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Date: Mon, 02 Dec 2002 09:35:53 +0530
From: Prasanna Venkatesh <prasanna@huawei.com>
Subject: RE: [Sip] B2BUA
In-reply-to: <000f01c29825$4ed3a3e0$0100a8c0@txdwillis>
To: Dean Willis <dean.willis@softarmor.com>, Juan-Carlos.Rojas@alcatel.fr,
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May be in place of "n" "two or more" is better..

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Dean
Willis
Sent: Saturday, November 30, 2002 9:32 AM
To: Juan-Carlos.Rojas@alcatel.fr; sip@ietf.org
Subject: RE: [Sip] B2BUA



Juan-Carlos said:
> Back-to-Back User Agent: A SIP element being able to handle a 
> collection of "n" User Agents (behaving each one as UAC and 
> UAS, according to SIP rules), which are linked by some 
> application logic that is fully independent of the SIP protocol rules.

This seems like a reasonable definition. 

Poll: Would it be worth including this in 3261bis? Do we need to say
N>1?

--
Dean



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From mailnull@www1.ietf.org  Sun Dec  1 23:10:55 2002
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Date: Mon, 02 Dec 2002 09:38:01 +0530
From: Prasanna Venkatesh <prasanna@huawei.com>
Subject: RE: [Sip] different messages with same call-id before 100 is sent out
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Hi Gonzalo,
    I also agree to Mike's suggestion.  Can't a recommendation be made on
the INGRESS GW that the CSeq for an overlapped dial message should
monotonically increase, though each of the INVITEs constitute a different
dialog.  This can help the EGRESS GW to drop out-of-order INVITEs.
Regards,
Prasanna
Huawei Technologies.
-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
Mpierce1@aol.com
Sent: Friday, November 29, 2002 11:38 PM
To: Gonzalo.Camarillo@lmf.ericsson.se; prasanna@huawei.com
Cc: rshiroor@aplion.stpn.soft.net; sip@ietf.org
Subject: Re: [Sip] different messages with same call-id before 100 is sent
out


In a message dated 11/28/2002 5:08:11 AM Eastern Standard Time,
Gonzalo.Camarillo@lmf.ericsson.se writes:



in the overlap case we do not care if the nth INVITE reaches the other
end before the (n-1)th INVITe, so you do not have to wait for anything
before sending the new INVITE.




[MAP] This may be true as long as the "other end" knows to throw away the
old INVITE (with fewer digits). However, I believe that the very fact that
INVITE's can arrive out of order leads to significant difficulty, since
their correct interpretation depends on the order.

Going back to some earlier comments, I believe there must be a requirement
that, for the PSTN-IP-PSTN case (which I think is the focus in this whole
discussion), that a single IAM received on the ingress side (of IP) must
result in exactly one IAM being sent on the egress side back to the PSTN.
There may be fewer (or no) SAMs sent than received, for example, if the IP
portion, at either the ingress or egress GW, does overlap to en-bloc
operation, or simply has the smarts to put multiple digits (from SAMs)
together.

I believe that a solution to meeting this requirement (single IAM) has a
significant impact on all the other discussions about out-of-order INVITES,
same or different Call-IDs, etc. Therefore, unless it can be agreed that
this is a requirement, all the other discussion is meaningless.

I raise this point especially with relation to my proposals in IEPREP for a
priority-based service for a military application which includes preemption
in the circuit switched world using SS#7 (MLPP). Multiple IAMs (for high
priority calls) would have the unacceptable result of preempting calls and
then releasing the IAM when another arrived (or took another path).

Is it possible to agree that this single IAM rule should be followed?

Mike Pierce
Artel

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Hi,

It has been explained several times why you cannot force the subsequent
INVITEs to follow the rute that the first one followed. You can find the
discussions in the archives or in the slides I presented in Australia.

Thanks,

Gonzalo
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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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To: Prasanna Venkatesh <prasanna@huawei.com>
CC: Mpierce1@aol.com, rshiroor@aplion.stpn.soft.net, sip@ietf.org,
        "Sip-Implementors@Cs.Columbia.Edu" <sip-implementors@cs.columbia.edu>
Subject: Re: [Sip] different messages with same call-id before 100 is sent out
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Hi,

The recommendation about using increasing Cseqs is already in the draft.

5th paragraph of section 3.3

Gonzalo

Prasanna Venkatesh wrote:
> 
> Hi Gonzalo,
>     I also agree to Mike's suggestion.  Can't a recommendation be made on
> the INGRESS GW that the CSeq for an overlapped dial message should
> monotonically increase, though each of the INVITEs constitute a different
> dialog.  This can help the EGRESS GW to drop out-of-order INVITEs.
> Regards,
> Prasanna
> Huawei Technologies.
> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> Mpierce1@aol.com
> Sent: Friday, November 29, 2002 11:38 PM
> To: Gonzalo.Camarillo@lmf.ericsson.se; prasanna@huawei.com
> Cc: rshiroor@aplion.stpn.soft.net; sip@ietf.org
> Subject: Re: [Sip] different messages with same call-id before 100 is sent
> out
> 
> In a message dated 11/28/2002 5:08:11 AM Eastern Standard Time,
> Gonzalo.Camarillo@lmf.ericsson.se writes:
> 
> in the overlap case we do not care if the nth INVITE reaches the other
> end before the (n-1)th INVITe, so you do not have to wait for anything
> before sending the new INVITE.
> 
> [MAP] This may be true as long as the "other end" knows to throw away the
> old INVITE (with fewer digits). However, I believe that the very fact that
> INVITE's can arrive out of order leads to significant difficulty, since
> their correct interpretation depends on the order.
> 
> Going back to some earlier comments, I believe there must be a requirement
> that, for the PSTN-IP-PSTN case (which I think is the focus in this whole
> discussion), that a single IAM received on the ingress side (of IP) must
> result in exactly one IAM being sent on the egress side back to the PSTN.
> There may be fewer (or no) SAMs sent than received, for example, if the IP
> portion, at either the ingress or egress GW, does overlap to en-bloc
> operation, or simply has the smarts to put multiple digits (from SAMs)
> together.
> 
> I believe that a solution to meeting this requirement (single IAM) has a
> significant impact on all the other discussions about out-of-order INVITES,
> same or different Call-IDs, etc. Therefore, unless it can be agreed that
> this is a requirement, all the other discussion is meaningless.
> 
> I raise this point especially with relation to my proposals in IEPREP for a
> priority-based service for a military application which includes preemption
> in the circuit switched world using SS#7 (MLPP). Multiple IAMs (for high
> priority calls) would have the unacceptable result of preempting calls and
> then releasing the IAM when another arrived (or took another path).
> 
> Is it possible to agree that this single IAM rule should be followed?
> 
> Mike Pierce
> Artel

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo
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Hi,

According to RFC 3261,  a request is a merged request if ...

"8.2.2.2 Merged Requests

    If the request has no tag in the To header field, the UAS core MUST
    check the request against ongoing transactions.  If the From tag,
    Call-ID, and CSeq exactly match those associated with an ongoing
    transaction, but the request does not match that transaction (based
    on the matching rules in Section 17.2.3), the UAS core SHOULD
    generate a 482 (Loop Detected) response and pass it to the server
    transaction."

Suppose if there are 2 UAS application sharing the same SIP stack
(same process). They are listening for incoming calls for the same
user. One application is handling the fax machine for user_a and the
other the phone for user_a.

user_a's fax machine can be reached at  fax_user_a@test.com
And user_a's phone can be reached at    phone_user_a@test.com


A call made to user_a@test.com gets forked at a proxy. The
2 forked request arrives at the UAS with the respective request
URI's

INVITE sip:fax_user_a@test.com
...

INVITE sip:phone_user_a@test.com
...

Now, according to the RFC, the first INVITE would be processed and the
second would be responded with 482 "loop detected". Shouldn't the
second INVITE also be processed? Logically, these are 2 different
application handling 2 different devices for the same user. The fact that
they are sharing the same process should not interfere each other.

I think merged request matching should also take the Request-Uri
into consideration. Request-Uri  identifies the end point at UAS. If
a UAS gets 2 INVITES  for the same request-uri with the same Call-ID,
Cseq and From-Tag, and with no To-Tag then it should be considered
as merged request.

Thanks
Sachin

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HI

RFC 3261
16.4 section.

If the Request-URI contains a maddr parameter, the proxy MUST check
to see if its value is in the set of addresses or domains the proxy
is configured to be responsible for. If the Request-URI has a maddr
parameter with a value the proxy is responsible for, and the request
was received using the port and transport indicated (explicitly or by
default) in the Request-URI, the proxy MUST strip the maddr and any
non-default port or transport parameter and continue processing as if
those values had not been present in the request.

A request may arrive with a maddr matching the proxy, but on a
port or transport different from that indicated in the URI. Such
a request needs to be forwarded to the proxy using the indicated
port and transport.

What if the Request URI is like this
 sips:abc@def.com:33445;transport=tcp

and  if request is received at def.com on TCP. Should we forward the request on
TLS??

or

 if request is received at def.com on TLS. since transport is TCP......Should we
 forward the request on TCP.

With Regds
Ajay Kumar



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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        Adam Roach
	 <adam@dynamicsoft.com>
Cc: "'Dean Willis'" <dean.willis@softarmor.com>, sip@ietf.org
Subject: RE: [Sip] DISCUSS on Service Route Discovery
Date: Mon, 2 Dec 2002 07:42:26 -0500 
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I do agree that there are meaningful threats against registrars that emerge
when a registering UAC cannot establish a direct, secure (per RFC3261, TLS)
connection to the registrar. While service disco is not the first draft to
condone architectures in which there are intermediaries in the registration
process, I do think it is a reasonable place to stage some security
considerations that focus on this case. It could be said that the RFC3261
security philosophy is "love it or leave it" - we show some architectures in
which the mechanisms we recommend will work, and the price of deviation from
these architectures is deriving your security properties from some other
mechanisms. Perhaps this leaves the ball squarely in the court of I-Ds using
different architectures.

As I'm sure we all recall, during the RFC3261 review, the IESG suggested
that over time, the normative language about S/MIME in the SIP standard
might have to be strengthened. I don't think there's anything wrong with
revisiting our security situation incrementally, as we come to understand
more vulnerabilities in the light of proposed architectures or deployments
in I-Ds that go before the SIP groups. Security considerations seem to be
one of the few places in which the IESG welcomes redundancy - if multiple
documents that consider these sorts of architectures need to have more or
less identical language to this effect, I don't think that will be a problem
(not Obviously Wrong, anyway).

So from my perspective, I support the change recommended by the Discuss.
RFC3261 is what it is - when we have the opportunity to reconsider some of
its statutes, we will undoubtedly incorporate a number of changes influenced
by drafts that have passed in the interim. Perhaps we can say that we're
issuing an IOU to revisiting registrar security in the base spec later.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Wednesday, November 27, 2002 6:28 AM
> To: Adam Roach
> Cc: 'Dean Willis'; sip@ietf.org
> Subject: Re: [Sip] DISCUSS on Service Route Discovery
> 
> 
> inline.
> 
> Adam Roach wrote:
> >>-----Original Message-----
> >>From: Dean Willis [mailto:dean.willis@softarmor.com]
> >>
> >>The second concern seems to be more interesting. Is it reasonable to
> >>mandate that registrars implementing the Service-Route header 
> >>field also
> >>MUST implement S/MIME? This doesn't say they have to USE 
> it, just that
> >>they must IMPLEMENT it. 
> >>
> >>My personal feeling is that implementing this in a 
> registrar is fairly
> >>straightforward, and even the key management is easy, so it "does no
> >>harm" to include as a MUST.
> > 
> > 
> > I'm inclined to agree; first, though, I'd like to 
> understand the threat
> > posed by a registrar that doesn't/can't use S/MIME versus one that
> > can. Not in the general context of registering, of course -- only
> > in the context of the extra functionality introduced by srvrtdisco.
> > 
> > It's not obvious to me that this point is addressing an issue raised
> > by *this* draft, versus a general issue with SIP -- and it would
> > be Obviously Wrong (tm) to fix problems in the base spec in every
> > other document that mentions it, as opposed to just recording it
> > as a bug and fixing it next go round.
> 
> I agree. I don't believe the threats specific to service 
> route are worse 
> than the general threats imposed through impersonating a registrar, 
> which are explicitly called out in section 26.1.2 of bis. Indeed, the 
> insertion of a fake 301 response to a REGISTER request is much more 
> damaging than a fake service route, I believe.
> 
> So, if we think that S/MIME is a MUST to protect service 
> route, then its 
> also a MUST for rfc 3261.
> 
> I suppose the counter-argument might be this. Bis more or 
> less assumes 
> that there is the possibility of a direct connection between 
> the client 
> and registrar, as it talks about using TLS to secure registrations 
> (Section 26.3.2.1). Almost by definition, that approach 
> doesn't work for 
> the problem space of service route, because of the existence 
> of proxies 
> between the UA and the registrar, not a common case. Thus, 
> this draft is 
> addressing a specific architecture where the normal bis security 
> recommendations don't apply, and thus a different 
> recommendation is needed.
> 
> I'm not entirely sure I believe this counter-argument....
> 
> -Jonathan R.
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
> _______________________________________________
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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
> 
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From mailnull@www1.ietf.org  Mon Dec  2 08:18:37 2002
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'stoshniwal@hss.hns.com'" <stoshniwal@hss.hns.com>, sip@ietf.org
Subject: RE: SIPS related clarifications (was: [Sip] SIPS question)
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We did discuss this issue a bit in Atlanta. Unfortunately, I think all we
agreed on is that we all need to think about it some more.

Some more specific suggestions are offered below.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: stoshniwal@hss.hns.com [mailto:stoshniwal@hss.hns.com]
> Sent: Thursday, November 28, 2002 7:09 AM
> To: sip@ietf.org
> Subject: SIPS related clarifications (was: [Sip] SIPS question)
> 
> 
> 
> 
> 
> Hi all,
> 
> There were couple of interesting questions raised in
> this thread by Christer earlier this month that perhaps
> got left out in the middle of the conversation. Wanted
> to get my understanding regarding inter-mingling of
> "sip:" and "sips:" URLs within the spec clear. Sorry
> if any questions had got resolved earlier and I missed
> the mail as I'm entering the discussion late. Any
> pointers are welcome.
> 
> * Is it allowed for a SIP proxy that got a request on
>   TLS to forward the request downstream on a non-TLS
>   protocol? 

Depends on what you mean by 'got a request'. This also doesn't say whether
or not SIPS appears in the Request-URI or not; it is possible for a proxy
server to receive a request over TLS that does not have the SIPS URI scheme
in the Request-URI.

> Jon Peterson had suggested that "sips" was
>   designed to allow selective hops in the path to be
>   secure,

I suggested that SIPS could appear, for example, in a Route header, in which
case it would only be intended to secure a particular hop.

>  while Rosenberg had mentioned that the presence
>   of "sips:" should imply that the end-to-end path is
>   secure.

If SIPS appears in the Request-URI of the original request sent by the UAC,
it implies this end-to-end path security.

I don't think these two ways of using the SIPS URI are mutually exclusive -
we don't have to say that SIPS is only usable in one or the other of these
contexts.

>  I guess the way I understand the spec stands
>   at present is that if the Req-URI is re-written to a
>   "sips:" URL at any stage of the routing, then any hop
>   downstream _from_this_hop_onwards_ has to forward the
>   request on TLS.

I agree, that is how we should understand it - with the caveat that SIPS
only applies to the entire path until it reaches the domain indicated by the
hostname portion of the Request-URI. Once a proxy server in the destination
gets hold of the request, it can retarget it arbitrarily (to non-SIPS
resources, if it wishes).

Also note that proxies should not 're-write' the Request-URI of a SIP
request at just any stage of the routing. In fact, there are rules in
RFC3261 about when the Request-URI can be retargeted, and they are somewhat
restrictive.

>  The entire path (from UA to UA) need
>   not be secure. Is this correct?

Yes.

>  Or can the request
>   path be set-up with some hops on TLS and others on
>   non-TLS in any combination of them?
> 

Not when the Request-URI contains a SIPS URI.

> * What is the response code to be used by a UAS to
>   inform the UAC that it is not willing to accept
>   insecure calls and that the UAC should retry the
>   request on TLS?
> 

Although 416 might look like an attractive candidate, the language in
8.1.3.5 makes it a little ugly. I would suggest 403.

> * Also, another question Christer had raised in the
>   discussion was end-points should not be allowed to
>   send REGISTER with AoR as "sip:" and Contacts as "sips:"
>   since end-to-end security (if that's what "sips:"
>   implies always) would not be assured if a request
>   for that end-point came with Req-URI of "sip:". Is
>   this the case?
> 

Mr. Holmberg's cases got a lot nastier than that - for example, if I use a
SIP URI scheme in the Contact header I send when I register, but a SIPS URI
scheme in the Contact header that I send in (say) 200 OK responses to
requests. This creates some tricky cases, ones we are still mulling, I
think. Just for the sake of simplicity, it might be nice to forbid this, but
the spec doesn't forbid it today.

> * Finally, the conclusion of the earlier thread seemed
>   to be that the last sentence of Section 8.1.2
>   (paragraph 1) is incorrect and the UAC's choice for
>   the transport protocol should not rest on the req-uri's
>   scheme being "sips:" alone. But this was not logged
>   into bugzilla for resolution? Did I miss something?
> 

I think we do need to fix the 8.1.2 text, yes. We should log this into
bugzilla. I think the only reason this may not have happened yet is that the
answer to all of this may rest on some more fundamental principles that we
want to consider further before we propose any new language. 

> Thanks for your time.
> 
> Regards,
> Siddharth.
> 
> --------------------------------------------------------
> Siddharth Toshniwal
> Hughes Software Systems
> Prestige Opal                    http://www.hssworld.com
> 146, Infantry Road          Ph (O): +91-80-2286390 (7094)
> Bangalore-560001, India           Mobile: +91-9845154068
> --------------------------------------------------------
> 
> 
> _______________________________________________
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> 
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Subject: RE: [Sip] B2BUA
To: "Dean Willis" <dean.willis@softarmor.com>
Cc: <sip@ietf.org>
Date: Mon, 2 Dec 2002 14:39:14 +0100
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Dean et al,

The main reason to propose this definition is to clearly precise that User
Agents being involved in a B2BUA are "linked by some application logic that
is fully independent of the SIP protocol rules"
This is currently not precised in RFC 3261, and sometimes brings confusion
on what can or cannot be done by a B2BUA.

Best regards
Juan Carlos





"Dean Willis" <dean.willis@softarmor.com> on 30/11/2002 05:02:10

To:   Juan-Carlos ROJAS/FR/ALCATEL@ALCATEL, <sip@ietf.org>
cc:
Subject:  RE: [Sip] B2BUA



Juan-Carlos said:
> Back-to-Back User Agent: A SIP element being able to handle a
> collection of "n" User Agents (behaving each one as UAC and
> UAS, according to SIP rules), which are linked by some
> application logic that is fully independent of the SIP protocol rules.

This seems like a reasonable definition.

Poll: Would it be worth including this in 3261bis? Do we need to say
N>1?

--
Dean







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Dean,

you wrote:
> Juan-Carlos said:
> > Back-to-Back User Agent: A SIP element being able to handle a 
> > collection of "n" User Agents (behaving each one as UAC and 
> > UAS, according to SIP rules), which are linked by some 
> > application logic that is fully independent of the SIP protocol rules.
> 
> This seems like a reasonable definition. 
> 

I dislike the use of "is" in the last line.  Many actual B2BUAs are
not "fully independent of the SIP protocol rules", and the above
definition seems to say they must be.  Perhaps "may be"?

In particular, I don't think a firewall ALG will fit in the above
definition of a B2BUA.  Neither will the 3GPP P-CSCF.  If the above
definition captures the common understanding of what a B2BUA is 
(and it certainly captures the problems of the general case), there
needs to be another type of SIP entity defined for what exists in 
practice.

The term "transparent B2BUA" has been used in the past.  I don't think
those can be considered a subcase of the general definition above.

Bill Marshall
wtm@research.att.com
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Subject: Re: [Sip] B2BUA
To: William Marshall <wtm@research.att.com>
Cc: dean.willis@softarmor.com, sip@ietf.org
Date: Mon, 2 Dec 2002 15:19:40 +0100
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William,

"Many actual B2BUAs are not "fully independent of the SIP protocol rules",
and the above definition seems to say they must be..."
The proposed definition states that User Agents MUST behave as UAC ans UAS
according to SIP rules (this MUST could be enforced in the definition).
This is SIP standard, nothing else.
All what is added is that the way of linking User Agents in a B2BUA is not
related to SIP rules, and is not subject to SIP standardisation. As Dean
pointed out in his mail, they are User Agents, they can do anything they
want to do, and we don't want to standardise their behaviour.
You can define a B2BUA whose behaviour is based on SIP rules, but this is a
special case for a specific B2BUA implementing a well defined application.
However, this cannot be generalised to any B2BUA.

Concerning your statement "there needs to be another type of SIP entity
defined for what exists in practice", my feeling is that today there are
two SIP entity types: Proxy and B2BUA, and that seems to be enough (Am I
missed something ?)


Best regards
Juan Carlos





William Marshall <wtm@research.att.com>@ietf.org on 02/12/2002 15:06:13

Sent by:  sip-admin@ietf.org


To:   dean.willis@softarmor.com
cc:   sip@ietf.org
Subject:  Re: [Sip] B2BUA


Dean,

you wrote:
> Juan-Carlos said:
> > Back-to-Back User Agent: A SIP element being able to handle a
> > collection of "n" User Agents (behaving each one as UAC and
> > UAS, according to SIP rules), which are linked by some
> > application logic that is fully independent of the SIP protocol rules.
>
> This seems like a reasonable definition.
>



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Subject: [Sip] Transcoding Invocation: Callee's capacities?
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Hi all,

I have a small question about transcoding services invocation, described 
in draft-camarillo-sip-deaf-01 :
In Figures 2, 4 and 5, The 3pcc sends transcoded descriptions without 
having the original ones (Ex.  SDP B in figure 2).
Did this implies that T uses the OPTIONS message to have the missing 
description?

Thank's in advance

---
Oussama Layaida, PhD student
LSR-IMAG Laboratory, SARDES Project (INPG, INRIA, UJF)
Web: <http://sardes.inrialpes.fr/~olayaida/>
E-mail: Oussama.Layaida@inria.fr <mailto:Oussama.Layaida@inria.fr>
Phone: +33 4 76 61 52 79, Fax: +33 4 76 61 52 52
Address: INRIA Rhone-Alpes, SARDES, Oussama Layaida
655 av. de l'Europe, Montbonnot 38334 St Ismier Cedex France



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Can we not define B2BUAs as a SIP entity that exhibits the behaviour of a
UAC and UAS combination. i.e. it can not be distinguished externally from a
properly conforming UA, thus not constraining it's internal functionality or
implementation.

-----Original Message-----
From: Juan-Carlos.Rojas@alcatel.fr [mailto:Juan-Carlos.Rojas@alcatel.fr] 
Sent: 02 December 2002 14:20
To: William Marshall
Cc: dean.willis@softarmor.com; sip@ietf.org
Subject: Re: [Sip] B2BUA



William,

"Many actual B2BUAs are not "fully independent of the SIP protocol rules",
and the above definition seems to say they must be..." The proposed
definition states that User Agents MUST behave as UAC ans UAS according to
SIP rules (this MUST could be enforced in the definition). This is SIP
standard, nothing else. All what is added is that the way of linking User
Agents in a B2BUA is not related to SIP rules, and is not subject to SIP
standardisation. As Dean pointed out in his mail, they are User Agents, they
can do anything they want to do, and we don't want to standardise their
behaviour. You can define a B2BUA whose behaviour is based on SIP rules, but
this is a special case for a specific B2BUA implementing a well defined
application. However, this cannot be generalised to any B2BUA.

Concerning your statement "there needs to be another type of SIP entity
defined for what exists in practice", my feeling is that today there are two
SIP entity types: Proxy and B2BUA, and that seems to be enough (Am I missed
something ?)


Best regards
Juan Carlos





William Marshall <wtm@research.att.com>@ietf.org on 02/12/2002 15:06:13

Sent by:  sip-admin@ietf.org


To:   dean.willis@softarmor.com
cc:   sip@ietf.org
Subject:  Re: [Sip] B2BUA


Dean,

you wrote:
> Juan-Carlos said:
> > Back-to-Back User Agent: A SIP element being able to handle a 
> > collection of "n" User Agents (behaving each one as UAC and UAS, 
> > according to SIP rules), which are linked by some application logic 
> > that is fully independent of the SIP protocol rules.
>
> This seems like a reasonable definition.
>



_______________________________________________
Sip mailing list  https://www1.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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>
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version 5.5.2653.12"=
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<TITLE>RE: [Sip] B2BUA</TITLE>
</HEAD>
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<P><FONT SIZE=3D2>Can we not define B2BUAs as a SIP entity that exhibits th=
e behaviour of a UAC and UAS combination. i.e. it can not be distinguished =
externally from a properly conforming UA, thus not constraining it's intern=
al functionality or implementation.</FONT></P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Juan-Carlos.Rojas@alcatel.fr [<A HREF=3D"mailto:Ju=
an-Carlos.Rojas@alcatel.fr">mailto:Juan-Carlos.Rojas@alcatel.fr</A>] </FONT=
>
<BR><FONT SIZE=3D2>Sent: 02 December 2002 14:20</FONT>
<BR><FONT SIZE=3D2>To: William Marshall</FONT>
<BR><FONT SIZE=3D2>Cc: dean.willis@softarmor.com; sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: Re: [Sip] B2BUA</FONT>
</P>
<BR>
<BR>

<P><FONT SIZE=3D2>William,</FONT>
</P>

<P><FONT SIZE=3D2>&quot;Many actual B2BUAs are not &quot;fully independent =
of the SIP protocol rules&quot;, and the above definition seems to say they=
 must be...&quot; The proposed definition states that User Agents MUST beha=
ve as UAC ans UAS according to SIP rules (this MUST could be enforced in th=
e definition). This is SIP standard, nothing else. All what is added is tha=
t the way of linking User Agents in a B2BUA is not related to SIP rules, an=
d is not subject to SIP standardisation. As Dean pointed out in his mail, t=
hey are User Agents, they can do anything they want to do, and we don't wan=
t to standardise their behaviour. You can define a B2BUA whose behaviour is=
 based on SIP rules, but this is a special case for a specific B2BUA implem=
enting a well defined application. However, this cannot be generalised to a=
ny B2BUA.</FONT></P>

<P><FONT SIZE=3D2>Concerning your statement &quot;there needs to be another=
 type of SIP entity defined for what exists in practice&quot;, my feeling i=
s that today there are two SIP entity types: Proxy and B2BUA, and that seem=
s to be enough (Am I missed something ?)</FONT></P>
<BR>

<P><FONT SIZE=3D2>Best regards</FONT>
<BR><FONT SIZE=3D2>Juan Carlos</FONT>
</P>
<BR>
<BR>
<BR>
<BR>

<P><FONT SIZE=3D2>William Marshall &lt;wtm@research.att.com&gt;@ietf.org on=
 02/12/2002 15:06:13</FONT>
</P>

<P><FONT SIZE=3D2>Sent by:&nbsp; sip-admin@ietf.org</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>To:&nbsp;&nbsp; dean.willis@softarmor.com</FONT>
<BR><FONT SIZE=3D2>cc:&nbsp;&nbsp; sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject:&nbsp; Re: [Sip] B2BUA</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Dean,</FONT>
</P>

<P><FONT SIZE=3D2>you wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; Juan-Carlos said:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Back-to-Back User Agent: A SIP element being a=
ble to handle a </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; collection of &quot;n&quot; User Agents (behav=
ing each one as UAC and UAS, </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; according to SIP rules), which are linked by s=
ome application logic </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; that is fully independent of the SIP protocol =
rules.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; This seems like a reasonable definition.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
</P>
<BR>
<BR>

<P><FONT SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>Sip mailing list&nbsp; <A HREF=3D"https://www1.ietf.org/=
mailman/listinfo/sip" TARGET=3D"_blank">https://www1.ietf.org/mailman/listi=
nfo/sip</A></FONT>
<BR><FONT SIZE=3D2>This list is for NEW development of the core SIP Protoco=
l</FONT>
<BR><FONT SIZE=3D2>Use sip-implementors@cs.columbia.edu for questions on cu=
rrent sip Use sipping@ietf.org for new developments on the application of s=
ip</FONT></P>

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From mailnull@www1.ietf.org  Mon Dec  2 11:57:03 2002
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From: "Mary Barnes" <mbarnes@nortelnetworks.com>
To: "'Phil Reynolds'" <preynolds@ridgewaysystems.com>, sip@ietf.org
Subject: RE: [Sip] B2BUA
Date: Mon, 2 Dec 2002 10:53:50 -0600 
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Hi all,

I guess I have to ask, what's the problem with the definition in RFC 3261? 

Back-to-Back user agent: A back-to-back user agent (B2BUA) is a
             logical entity that receives a request, and processes it as
             a UAS. In order to determine how the request should be
             answered, it acts as a UAC and generates requests. Unlike a
             proxy server, it maintains dialog state, and must
             participate in all requests sent on the dialogs it has
             established. Since it is a concatenation of a UAC and UAS,
             no explicit definitions are needed for its behavior.

Regards,
Mary. 

