not 
> just building a phone-tapper protocol here . . .

I haven't been working on a phone-tapper, even though a media-relay or
transcoder could be labeled so for marketing purposes :-) zRTP actually
makes it gracefuly through media relay while confidentiality remains
preserved.

-jiri

> 
> -- 
> Dean
> 


From jiri@iptel.org  Mon Mar 30 11:24:47 2009
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Cc: SIP List <sip@ietf.org>, Jon Peterson <jon.peterson@neustar.biz>, Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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Francois Audet wrote:
>  
>> No, that's not at all the main reason.  RFC4474 is already 
>> not end-to-end.  It's signed by a middlebox in the 
>> originating domain, and verified by a middlebox in the 
>> terminating domain.  There is ample opportunity to change the 
>> SDP at either of those domains to perform lawful intercept. 
> 
> End-to-end means from end domain to end domain. So 4474 is
> end-to-end.
> 
> Again, if enterprise A wants to talk to enterprise B, without
> some service provider spying or redirecting media for their
> own potentially nefarious purposes, then 4474 is a perfect
> way to ensure that you are talking to who you think you are
> talking to.
> 
> The fact that these enterprise may or may not be using their
> own SBCs to deal with NAT traversal and topology hiding is
> not relevant.
> 
>> The main reason to change SDP is to steer the media, for 
>> numerous reasons.
> 
> It should be up to the Enterprise to decide if it is willing
> to deal with a man-in-the-middle. If it does not, then it
> can use 4474.
> 
> If it does, then it's not end-to-end security. This is probably
> acceptable to many enterprises, provided that they can trust
> their service provider.
> 
> So again, perhaps what we need is a non end-to-end secure identity.
> Perhaps something that requires a broker/service provider.

I think that there is a lot of confusion around putting together
identity and integrity. It is not that I must have one for the other.
I can have identity without integrity of SDP and still be in position
to accept/decline a call based on who is calling me. If I'm worried
about media confidentiality, I can add zRTP.

As long as identity remains clearly separated from other stuff, we can
do quite some things. Most of the confusion seems to come from packaging
it with other functionalities. 4474 without integrity of the IP:port
in SDP would work, DERIVE (as e2e representaive) would work, certainly
many other things as well.

Let's face it,  integrity of whole SDP payload is illusive and any
functionality  relying upon it undeployable.

-jiri


> 


From dean.willis@softarmor.com  Mon Mar 30 11:26:43 2009
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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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On Mar 30, 2009, at 12:30 AM, Dan Wing wrote:

>>
>> Would we be willing to accept a model where the fingerprint is
>> preserved across SBCs but the IP address/port number information is
>> not? I'm not saying that would be easy to produce; it would seem to
>> require a very complex modification to RFC 4474's handling of
>> SDP, or
>> complementary fundamental changes to RFC 4474 and the DTLS draft
>> family. But would it get us past this this stone wall?
>
> It would not get us past this stone wall, because this very idea
> was already proposed and already rejected,
> http://tools.ietf.org/html/draft-wing-sip-identity- 
> media-02#section-4.2

I'm hoping that we've developed a broader understanding of the problem  
since the last time this was circulated,


> The reasons for the rejection, as I recall, were along the lines of  
> "we
> don't want to require SRTP to provide identity".  That argument is
> countered with the ICE technique in Section 4.3 of the same draft;
> we could change that from ICE to something else that does not
> overload ICE semantics of course.  Other reasons included "but SBCs
> might [destroy | modify | manipulate]" the messages.  The counter
> argument to that is that the operators of SBCs have an incentive
> to allow identity through (namely, so that people will accept
> incoming SIP calls), very much like the long distance phone
> companies in the 1980's added echo cancellation tone detectors
> (because it was something needed by the end users).  Other arguments
> questioned how non-Invite requests would get protection (also
> answered in that same draft).

Given that a clever on-path attacker could disrupt "identity" without  
changing the port/address presentation, I fail to see how protecting  
them proves anything.




> It is not a lack of solutions in this space.  It is a lack of
> participation by interested parties and an unwillingness to admit
> that voice networks are using SBCs -- as if SBCs operated by service
> providers are as real as the tooth fairy or Santa Claus.
>
>> If not, why not? And you're going to have to convince me that you're
>> not just building a phone-tapper protocol here . . .
>
> Strong identity makes it *harder* to build a phone-tapping system.   
> This
> hop-by-hop signing thing that you are describing above, and Jon  
> described at
> the microphone, is transitive trust with extra CPU horsepower and  
> something
> you could fight better in a lawsuit when you learned that a service  
> provider
> lied.  I don't see a service provider getting very interested in the
> additional capex, opex, and business risk of doing such signing,  
> especially
> compared to the three proposals on the table which require the service
> provider do _no_ signing and no stronger attestation than today  
> (same business
> risk as today).


You misunderstand the hop-by-hop signing approach I described. By  
adding signatures, rather than replacing them, the original media and  
DTLS key identifier properties are preserved. The stack of signatures  
and new SDPs provide an audit chain for figuring out what happened and  
for a basis of transitive trust where end-to-end trust is not  
existent, but do not impede the ability for end-to-end when the  
endpoints have that capability.


--
Dean


From dean.willis@softarmor.com  Mon Mar 30 11:28:48 2009
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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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On Mar 30, 2009, at 10:46 AM, Francois Audet wrote:

>
>> No, that's not at all the main reason.  RFC4474 is already
>> not end-to-end.  It's signed by a middlebox in the
>> originating domain, and verified by a middlebox in the
>> terminating domain.  There is ample opportunity to change the
>> SDP at either of those domains to perform lawful intercept.
>
> End-to-end means from end domain to end domain. So 4474 is
> end-to-end.

RFC 4474 can also be used user-agent to user-agent, phone to phone.
>
> So again, perhaps what we need is a non end-to-end secure identity.
> Perhaps something that requires a broker/service provider.

How is this not P-Asserted-Identity? What real property does such an  
identity assertion have that PAID lacks?

--
Dean


From dean.willis@softarmor.com  Mon Mar 30 11:39:48 2009
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From: Dean Willis <dean.willis@softarmor.com>
To: Jiri Kuthan <jiri@iptel.org>
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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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On Mar 30, 2009, at 11:59 AM, Jiri Kuthan wrote:

> Dean Willis wrote:
>> On Mar 28, 2009, at 3:02 PM, Jiri Kuthan wrote:
>>>
>>>
>>> I'm worried this is only a wishful thinking. While perfectly
>>> logical, still even in such constrained setups some bizzar
>>> ALGs do in my experience appear in the middle, change SDP
>>> and make thus the identity worthless.
>> If they change the SDP,  they've changed the identity assertion.  
>> They need to re-assert. There might reasonably be a need to have a  
>> stack of assertions with diffs, so one could go back and see the  
>> original SDP, with its original key fingerprint and original  
>> verifiable signature, but just changing stuff and pretending the  
>> associated identity assertion hasn't changed is fundamentally bogus.
>
> My argument has been that including SDP's IP address and port numbers
> in message integrity check is not an identity assertion, but a  
> protocol
> shortcoming (certainly well-meant, but making things hard to deploy).
> I really think that identity and integrity of SDP are two different  
> things.
>
> Or do you think that these elements somehow belong to identity  
> assertion?

The argument has been made that these elements might provide  
additional clues to identity. For example, if I see something that is  
coming from what I know is your IP address, and see that it was signed  
by your proxy, whose cert I trust, I might be inclined to believe this  
assertion more than I would believe it if the source address  were  
coming from a node that had a reverse record in alqueda,org.

In the absence of a DTLS or zRTP protection of the media path, I'd be  
likely to give fairly heavy weighting to this IP address comparison,  
even though we know it's not 100% it is MUCH better than nothing.

Of course, I'd rather have an identity mechanism that actually works  
with DTLS than one than doesn't but "works just a little" for  
unprotected media.

So personally, I'd be willing to give up signing of IP address/port  
numbers when a media key fingerprint is present. But if there's no  
media key fingerprint in the signaling (or in the protocol, as zRTP),  
then I  want to see the address and port numbers included.

--
Dean


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Thread-Topic: [Sip] francois' comments and why RFC4474 not used in the field
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From: "Francois Audet" <audet@nortel.com>
To: "Jiri Kuthan" <jiri@iptel.org>
Cc: SIP List <sip@ietf.org>, Jon Peterson <jon.peterson@neustar.biz>, Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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=20
> I think that there is a lot of confusion around putting=20
> together identity and integrity. It is not that I must have=20
> one for the other.
> I can have identity without integrity of SDP and still be in=20
> position to accept/decline a call based on who is calling me.=20
> If I'm worried about media confidentiality, I can add zRTP.

If relying on dead-beat end-users for enforcing security, then
sure, you can use ZRTP. Those end-users are the same guys who
keep clicking accept anytime theirs browsers tell them the certificate
is invalid.

When the private conversations unlawfully recorded gets leaked,
or when somebody impersonates an important person, the individual who=20
gets fired is the IT guy who recommended using ZRTP.

Anyways, I digress.

I think the argument for the proponents of RFC 4474 is that
you can not separate identity from media security. If 4474
allows for intercepting the media, than the identity of the
interlocutor is wrong.

What you are arguing for is a mechanism that asserts the
identity of the sender of the signalling only (SIP), and not the
media.

Perhaps this what we need. I believe that the current DTLS proposal
actually relies on 4474 for the fingerprinting. I guess we would
need something else.

It's almost as if we would need something like 4474 between originating
domain and Service Provider, and then 4474 again between Service =
provider
and destination domain.

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Cc: SIP List <sip@ietf.org>, Francois Audet <audet@nortel.com>, Jon Peterson <jon.peterson@neustar.biz>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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<FONT face=3D"Default Sans Serif,Verdana,Arial,Helvetica,sans-serif" size=
=3D2>inline<br><br><span style=3D"FONT-SIZE: 10px">This is a PRIVATE messag=
e. If you are not the intended recipient, please delete without copying and=
 kindly advise us by e-mail of the mistake in delivery. <br>NOTE: Regardles=
s of content, this e-mail shall not operate to bind CSC to any order or oth=
er contract unless pursuant to explicit written agreement or government ini=
tiative expressly permitting the use of e-mail for such purpose.</span><spa=
n style=3D"FONT-SIZE: 10px">&nbsp;=E2=80=A2&nbsp;</span><DIV><BR></DIV><FON=
T color=3D#990099>-----sip-bounces@ietf.org wrote: -----<BR><BR></FONT><blo=
ckquote style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; B=
ORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">  <P>To: Hadriel Kaplan &=
lt;HKaplan@acmepacket.com&gt;<BR>From: Dean Willis &lt;dean.willis@softarmo=
r.com&gt;<BR>Sent by: sip-bounces@ietf.org<BR>Date: 03/30/2009 02:18PM<BR>c=
c: SIP List &lt;sip@ietf.org&gt;, Francois Audet &lt;audet@nortel.com&gt;, =
Jon Peterson &lt;jon.peterson@neustar.biz&gt;<BR>Subject: Re: [Sip] francoi=
s' comments and why RFC4474 not used in the field<BR><BR><FONT face=3D"Defa=
ult Monospace,Courier New,Courier,monospace" size=3D2><BR>On Mar 30, 2009, =
at 1:00 AM, Hadriel Kaplan wrote:<BR><BR>&gt;<BR>&gt;<BR>&gt;&gt; -----Orig=
inal Message-----<BR>&gt;&gt; From: sip-bounces@ietf.org [<A href=3D"mailto=
:sip-bounces@ietf.org" target=3Dblank >mailto:sip-bounces@ietf.org</A>] On =
Behalf &nbsp;<BR>&gt;&gt; Of Dean<BR>&gt;&gt; Willis<BR>&gt;&gt; Sent: Mond=
ay, March 30, 2009 12:09 AM<BR>&gt;&gt;<BR>&gt;&gt; The MAIN reason, it see=
ms to me, to want to change the SDP (and<BR>&gt;&gt; especially the key fin=
gerprint) undetectably is to enable intercept<BR>&gt;&gt; (lawful or otherw=
ise) without detectability. That just won't do.<BR>&gt;<BR>&gt; No, that's =
not at all the main reason. &nbsp;RFC4474 is already not end- <BR>&gt; to-e=
nd. &nbsp;It's signed by a middlebox in the originating domain, and &nbsp;<=
BR>&gt; verified by a middlebox in the terminating domain. &nbsp;There is a=
mple &nbsp;<BR>&gt; opportunity to change the SDP at either of those domain=
s to perform &nbsp;<BR>&gt; lawful intercept.<BR><BR>RFC 4474 can be used e=
nd to end. It all depends on which cert is used &nbsp;<BR>to sign the ident=
ity, where that signing occurs, and where it gets &nbsp;<BR>checked. For st=
upid legacy phones, it can be signed and checked at &nbsp;<BR>middleboxes o=
perating under some level of transitive trust. For &nbsp;<BR>smarter device=
s, it happens in the endpoint.<BR><BR>Signing and checking at the endpoint,=
 coupled with the DTLS &nbsp;<BR>fingerprint check. enables end-to-end veri=
fication of media integrity &nbsp;<BR>and privacy. Once you start ripping t=
hese signatures out in the middle &nbsp;<BR>of the network and replacing th=
em, we have lost this absolutely &nbsp;<BR>critical property. If transit pr=
oviders even think it MIGHT be &nbsp;<BR>reasonable to rip out the signatur=
es, then we have lost the key &nbsp;<BR>functionality. Removing signatures =
MUST BE FORBIDDEN. Adding new ones &nbsp;<BR>might be ok.</FONT></P>  <P><F=
ONT face=3D"Default Monospace,Courier New,Courier,monospace" size=3D2>I am =
  100% in agreement with this normative text above. I also believe that Bot=
h   the&nbsp;Signalling and Media&nbsp;need to be asserted ( this is not th=
e case   in 4474), the mechanism is TBD, but I believe this should be a req=
uirement.   Think of your children I have a 5 year old girl growing up in a=
 next   generation environment, if Identity is not completely solved in the=
 next few   years, then shame on us for not helping to protect the next gen=
eration! My   0.02c worth.<BR><BR>I might be willing to compromise on the m=
echanism (I've already &nbsp;<BR>proposed several potential approaches to d=
oing so), but I'm unwilling &nbsp;<BR>to compromise on the fundamental requ=
irement for this end-to-end &nbsp;<BR>applicability.<BR><BR>--<BR>Dean<BR>=
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F<BR>Sip m=
ailing list &nbsp;<A href=3D"https://www.ietf.org/mailman/listinfo/sip" tar=
get=3Dblank >https://www.ietf.org/mailman/listinfo/sip</A><BR>This list is =
for NEW development of the core SIP Protocol<BR>Use sip-implementors@cs.col=
umbia.edu for questions on current sip<BR>Use sipping@ietf.org for new deve=
lopments on the application of sip<BR></P></FONT></blockquote><br></FONT>=