-----Original Message-----
From: Phil Reynolds [mailto:preynolds@ridgewaysystems.com]
Sent: Monday, December 02, 2002 10:20 AM
To: sip@ietf.org
Subject: RE: [Sip] B2BUA


Can we not define B2BUAs as a SIP entity that exhibits the behaviour of a
UAC and UAS combination. i.e. it can not be distinguished externally from a
properly conforming UA, thus not constraining it's internal functionality or
implementation.
-----Original Message----- 
From: Juan-Carlos.Rojas@alcatel.fr [mailto:Juan-Carlos.Rojas@alcatel.fr] 
Sent: 02 December 2002 14:20 
To: William Marshall 
Cc: dean.willis@softarmor.com; sip@ietf.org 
Subject: Re: [Sip] B2BUA 



William, 
"Many actual B2BUAs are not "fully independent of the SIP protocol rules",
and the above definition seems to say they must be..." The proposed
definition states that User Agents MUST behave as UAC ans UAS according to
SIP rules (this MUST could be enforced in the definition). This is SIP
standard, nothing else. All what is added is that the way of linking User
Agents in a B2BUA is not related to SIP rules, and is not subject to SIP
standardisation. As Dean pointed out in his mail, they are User Agents, they
can do anything they want to do, and we don't want to standardise their
behaviour. You can define a B2BUA whose behaviour is based on SIP rules, but
this is a special case for a specific B2BUA implementing a well defined
application. However, this cannot be generalised to any B2BUA.
Concerning your statement "there needs to be another type of SIP entity
defined for what exists in practice", my feeling is that today there are two
SIP entity types: Proxy and B2BUA, and that seems to be enough (Am I missed
something ?)


Best regards 
Juan Carlos 





William Marshall <wtm@research.att.com>@ietf.org on 02/12/2002 15:06:13 
Sent by:  sip-admin@ietf.org 


To:   dean.willis@softarmor.com 
cc:   sip@ietf.org 
Subject:  Re: [Sip] B2BUA 


Dean, 
you wrote: 
> Juan-Carlos said: 
> > Back-to-Back User Agent: A SIP element being able to handle a 
> > collection of "n" User Agents (behaving each one as UAC and UAS, 
> > according to SIP rules), which are linked by some application logic 
> > that is fully independent of the SIP protocol rules. 
> 
> This seems like a reasonable definition. 
> 



_______________________________________________ 
Sip mailing list  https://www1.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://www1.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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Hello,
Well, if from times to times there are questions and debates on a given
subject, it is surely because the definition needs to be improved, or at
least clarified.

"Since it is a concatenation of a UAC and UAS"
1.- A B2BUA is not a "concatenation of a UAC and UAS", but a collection of
several UAs (two, three or more), which behave individually as UAS or UAC
according to SIP rules (and are seen as such from the outside of the
B2BUA). For example, a conference server where all the participants send
INVITE messages to the server could be made with UAS only (do note that
this is strictly a matter of application definition, and cannot be subject
to standardisation).
2.- UAs are not "concateneted" in the B2BUA, but they are linked by some
application logic which is by definition, a non-standard element. This is
the key element that should be precised in the RFC3261 to avoid confusion.
You cannot define the behaviour of a B2BUA, you can define only the
behaviour of each UA (and this is actually done by the SIP specification).

Hoping this answers your questions
Best regards
Juan Carlos




"Mary Barnes" <mbarnes@nortelnetworks.com>@ietf.org on 02/12/2002 17:53:50

Sent by:  sip-admin@ietf.org


To:   "'Phil Reynolds'" <preynolds@ridgewaysystems.com>, sip@ietf.org
cc:
Subject:  RE: [Sip] B2BUA


Hi all,

I guess I have to ask, what's the problem with the definition in RFC 3261?

Back-to-Back user agent: A back-to-back user agent (B2BUA) is a
             logical entity that receives a request, and processes it as
             a UAS. In order to determine how the request should be
             answered, it acts as a UAC and generates requests. Unlike a
             proxy server, it maintains dialog state, and must
             participate in all requests sent on the dialogs it has
             established. Since it is a concatenation of a UAC and UAS,
             no explicit definitions are needed for its behavior.

Regards,
Mary.




_______________________________________________
Sip mailing list  https://www1.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: "Mary Barnes" <mbarnes@nortelnetworks.com>
To: "'Juan-Carlos.Rojas@alcatel.fr'" <Juan-Carlos.Rojas@alcatel.fr>
Cc: sip@ietf.org
Subject: RE: [Sip] B2BUA
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------_=_NextPart_001_01C29A2A.CA093460
Content-Type: text/plain;
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Juan-Carlos,

So, I guess you're suggesting that the term "concatenation" is not accurate
or is misleading and I guess that depends upon how loosely one applies the
term concatenation here; my assumption had been that it meant that they were
"linked" (consistent with Webster's definition of concatenate) in some
manner which is unprescribed, so I think your point 2 below isn't a problem
with the current definition.  My understanding of the current definition is
that there isn't anything more for the SIP protocol to prescribe about the
operation of a B2BUA other than what is required from a protocol perspective
to function as a UAC and a UAS. From a protocol perspective, it is a logical
entity and thus whether there's one or multiple doesn't seem to apply.  I
think what you (and perhaps others) are suggesting is that there's a
definition of a B2BUA that more closely parallels the definition of a Proxy
Server since in many cases the physical rendition of the logical B2BUA
functionality would be replacing a Proxy Server in a network. And I think
this is where the problem begins, as many are opposed to this model (as it
increases the complexity, etc.) and this is why the logical definition has
remained. 

On your first point, however, I think we need to wait until the conferencing
work in SIPPING is more complete before we can suggest that a conference
server would be a B2BUA. 

Mary.

-----Original Message-----
From: Juan-Carlos.Rojas@alcatel.fr [mailto:Juan-Carlos.Rojas@alcatel.fr]
Sent: Monday, December 02, 2002 11:20 AM
To: Barnes, Mary [NGC:B602:EXCH]
Cc: sip@ietf.org
Subject: RE: [Sip] B2BUA



Hello,
Well, if from times to times there are questions and debates on a given
subject, it is surely because the definition needs to be improved, or at
least clarified.

"Since it is a concatenation of a UAC and UAS"
1.- A B2BUA is not a "concatenation of a UAC and UAS", but a collection of
several UAs (two, three or more), which behave individually as UAS or UAC
according to SIP rules (and are seen as such from the outside of the
B2BUA). For example, a conference server where all the participants send
INVITE messages to the server could be made with UAS only (do note that
this is strictly a matter of application definition, and cannot be subject
to standardisation).
2.- UAs are not "concateneted" in the B2BUA, but they are linked by some
application logic which is by definition, a non-standard element. This is
the key element that should be precised in the RFC3261 to avoid confusion.
You cannot define the behaviour of a B2BUA, you can define only the
behaviour of each UA (and this is actually done by the SIP specification).

Hoping this answers your questions
Best regards
Juan Carlos




"Mary Barnes" <mbarnes@nortelnetworks.com>@ietf.org on 02/12/2002 17:53:50

Sent by:  sip-admin@ietf.org


To:   "'Phil Reynolds'" <preynolds@ridgewaysystems.com>, sip@ietf.org
cc:
Subject:  RE: [Sip] B2BUA


Hi all,

I guess I have to ask, what's the problem with the definition in RFC 3261?

Back-to-Back user agent: A back-to-back user agent (B2BUA) is a
             logical entity that receives a request, and processes it as
             a UAS. In order to determine how the request should be
             answered, it acts as a UAC and generates requests. Unlike a
             proxy server, it maintains dialog state, and must
             participate in all requests sent on the dialogs it has
             established. Since it is a concatenation of a UAC and UAS,
             no explicit definitions are needed for its behavior.

Regards,
Mary.





------_=_NextPart_001_01C29A2A.CA093460
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

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charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2654.89">
<TITLE>RE: [Sip] B2BUA</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Juan-Carlos,</FONT>
</P>

<P><FONT SIZE=3D2>So, I guess you're suggesting that the term =
&quot;concatenation&quot; is not accurate or is misleading and I guess =
that depends upon how loosely one applies the term concatenation here; =
my assumption had been that it meant that they were &quot;linked&quot; =
(consistent with Webster's definition of concatenate) in some manner =
which is unprescribed, so I think your point 2 below isn't a problem =
with the current definition.&nbsp; My understanding of the current =
definition is that there isn't anything more for the SIP protocol to =
prescribe about the operation of a B2BUA other than what is required =
from a protocol perspective to function as a UAC and a UAS. From a =
protocol perspective, it is a logical entity and thus whether there's =
one or multiple doesn't seem to apply.&nbsp; I think what you (and =
perhaps others) are suggesting is that there's a definition of a B2BUA =
that more closely parallels the definition of a Proxy Server since in =
many cases the physical rendition of the logical B2BUA functionality =
would be replacing a Proxy Server in a network. And I think this is =
where the problem begins, as many are opposed to this model (as it =
increases the complexity, etc.) and this is why the logical definition =
has remained. </FONT></P>

<P><FONT SIZE=3D2>On your first point, however, I think we need to wait =
until the conferencing work in SIPPING is more complete before we can =
suggest that a conference server would be a B2BUA. </FONT></P>

<P><FONT SIZE=3D2>Mary.</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Juan-Carlos.Rojas@alcatel.fr [<A =
HREF=3D"mailto:Juan-Carlos.Rojas@alcatel.fr">mailto:Juan-Carlos.Rojas@al=
catel.fr</A>]</FONT>
<BR><FONT SIZE=3D2>Sent: Monday, December 02, 2002 11:20 AM</FONT>
<BR><FONT SIZE=3D2>To: Barnes, Mary [NGC:B602:EXCH]</FONT>
<BR><FONT SIZE=3D2>Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: RE: [Sip] B2BUA</FONT>
</P>
<BR>
<BR>

<P><FONT SIZE=3D2>Hello,</FONT>
<BR><FONT SIZE=3D2>Well, if from times to times there are questions and =
debates on a given</FONT>
<BR><FONT SIZE=3D2>subject, it is surely because the definition needs =
to be improved, or at</FONT>
<BR><FONT SIZE=3D2>least clarified.</FONT>
</P>

<P><FONT SIZE=3D2>&quot;Since it is a concatenation of a UAC and =
UAS&quot;</FONT>
<BR><FONT SIZE=3D2>1.- A B2BUA is not a &quot;concatenation of a UAC =
and UAS&quot;, but a collection of</FONT>
<BR><FONT SIZE=3D2>several UAs (two, three or more), which behave =
individually as UAS or UAC</FONT>
<BR><FONT SIZE=3D2>according to SIP rules (and are seen as such from =
the outside of the</FONT>
<BR><FONT SIZE=3D2>B2BUA). For example, a conference server where all =
the participants send</FONT>
<BR><FONT SIZE=3D2>INVITE messages to the server could be made with UAS =
only (do note that</FONT>
<BR><FONT SIZE=3D2>this is strictly a matter of application definition, =
and cannot be subject</FONT>
<BR><FONT SIZE=3D2>to standardisation).</FONT>
<BR><FONT SIZE=3D2>2.- UAs are not &quot;concateneted&quot; in the =
B2BUA, but they are linked by some</FONT>
<BR><FONT SIZE=3D2>application logic which is by definition, a =
non-standard element. This is</FONT>
<BR><FONT SIZE=3D2>the key element that should be precised in the =
RFC3261 to avoid confusion.</FONT>
<BR><FONT SIZE=3D2>You cannot define the behaviour of a B2BUA, you can =
define only the</FONT>
<BR><FONT SIZE=3D2>behaviour of each UA (and this is actually done by =
the SIP specification).</FONT>
</P>

<P><FONT SIZE=3D2>Hoping this answers your questions</FONT>
<BR><FONT SIZE=3D2>Best regards</FONT>
<BR><FONT SIZE=3D2>Juan Carlos</FONT>
</P>
<BR>
<BR>
<BR>

<P><FONT SIZE=3D2>&quot;Mary Barnes&quot; =
&lt;mbarnes@nortelnetworks.com&gt;@ietf.org on 02/12/2002 =
17:53:50</FONT>
</P>

<P><FONT SIZE=3D2>Sent by:&nbsp; sip-admin@ietf.org</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>To:&nbsp;&nbsp; &quot;'Phil Reynolds'&quot; =
&lt;preynolds@ridgewaysystems.com&gt;, sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>cc:</FONT>
<BR><FONT SIZE=3D2>Subject:&nbsp; RE: [Sip] B2BUA</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Hi all,</FONT>
</P>

<P><FONT SIZE=3D2>I guess I have to ask, what's the problem with the =
definition in RFC 3261?</FONT>
</P>

<P><FONT SIZE=3D2>Back-to-Back user agent: A back-to-back user agent =
(B2BUA) is a</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; logical entity that receives a request, and processes it =
as</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; a UAS. In order to determine how the request should be</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; answered, it acts as a UAC and generates requests. Unlike =
a</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; proxy server, it maintains dialog state, and must</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; participate in all requests sent on the dialogs it has</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; established. Since it is a concatenation of a UAC and =
UAS,</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; no explicit definitions are needed for its behavior.</FONT>
</P>

<P><FONT SIZE=3D2>Regards,</FONT>
<BR><FONT SIZE=3D2>Mary.</FONT>
</P>
<BR>
<BR>
<BR>

</BODY>
</HTML>
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Subject: Re: [Sip] different messages with same call-id before 100 is sent out
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In a message dated 12/2/2002 2:38:32 AM Eastern Standard Time, 
Gonzalo.Camarillo@lmf.ericsson.se writes:


> Hi,
> 
> It has been explained several times why you cannot force the subsequent
> INVITEs to follow the rute that the first one followed. You can find the
> discussions in the archives or in the slides I presented in Australia.
> 
> Thanks,
> 
> 


[MAP] I understand that you can not "force" anything to take the same route, 
or even to arrive at the same egress point.

But my previous comments were generally the following:

1. Define the procedures for overlap based on a presumption that all INVITEs 
for one call will follow the same route.
2. Strongly "suggest" that the ingress point should not send the first INVITE 
until it has enough digits for proper routing (generally to the egress point).
3. The ingress clearly marks the first INVITE as incomplete and subsequent 
INVITEs (call id, seq number, or whatever) to indicate that they belong 
together
4. The egress doesn't handle anything that isn't correct, i.e., throws away 
an initial incomplete INVITE if it doesn't understand how to do overlap, 
stores it if it doesn't want to do overlap, or throws away a subsequent 
INVITE if it didn't get the previous.

I realize that 4. could be a problem if a subsequent INVITE looks just like a 
normal one and the egress GW knows nothing about handling overlap, that is, 
it doesn't understand that it is supposed to do what 4. says. But this is the 
whole point of my comments. There needs to be distinguishing information in 
the INVITE and if the egress GW doesn't understand the overlap procedures, 
then it shouldn't be doing it.

If SIP doesn't have a good way to tell the other end to ignore a message if 
it doesn't understand some specific part, then there is a bigger problem.

Mike Pierce
Artel


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<HTML><FONT FACE=arial,helvetica><FONT  SIZE=2>In a message dated 12/2/2002 2:38:32 AM Eastern Standard Time, Gonzalo.Camarillo@lmf.ericsson.se writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">Hi,
<BR>
<BR>It has been explained several times why you cannot force the subsequent
<BR>INVITEs to follow the rute that the first one followed. You can find the
<BR>discussions in the archives or in the slides I presented in Australia.
<BR>
<BR>Thanks,
<BR>
<BR></FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR>
<BR>[MAP] I understand that you can not "force" anything to take the same route, or even to arrive at the same egress point.
<BR>
<BR>But my previous comments were generally the following:
<BR>
<BR>1. Define the procedures for overlap based on a presumption that all INVITEs for one call will follow the same route.
<BR>2. Strongly "suggest" that the ingress point should not send the first INVITE until it has enough digits for proper routing (generally to the egress point).
<BR>3. The ingress clearly marks the first INVITE as incomplete and subsequent INVITEs (call id, seq number, or whatever) to indicate that they belong together
<BR>4. The egress doesn't handle anything that isn't correct, i.e., throws away an initial incomplete INVITE if it doesn't understand how to do overlap, stores it if it doesn't want to do overlap, or throws away a subsequent INVITE if it didn't get the previous.
<BR>
<BR>I realize that 4. could be a problem if a subsequent INVITE looks just like a normal one and the egress GW knows nothing about handling overlap, that is, it doesn't understand that it is supposed to do what 4. says. But this is the whole point of my comments. There needs to be distinguishing information in the INVITE and if the egress GW doesn't understand the overlap procedures, then it shouldn't be doing it.
<BR>
<BR>If SIP doesn't have a good way to tell the other end to ignore a message if it doesn't understand some specific part, then there is a bigger problem.
<BR>
<BR>Mike Pierce
<BR>Artel
<BR></FONT></HTML>

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Mary,
Debates about what a B2BUA can or cannot do come back regularly. That's why
I suggested that to avoid confusion, the definition needs maybe to be
improved.
See other comments inline in your text.
Best regards
Juan Carlos





"Mary Barnes" <mbarnes@nortelnetworks.com> on 02/12/2002 18:46:48

To:   Juan-Carlos ROJAS/FR/ALCATEL@ALCATEL
cc:   sip@ietf.org
Subject:  RE: [Sip] B2BUA




Juan-Carlos,

So, I guess you're suggesting that the term "concatenation" is not accurate
or is misleading and I guess that depends upon how loosely one applies the
term concatenation here; my assumption had been that it meant that they
were "linked" (consistent with Webster's definition of concatenate) in some
manner which is unprescribed, so I think your point 2 below isn't a problem
with the current definition.  My understanding of the current definition is
that there isn't anything more for the SIP protocol to prescribe about the
operation of a B2BUA other than what is required from a protocol
perspective to function as a UAC and a UAS. From a protocol perspective, it
is a logical entity and thus whether there's one or multiple doesn't seem
to apply.

JC: you say "...they were "linked" ... in some manner which is
unprescribed". Fully agree with you, and in my feeling, the word
"unprescribed" or any other synonym should be used explicitly in the
definition of a B2BUA (this is not currently the case), because very often
this is the reason of confusion.

I think what you (and perhaps others) are suggesting is that there's a
definition of a B2BUA that more closely parallels the definition of a Proxy
Server since in many cases the physical rendition of the logical B2BUA
functionality would be replacing a Proxy Server in a network. And I think
this is where the problem begins, as many are opposed to this model (as it
increases the complexity, etc.) and this is why the logical definition has
remained.

JC: it is definitely not my case to search for any parallel with the
definition of a Proxy Server (whose behaviour is well defined in RFC3261).
Do mind that this discusion thread started because somebody asked what a
B2BUA can or cannot do. The central problem is once again this
"unprescribed" relation of UAs in a B2BUA.

On your first point, however, I think we need to wait until the
conferencing work in SIPPING is more complete before we can suggest that a
conference server would be a B2BUA.

JC: you can define a conference server as a B2BUA or not, no problem with
that. But this is only one type of application; I'm not suggesting it, I
used only as an example.

Mary.




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To: Juan-Carlos.Rojas@alcatel.fr
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Subject: RE: [Sip] B2BUA
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Juan Carlow wrote, in part:
> 2.- UAs are not "concateneted" in the B2BUA, but they are linked by some
> application logic which is by definition, a non-standard element. This is
                    ~~~~~~~~~~~~~~~~~~~~~~
> the key element that should be precised in the RFC3261 to avoid confusion.

This is the exact point that I disagree with.  

The entity that we are describing receives SIP requests on one side,
processes them internally, and generates SIP requests on the other side.
Responses are handled in the reverse path.  I don't think there is any
disagreement so far.

There is wide variation possible in the algorithms used internally
to process the requests and responses.  But they are all B2BUAs.
One of the possible algorithms is defined in section 16 of rfc3261.
Many other possible algorithms exist, and are not covered in rfc3261.
The definition needs to cover all of these possibilities, not just
the non-standard ones.

At some point we need to define exactly what the requirements are
for the internal processing algorithm, such that new services will
work through it.  Section 16 is one example.  But I don't think we 
are ready yet to generalize this.

Bill Marshall
wtm@research.att.com

-----original message-----
From: Juan-Carlos.Rojas@alcatel.fr [mailto:Juan-Carlos.Rojas@alcatel.fr]
Sent: Monday, December 02, 2002 11:20 AM
To: Barnes, Mary [NGC:B602:EXCH]
Cc: sip@ietf.org
Subject: RE: [Sip] B2BUA



Hello,
Well, if from times to times there are questions and debates on a given
subject, it is surely because the definition needs to be improved, or at
least clarified.

"Since it is a concatenation of a UAC and UAS"
1.- A B2BUA is not a "concatenation of a UAC and UAS", but a collection of
several UAs (two, three or more), which behave individually as UAS or UAC
according to SIP rules (and are seen as such from the outside of the
B2BUA). For example, a conference server where all the participants send
INVITE messages to the server could be made with UAS only (do note that
this is strictly a matter of application definition, and cannot be subject
to standardisation).
2.- UAs are not "concateneted" in the B2BUA, but they are linked by some
application logic which is by definition, a non-standard element. This is
the key element that should be precised in the RFC3261 to avoid confusion.
You cannot define the behaviour of a B2BUA, you can define only the
behaviour of each UA (and this is actually done by the SIP specification).

Hoping this answers your questions
Best regards
Juan Carlos

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'William Marshall'" <wtm@research.att.com>
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Subject: RE: [Sip] B2BUA
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> > Juan-Carlos said:
> > > Back-to-Back User Agent: A SIP element being able to handle a
> > > collection of "n" User Agents (behaving each one as UAC and 
> > > UAS, according to SIP rules), which are linked by some 
> > > application logic that is fully independent of the SIP 
> protocol rules.
> 
> I dislike the use of "is" in the last line.  Many actual B2BUAs are
> not "fully independent of the SIP protocol rules", and the above
> definition seems to say they must be.  Perhaps "may be"?
> 
> In particular, I don't think a firewall ALG will fit in the above
> definition of a B2BUA.  Neither will the 3GPP P-CSCF.  If the above
> definition captures the common understanding of what a B2BUA is 
> (and it certainly captures the problems of the general case), there
> needs to be another type of SIP entity defined for what exists in 
> practice.

Well, "fully independent" doen't seem the same to me as "must ignore",
but perhaps "may be" would be better. Honestly, the existing text
satisfies me quite nicely, but I have noticed that it seems to confuse
people which raises the question of "how can we keep this confusion from
occurring in the mind of the reader."

> The term "transparent B2BUA" has been used in the past.  I don't think
> those can be considered a subcase of the general definition above.
> 

Oh no -- I'll have to dig up my "Transparency Top 10" list again. Maybe
that should become a yearly post, sent out the week after my "B2BUA"
post . . .;-)

Really, I think that almost any of the "Top 10" transparency modes can
be violated by a B2BUA, whereas, if I recall my post, NONE are violated
by a conformant proxy. Something like a Firewall Control Proxy preserves
many of the transparency modes, while violating something fundamental
like "media transparency". It's still not really a "proxy", so it's
"some kind of B2BUA that is almost but not quite exactly a proxy".

Just because the FCP's application logic implements something that is
SIMILAR TO the SIP protocol rules, it doesn't mean that it isn't
INDEPENDENT OF thos same rules. But hey, the 3261 def works for me, so
what do I know?

--
Dean

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Mary Barnes'" <mbarnes@nortelnetworks.com>,
        <Juan-Carlos.Rojas@alcatel.fr>
Cc: <sip@ietf.org>
Subject: RE: [Sip] B2BUA
Date: Tue, 3 Dec 2002 00:00:19 -0600
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Mary said:
 -----
 So, I guess you're suggesting that the term "concatenation" is not
accurate or is misleading and I guess that depends upon how loosely one
applies the term concatenation here; my assumption had been that it
meant that they were "linked" (consistent with Webster's definition of
concatenate) in some manner which is unprescribed, so I think your point
2 below isn't a problem with the current definition.  My understanding
of the current definition is that there isn't anything more for the SIP
protocol to prescribe about the operation of a B2BUA other than what is
required from a protocol perspective to function as a UAC and a UAS.
From a protocol perspective, it is a logical entity and thus whether
there's one or multiple doesn't seem to apply.  I think what you (and
perhaps others) are suggesting is that there's a definition of a B2BUA
that more closely parallels the definition of a Proxy Server since in
many cases the physical rendition of the logical B2BUA functionality
would be replacing a Proxy Server in a network. And I think this is
where the problem begins, as many are opposed to this model (as it
increases the complexity, etc.) and this is why the logical definition
has remained.  
------ 
 
Well, I think the argument is being made that this definition doesn't
(for some reason that I seem incapable of comprehending) work for some
people, and that therefore we should clarify it. At the fundamental
level, I think some people do not find the implications of this
definition obvious. There are several ways we might address this. We
might add examples. We might add "In other words, the only relationship
between the UAC and the UAS is the application logic, and that can do
anything that one can imagine an application doing", or we might just
use better wording.
 
I don't think the argument is being made that the definition in 3261 is
wrong, just that we can make it better for those who don't get it the
way it is written. Of course, if we discuss it here long enough, maybe
most of them WILL get it -- but that doesn't help the ones who aren't
"here" yet.
 
I think that Bill is suggesting that there is also something BETWEEN the
B2BUA and the conformant proxy, and that the world would be a better
place (ok, maybe not THAT much better) if our taxonomy also included
this "between" thing.
 
--
Dean
 

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 6.00.2800.1126" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT size=3D2><SPAN class=3D320285205-03122002>Mary =
said:</SPAN></FONT></DIV>
<DIV><SPAN class=3D320285205-03122002></SPAN><FONT size=3D2><SPAN=20
class=3D320285205-03122002><FONT face=3DArial=20
color=3D#0000ff>&nbsp;-----</FONT></SPAN></FONT></DIV>
<DIV><FONT size=3D2><SPAN class=3D320285205-03122002>&nbsp;</SPAN>So, I =
guess you're=20
suggesting that the term "concatenation" is not accurate or is =
misleading and I=20
guess that depends upon how loosely one applies the term concatenation =
here; my=20
assumption had been that it meant that they were "linked" (consistent =
with=20
Webster's definition of concatenate) in some manner which is =
unprescribed, so I=20
think your point 2 below isn't a problem with the current =
definition.&nbsp; My=20
understanding of the current definition is that there isn't anything =
more for=20
the SIP protocol to prescribe about the operation of a B2BUA other than =
what is=20
required from a protocol perspective to function as a UAC and a UAS. =
From a=20
protocol perspective, it is a logical entity and thus whether there's =
one or=20
multiple doesn't seem to apply.&nbsp; I think what you (and perhaps =
others) are=20
suggesting is that there's a definition of a B2BUA that more closely =
parallels=20
the definition of a Proxy Server since in many cases the physical =
rendition of=20
the logical B2BUA functionality would be replacing a Proxy Server in a =
network.=20
And I think this is where the problem begins, as many are opposed to =
this model=20
(as it increases the complexity, etc.) and this is why the logical =
definition=20
has remained.&nbsp;<SPAN class=3D320285205-03122002><FONT face=3DArial=20
color=3D#0000ff>&nbsp;</FONT></SPAN></FONT></DIV>
<DIV><FONT size=3D2><SPAN class=3D320285205-03122002><FONT face=3DArial=20
color=3D#0000ff>------</FONT>&nbsp;</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D320285205-03122002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D320285205-03122002>Well,=20
I think the argument is being made that this definition doesn't (for =
some reason=20
that I seem incapable of comprehending) work for some people, and that =
therefore=20
we should clarify it. At the fundamental level, I think some people do =
not find=20
the implications of this definition obvious. There are several ways we =
might=20
address this. We might add examples. We might add "In other words, the =
only=20
relationship between the UAC and the UAS is the application logic, and =
that can=20
do anything that one can imagine an application doing", or we might just =
use=20
better wording.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D320285205-03122002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D320285205-03122002>I=20
don't think the argument is being made that the definition in 3261 is =
wrong,=20
just that we can make it better for those who don't get it the way it is =

written. Of course, if we discuss it here long enough, maybe most of =
them WILL=20
get it -- but that doesn't help the ones who aren't "here"=20
yet.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D320285205-03122002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D320285205-03122002>I=20
think that Bill is suggesting that there is also something BETWEEN the =
B2BUA and=20
the conformant proxy, and that the world would be a better place (ok, =
maybe not=20
THAT much better) if our taxonomy also included this "between"=20
thing.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D320285205-03122002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D320285205-03122002>--<BR>Dean</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D320285205-03122002></SPAN></FONT>&nbsp;</DIV></BODY></HTML>

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Hi,

Any news on this issue?

Was this discussed at the Atlanta IETF?

Regards,

Christer Holmberg
Ericsson Finland


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CC: prasanna@huawei.com, rshiroor@aplion.stpn.soft.net, sip@ietf.org
Subject: Re: [Sip] different messages with same call-id before 100 is sent out
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Hi,

(I cannot "quote" your mail to answer more specifically because you sent
in in html format.)


Anyway, what you are proposing is a variant of what it has been proposed
a (large) number of times in the mailing list. It was also discussed in
the Australian IETF.

Thanks,

Gonzalo
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Date: Tue, 03 Dec 2002 14:31:12 +0530
From: Prasanna Venkatesh <prasanna@huawei.com>
Subject: RE: [Sip] different messages with same call-id before 100 is sent out
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One more query Gonzalo.
Can the draft define some behaviour on the UAS (egress GW) for out-of-order
messages?  There may be typical issues for SIP-T like, the first message
bore the IAM and a subsequent one had the SAM.  If they come out of order,
the egress GW's behaviour is confusing.
Regards,
Prasanna
Huawei Technologies

-----Original Message-----
From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@lmf.ericsson.se]
Sent: Tuesday, December 03, 2002 2:31 PM
To: Mpierce1@aol.com
Cc: prasanna@huawei.com; rshiroor@aplion.stpn.soft.net; sip@ietf.org
Subject: Re: [Sip] different messages with same call-id before 100 is
sent out


Hi,

(I cannot "quote" your mail to answer more specifically because you sent
in in html format.)


Anyway, what you are proposing is a variant of what it has been proposed
a (large) number of times in the mailing list. It was also discussed in
the Australian IETF.

Thanks,

Gonzalo

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Hi,

> Can the draft define some behaviour on the UAS (egress GW) for out-of-order
> messages?  There may be typical issues for SIP-T like, the first message
> bore the IAM and a subsequent one had the SAM.

All the INVITEs contain the IAM and all the subsequent SAMs.

Anyway, I will see if I can clarify the behavior in the draft so that it
is clearer.

Thanks,

Gonzalo
_______________________________________________
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From: Prasanna Venkatesh <prasanna@huawei.com>
Subject: RE: [Sip] different messages with same call-id before 100 is sent out
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Is this behaviour that all INVITEs contain all the IAMs and SAMs mentioned
in the draft.  I am very sorry if I have overlooked it.
regards,
Prasanna

-----Original Message-----
From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@lmf.ericsson.se]
Sent: Tuesday, December 03, 2002 3:00 PM
To: Prasanna Venkatesh
Cc: Mpierce1@aol.com; rshiroor@aplion.stpn.soft.net; sip@ietf.org
Subject: Re: [Sip] different messages with same call-id before 100 is
sent out


Hi,

> Can the draft define some behaviour on the UAS (egress GW) for
out-of-order
> messages?  There may be typical issues for SIP-T like, the first message
> bore the IAM and a subsequent one had the SAM.

All the INVITEs contain the IAM and all the subsequent SAMs.

Anyway, I will see if I can clarify the behavior in the draft so that it
is clearer.

Thanks,

Gonzalo

_______________________________________________
Sip mailing list  https://www1.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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Hi Peterson,

Thanks for your response. Please find some comments inline.

Regards,
Siddharth.

-------------------------------------------------------
Siddharth Toshniwal
Hughes Software Systems
Prestige Opal                    http://www.hssworld.com
146, Infantry Road          Ph (O): +91-80-2286390 (7094)
Bangalore-560001, India           Mobile: +91-9845154068
-------------------------------------------------------


> We did discuss this issue a bit in Atlanta.
> Unfortunately, I think all we agreed on is that we
> all need to think about it some more.

Aside from this question, can someone suggest what is
the best way to track open issues related to the spec?

I tried the bugzilla this time around, but as you
mentioned some issues (like this one) may take a while
to reach there if there are other significant issues
surrounding the original question that you wish to
discuss before proposing a solution.

If the spec is going to undergo rewording in some aspect,
I would request the authors to post to the discussion
list while they discussed the changes... else people
may implement incorrect behaviour and would need to go
back and change the same when the draft gets fixed.

It would help if you could give me a pointer for this
IETF's minutes regarding SIP issues that were discussed.


>> * Is it allowed for a SIP proxy that got a request on
>>   TLS to forward the request downstream on a non-TLS
>>   protocol?
>
> Depends on what you mean by 'got a request'. This also
> doesn't say whether or not SIPS appears in the Request-URI
> or not; it is possible for a proxy server to receive a
> request over TLS that does not have the SIPS URI scheme
> in the Request-URI.

I'm sorry for the ambiguity in my question. I had
referred to the foll scenario:

Lets say that the request-uri does not contain
"sips:" but the request was received on TLS (perhaps
because of a Route header configured with a previous
hop). Can the proxy send it downstream on a non-TLS
protocol if there are no more Route headers present
in the message?

From your mail, I gather this is allowed.

<snip>
>>  while Rosenberg had mentioned that the presence
>>   of "sips:" should imply that the end-to-end path is
>>   secure.
>
> If SIPS appears in the Request-URI of the original
> request sent by the UAC, it implies this end-to-end
> path security.
</snip>

<snip>
>> I guess the way I understand the spec stands
>> at present is that if the Req-URI is re-written to a
>> "sips:" URL at any stage of the routing, then any hop
>> downstream _from_this_hop_onwards_ has to forward the
>> request on TLS.
>
> I agree, that is how we should understand it - with the
> caveat that SIPS only applies to the entire path until
> it reaches the domain indicated by the hostname portion
> of the Request-URI. Once a proxy server in the destination
> gets hold of the request, it can retarget it arbitrarily
> (to non-SIPS resources, if it wishes).
</snip>

Aren't the 2 texts quoted by you at opposite ends?

a) the UAS cannot be sure of end-to-end security if it
   received a INVITE with the request-URI as "sips:" ...
   Some hop in the middle could have modified the req-URI
   for "sips:". It may even be the proxy with whom the UAS
   registered with the Contact as "sips:" ?

b) (If the caveat that you mentioned holds) the UAC
   cannot be sure that the UAS received the request
   on TLS. Shouldn't the destination's proxy be
   disallowed to route a request received on TLS to
   a non-TLS client?


>> * What is the response code to be used by a UAS to
>>   inform the UAC that it is not willing to accept
>>   insecure calls and that the UAC should retry the
>>   request on TLS?
>>
>
> Although 416 might look like an attractive candidate, the language in
> 8.1.3.5 makes it a little ugly. I would suggest 403.

can we add this text to the spec too if everyone agrees on it?


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Hi,

7th paragraph of section 3.3:

If the gateway is encapsulating ISUP messages as SIP bodies, it
  should place the IAM and all the SAMs received so far in this INVITE.

Regards,

Gonzalo

Prasanna Venkatesh wrote:
> 
> Is this behaviour that all INVITEs contain all the IAMs and SAMs mentioned
> in the draft.  I am very sorry if I have overlooked it.
> regards,
> Prasanna
> 
> -----Original Message-----
> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@lmf.ericsson.se]
> Sent: Tuesday, December 03, 2002 3:00 PM
> To: Prasanna Venkatesh
> Cc: Mpierce1@aol.com; rshiroor@aplion.stpn.soft.net; sip@ietf.org
> Subject: Re: [Sip] different messages with same call-id before 100 is
> sent out
> 
> Hi,
> 
> > Can the draft define some behaviour on the UAS (egress GW) for
> out-of-order
> > messages?  There may be typical issues for SIP-T like, the first message
> > bore the IAM and a subsequent one had the SAM.
> 
> All the INVITEs contain the IAM and all the subsequent SAMs.
> 
> Anyway, I will see if I can clarify the behavior in the draft so that it
> is clearer.
> 
> Thanks,
> 
> Gonzalo

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
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The problem as I see it is that the RFC defines 2 distinct intermediary
entities a "SIP Proxy" which has limited functionality, i.e. it is not
allowed to modify the body of the message, and a B2BUA which is a full blown
pair of Uas with all the transactional and dialogue stuff.

What we build is something in the middle... We rely on the "real" UAs to do
all the transactions, retries dialogs etc, and we modify the messages and
SDP bodies as they go through.

Personally I don't care what the spec says. We build a system that does a
useful function, if it doesn't fit into a category that is defined by the
spec -- so what? However the marketing types don't quite see it that way ;-)

If the spec is going to define the permissible intermediaries, it SHOULD
allow for things that are not implemented as UAs but externally appear to be
so.

-----Original Message-----
From: Mary Barnes [mailto:mbarnes@nortelnetworks.com] 
Sent: 02 December 2002 16:54
To: 'Phil Reynolds'; sip@ietf.org
Subject: RE: [Sip] B2BUA


Hi all,

I guess I have to ask, what's the problem with the definition in RFC 3261? 


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<P><FONT SIZE=3D2>The problem as I see it is that the RFC defines 2 distinc=
t intermediary entities a &quot;SIP Proxy&quot; which has limited functiona=
lity, i.e. it is not allowed to modify the body of the message, and a B2BUA=
 which is a full blown pair of Uas with all the transactional and dialogue =
stuff.</FONT></P>

<P><FONT SIZE=3D2>What we build is something in the middle... We rely on th=
e &quot;real&quot; UAs to do all the transactions, retries dialogs etc, and=
 we modify the messages and SDP bodies as they go through.</FONT></P>

<P><FONT SIZE=3D2>Personally I don't care what the spec says. We build a sy=
stem that does a useful function, if it doesn't fit into a category that is=
 defined by the spec -- so what? However the marketing types don't quite se=
e it that way ;-)</FONT></P>

<P><FONT SIZE=3D2>If the spec is going to define the permissible intermedia=
ries, it SHOULD allow for things that are not implemented as UAs but extern=
ally appear to be so.</FONT></P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Mary Barnes [<A HREF=3D"mailto:mbarnes@nortelnetwo=
rks.com">mailto:mbarnes@nortelnetworks.com</A>] </FONT>
<BR><FONT SIZE=3D2>Sent: 02 December 2002 16:54</FONT>
<BR><FONT SIZE=3D2>To: 'Phil Reynolds'; sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: RE: [Sip] B2BUA</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Hi all,</FONT>
</P>

<P><FONT SIZE=3D2>I guess I have to ask, what's the problem with the defini=
tion in RFC 3261? </FONT>
</P>

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_______________________________________________
Sip mailing list  https://www1.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: William Marshall <wtm@research.att.com>
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To: dean.willis@softarmor.com
Cc: mbarnes@nortelnetworks.com, Juan-Carlos.Rojas@alcatel.fr, sip@ietf.org
Subject: RE: [Sip] B2BUA
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Dean wrote:
> I think that Bill is suggesting that there is also something BETWEEN the
> B2BUA and the conformant proxy, and that the world would be a better
> place (ok, maybe not THAT much better) if our taxonomy also included
> this "between" thing.

To further elaborate, there certainly exists on paper a box that does
some abritrary and weird algorithm, not conformant with any of the
rules in rfc3261, that transforms requests and response and breaks
everything that goes through it.  Call it "Dean's devilish-B2BUA".

But I don't think anyone builds such a box.  At least not intentionally.

What is built instead are boxes that are close to the definition of
proxies, but not quite.  They serve a real need in the market (and
vendors tend to build what sells).  But due to various details, like
looking at the attachment (a proxy-no-no) to open pinholes in the firewall,
they don't strictly qualify as proxies.

Saying there are only two types of boxes, proxies and devilish-B2BUAs,
ignores the reality.

The writing of an accurate definition of a B2BUA is only the first step.
Unfortunately there is interaction with definition of new services.  The
definition of proxy is very restrictive for a reason, so that it leaves
wide open the possibilities for designs of new services.  But the
not-quite-proxy boxes will exist, and endpoints will need to interoperate
with them.  So services will also need to work through them.  This will
impose some restrictions on how new services are defined.  I think the
objective should be to minimize those restrictions; to do that the variations
in not-quite-proxy behavior need to be documented and made predictable so
that services can be defined to interwork with them.

Having services work through the "something BETWEEN the B2BUA and the
conformant proxy" would certainly make the world a much better place
(yes, THAT much better).

Bill Marshall
wtm@research.att.com

-----original message-----
From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Mary Barnes'" <mbarnes@nortelnetworks.com>,
        <Juan-Carlos.Rojas@alcatel.fr>
Cc: <sip@ietf.org>
Subject: RE: [Sip] B2BUA
Date: Tue, 3 Dec 2002 00:00:19 -0600

Mary said:
 -----
 So, I guess you're suggesting that the term "concatenation" is not
accurate or is misleading and I guess that depends upon how loosely one
applies the term concatenation here; my assumption had been that it
meant that they were "linked" (consistent with Webster's definition of
concatenate) in some manner which is unprescribed, so I think your point
2 below isn't a problem with the current definition.  My understanding
of the current definition is that there isn't anything more for the SIP
protocol to prescribe about the operation of a B2BUA other than what is
required from a protocol perspective to function as a UAC and a UAS.
>From a protocol perspective, it is a logical entity and thus whether
there's one or multiple doesn't seem to apply.  I think what you (and
perhaps others) are suggesting is that there's a definition of a B2BUA
that more closely parallels the definition of a Proxy Server since in
many cases the physical rendition of the logical B2BUA functionality
would be replacing a Proxy Server in a network. And I think this is
where the problem begins, as many are opposed to this model (as it
increases the complexity, etc.) and this is why the logical definition
has remained.  
------ 
 
Well, I think the argument is being made that this definition doesn't
(for some reason that I seem incapable of comprehending) work for some
people, and that therefore we should clarify it. At the fundamental
level, I think some people do not find the implications of this
definition obvious. There are several ways we might address this. We
might add examples. We might add "In other words, the only relationship
between the UAC and the UAS is the application logic, and that can do
anything that one can imagine an application doing", or we might just
use better wording.
 
I don't think the argument is being made that the definition in 3261 is
wrong, just that we can make it better for those who don't get it the
way it is written. Of course, if we discuss it here long enough, maybe
most of them WILL get it -- but that doesn't help the ones who aren't
"here" yet.
 
I think that Bill is suggesting that there is also something BETWEEN the
B2BUA and the conformant proxy, and that the world would be a better
place (ok, maybe not THAT much better) if our taxonomy also included
this "between" thing.
 
--
Dean
 
_______________________________________________
Sip mailing list  https://www1.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: "Mary Barnes" <mbarnes@nortelnetworks.com>
To: "'William Marshall'" <wtm@research.att.com>, dean.willis@softarmor.com
Cc: Juan-Carlos.Rojas@alcatel.fr, sip@ietf.org
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Bill et al, 

My comments are embedded below [MB]:

-----Original Message-----
From: William Marshall [mailto:wtm@research.att.com]
Sent: Tuesday, December 03, 2002 7:29 AM
To: dean.willis@softarmor.com
Cc: Barnes, Mary [NGC:B602:EXCH]; Juan-Carlos.Rojas@alcatel.fr;
sip@ietf.org
Subject: RE: [Sip] B2BUA


Dean wrote:
> I think that Bill is suggesting that there is also something BETWEEN the
> B2BUA and the conformant proxy, and that the world would be a better
> place (ok, maybe not THAT much better) if our taxonomy also included
> this "between" thing.

To further elaborate, there certainly exists on paper a box that does
some abritrary and weird algorithm, not conformant with any of the
rules in rfc3261, that transforms requests and response and breaks
everything that goes through it.  Call it "Dean's devilish-B2BUA".

But I don't think anyone builds such a box.  At least not intentionally.

What is built instead are boxes that are close to the definition of
proxies, but not quite.  They serve a real need in the market (and
vendors tend to build what sells).  But due to various details, like
looking at the attachment (a proxy-no-no) to open pinholes in the firewall,
they don't strictly qualify as proxies.

[MB]: And I think this is where we get into problems trying to agree what
defines this in-between thing, as everyone has something that is likely
slightly different.  

Saying there are only two types of boxes, proxies and devilish-B2BUAs,
ignores the reality.

The writing of an accurate definition of a B2BUA is only the first step.
Unfortunately there is interaction with definition of new services.  The
definition of proxy is very restrictive for a reason, so that it leaves
wide open the possibilities for designs of new services.  But the
not-quite-proxy boxes will exist, and endpoints will need to interoperate
with them.  So services will also need to work through them.  This will
impose some restrictions on how new services are defined.  I think the
objective should be to minimize those restrictions; to do that the
variations
in not-quite-proxy behavior need to be documented and made predictable so
that services can be defined to interwork with them.

[MB]: Well, perhaps the most reasonable way forward would be to put together
an informational draft defining the various "flavors" of these in-between
things and categorizing the issues for service interworking introduced by
each of the variations.  Some of this, of course, will lead us back to the
list of the reasons why some folks consider these things "evil", but would
this also not lead us down the road of being very prescriptive about how
services work in SIP? 

Having services work through the "something BETWEEN the B2BUA and the
conformant proxy" would certainly make the world a much better place
(yes, THAT much better).

Bill Marshall
wtm@research.att.com

-----original message-----
From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Mary Barnes'" <mbarnes@nortelnetworks.com>,
        <Juan-Carlos.Rojas@alcatel.fr>
Cc: <sip@ietf.org>
Subject: RE: [Sip] B2BUA
Date: Tue, 3 Dec 2002 00:00:19 -0600

Mary said:
 -----
 So, I guess you're suggesting that the term "concatenation" is not
accurate or is misleading and I guess that depends upon how loosely one
applies the term concatenation here; my assumption had been that it
meant that they were "linked" (consistent with Webster's definition of
concatenate) in some manner which is unprescribed, so I think your point
2 below isn't a problem with the current definition.  My understanding
of the current definition is that there isn't anything more for the SIP
protocol to prescribe about the operation of a B2BUA other than what is
required from a protocol perspective to function as a UAC and a UAS.
>From a protocol perspective, it is a logical entity and thus whether
there's one or multiple doesn't seem to apply.  I think what you (and
perhaps others) are suggesting is that there's a definition of a B2BUA
that more closely parallels the definition of a Proxy Server since in
many cases the physical rendition of the logical B2BUA functionality
would be replacing a Proxy Server in a network. And I think this is
where the problem begins, as many are opposed to this model (as it
increases the complexity, etc.) and this is why the logical definition
has remained.  
------ 
 
Well, I think the argument is being made that this definition doesn't
(for some reason that I seem incapable of comprehending) work for some
people, and that therefore we should clarify it. At the fundamental
level, I think some people do not find the implications of this
definition obvious. There are several ways we might address this. We
might add examples. We might add "In other words, the only relationship
between the UAC and the UAS is the application logic, and that can do
anything that one can imagine an application doing", or we might just
use better wording.
 
I don't think the argument is being made that the definition in 3261 is
wrong, just that we can make it better for those who don't get it the
way it is written. Of course, if we discuss it here long enough, maybe
most of them WILL get it -- but that doesn't help the ones who aren't
"here" yet.
 
I think that Bill is suggesting that there is also something BETWEEN the
B2BUA and the conformant proxy, and that the world would be a better
place (ok, maybe not THAT much better) if our taxonomy also included
this "between" thing.
 
--
Dean
 

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<P><FONT SIZE=3D2>Bill et al, </FONT>
</P>

<P><FONT SIZE=3D2>My comments are embedded below [MB]:</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: William Marshall [<A =
HREF=3D"mailto:wtm@research.att.com">mailto:wtm@research.att.com</A>]</F=
ONT>
<BR><FONT SIZE=3D2>Sent: Tuesday, December 03, 2002 7:29 AM</FONT>
<BR><FONT SIZE=3D2>To: dean.willis@softarmor.com</FONT>
<BR><FONT SIZE=3D2>Cc: Barnes, Mary [NGC:B602:EXCH]; =
Juan-Carlos.Rojas@alcatel.fr;</FONT>
<BR><FONT SIZE=3D2>sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: RE: [Sip] B2BUA</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Dean wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; I think that Bill is suggesting that there is =
also something BETWEEN the</FONT>
<BR><FONT SIZE=3D2>&gt; B2BUA and the conformant proxy, and that the =
world would be a better</FONT>
<BR><FONT SIZE=3D2>&gt; place (ok, maybe not THAT much better) if our =
taxonomy also included</FONT>
<BR><FONT SIZE=3D2>&gt; this &quot;between&quot; thing.</FONT>
</P>

<P><FONT SIZE=3D2>To further elaborate, there certainly exists on paper =
a box that does</FONT>
<BR><FONT SIZE=3D2>some abritrary and weird algorithm, not conformant =
with any of the</FONT>
<BR><FONT SIZE=3D2>rules in rfc3261, that transforms requests and =
response and breaks</FONT>
<BR><FONT SIZE=3D2>everything that goes through it.&nbsp; Call it =
&quot;Dean's devilish-B2BUA&quot;.</FONT>
</P>

<P><FONT SIZE=3D2>But I don't think anyone builds such a box.&nbsp; At =
least not intentionally.</FONT>
</P>

<P><FONT SIZE=3D2>What is built instead are boxes that are close to the =
definition of</FONT>
<BR><FONT SIZE=3D2>proxies, but not quite.&nbsp; They serve a real need =
in the market (and</FONT>
<BR><FONT SIZE=3D2>vendors tend to build what sells).&nbsp; But due to =
various details, like</FONT>
<BR><FONT SIZE=3D2>looking at the attachment (a proxy-no-no) to open =
pinholes in the firewall,</FONT>
<BR><FONT SIZE=3D2>they don't strictly qualify as proxies.</FONT>
</P>

<P><FONT SIZE=3D2>[MB]: And I think this is where we get into problems =
trying to agree what defines this in-between thing, as everyone has =
something that is likely slightly different.&nbsp; </FONT></P>

<P><FONT SIZE=3D2>Saying there are only two types of boxes, proxies and =
devilish-B2BUAs,</FONT>
<BR><FONT SIZE=3D2>ignores the reality.</FONT>
</P>

<P><FONT SIZE=3D2>The writing of an accurate definition of a B2BUA is =
only the first step.</FONT>
<BR><FONT SIZE=3D2>Unfortunately there is interaction with definition =
of new services.&nbsp; The</FONT>
<BR><FONT SIZE=3D2>definition of proxy is very restrictive for a =
reason, so that it leaves</FONT>
<BR><FONT SIZE=3D2>wide open the possibilities for designs of new =
services.&nbsp; But the</FONT>
<BR><FONT SIZE=3D2>not-quite-proxy boxes will exist, and endpoints will =
need to interoperate</FONT>
<BR><FONT SIZE=3D2>with them.&nbsp; So services will also need to work =
through them.&nbsp; This will</FONT>
<BR><FONT SIZE=3D2>impose some restrictions on how new services are =
defined.&nbsp; I think the</FONT>
<BR><FONT SIZE=3D2>objective should be to minimize those restrictions; =
to do that the variations</FONT>
<BR><FONT SIZE=3D2>in not-quite-proxy behavior need to be documented =
and made predictable so</FONT>
<BR><FONT SIZE=3D2>that services can be defined to interwork with =
them.</FONT>
</P>

<P><FONT SIZE=3D2>[MB]: Well, perhaps the most reasonable way forward =
would be to put together an informational draft defining the various =
&quot;flavors&quot; of these in-between things and categorizing the =
issues for service interworking introduced by each of the variations.&nb=
sp; Some of this, of course, will lead us back to the list of the =
reasons why some folks consider these things &quot;evil&quot;, but =
would this also not lead us down the road of being very prescriptive =
about how services work in SIP? </FONT></P>

<P><FONT SIZE=3D2>Having services work through the &quot;something =
BETWEEN the B2BUA and the</FONT>
<BR><FONT SIZE=3D2>conformant proxy&quot; would certainly make the =
world a much better place</FONT>
<BR><FONT SIZE=3D2>(yes, THAT much better).</FONT>
</P>

<P><FONT SIZE=3D2>Bill Marshall</FONT>
<BR><FONT SIZE=3D2>wtm@research.att.com</FONT>
</P>

<P><FONT SIZE=3D2>-----original message-----</FONT>
<BR><FONT SIZE=3D2>From: &quot;Dean Willis&quot; =
&lt;dean.willis@softarmor.com&gt;</FONT>
<BR><FONT SIZE=3D2>To: &quot;'Mary Barnes'&quot; =
&lt;mbarnes@nortelnetworks.com&gt;,</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&lt;Juan-Carlos.Rojas@alcatel.fr&gt;</FONT>
<BR><FONT SIZE=3D2>Cc: &lt;sip@ietf.org&gt;</FONT>
<BR><FONT SIZE=3D2>Subject: RE: [Sip] B2BUA</FONT>
<BR><FONT SIZE=3D2>Date: Tue, 3 Dec 2002 00:00:19 -0600</FONT>
</P>

<P><FONT SIZE=3D2>Mary said:</FONT>
<BR><FONT SIZE=3D2>&nbsp;-----</FONT>
<BR><FONT SIZE=3D2>&nbsp;So, I guess you're suggesting that the term =
&quot;concatenation&quot; is not</FONT>
<BR><FONT SIZE=3D2>accurate or is misleading and I guess that depends =
upon how loosely one</FONT>
<BR><FONT SIZE=3D2>applies the term concatenation here; my assumption =
had been that it</FONT>
<BR><FONT SIZE=3D2>meant that they were &quot;linked&quot; (consistent =
with Webster's definition of</FONT>
<BR><FONT SIZE=3D2>concatenate) in some manner which is unprescribed, =
so I think your point</FONT>
<BR><FONT SIZE=3D2>2 below isn't a problem with the current =
definition.&nbsp; My understanding</FONT>
<BR><FONT SIZE=3D2>of the current definition is that there isn't =
anything more for the SIP</FONT>
<BR><FONT SIZE=3D2>protocol to prescribe about the operation of a B2BUA =
other than what is</FONT>
<BR><FONT SIZE=3D2>required from a protocol perspective to function as =
a UAC and a UAS.</FONT>
<BR><FONT SIZE=3D2>&gt;From a protocol perspective, it is a logical =
entity and thus whether</FONT>
<BR><FONT SIZE=3D2>there's one or multiple doesn't seem to apply.&nbsp; =
I think what you (and</FONT>
<BR><FONT SIZE=3D2>perhaps others) are suggesting is that there's a =
definition of a B2BUA</FONT>
<BR><FONT SIZE=3D2>that more closely parallels the definition of a =
Proxy Server since in</FONT>
<BR><FONT SIZE=3D2>many cases the physical rendition of the logical =
B2BUA functionality</FONT>
<BR><FONT SIZE=3D2>would be replacing a Proxy Server in a network. And =
I think this is</FONT>
<BR><FONT SIZE=3D2>where the problem begins, as many are opposed to =
this model (as it</FONT>
<BR><FONT SIZE=3D2>increases the complexity, etc.) and this is why the =
logical definition</FONT>
<BR><FONT SIZE=3D2>has remained.&nbsp; </FONT>
<BR><FONT SIZE=3D2>------ </FONT>
<BR><FONT SIZE=3D2>&nbsp;</FONT>
<BR><FONT SIZE=3D2>Well, I think the argument is being made that this =
definition doesn't</FONT>
<BR><FONT SIZE=3D2>(for some reason that I seem incapable of =
comprehending) work for some</FONT>
<BR><FONT SIZE=3D2>people, and that therefore we should clarify it. At =
the fundamental</FONT>
<BR><FONT SIZE=3D2>level, I think some people do not find the =
implications of this</FONT>
<BR><FONT SIZE=3D2>definition obvious. There are several ways we might =
address this. We</FONT>
<BR><FONT SIZE=3D2>might add examples. We might add &quot;In other =
words, the only relationship</FONT>
<BR><FONT SIZE=3D2>between the UAC and the UAS is the application =
logic, and that can do</FONT>
<BR><FONT SIZE=3D2>anything that one can imagine an application =
doing&quot;, or we might just</FONT>
<BR><FONT SIZE=3D2>use better wording.</FONT>
<BR><FONT SIZE=3D2>&nbsp;</FONT>
<BR><FONT SIZE=3D2>I don't think the argument is being made that the =
definition in 3261 is</FONT>
<BR><FONT SIZE=3D2>wrong, just that we can make it better for those who =
don't get it the</FONT>
<BR><FONT SIZE=3D2>way it is written. Of course, if we discuss it here =
long enough, maybe</FONT>
<BR><FONT SIZE=3D2>most of them WILL get it -- but that doesn't help =
the ones who aren't</FONT>
<BR><FONT SIZE=3D2>&quot;here&quot; yet.</FONT>
<BR><FONT SIZE=3D2>&nbsp;</FONT>
<BR><FONT SIZE=3D2>I think that Bill is suggesting that there is also =
something BETWEEN the</FONT>
<BR><FONT SIZE=3D2>B2BUA and the conformant proxy, and that the world =
would be a better</FONT>
<BR><FONT SIZE=3D2>place (ok, maybe not THAT much better) if our =
taxonomy also included</FONT>
<BR><FONT SIZE=3D2>this &quot;between&quot; thing.</FONT>
<BR><FONT SIZE=3D2>&nbsp;</FONT>
<BR><FONT SIZE=3D2>--</FONT>
<BR><FONT SIZE=3D2>Dean</FONT>
<BR><FONT SIZE=3D2>&nbsp;</FONT>
</P>

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To: mbarnes@nortelnetworks.com
Cc: dean.willis@softarmor.com, Juan-Carlos.Rojas@alcatel.fr, sip@ietf.org
Subject: RE: [Sip] B2BUA
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Mary wrote:
> [MB]: Well, perhaps the most reasonable way forward would be to put together
> an informational draft defining the various "flavors" of these in-between
> things and categorizing the issues for service interworking introduced by
> each of the variations.  Some of this, of course, will lead us back to the
> list of the reasons why some folks consider these things "evil", but would
> this also not lead us down the road of being very prescriptive about how
> services work in SIP? 

Sounds good.

In all these different "flavors" there are likely lots of dont-cares in
how to handle requests/responses/errors not directly related to their function.
I think this could also serve as a BCP-like document listing how these
issues should be resolved to maximize interoperability.

Bill Marshall
wtm@research.att.com

-----original message-----
From: "Mary Barnes" <mbarnes@nortelnetworks.com>
To: "'William Marshall'" <wtm@research.att.com>, dean.willis@softarmor.com
Cc: Juan-Carlos.Rojas@alcatel.fr, sip@ietf.org
Subject: RE: [Sip] B2BUA
Date: Tue, 3 Dec 2002 07:35:26 -0600 

Bill et al, 

My comments are embedded below [MB]:

-----Original Message-----
From: William Marshall [mailto:wtm@research.att.com]
Sent: Tuesday, December 03, 2002 7:29 AM
To: dean.willis@softarmor.com
Cc: Barnes, Mary [NGC:B602:EXCH]; Juan-Carlos.Rojas@alcatel.fr;
sip@ietf.org
Subject: RE: [Sip] B2BUA


Dean wrote:
> I think that Bill is suggesting that there is also something BETWEEN the
> B2BUA and the conformant proxy, and that the world would be a better
> place (ok, maybe not THAT much better) if our taxonomy also included
> this "between" thing.

To further elaborate, there certainly exists on paper a box that does
some abritrary and weird algorithm, not conformant with any of the
rules in rfc3261, that transforms requests and response and breaks
everything that goes through it.  Call it "Dean's devilish-B2BUA".

But I don't think anyone builds such a box.  At least not intentionally.

What is built instead are boxes that are close to the definition of
proxies, but not quite.  They serve a real need in the market (and
vendors tend to build what sells).  But due to various details, like
looking at the attachment (a proxy-no-no) to open pinholes in the firewall,
they don't strictly qualify as proxies.

[MB]: And I think this is where we get into problems trying to agree what
defines this in-between thing, as everyone has something that is likely
slightly different.  

Saying there are only two types of boxes, proxies and devilish-B2BUAs,
ignores the reality.

The writing of an accurate definition of a B2BUA is only the first step.
Unfortunately there is interaction with definition of new services.  The
definition of proxy is very restrictive for a reason, so that it leaves
wide open the possibilities for designs of new services.  But the
not-quite-proxy boxes will exist, and endpoints will need to interoperate
with them.  So services will also need to work through them.  This will
impose some restrictions on how new services are defined.  I think the
objective should be to minimize those restrictions; to do that the
variations
in not-quite-proxy behavior need to be documented and made predictable so
that services can be defined to interwork with them.