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From: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
To: Jiri Kuthan <jiri@iptel.org>, Dean Willis <dean.willis@softarmor.com>
Date: Mon, 30 Mar 2009 15:25:11 -0400
Thread-Topic: [Sip] francois' comments and why RFC4474 not used in the field
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Cc: SIP List <sip@ietf.org>, Francois Audet <audet@nortel.com>, Jon Peterson <jon.peterson@neustar.biz>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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Francois has said it, and so has Jiri. =20

" I think that there is a lot of confusion around putting together identity=
 and integrity."=20

" I really think that identity and integrity of SDP are two different thing=
s."

Put me down as +1

The need is IDENTITY first and foremost.

Adam



From dean.willis@softarmor.com  Mon Mar 30 12:48:04 2009
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From: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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On Mar 30, 2009, at 2:25 PM, Uzelac, Adam wrote:
>
>
> The need is IDENTITY first and foremost.

Is that signaling identity, or media identity, or both?

Is either useful alone?

I'm thinking that a lot of people are demanding signaling identity  
without media identity, and haven't really thought about the  
consequences of that decision because they grew up with a telephone  
system that  essentially lacks the distinction.

--
Dean


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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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> > Strong identity makes it *harder* to build a phone-tapping 
> system.   
> > This
> > hop-by-hop signing thing that you are describing above, and Jon  
> > described at
> > the microphone, is transitive trust with extra CPU horsepower and  
> > something
> > you could fight better in a lawsuit when you learned that a 
> service  
> > provider
> > lied.  I don't see a service provider getting very interested in the
> > additional capex, opex, and business risk of doing such signing,  
> > especially
> > compared to the three proposals on the table which require 
> the service
> > provider do _no_ signing and no stronger attestation than today  
> > (same business
> > risk as today).
> 
> 
> You misunderstand the hop-by-hop signing approach I described. By  
> adding signatures, rather than replacing them, the original 
> media and  
> DTLS key identifier properties are preserved. The stack of 
> signatures  
> and new SDPs provide an audit chain for figuring out what 
> happened and  
> for a basis of transitive trust where end-to-end trust is not  
> existent, but do not impede the ability for end-to-end when the  
> endpoints have that capability.

You are right, I had misunderstood.  Your proposal is different
from the proposal that Jon described at the microphone.  However,
it would disclose the entire path which SP's appear reluctant to
do -- else they would not remove Via headers (I understand most
SPs, using SBCs, remove Via headers).

-d



From michael@voip.co.uk  Mon Mar 30 13:00:38 2009
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Cc: SIP List <sip@ietf.org>, "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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Dean Willis wrote:
> On Mar 30, 2009, at 2:25 PM, Uzelac, Adam wrote:
>> The need is IDENTITY first and foremost.
>
> Is that signaling identity, or media identity, or both?
>
> Is either useful alone?

I think it is at least a bit useful.  If Alice calls Bob, Bob can ensure
that the signalling for the call originated from Alice.  He can't assert
that the media is to/from Alice yet, but it is a start.  More usefully,
if Eve calls Bob, Bob can see that it isn't Alice.  A positive result
from signalling-only identity isn't as useful as you might hope.  But a
negative result is still interesting.

Is that enough of a solution to still be useful?  There are still
several holes in it, but it is better than we have at present.

> I'm thinking that a lot of people are demanding signaling identity
> without media identity, and haven't really thought about the
> consequences of that decision because they grew up with a telephone
> system that  essentially lacks the distinction.

Indeed.

Michael


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Cc: SIP <sip@ietf.org>, Adam Roach <adam@nostrum.com>
Subject: Re: [Sip] I-D Inaction: draft-kaplan-sip-secure-call-id-00.txt
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Hadriel Kaplan wrote:
> What's non-backward-compatible about it?  The text would say a UAC should generate a call-id without the IP/host (which is completely in-line with 3261), and that a UAS must continue to accept a call-id with ip/host.

OK. I thought you were proposing to go further.

What you say above here seems fine. (But of course some won't implement 
SHOULD either.)

	Thanks,
	Paul

> I'll remove the ABNF change piece since people are worried that may affect backwards compatibility, due to a future strict-compliance parser checking for this update's ABNF and rejecting legacy 3261 ABNF.  So this update would only affect the normative text and not the ABNF.
> 
> -hadriel
> 
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Monday, March 30, 2009 12:55 PM
>> To: Hadriel Kaplan
>> Cc: Adam Roach; SIP
>> Subject: Re: [Sip] I-D Inaction: draft-kaplan-sip-secure-call-id-00.txt
>>
>> Hadriel,
>>
>> I understand your logic. BUT are we agreed that we should be making
>> non-backward-compatible normative changes to 3261 (or any of the key
>> RFCs) for reasons other than correcting out and out errors and
>> ambiguities? Seems like that is the start of SIP/3.0.
>>
>> 	Paul
>>
>> Hadriel Kaplan wrote:
>>> Yeah I've been thinking about what you said the other day about this,
>> and I think I understand your point, but I still think MUST's and an
>> update to 3261 would be good, for the following reasons:
>>> 1) We really should want to change 3261 to say MUST NOT put an ip-
>> address/hostname in the call-id.  Really in 20/20 hindsight I don't think
>> it should have had one to begin with.
>>> 2) It's easier for customers to force vendors to do something if there's
>> a MUST statement, standards-track RFC, vs. an informational
>> recommendation/should.
>>> 3) Endpoint UAC vendors should do it *before* getting deployed and being
>> told by customers to change after-wards.  Specifically I'm thinking about
>> vendors who don't attend IETF and aren't that up-to-speed on all the
>> nuances and such.  An update to 3261 with normative mandatory language is
>> clearer, I think.
>>> -hadriel
>>>
>>>
>>>> -----Original Message-----
>>>> From: Adam Roach [mailto:adam@nostrum.com]
>>>> Sent: Friday, March 27, 2009 4:20 PM
>>>>
>>>> Which is why an informational draft explaining the problem is useful,
>>>> while a normative document... not quite so much.
>>>>
>>>> Tell UACs: "If you include machine names in your Call-IDs, B2BUAs will
>>>> have reason to change them." There's no MUST or MUST NOT here, just an
>>>> explanation of rationale.
>>>>
>>>> Tell B2BUAs: "If a Call-ID doesn't contain an @ sign, you'll do
>> everyone
>>>> a favor by not changing them, because it allows proper correlation of
>>>> related transactions an dialogs." Again, no MUST; no MUST NOT. Just
>>>> cause and effect.
>>>>
>>>> Then see if people want to play nice. The market will sort out if
>> that's
>>>> valuable or not.
>>>>
>>>> /a
>>> _______________________________________________
>>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
>>>
> 

From john.elwell@siemens-enterprise.com  Mon Mar 30 13:54:58 2009
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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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Jiri,=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Jiri Kuthan
> Sent: 28 March 2009 20:02
> To: sip@ietf.org
> Cc: Francois Audet
> Subject: [Sip] francois' comments and why RFC4474 not used in=20
> the field
>=20
> As we were running out of time in the meeting, I wanted to
> follow up on an argument which Francois has made: RFC 4474
> works if the identity provider and verifiers are "next to
> each other", thus we have no problem with this RFC.
>=20
> /* hope I didn't misinterpret, let me know should that be
> the case */
>=20
> I'm worried this is only a wishful thinking. While perfectly
> logical, still even in such constrained setups some bizzar
> ALGs do in my experience appear in the middle, change SDP
> and make thus the identity worthless.
>=20
> I think the bottom line is the protocol is not really end-to-end.
> It makes quite some assumptions about the quality of underlying
> network (particularly about not changing SDPs) as opposed to
> trying to deal with lack of it. In fact, SDP payloads are
> changed at quite a number of devices: proxy servers, B2BUAs,
> SBCs, you name it. Assuming there is nothing in the way
> which won't change SDP is just NOT ROBUST.
>=20
> Other way to look at it is it is function overloading. You ask
> for identity and get message integrity -- consequently when
> the latter fails, the former, which you really initially wanted,
> fails too. ekr's argument is certainly right that a message
> of certain origin but uncertain content is close to worthless.
> OTOH, the particular piece, IP address and port numbers in SDP
> are certainly not to be covered by a message integrity checks.
> In fact, for accepting calls based on who is calling, as
> a minimalistic application example, one does  not need
> message integrity.
>=20
> In conclusion, I do not think that RFC4474 can be of any help
> (other than in controlled environments which beats its purpose)
> as long as IP/port in SDP are message-integrity-protected.
[JRE] Exactly. Moreover, when media are secured by some means (SRTP in
the case of RTP) and if SDP contains a fingerprint of the certificate
used to secure the media, there is absolutely no need to sign the IP
address/port. They can quite happily change without impacting the
security of media. Of course, if media isn't secured, it is a slightly
different matter, but I doubt that signing the IP address and port
really buys us much. So you might as well not include IP address and
port in the signature, and if they are changed by intermediate SBCs, no
problem.