[MB]: Well, perhaps the most reasonable way forward would be to put together
an informational draft defining the various "flavors" of these in-between
things and categorizing the issues for service interworking introduced by
each of the variations.  Some of this, of course, will lead us back to the
list of the reasons why some folks consider these things "evil", but would
this also not lead us down the road of being very prescriptive about how
services work in SIP? 

Having services work through the "something BETWEEN the B2BUA and the
conformant proxy" would certainly make the world a much better place
(yes, THAT much better).

Bill Marshall
wtm@research.att.com

-----original message-----
From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Mary Barnes'" <mbarnes@nortelnetworks.com>,
        <Juan-Carlos.Rojas@alcatel.fr>
Cc: <sip@ietf.org>
Subject: RE: [Sip] B2BUA
Date: Tue, 3 Dec 2002 00:00:19 -0600

Mary said:
 -----
 So, I guess you're suggesting that the term "concatenation" is not
accurate or is misleading and I guess that depends upon how loosely one
applies the term concatenation here; my assumption had been that it
meant that they were "linked" (consistent with Webster's definition of
concatenate) in some manner which is unprescribed, so I think your point
2 below isn't a problem with the current definition.  My understanding
of the current definition is that there isn't anything more for the SIP
protocol to prescribe about the operation of a B2BUA other than what is
required from a protocol perspective to function as a UAC and a UAS.
>From a protocol perspective, it is a logical entity and thus whether
there's one or multiple doesn't seem to apply.  I think what you (and
perhaps others) are suggesting is that there's a definition of a B2BUA
that more closely parallels the definition of a Proxy Server since in
many cases the physical rendition of the logical B2BUA functionality
would be replacing a Proxy Server in a network. And I think this is
where the problem begins, as many are opposed to this model (as it
increases the complexity, etc.) and this is why the logical definition
has remained.  
------ 
 
Well, I think the argument is being made that this definition doesn't
(for some reason that I seem incapable of comprehending) work for some
people, and that therefore we should clarify it. At the fundamental
level, I think some people do not find the implications of this
definition obvious. There are several ways we might address this. We
might add examples. We might add "In other words, the only relationship
between the UAC and the UAS is the application logic, and that can do
anything that one can imagine an application doing", or we might just
use better wording.
 
I don't think the argument is being made that the definition in 3261 is
wrong, just that we can make it better for those who don't get it the
way it is written. Of course, if we discuss it here long enough, maybe
most of them WILL get it -- but that doesn't help the ones who aren't
"here" yet.
 
I think that Bill is suggesting that there is also something BETWEEN the
B2BUA and the conformant proxy, and that the world would be a better
place (ok, maybe not THAT much better) if our taxonomy also included
this "between" thing.
 
--
Dean
 
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Subject: RE: [Sip] B2BUA
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The trouble with that is that the RFC stipulates the way these
intermediaries work and unless your intermediaries work in that way, you
cant put "CONFORMS TO RFCxyz" on the box and hence the spec is stifling my
creativity (and more importantly my marketability).
 
The spec should stick to defining the "on wire" protocols and let us
"creative types" come up with imaginative ways to subvert them :-)

-----Original Message-----
From: Mary Barnes [mailto:mbarnes@nortelnetworks.com] 
Sent: 03 December 2002 13:20
To: 'Phil Reynolds'
Subject: RE: [Sip] B2BUA


Phil,
 
I do see both sides of this debate, but I've yet to see it come to a
resolution to which everyone can agree, as everyone has built something
that's slightly different (and yes our marketing folks want each of ours to
be the best of such different things). Realistically, building anything
beyond a basic proxy is a tradeoff between desired (and many believe
required) functionality versus the complexity and the potential for
interworking problems. I think it's the fundamental problem that the IETF
doesn't define architectures, but rather protocols.  However, to define the
protocols, you must define some architectural elements to set the context
and that's where the problems begin, as everyone's implementation of the
architectural elements is always different because the specs don't provide
enough detail. 
 
So, perhaps the best way around this is to leave the basic SIP spec as is
and write an informational RFC defining the various flavors of entities
"Beyond SIP Proxys"  and "Towards B2BUAs" so that we can give our marketing
folks a pointer to a document that describes at some level the "flavor" of
system we provide.  I say this rather facetiously, but realistically, it
might be the most effective way to resolve this problem and end this long
debates. 
 
Regards,
Mary. 
 
 
 -----Original Message-----
From: Phil Reynolds [mailto:preynolds@ridgewaysystems.com]
Sent: Tuesday, December 03, 2002 4:05 AM
To: Barnes, Mary [NGC:B602:EXCH]
Subject: RE: [Sip] B2BUA



The problem as I see it is that the RFC defines 2 distinct intermediary
entities a "SIP Proxy" which has limited functionality, i.e. it is not
allowed to modify the body of the message, and a B2BUA which is a full blown
pair of Uas with all the transactional and dialogue stuff.

What we build is something in the middle... We rely on the "real" UAs to do
all the transactions, retries dialogs etc, and we modify the messages and
SDP bodies as they go through.

Personally I don't care what the spec says. We build a system that does a
useful function, if it doesn't fit into a category that is defined by the
spec -- so what? However the marketing types don't quite see it that way ;-)

If the spec is going to define the permissible intermediaries, it SHOULD
allow for things that are not implemented as UAs but externally appear to be
so.

-----Original Message----- 
From: Mary Barnes [mailto:mbarnes@nortelnetworks.com
<mailto:mbarnes@nortelnetworks.com> ] 
Sent: 02 December 2002 16:54 
To: 'Phil Reynolds'; sip@ietf.org 
Subject: RE: [Sip] B2BUA 


Hi all, 

I guess I have to ask, what's the problem with the definition in RFC 3261? 


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<DIV><FONT face=Arial color=#008080 size=2><SPAN class=652084413-03122002>The 
trouble with that is that the RFC stipulates the way these intermediaries work 
and unless your intermediaries work in that way, you cant put "CONFORMS TO 
RFCxyz" on the box and hence the spec is stifling my creativity (and more 
importantly my marketability).</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#008080 size=2><SPAN 
class=652084413-03122002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial color=#008080 size=2><SPAN class=652084413-03122002>The 
spec should stick to defining the "on wire"&nbsp;protocols and let us "creative 
types" come up with imaginative ways to subvert them :-)</SPAN></FONT></DIV>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left><FONT 
  face=Tahoma size=2>-----Original Message-----<BR><B>From:</B> Mary Barnes 
  [mailto:mbarnes@nortelnetworks.com] <BR><B>Sent:</B> 03 December 2002 
  13:20<BR><B>To:</B> 'Phil Reynolds'<BR><B>Subject:</B> RE: [Sip] 
  B2BUA<BR><BR></FONT></DIV>
  <DIV><FONT color=#0000ff size=2><SPAN 
  class=614313513-03122002>Phil,</SPAN></FONT></DIV>
  <DIV><FONT color=#0000ff size=2><SPAN 
  class=614313513-03122002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT color=#0000ff size=2><SPAN class=614313513-03122002>I do see both 
  sides of this debate, but I've yet to see it come to a resolution to which 
  everyone can agree, as everyone has built something that's slightly different 
  (and yes our marketing folks want each of ours to be the best of such 
  different things).&nbsp;Realistically, building anything beyond a basic proxy 
  is a tradeoff between desired (and many believe required) functionality versus 
  the complexity and the potential for interworking problems. I think it's the 
  fundamental problem that the IETF doesn't define architectures, but rather 
  protocols.&nbsp; However, to define the protocols, you must define some 
  architectural elements to set the context and that's where the problems begin, 
  as everyone's implementation&nbsp;of the architectural elements is always 
  different because the specs don't provide enough detail. </SPAN></FONT></DIV>
  <DIV><FONT color=#0000ff size=2><SPAN 
  class=614313513-03122002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT color=#0000ff size=2><SPAN class=614313513-03122002>So, perhaps the 
  best way around this is to leave the basic SIP spec as is and write an 
  informational RFC defining the various flavors of entities "Beyond SIP 
  Proxys"&nbsp; and "Towards B2BUAs" so that we can give our marketing folks a 
  pointer to a document that describes at some level the "flavor" of system we 
  provide.&nbsp; I say this rather facetiously, but realistically, it might be 
  the most effective way to resolve this problem and end this long debates. 
  </SPAN></FONT></DIV>
  <DIV><FONT color=#0000ff size=2><SPAN 
  class=614313513-03122002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT color=#0000ff size=2><SPAN 
  class=614313513-03122002>Regards,</SPAN></FONT></DIV>
  <DIV><FONT color=#0000ff size=2><SPAN class=614313513-03122002>Mary. 
  </SPAN></FONT></DIV>
  <DIV><SPAN class=614313513-03122002></SPAN><FONT face=Tahoma><FONT 
  size=2><SPAN class=614313513-03122002><FONT face=Arial 
  color=#0000ff></FONT></SPAN></FONT></FONT>&nbsp;</DIV>
  <DIV><FONT face=Tahoma><FONT size=2><SPAN 
  class=614313513-03122002></SPAN></FONT></FONT>&nbsp;</DIV>
  <DIV><FONT face=Tahoma><FONT size=2><SPAN 
  class=614313513-03122002>&nbsp;</SPAN>-----Original 
  Message-----<BR><B>From:</B> Phil Reynolds 
  [mailto:preynolds@ridgewaysystems.com]<BR><B>Sent:</B> Tuesday, December 03, 
  2002 4:05 AM<BR><B>To:</B> Barnes, Mary [NGC:B602:EXCH]<BR><B>Subject:</B> RE: 
  [Sip] B2BUA<BR><BR></DIV></FONT>
  <BLOCKQUOTE style="MARGIN-RIGHT: 0px"></FONT>
    <P><FONT size=2>The problem as I see it is that the RFC defines 2 distinct 
    intermediary entities a "SIP Proxy" which has limited functionality, i.e. it 
    is not allowed to modify the body of the message, and a B2BUA which is a 
    full blown pair of Uas with all the transactional and dialogue 
    stuff.</FONT></P>
    <P><FONT size=2>What we build is something in the middle... We rely on the 
    "real" UAs to do all the transactions, retries dialogs etc, and we modify 
    the messages and SDP bodies as they go through.</FONT></P>
    <P><FONT size=2>Personally I don't care what the spec says. We build a 
    system that does a useful function, if it doesn't fit into a category that 
    is defined by the spec -- so what? However the marketing types don't quite 
    see it that way ;-)</FONT></P>
    <P><FONT size=2>If the spec is going to define the permissible 
    intermediaries, it SHOULD allow for things that are not implemented as UAs 
    but externally appear to be so.</FONT></P>
    <P><FONT size=2>-----Original Message-----</FONT> <BR><FONT size=2>From: 
    Mary Barnes [<A 
    href="mailto:mbarnes@nortelnetworks.com">mailto:mbarnes@nortelnetworks.com</A>] 
    </FONT><BR><FONT size=2>Sent: 02 December 2002 16:54</FONT> <BR><FONT 
    size=2>To: 'Phil Reynolds'; sip@ietf.org</FONT> <BR><FONT size=2>Subject: 
    RE: [Sip] B2BUA</FONT> </P><BR>
    <P><FONT size=2>Hi all,</FONT> </P>
    <P><FONT size=2>I guess I have to ask, what's the problem with the 
    definition in RFC 3261? </FONT></P></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

----=_NextPart_ST_13_54_22_Tuesday_December_03_2002_30999--

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Hi,

I agree with Mary that concatenation is the right word for it. But the source of confusion is probably this:
 
UA = UAC + UAS

B2BUA = UA | UA

=> B2BUA = (UAC + UAS) | (UAC + UAS) 

So the B2BUA is a concatenation of two User Agents, each of which acts both as UAC and UAS. A conference server will probably look like a UA, definitely not a B2BUA, IMHO.
   
Cheers,
Aki

-----Original Message-----
From: ext Mary Barnes [mailto:mbarnes@nortelnetworks.com]
Sent: Monday, December 02, 2002 7:47 PM
To: 'Juan-Carlos.Rojas@alcatel.fr'
Cc: sip@ietf.org
Subject: RE: [Sip] B2BUA


Juan-Carlos, 
So, I guess you're suggesting that the term "concatenation" is not accurate or is misleading and I guess that depends upon how loosely one applies the term concatenation here; my assumption had been that it meant that they were "linked" (consistent with Webster's definition of concatenate) in some manner which is unprescribed, so I think your point 2 below isn't a problem with the current definition.  My understanding of the current definition is that there isn't anything more for the SIP protocol to prescribe about the operation of a B2BUA other than what is required from a protocol perspective to function as a UAC and a UAS. From a protocol perspective, it is a logical entity and thus whether there's one or multiple doesn't seem to apply.  I think what you (and perhaps others) are suggesting is that there's a definition of a B2BUA that more closely parallels the definition of a Proxy Server since in many cases the physical rendition of the logical B2BUA functionality would!
 be replacing a Proxy Server in a network. And I think this is where the problem begins, as many are opposed to this model (as it increases the complexity, etc.) and this is why the logical definition has remained. 
On your first point, however, I think we need to wait until the conferencing work in SIPPING is more complete before we can suggest that a conference server would be a B2BUA. 
Mary. 
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From: "Cullen Jennings" <fluffy@cisco.com>
To: <sip@ietf.org>
Subject: RE: [Sip] B2BUA
Date: Mon, 2 Dec 2002 22:12:17 -0800
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One problem is that people often want a definition of what a B2BUA can or
can not do. It is not clear that it is possible to define a SIP speaking
systems this way.

What function can a proxy provide? Well just about anything as long as the
responses it sends are compliant with the constraints of a thing called a
proxy in 3261. How a proxy determines the next hop for an requested targeted
to it is completely up to the proxy and not specified in SIP.

We run into the same problem with defining what can a B2BUA do. The answer
is, well lots of things. It can handle multiple different dialogs at the
same time and the message sent on each dialogs will meet the constraints of
a what 3261 calls a UA. How the message on one dialog relate to the message
on another dialog is completely up to the B2BUA and not specified in SIP.

I recognize this is not a very satisfactory way to help people understand
B2BUA so I have a proposal. Trying to have an all encompassing functional
definition of a B2BUA is very hard. However, it is possible to have call
flows that show some common application of B2BUA. Perhaps even form some sub
classification of a B2BUA that specify how a given class of B2B will
interact with messages.

If you think about it, this is similar to the SIP to QSIG mapping. We say
how messages on one side will get mapped to message on the other side. It's
not a normative part of SIP.

I think a few folks are already working on this - perhaps we will get a
whole taxonomy of different variants of the B2BUA beast.

Cullen


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From: "F.S.Salloum" <ssal@intracom.gr>
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Subject: RE: SIPS related clarifications (was: [Sip] SIPS question)
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Dear all,


<Siddharth>
	>Lets say that the request-uri does not contain
	>"sips:" but the request was received on TLS (perhaps
	>because of a Route header configured with a previous
	>hop). Can the proxy send it downstream on a non-TLS
	>protocol if there are no more Route headers present
	>in the message?
	>
	>From your mail, I gather this is allowed.
</Siddharth>

I believe that the difference between sip over TLS and
sips over TLS is the following

Using sip: over TLS the UA that sends the requests over
TLS to a proxy server has no assurance that TLS will be used
end-to-end  (RFC3261:pp238)

Using sips: over TLS the SIPS scheme signifies that each
hop over which the request is forwarded, until the request
reaches the SIP entity responsible for the domain portion of the
Request URI must be secured with TLS; once it reaches the domain
in question it is handled in accordance with local security and
routing policy, quite possibly using TLS for any last hop to a UAS.
(RFC3261:pp238)


<Siddharth>
	<snip>
	>>  while Rosenberg had mentioned that the presence
	>>   of "sips:" should imply that the end-to-end path is
	>>   secure.
	>
	> If SIPS appears in the Request-URI of the original
	> request sent by the UAC, it implies this end-to-end
	> path security.
	</snip>
</Siddharth>


In this case end-to-end actually referes the UAC and the first entity
in the domain in question then it depends on the local security and routing
policy
of that domain.

<Siddharth>
	<snip>
	>> I guess the way I understand the spec stands
	>> at present is that if the Req-URI is re-written to a
	>> "sips:" URL at any stage of the routing, then any hop
	>> downstream _from_this_hop_onwards_ has to forward the
	>> request on TLS.
	>
	> I agree, that is how we should understand it - with the
	> caveat that SIPS only applies to the entire path until
	> it reaches the domain indicated by the hostname portion
	> of the Request-URI. Once a proxy server in the destination
	> gets hold of the request, it can retarget it arbitrarily
	> (to non-SIPS resources, if it wishes).
	</snip>
	Aren't the 2 texts quoted by you at opposite ends?

	a) the UAS cannot be sure of end-to-end security if it
	   received a INVITE with the request-URI as "sips:" ...
	   Some hop in the middle could have modified the req-URI
	   for "sips:". It may even be the proxy with whom the UAS
	   registered with the Contact as "sips:" ?

	The UAS by parsing the sips is sure that security exist
	between the previous hop and him.
	The UAC initiator is the only one sure for end-to-end security (till the
domain etc)
	because he has initiated a session using sips:

	b) (If the caveat that you mentioned holds) the UAC
	   cannot be sure that the UAS received the request
	   on TLS. Shouldn't the destination's proxy be
	   disallowed to route a request received on TLS to
	   a non-TLS client?
</Siddharth>

I believe that end-to-end security is reffered between the UAC and
the first entity in the domain in question and not the final UAS this
depends on routing and local security policy. For example the edge proxy
may connect with the UAs or the final UAC using IPSec since maybe the UAs
have a pre-shared kwying relationship with their first-hop proxy server.

The sure thing is that when something is sent with sips then the sender is
sure that
it will be sent securely till the referenced domain ...

if a UA receives something with sips it is sure that has been sent securely
from the previous hop...




Regards
F.S.Salloum
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Mary wrote:
> [MB]: Some of this, of course, will lead us back to the
> list of the reasons why some folks consider these things "evil", ... 

I do not remember whether anyone has ever said that B2BUA is evil
things. What I remember is that the dumping of the B2BUA functionality
in the name of SIP Proxy is an evil thing.
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From: "F.S.Salloum" <ssal@intracom.gr>
To: "Sip@Ietf. Org" <sip@ietf.org>
Date: Tue, 3 Dec 2002 17:21:04 +0200
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Subject: [Sip] Missing TLS?
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Dear All,

On page RFC 3261/pp151 on comment (1) I think
its missing an TLS, it should be changed from

"The default is 5061 for sip: using TLS over TCP and
sips: over TCP."

To 

"The default is 5061 for sip: using TLS over TCP and
sips: TLS over TCP."


Regards 
F.S.Salloum

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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'aki.niemi@nokia.com'" <aki.niemi@nokia.com>
Cc: sip@ietf.org
Subject: RE: [Sip] B2BUA
Date: Tue, 3 Dec 2002 11:29:15 -0500 
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> So the B2BUA is a concatenation of two User Agents, each of 
> which acts both as UAC and UAS. A conference server will 
> probably look like a UA, definitely not a B2BUA, IMHO.
>    
This thread is, hopefully reaching an end (with an agreement to
do some examples with call flows).  However, the above statement
is yet one more example of the confusion people have with B2BUA.

The key thing about a B2BUA is that it looks like a UA to whatever
it is connected to (on both sides).  So, a conference server
looks like a UA to each of it's ports, and it has at least two of
them, and there is some "magic" between them.  If you look at the
concept of the "focus" from the conferencing work, it is a
thing that has a bunch of two ended SIP calls, one with each
participant.  Each participant thinks it is talking to a UA.
While some messages sent from a participant to a UA might be
consumed by the UA (state changes), often, a message will trigger
messages from the focus to other participants.

This seems, to me, to meet the definition of B2BUA in 3261,
and more strongly meets proposed changes in the definition.
Of course, it really doesn't matter if you consider the focus to
be a B2BUA or not.

Brian 
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From: "Christian Huitema" <huitema@windows.microsoft.com>
To: "Roy, Radhika R, ALASO" <rrroy@att.com>, <mbarnes@nortelnetworks.com>
Cc: <dean.willis@softarmor.com>, <Juan-Carlos.Rojas@alcatel.fr>,
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> > [MB]: Some of this, of course, will lead us back to the
> > list of the reasons why some folks consider these things "evil", ...
> 
> I do not remember whether anyone has ever said that B2BUA is evil
> things. What I remember is that the dumping of the B2BUA functionality
> in the name of SIP Proxy is an evil thing.

The real issue in all this is security. Many security folks believe
that, in order to get proper security, the body of the signaling
messages should be signed, and possibly encrypted. In that logic, the
proxies only manipulate the information exposed in the SIP header, which
can be as little as the pair of source and destination URL.

Any intermediary that attempts to "rewrite the SDP" has to somehow
insert itself in the end-to-end security exchange; many security folks
will resent that as "breaking end-to-end security". Insofar as SDP
rewriting attempts to provide a legitimate service, the challenge is
thus to provide the service in such a way that it does not require
breaking end-to-end encryption.

-- Christian Huitema
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From mailnull@www1.ietf.org  Tue Dec  3 11:55:08 2002
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From: Robert Sparks <rsparks@dynamicsoft.com>
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Subject: [Sip] Constructing the branch parameter at stateless proxies
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This list has seen a couple of requests to clarify the language in
RFC 3261 concerning how a stateless proxy creates a branch identifier.

The current text requires uniqueness, dependency on route decision
affecting parameters and care to be sure that those parameters are
retransmission invariant. It goes on to recommend a hash over a
set of fields (this set has been the subject of the requests for
clarification).

I think there's a much simpler solution. The purpose of this 
branch parameter is to act as a transaction identifier. Who
cares about this identifier? Only the stateful elements at
each end of the transaction. Any stateless proxies between
these elements don't use that identifier in any of their
decisions, so I assert that it is not useful to change it
as it crosses the stateless proxy.

I propose, then, that a stateless proxy simply copy
the branch parameter from the topmost Via header field
value it receives into the Via header field value it adds
to the request. 

I don't believe it matters that this parameter contain the
magic cookie. If it doesn't, the next stateful hop will use
the 2543-style transaction identifiers provided by the sending
stateful element. The stateless elements are by definition
transaction transparent.

So - can anyone construct a scenario where this doesn't 
function correctly? If not, we can remove several convoluted
paragraphs from the spec and replace them with a very simple
couple of sentences.

RjS



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Subject: RE: [Sip] B2BUA
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Inline.

> The key thing about a B2BUA is that it looks like a UA to whatever
> it is connected to (on both sides).  So, a conference server
> looks like a UA to each of it's ports, and it has at least two of
> them, and there is some "magic" between them.  If you look at the
> concept of the "focus" from the conferencing work, it is a
> thing that has a bunch of two ended SIP calls, one with each
> participant.  Each participant thinks it is talking to a UA.
> While some messages sent from a participant to a UA might be
> consumed by the UA (state changes), often, a message will trigger
> messages from the focus to other participants.

This just generalized the B2BUA to cover any normal UA behavior. Example, I get a MESSAGE from a friend to my UA, based on which decide to call another friend by sending an INVITE from that UA. This I think is quite normal UA operation, the "magic" being me via my UI.

> This seems, to me, to meet the definition of B2BUA in 3261,
> and more strongly meets proposed changes in the definition.
> Of course, it really doesn't matter if you consider the focus to
> be a B2BUA or not.

Hmm... Let's think of the B2BUA as 'in-UA' concatenated with 'out-UA'. By definition:

1) in-UAS receives request,
   <Magic happens>
2) out-UAC sends request'
3) out-UAC receives response'
   <Magic happens>
4) in-UAS sends response

(The same process happens with requests going from 'out' to 'in')

Within this B2BUA, 'in-UAS' shares magic with 'out-UAC', and 'in-UAC' shares magic with 'out-UAS'. However, if (as you state above), 4) follows immediately after 1), it's no longer a B2BUA, but a plain old UA, IMHO. 

Cheers,
Aki
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Subject: RE: [Sip] B2BUA
To: "Dean Willis" <dean.willis@softarmor.com>
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Dean,

I fully agree with you. I've never said that the definition in 3261 is
wrong, just that it could be completed (or improved, etc.) to avoid
confusion.
After reading the mails in this discusion thread, what could be added is
that:
1) A B2BUA handles two or more UAs (each one behaving as UAC/UAS, as
described in 3261 for UAs)
2) The SIP protocol specification does not mandate for any behaviour of
such B2BUA, i.e. on how to "link" (or "concatenate") two or more UAs.

As a consequence, you cannot standardise the behaviour of a B2BUA, since a
B2BUA is an application "and that can  do anything that one can imagine an
application doing".
But you can still standardise an application or a class of applications
that behave as B2BUA. These applications exhibit a UA behaviour for each
concerned UA, according to SIP rules defined in 3261. But this is yet
another story.

Best regards
Juan Carlos

P.S.: The "something BETWEEN the B2BUA and  the conformant proxy" story
should be handled in a separate discusion thread, to avoid to mix all
problems in the same thread.





"Dean Willis" <dean.willis@softarmor.com>@ietf.org on 03/12/2002 07:00:19

Sent by:  sip-admin@ietf.org


To:   "'Mary Barnes'" <mbarnes@nortelnetworks.com>, Juan-Carlos
      ROJAS/FR/ALCATEL@ALCATEL
cc:   <sip@ietf.org>
Subject:  RE: [Sip] B2BUA



Mary said:
 -----
 So, I guess you're  suggesting that the term "concatenation" is not
accurate or is misleading and I  guess that depends upon how loosely one
applies the term concatenation here; my  assumption had been that it meant
that they were "linked" (consistent with  Webster's definition of
concatenate) in some manner which is unprescribed, so I  think your point 2
below isn't a problem with the current definition.  My  understanding of
the current definition is that there isn't anything more for  the SIP
protocol to prescribe about the operation of a B2BUA other than what is
required from a protocol perspective to function as a UAC and a UAS. From a
protocol perspective, it is a logical entity and thus whether there's one
or  multiple doesn't seem to apply.  I think what you (and perhaps others)
are  suggesting is that there's a definition of a B2BUA that more closely
parallels  the definition of a Proxy Server since in many cases the
physical rendition of  the logical B2BUA functionality would be replacing a
Proxy Server in a network.  And I think this is where the problem begins,
as many are opposed to this model  (as it increases the complexity, etc.)
and this is why the logical definition  has remained.
------

Well,  I think the argument is being made that this definition doesn't (for
some reason  that I seem incapable of comprehending) work for some people,
and that therefore  we should clarify it. At the fundamental level, I think
some people do not find  the implications of this definition obvious. There
are several ways we might  address this. We might add examples. We might
add "In other words, the only  relationship between the UAC and the UAS is
the application logic, and that can  do anything that one can imagine an
application doing", or we might just use  better wording.

I  don't think the argument is being made that the definition in 3261 is
wrong,  just that we can make it better for those who don't get it the way
it is  written. Of course, if we discuss it here long enough, maybe most of
them WILL  get it -- but that doesn't help the ones who aren't "here"  yet.

I  think that Bill is suggesting that there is also something BETWEEN the
B2BUA and  the conformant proxy, and that the world would be a better place
(ok, maybe not  THAT much better) if our taxonomy also included this
"between"  thing.

--
Dean




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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'aki.niemi@nokia.com'" <aki.niemi@nokia.com>
Cc: sip@ietf.org
Subject: RE: [Sip] B2BUA
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Of course these things are never so simple.
On an "end system mixed" conference, an INVITE
from a new participant triggers Re-INVITES to
existing participants.  Depending on details,
a focus could hold the response from the INVITE
of the new participant until it receives responses
from all existing participants (new CODEC negotiation
for example).  That meets your definition of B2BUA.

On the other hand, a centrally mixed conference
would not normally trigger Re-INVITES.  Does that
mean that an end-system mixed focus is a B2BUA
but a centrally mixed focus is not?  Hard to
differentiate.  With both of them, a state
change, signaled by, say a NOTIFY from one to
the focus triggers NOTIFIES to (possibly a
subset of) the other participants, so that
would meet the definition.  Would you call
that a change in roles?

Brian

> -----Original Message-----
> From: aki.niemi@nokia.com [mailto:aki.niemi@nokia.com]
> Sent: Tuesday, December 03, 2002 12:13 PM
> To: Brian.Rosen@marconi.com
> Cc: sip@ietf.org
> Subject: RE: [Sip] B2BUA
> 
> 
> Inline.
> 
> > The key thing about a B2BUA is that it looks like a UA to whatever
> > it is connected to (on both sides).  So, a conference server
> > looks like a UA to each of it's ports, and it has at least two of
> > them, and there is some "magic" between them.  If you look at the
> > concept of the "focus" from the conferencing work, it is a
> > thing that has a bunch of two ended SIP calls, one with each
> > participant.  Each participant thinks it is talking to a UA.
> > While some messages sent from a participant to a UA might be
> > consumed by the UA (state changes), often, a message will trigger
> > messages from the focus to other participants.
> 
> This just generalized the B2BUA to cover any normal UA 
> behavior. Example, I get a MESSAGE from a friend to my UA, 
> based on which decide to call another friend by sending an 
> INVITE from that UA. This I think is quite normal UA 
> operation, the "magic" being me via my UI.
> 
> > This seems, to me, to meet the definition of B2BUA in 3261,
> > and more strongly meets proposed changes in the definition.
> > Of course, it really doesn't matter if you consider the focus to
> > be a B2BUA or not.
> 
> Hmm... Let's think of the B2BUA as 'in-UA' concatenated with 
> 'out-UA'. By definition:
> 
> 1) in-UAS receives request,
>    <Magic happens>
> 2) out-UAC sends request'
> 3) out-UAC receives response'
>    <Magic happens>
> 4) in-UAS sends response
> 
> (The same process happens with requests going from 'out' to 'in')
> 
> Within this B2BUA, 'in-UAS' shares magic with 'out-UAC', and 
> 'in-UAC' shares magic with 'out-UAS'. However, if (as you 
> state above), 4) follows immediately after 1), it's no longer 
> a B2BUA, but a plain old UA, IMHO. 
> 
> Cheers,
> Aki
> 
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Subject: RE: [Sip] B2BUA
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Brian,

inline.