John

From john.elwell@siemens-enterprise.com  Mon Mar 30 13:55:34 2009
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Date: Mon, 30 Mar 2009 21:56:30 +0100
From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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To: Jiri Kuthan <jiri@iptel.org>, Francois Audet <audet@nortel.com>
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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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Jiri,=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Jiri Kuthan
> Sent: 29 March 2009 14:38
> To: Francois Audet
> Cc: sip@ietf.org
> Subject: Re: [Sip] francois' comments and why RFC4474 not=20
> used in the field
>=20
> Francois Audet wrote:
> > I think our views are not that appart.
> >=20
> > See belows. =20
> >=20
> >> -----Original Message-----
> >> From: Jiri Kuthan [mailto:jiri@iptel.org]=20
> >> Sent: Saturday, March 28, 2009 13:02
> >> To: sip@ietf.org
> >> Cc: Audet, Francois (SC100:3055)
> >> Subject: francois' comments and why RFC4474 not used in the field
> >>
> >> As we were running out of time in the meeting, I wanted to=20
> >> follow up on an argument which Francois has made: RFC 4474=20
> >> works if the identity provider and verifiers are "next to=20
> >> each other", thus we have no problem with this RFC.
> >>
> >> /* hope I didn't misinterpret, let me know should that be=20
> the case */
> >=20
> > First part is currect, but I didn't say "thus we have no=20
> problem with
> > this RFC."
> >=20
> > As you say, it only works if the identity provider and the=20
> verifier are
> > "next to each other". A federation of Enterprises for=20
> example. Or between
> > service providers.
> >=20
> > However, one of the most predominant model today doesn't=20
> work like this.
> > A domain is often relying on a third party, a "service provider".
> >=20
> > So it's really not "end-to-end". That was the point I was=20
> trying to make.
> >=20
> > RFC 4474 appears to work fine for end-to-end identity:=20
> unfortunately,
> > many deployment models (perhaps the majority) and in fact,=20
> not end-to-end,
> > but mediated. In a typical scenario, each end would have it's own
> > "service provider", so 2 intermediaries.
> >=20
> >> I'm worried this is only a wishful thinking. While perfectly=20
> >> logical, still even in such constrained setups some bizzar=20
> >> ALGs do in my experience appear in the middle, change SDP and=20
> >> make thus the identity worthless.
> >=20
> > Yes, I agree 4474 doesn't work if there are intermidiaries.
> >=20
> >> I think the bottom line is the protocol is not really end-to-end.
> >=20
> > EXACTLY!
> >=20
> >=20
> >> It makes quite some assumptions about the quality of=20
> >> underlying network (particularly about not changing SDPs) as=20
> >> opposed to trying to deal with lack of it. In fact, SDP=20
> >> payloads are changed at quite a number of devices: proxy=20
> >> servers, B2BUAs, SBCs, you name it. Assuming there is nothing=20
> >> in the way which won't change SDP is just NOT ROBUST.
> >>
> >> Other way to look at it is it is function overloading. You=20
> >> ask for identity and get message integrity -- consequently=20
> >> when the latter fails, the former, which you really initially=20
> >> wanted, fails too. ekr's argument is certainly right that a=20
> >> message of certain origin but uncertain content is close=20
> to worthless.
> >> OTOH, the particular piece, IP address and port numbers in=20
> >> SDP are certainly not to be covered by a message integrity checks.
> >> In fact, for accepting calls based on who is calling, as a=20
> >> minimalistic application example, one does  not need message=20
> >> integrity.
> >>
> >> In conclusion, I do not think that RFC4474 can be of any help=20
> >> (other than in controlled environments which beats its=20
> >> purpose) as long as IP/port in SDP are message-integrity-protected.
>=20
> Thanks for the clarifications. I think we are on the same page then.
>=20
> >=20
> > Here is what I think:
> >=20
> > RFC 4474 can work in the following situations:
> >=20
> > 1 - Enterprise-to-Enterprise: i.e., the network as a dumb=20
> pipe. The enterprises
> > have their own SBCs modifying SIP/SDP (or they can use ICE).
> >=20
> > 2 - Subscriber on Service provider A to Subscriber on=20
> service provider B:=20
> > i.e., the "domain" for the first subscriber is actually=20
> that of the subscriber.
> > This can work for "residential users" and small enterprises.
> >=20
> > 3 - Any permutation of the above.
> >=20
> > In all those cases, there is effectively end-to-end
> >=20
> > RFC 4474 does NOT work if it's not end-to-end. For example, if=20
> > the "Enterprise" has it's own domain, but it relies on it's=20
> > service provider to perform a number of functions. NAT traversal,
> > topology hiding, etc.
> >=20
> > So, what can we do?  I see 2 possibilities:
> >=20
> > 1 - Make it end-to-end. This means tackling one by one the=20
> problem areas.
> >     Use TURN/ICE instead for relying on SDP payload=20
> manipulation. Use
> >     Call-ID and Contact headers that don't reveal topology, etc.
> >=20
> > 2 - Create a non end-to-end Identity assertion mechanism=20
> that is different
> >     from RFC 4474 and more secure than P-Asserted-Identity.=20
> I have a=20
> >     feeling that it won't be trivial. We may have to define the=20
> >     trust relationship between the ends and the intermediaries.=20
> >=20
> > What I was trying to say in the meeting is that for "end-to-end
> > identity", 4474 appears to work. But if the problem we are trying to
> > solve is NOT end-to-end identity, but strong mediated=20
> identity, then,
> > let's just say so. It's just a DIFFERENT problem to solve.
>=20
> Well, I'm afraid that 1 is too complicated to work. Bringing=20
> TURN/ICE to
> work is already quite an enterprise, and as much success I wish to it,
> I would be careful and avoid dependency on it (which is generally
> a good protocol design practice anyhow).
>=20
> 2 is then probably the safer direction to me. The point is really that
> RFC4474 is not having as broad applicability as we have hoped, and
> leaving SIP without Identity is going to be real-time irritating,
> compared to convenience of Email's off-line irritation.
>=20
> Some of the RFC 4474 applicability limitations could be lifted by
> reducing its integrity protection scope. I'm not sure how popular
> will it be then (some have had different opinions on adoptability
> of things relying on CA) but it  seems worth trying.
>=20
> I would not dare to restrain ourselves to fixing RFC4474 though.
> I'm very much in favor of testing other alternatives. IETF has
> had a very successful tradition of testing several approaches.
> A more ellaborated email on the topic was BTW recently written
> by Dave Crocker:=20
> http://www.ietf.org/mail-archive/web/mmox/current/msg01233.html
>=20
> Obviously DERIVE (or perhaps non-SIP based reverse verifications)
> appears a feasible alternative to me (because  easy to deploy and
> offering incentives to do so), but given the interest it is quite
> certain there will be more different alternatives.
[JRE] I am open to such alternative suggestions. I think the problem
with DERIVE was that it still was impacted by the idiosyncrasies of SIP,
i.e., B2BUAs and backwards compatibility with UACs. Perhaps we need a
non-SIP-based solution.

John

From john.elwell@siemens-enterprise.com  Mon Mar 30 13:56:11 2009
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Date: Mon, 30 Mar 2009 21:57:08 +0100
From: "Elwell, John" <john.elwell@siemens-enterprise.com>
In-reply-to: <0C5FC0C7-6EF1-4CCA-8FB8-D82E4B546E1B@softarmor.com>
To: Dean Willis <dean.willis@softarmor.com>, Hadriel Kaplan <HKaplan@acmepacket.com>
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Cc: SIP List <sip@ietf.org>, Francois Audet <audet@nortel.com>, Jon Peterson <jon.peterson@neustar.biz>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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Dean,=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Dean Willis
> Sent: 30 March 2009 19:18
> To: Hadriel Kaplan
> Cc: SIP List; Francois Audet; Jon Peterson
> Subject: Re: [Sip] francois' comments and why RFC4474 not=20
> used in the field
>=20
>=20
> On Mar 30, 2009, at 1:00 AM, Hadriel Kaplan wrote:
>=20
> >
> >
> >> -----Original Message-----
> >> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]=20
> On Behalf =20
> >> Of Dean
> >> Willis
> >> Sent: Monday, March 30, 2009 12:09 AM
> >>
> >> The MAIN reason, it seems to me, to want to change the SDP (and
> >> especially the key fingerprint) undetectably is to enable intercept
> >> (lawful or otherwise) without detectability. That just won't do.
> >
> > No, that's not at all the main reason.  RFC4474 is already not end-=20
> > to-end.  It's signed by a middlebox in the originating domain, and =20
> > verified by a middlebox in the terminating domain.  There is ample =20
> > opportunity to change the SDP at either of those domains to=20
> perform =20
> > lawful intercept.
>=20
> RFC 4474 can be used end to end. It all depends on which cert=20
> is used =20
> to sign the identity, where that signing occurs, and where it gets =20
> checked. For stupid legacy phones, it can be signed and checked at =20
> middleboxes operating under some level of transitive trust. For =20
> smarter devices, it happens in the endpoint.
>=20
> Signing and checking at the endpoint, coupled with the DTLS =20
> fingerprint check. enables end-to-end verification of media=20
> integrity =20
> and privacy. Once you start ripping these signatures out in=20
> the middle =20
> of the network and replacing them, we have lost this absolutely =20
> critical property. If transit providers even think it MIGHT be =20
> reasonable to rip out the signatures, then we have lost the key =20
> functionality. Removing signatures MUST BE FORBIDDEN. Adding=20
> new ones =20
> might be ok.
>=20
> I might be willing to compromise on the mechanism (I've already =20
> proposed several potential approaches to doing so), but I'm=20
> unwilling =20
> to compromise on the fundamental requirement for this end-to-end =20
> applicability.
[JRE] Yes, that is a positive feature of RFC 4474, the fact that it can
be used end-to-end, end-domain-to-end-domain, or even end-domain-to-end
or end-to-end-domain. In other words, the authentication and
verification functions can be in the endpoint or an end domain
intermediary.

John

>=20
> --
> Dean
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Jiri,=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Jiri Kuthan
> Sent: 30 March 2009 18:00
> To: Dean Willis
> Cc: SIP List; Francois Audet; Jon Peterson
> Subject: Re: [Sip] francois' comments and why RFC4474 not=20
> used in the field
>=20
> Dean Willis wrote:
> >=20
> > On Mar 28, 2009, at 3:02 PM, Jiri Kuthan wrote:
> >=20
> >>
> >>
> >> I'm worried this is only a wishful thinking. While perfectly
> >> logical, still even in such constrained setups some bizzar
> >> ALGs do in my experience appear in the middle, change SDP
> >> and make thus the identity worthless.
> >=20
> >=20
> > If they change the SDP,  they've changed the identity=20
> assertion. They=20
> > need to re-assert. There might reasonably be a need to have=20
> a stack of=20
> > assertions with diffs, so one could go back and see the=20
> original SDP,=20
> > with its original key fingerprint and original verifiable=20
> signature, but=20
> > just changing stuff and pretending the associated identity=20
> assertion=20
> > hasn't changed is fundamentally bogus.
>=20
> Hi Dean,
>=20
> My argument has been that including SDP's IP address and port numbers
> in message integrity check is not an identity assertion, but=20
> a protocol
> shortcoming (certainly well-meant, but making things hard to deploy).
> I really think that identity and integrity of SDP are two=20
> different things.
>=20
> Or do you think that these elements somehow belong to=20
> identity assertion?
>=20
> >=20
> > The MAIN reason, it seems to me, to want to change the SDP (and=20
> > especially the key fingerprint) undetectably is to enable intercept=20
> > (lawful or otherwise) without detectability. That just won't do.
>=20
> In the field, the desire to change SDP I have witnessed is=20
> due to NATs.
> We don't have TURN & ICE in the field and I'm quite worried that very
> few care today. In fact, SDP rewriting has been in use for so=20
> long, that
> I consider TURN's success chances very marginal. Other I have=20
> witnessed
> is transcoding.
>=20
> To make it clear: I'm not opposed to all sorts of integrity=20
> protection.
> I'm just worried that if we try to solve identity in a single package
> with integrity of ever changing SDP, we will unlikely succeed.
>=20
> >=20
> > Perhaps we only need are two different identity assertions;=20
> one on the=20
> > original SDP with fingerprint, and one on whatever has resulted=20
> > post-modification by the MITM attackers. Intermediate=20
> manipulations by=20
> > yet more MITN attackers might well be worthless.
> >=20
> > Let's put it another way: One of the very important things=20
> that RFC 4474=20
> > does is protect the DTLS/SRTP key fingerprint. Arguably,=20
> its protection=20
> > of the other stuff in the SDP, such as the port numbers and=20
> IP addresses=20
> > used for media, is just a side effect. Of course, changing=20
> the codec=20
> > (transcoding) is going to break DTLS/SRTP quite horribly,=20
> so the codec=20
> > information might as well be protected by Identity.
> >=20
> > Would we be willing to accept a model where the fingerprint=20
> is preserved=20
> > across SBCs but the IP address/port number information is=20
> not? I'm not=20
> > saying that would be easy to produce; it would seem to=20
> require a very=20
> > complex modification to RFC 4474's handling of SDP, or=20
> complementary=20
> > fundamental changes to RFC 4474 and the DTLS draft family.=20
> But would it=20
> > get us past this this stone wall?
>=20
> Logically yes, integrity of some pieces that are known not to=20
> be useful
> to be rewritten would be ok.
>=20
> The practicability is of course another question, on which=20
> many may have
> different opinions. Should mine be asked, I would be in favor of using
> confidentiality protocols that do not have dependencies in general,
> and dependencies on protocols with unenforceable identity in=20
> particular:
> zRTP.
>=20
> >=20
> > If not, why not? And you're going to have to convince me=20
> that you're not=20
> > just building a phone-tapper protocol here . . .
>=20
> I haven't been working on a phone-tapper, even though a media-relay or
> transcoder could be labeled so for marketing purposes :-)=20
> zRTP actually
> makes it gracefuly through media relay while confidentiality remains
> preserved.
[JRE] I am concerned about more than just confidentiality of media. I
believe integrity protection (of certain data) and authentication are
also important. There is only limited value on agreeing an encryption
key with an unknown entity.

John


>=20
> -jiri
>=20
> >=20
> > --=20
> > Dean
> >=20
>=20
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

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Cc: SIP List <sip@ietf.org>, "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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inline:

Dean Willis wrote:
> 
> On Mar 30, 2009, at 2:25 PM, Uzelac, Adam wrote:
>>
>>
>> The need is IDENTITY first and foremost.
> 
> Is that signaling identity, or media identity, or both?
> 
> Is either useful alone?
> 
> I'm thinking that a lot of people are demanding signaling identity 
> without media identity, and haven't really thought about the 
> consequences of that decision because they grew up with a telephone 
> system that  essentially lacks the distinction.

 From an end user perspective, I would assert that the most important 
thing is probably the media. If the callerID says, "this is bob", what 
is important to the user, is that when I pick up the phone and start 
talking, it will be Bob who hears me, and Bob that I hear.

Consider this litmus test:

If the signaling actually came from Mary (perhaps as a third party), but 
the media goes/comes to/from Bob, who should appear on the caller ID? I 
say - Bob.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
Cisco Fellow                                   Iselin, NJ 08830
Cisco, Voice Technology Group
jdrosen@cisco.com
http://www.jdrosen.net                         PHONE: (408) 902-3084
http://www.cisco.com

From jiri@iptel.org  Mon Mar 30 16:47:16 2009
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Cc: SIP List <sip@ietf.org>, Francois Audet <audet@nortel.com>, Jon Peterson <jon.peterson@neustar.biz>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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Dean Willis wrote:
> 
> On Mar 30, 2009, at 11:59 AM, Jiri Kuthan wrote:
> 
>> Dean Willis wrote:
>>> On Mar 28, 2009, at 3:02 PM, Jiri Kuthan wrote:
>>>>
>>>>
>>>> I'm worried this is only a wishful thinking. While perfectly
>>>> logical, still even in such constrained setups some bizzar
>>>> ALGs do in my experience appear in the middle, change SDP
>>>> and make thus the identity worthless.
>>> If they change the SDP,  they've changed the identity assertion. They 
>>> need to re-assert. There might reasonably be a need to have a stack 
>>> of assertions with diffs, so one could go back and see the original 
>>> SDP, with its original key fingerprint and original verifiable 
>>> signature, but just changing stuff and pretending the associated 
>>> identity assertion hasn't changed is fundamentally bogus.
>>
>> My argument has been that including SDP's IP address and port numbers
>> in message integrity check is not an identity assertion, but a protocol
>> shortcoming (certainly well-meant, but making things hard to deploy).
>> I really think that identity and integrity of SDP are two different 
>> things.
>>
>> Or do you think that these elements somehow belong to identity assertion?
> 
> The argument has been made that these elements might provide additional 
> clues to identity. For example, if I see something that is coming from 
> what I know is your IP address, and see that it was signed by your 
> proxy, whose cert I trust, I might be inclined to believe this assertion 
> more than I would believe it if the source address  were coming from a 
> node that had a reverse record in alqueda,org.
> 
> In the absence of a DTLS or zRTP protection of the media path, I'd be 
> likely to give fairly heavy weighting to this IP address comparison, 
> even though we know it's not 100% it is MUCH better than nothing.
> 
> Of course, I'd rather have an identity mechanism that actually works 
> with DTLS than one than doesn't but "works just a little" for 
> unprotected media.
> 
> So personally, I'd be willing to give up signing of IP address/port 
> numbers when a media key fingerprint is present. But if there's no media 
> key fingerprint in the signaling (or in the protocol, as zRTP), then I  
> want to see the address and port numbers included.

I perfectly agree with the suggestion "relax about changed IP/port in SDP
if media secured".

-jiri

> 
> -- 
> Dean
> 

From jiri@iptel.org  Mon Mar 30 16:54:44 2009
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Cc: SIP List <sip@ietf.org>, Jon Peterson <jon.peterson@neustar.biz>, Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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Francois Audet wrote:
>  
>> I think that there is a lot of confusion around putting 
>> together identity and integrity. It is not that I must have 
>> one for the other.
>> I can have identity without integrity of SDP and still be in 
>> position to accept/decline a call based on who is calling me. 
>> If I'm worried about media confidentiality, I can add zRTP.
> 
> If relying on dead-beat end-users for enforcing security, then
> sure, you can use ZRTP. Those end-users are the same guys who
> keep clicking accept anytime theirs browsers tell them the certificate
> is invalid.
> 
> When the private conversations unlawfully recorded gets leaked,
> or when somebody impersonates an important person, the individual who 
> gets fired is the IT guy who recommended using ZRTP.
> 
> Anyways, I digress.
> 
> I think the argument for the proponents of RFC 4474 is that
> you can not separate identity from media security. If 4474
> allows for intercepting the media, than the identity of the
> interlocutor is wrong.

That's indeed the argument. In other words, how can be message
of certain origin valuable if of uncertain content. Are you
sure to whom you are sending your RTP packets to? (Ironic
answer: with 99% probability to your enterprise's ALG ;-))

The argument is valid, but unfortunately I believe SDP rewriting
is here to stay, in places we have not anticipated. These SDP
rewriters render the extra security value much less useful then
we have hoped for.


> 
> What you are arguing for is a mechanism that asserts the
> identity of the sender of the signalling only (SIP), and not the
> media.

Yes.

> 
> Perhaps this what we need. I believe that the current DTLS proposal
> actually relies on 4474 for the fingerprinting. I guess we would
> need something else.

That's indeed a problem. Could be addressed by DTLS doing it somewhere
else or 4474 changed to cover less.

-jiri

> 
> It's almost as if we would need something like 4474 between originating
> domain and Service Provider, and then 4474 again between Service provider
> and destination domain.
> 

From jiri@iptel.org  Mon Mar 30 16:57:36 2009
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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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Elwell, John wrote:
>> In conclusion, I do not think that RFC4474 can be of any help
>> (other than in controlled environments which beats its purpose)
>> as long as IP/port in SDP are message-integrity-protected.
> [JRE] Exactly. Moreover, when media are secured by some means (SRTP in
> the case of RTP) and if SDP contains a fingerprint of the certificate
> used to secure the media, there is absolutely no need to sign the IP
> address/port. They can quite happily change without impacting the
> security of media. Of course, if media isn't secured, it is a slightly
> different matter, but I doubt that signing the IP address and port
> really buys us much. So you might as well not include IP address and
> port in the signature, and if they are changed by intermediate SBCs, no
> problem.
> 
> John
> 

Yes, that's the point. -jiri

From jiri@iptel.org  Mon Mar 30 17:07:25 2009
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Cc: sip@ietf.org, Francois Audet <audet@nortel.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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Elwell, John wrote:
>> Obviously DERIVE (or perhaps non-SIP based reverse verifications)
>> appears a feasible alternative to me (because  easy to deploy and
>> offering incentives to do so), but given the interest it is quite
>> certain there will be more different alternatives.
> [JRE] I am open to such alternative suggestions. I think the problem
> with DERIVE was that it still was impacted by the idiosyncrasies of SIP,
> i.e., B2BUAs and backwards compatibility with UACs. Perhaps we need a
> non-SIP-based solution.
> 
> John

Hi John,

that may be very well true, in the end SIP is quite complex nowadays and
overloading it with more stuff may make things possibly worse.

I'm just wondering what folks thing about reverse verification per DNS.
While not providing such a good granularity as SIP does, there is still
some better-than-nothing value in it. What I find appealing about it is
there is some prior experience (our domain for example is using SPF).
Also, identity established via DNS is not worthless -- changing DNS
names to change identity is associated with some prohibitive cost.

Any other ideas?

-jiri

From jiri@iptel.org  Mon Mar 30 18:02:36 2009
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Cc: SIP List <sip@ietf.org>, "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>, Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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Jonathan Rosenberg wrote:
> inline:
> 
> Dean Willis wrote:
>>
>> On Mar 30, 2009, at 2:25 PM, Uzelac, Adam wrote:
>>>
>>>
>>> The need is IDENTITY first and foremost.
>>
>> Is that signaling identity, or media identity, or both?
>>
>> Is either useful alone?
>>
>> I'm thinking that a lot of people are demanding signaling identity 
>> without media identity, and haven't really thought about the 
>> consequences of that decision because they grew up with a telephone 
>> system that  essentially lacks the distinction.
> 
>  From an end user perspective, I would assert that the most important 
> thing is probably the media. If the callerID says, "this is bob", what 
> is important to the user, is that when I pick up the phone and start 
> talking, it will be Bob who hears me, and Bob that I hear.
> 
> Consider this litmus test:
> 
> If the signaling actually came from Mary (perhaps as a third party), but 
> the media goes/comes to/from Bob, who should appear on the caller ID? I 
> say - Bob.

There is a timing aspect in favor of placing identity in signaling --
I would like to know whose call is ringing before I answer (if I do).

-jiri


> 
> -Jonathan R.
> 
> 
> 

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Cc: SIP List <sip@ietf.org>, "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>, Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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inline:

Jiri Kuthan wrote:


>>  From an end user perspective, I would assert that the most important 
>> thing is probably the media. If the callerID says, "this is bob", what 
>> is important to the user, is that when I pick up the phone and start 
>> talking, it will be Bob who hears me, and Bob that I hear.
>>
>> Consider this litmus test:
>>
>> If the signaling actually came from Mary (perhaps as a third party), 
>> but the media goes/comes to/from Bob, who should appear on the caller 
>> ID? I say - Bob.
> 
> There is a timing aspect in favor of placing identity in signaling --
> I would like to know whose call is ringing before I answer (if I do).

You can still have that. Just don't ring the phone until early media has 
been exchanged and verified. Indeed if you were doing an ICE-style thing 
per Dan's draft, you'd get that for free.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
Cisco Fellow                                   Iselin, NJ 08830
Cisco, Voice Technology Group
jdrosen@cisco.com
http://www.jdrosen.net                         PHONE: (408) 902-3084
http://www.cisco.com

From AUDET@nortel.com  Mon Mar 30 19:52:29 2009
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Cc: SIP IETF <sip@ietf.org>, "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>, Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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For everybody's reference, can you point to Dan's draft?


On Mar30 2009 19:12 , "Jonathan Rosenberg" <jdrosen@cisco.com> wrote:

> inline:
> 
> Jiri Kuthan wrote:
> 
> 
>>>  From an end user perspective, I would assert that the most important
>>> thing is probably the media. If the callerID says, "this is bob", what
>>> is important to the user, is that when I pick up the phone and start
>>> talking, it will be Bob who hears me, and Bob that I hear.
>>> 
>>> Consider this litmus test:
>>> 
>>> If the signaling actually came from Mary (perhaps as a third party),
>>> but the media goes/comes to/from Bob, who should appear on the caller
>>> ID? I say - Bob.
>> 
>> There is a timing aspect in favor of placing identity in signaling --
>> I would like to know whose call is ringing before I answer (if I do).
> 
> You can still have that. Just don't ring the phone until early media has
> been exchanged and verified. Indeed if you were doing an ICE-style thing
> per Dan's draft, you'd get that for free.
> 
> -Jonathan R.
> 
> 


From jiri@iptel.org  Mon Mar 30 20:19:47 2009
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Cc: SIP List <sip@ietf.org>, "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>, Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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Jonathan Rosenberg wrote:
> inline:
> 
> Jiri Kuthan wrote:
> 
> 
>>>  From an end user perspective, I would assert that the most important 
>>> thing is probably the media. If the callerID says, "this is bob", 
>>> what is important to the user, is that when I pick up the phone and 
>>> start talking, it will be Bob who hears me, and Bob that I hear.
>>>
>>> Consider this litmus test:
>>>
>>> If the signaling actually came from Mary (perhaps as a third party), 
>>> but the media goes/comes to/from Bob, who should appear on the caller 
>>> ID? I say - Bob.
>>
>> There is a timing aspect in favor of placing identity in signaling --
>> I would like to know whose call is ringing before I answer (if I do).
> 
> You can still have that. Just don't ring the phone until early media has 
> been exchanged and verified. Indeed if you were doing an ICE-style thing 
> per Dan's draft, you'd get that for free.