> Of course these things are never so simple.
> On an "end system mixed" conference, an INVITE
> from a new participant triggers Re-INVITES to
> existing participants.  Depending on details,
> a focus could hold the response from the INVITE
> of the new participant until it receives responses
> from all existing participants (new CODEC negotiation
> for example).  That meets your definition of B2BUA.
> 
> On the other hand, a centrally mixed conference
> would not normally trigger Re-INVITES.  Does that
> mean that an end-system mixed focus is a B2BUA
> but a centrally mixed focus is not?  Hard to
> differentiate.  With both of them, a state
> change, signaled by, say a NOTIFY from one to
> the focus triggers NOTIFIES to (possibly a
> subset of) the other participants, so that
> would meet the definition.  Would you call
> that a change in roles?

I agree, a conference server could assume both UA and B2BUA roles, as well as proxy and redirect server roles depending on the system, application, request, and so on.  But whether a focus is a UA, a B2BUA, or both is somewhat besides the point.  The roles need to be defined so that B2BUA behavior is distinct from UA behavior.  The end-point can then use one that best fits its purpose.

Cheers,
Aki

> 
> Brian
> 
> > -----Original Message-----
> > From: aki.niemi@nokia.com [mailto:aki.niemi@nokia.com]
> > Sent: Tuesday, December 03, 2002 12:13 PM
> > To: Brian.Rosen@marconi.com
> > Cc: sip@ietf.org
> > Subject: RE: [Sip] B2BUA
> > 
> > 
> > Inline.
> > 
> > > The key thing about a B2BUA is that it looks like a UA to whatever
> > > it is connected to (on both sides).  So, a conference server
> > > looks like a UA to each of it's ports, and it has at least two of
> > > them, and there is some "magic" between them.  If you look at the
> > > concept of the "focus" from the conferencing work, it is a
> > > thing that has a bunch of two ended SIP calls, one with each
> > > participant.  Each participant thinks it is talking to a UA.
> > > While some messages sent from a participant to a UA might be
> > > consumed by the UA (state changes), often, a message will trigger
> > > messages from the focus to other participants.
> > 
> > This just generalized the B2BUA to cover any normal UA 
> > behavior. Example, I get a MESSAGE from a friend to my UA, 
> > based on which decide to call another friend by sending an 
> > INVITE from that UA. This I think is quite normal UA 
> > operation, the "magic" being me via my UI.
> > 
> > > This seems, to me, to meet the definition of B2BUA in 3261,
> > > and more strongly meets proposed changes in the definition.
> > > Of course, it really doesn't matter if you consider the focus to
> > > be a B2BUA or not.
> > 
> > Hmm... Let's think of the B2BUA as 'in-UA' concatenated with 
> > 'out-UA'. By definition:
> > 
> > 1) in-UAS receives request,
> >    <Magic happens>
> > 2) out-UAC sends request'
> > 3) out-UAC receives response'
> >    <Magic happens>
> > 4) in-UAS sends response
> > 
> > (The same process happens with requests going from 'out' to 'in')
> > 
> > Within this B2BUA, 'in-UAS' shares magic with 'out-UAC', and 
> > 'in-UAC' shares magic with 'out-UAS'. However, if (as you 
> > state above), 4) follows immediately after 1), it's no longer 
> > a B2BUA, but a plain old UA, IMHO. 
> > 
> > Cheers,
> > Aki
> > 
> 
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From mailnull@www1.ietf.org  Tue Dec  3 15:54:55 2002
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Subject: [Sip] SIP servers interconnection
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Hello,

I have a question about SIP Servers (Agents)
interconnection question.  It would be appreciated if
you could give me some point or suggestions.

The question is, if a service provider1(SP1) has a SIP
server in its network and it wants to connect to its
partner service provider2(SP2)'s SIP server so that
the user can roam between these 2 SPs, what kind of
protocol or solution will be applied between these 2
SIP servers to communicate so that the user(who is
registed in SP1) can use SP1's SIP server when he/she
is in SP1 service coverage and use SP2's SIP server
when in SP2 service coverage? What if these 2 SIP
servers are using different databases (e.g. oracle and
infomix)?

Thanks a lot and I look forward to your help!

B.
 

______________________________________________________________________ 
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One issue that the recent B2BUA discussion has raised in my mind is what a
B2BUA (whatever its decided that should be) should do with Max-Forwards.

This may have been discussed before, but while RFC 3261 says a B2BUA is a
back to back UAS / UAC, and the coupling is not specified in the RFC,
perhaps it ought to say that the max-forwards parameter MUST be propagated
from the incoming request to the outgoing request using the same rules as a
regular proxy.  This would seem helpful for the avoidance of loops.

Any comments?

Pete.



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> This thread is, hopefully reaching an end (with an agreement to
> do some examples with call flows).  However, the above statement
> is yet one more example of the confusion people have with B2BUA.


A group of us have been contemplating writing an informational
draft in sipping that would provide examples with call flows among other 
things.
We floated this idea with several people at the recent IETF meeting.

In particular, while recognizing that little can be said about the 
behavior of the most
general form of B2BUA, it seems possible to classify different kinds 
(sub-classes)
of B2BUAs and then take a stab at specifying "suggested" behavior for 
these sub-classes
by means of call flows. To be clear, we recognize that any 
"specification" of behavior
of any sort of B2BUA would not be normative and would perhaps ultimately
take the form of a BCP.

Thinking to start things simply, we considered looking at the case of a
"transparent" B2BUA with a UAC on one side and a UAS on the other. While 
not
functionally interesting, this case would provide a baseline of
behavior to serve as a launching point for more interesting B2BUAs types.
Following an examination of the "transparent" B2BUA, we would hopefully 
move
on to looking at a B2BUA that serves as a firewall gateway.

Naturally, my colleagues and I welcome feedback on this proposal.

I don't know if it's appropriate to move any discussion of this proposal 
to the
sipping list since it seems to belong there. I'm happy to follow the 
group consensus
regarding where the discussion should be held.


Hal Purdy


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Pete Cordell wrote:
> One issue that the recent B2BUA discussion has raised in my mind is what a
> B2BUA (whatever its decided that should be) should do with Max-Forwards.
> 
> This may have been discussed before, but while RFC 3261 says a B2BUA is a
> back to back UAS / UAC, and the coupling is not specified in the RFC,
> perhaps it ought to say that the max-forwards parameter MUST be propagated
> from the incoming request to the outgoing request using the same rules 
> as a regular proxy.  This would seem helpful for the avoidance of loops.

I have a feeling that specialized handling of headers for B2BUAs is yet
another nightmare in the making.

A B2BUA is a UA, period.  So long as the request made it to one half
of the B2BUA before Max-Forwards reached 0, the protocol constraints
on that request have been satisfied.  What the other half of the
B2BUA wants to do with the Max-Forwards header is entirely up to it.

Regards,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216

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inline [RRR]

-----Original Message-----
From: Pete Cordell [mailto:pete@tech-know-ware.com]
Sent: Tuesday, December 03, 2002 12:44 PM
To: sip@ietf.org
Subject: [Sip] B2BUA & Max-Forwards


One issue that the recent B2BUA discussion has raised in my mind is what a
B2BUA (whatever its decided that should be) should do with Max-Forwards.

This may have been discussed before, but while RFC 3261 says a B2BUA is a
back to back UAS / UAC, and the coupling is not specified in the RFC,
perhaps it ought to say that the max-forwards parameter MUST be propagated
from the incoming request to the outgoing request using the same rules as a
regular proxy.  This would seem helpful for the avoidance of loops.

[RRR] It depends. Proxy MUST abide by its own rules as specified in RFC 3261. Similarly, UAs also MUST need to obey all SIP rules. These two (i.e., Proxy and B2BUA) act independently as per their SIP rules. The question then comes: How do people build services/applications that have the dependency between B2BUA and proxy function? SIP does not define these rules. That is why, people are coming to the conclusion let there should an informational RFC in defining B2BUA functionalities along with call flows (as Brain and others pointed out).

Any comments?

Pete.

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Subject: Re: [Sip] B2BUA & Max-Forwards
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<html><div style='background-color:'><DIV>
<P>Vijay,</P>
<P>&nbsp;&nbsp;&nbsp;&nbsp; You are right - </P>
<P>&nbsp;&nbsp;&nbsp; A B2BUA needs to propagate some of the headers received in a request/response to the other endpoint.</P>
<P>&nbsp;&nbsp;&nbsp; Example: Require, Expires to name a few. This is because these headers actually are relevant end to end. </P>
<P>&nbsp;&nbsp; However, some of the other headers like Contact should be set by the B2BUA as itself. &nbsp;</P>
<P>&nbsp;&nbsp;&nbsp;Padma <BR><BR></P></DIV>
<DIV></DIV>
<DIV></DIV>&gt;From: "Vijay K. Gurbani" <VKG@LUCENT.COM>
<DIV></DIV>&gt;To: Pete Cordell <PETE@TECH-KNOW-WARE.COM>
<DIV></DIV>&gt;CC: sip@ietf.org 
<DIV></DIV>&gt;Subject: Re: [Sip] B2BUA &amp; Max-Forwards 
<DIV></DIV>&gt;Date: Tue, 03 Dec 2002 15:11:37 -0600 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;Pete Cordell wrote: 
<DIV></DIV>&gt;&gt;One issue that the recent B2BUA discussion has raised in my mind is 
<DIV></DIV>&gt;&gt;what a 
<DIV></DIV>&gt;&gt;B2BUA (whatever its decided that should be) should do with 
<DIV></DIV>&gt;&gt;Max-Forwards. 
<DIV></DIV>&gt;&gt; 
<DIV></DIV>&gt;&gt;This may have been discussed before, but while RFC 3261 says a 
<DIV></DIV>&gt;&gt;B2BUA is a 
<DIV></DIV>&gt;&gt;back to back UAS / UAC, and the coupling is not specified in the 
<DIV></DIV>&gt;&gt;RFC, 
<DIV></DIV>&gt;&gt;perhaps it ought to say that the max-forwards parameter MUST be 
<DIV></DIV>&gt;&gt;propagated 
<DIV></DIV>&gt;&gt;from the incoming request to the outgoing request using the same 
<DIV></DIV>&gt;&gt;rules as a regular proxy. This would seem helpful for the 
<DIV></DIV>&gt;&gt;avoidance of loops. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;I have a feeling that specialized handling of headers for B2BUAs is 
<DIV></DIV>&gt;yet 
<DIV></DIV>&gt;another nightmare in the making. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;A B2BUA is a UA, period. So long as the request made it to one half 
<DIV></DIV>&gt;of the B2BUA before Max-Forwards reached 0, the protocol constraints 
<DIV></DIV>&gt;on that request have been satisfied. What the other half of the 
<DIV></DIV>&gt;B2BUA wants to do with the Max-Forwards header is entirely up to it. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;Regards, 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;- vijay 
<DIV></DIV>&gt;-- 
<DIV></DIV>&gt;Vijay K. Gurbani vkg@{lucent.com,research.bell-labs.com,acm.org} 
<DIV></DIV>&gt;Wireless Networks Group/Internet Software and Services 
<DIV></DIV>&gt;Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 
<DIV></DIV>&gt;6G-440 
<DIV></DIV>&gt;Naperville, Illinois 60566 Voice: +1 630 224 0216 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;_______________________________________________ 
<DIV></DIV>&gt;Sip mailing list https://www1.ietf.org/mailman/listinfo/sip 
<DIV></DIV>&gt;This list is for NEW development of the core SIP Protocol 
<DIV></DIV>&gt;Use sip-implementors@cs.columbia.edu for questions on current sip 
<DIV></DIV>&gt;Use sipping@ietf.org for new developments on the application of sip 
<DIV></DIV></div><br clear=all><hr>Add photos to your e-mail with <a href="http://g.msn.com/8HMSEN/2022">MSN 8.</a> Get 2 months FREE*.</html>
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To: "Vijay K. Gurbani" <vkg@lucent.com>
Cc: Pete Cordell <pete@tech-know-ware.com>, sip@ietf.org, sip-admin@ietf.org
Subject: Re: [Sip] B2BUA & Max-Forwards
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I've been following two threads on the B'to'BUA definition and looks like 
both are pretty similar. So Here are my 2c:

* First, hope I am not offending someone by using BtoBUA instead of B2BUA. 
Beleive it or not, this '2' also is another source
of confusion for many people who seem to think '2' means the number two 
and implies '1 UA and 1 UAS' and therefore should
be really called BnBUA :-)

* How a  BtoBUA functions depends on what application is being implemented 
by it. Therefore, it does not make sense
to me trying to dictate or suggest how the BtoB associates one or more UAS 
instances with one or more UAC instances.

Therefore, a BtoB can only be defined as how its seen from the 'outside' 
-> i.e. as Vijay points out, 

        <quote>
        A B2BUA is a UA, period.  So long as the request made it to one 
half
        of the B2BUA before Max-Forwards reached 0, the protocol 
constraints
        on that request have been satisfied.  What the other half of the
        B2BUA wants to do with the Max-Forwards header is entirely up to 
it.
        </quote>

* How the BtoB 'associates' one or more UAS instances with one or more UAC 
instances is dependent on an algorithm
which MAY associate using defined SIP protocol semantics or some other 
application level semantic which cannot be standardized.

* 'transparent' , 'semi-transparent', 'evil' <and other such terms> 
BtoBUAs reflect different algorithms that a BtoB
applies to a message after a inbound SIP message is received and before an 
associated SIP message is sent out.

* Therefore, trying to define what a BtoBUA should do with a message 
without defining what is the application that is using this
model seems like it will simply add to the confusion instead of trying to 
solve it.

In summary, as suggested by some  others, the only way to 'describe' what 
a BtoB may do, makes sense only if we define
call-flows with specific applications in mind. Trying to generalize an 
application-specific entity is going to confuse implementors
much more. 

In addition, these call flows should describe how one can achieve certain 
end services (specific)  instead of trying to project the message 
that this is how a BtoB should really work (generic). 

Atleast now implementors know they can do what they want with the B2B 
internally as long as external interfaces are standard.
While this may seem like sacrilege, it makes a lot of sense to keep it 
this way . Opinions on what should or should not be done
really depends on a particular scenario and deployment case (as proven 
time and again by 3G CSCFs as well as FCPs, media proxies et al).


regds
arjun




--
Arjun Roychowdhury @ Hughes Software Systems
11717 Exploration Lane, Germantown MD 20876
(O): 301-212-7860        (M): 240-997-0066




"Vijay K. Gurbani" <vkg@lucent.com>
Sent by: sip-admin@ietf.org
12/03/2002 04:11 PM

 
        To:     Pete Cordell <pete@tech-know-ware.com>
        cc:     sip@ietf.org
        Subject:        Re: [Sip] B2BUA & Max-Forwards


Pete Cordell wrote:
> One issue that the recent B2BUA discussion has raised in my mind is what 
a
> B2BUA (whatever its decided that should be) should do with Max-Forwards.
> 
> This may have been discussed before, but while RFC 3261 says a B2BUA is 
a
> back to back UAS / UAC, and the coupling is not specified in the RFC,
> perhaps it ought to say that the max-forwards parameter MUST be 
propagated
> from the incoming request to the outgoing request using the same rules 
> as a regular proxy.  This would seem helpful for the avoidance of loops.

I have a feeling that specialized handling of headers for B2BUAs is yet
another nightmare in the making.

A B2BUA is a UA, period.  So long as the request made it to one half
of the B2BUA before Max-Forwards reached 0, the protocol constraints
on that request have been satisfied.  What the other half of the
B2BUA wants to do with the Max-Forwards header is entirely up to it.

Regards,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216

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Subject: RE: [Sip] B2BUA & Max-Forwards
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-----Original Message-----
From: Padma Suresh [mailto:padma_suresh@hotmail.com]
 
> A B2BUA needs to propagate some of the headers received in a
> request/response to the other endpoint. Example: Require...

No. As emphatically as possible: NO. This is completely wrong,
and it appears to be a growing -- and dangerous -- misconception.

A B2BUA cannot transparently pass Require through. It must
understand and invoke the extension indicated.

  "The Require header field contains a list of option tags, described in
   Section 19.2.  Each option tag defines a SIP extension that MUST be
   understood to process the request."

A back to back user agent *is* a UAS, and *is* bound by this
provision.

Whether the UAC portion then turns around and imposes the same
requirement on the UAS it contacts is a local decision.

/a
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From: "Cullen Jennings" <fluffy@cisco.com>
To: "Hal Purdy" <khp@research.att.com>, "SIP Mailng List" <sip@ietf.org>
Subject: RE: [Sip] Re: B2BUA draft proposal
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Sounds good - I look forward to seeing it - and since this will not change
the behavior of SIP but just show some ways to use it, I think the sipping
list would be the place to discuss it.

Cullen


> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Hal
> Purdy
> Sent: Tuesday, December 03, 2002 12:45 PM
> To: SIP Mailng List
> Subject: [Sip] Re: B2BUA draft proposal
>
>
> > This thread is, hopefully reaching an end (with an agreement to
> > do some examples with call flows).  However, the above statement
> > is yet one more example of the confusion people have with B2BUA.
>
>
> A group of us have been contemplating writing an informational
> draft in sipping that would provide examples with call flows among other
> things.
> We floated this idea with several people at the recent IETF meeting.
>
> In particular, while recognizing that little can be said about the
> behavior of the most
> general form of B2BUA, it seems possible to classify different kinds
> (sub-classes)
> of B2BUAs and then take a stab at specifying "suggested" behavior for
> these sub-classes
> by means of call flows. To be clear, we recognize that any
> "specification" of behavior
> of any sort of B2BUA would not be normative and would perhaps ultimately
> take the form of a BCP.
>
> Thinking to start things simply, we considered looking at the case of a
> "transparent" B2BUA with a UAC on one side and a UAS on the other. While
> not
> functionally interesting, this case would provide a baseline of
> behavior to serve as a launching point for more interesting B2BUAs types.
> Following an examination of the "transparent" B2BUA, we would hopefully
> move
> on to looking at a B2BUA that serves as a firewall gateway.
>
> Naturally, my colleagues and I welcome feedback on this proposal.
>
> I don't know if it's appropriate to move any discussion of this proposal
> to the
> sipping list since it seems to belong there. I'm happy to follow the
> group consensus
> regarding where the discussion should be held.
>
>
> Hal Purdy
>
>
> _______________________________________________
> Sip mailing list  https://www1.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: "Cullen Jennings" <fluffy@cisco.com>
To: "Pete Cordell" <pete@tech-know-ware.com>, <sip@ietf.org>
Subject: RE: [Sip] B2BUA & Max-Forwards
Date: Tue, 3 Dec 2002 18:44:06 -0800
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A related question to this was how to stop routing loops that included some
PSTN hops. One solution to this was to make sure that as a call was passed
from SIP to SS7 there was a mapping of the number of hops. A trivial 1:1
mapping would be unlikely to work.

I really don't really care that much if something like this is documented or
not but I imagine that some devices will do it to stop routing loops that
eat infinite amounts of resources.

Cullen


> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Pete
> Cordell
> Sent: Tuesday, December 03, 2002 9:44 AM
> To: sip@ietf.org
> Subject: [Sip] B2BUA & Max-Forwards
>
>
> One issue that the recent B2BUA discussion has raised in my mind is what a
> B2BUA (whatever its decided that should be) should do with Max-Forwards.
>
> This may have been discussed before, but while RFC 3261 says a B2BUA is a
> back to back UAS / UAC, and the coupling is not specified in the RFC,
> perhaps it ought to say that the max-forwards parameter MUST be propagated
> from the incoming request to the outgoing request using the same
> rules as a
> regular proxy.  This would seem helpful for the avoidance of loops.
>
> Any comments?
>
> Pete.
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.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] Re: B2BUA draft proposal
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Hello,

"In particular, while recognizing that little can be said about the
behavior of the most general form of B2BUA, it seems possible to classify
different kinds (sub-classes)"
Good idea to give examples of implementing an application using a B2BUA,
this can illustrate what could be done with a B2BUA (even if is very likely
that this will generate a lot of debate).
But please *do not* classify B2BUA, because there will always be a new
application not fitting in your classification.

Best regards
Juan Carlos





Hal Purdy <khp@research.att.com>@ietf.org on 03/12/2002 21:45:21

Sent by:  sip-admin@ietf.org


To:   SIP Mailng List <sip@ietf.org>
cc:
Subject:  [Sip] Re: B2BUA draft proposal


> This thread is, hopefully reaching an end (with an agreement to
> do some examples with call flows).  However, the above statement
> is yet one more example of the confusion people have with B2BUA.


A group of us have been contemplating writing an informational
draft in sipping that would provide examples with call flows among other
things.
We floated this idea with several people at the recent IETF meeting.

In particular, while recognizing that little can be said about the
behavior of the most
general form of B2BUA, it seems possible to classify different kinds
(sub-classes)
of B2BUAs and then take a stab at specifying "suggested" behavior for
these sub-classes
by means of call flows. To be clear, we recognize that any
"specification" of behavior
of any sort of B2BUA would not be normative and would perhaps ultimately
take the form of a BCP.

Thinking to start things simply, we considered looking at the case of a
"transparent" B2BUA with a UAC on one side and a UAS on the other. While
not
functionally interesting, this case would provide a baseline of
behavior to serve as a launching point for more interesting B2BUAs types.
Following an examination of the "transparent" B2BUA, we would hopefully
move
on to looking at a B2BUA that serves as a firewall gateway.

Naturally, my colleagues and I welcome feedback on this proposal.

I don't know if it's appropriate to move any discussion of this proposal
to the
sipping list since it seems to belong there. I'm happy to follow the
group consensus
regarding where the discussion should be held.


Hal Purdy


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Subject: RE: [Sip] B2BUA & Max-Forwards
To: Adam Roach <adam@dynamicsoft.com>
Cc: "'Padma Suresh'" <padma_suresh@hotmail.com>, vkg@lucent.com,
        pete@tech-know-ware.com, sip@ietf.org
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Hello,

As it was already stated in this discusion thread, a B2BUA is an
*application* linking several UAs
As such, the behaviour of a B2BUA cannot be standardised.

Therefore, one cannot say that "A B2BUA needs to propagate some of the
headers received".
However, you can say that the application "x" behaves as a B2BUA, and that
the application "x" "needs to propagate" the a-header or the b-header etc.
for the needs of the application. This is the choice of the application,
that's all.

Best regards
Juan Carlos





Adam Roach <adam@dynamicsoft.com>@ietf.org on 04/12/2002 01:36:50

Sent by:  sip-admin@ietf.org


To:   "'Padma Suresh'" <padma_suresh@hotmail.com>, vkg@lucent.com,
      pete@tech-know-ware.com
cc:   sip@ietf.org
Subject:  RE: [Sip] B2BUA & Max-Forwards


-----Original Message-----
From: Padma Suresh [mailto:padma_suresh@hotmail.com]

> A B2BUA needs to propagate some of the headers received in a
> request/response to the other endpoint. Example: Require...

No. As emphatically as possible: NO. This is completely wrong,
and it appears to be a growing -- and dangerous -- misconception.

A B2BUA cannot transparently pass Require through. It must
understand and invoke the extension indicated.

  "The Require header field contains a list of option tags, described in
   Section 19.2.  Each option tag defines a SIP extension that MUST be
   understood to process the request."

A back to back user agent *is* a UAS, and *is* bound by this
provision.

Whether the UAC portion then turns around and imposes the same
requirement on the UAS it contacts is a local decision.

/a
_______________________________________________
Sip mailing list  https://www1.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] B2BUA & Max-Forwards
To: "Pete Cordell" <pete@tech-know-ware.com>
Cc: <sip@ietf.org>
Date: Wed, 4 Dec 2002 09:01:56 +0100
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Hello,

The following text comes from RFC 3261, page 39
"A UAC MUST insert a Max-Forwards header field into each request it
originates with a value that SHOULD be 70"

Now, a B2BUA handles several UAs.
For a given UA, when behaving as UAC, the Max-Forwards header has to be
*initialised* by the UA. The RFC3261 recommends to use the value 70, but
this is not mandatory (to fixe the initial value of Max-Forwards, RFC3261
uses a SHOULD and not a MUST).

Therefore, a B2BUA acting as an application can freely choose for a UAC
which it is handling:
- Initialise Max-Forwards to 70, as recommended by RFC3261 for a UAC
- Initialise Max-Forwards with the value possibly received in a UA acting
as UAS.
- Initialise Max-Forwards to any other value, according to the application
definition

Once again, this is a question of application defintion.

Best regards
Juan Carlos





"Pete Cordell" <pete@tech-know-ware.com>@ietf.org on 03/12/2002 18:44:21

Sent by:  sip-admin@ietf.org


To:   <sip@ietf.org>
cc:
Subject:  [Sip] B2BUA & Max-Forwards


One issue that the recent B2BUA discussion has raised in my mind is what a
B2BUA (whatever its decided that should be) should do with Max-Forwards.

This may have been discussed before, but while RFC 3261 says a B2BUA is a
back to back UAS / UAC, and the coupling is not specified in the RFC,
perhaps it ought to say that the max-forwards parameter MUST be propagated
from the incoming request to the outgoing request using the same rules as a
regular proxy.  This would seem helpful for the avoidance of loops.

Any comments?

Pete.



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This list is for NEW development of the core SIP Protocol
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From: "Ravi Shiroor" <rshiroor@aplion.stpn.soft.net>
To: <sip@ietf.org>
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Subject: [Sip] response to merged requests
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posting this on sip as there were no responses on sip-implementors.

--------->

section 8.2.2.2 in RFC 3261 says that when a UAS receives
merged requests, it should respond to the INVITEs receieved
after the first one (those detected as merged) with 482 (loop detected).

i was wondering why "loop detected" is being being used when
merged requests are receieved? wouldnt it be appropriate to
define a specific 4xx response for merged requests?

can somebody please clarify if there is a valid reason for
using "loop detected" in this scenario?

TIA,
ravi.

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Subject: RE: [Sip] B2BUA & Max-Forwards
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Now your back to restricting what I can and can't do in an intermediary. It
can't be a B2BUA 'cos it doesn't need to honour require headers (the UA at
the other end does that, if they were meant for my intermediary then they
would be proxy-require?), and it can't be a proxy 'cos of the rules about
not changing the body so what is it?

It's not sanctioned by the RFC, yet id does an important and useful job.

-----Original Message-----
From: Adam Roach [mailto:adam@dynamicsoft.com] 
Sent: 04 December 2002 00:37
To: 'Padma Suresh'; vkg@lucent.com; pete@tech-know-ware.com
Cc: sip@ietf.org
Subject: RE: [Sip] B2BUA & Max-Forwards


-----Original Message-----
From: Padma Suresh [mailto:padma_suresh@hotmail.com]
 
> A B2BUA needs to propagate some of the headers received in a 
> request/response to the other endpoint. Example: Require...

No. As emphatically as possible: NO. This is completely wrong, and it
appears to be a growing -- and dangerous -- misconception.

A B2BUA cannot transparently pass Require through. It must understand and
invoke the extension indicated.

  "The Require header field contains a list of option tags, described in
   Section 19.2.  Each option tag defines a SIP extension that MUST be
   understood to process the request."

A back to back user agent *is* a UAS, and *is* bound by this provision.

Whether the UAC portion then turns around and imposes the same requirement
on the UAS it contacts is a local decision.

/a
_______________________________________________
Sip mailing list  https://www1.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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>
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version 5.5.2653.12"=
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<TITLE>RE: [Sip] B2BUA &amp; Max-Forwards</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Now your back to restricting what I can and can't do in a=
n intermediary. It can't be a B2BUA 'cos it doesn't need to honour require =
headers (the UA at the other end does that, if they were meant for my inter=
mediary then they would be proxy-require?), and it can't be a proxy 'cos of=
 the rules about not changing the body so what is it?</FONT></P>

<P><FONT SIZE=3D2>It's not sanctioned by the RFC, yet id does an important =
and useful job.</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Adam Roach [<A HREF=3D"mailto:adam@dynamicsoft.com=
">mailto:adam@dynamicsoft.com</A>] </FONT>
<BR><FONT SIZE=3D2>Sent: 04 December 2002 00:37</FONT>
<BR><FONT SIZE=3D2>To: 'Padma Suresh'; vkg@lucent.com; pete@tech-know-ware.=
com</FONT>
<BR><FONT SIZE=3D2>Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: RE: [Sip] B2BUA &amp; Max-Forwards</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Padma Suresh [<A HREF=3D"mailto:padma_suresh@hotma=
il.com">mailto:padma_suresh@hotmail.com</A>]</FONT>
<BR><FONT SIZE=3D2>&nbsp;</FONT>
<BR><FONT SIZE=3D2>&gt; A B2BUA needs to propagate some of the headers rece=
ived in a </FONT>
<BR><FONT SIZE=3D2>&gt; request/response to the other endpoint. Example: Re=
quire...</FONT>
</P>

<P><FONT SIZE=3D2>No. As emphatically as possible: NO. This is completely w=
rong, and it appears to be a growing -- and dangerous -- misconception.</FO=
NT></P>

<P><FONT SIZE=3D2>A B2BUA cannot transparently pass Require through. It mus=
t understand and invoke the extension indicated.</FONT>
</P>

<P><FONT SIZE=3D2>&nbsp; &quot;The Require header field contains a list of =
option tags, described in</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; Section 19.2.&nbsp; Each option tag defines=
 a SIP extension that MUST be</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; understood to process the request.&quot;</F=
ONT>
</P>

<P><FONT SIZE=3D2>A back to back user agent *is* a UAS, and *is* bound by t=
his provision.</FONT>
</P>

<P><FONT SIZE=3D2>Whether the UAC portion then turns around and imposes the=
 same requirement on the UAS it contacts is a local decision.</FONT>
</P>

<P><FONT SIZE=3D2>/a</FONT>
<BR><FONT SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>Sip mailing list&nbsp; <A HREF=3D"https://www1.ietf.org/=
mailman/listinfo/sip" TARGET=3D"_blank">https://www1.ietf.org/mailman/listi=
nfo/sip</A></FONT>
<BR><FONT SIZE=3D2>This list is for NEW development of the core SIP Protoco=
l</FONT>
<BR><FONT SIZE=3D2>Use sip-implementors@cs.columbia.edu for questions on cu=
rrent sip Use sipping@ietf.org for new developments on the application of s=
ip</FONT></P>

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Subject: Re: [Sip] Constructing the branch parameter at stateless proxies
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Hi Robert,
     Here's one issue i could think of:
Insertion of branch parameter in Via header has been
mandatory for proxies all along. However, for UA's it
became mandatory only from bis-06 onwards. So if we
have a pre-bis-06 UA sending a request to a stateless
proxy (i.e. first hop is a SLP), the proxy would end
up forwarding it without a branch parameter. Ofcourse,
things would still work fine at the next stateful hop.
But, by not inserting a branch parameter, the SLP would
not be compliant to the spec. (Proxies MUST insert a
branch).