Hi Jonathan,

Ok, that's feasible in a sense. And there is certain beauty in getting the
security attributes associated with RTP packets, to which they belong.
(I like this about zRTP). A particular example where this would perform
well is today's PSTN directory service, which connects me to the final
recipient, while still the directory's service phone number appears on
my phone. (well, with connected identity one could actually handle
it using signaling too...)

What I'm anxious of is the idea of to do "x" you also have to do
"y" which you need for "z" anyhow -- doesn't that impair the feasibility?
Could not we find something that is just as easy as to do "x" do "x"?


-jiri

From dean.willis@softarmor.com  Mon Mar 30 21:06:57 2009
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From: Dean Willis <dean.willis@softarmor.com>
To: Jiri Kuthan <jiri@iptel.org>
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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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On Mar 30, 2009, at 8:03 PM, Jiri Kuthan wrote:
>
> There is a timing aspect in favor of placing identity in signaling --
> I would like to know whose call is ringing before I answer (if I do).

So, we need identity in signaling, we need identity in media, and we  
need the two identity expressions to match.

That's what we get with RFC 4474 and DTLS/SRTP.

Why were we trying to change it to protect something else?

--
Dean


From dwing@cisco.com  Mon Mar 30 21:14:51 2009
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From: "Dan Wing" <dwing@cisco.com>
To: "'Francois Audet'" <audet@nortel.com>, "'Jonathan Rosenberg'" <jdrosen@cisco.com>, "'Jiri Kuthan'" <jiri@iptel.org>
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Cc: 'SIP IETF' <sip@ietf.org>, "'Uzelac, Adam'" <Adam.Uzelac@globalcrossing.com>, 'Dean Willis' <dean.willis@softarmor.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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> -----Original Message-----
> From: Francois Audet [mailto:audet@nortel.com] 
> Sent: Monday, March 30, 2009 7:53 PM
> To: Jonathan Rosenberg; Jiri Kuthan; Dan WING
> Cc: SIP IETF; Uzelac, Adam; Dean Willis
> Subject: Re: [Sip] francois' comments and why RFC4474 not 
> used in the field
> 
> For everybody's reference, can you point to Dan's draft?

It is
http://tools.ietf.org/html/draft-wing-sip-identity-media-02

other proposals in the same sphere,
http://tools.ietf.org/html/draft-fischer-sip-e2e-sec-media-00
http://tools.ietf.org/html/draft-kaplan-sip-asserter-identity-01

-d


> On Mar30 2009 19:12 , "Jonathan Rosenberg" <jdrosen@cisco.com> wrote:
> 
> > inline:
> > 
> > Jiri Kuthan wrote:
> > 
> > 
> >>>  From an end user perspective, I would assert that the 
> most important
> >>> thing is probably the media. If the callerID says, "this 
> is bob", what
> >>> is important to the user, is that when I pick up the 
> phone and start
> >>> talking, it will be Bob who hears me, and Bob that I hear.
> >>> 
> >>> Consider this litmus test:
> >>> 
> >>> If the signaling actually came from Mary (perhaps as a 
> third party),
> >>> but the media goes/comes to/from Bob, who should appear 
> on the caller
> >>> ID? I say - Bob.
> >> 
> >> There is a timing aspect in favor of placing identity in 
> signaling --
> >> I would like to know whose call is ringing before I answer 
> (if I do).
> > 
> > You can still have that. Just don't ring the phone until 
> early media has
> > been exchanged and verified. Indeed if you were doing an 
> ICE-style thing
> > per Dan's draft, you'd get that for free.
> > 
> > -Jonathan R.
> > 
> > 
> 


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Cc: "'Uzelac, Adam'" <Adam.Uzelac@globalcrossing.com>, 'SIP IETF' <sip@ietf.org>, 'Francois Audet' <audet@nortel.com>, 'Dean Willis' <dean.willis@softarmor.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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<FONT face=3D"Default Sans Serif,Verdana,Arial,Helvetica,sans-serif" size=
=3D2>This general approach looks interesting I need time to digest the Kapl=
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From ietf.hanserik@gmail.com  Tue Mar 31 00:48:10 2009
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Date: Tue, 31 Mar 2009 09:49:06 +0200
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From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: Jonathan Rosenberg <jdrosen@cisco.com>
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Cc: SIP List <sip@ietf.org>, "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>, Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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This litmus test seems to mixup presentation with Identity.

All identity presentation services I know of are based on signaled identity.
Of course a more advanced identity presentation service can be devised that
also incorporates media identity. But this can become quite complicated,
imagine 1 session containing multiple media streams coming from different
entities, what do I present?  Or will this not be allowed anymore?

The first thing I would be interested in is who is trying to setup a session
with me, and if I trust that someone. If I have more stringent requirements
also on the media integrity then I need something additional.

/Hans Erik van Elburg


On Tue, Mar 31, 2009 at 4:12 AM, Jonathan Rosenberg <jdrosen@cisco.com>wrote:

> inline:
>
> Jiri Kuthan wrote:
>
>
>   From an end user perspective, I would assert that the most important
>>> thing is probably the media. If the callerID says, "this is bob", what is
>>> important to the user, is that when I pick up the phone and start talking,
>>> it will be Bob who hears me, and Bob that I hear.
>>>
>>> Consider this litmus test:
>>>
>>> If the signaling actually came from Mary (perhaps as a third party), but
>>> the media goes/comes to/from Bob, who should appear on the caller ID? I say
>>> - Bob.
>>>
>>
>> There is a timing aspect in favor of placing identity in signaling --
>> I would like to know whose call is ringing before I answer (if I do).
>>
>
> You can still have that. Just don't ring the phone until early media has
> been exchanged and verified. Indeed if you were doing an ICE-style thing per
> Dan's draft, you'd get that for free.
>
> -Jonathan R.
>
>
>
> --
> Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
> Cisco Fellow                                   Iselin, NJ 08830
> Cisco, Voice Technology Group
> jdrosen@cisco.com
> http://www.jdrosen.net                         PHONE: (408) 902-3084
> http://www.cisco.com
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
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--001485f27f5eff46a80466656ead
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This litmus test seems to mixup presentation with Identity.<br><br>All iden=
tity presentation services I know of are based on signaled identity. Of cou=
rse a more advanced identity presentation service can be devised that also =
incorporates media identity. But this can become quite complicated, imagine=
 1 session containing multiple media streams coming from different entities=
, what do I present?=A0 Or will this not be allowed anymore?<br>
<br>The first thing I would be interested in is who is trying to setup a se=
ssion with me, and if I trust that someone. If I have more stringent requir=
ements also on the media integrity then I need something additional. <br>
<br clear=3D"all">/Hans Erik van Elburg<br>
<br><br><div class=3D"gmail_quote">On Tue, Mar 31, 2009 at 4:12 AM, Jonatha=
n Rosenberg <span dir=3D"ltr">&lt;<a href=3D"mailto:jdrosen@cisco.com">jdro=
sen@cisco.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" st=
yle=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex=
; padding-left: 1ex;">
inline:<div class=3D"im"><br>
<br>
Jiri Kuthan wrote:<br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><blockquote class=
=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, 204, 204); margin=
: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">

=A0From an end user perspective, I would assert that the most important thi=
ng is probably the media. If the callerID says, &quot;this is bob&quot;, wh=
at is important to the user, is that when I pick up the phone and start tal=
king, it will be Bob who hears me, and Bob that I hear.<br>

<br>
Consider this litmus test:<br>
<br>
If the signaling actually came from Mary (perhaps as a third party), but th=
e media goes/comes to/from Bob, who should appear on the caller ID? I say -=
 Bob.<br>
</blockquote>
<br>
There is a timing aspect in favor of placing identity in signaling --<br>
I would like to know whose call is ringing before I answer (if I do).<br>
</blockquote>
<br></div>
You can still have that. Just don&#39;t ring the phone until early media ha=
s been exchanged and verified. Indeed if you were doing an ICE-style thing =
per Dan&#39;s draft, you&#39;d get that for free.<div class=3D"im"><br>
<br>
-Jonathan R.<br>
<br>
<br>
<br>
-- <br>
Jonathan D. Rosenberg, Ph.D. =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 111 Wood A=
venue South<br>
Cisco Fellow =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 Iselin, NJ 08830<br>
Cisco, Voice Technology Group<br>
<a href=3D"mailto:jdrosen@cisco.com" target=3D"_blank">jdrosen@cisco.com</a=
><br>
<a href=3D"http://www.jdrosen.net" target=3D"_blank">http://www.jdrosen.net=
</a> =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 PHONE: (408) 902-3084<=
br>
<a href=3D"http://www.cisco.com" target=3D"_blank">http://www.cisco.com</a>=
<br>
_______________________________________________<br></div><div><div></div><d=
iv class=3D"h5">
Sip mailing list =A0<a href=3D"https://www.ietf.org/mailman/listinfo/sip" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
This list is for NEW development of the core SIP Protocol<br>
Use <a href=3D"mailto:sip-implementors@cs.columbia.edu" target=3D"_blank">s=
ip-implementors@cs.columbia.edu</a> for questions on current sip<br>
Use <a href=3D"mailto:sipping@ietf.org" target=3D"_blank">sipping@ietf.org<=
/a> for new developments on the application of sip<br>
</div></div></blockquote></div><br>

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Cc: SIP List <sip@ietf.org>, "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>, Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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Hans Erik,
=20
I think the issue is to verify that the media is to/from the same user
as the signalling. I agree it would be complex if we were dealing with
different users.
=20
John


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Hans Erik van Elburg
	Sent: 31 March 2009 08:49
	To: Jonathan Rosenberg
	Cc: SIP List; Uzelac, Adam; Dean Willis
	Subject: Re: [Sip] francois' comments and why RFC4474 not used
in the field
=09
=09
	This litmus test seems to mixup presentation with Identity.
=09
	All identity presentation services I know of are based on
signaled identity. Of course a more advanced identity presentation
service can be devised that also incorporates media identity. But this
can become quite complicated, imagine 1 session containing multiple
media streams coming from different entities, what do I present?  Or
will this not be allowed anymore?
=09
	The first thing I would be interested in is who is trying to
setup a session with me, and if I trust that someone. If I have more
stringent requirements also on the media integrity then I need something
additional.=20
=09
	/Hans Erik van Elburg
=09
=09
=09
	On Tue, Mar 31, 2009 at 4:12 AM, Jonathan Rosenberg
<jdrosen@cisco.com> wrote:
=09

		inline:=20


		Jiri Kuthan wrote:
	=09
	=09
	=09

				 From an end user perspective, I would
assert that the most important thing is probably the media. If the
callerID says, "this is bob", what is important to the user, is that
when I pick up the phone and start talking, it will be Bob who hears me,
and Bob that I hear.
			=09
				Consider this litmus test:
			=09
				If the signaling actually came from Mary
(perhaps as a third party), but the media goes/comes to/from Bob, who
should appear on the caller ID? I say - Bob.
			=09


			There is a timing aspect in favor of placing
identity in signaling --
			I would like to know whose call is ringing
before I answer (if I do).
		=09


		You can still have that. Just don't ring the phone until
early media has been exchanged and verified. Indeed if you were doing an
ICE-style thing per Dan's draft, you'd get that for free.=20


		-Jonathan R.
	=09
	=09
	=09
		--=20
		Jonathan D. Rosenberg, Ph.D.                   111 Wood
Avenue South
		Cisco Fellow                                   Iselin,
NJ 08830
		Cisco, Voice Technology Group
		jdrosen@cisco.com
		http://www.jdrosen.net                         PHONE:
(408) 902-3084
		http://www.cisco.com
		_______________________________________________
	=09
		Sip mailing list
https://www.ietf.org/mailman/listinfo/sip
		This list is for NEW development of the core SIP
Protocol
		Use sip-implementors@cs.columbia.edu for questions on
current sip
		Use sipping@ietf.org for new developments on the
application of sip
	=09