     This would happen only if a SLP is at the first
hop from such a UA. Thereafter, even if the request
loops or spirals through, the SLP would end up adding
different branches by following your approach. So we
could special-case this with a simpler logic so that
if the SLP doesn't see a branch in the topmost Via,
it could insert a simple one (concatenating the
Tags-CallId-CseqNum for example).


Subhash Nayak
Hughes Software Systems
http://www.hssworld.com







Robert Sparks <rsparks@dynamicsoft.com> on 12/03/2002 10:05:01 PM

To:   sip@ietf.org
cc:    (bcc: Subhash Ullal Nayak/HSSBLR)

Subject:  [Sip] Constructing the branch parameter at stateless proxies




This list has seen a couple of requests to clarify the language in
RFC 3261 concerning how a stateless proxy creates a branch identifier.

The current text requires uniqueness, dependency on route decision
affecting parameters and care to be sure that those parameters are
retransmission invariant. It goes on to recommend a hash over a
set of fields (this set has been the subject of the requests for
clarification).

I think there's a much simpler solution. The purpose of this
branch parameter is to act as a transaction identifier. Who
cares about this identifier? Only the stateful elements at
each end of the transaction. Any stateless proxies between
these elements don't use that identifier in any of their
decisions, so I assert that it is not useful to change it
as it crosses the stateless proxy.

I propose, then, that a stateless proxy simply copy
the branch parameter from the topmost Via header field
value it receives into the Via header field value it adds
to the request.

I don't believe it matters that this parameter contain the
magic cookie. If it doesn't, the next stateful hop will use
the 2543-style transaction identifiers provided by the sending
stateful element. The stateless elements are by definition
transaction transparent.

So - can anyone construct a scenario where this doesn't
function correctly? If not, we can remove several convoluted
paragraphs from the spec and replace them with a very simple
couple of sentences.

RjS



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To: "Christian Huitema" <huitema@windows.microsoft.com>,
        "Roy, Radhika R, ALASO" <rrroy@att.com>, <mbarnes@nortelnetworks.com>
From: Jiri Kuthan <jiri.kuthan@fokus.fraunhofer.de>
Subject: RE: [Sip] B2BUA
Cc: <dean.willis@softarmor.com>, <Juan-Carlos.Rojas@alcatel.fr>,
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At 05:35 PM 12/3/2002, Christian Huitema wrote:
>> > [MB]: Some of this, of course, will lead us back to the
>> > list of the reasons why some folks consider these things "evil", ...
>> 
>> I do not remember whether anyone has ever said that B2BUA is evil
>> things. What I remember is that the dumping of the B2BUA functionality
>> in the name of SIP Proxy is an evil thing.
>
>The real issue in all this is security. Many security folks believe
>that, in order to get proper security, the body of the signaling
>messages should be signed, and possibly encrypted. In that logic, the
>proxies only manipulate the information exposed in the SIP header, which
>can be as little as the pair of source and destination URL.
>
>Any intermediary that attempts to "rewrite the SDP" has to somehow
>insert itself in the end-to-end security exchange; many security folks
>will resent that as "breaking end-to-end security". Insofar as SDP
>rewriting attempts to provide a legitimate service, the challenge is
>thus to provide the service in such a way that it does not require
>breaking end-to-end encryption.
>
>-- Christian Huitema

I think there are more issues with B2BUA. A B2BUA's failure affects all
current calls :-(

-Jiri 

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From: Phil Reynolds <preynolds@ridgewaysystems.com>
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Subject: RE: [Sip] B2BUA
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If a telephone switch crashes you loose all calls...  The telecoms people
are clever enough to have thought about that and build redundancy, fault
tolerance and failover into their switches.

Maybe some of us IP-telecoms people have thought of similar techniques for
SIP intermediaries?

-----Original Message-----
From: Jiri Kuthan [mailto:jiri.kuthan@fokus.fraunhofer.de] 
Sent: 04 December 2002 12:32
To: Christian Huitema; Roy, Radhika R, ALASO; mbarnes@nortelnetworks.com
Cc: dean.willis@softarmor.com; Juan-Carlos.Rojas@alcatel.fr; sip@ietf.org
Subject: RE: [Sip] B2BUA

[...snip...]

I think there are more issues with B2BUA. A B2BUA's failure affects all
current calls :-(

-Jiri 

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<P><FONT SIZE=3D2>If a telephone switch crashes you loose all calls...&nbsp=
; The telecoms people are clever enough to have thought about that and buil=
d redundancy, fault tolerance and failover into their switches.</FONT></P>

<P><FONT SIZE=3D2>Maybe some of us IP-telecoms people have thought of simil=
ar techniques for SIP intermediaries?</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Jiri Kuthan [<A HREF=3D"mailto:jiri.kuthan@fokus.f=
raunhofer.de">mailto:jiri.kuthan@fokus.fraunhofer.de</A>] </FONT>
<BR><FONT SIZE=3D2>Sent: 04 December 2002 12:32</FONT>
<BR><FONT SIZE=3D2>To: Christian Huitema; Roy, Radhika R, ALASO; mbarnes@no=
rtelnetworks.com</FONT>
<BR><FONT SIZE=3D2>Cc: dean.willis@softarmor.com; Juan-Carlos.Rojas@alcatel=
=2Efr; sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: RE: [Sip] B2BUA</FONT>
</P>

<P><FONT SIZE=3D2>[...snip...]</FONT>
</P>

<P><FONT SIZE=3D2>I think there are more issues with B2BUA. A B2BUA's failu=
re affects all current calls :-(</FONT>
</P>

<P><FONT SIZE=3D2>-Jiri </FONT>
</P>

<P><FONT SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>Sip mailing list&nbsp; <A HREF=3D"https://www1.ietf.org/=
mailman/listinfo/sip" TARGET=3D"_blank">https://www1.ietf.org/mailman/listi=
nfo/sip</A></FONT>
<BR><FONT SIZE=3D2>This list is for NEW development of the core SIP Protoco=
l</FONT>
<BR><FONT SIZE=3D2>Use sip-implementors@cs.columbia.edu for questions on cu=
rrent sip Use sipping@ietf.org for new developments on the application of s=
ip</FONT></P>

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From mailnull@www1.ietf.org  Wed Dec  4 09:29:50 2002
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Date: Wed, 04 Dec 2002 15:16:25 +0100
To: Phil Reynolds <preynolds@ridgewaysystems.com>, sip@ietf.org
From: Jiri Kuthan <jiri@iptel.org>
Subject: RE: [Sip] B2BUA
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I think a better strategy to design reliable IP networks is not to
have stateful elements in the middle. It is sometimes not
easy to achieve, but you don't need to solve the problem what
happens when they fail. 

Besides that, all sort of middleboxes (B2BUA can IMHO be labeled
as app-layer middlebox) have e2e security issues as pointed out
by Christian, and are not free of scalability issues.

I suggest you read RFC3234 -- it covers various issues middleboxes
have, including failure handling.

-Jiri

At 02:36 PM 12/4/2002, Phil Reynolds wrote:

>If a telephone switch crashes you loose all calls...  The telecoms people are clever enough to have thought about that and build redundancy, fault tolerance and failover into their switches.
>
>Maybe some of us IP-telecoms people have thought of similar techniques for SIP intermediaries? 
>
>-----Original Message----- 
>From: Jiri Kuthan [<mailto:jiri.kuthan@fokus.fraunhofer.de>mailto:jiri.kuthan@fokus.fraunhofer.de] 
>Sent: 04 December 2002 12:32 
>To: Christian Huitema; Roy, Radhika R, ALASO; mbarnes@nortelnetworks.com 
>Cc: dean.willis@softarmor.com; Juan-Carlos.Rojas@alcatel.fr; sip@ietf.org 
>Subject: RE: [Sip] B2BUA 
>
>[...snip...] 
>
>I think there are more issues with B2BUA. A B2BUA's failure affects all current calls :-( 
>
>-Jiri 
>
>_______________________________________________ 
>Sip mailing list  <https://www1.ietf.org/mailman/listinfo/sip>https://www1.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've just submitted 
http://www.nostrum.com/~rjsparks/draft-ietf-sip-refer-07.txt
to the I-D repository.

This revision incorporates feedback from the IESG. The changes
were additions to the security considerations to discuss replay
and cut-and-paste issues with the use of message/sipfrag, and 
a more detailed discussion of the risks associated with acting
on a REFER request, particularly when the reference is to a 
non-SIP URI.

The updated sections are 5.2 and 5.3.

RjS 



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Padma Suresh wrote:
> Vijay,
> 
>      You are right -
> 
>     A B2BUA needs to propagate some of the headers received in a 
> request/response to the other endpoint.

Choosing which headers should be passed by a B2BUA and which not is
exactly the ambiguity that we should AVOID, as I pointed out in my
my response to the original poster.  As others have expressed, we may
try to taxonomize different types of B2BUAs, but each will have a
different manner of handling the headers (depending on the application
being realized by that particular B2BUA).

An anonymizer B2BUA may change To, From, Contact, etc, for instance (and
maintain internal state).  A network-hiding B2BUA may scrap all Via
headers coming in and only expose one Via header going out.  The
possibilities are endless.

>     Example: Require, Expires to name a few. This is because these 
> headers actually are relevant end to end.

A B2BUA may decide to consume the Require and change the Expires header
(maybe decrease its value) before sending it downstream, thus causing
havoc with the expectations of the sending UAC.

>    However, some of the other headers like Contact should be set by the 
> B2BUA as itself.  

Likewise, I can imagine a B2BUA which rendezvous two users and steps
out of subsequent signaling.  Thus the Contact it sends to each of
the users is that of the other peer.

I would strongly *stay away* from prescribing which headers must be
passed by a B2BUA and which must not.

Regards,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216

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Folks,

If my UA receives a request, it uses an API to interact with me thru the
GUI. Based on that, it will send a response.

If my SIP-PSTN gateway receives a request, it uses ISUP, Q.931, or what
have you to interact with me. Based on that, it will send a response.

If my SIP entity happens to use SIP instead of an API or ISUP to
interact with whoever of whatever it has to interact with, we call it
"B2BUA". Based on that, it will send a response. That simple. Every time
you drive your state machinery using SIP to communicate with another
entity, you are a B2BUA.

I do not think we need more definitions or examples. Things that can be
done with a B2BUA depend on how innovative are the engineers in your
company. We should not standardize them.

About a draft with B2BUA examples, it might be valuable to point out at
some stupid things you do not want to do because they may be harmful for
the network... but providing an exhaustive list of B2BUA types is a
terrible idea.

Regards,

Gonzalo
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Subject: Re: [Sip] Constructing the branch parameter at stateless proxies
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Hi Robert,

Sounds like a very good idea. One special case to consider is what 
happens if the request spirals back to the sending proxy (which 
presumably is stateful) -- either directly from the stateless proxy or 
via a number of other stateless proxies.  The stateful proxy in this 
case will have a server transaction and a client transaction with the 
same id, which might confuse some implementations.

This can be addressed simply by appending a fixed string to the branch 
value at each stateless proxy doing this trick, e.g. a single character. 
This guarantees the uniqueness properties required by rfc 3261, which 
seems desirable.

Anders


Robert Sparks wrote:
> This list has seen a couple of requests to clarify the language in
> RFC 3261 concerning how a stateless proxy creates a branch identifier.
> 
> The current text requires uniqueness, dependency on route decision
> affecting parameters and care to be sure that those parameters are
> retransmission invariant. It goes on to recommend a hash over a
> set of fields (this set has been the subject of the requests for
> clarification).
> 
> I think there's a much simpler solution. The purpose of this 
> branch parameter is to act as a transaction identifier. Who
> cares about this identifier? Only the stateful elements at
> each end of the transaction. Any stateless proxies between
> these elements don't use that identifier in any of their
> decisions, so I assert that it is not useful to change it
> as it crosses the stateless proxy.
> 
> I propose, then, that a stateless proxy simply copy
> the branch parameter from the topmost Via header field
> value it receives into the Via header field value it adds
> to the request. 
> 
> I don't believe it matters that this parameter contain the
> magic cookie. If it doesn't, the next stateful hop will use
> the 2543-style transaction identifiers provided by the sending
> stateful element. The stateless elements are by definition
> transaction transparent.
> 
> So - can anyone construct a scenario where this doesn't 
> function correctly? If not, we can remove several convoluted
> paragraphs from the spec and replace them with a very simple
> couple of sentences.
> 
> RjS
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.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 mailnull@www1.ietf.org  Wed Dec  4 11:13:38 2002
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Subject: Re: [Sip] Constructing the branch parameter at stateless proxies
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Anders Kristensen <akristensen@dynamicsoft.com>
Cc: sip@ietf.org
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On Wed, 2002-12-04 at 09:54, Anders Kristensen wrote:
> Hi Robert,
> 
> Sounds like a very good idea. One special case to consider is what 
> happens if the request spirals back to the sending proxy (which 
> presumably is stateful) -- either directly from the stateless proxy or 
> via a number of other stateless proxies.  The stateful proxy in this 
> case will have a server transaction and a client transaction with the 
> same id, which might confuse some implementations.

This is no different from the same proxy spiraling directly to itself
is it?

> 
> This can be addressed simply by appending a fixed string to the branch 
> value at each stateless proxy doing this trick, e.g. a single character. 
> This guarantees the uniqueness properties required by rfc 3261, which 
> seems desirable.

There are several things we could do here - append the count of present
via header fields for example to make sure you get different values as
you spiral through the stateless elements, etc. I don't think any of
this is necessary though - the stateless elements are supposed to be
transaction invisible. If they _can_ do something that botches the
transaction, we've got a weak spot in the protocol. I propose that
preserving the transaction identifier provided by the stateful element
is exactly the right thing to do, and that any mucking with it brings
nothing good to the table, and has the potential to bring harm.

> 
> Anders
> 
> 
> Robert Sparks wrote:
> > This list has seen a couple of requests to clarify the language in
> > RFC 3261 concerning how a stateless proxy creates a branch identifier.
> > 
> > The current text requires uniqueness, dependency on route decision
> > affecting parameters and care to be sure that those parameters are
> > retransmission invariant. It goes on to recommend a hash over a
> > set of fields (this set has been the subject of the requests for
> > clarification).
> > 
> > I think there's a much simpler solution. The purpose of this 
> > branch parameter is to act as a transaction identifier. Who
> > cares about this identifier? Only the stateful elements at
> > each end of the transaction. Any stateless proxies between
> > these elements don't use that identifier in any of their
> > decisions, so I assert that it is not useful to change it
> > as it crosses the stateless proxy.
> > 
> > I propose, then, that a stateless proxy simply copy
> > the branch parameter from the topmost Via header field
> > value it receives into the Via header field value it adds
> > to the request. 
> > 
> > I don't believe it matters that this parameter contain the
> > magic cookie. If it doesn't, the next stateful hop will use
> > the 2543-style transaction identifiers provided by the sending
> > stateful element. The stateless elements are by definition
> > transaction transparent.
> > 
> > So - can anyone construct a scenario where this doesn't 
> > function correctly? If not, we can remove several convoluted
> > paragraphs from the spec and replace them with a very simple
> > couple of sentences.
> > 
> > RjS
> > 
> > 
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.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: Anders Kristensen <akristensen@dynamicsoft.com>
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Subject: Re: [Sip] Constructing the branch parameter at stateless proxies
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Robert Sparks wrote:
> On Wed, 2002-12-04 at 09:54, Anders Kristensen wrote:
> 
>>Hi Robert,
>>
>>Sounds like a very good idea. One special case to consider is what 
>>happens if the request spirals back to the sending proxy (which 
>>presumably is stateful) -- either directly from the stateless proxy or 
>>via a number of other stateless proxies.  The stateful proxy in this 
>>case will have a server transaction and a client transaction with the 
>>same id, which might confuse some implementations.
> 
> 
> This is no different from the same proxy spiraling directly to itself
> is it?

Well, it's different in that it is NOT spiraling directly to itself and 
so is currently guaranteed to see a different branch ID in a spiralled 
request. It may be the case that noone depends on this but who knows?

> 
> 
>>This can be addressed simply by appending a fixed string to the branch 
>>value at each stateless proxy doing this trick, e.g. a single character. 
>>This guarantees the uniqueness properties required by rfc 3261, which 
>>seems desirable.
> 
> 
> There are several things we could do here - append the count of present
> via header fields for example to make sure you get different values as
> you spiral through the stateless elements, etc. I don't think any of
> this is necessary though - the stateless elements are supposed to be
> transaction invisible. If they _can_ do something that botches the
> transaction, we've got a weak spot in the protocol. I propose that
> preserving the transaction identifier provided by the stateful element
> is exactly the right thing to do, and that any mucking with it brings
> nothing good to the table, and has the potential to bring harm.

Like what?

Anders

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Subject: Re: [Sip] Constructing the branch parameter at stateless proxies
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Anders Kristensen <akristensen@dynamicsoft.com>
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On Wed, 2002-12-04 at 10:11, Anders Kristensen wrote:
> 
> 
> Robert Sparks wrote:
> > On Wed, 2002-12-04 at 09:54, Anders Kristensen wrote:
> > 
> >>Hi Robert,
> >>
> >>Sounds like a very good idea. One special case to consider is what 
> >>happens if the request spirals back to the sending proxy (which 
> >>presumably is stateful) -- either directly from the stateless proxy or 
> >>via a number of other stateless proxies.  The stateful proxy in this 
> >>case will have a server transaction and a client transaction with the 
> >>same id, which might confuse some implementations.
> > 
> > 
> > This is no different from the same proxy spiraling directly to itself
> > is it?
> 
> Well, it's different in that it is NOT spiraling directly to itself and 
> so is currently guaranteed to see a different branch ID in a spiralled 
> request. It may be the case that noone depends on this but who knows?

Ok - so an implementation _may_ have gone to some effort to ensure
that it wasn't sending a message to itself and is counting on not
seeing the same transaction identifier come back to it. Not a choice
I would recommend, but I can see how someone might have made it.

We could avoid this with the suffix mechanism you suggest, or we
could add an admonishment to the spec to tell systems to be prepared
to have their transaction ID come back. I'd rather do the second.

> 
> > 
> > 
> >>This can be addressed simply by appending a fixed string to the branch 
> >>value at each stateless proxy doing this trick, e.g. a single character. 
> >>This guarantees the uniqueness properties required by rfc 3261, which 
> >>seems desirable.
> > 
> > 
> > There are several things we could do here - append the count of present
> > via header fields for example to make sure you get different values as
> > you spiral through the stateless elements, etc. I don't think any of
> > this is necessary though - the stateless elements are supposed to be
> > transaction invisible. If they _can_ do something that botches the
> > transaction, we've got a weak spot in the protocol. I propose that
> > preserving the transaction identifier provided by the stateful element
> > is exactly the right thing to do, and that any mucking with it brings
> > nothing good to the table, and has the potential to bring harm.
> 
> Like what?

Well, nothing earth-shattering. Right now, just an increased probability
that implementation error will lead to protocol failure. Its hard to get
this right, and we can greatly simplify it.

My growing distaste for what the stateless proxies are currently doing
comes from seeing it somewhat like a NAT. The transaction is really 
between the two stateful elements reguardless of how many stateless
elements lie between. Right now, the stateless elements are 
"translating" the transaction identifier as the message goes through.
We haven't identified anything in the protocol that breaks as long
as they do this consistently, but I haven't been able to identify any
benefit to the behavior. 

Now I'll go out on a limb. In the future, we might want as part of 
diagnostics, error recovery, or some similar function, to be able
to refer to the transaction by its identifier. It would be very
nice if that identifier were the same at the stateful elements on
each end of the transaction. 

> 
> Anders


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From: "Padma Suresh" <padma_suresh@hotmail.com>
To: vkg@lucent.com
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Subject: Re: [Sip] B2BUA & Max-Forwards
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<html><div style='background-color:'><DIV>
<P>A B2BUA is an intermediate network element and MUST maintain the expectations of the endpoints involved in the session.</P>
<P>Not forwarding the Require or Expires headers (and some more) would break the above which would not be quite acceptable.</P>
<P>However, yes, the B2BUA can choose to step out of the signalling path by setting the contact header to the appropriate endpoint.</P>
<P>Vijay I agree that the ambiguity of choosing which header to pass on and which not, could be well avoided.</P>
<P>Regards</P>
<P>Padma<BR><BR></P></DIV>
<DIV></DIV>
<DIV></DIV>&gt;From: "Vijay K. Gurbani" <VKG@LUCENT.COM>
<DIV></DIV>&gt;To: Padma Suresh <PADMA_SURESH@HOTMAIL.COM>
<DIV></DIV>&gt;CC: pete@tech-know-ware.com, sip@ietf.org 
<DIV></DIV>&gt;Subject: Re: [Sip] B2BUA &amp; Max-Forwards 
<DIV></DIV>&gt;Date: Wed, 04 Dec 2002 09:21:11 -0600 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;Padma Suresh wrote: 
<DIV></DIV>&gt;&gt;Vijay, 
<DIV></DIV>&gt;&gt; 
<DIV></DIV>&gt;&gt; You are right - 
<DIV></DIV>&gt;&gt; 
<DIV></DIV>&gt;&gt; A B2BUA needs to propagate some of the headers received in a 
<DIV></DIV>&gt;&gt;request/response to the other endpoint. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;Choosing which headers should be passed by a B2BUA and which not is 
<DIV></DIV>&gt;exactly the ambiguity that we should AVOID, as I pointed out in my 
<DIV></DIV>&gt;my response to the original poster. As others have expressed, we 
<DIV></DIV>&gt;may 
<DIV></DIV>&gt;try to taxonomize different types of B2BUAs, but each will have a 
<DIV></DIV>&gt;different manner of handling the headers (depending on the 
<DIV></DIV>&gt;application 
<DIV></DIV>&gt;being realized by that particular B2BUA). 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;An anonymizer B2BUA may change To, From, Contact, etc, for instance 
<DIV></DIV>&gt;(and 
<DIV></DIV>&gt;maintain internal state). A network-hiding B2BUA may scrap all Via 
<DIV></DIV>&gt;headers coming in and only expose one Via header going out. The 
<DIV></DIV>&gt;possibilities are endless. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;&gt; Example: Require, Expires to name a few. This is because these 
<DIV></DIV>&gt;&gt;headers actually are relevant end to end. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;A B2BUA may decide to consume the Require and change the Expires 
<DIV></DIV>&gt;header 
<DIV></DIV>&gt;(maybe decrease its value) before sending it downstream, thus 
<DIV></DIV>&gt;causing 
<DIV></DIV>&gt;havoc with the expectations of the sending UAC. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;&gt; However, some of the other headers like Contact should be set 
<DIV></DIV>&gt;&gt;by the B2BUA as itself. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;Likewise, I can imagine a B2BUA which rendezvous two users and steps 
<DIV></DIV>&gt;out of subsequent signaling. Thus the Contact it sends to each of 
<DIV></DIV>&gt;the users is that of the other peer. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;I would strongly *stay away* from prescribing which headers must be 
<DIV></DIV>&gt;passed by a B2BUA and which must not. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;Regards, 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;- vijay 
<DIV></DIV>&gt;-- 
<DIV></DIV>&gt;Vijay K. Gurbani vkg@{lucent.com,research.bell-labs.com,acm.org} 
<DIV></DIV>&gt;Wireless Networks Group/Internet Software and Services 
<DIV></DIV>&gt;Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 
<DIV></DIV>&gt;6G-440 
<DIV></DIV>&gt;Naperville, Illinois 60566 Voice: +1 630 224 0216 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;_______________________________________________ 
<DIV></DIV>&gt;Sip mailing list https://www1.ietf.org/mailman/listinfo/sip 
<DIV></DIV>&gt;This list is for NEW development of the core SIP Protocol 
<DIV></DIV>&gt;Use sip-implementors@cs.columbia.edu for questions on current sip 
<DIV></DIV>&gt;Use sipping@ietf.org for new developments on the application of sip 
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REGISTER does not create call state, that's it.

If you create call state with INVITE, the B2BUA
will keep it until the call is gone or a failure
occurs.

-Jiri

At 06:14 PM 12/4/2002, Kedar Patil wrote:

>BtoBUA need not always be call stateful. 
>e.g. a SIP REGISTER request reaches a BtoBUA, 
>it sends a new request to appropriate Registrar 
>based on some local application specific 
>configuration. 
>
>There isn't any network state here (except the 
>transaction state, like sip proxy). 
>
>Pls. let me know, if you foresee any problem here. 
>
>thanks, 
>= Kedar = 

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Date: Wed, 04 Dec 2002 13:02:38 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sip] B2BUA
In-reply-to: 
 <00533D13955AD411AF3800A0C9B4263994D87F@ridgeway.internal.ridgewaysystems.com>
To: "'Phil Reynolds'" <preynolds@ridgewaysystems.com>, sip@ietf.org
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>The telecoms people are clever enough to have thought 
>about that and build redundancy, fault tolerance and failover into
their switches.

This has proven unfortunately not to be the case on September 11 when
telecom service failed for days, while Internet traffic continued
smoothly. I suggest you to look at the central control model of the
switch network versus the e2e architecture of the Internet. SIP enabled
networks are also fully distributed and the e2e control principle
maintained - that's why state in placed in SIP endpoints.

I am not sure if a few words here can give full justice to the superior
resilience of the Internet and to SIP enabled services,...

Thanks, Henry

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Phil
Reynolds
Sent: Wednesday, December 04, 2002 7:37 AM
To: sip@ietf.org
Subject: RE: [Sip] B2BUA


If a telephone switch crashes you loose all calls...  The telecoms
people are clever enough to have thought about that and build
redundancy, fault tolerance and failover into their switches.
Maybe some of us IP-telecoms people have thought of similar techniques
for SIP intermediaries? 
-----Original Message----- 
From: Jiri Kuthan [mailto:jiri.kuthan@fokus.fraunhofer.de] 
Sent: 04 December 2002 12:32 
To: Christian Huitema; Roy, Radhika R, ALASO; mbarnes@nortelnetworks.com

Cc: dean.willis@softarmor.com; Juan-Carlos.Rojas@alcatel.fr;
sip@ietf.org 
Subject: RE: [Sip] B2BUA 
[...snip...] 
I think there are more issues with B2BUA. A B2BUA's failure affects all
current calls :-( 
-Jiri 
_______________________________________________ 
Sip mailing list  https://www1.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: "Cullen Jennings" <fluffy@cisco.com>
To: "Phil Reynolds" <preynolds@ridgewaysystems.com>, <sip@ietf.org>
Subject: RE: [Sip] B2BUA & Max-Forwards
Date: Wed, 4 Dec 2002 12:04:17 -0800
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RE: [Sip] B2BUA & Max-ForwardsI think the answer to what it is is fairly
simple - it's not a proxy and it's not a B2BUA.

And in some call flows, it will probably be called broken :-)

Cullen


  -----Original Message-----
  From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Phil
Reynolds
  Sent: Wednesday, December 04, 2002 1:37 AM
  To: sip@ietf.org
  Subject: RE: [Sip] B2BUA & Max-Forwards


  Now your back to restricting what I can and can't do in an intermediary.
It can't be a B2BUA 'cos it doesn't need to honour require headers (the UA
at the other end does that, if they were meant for my intermediary then they
would be proxy-require?), and it can't be a proxy 'cos of the rules about
not changing the body so what is it?

  It's not sanctioned by the RFC, yet id does an important and useful job.

  -----Original Message-----
  From: Adam Roach [mailto:adam@dynamicsoft.com]
  Sent: 04 December 2002 00:37
  To: 'Padma Suresh'; vkg@lucent.com; pete@tech-know-ware.com
  Cc: sip@ietf.org
  Subject: RE: [Sip] B2BUA & Max-Forwards



  -----Original Message-----
  From: Padma Suresh [mailto:padma_suresh@hotmail.com]

  > A B2BUA needs to propagate some of the headers received in a
  > request/response to the other endpoint. Example: Require...

  No. As emphatically as possible: NO. This is completely wrong, and it
appears to be a growing -- and dangerous -- misconception.

  A B2BUA cannot transparently pass Require through. It must understand and
invoke the extension indicated.

    "The Require header field contains a list of option tags, described in
     Section 19.2.  Each option tag defines a SIP extension that MUST be
     understood to process the request."

  A back to back user agent *is* a UAS, and *is* bound by this provision.

  Whether the UAC portion then turns around and imposes the same requirement
on the UAS it contacts is a local decision.