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<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.3492" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D538190509-31032009><FONT =
face=3DArial=20
color=3D#000080>Hans Erik,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D538190509-31032009><FONT =
face=3DArial=20
color=3D#000080></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D538190509-31032009><FONT =
face=3DArial=20
color=3D#000080>I think the issue is to verify that the media is to/from =
the same=20
user as the signalling. I agree it would be complex if we were dealing =
with=20
different users.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D538190509-31032009><FONT =
face=3DArial=20
color=3D#000080></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D538190509-31032009><FONT =
face=3DArial=20
color=3D#000080>John</FONT></SPAN></DIV><BR>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000080 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Hans Erik van=20
  Elburg<BR><B>Sent:</B> 31 March 2009 08:49<BR><B>To:</B> Jonathan=20
  Rosenberg<BR><B>Cc:</B> SIP List; Uzelac, Adam; Dean =
Willis<BR><B>Subject:</B>=20
  Re: [Sip] francois' comments and why RFC4474 not used in the=20
  field<BR></FONT><BR></DIV>
  <DIV></DIV>This litmus test seems to mixup presentation with=20
  Identity.<BR><BR>All identity presentation services I know of are =
based on=20
  signaled identity. Of course a more advanced identity presentation =
service can=20
  be devised that also incorporates media identity. But this can become =
quite=20
  complicated, imagine 1 session containing multiple media streams =
coming from=20
  different entities, what do I present?&nbsp; Or will this not be =
allowed=20
  anymore?<BR><BR>The first thing I would be interested in is who is =
trying to=20
  setup a session with me, and if I trust that someone. If I have more =
stringent=20
  requirements also on the media integrity then I need something =
additional.=20
  <BR><BR clear=3Dall>/Hans Erik van Elburg<BR><BR><BR>
  <DIV class=3Dgmail_quote>On Tue, Mar 31, 2009 at 4:12 AM, Jonathan =
Rosenberg=20
  <SPAN dir=3Dltr>&lt;<A=20
  href=3D"mailto:jdrosen@cisco.com">jdrosen@cisco.com</A>&gt;</SPAN> =
wrote:<BR>
  <BLOCKQUOTE class=3Dgmail_quote=20
  style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">inline:
    <DIV class=3Dim><BR><BR>Jiri Kuthan wrote:<BR><BR><BR>
    <BLOCKQUOTE class=3Dgmail_quote=20
    style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">
      <BLOCKQUOTE class=3Dgmail_quote=20
      style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; =
BORDER-LEFT: rgb(204,204,204) 1px solid">&nbsp;From=20
        an end user perspective, I would assert that the most important =
thing is=20
        probably the media. If the callerID says, "this is bob", what is =

        important to the user, is that when I pick up the phone and =
start=20
        talking, it will be Bob who hears me, and Bob that I=20
        hear.<BR><BR>Consider this litmus test:<BR><BR>If the signaling =
actually=20
        came from Mary (perhaps as a third party), but the media =
goes/comes=20
        to/from Bob, who should appear on the caller ID? I say -=20
      Bob.<BR></BLOCKQUOTE><BR>There is a timing aspect in favor of =
placing=20
      identity in signaling --<BR>I would like to know whose call is =
ringing=20
      before I answer (if I do).<BR></BLOCKQUOTE><BR></DIV>You can still =
have=20
    that. Just don't ring the phone until early media has been exchanged =
and=20
    verified. Indeed if you were doing an ICE-style thing per Dan's =
draft, you'd=20
    get that for free.
    <DIV class=3Dim><BR><BR>-Jonathan R.<BR><BR><BR><BR>-- <BR>Jonathan =
D.=20
    Rosenberg, Ph.D. &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; 111 Wood Avenue South<BR>Cisco Fellow &nbsp; &nbsp; &nbsp; =
&nbsp;=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp; &nbsp; Iselin, NJ 08830<BR>Cisco, Voice Technology =
Group<BR><A=20
    href=3D"mailto:jdrosen@cisco.com" =
target=3D_blank>jdrosen@cisco.com</A><BR><A=20
    href=3D"http://www.jdrosen.net" =
target=3D_blank>http://www.jdrosen.net</A>=20
    &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
    &nbsp; PHONE: (408) 902-3084<BR><A href=3D"http://www.cisco.com"=20
    =
target=3D_blank>http://www.cisco.com</A><BR>_____________________________=
__________________<BR></DIV>
    <DIV>
    <DIV></DIV>
    <DIV class=3Dh5>Sip mailing list &nbsp;<A=20
    href=3D"https://www.ietf.org/mailman/listinfo/sip"=20
    =
target=3D_blank>https://www.ietf.org/mailman/listinfo/sip</A><BR>This =
list is=20
    for NEW development of the core SIP Protocol<BR>Use <A=20
    href=3D"mailto:sip-implementors@cs.columbia.edu"=20
    target=3D_blank>sip-implementors@cs.columbia.edu</A> for questions =
on current=20
    sip<BR>Use <A href=3D"mailto:sipping@ietf.org"=20
    target=3D_blank>sipping@ietf.org</A> for new developments on the =
application=20
    of =
sip<BR></DIV></DIV></BLOCKQUOTE></DIV><BR></BLOCKQUOTE></BODY></HTML>

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From: =?UTF-8?Q?Victor_Pascual_=C3=81vila?= <victor.pascual.avila@gmail.com>
To: Jonathan Rosenberg <jdrosen@cisco.com>
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Cc: SIP List <sip@ietf.org>
Subject: [Sip] Media vs Signaling identity (was Re: francois' comments and why RFC4474 not used in the field)
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On Tue, Mar 31, 2009 at 4:12 AM, Jonathan Rosenberg <jdrosen@cisco.com> wro=
te:
> inline:
>
> Jiri Kuthan wrote:
>
>
>>> =C2=A0From an end user perspective, I would assert that the most import=
ant
>>> thing is probably the media. If the callerID says, "this is bob", what =
is
>>> important to the user, is that when I pick up the phone and start talki=
ng,
>>> it will be Bob who hears me, and Bob that I hear.
>>>
>>> Consider this litmus test:
>>>
>>> If the signaling actually came from Mary (perhaps as a third party), bu=
t
>>> the media goes/comes to/from Bob, who should appear on the caller ID? I=
 say
>>> - Bob.
>>
>> There is a timing aspect in favor of placing identity in signaling --
>> I would like to know whose call is ringing before I answer (if I do).
>
> You can still have that. Just don't ring the phone until early media has
> been exchanged and verified. Indeed if you were doing an ICE-style thing =
per
> Dan's draft, you'd get that for free.

Are we restricting the identity assertion to telephony-like sessions?

IMHO, identity assertion should also work for the following scenarios
(among others):

-rfc3725, figure 1, message 1 (INVITE no SDP): Upon receipt of the
initial INVITE (note there's no session description at all), "A"
decides to authorize or reject the call based on the delivered
identifier.

-rfc3428, figure 1, message 2 (F2): "user2" decides to answer or
ignore the message based on the delivered identifier.

-rfc3515, example 4.1, message 1 (F1): "agent B" decides to accept or
reject the refer based on the delivered identifier

-Any other application based on white/black lists (where the identity
verification must happen before accepting a request).


While "signaling identity" seems to be a common denominator, I'm not
sure about the role of "media identity" in the above listed scenarios.
Any clarification will be appreciated.

Cheers,
--=20
Victor Pascual =C3=81vila

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To: Jonathan Rosenberg <jdrosen@cisco.com>, Dean Willis <dean.willis@softarmor.com>
Date: Tue, 31 Mar 2009 08:55:41 -0400
Thread-Topic: [Sip] francois' comments and why RFC4474 not used in the field
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Cc: SIP List <sip@ietf.org>, "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Jonathan Rosenberg
> Sent: Monday, March 30, 2009 5:26 PM
>=20
> Consider this litmus test:
> If the signaling actually came from Mary (perhaps as a third party), but
> the media goes/comes to/from Bob, who should appear on the caller ID? I
> say - Bob.

So what does 10.0.200.6:45001 mean to you?  Bob or Mary?
And since the SDP only indicates where to send your media to, not where it =
comes from, what does it mean to you to get a session request where the cal=
ler asks you to send media to 10.0.200.6:45001, but it can come from anywhe=
re?

The thing is an IP:port is NOT an Identity.  It's not a secure assertion of=
 anything.  Signing it doesn't *prove* anything.  It neither prevents media=
 spoofing injection nor interception.  It's only the signing of the dtls fi=
ngerprint by binding it to the signaling identity that provides strong "med=
ia" identity.

-hadriel

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dean Willis <dean.willis@softarmor.com>, Jiri Kuthan <jiri@iptel.org>
Date: Tue, 31 Mar 2009 09:02:25 -0400
Thread-Topic: [Sip] francois' comments and why RFC4474 not used in the field
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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Dea=
n
> Willis
> Sent: Tuesday, March 31, 2009 12:08 AM
>=20
> So, we need identity in signaling, we need identity in media, and we
> need the two identity expressions to match.
>=20
> That's what we get with RFC 4474 and DTLS/SRTP.
>=20
> Why were we trying to change it to protect something else?

Well, for one, you can get identity in signaling and identity in media usin=
g DTLS/SRTP *without* using RFC-4474 as currently defined.  We're trying to=
 get it to NOT protect things that don't need to be protected for identity =
to be attained.

-hadriel

From christer.holmberg@ericsson.com  Tue Mar 31 06:12:15 2009
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Thread-Topic: [Sip] I-D Inaction: draft-kaplan-sip-secure-call-id-00.txt
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>, "Hadriel Kaplan" <HKaplan@acmepacket.com>
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Subject: Re: [Sip] I-D Inaction: draft-kaplan-sip-secure-call-id-00.txt
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Hi,

I think Hadriel's proposal is fine.

Regards,

Christer=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Paul Kyzivat
> Sent: 30. maaliskuuta 2009 23:48
> To: Hadriel Kaplan
> Cc: SIP; Adam Roach
> Subject: Re: [Sip] I-D Inaction:=20
> draft-kaplan-sip-secure-call-id-00.txt
>=20
>=20
>=20
> Hadriel Kaplan wrote:
> > What's non-backward-compatible about it?  The text would=20
> say a UAC should generate a call-id without the IP/host=20
> (which is completely in-line with 3261), and that a UAS must=20
> continue to accept a call-id with ip/host.
>=20
> OK. I thought you were proposing to go further.
>=20
> What you say above here seems fine. (But of course some won't=20
> implement SHOULD either.)
>=20
> 	Thanks,
> 	Paul
>=20
> > I'll remove the ABNF change piece since people are worried=20
> that may affect backwards compatibility, due to a future=20
> strict-compliance parser checking for this update's ABNF and=20
> rejecting legacy 3261 ABNF.  So this update would only affect=20
> the normative text and not the ABNF.
> >=20
> > -hadriel
> >=20
> >> -----Original Message-----
> >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> >> Sent: Monday, March 30, 2009 12:55 PM
> >> To: Hadriel Kaplan
> >> Cc: Adam Roach; SIP
> >> Subject: Re: [Sip] I-D Inaction:=20
> >> draft-kaplan-sip-secure-call-id-00.txt
> >>
> >> Hadriel,
> >>
> >> I understand your logic. BUT are we agreed that we should=20
> be making=20
> >> non-backward-compatible normative changes to 3261 (or any=20
> of the key
> >> RFCs) for reasons other than correcting out and out errors and=20
> >> ambiguities? Seems like that is the start of SIP/3.0.
> >>
> >> 	Paul
> >>
> >> Hadriel Kaplan wrote:
> >>> Yeah I've been thinking about what you said the other day about=20
> >>> this,
> >> and I think I understand your point, but I still think=20
> MUST's and an=20
> >> update to 3261 would be good, for the following reasons:
> >>> 1) We really should want to change 3261 to say MUST NOT put an ip-
> >> address/hostname in the call-id.  Really in 20/20=20
> hindsight I don't=20
> >> think it should have had one to begin with.
> >>> 2) It's easier for customers to force vendors to do something if=20
> >>> there's
> >> a MUST statement, standards-track RFC, vs. an informational=20
> >> recommendation/should.
> >>> 3) Endpoint UAC vendors should do it *before* getting=20
> deployed and=20
> >>> being
> >> told by customers to change after-wards.  Specifically I'm=20
> thinking=20
> >> about vendors who don't attend IETF and aren't that up-to-speed on=20
> >> all the nuances and such.  An update to 3261 with=20
> normative mandatory=20
> >> language is clearer, I think.
> >>> -hadriel
> >>>
> >>>
> >>>> -----Original Message-----
> >>>> From: Adam Roach [mailto:adam@nostrum.com]
> >>>> Sent: Friday, March 27, 2009 4:20 PM
> >>>>
> >>>> Which is why an informational draft explaining the problem is=20
> >>>> useful, while a normative document... not quite so much.
> >>>>
> >>>> Tell UACs: "If you include machine names in your=20
> Call-IDs, B2BUAs=20
> >>>> will have reason to change them." There's no MUST or=20
> MUST NOT here,=20
> >>>> just an explanation of rationale.
> >>>>
> >>>> Tell B2BUAs: "If a Call-ID doesn't contain an @ sign, you'll do
> >> everyone
> >>>> a favor by not changing them, because it allows proper=20
> correlation=20
> >>>> of related transactions an dialogs." Again, no MUST; no=20
> MUST NOT.=20
> >>>> Just cause and effect.
> >>>>
> >>>> Then see if people want to play nice. The market will sort out if
> >> that's
> >>>> valuable or not.
> >>>>
> >>>> /a
> >>> _______________________________________________
> >>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> >>> This list is for NEW development of the core SIP Protocol Use=20
> >>> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >>> sipping@ietf.org for new developments on the application of sip
> >>>
> >=20
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

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Cc: SIP List <sip@ietf.org>, "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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Hans beat me to it with his reply - if there is more than one media 
stream (e.g. audio/video), and they come from different sources, then 
what to you display?

Dean says these and the signaling identity should match. Modulo the 
problems we are trying to solve with intermediaries, that will be the 
situation in the majority of cases. But the general case is:
- one, none, or maybe more than one, identity for the signaling
- one, none, (or maybe more than one?), identity for each media stream

With that jumble of stuff you need to decide what to present to the 
user. If all are the same then no problem. If you have a UI that permits 
display of them all, then no problem. (Except user confusion.) But if 
there is disagreement among the sources and you can only present one, 
then do you: pick one, or pick none?

At the moment that seems to be a common case. We may be able to get some 
sort of identity for one of these, but can't verify that it applies to 
all (where "all" usually means signaling + audio) and we can at most 
display one.

	Thanks,
	Paul

Dean Willis wrote:
> 
> On Mar 30, 2009, at 8:03 PM, Jiri Kuthan wrote:
>>
>> There is a timing aspect in favor of placing identity in signaling --
>> I would like to know whose call is ringing before I answer (if I do).
> 
> So, we need identity in signaling, we need identity in media, and we 
> need the two identity expressions to match.
> 
> That's what we get with RFC 4474 and DTLS/SRTP.
> 
> Why were we trying to change it to protect something else?
> 
> -- 
> Dean
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

From HKaplan@acmepacket.com  Tue Mar 31 06:26:59 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dean Willis <dean.willis@softarmor.com>
Date: Tue, 31 Mar 2009 09:27:50 -0400
Thread-Topic: [Sip] francois' comments and why RFC4474 not used in the field
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Cc: List <sip@ietf.org>, Francois Audet <audet@nortel.com>, SIP, Jon Peterson <jon.peterson@neustar.biz>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Monday, March 30, 2009 2:18 PM
> To: Hadriel Kaplan
>=20
> RFC 4474 can be used end to end. It all depends on which cert is used
> to sign the identity, where that signing occurs, and where it gets
> checked. For stupid legacy phones, it can be signed and checked at
> middleboxes operating under some level of transitive trust. For
> smarter devices, it happens in the endpoint.

Ummm.... nafaik.  If your AoR is of the form user@domain.com, then even if =
your UA endpoint has a cert and signs the request, any middlebox which has =
a cert for "domain.com", or "*.domain.com", can willy-nilly replace your si=
gnature and do lawful interception. =20


> Signing and checking at the endpoint, coupled with the DTLS
> fingerprint check. enables end-to-end verification of media integrity
> and privacy. Once you start ripping these signatures out in the middle
> of the network and replacing them, we have lost this absolutely
> critical property. If transit providers even think it MIGHT be
> reasonable to rip out the signatures, then we have lost the key
> functionality. Removing signatures MUST BE FORBIDDEN. Adding new ones
> might be ok.

I don't think anyone wants to remove signatures.  (of course mandating they=
 don't get removed is like mandating to set the evil bit if you're bad :)

-hadriel

From HKaplan@acmepacket.com  Tue Mar 31 07:26:05 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: SIP List <sip@ietf.org>
Date: Tue, 31 Mar 2009 10:26:51 -0400
Thread-Topic: secure-call-id vs. session-id
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Subject: [Sip] secure-call-id vs. session-id
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Howdy,
Several people asked me after the meeting about why we would need a session=
-id if people just implemented secure-call-id.  Clearly I didn't articulate=
 it well enough, so here goes another try...

ISTM, there are numerous "types" of B2BUA's in the World.  Some of those B2=
BUA's replace the Call-ID for a single reason: security/privacy, because th=
e Call-ID had an IP/host in it.  For example SBC's are of such a type, typi=
cally.  Using a secure-call-id should remove the incentive they have for re=
placing it, which will improve scenarios for dialog-matching in out-of-dial=
og requests. =20

There are also other B2BUA types, which replace the Call-ID for different r=
easons.  IP-PBX's, App-Servers, SoftSwitches, etc., I would put in that cam=
p.  I don't know why they replace the Call-ID, but clearly their designers =
feel they need to.  It may just be for strict compliance with RFC-3261, whi=
ch may well be the right thing to do for their case.  I believe the designe=
rs of those devices fully expect that the out-of-dialog request reaches the=
m, and that it's their Call-ID that is in it.  They just didn't expect *oth=
er* B2BUA's to be in the path, such as SBC's.

So the secure-call-id is to provide a better path forward for B2BAU's which=
 only change the call-id for the privacy reason. =20

For troubleshooting purposes, however, that's not enough.  People want to c=
orrelate the session as it crosses as many B2BUA's as possible in their log=
s/monitoring-equipment, and thus the session-id draft.

-hadriel


From L.Liess@telekom.de  Tue Mar 31 08:07:46 2009
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Subject: Re: [Sip] secure-call-id vs. session-id
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Another reason for the session-ID is that, different from the
secure-call-id, the mechanism  does not rely on the end devices.=20

So service providers=20
- can use it as soon as they implement it within their B2BUAs directly
connected to end devices and=20
-  can use it for tracing some kind of attacks (we do not expect that
people to upgrade their end devices soon)

Laura=20

=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Hadriel Kaplan
> Sent: Tuesday, March 31, 2009 4:27 PM
> To: SIP List
> Subject: [Sip] secure-call-id vs. session-id
>=20
> Howdy,
> Several people asked me after the meeting about why we would=20
> need a session-id if people just implemented secure-call-id. =20
> Clearly I didn't articulate it well enough, so here goes=20
> another try...
>=20
> ISTM, there are numerous "types" of B2BUA's in the World. =20
> Some of those B2BUA's replace the Call-ID for a single=20
> reason: security/privacy, because the Call-ID had an IP/host=20
> in it.  For example SBC's are of such a type, typically. =20
> Using a secure-call-id should remove the incentive they have=20
> for replacing it, which will improve scenarios for=20
> dialog-matching in out-of-dialog requests. =20
>=20
> There are also other B2BUA types, which replace the Call-ID=20
> for different reasons.  IP-PBX's, App-Servers, SoftSwitches,=20
> etc., I would put in that camp.  I don't know why they=20
> replace the Call-ID, but clearly their designers feel they=20
> need to.  It may just be for strict compliance with RFC-3261,=20
> which may well be the right thing to do for their case.  I=20
> believe the designers of those devices fully expect that the=20
> out-of-dialog request reaches them, and that it's their=20
> Call-ID that is in it.  They just didn't expect *other*=20
> B2BUA's to be in the path, such as SBC's.
>=20
> So the secure-call-id is to provide a better path forward for=20
> B2BAU's which only change the call-id for the privacy reason. =20
>=20
> For troubleshooting purposes, however, that's not enough. =20
> People want to correlate the session as it crosses as many=20
> B2BUA's as possible in their logs/monitoring-equipment, and=20
> thus the session-id draft.
>=20
> -hadriel
>=20
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

From dean.willis@softarmor.com  Tue Mar 31 09:40:22 2009
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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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On Mar 31, 2009, at 7:55 AM, Hadriel Kaplan wrote:

>
>
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
>> Jonathan Rosenberg
>> Sent: Monday, March 30, 2009 5:26 PM
>>
>> Consider this litmus test:
>> If the signaling actually came from Mary (perhaps as a third  
>> party), but
>> the media goes/comes to/from Bob, who should appear on the caller  
>> ID? I
>> say - Bob.
>
> So what does 10.0.200.6:45001 mean to you?  Bob or Mary?

It means "Not Dean's home phone, which I know uses a 192  
address . . ." If I knew Bob and Mary well, I might have a clue there  
too.

Not perfect, but better than nothing.

Enough better than nothing to bother with? I don't know. I'd like to  
hear Jon's take on that.

--
Dean




From dean.willis@softarmor.com  Tue Mar 31 09:42:05 2009
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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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On Mar 31, 2009, at 8:02 AM, Hadriel Kaplan wrote:

>
> Well, for one, you can get identity in signaling and identity in  
> media using DTLS/SRTP *without* using RFC-4474 as currently  
> defined.  We're trying to get it to NOT protect things that don't  
> need to be protected for identity to be attained.

umm, I believe RFC 4474 is what correlates the identity-in-media with  
the identity-in-signaling. The media identity expression is not  
particularly useful without its correlation to signaling and thence to  
a "human readable" identity expression.

--
Dean


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dean Willis <dean.willis@softarmor.com>
Date: Tue, 31 Mar 2009 12:51:35 -0400
Thread-Topic: [Sip] francois' comments and why RFC4474 not used in the field
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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Tuesday, March 31, 2009 12:43 PM
> To: Hadriel Kaplan
>=20
> On Mar 31, 2009, at 8:02 AM, Hadriel Kaplan wrote:
>=20
> > Well, for one, you can get identity in signaling and identity in
> > media using DTLS/SRTP *without* using RFC-4474 as currently
> > defined.  We're trying to get it to NOT protect things that don't
> > need to be protected for identity to be attained.
>=20
> umm, I believe RFC 4474 is what correlates the identity-in-media with
> the identity-in-signaling. The media identity expression is not
> particularly useful without its correlation to signaling and thence to
> a "human readable" identity expression.


Right, sorry I should have been clearer.  You had said:
> That's what we get with RFC 4474 and DTLS/SRTP.
> Why were we trying to change it to protect something else?

I meant we don't have to use 4474 in particular to accomplish the same goal=
 - for example, any one of the other proposals thus far provide you that sa=
me binding.  Just because 4474 gets you that binding, doesn't mean another =
mechanism can't, nor that changing what 4474 protects removes that binding.=
  We're trying to protect only what actually needs to be protected to get y=
ou that binding.

-hadriel

From dean.willis@softarmor.com  Tue Mar 31 09:51:55 2009
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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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On Mar 31, 2009, at 8:27 AM, Hadriel Kaplan wrote:
>>
>> RFC 4474 can be used end to end. It all depends on which cert is used
>> to sign the identity, where that signing occurs, and where it gets
>> checked. For stupid legacy phones, it can be signed and checked at
>> middleboxes operating under some level of transitive trust. For
>> smarter devices, it happens in the endpoint.
>
> Ummm.... nafaik.  If your AoR is of the form user@domain.com, then  
> even if your UA endpoint has a cert and signs the request, any  
> middlebox which has a cert for "domain.com", or "*.domain.com", can  
> willy-nilly replace your signature and do lawful interception.

Not quite.

Here's how I understand it to work:

If the signing cert is user@domain.com instead of domain.com, then the  
middlebox would need to work with a CA to mint a cert having that full  
name. A more typical scenario would have the middlebox use a cert with  
name "domain.com". The verifying node can tell the difference.

But even better, with RFC 4474 I might have a connection to AT&T. I  
might use this connection to send a call with an Identity header for "dean.willis@softarmor.com 
". If you have any relationship with me at all, you know AT&T isn't  
likely to have the private key for the softarmor.com CA certificate or  
for my private certificate issued by that CA. So AT&T isn't a MITM  
attacker in this scenario.

Of course, if AT&T's accursed SBC strips off my RFC 4474 Identity  
header and inserts their own useless declamation asserting some bogus  
phone number then my assertion is useless, we could have a MITM, and  
I'm changing carriers.

--
dean



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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dean Willis <dean.willis@softarmor.com>
Date: Tue, 31 Mar 2009 13:14:56 -0400
Thread-Topic: [Sip] francois' comments and why RFC4474 not used in the field
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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Tuesday, March 31, 2009 12:53 PM
>=20
> Here's how I understand it to work:
>=20
> If the signing cert is user@domain.com instead of domain.com, then the
> middlebox would need to work with a CA to mint a cert having that full
> name. A more typical scenario would have the middlebox use a cert with
> name "domain.com". The verifying node can tell the difference.

Not as I read 4474.  The verifier will be getting a From AoR of user@domain=
.com, and a cert for domain.com.  That passes the verification check.  The =
verifier had no idea that the original cert was for "user@domain.com", beca=
use it won't be downloading that cert, because the Identity-Info header wou=
ld have been changed to point to the domain-wide cert, and the signature wo=
uld have been recalculated using the domain-wide cert. (assuming a middlebo=
x in the domain wanted to do this)

Again, this isn't really a problem - presumably you trust your own domain. =
 I was just pointing out that the protected portion of this is domain-to-do=
main, not end-to-end.  Most people are fine with that, as am I.