  /a
  _______________________________________________
  Sip mailing list  https://www1.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><HEAD><TITLE>RE: [Sip] B2BUA & Max-Forwards</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 5.50.4807.2300" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D009324119-04122002><FONT face=3DArial color=3D#0000ff =
size=3D2>I=20
think the answer to what it is is fairly simple - it's not a proxy and =
it's not=20
a B2BUA. </FONT></SPAN></DIV>
<DIV><SPAN class=3D009324119-04122002><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D009324119-04122002><FONT face=3DArial color=3D#0000ff =
size=3D2>And in=20
some call flows, it will probably be called broken :-) =
</FONT></SPAN></DIV>
<DIV><SPAN class=3D009324119-04122002><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D009324119-04122002><FONT face=3DArial color=3D#0000ff =

size=3D2>Cullen</FONT></SPAN></DIV>
<DIV><SPAN class=3D009324119-04122002></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D009324119-04122002><FONT face=3DArial 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 dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> sip-admin@ietf.org =

  [mailto:sip-admin@ietf.org]<B>On Behalf Of </B>Phil =
Reynolds<BR><B>Sent:</B>=20
  Wednesday, December 04, 2002 1:37 AM<BR><B>To:</B>=20
  sip@ietf.org<BR><B>Subject:</B> RE: [Sip] B2BUA &amp;=20
  Max-Forwards<BR><BR></FONT></DIV>
  <P><FONT size=3D2>Now your back to restricting what I can and can't do =
in an=20
  intermediary. It can't be a B2BUA 'cos it doesn't need to honour =
require=20
  headers (the UA at the other end does that, if they were meant for my=20
  intermediary then they would be proxy-require?), and it can't be a =
proxy 'cos=20
  of the rules about not changing the body so what is it?</FONT></P>
  <P><FONT size=3D2>It's not sanctioned by the RFC, yet id does an =
important and=20
  useful job.</FONT> </P>
  <P><FONT size=3D2>-----Original Message-----</FONT> <BR><FONT =
size=3D2>From: Adam=20
  Roach [<A =
href=3D"mailto:adam@dynamicsoft.com">mailto:adam@dynamicsoft.com</A>]=20
  </FONT><BR><FONT size=3D2>Sent: 04 December 2002 00:37</FONT> =
<BR><FONT=20
  size=3D2>To: 'Padma Suresh'; vkg@lucent.com; =
pete@tech-know-ware.com</FONT>=20
  <BR><FONT size=3D2>Cc: sip@ietf.org</FONT> <BR><FONT size=3D2>Subject: =
RE: [Sip]=20
  B2BUA &amp; Max-Forwards</FONT> </P><BR>
  <P><FONT size=3D2>-----Original Message-----</FONT> <BR><FONT =
size=3D2>From: Padma=20
  Suresh [<A=20
  =
href=3D"mailto:padma_suresh@hotmail.com">mailto:padma_suresh@hotmail.com<=
/A>]</FONT>=20
  <BR><FONT size=3D2>&nbsp;</FONT> <BR><FONT size=3D2>&gt; A B2BUA needs =
to=20
  propagate some of the headers received in a </FONT><BR><FONT =
size=3D2>&gt;=20
  request/response to the other endpoint. Example: Require...</FONT> =
</P>
  <P><FONT size=3D2>No. As emphatically as possible: NO. This is =
completely wrong,=20
  and it appears to be a growing -- and dangerous -- =
misconception.</FONT></P>
  <P><FONT size=3D2>A B2BUA cannot transparently pass Require through. =
It must=20
  understand and invoke the extension indicated.</FONT> </P>
  <P><FONT size=3D2>&nbsp; "The Require header field contains a list of =
option=20
  tags, described in</FONT> <BR><FONT size=3D2>&nbsp;&nbsp; Section =
19.2.&nbsp;=20
  Each option tag defines a SIP extension that MUST be</FONT> <BR><FONT=20
  size=3D2>&nbsp;&nbsp; understood to process the request."</FONT> </P>
  <P><FONT size=3D2>A back to back user agent *is* a UAS, and *is* bound =
by this=20
  provision.</FONT> </P>
  <P><FONT size=3D2>Whether the UAC portion then turns around and =
imposes the same=20
  requirement on the UAS it contacts is a local decision.</FONT> </P>
  <P><FONT size=3D2>/a</FONT> <BR><FONT=20
  size=3D2>_______________________________________________</FONT> =
<BR><FONT=20
  size=3D2>Sip mailing list&nbsp; <A target=3D_blank=20
  =
href=3D"https://www1.ietf.org/mailman/listinfo/sip">https://www1.ietf.org=
/mailman/listinfo/sip</A></FONT>=20
  <BR><FONT size=3D2>This list is for NEW development of the core SIP=20
  Protocol</FONT> <BR><FONT size=3D2>Use =
sip-implementors@cs.columbia.edu for=20
  questions on current sip Use sipping@ietf.org for new developments on =
the=20
  application of sip</FONT></P></BLOCKQUOTE></BODY></HTML>

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Kedar Patil wrote:
> This discussion raises one more doubt for me, pls. clarify.
> 
> UA1------->BtoBUA---------->UA2
 >
> Is the BtoBUA a SIP endpoint from UA1's point of view? In other
> words, are the the "Require:" etc end-to-end headers sent by UA1,
> meant for B2BUA or UA2?

As Adam pointed out in an earlier email in this thread, "A B2BUA cannot
transparently pass Require through. It must understand and invoke the
extension indicated." (rfc3261 says the same; see sec 20.32)  Having
said that, I realize that there exist what are called transparent B2BUAs
that pass e2e headers through without operating on them.  Time and
experience will decide if they are a boon or a bane; right now my
personal feeling is that a proxy is a far more better tool than a
transparent B2BUA.  It plays by the rules and is unlikely to break
e2e feature support.

> Is the UAS part in in BtoBUA is bound by the definition and rules
> in 3261?

Yes, it is a UA.

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216

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-----Original Message-----
From: Phil Reynolds [mailto:preynolds@ridgewaysystems.com]

> Now your back to restricting what I can and can't do in an
> intermediary. It can't be a B2BUA 'cos it doesn't need to
> honour require headers (the UA at the other end does that,
> if they were meant for my intermediary then they would be
> proxy-require?), and it can't be a proxy 'cos of the rules
> about not changing the body so what is it?

Broken.

The restrictions placed on nodes by 3261 (which I'm merely
reiterating, not imposing) aren't an authoritarian attempt
at unnecessary proscription. They serve a purpose.

I'll describe one of the (several) purposes that applies
to the situation you propose.

One of the things that UAs are required to be able to do is
sign and/or encrypt the bodies and/or some subset of the
headers to ensure confidentiality and/or integrity. The
protocol places restrictions on various nodes to ensure
that this can be done. Your box is broken precisely because
it interferes with these abilities (or, more precisely,
these abilities interfere with it).

/a
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From mailnull@www1.ietf.org  Wed Dec  4 15:48:08 2002
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From: Adam Roach <adam@dynamicsoft.com>
To: "'Kedar Patil'" <kpatil@sta.samsung.com>,
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	 <adam@dynamicsoft.com>,
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Subject: RE: [Sip] B2BUA & Max-Forwards
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-----Original Message-----
From: Kedar Patil [mailto:kpatil@sta.samsung.com]

> This discussion raises one more doubt for me, pls. clarify. 
>
> UA1------->BtoBUA---------->UA2 
>
> Is the BtoBUA a SIP endpoint from UA1's point of view?

Yes.

> In other 
> words, are the the "Require:" etc end-to-end headers sent by UA1, 
> meant for B2BUA or UA2? 

B2BUA.

> Is the UAS part in in BtoBUA is bound by the definition and rules 
> in 3261? 

Yes. I'll revise your drawing to more clearly show what is going on here.

UAC(a)--->UAS(b)===[application logic here]===UAC(b)--->UAS(c)

/a
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From mailnull@www1.ietf.org  Wed Dec  4 15:54:14 2002
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From: Adam Roach <adam@dynamicsoft.com>
To: "'Juan-Carlos.Rojas@alcatel.fr'" <Juan-Carlos.Rojas@alcatel.fr>,
        Adam Roach <adam@dynamicsoft.com>
Cc: "'Padma Suresh'" <padma_suresh@hotmail.com>, vkg@lucent.com,
        pete@tech-know-ware.com, sip@ietf.org
Subject: RE: [Sip] B2BUA & Max-Forwards
Date: Wed, 4 Dec 2002 14:55:15 -0600 
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Unless you were merely trying to agree at length with the
final paragraph of my previous message, I'm afraid that
you missed my point rather broadly.

I'll reiterate the main point concisely: a B2BUA is a UAS
attached to a UAC by application logic.

Read that carefully: Part of a B2BUA is a UAS.

That UAS is bound by RFC 3261 to certain behaviors.

One of those behaviors is honoring "Require" headers.

/a

> -----Original Message-----
> From: Juan-Carlos.Rojas@alcatel.fr 
> [mailto:Juan-Carlos.Rojas@alcatel.fr]
> Sent: Wednesday, December 04, 2002 1:51
> To: Adam Roach
> Cc: 'Padma Suresh'; vkg@lucent.com; pete@tech-know-ware.com;
> sip@ietf.org
> Subject: RE: [Sip] B2BUA & Max-Forwards
> 
> 
> 
> Hello,
> 
> As it was already stated in this discusion thread, a B2BUA is an
> *application* linking several UAs
> As such, the behaviour of a B2BUA cannot be standardised.
> 
> Therefore, one cannot say that "A B2BUA needs to propagate some of the
> headers received".
> However, you can say that the application "x" behaves as a 
> B2BUA, and that
> the application "x" "needs to propagate" the a-header or the 
> b-header etc.
> for the needs of the application. This is the choice of the 
> application,
> that's all.
> 
> Best regards
> Juan Carlos
> 
> 
> 
> 
> 
> Adam Roach <adam@dynamicsoft.com>@ietf.org on 04/12/2002 01:36:50
> 
> Sent by:  sip-admin@ietf.org
> 
> 
> To:   "'Padma Suresh'" <padma_suresh@hotmail.com>, vkg@lucent.com,
>       pete@tech-know-ware.com
> cc:   sip@ietf.org
> Subject:  RE: [Sip] B2BUA & Max-Forwards
> 
> 
> -----Original Message-----
> From: Padma Suresh [mailto:padma_suresh@hotmail.com]
> 
> > A B2BUA needs to propagate some of the headers received in a
> > request/response to the other endpoint. Example: Require...
> 
> No. As emphatically as possible: NO. This is completely wrong,
> and it appears to be a growing -- and dangerous -- misconception.
> 
> A B2BUA cannot transparently pass Require through. It must
> understand and invoke the extension indicated.
> 
>   "The Require header field contains a list of option tags, 
> described in
>    Section 19.2.  Each option tag defines a SIP extension that MUST be
>    understood to process the request."
> 
> A back to back user agent *is* a UAS, and *is* bound by this
> provision.
> 
> Whether the UAC portion then turns around and imposes the same
> requirement on the UAS it contacts is a local decision.
> 
> /a
> _______________________________________________
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> 
> 
> 
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From mailnull@www1.ietf.org  Wed Dec  4 16:02:55 2002
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        Kedar Patil
	 <kpatil@sta.samsung.com>
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Subject: RE: [Sip] B2BUA & Max-Forwards
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> -----Original Message-----
> From: Vijay K. Gurbani [mailto:vkg@lucent.com]
>
> I realize that there exist what are called 
> transparent B2BUAs that pass e2e headers through
> without operating on them.

And the point that I'm trying to make is that such network
elements are out of spec. The will run into interoperability
problems with compliant nodes.

/a
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From mailnull@www1.ietf.org  Wed Dec  4 16:33:54 2002
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From: Adam Roach <adam@dynamicsoft.com>
To: "'William Marshall'" <wtm@research.att.com>, dean.willis@softarmor.com
Cc: mbarnes@nortelnetworks.com, Juan-Carlos.Rojas@alcatel.fr, sip@ietf.org
Subject: RE: [Sip] B2BUA
Date: Wed, 4 Dec 2002 15:34:22 -0600 
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> -----Original Message-----
> From: William Marshall [mailto:wtm@research.att.com]
> 
> What is built instead are boxes that are close to the definition of
> proxies, but not quite.  They serve a real need in the market (and
> vendors tend to build what sells).  But due to various details, like
> looking at the attachment (a proxy-no-no) to open pinholes in 
> the firewall,
> they don't strictly qualify as proxies.
> 
> Saying there are only two types of boxes, proxies and devilish-B2BUAs,
> ignores the reality.

Right. And no one from the ISOC, IAB, IESG, or the working group is
going to show up with tanks and guns to force you to not build
these.

However, there are demonstratable cases in which these nodes break
when communicating with other nodes that follow 3261 to the letter.

Our goal is not to make protocols that kinda interoperate under some
circumstances... sorta... maybe. Our goal is to make protocols that
interoperate. Period.

So, while you might build, deploy, and sell such boxes that fail
to interoperate with fully compliant nodes under certain circumstances
(encrypted SDP, in your example), that does NOT mean that condoning
such partially-interoperable behavior would be appropriate.

In fact, the illusion of implied interoperability that would
result from codifying such partially-interoperable nodes would be
dishonest. Doing so would damage the credibility of the protocols
generated by the IETF in general, and the SIP-related working
groups in particular.

/a
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
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The thread on the Discuss for service discovery related to the requirement
for S/MIME seems to have petered out - it would be nice if we could come to
a conclusion and go forward with the draft.

From my own reading of the thread, I think the predominant consensus was
that requiring S/MIME for registrars (in the context of architectures in
which hop-by-hop security is unavailable or insufficient) was reasonable.
There were concerns that this was a broader issue than just service
discovery - but ultimately, I don't think that argues against the suggested
change from the IESG, it merely argues that this fix may have even broader
utility.

Can we go forward with the requested fix?

Jon Peterson
NeuStar, Inc.
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        RFC 3323
                            
        Title:      A Privacy Mechanism for the Session Initiation
                    Protocol (SIP)
        Author(s):  J. Peterson
        Status:     Standards Track
        Date:       November 2002
        Mailbox:    jon.peterson@neustar.biz
        Pages:      22
        Characters: 54116
        Updates/Obsoletes/SeeAlso:   None
                            
        I-D Tag:    draft-ietf-sip-privacy-general-01.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3323.txt


This document defines new mechanisms for the Session Initiation
Protocol (SIP) in support of privacy.  Specifically, guidelines are
provided for the creation of messages that do not divulge personal
identity information.  A new "privacy service" logical role for
intermediaries is defined to answer some privacy requirements that
user agents cannot satisfy themselves.  Finally, means are presented
by which a user can request particular functions from a privacy
service.

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

This is now a Proposed Standard Protocol.

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
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        RFC 3324
                            
        Title:      Short Term Requirements for Network Asserted
                    Identity
        Author(s):  M. Watson
        Status:     Informational
        Date:       November 2002
        Mailbox:    mwatson@nortelnetworks.com
        Pages:      11
        Characters: 21964
        Updates/Obsoletes/SeeAlso:   None
                            
        I-D Tag:    draft-ietf-sipping-nai-reqs-02.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3324.txt


A Network Asserted Identity is an identity initially derived by
a Session Initiation Protocol (SIP) network intermediary as a result
of an authentication process.  This document describes short term
requirements for the exchange of Network Asserted Identities within
networks of securely interconnected trusted nodes and to User Agents
securely connected to such networks.

There is no requirement for identities asserted by a UA in a SIP
message to be anything other than the user's desired alias.

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

This memo provides information for the Internet community.  It does
not specify an Internet standard of any kind.  Distribution of this
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        RFC 3325
                            
        Title:      Private Extensions to the Session Initiation
                    Protocol (SIP) for Asserted Identity within
                    Trusted Networks
        Author(s):  C. Jennings, J. Peterson, M. Watson
        Status:     Informational
        Date:       November 2002
        Mailbox:    fluffy@cisco.com, Jon.Peterson@NeuStar.biz,
                    mwatson@nortelnetworks.com
        Pages:      18
        Characters: 36170
        Updates/Obsoletes/SeeAlso:   None
                            
        I-D Tag:    draft-ietf-sip-asserted-identity-02.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3325.txt


This document describes private extensions to the Session Initiation
Protocol (SIP) that enable a network of trusted SIP servers to assert
the identity of authenticated users, and the application of existing
privacy mechanisms to the identity problem.  The use of these
extensions is only applicable inside an administrative domain with
previously agreed-upon policies for generation, transport and usage of
such information.  This document does NOT offer a general privacy or
identity model suitable for use between different trust domains, or
use in the Internet at large.

This document is a product of the Session Initiation Protocol Working
Group and the Session Initiation Proposal Investigation Working Groups
of the IETF.  

This memo provides information for the Internet community.  It does
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From mailnull@www1.ietf.org  Wed Dec  4 17:42:57 2002
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From: "Roy, Radhika R, ALASO" <rrroy@att.com>
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Adam:

I would appreciate if you would clarify the following questions provided inline [RRR].

Radhika


> Is the UAS part in in BtoBUA is bound by the definition and rules 
> in 3261? 

Yes. I'll revise your drawing to more clearly show what is going on here.

UAC(a)--->UAS(b)===[application logic here]===UAC(b)--->UAS(c)

/a

[RRR] I like to know who has been the called party when UAC(a)) sent the INVITE message? Has it been UAS(b) or UAS(c)?

[RRR] If the answer is UAS(c), the question is: Why the call has been routed to UAS(b) by UAC(a) in the first place?

[RRR] If the answer is UAS(b), probably UAC(a) does not care.
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To: "'Peterson, Jon'" <jon.peterson@neustar.biz>,
        "'sip@ietf.org'"
	 <sip@ietf.org>
Subject: RE: [Sip] Discuss on service discovery
Date: Wed, 4 Dec 2002 16:44:49 -0600 
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... 
> Can we go forward with the requested fix?

In light of the conclusions of the previous discussions on the
topic, I have no further objections.

I would, however, like to see a bug opened against 3261 to
address this issue so that the solution can eventually appear
in a centralized place.

/a
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Adam Roach'" <adam@dynamicsoft.com>, "'sip@ietf.org'" <sip@ietf.org>
Subject: RE: [Sip] Discuss on service discovery
Date: Wed, 4 Dec 2002 17:47:15 -0500 
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I think that's quite reasonable.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Adam Roach [mailto:adam@dynamicsoft.com]
> Sent: Wednesday, December 04, 2002 2:45 PM
> To: 'Peterson, Jon'; 'sip@ietf.org'
> Subject: RE: [Sip] Discuss on service discovery
> 
> 
> ... 
> > Can we go forward with the requested fix?
> 
> In light of the conclusions of the previous discussions on the
> topic, I have no further objections.
> 
> I would, however, like to see a bug opened against 3261 to
> address this issue so that the solution can eventually appear
> in a centralized place.
> 
> /a
> 
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From: Adam Roach <adam@dynamicsoft.com>
To: "'Roy, Radhika R, ALASO'" <rrroy@att.com>,
        Adam Roach
	 <adam@dynamicsoft.com>
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Subject: RE: [Sip] B2BUA & Max-Forwards
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> UAC(a)--->UAS(b)===[application logic here]===UAC(b)--->UAS(c)
> 
> [RRR] I like to know who has been the called party when 
> UAC(a)) sent the INVITE message? Has it been UAS(b) or UAS(c)?

I'm going to first need to clarify your question.

The question I will answer is: "If an INVITE is sent through the
network described in this diagram, what is the value of the
Request-URI when UAC(a) sends it?"

The answer is: it depends. In general, yes, it will probably be
either of UAS(b) or UAS(c).

> [RRR] If the answer is UAS(c), the question is: Why the call 
> has been routed to UAS(b) by UAC(a) in the first place?

The most obvious of several plausible answers is: UAC(a)
has UAS(b) configured as a local outbound proxy.

> [RRR] If the answer is UAS(b), probably UAC(a) does not care.

About what?

/a
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From mailnull@www1.ietf.org  Wed Dec  4 18:36:51 2002
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From: "Paul Long" <plong@packetizer.com>
To: sip@ietf.org
Subject: RE: [Sip] B2BUA & Max-Forwards
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Adam,

FWIW, a B2BUA operates identically in this regard to an H.323 routing 
gatekeeper. The gatekeeper must contain H.323 endpoint functionality 
facing each of the two "real" endpoints with "application logic" in 
between the two functional endpoints, although this is not made 
explicit in any of the Recommendations. Vendors have wedged all sorts 
of applications in between these two virtual endpoints within the 
gatekeeper. Interestingly, some H.323 vendors have undergone the same 
initial naivete as some of the posters on this reflector. The bottom 
line is that you must fully terminate the connections on both sides and 
not simply pass everything or even selectively some things through and 
hope for the best.

Paul

> Unless you were merely trying to agree at length with the
> final paragraph of my previous message, I'm afraid that
> you missed my point rather broadly.
> 
> I'll reiterate the main point concisely: a B2BUA is a UAS
> attached to a UAC by application logic.
> 
> Read that carefully: Part of a B2BUA is a UAS.
> 
> That UAS is bound by RFC 3261 to certain behaviors.
> 
> One of those behaviors is honoring "Require" headers.
> 
> /a
> 
> > -----Original Message-----
> > From: Juan-Carlos.Rojas@alcatel.fr 
> > [mailto:Juan-Carlos.Rojas@alcatel.fr]
> > Sent: Wednesday, December 04, 2002 1:51
> > To: Adam Roach
> > Cc: 'Padma Suresh'; vkg@lucent.com; pete@tech-know-ware.com;
> > sip@ietf.org
> > Subject: RE: [Sip] B2BUA & Max-Forwards
> > 
> > 
> > 
> > Hello,
> > 
> > As it was already stated in this discusion thread, a B2BUA is an
> > *application* linking several UAs
> > As such, the behaviour of a B2BUA cannot be standardised.
> > 
> > Therefore, one cannot say that "A B2BUA needs to propagate some of 
the
> > headers received".
> > However, you can say that the application "x" behaves as a 
> > B2BUA, and that
> > the application "x" "needs to propagate" the a-header or the 
> > b-header etc.
> > for the needs of the application. This is the choice of the 
> > application,
> > that's all.
> > 
> > Best regards
> > Juan Carlos
> > 
> > 
> > 
> > 
> > 
> > Adam Roach <adam@dynamicsoft.com>@ietf.org on 04/12/2002 01:36:50
> > 
> > Sent by:  sip-admin@ietf.org
> > 
> > 
> > To:   "'Padma Suresh'" <padma_suresh@hotmail.com>, vkg@lucent.com,
> >       pete@tech-know-ware.com
> > cc:   sip@ietf.org
> > Subject:  RE: [Sip] B2BUA & Max-Forwards
> > 
> > 
> > -----Original Message-----
> > From: Padma Suresh [mailto:padma_suresh@hotmail.com]
> > 
> > > A B2BUA needs to propagate some of the headers received in a
> > > request/response to the other endpoint. Example: Require...
> > 
> > No. As emphatically as possible: NO. This is completely wrong,
> > and it appears to be a growing -- and dangerous -- misconception.
> > 
> > A B2BUA cannot transparently pass Require through. It must
> > understand and invoke the extension indicated.
> > 
> >   "The Require header field contains a list of option tags, 
> > described in
> >    Section 19.2.  Each option tag defines a SIP extension that MUST 
be
> >    understood to process the request."
> > 
> > A back to back user agent *is* a UAS, and *is* bound by this
> > provision.
> > 
> > Whether the UAC portion then turns around and imposes the same
> > requirement on the UAS it contacts is a local decision.
> > 
> > /a
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From: "Padma Suresh" <padma_suresh@hotmail.com>
To: adam@dynamicsoft.com, rrroy@att.com
Cc: sip@ietf.org
Subject: RE: [Sip] B2BUA & Max-Forwards
Date: Thu, 05 Dec 2002 00:16:34 +0000
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<html><div style='background-color:'><DIV>
<P>For UAC (a) UAS (c) is the intended called party in the call. We have to define&nbsp;the intended party in a call when a B2BUA is involved. B2BUA IS an intermediate network element. It&nbsp;acts as some kind of a controller for a domain.&nbsp;It is not the called party in any call. </P>
<P>Regards</P>
<P>Padma</P></DIV>
<DIV></DIV>
<DIV></DIV>&gt;From: Adam Roach <ADAM@DYNAMICSOFT.COM>
<DIV></DIV>&gt;To: "'Roy, Radhika R, ALASO'" <RRROY@ATT.COM>, Adam Roach <ADAM@DYNAMICSOFT.COM>
<DIV></DIV>&gt;CC: sip@ietf.org 
<DIV></DIV>&gt;Subject: RE: [Sip] B2BUA &amp; Max-Forwards 
<DIV></DIV>&gt;Date: Wed, 4 Dec 2002 16:58:14 -0600 
<DIV></DIV>&gt; 
<DIV></DIV>&gt; &gt; UAC(a)---&gt;UAS(b)===[application logic here]===UAC(b)---&gt;UAS(c) 
<DIV></DIV>&gt; &gt; 
<DIV></DIV>&gt; &gt; [RRR] I like to know who has been the called party when 
<DIV></DIV>&gt; &gt; UAC(a)) sent the INVITE message? Has it been UAS(b) or UAS(c)? 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;I'm going to first need to clarify your question. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;The question I will answer is: "If an INVITE is sent through the 
<DIV></DIV>&gt;network described in this diagram, what is the value of the 
<DIV></DIV>&gt;Request-URI when UAC(a) sends it?" 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;The answer is: it depends. In general, yes, it will probably be 
<DIV></DIV>&gt;either of UAS(b) or UAS(c). 
<DIV></DIV>&gt; 
<DIV></DIV>&gt; &gt; [RRR] If the answer is UAS(c), the question is: Why the call 
<DIV></DIV>&gt; &gt; has been routed to UAS(b) by UAC(a) in the first place? 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;The most obvious of several plausible answers is: UAC(a) 
<DIV></DIV>&gt;has UAS(b) configured as a local outbound proxy. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt; &gt; [RRR] If the answer is UAS(b), probably UAC(a) does not care. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;About what? 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;/a 
<DIV></DIV>&gt;_______________________________________________ 
<DIV></DIV>&gt;Sip mailing list https://www1.ietf.org/mailman/listinfo/sip 
<DIV></DIV>&gt;This list is for NEW development of the core SIP Protocol 
<DIV></DIV>&gt;Use sip-implementors@cs.columbia.edu for questions on current sip 
<DIV></DIV>&gt;Use sipping@ietf.org for new developments on the application of sip 
<DIV></DIV></div><br clear=all><hr>Add photos to your e-mail with <a href="http://g.msn.com/8HMIEN/2022">MSN 8.</a> Get 2 months FREE*.</html>
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From mailnull@www1.ietf.org  Wed Dec  4 20:09:41 2002
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        Adam Roach
	 <adam@dynamicsoft.com>, rrroy@att.com
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Subject: RE: [Sip] B2BUA & Max-Forwards
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You see, that's why I rephrased the question to be more
precise. The original question didn't make a whole lot of
sense, because the term "called party" is somewhat
counter-intuitive when applied to B2BUAs.

      Invitee, Invited User, Called Party, Callee: The party that
         receives an INVITE request for the purpose of establishing a
         new session.  A callee retains this role from the time it
         receives the INVITE until the termination of the dialog
         established by that INVITE.

In RFC3261 terms, the answer to Radhika's question, "who has been the
called party?" is "both UAS(b) and UAS(c)". They both receive the
INVITE, and establish a session. These sessions are, from a protocol
perspective, unrelated -- although they're obviously related by
some application logic.

But "both" would have been a flip answer, and it would be answering
a question that Radhika obviously did not intend to ask.

That notwithstanding, even using the colloquial definition you
seem to have in mind (which apparently has the constraint that
the "called party" device has a user sitting in front of it, or
has something to do with some personal desire held by the calling
party, or something like that), your assertion is demonstratably
false.

Take, for example, the situation in which User A (at UAC(a))
calls User B (at UAS(b)). They have a brief discussion, and
user B decides that he wants to add User C to the call;
using whatever interface is available, User B instructs his
client do so: User B (at UAC(b)) sends an INVITE request to
User C (at UAS(c)). User B decides to drop out of the call,
but instructs his device to keep the call up until one of
the parties disconnects.

So User B's device, which was obviously the called party in
the initial phase of the call (by either the 3261 or any
reasonable and most unreasonable colloquial definitions), is
rather clearly acting as a B2BUA.

/a

-----Original Message-----
From: Padma Suresh [mailto:padma_suresh@hotmail.com]
Sent: Wednesday, December 04, 2002 18:17
To: adam@dynamicsoft.com; rrroy@att.com
Cc: sip@ietf.org
Subject: RE: [Sip] B2BUA & Max-Forwards


For UAC (a) UAS (c) is the intended called party in the call. We have to
define the intended party in a call when a B2BUA is involved. B2BUA IS an
intermediate network element. It acts as some kind of a controller for a
domain. It is not the called party in any call. 
Regards
Padma
>From: Adam Roach 
>To: "'Roy, Radhika R, ALASO'" , Adam Roach 
>CC: sip@ietf.org 
>Subject: RE: [Sip] B2BUA & Max-Forwards 
>Date: Wed, 4 Dec 2002 16:58:14 -0600 
> 
> > UAC(a)--->UAS(b)===[application logic here]===UAC(b)--->UAS(c) 
> > 
> > [RRR] I like to know who has been the called party when 
> > UAC(a)) sent the INVITE message? Has it been UAS(b) or UAS(c)? 
> 
>I'm going to first need to clarify your question. 
> 
>The question I will answer is: "If an INVITE is sent through the 
>network described in this diagram, what is the value of the 
>Request-URI when UAC(a) sends it?" 
> 
>The answer is: it depends. In general, yes, it will probably be 
>either of UAS(b) or UAS(c). 
> 
> > [RRR] If the answer is UAS(c), the question is: Why the call 
> > has been routed to UAS(b) by UAC(a) in the first place? 
> 
>The most obvious of several plausible answers is: UAC(a) 
>has UAS(b) configured as a local outbound proxy. 
> 
> > [RRR] If the answer is UAS(b), probably UAC(a) does not care. 
> 
>About what? 
> 
>/a 
>_______________________________________________ 
>Sip mailing list https://www1.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: "Dean Willis" <dwillis@dynamicsoft.com>
To: <sip@ietf.org>
Date: Wed, 4 Dec 2002 19:23:23 -0600
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Subject: [Sip] Back-to-Back User Agent Ttransparency Top 10 list
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Apologies to those of you who have seen this before. It's a couple of
years old, which explains my frustration with this particular topic.

Note that "conformant" proxies satisfy all 10. Strict UAs satisfy none.

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

1) Dialog transparency: The call-ID is the same on "both sides" of the 
B2BUA. Proxies have this characteristic. A B2BUA that actually 
terminates a dialog on one side and generates a new dialog on the other 
doesn't. Some 3GPP AS functions fall into this category, as do 3PCC 
controllers.

2) Identity transparency: The user appears to have the same identity on 
both sides of the B2BUA. A proxy has this property. An identity 
anonymizer doesn't.

3) CSEQ transparency: The B2BUA does not alter CSEQ numbers. A 
traditional proxy has this property. A 3GPP P-CSCF does not, as it can 
generate a BYE message and then have to manage disjoint CSEQ spaces on 
each side.