-hadriel

From jdrosen@cisco.com  Tue Mar 31 10:33:15 2009
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Subject: Re: [Sip] Media vs Signaling identity (was Re: francois' comments and why RFC4474 not used in the field)
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inline:

Victor Pascual Ã�vila wrote:
> On Tue, Mar 31, 2009 at 4:12 AM, Jonathan Rosenberg <jdrosen@cisco.com> wrote:
>> inline:
>>
>> Jiri Kuthan wrote:
>>
>>
>>>>  From an end user perspective, I would assert that the most important
>>>> thing is probably the media. If the callerID says, "this is bob", what is
>>>> important to the user, is that when I pick up the phone and start talking,
>>>> it will be Bob who hears me, and Bob that I hear.
>>>>
>>>> Consider this litmus test:
>>>>
>>>> If the signaling actually came from Mary (perhaps as a third party), but
>>>> the media goes/comes to/from Bob, who should appear on the caller ID? I say
>>>> - Bob.
>>> There is a timing aspect in favor of placing identity in signaling --
>>> I would like to know whose call is ringing before I answer (if I do).
>> You can still have that. Just don't ring the phone until early media has
>> been exchanged and verified. Indeed if you were doing an ICE-style thing per
>> Dan's draft, you'd get that for free.
> 
> Are we restricting the identity assertion to telephony-like sessions?
> 
> IMHO, identity assertion should also work for the following scenarios
> (among others):
> 
> -rfc3725, figure 1, message 1 (INVITE no SDP): Upon receipt of the
> initial INVITE (note there's no session description at all), "A"
> decides to authorize or reject the call based on the delivered
> identifier.

The identity is delivered via the signaling; you just need the media to 
verify it. This then enables a 'trust but verify model' - an attacker 
might get that initial call in but will be immediately detected and 
rejected when their media comes through.


> 
> -rfc3428, figure 1, message 2 (F2): "user2" decides to answer or
> ignore the message based on the delivered identifier.

Its not obvious to me at this point, that we should be using exactly the 
same solution for MESSAGE as INVITE-based media sessions.

The general idea with 'media' is that the media stream itself - the true 
  user content - is what we are binding the identity to. User content is 
more difficult to modify because it is detectable by the user's 
themselves and disrupts the user's perception of the service. In the 
case of MESSAGE, it would be the content of the MESSAGE that we'd 
correlate the identity with. In that way, it doesn't matter what 
manipulations were made on any other aspect of the MESSAGE, but as long 
as the content is not manipulated - the identity matches.

For MSRP, something akin to DTLS-SRTP could be done, since its a media 
stream like RTP.



> 
> -rfc3515, example 4.1, message 1 (F1): "agent B" decides to accept or
> reject the refer based on the delivered identifier

This one is tricky. Arguably the thing that represents the 'user 
content' and should be immutable is the Refer-To URI, but I have no 
doubt that b2bua muck with that header too.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   111 Wood Avenue South
Cisco Fellow                                   Iselin, NJ 08830
Cisco, Voice Technology Group
jdrosen@cisco.com
http://www.jdrosen.net                         PHONE: (408) 902-3084
http://www.cisco.com

From fluffy@cisco.com  Tue Mar 31 15:30:50 2009
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From: Cullen Jennings <fluffy@cisco.com>
To: Jiri Kuthan <jiri@iptel.org>
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Cc: "sip@ietf.org" <sip@ietf.org>, Francois Audet <audet@nortel.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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On Mar 28, 2009, at 2:02 PM, Jiri Kuthan wrote:

>
> I'm worried this is only a wishful thinking. While perfectly
> logical, still even in such constrained setups some bizzar
> ALGs do in my experience appear in the middle, change SDP
> and make thus the identity worthless.

The thing I keep asking is can we make a list of reason why SDP and  
headers get changes and in what scenarios they do this. I think it  
will be hard to sort how to fix this without being clear what needs to  
be fixed.

For example, one of the things we might want to say is something like:  
the Phone is behind a NAT and connects to it's proxy / registrar for  
its' domain. That proxy/b2bua whatever mucks with IP/ports in the SDP  
for NAT traversal. Then we could ask if 4474 is broken in this case or  
not and what might be a good way of solving the problem of having UAs  
behind NATs.

Cullen <in my individual contributor role>


From drage@alcatel-lucent.com  Tue Mar 31 18:00:26 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: Cullen Jennings <fluffy@cisco.com>, Jiri Kuthan <jiri@iptel.org>
Date: Wed, 1 Apr 2009 03:00:54 +0200
Thread-Topic: [Sip] francois' comments and why RFC4474 not used in the field
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Cc: "sip@ietf.org" <sip@ietf.org>, Francois Audet <audet@nortel.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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Cullen wrote:

> The thing I keep asking is can we make a list of reason why=20
> SDP and headers get changes and in what scenarios they do=20
> this. I think it will be hard to sort how to fix this without=20
> being clear what needs to be fixed.
> =20

Isn't that the function of the text that was placed in section 8 of

http://tools.ietf.org/html/draft-elwell-sip-e2e-identity-important-03

If it is not, then what do you think is missing.

regards

Keith

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Cullen Jennings
> Sent: Tuesday, March 31, 2009 11:32 PM
> To: Jiri Kuthan
> Cc: sip@ietf.org; Francois Audet
> Subject: Re: [Sip] francois' comments and why RFC4474 not=20
> used in the field
>=20
>=20
> On Mar 28, 2009, at 2:02 PM, Jiri Kuthan wrote:
>=20
> >
> > I'm worried this is only a wishful thinking. While=20
> perfectly logical,=20
> > still even in such constrained setups some bizzar ALGs do in my=20
> > experience appear in the middle, change SDP and make thus=20
> the identity=20
> > worthless.
>=20
> The thing I keep asking is can we make a list of reason why=20
> SDP and headers get changes and in what scenarios they do=20
> this. I think it will be hard to sort how to fix this without=20
> being clear what needs to be fixed.
>=20
> For example, one of the things we might want to say is=20
> something like: =20
> the Phone is behind a NAT and connects to it's proxy /=20
> registrar for its' domain. That proxy/b2bua whatever mucks=20
> with IP/ports in the SDP for NAT traversal. Then we could ask=20
> if 4474 is broken in this case or not and what might be a=20
> good way of solving the problem of having UAs behind NATs.
>=20
> Cullen <in my individual contributor role>
>=20
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
> =

From fluffy@cisco.com  Tue Mar 31 18:23:11 2009
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From: Cullen Jennings <fluffy@cisco.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
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Cc: sip@ietf.org, Francois Audet <audet@nortel.com>
Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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In all of section 8, the only reason I find of why things are changed is

"For media steering purposes, B2BUAs in intermediate domains need to  
modify the IP address c-lines and the port in m-lines."
Even this one line leaves me confused about what intermediate domains  
are in the deployment cases where this happens and if we are talking  
about email style address or e.164 style addresses. What an example is  
of a deployment where things like this happen and why the middle (not  
the end domains) do the media steering. When I ask this of example  
people give me, if often turns out what is really wanted is not media  
steering but the middle to be able to hide the fact that they actually  
delivered the call to a third provider for PSTN termination.

Pretend 4474 does not even exist for a minute. I don't think it is  
unrealizable of me to ask what the problem is we are trying to solve.  
Saying 4474 is does not solve the problem may or may not be true but  
we are unlikely to have a good conversation about about what we should  
do until we understand what we are trying to accomplish.


On Mar 31, 2009, at 7:00 PM, DRAGE, Keith (Keith) wrote:

> Cullen wrote:
>
> > The thing I keep asking is can we make a list of reason why
> > SDP and headers get changes and in what scenarios they do
> > this. I think it will be hard to sort how to fix this without
> > being clear what needs to be fixed.
> >
>
> Isn't that the function of the text that was placed in section 8 of
>
> http://tools.ietf.org/html/draft-elwell-sip-e2e-identity-important-03
>
> If it is not, then what do you think is missing.
>
> regards
>
> Keith
>
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> > Behalf Of Cullen Jennings
> > Sent: Tuesday, March 31, 2009 11:32 PM
> > To: Jiri Kuthan
> > Cc: sip@ietf.org; Francois Audet
> > Subject: Re: [Sip] francois' comments and why RFC4474 not
> > used in the field
> >
> >
> > On Mar 28, 2009, at 2:02 PM, Jiri Kuthan wrote:
> >
> > >
> > > I'm worried this is only a wishful thinking. While
> > perfectly logical,
> > > still even in such constrained setups some bizzar ALGs do in my
> > > experience appear in the middle, change SDP and make thus
> > the identity
> > > worthless.
> >
> > The thing I keep asking is can we make a list of reason why
> > SDP and headers get changes and in what scenarios they do
> > this. I think it will be hard to sort how to fix this without
> > being clear what needs to be fixed.
> >
> > For example, one of the things we might want to say is
> > something like:
> > the Phone is behind a NAT and connects to it's proxy /
> > registrar for its' domain. That proxy/b2bua whatever mucks
> > with IP/ports in the SDP for NAT traversal. Then we could ask
> > if 4474 is broken in this case or not and what might be a
> > good way of solving the problem of having UAs behind NATs.
> >
> > Cullen <in my individual contributor role>
> >
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use
> > sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >


From Martin.Thomson@andrew.com  Tue Mar 31 20:02:56 2009
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From: "Thomson, Martin" <Martin.Thomson@andrew.com>
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From dean.willis@softarmor.com  Tue Mar 31 23:51:55 2009
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From: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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On Mar 31, 2009, at 5:31 PM, Cullen Jennings wrote:

>
> On Mar 28, 2009, at 2:02 PM, Jiri Kuthan wrote:
>
>>
>> I'm worried this is only a wishful thinking. While perfectly
>> logical, still even in such constrained setups some bizzar
>> ALGs do in my experience appear in the middle, change SDP
>> and make thus the identity worthless.
>
> The thing I keep asking is can we make a list of reason why SDP and  
> headers get changes and in what scenarios they do this. I think it  
> will be hard to sort how to fix this without being clear what needs  
> to be fixed.
>
> For example, one of the things we might want to say is something  
> like: the Phone is behind a NAT and connects to it's proxy /  
> registrar for its' domain. That proxy/b2bua whatever mucks with IP/ 
> ports in the SDP for NAT traversal. Then we could ask if 4474 is  
> broken in this case or not and what might be a good way of solving  
> the problem of having UAs behind NATs.
>

You seem to be trying to construct an argument for "why 4474 isn't  
broken" based on being able to transform every use case where RFC 4474  
breaks to one where it doesn't, generally by mandating a different  
approach to solving problems (TURN vs, SDP fixup, for example). You  
seem to expect this to prove that RFC 4474 is fine as-is, needing no  
modification to its A/M signing.

It might be useful to counter this with "If RFC 4474 isn't broken, why  
isn't it being deployed?" I expect that the answer is "because users  
currently expect to correct their topological issues using SBCs  
instead of the tools that work with 4474. You might not think they  
have legitimate reasons for doing it this way, but the fact remains --  
that's how they do it. Technically correct or not, ordering this to  
flow out is likely to give you wet feet.

--
Dean


From dean.willis@softarmor.com  Tue Mar 31 23:57:24 2009
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From: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: [Sip] francois' comments and why RFC4474 not used in the field
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On Mar 31, 2009, at 8:24 PM, Cullen Jennings wrote:

>
> In all of section 8, the only reason I find of why things are  
> changed is
>
> "For media steering purposes, B2BUAs in intermediate domains need to  
> modify the IP address c-lines and the port in m-lines."
> Even this one line leaves me confused about what intermediate  
> domains are in the deployment cases where this happens and if we are  
> talking about email style address or e.164 style addresses. What an  
> example is of a deployment where things like this happen and why the  
> middle (not the end domains) do the media steering. When I ask this  
> of example people give me, if often turns out what is really wanted  
> is not media steering but the middle to be able to hide the fact  
> that they actually delivered the call to a third provider for PSTN  
> termination.
>

I'll agree that topology hiding is a use case for port/address  
obfuscation that results in RFC 4474 not working, thereby preventing  
DTLS/SRTP key fingerprinting when topology-hiding occurs.

Design alternatives include tunnels and STUN relays combined with  
header obfuscation  and UA support for selective obfuscation for the  
offer. While these sorts of triggerworks might work, they're a lot  
more complex than just editing the SDP.

--
Dean