4) Header transparency: The B2BUA doesn't change headers that transit 
it. Proxies have mostly this property, changing onlya few headers that 
are specifically changed by the proxy function. Proxies don't change 
headers they don't understand. 3GPP "proxies" are likely to go to town 
changing all SORTS of headers. All with the best of intentions, mind 
you.

5) Body transparency: The MIME bodies on the SIP message are not 
altered by the B2BUA. Real SIP proxies have this property. Firewall 
proxies that modify SDP, and most everything in 3GPP, doesn't.

6) Media transparency: AS with body transparency, but considering only 
the media aspects. The 3GPP P-CSCF breaks this transparency rule by 
editing one's SDP to meet operator preferences.

7) Topology transparency. Via, Route, Record-Route, Path, 
P-Service-Route, and other headers that reveal topology are roughly the 
same on both sides of the B2BUA, save for any which uniquely identify 
that B2BUA. Proxies have this characteristic. Topology hiding devices 
like the dynamicosft "firewall proxy in edge proxy mode" or the 3GPP 
THIG don't.

8) Security transparency: The B2BUA doesn't alter security aspects, 
such as encryptions, authorization headers, etc. Proxies generally have 
this property, but 3GPP stuff  doesn't.

9) Accounting transparency: Stuff used to generate records or track 
usage is not altered by a B2BUA with this property. Proxies have this 
property. 3GPP is trying to define a P-header (P-original-dialog-ID, I
believe) to provide for accounting transparency across 3GPP AS elements 
operating at a low level of transparency.

10) Functional transparency: The task that the user of the UA wants to 
accomplish actually works across the B2BUA, rather than being 
blocked/distorted by it. Proxies generally have this property, with the 
caveat the proxy can reject a request if needed. Some 3GPP entities can 
mutate functionality beyond all recognition. This is probably the 
single most important transparency concept. The rules of a "proxy" as 
defined in RFC3162 are there to preserve functional transparency, and 
anything that "breaks" those rules is likely to compromise 
functionality in some way.

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Pete Cordell'" <pete@tech-know-ware.com>, <sip@ietf.org>
Subject: RE: [Sip] B2BUA & Max-Forwards
Date: Wed, 4 Dec 2002 19:23:23 -0600
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> One issue that the recent B2BUA discussion has raised in my 
> mind is what a B2BUA (whatever its decided that should be) 
> should do with Max-Forwards.

It does the same thing as it does with any other header or body part,
which is the same thing that ANY 500-pound gorilla does to somebody's
body parts -- whatever it wants.

What you're asking is "what is the transparency of this node with
respect to max-forwards", which is "#4 -- Header Transaprency" in my
top-10 list. The specification of a back-to-back user agent doesn't
mandate any level of transparency on this or any other header.

--
Dean

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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Subject: RE: [Sip] Back-to-Back User Agent Ttransparency Top 10 list
Date: Wed, 4 Dec 2002 19:42:49 -0600
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By the way, 3GPP proxies are much more conformant now than they were
when I first posted this, so please don't beat them up about anything
here. They've done a great job improving the transparency of their
system. Thank you, 3GPP!

--
Dean

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Dean Willis
> Sent: Wednesday, December 04, 2002 7:23 PM
> To: sip@ietf.org
> Subject: [Sip] Back-to-Back User Agent Ttransparency Top 10 list
> 
> 
> 
> Apologies to those of you who have seen this before. It's a 
> couple of years old, which explains my frustration with this 
> particular topic.
> 
> Note that "conformant" proxies satisfy all 10. Strict UAs 
> satisfy none.
> 
> ----------------------------
> 
> 1) Dialog transparency: The call-ID is the same on "both 
> sides" of the 
> B2BUA. Proxies have this characteristic. A B2BUA that actually 
> terminates a dialog on one side and generates a new dialog on 
> the other 
> doesn't. Some 3GPP AS functions fall into this category, as do 3PCC 
> controllers.
> 
> 2) Identity transparency: The user appears to have the same 
> identity on 
> both sides of the B2BUA. A proxy has this property. An identity 
> anonymizer doesn't.
> 
> 3) CSEQ transparency: The B2BUA does not alter CSEQ numbers. A 
> traditional proxy has this property. A 3GPP P-CSCF does not, 
> as it can 
> generate a BYE message and then have to manage disjoint CSEQ 
> spaces on 
> each side.
> 
> 4) Header transparency: The B2BUA doesn't change headers that transit 
> it. Proxies have mostly this property, changing onlya few 
> headers that 
> are specifically changed by the proxy function. Proxies don't change 
> headers they don't understand. 3GPP "proxies" are likely to 
> go to town 
> changing all SORTS of headers. All with the best of intentions, mind 
> you.
> 
> 5) Body transparency: The MIME bodies on the SIP message are not 
> altered by the B2BUA. Real SIP proxies have this property. Firewall 
> proxies that modify SDP, and most everything in 3GPP, doesn't.
> 
> 6) Media transparency: AS with body transparency, but 
> considering only 
> the media aspects. The 3GPP P-CSCF breaks this transparency rule by 
> editing one's SDP to meet operator preferences.
> 
> 7) Topology transparency. Via, Route, Record-Route, Path, 
> P-Service-Route, and other headers that reveal topology are 
> roughly the 
> same on both sides of the B2BUA, save for any which uniquely identify 
> that B2BUA. Proxies have this characteristic. Topology hiding devices 
> like the dynamicosft "firewall proxy in edge proxy mode" or the 3GPP 
> THIG don't.
> 
> 8) Security transparency: The B2BUA doesn't alter security aspects, 
> such as encryptions, authorization headers, etc. Proxies 
> generally have 
> this property, but 3GPP stuff  doesn't.
> 
> 9) Accounting transparency: Stuff used to generate records or track 
> usage is not altered by a B2BUA with this property. Proxies have this 
> property. 3GPP is trying to define a P-header (P-original-dialog-ID, I
> believe) to provide for accounting transparency across 3GPP 
> AS elements 
> operating at a low level of transparency.
> 
> 10) Functional transparency: The task that the user of the UA 
> wants to 
> accomplish actually works across the B2BUA, rather than being 
> blocked/distorted by it. Proxies generally have this 
> property, with the 
> caveat the proxy can reject a request if needed. Some 3GPP 
> entities can 
> mutate functionality beyond all recognition. This is probably the 
> single most important transparency concept. The rules of a "proxy" as 
> defined in RFC3162 are there to preserve functional transparency, and 
> anything that "breaks" those rules is likely to compromise 
> functionality in some way.
> 
> _______________________________________________
> Sip mailing list  https://www1.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: "Dean Willis" <dean.willis@softarmor.com>
To: "'Phil Reynolds'" <preynolds@ridgewaysystems.com>, <sip@ietf.org>
Subject: RE: [Sip] B2BUA
Date: Wed, 4 Dec 2002 19:42:49 -0600
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-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Phil
Reynolds
Sent: Wednesday, December 04, 2002 7:37 AM
To: sip@ietf.org
Subject: RE: [Sip] B2BUA


If a telephone switch crashes you loose all calls...  The telecoms
people are clever enough to have thought about that and build
redundancy, fault tolerance and failover into their switches.
Maybe some of us IP-telecoms people have thought of similar techniques
for SIP intermediaries?   
-----

Yes. We call them "proxies".

--
Dean

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From mailnull@www1.ietf.org  Wed Dec  4 21:10:10 2002
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Subject: Re: [Sip] B2BUA & Max-Forwards
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> -----Original Message-----
> From: Kedar Patil [mailto:kpatil@sta.samsung.com]
> 
> > This discussion raises one more doubt for me, pls. clarify. 
> >
> > UA1------->BtoBUA---------->UA2 
> >
> > Is the BtoBUA a SIP endpoint from UA1's point of view?
> 
> Yes.
> 
I'm confused. Isn't the "BtoBUA" a concept of the combination of UACs and UASs? I think from the UA1's point of view, it doesn't know whether its peer is a B2BUA or a normal single UA. 
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Subject: RE: [Sip] B2BUA & Max-Forwards
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> -----Original Message-----
> From: Nicky Song [mailto:nicky.song@utstar.com]
> 
> > -----Original Message-----
> > From: Kedar Patil [mailto:kpatil@sta.samsung.com]
> > 
> > > This discussion raises one more doubt for me, pls. clarify. 
> > >
> > > UA1------->BtoBUA---------->UA2 
> > >
> > > Is the BtoBUA a SIP endpoint from UA1's point of view?
> > 
> > Yes.
> > 
> I'm confused. Isn't the "BtoBUA" a concept of the combination 
> of UACs and UASs? I think from the UA1's point of view, it 
> doesn't know whether its peer is a B2BUA or a normal single UA. 

Yep. It doesn't know whether its peer is a B2BUA or a "normal" UA.
So, it's a SIP endpoint from UA1's point of view.

It's not clear to me where the confusion comes in.

/a
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From: "Arunachalam Venkatraman" <arunvenk@cisco.com>
To: <sip@ietf.org>
Subject: RE: [Sip] Detecting merged requests?
Date: Wed, 4 Dec 2002 22:39:03 -0600
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Sachin
Your interpretation is correct. The request is not considered merged, unless
the ReqUri matches that on a previously received request on the same calId
(in addition to other headers specified in the RFC).
ReqUri is a primary parameter in a merged request. The other criteria come
into play only when the request uri indicates this as a possibility. In
transaction matching rules, used to detect retransmissions/merged requests
ReqUri, VIA, From Tag/ToTag, CSEQ and CallId are all used.

The correct thing for the two UASs to do is respond with a different To Tag
on each request and proceed to set up the calls independently(This is
distinct from 2 legs on the same call.) This scenario is also referred to,
loosely, as UAS forking.

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Sachin
Shenoy
Sent: Monday, December 02, 2002 5:10 AM
To: sip@ietf.org
Subject: [Sip] Detecting merged requests?


Hi,

According to RFC 3261,  a request is a merged request if ...

"8.2.2.2 Merged Requests

    If the request has no tag in the To header field, the UAS core MUST
    check the request against ongoing transactions.  If the From tag,
    Call-ID, and CSeq exactly match those associated with an ongoing
    transaction, but the request does not match that transaction (based
    on the matching rules in Section 17.2.3), the UAS core SHOULD
    generate a 482 (Loop Detected) response and pass it to the server
    transaction."

Suppose if there are 2 UAS application sharing the same SIP stack
(same process). They are listening for incoming calls for the same
user. One application is handling the fax machine for user_a and the
other the phone for user_a.

user_a's fax machine can be reached at  fax_user_a@test.com
And user_a's phone can be reached at    phone_user_a@test.com


A call made to user_a@test.com gets forked at a proxy. The
2 forked request arrives at the UAS with the respective request
URI's

INVITE sip:fax_user_a@test.com
...

INVITE sip:phone_user_a@test.com
...

Now, according to the RFC, the first INVITE would be processed and the
second would be responded with 482 "loop detected". Shouldn't the
second INVITE also be processed? Logically, these are 2 different
application handling 2 different devices for the same user. The fact that
they are sharing the same process should not interfere each other.

I think merged request matching should also take the Request-Uri
into consideration. Request-Uri  identifies the end point at UAS. If
a UAS gets 2 INVITES  for the same request-uri with the same Call-ID,
Cseq and From-Tag, and with no To-Tag then it should be considered
as merged request.

Thanks
Sachin

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From: "Padma Suresh" <padma_suresh@hotmail.com>
To: dwillis@dynamicsoft.com, sip@ietf.org
Subject: Re: [Sip] Back-to-Back User Agent Ttransparency Top 10 list
Date: Thu, 05 Dec 2002 06:16:13 +0000
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<html><div style='background-color:'><DIV>
<P>"4) Header transparency: The B2BUA doesn't change headers that transit it." </P>
<P>This mail seems to suggest that a B2BUA forwards the headers to the other endpoint (in other words does not change the headers that transit it) .</P>
<P>Am I wrong in this assumption?</P>
<P>The content of this mail would definitely be pretty useful in a draft on the B2BUA.</P>
<P>Regards</P>
<P>Padma<BR><BR></P></DIV>
<DIV></DIV>
<DIV></DIV>&gt;From: "Dean Willis" <DWILLIS@DYNAMICSOFT.COM>
<DIV></DIV>&gt;To: <SIP@IETF.ORG>
<DIV></DIV>&gt;Subject: [Sip] Back-to-Back User Agent Ttransparency Top 10 list 
<DIV></DIV>&gt;Date: Wed, 4 Dec 2002 19:23:23 -0600 
<DIV></DIV>&gt; 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;Apologies to those of you who have seen this before. It's a couple of 
<DIV></DIV>&gt;years old, which explains my frustration with this particular topic. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;Note that "conformant" proxies satisfy all 10. Strict UAs satisfy none. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;---------------------------- 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;1) Dialog transparency: The call-ID is the same on "both sides" of the 
<DIV></DIV>&gt;B2BUA. Proxies have this characteristic. A B2BUA that actually 
<DIV></DIV>&gt;terminates a dialog on one side and generates a new dialog on the other 
<DIV></DIV>&gt;doesn't. Some 3GPP AS functions fall into this category, as do 3PCC 
<DIV></DIV>&gt;controllers. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;2) Identity transparency: The user appears to have the same identity on 
<DIV></DIV>&gt;both sides of the B2BUA. A proxy has this property. An identity 
<DIV></DIV>&gt;anonymizer doesn't. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;3) CSEQ transparency: The B2BUA does not alter CSEQ numbers. A 
<DIV></DIV>&gt;traditional proxy has this property. A 3GPP P-CSCF does not, as it can 
<DIV></DIV>&gt;generate a BYE message and then have to manage disjoint CSEQ spaces on 
<DIV></DIV>&gt;each side. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;4) Header transparency: The B2BUA doesn't change headers that transit 
<DIV></DIV>&gt;it. Proxies have mostly this property, changing onlya few headers that 
<DIV></DIV>&gt;are specifically changed by the proxy function. Proxies don't change 
<DIV></DIV>&gt;headers they don't understand. 3GPP "proxies" are likely to go to town 
<DIV></DIV>&gt;changing all SORTS of headers. All with the best of intentions, mind 
<DIV></DIV>&gt;you. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;5) Body transparency: The MIME bodies on the SIP message are not 
<DIV></DIV>&gt;altered by the B2BUA. Real SIP proxies have this property. Firewall 
<DIV></DIV>&gt;proxies that modify SDP, and most everything in 3GPP, doesn't. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;6) Media transparency: AS with body transparency, but considering only 
<DIV></DIV>&gt;the media aspects. The 3GPP P-CSCF breaks this transparency rule by 
<DIV></DIV>&gt;editing one's SDP to meet operator preferences. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;7) Topology transparency. Via, Route, Record-Route, Path, 
<DIV></DIV>&gt;P-Service-Route, and other headers that reveal topology are roughly the 
<DIV></DIV>&gt;same on both sides of the B2BUA, save for any which uniquely identify 
<DIV></DIV>&gt;that B2BUA. Proxies have this characteristic. Topology hiding devices 
<DIV></DIV>&gt;like the dynamicosft "firewall proxy in edge proxy mode" or the 3GPP 
<DIV></DIV>&gt;THIG don't. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;8) Security transparency: The B2BUA doesn't alter security aspects, 
<DIV></DIV>&gt;such as encryptions, authorization headers, etc. Proxies generally have 
<DIV></DIV>&gt;this property, but 3GPP stuff doesn't. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;9) Accounting transparency: Stuff used to generate records or track 
<DIV></DIV>&gt;usage is not altered by a B2BUA with this property. Proxies have this 
<DIV></DIV>&gt;property. 3GPP is trying to define a P-header (P-original-dialog-ID, I 
<DIV></DIV>&gt;believe) to provide for accounting transparency across 3GPP AS elements 
<DIV></DIV>&gt;operating at a low level of transparency. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;10) Functional transparency: The task that the user of the UA wants to 
<DIV></DIV>&gt;accomplish actually works across the B2BUA, rather than being 
<DIV></DIV>&gt;blocked/distorted by it. Proxies generally have this property, with the 
<DIV></DIV>&gt;caveat the proxy can reject a request if needed. Some 3GPP entities can 
<DIV></DIV>&gt;mutate functionality beyond all recognition. This is probably the 
<DIV></DIV>&gt;single most important transparency concept. The rules of a "proxy" as 
<DIV></DIV>&gt;defined in RFC3162 are there to preserve functional transparency, and 
<DIV></DIV>&gt;anything that "breaks" those rules is likely to compromise 
<DIV></DIV>&gt;functionality in some way. 
<DIV></DIV>&gt; 
<DIV></DIV>&gt;_______________________________________________ 
<DIV></DIV>&gt;Sip mailing list https://www1.ietf.org/mailman/listinfo/sip 
<DIV></DIV>&gt;This list is for NEW development of the core SIP Protocol 
<DIV></DIV>&gt;Use sip-implementors@cs.columbia.edu for questions on current sip 
<DIV></DIV>&gt;Use sipping@ietf.org for new developments on the application of sip 
<DIV></DIV></div><br clear=all><hr> <a href="http://g.msn.com/8HMHEN/2015">the new MSN 8</a> and get 2 months FREE*</html>
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Adam Roach wrote:
> -----Original Message-----
> From: Phil Reynolds [mailto:preynolds@ridgewaysystems.com]
> 
> 
>>Now your back to restricting what I can and can't do in an
>>intermediary. It can't be a B2BUA 'cos it doesn't need to
>>honour require headers (the UA at the other end does that,
>>if they were meant for my intermediary then they would be
>>proxy-require?), and it can't be a proxy 'cos of the rules
>>about not changing the body so what is it?
> 
> 
> Broken.

Adam is correct.

> 
> The restrictions placed on nodes by 3261 (which I'm merely
> reiterating, not imposing) aren't an authoritarian attempt
> at unnecessary proscription. They serve a purpose.
> 
> I'll describe one of the (several) purposes that applies
> to the situation you propose.
> 
> One of the things that UAs are required to be able to do is
> sign and/or encrypt the bodies and/or some subset of the
> headers to ensure confidentiality and/or integrity. The
> protocol places restrictions on various nodes to ensure
> that this can be done. Your box is broken precisely because
> it interferes with these abilities (or, more precisely,
> these abilities interfere with it).

Let me add some more here.

Anything that deviates at all from proxy rules will need to be a b2bua 
and follow b2bua rules, including processing of Require. There is no 
black-or-white here. Let me give some examples.

Lets say my b2bua is a proxy in all ways, BUT it happens to muck with 
SDP to enable firewall traversal. Surely then it can ignore Require, 
right? WRONG. Lets say I define extension foo, which means "the content 
of this SDP is not what it seems - look at this header to instead find 
the real port/address for media". This extension needs a Require 
(because only a UA ever looks at bodies), but not a Proxy-Require, 
because proxies don't. Thus, your almost-a-proxy will fail in odd ways 
because it has ignored Require when it shouldn't have.

Indeed, for any operation prohibited to a proxy, but allowed for a UA, I 
can come up with an extension, indicated in a Require header, that would 
need to be understood by a b2bua which breaks the proxy rules and 
performs this operation. Indeed, the rules on what is a UA, and what is 
a proxy, are meant precisely to help define what goes in Require, and 
what goes in Proxy-Require.

Whether or not Require gets passed on by such a b2bua depends entirely 
on the application logic.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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William Marshall wrote:
> Dean wrote:
> 
>>I think that Bill is suggesting that there is also something BETWEEN the
>>B2BUA and the conformant proxy, and that the world would be a better
>>place (ok, maybe not THAT much better) if our taxonomy also included
>>this "between" thing.
> 
> 
> To further elaborate, there certainly exists on paper a box that does
> some abritrary and weird algorithm, not conformant with any of the
> rules in rfc3261, that transforms requests and response and breaks
> everything that goes through it.  Call it "Dean's devilish-B2BUA".
> 
> But I don't think anyone builds such a box.  At least not intentionally.
> 
> What is built instead are boxes that are close to the definition of
> proxies, but not quite.  They serve a real need in the market (and
> vendors tend to build what sells).  But due to various details, like
> looking at the attachment (a proxy-no-no) to open pinholes in the firewall,
> they don't strictly qualify as proxies.
> 
> Saying there are only two types of boxes, proxies and devilish-B2BUAs,
> ignores the reality.
> 
> The writing of an accurate definition of a B2BUA is only the first step.

No.

If, indeed, there is a strong need for elements that are ALMOST proxies, 
but are doing X,Y, and Z things that proxies cannot, there are two 
approaches. The first approach is, as you propose, to presume the 
existence of a new type of element and go about defining it. The second, 
is to consider how to add that capability to the repetoire of things a 
proxy can, in fact, legally do.

As an example, a very common offense is mucking with SDP, for example, 
to implement a firewall control proxy, ala midcom. Instead of defining a 
new element, called an SDP-munging proxy, and dealing with all the 
horrible security implications that will arise from it, I would propose 
to simply FIX THE PROBLEM and define a means by which a proxy can impose 
session policies without breaking SIP. Indeed, I wrote a draft on just 
such a topic:

http://www.jdrosen.net/papers/draft-rosenberg-sipping-session-policy-00.txt

Perhaps if you could list what is it that a "transparent b2bua" needs to 
do that a proxy cannot, we can figure out how to fix the problem instead 
of defining an entirely new protocol element.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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Done. Logged as:
http://www.sipwg.org/sipwg/show_bug.cgi?id=679

I would request, though, that service route at least call out that this 
issue is exacerbated by service route, because of the underlying 
assumption of intervening proxies between the client and registrar.

-Jonathan R.

Peterson, Jon wrote:
> I think that's quite reasonable.
> 
> Jon Peterson
> NeuStar, Inc.
> 
> 
>>-----Original Message-----
>>From: Adam Roach [mailto:adam@dynamicsoft.com]
>>Sent: Wednesday, December 04, 2002 2:45 PM
>>To: 'Peterson, Jon'; 'sip@ietf.org'
>>Subject: RE: [Sip] Discuss on service discovery
>>
>>
>>... 
>>
>>>Can we go forward with the requested fix?
>>
>>In light of the conclusions of the previous discussions on the
>>topic, I have no further objections.
>>
>>I would, however, like to see a bug opened against 3261 to
>>address this issue so that the solution can eventually appear
>>in a centralized place.
>>
>>/a
>>
> 
> _______________________________________________
> Sip mailing list  https://www1.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
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> 

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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inline.

Robert Sparks wrote:
> On Wed, 2002-12-04 at 10:11, Anders Kristensen wrote:
> 
>>
>>Robert Sparks wrote:
>>
>>>On Wed, 2002-12-04 at 09:54, Anders Kristensen wrote:
>>>
>>>
>>>>Hi Robert,
>>>>
>>>>Sounds like a very good idea. One special case to consider is what 
>>>>happens if the request spirals back to the sending proxy (which 
>>>>presumably is stateful) -- either directly from the stateless proxy or 
>>>>via a number of other stateless proxies.  The stateful proxy in this 
>>>>case will have a server transaction and a client transaction with the 
>>>>same id, which might confuse some implementations.
>>>
>>>
>>>This is no different from the same proxy spiraling directly to itself
>>>is it?
>>
>>Well, it's different in that it is NOT spiraling directly to itself and 
>>so is currently guaranteed to see a different branch ID in a spiralled 
>>request. It may be the case that noone depends on this but who knows?
> 
> 
> Ok - so an implementation _may_ have gone to some effort to ensure
> that it wasn't sending a message to itself and is counting on not
> seeing the same transaction identifier come back to it. Not a choice
> I would recommend, but I can see how someone might have made it.
> 
> We could avoid this with the suffix mechanism you suggest, or we
> could add an admonishment to the spec to tell systems to be prepared
> to have their transaction ID come back. I'd rather do the second.

I agree with Robert. Over time, our view of stateless proxies has 
changed a lot. Initially, we tried to treat them as the same as stateful 
ones, but it has become clear that there are important differences (for 
example, there are restrictions on the types of routing logic that are 
permitted).

One of the big changes in bis was the formal definition of a transaction 
  as being between a client transaction in one element and a server 
transaction in another (figure 4). Whats important is that the 
transaction is between transaction-stateful elements (by definition). It 
would seem very logical, then, that the transaction identifier (the 
branch ID) would be the same for all messages in this transaction. Since 
a stateless proxy doesn't muck with the transaction, just forwards 
messages, I would argue that the transaction ID should stay the same.

Indeed, the analogy is similar to the dialog identifiers in a stateful 
proxy; they too are not modified, since the proxy doesn't act as a 
termination point of the dialog. If it did (b2bua), it would need to 
modify those identifiers.


> 
> Well, nothing earth-shattering. Right now, just an increased probability
> that implementation error will lead to protocol failure. Its hard to get
> this right, and we can greatly simplify it.
> 
> My growing distaste for what the stateless proxies are currently doing
> comes from seeing it somewhat like a NAT. The transaction is really 
> between the two stateful elements reguardless of how many stateless
> elements lie between. Right now, the stateless elements are 
> "translating" the transaction identifier as the message goes through.
> We haven't identified anything in the protocol that breaks as long
> as they do this consistently, but I haven't been able to identify any
> benefit to the behavior. 

I agree. I think its really legacy at this point.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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By the way, I am now tracking this in bugzilla:
http://www.sipwg.org/sipwg/show_bug.cgi?id=680

-Jonathan R.

Robert Sparks wrote:
> On Wed, 2002-12-04 at 10:11, Anders Kristensen wrote:
> 
>>
>>Robert Sparks wrote:
>>
>>>On Wed, 2002-12-04 at 09:54, Anders Kristensen wrote:
>>>
>>>
>>>>Hi Robert,
>>>>
>>>>Sounds like a very good idea. One special case to consider is what 
>>>>happens if the request spirals back to the sending proxy (which 
>>>>presumably is stateful) -- either directly from the stateless proxy or 
>>>>via a number of other stateless proxies.  The stateful proxy in this 
>>>>case will have a server transaction and a client transaction with the 
>>>>same id, which might confuse some implementations.
>>>
>>>
>>>This is no different from the same proxy spiraling directly to itself
>>>is it?
>>
>>Well, it's different in that it is NOT spiraling directly to itself and 
>>so is currently guaranteed to see a different branch ID in a spiralled 
>>request. It may be the case that noone depends on this but who knows?
> 
> 
> Ok - so an implementation _may_ have gone to some effort to ensure
> that it wasn't sending a message to itself and is counting on not
> seeing the same transaction identifier come back to it. Not a choice
> I would recommend, but I can see how someone might have made it.
> 
> We could avoid this with the suffix mechanism you suggest, or we
> could add an admonishment to the spec to tell systems to be prepared
> to have their transaction ID come back. I'd rather do the second.
> 
> 
>>>
>>>>This can be addressed simply by appending a fixed string to the branch 
>>>>value at each stateless proxy doing this trick, e.g. a single character. 
>>>>This guarantees the uniqueness properties required by rfc 3261, which 
>>>>seems desirable.
>>>
>>>
>>>There are several things we could do here - append the count of present
>>>via header fields for example to make sure you get different values as
>>>you spiral through the stateless elements, etc. I don't think any of
>>>this is necessary though - the stateless elements are supposed to be
>>>transaction invisible. If they _can_ do something that botches the
>>>transaction, we've got a weak spot in the protocol. I propose that
>>>preserving the transaction identifier provided by the stateful element
>>>is exactly the right thing to do, and that any mucking with it brings
>>>nothing good to the table, and has the potential to bring harm.
>>
>>Like what?
> 
> 
> Well, nothing earth-shattering. Right now, just an increased probability
> that implementation error will lead to protocol failure. Its hard to get
> this right, and we can greatly simplify it.
> 
> My growing distaste for what the stateless proxies are currently doing
> comes from seeing it somewhat like a NAT. The transaction is really 
> between the two stateful elements reguardless of how many stateless
> elements lie between. Right now, the stateless elements are 
> "translating" the transaction identifier as the message goes through.
> We haven't identified anything in the protocol that breaks as long
> as they do this consistently, but I haven't been able to identify any
> benefit to the behavior. 
> 
> Now I'll go out on a limb. In the future, we might want as part of 
> diagnostics, error recovery, or some similar function, to be able
> to refer to the transaction by its identifier. It would be very
> nice if that identifier were the same at the stateful elements on
> each end of the transaction. 
> 
> 
>>Anders
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.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.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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Subject: Re: [Sip] response to merged requests
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I don't think there was an explicit decision to not define a new 
response code. The error condition seemed close enough to a loop (its 
another topology failure) that it was reasonable to reuse the code.

-Jonathan R.

Ravi Shiroor wrote:
> posting this on sip as there were no responses on sip-implementors.
> 
> --------->
> 
> section 8.2.2.2 in RFC 3261 says that when a UAS receives
> merged requests, it should respond to the INVITEs receieved
> after the first one (those detected as merged) with 482 (loop detected).
> 
> i was wondering why "loop detected" is being being used when
> merged requests are receieved? wouldnt it be appropriate to
> define a specific 4xx response for merged requests?
> 
> can somebody please clarify if there is a valid reason for
> using "loop detected" in this scenario?
> 
> TIA,
> ravi.
> 
> _______________________________________________
> Sip mailing list  https://www1.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.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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Remember, SIP talks about logical elements, and doesn't care about 
specific implementations. In your example, there are really two separate 
user agents, in the sense that there are two separate entities making 
decisions about whether to accept or reject calls. That these happen to 
share an internal stack and port is a matter of implementation. As such, 
  according to the spec, they would be treated as separate UA and thus 
not subject to the merged request rule. That rule is meant to deal 
explicitly with the case of a single UA, in the logical sense above, 
getting two requests. The issue is how a particular call acceptance 
decision - accept, reject, redirect - manifests itself in terms of 
responses across multiple transactions. That is a non-issue in your case.

-Jonathan R.

Sachin Shenoy wrote:
> Hi,
> 
> According to RFC 3261,  a request is a merged request if ...
> 
> "8.2.2.2 Merged Requests
> 
>    If the request has no tag in the To header field, the UAS core MUST
>    check the request against ongoing transactions.  If the From tag,
>    Call-ID, and CSeq exactly match those associated with an ongoing
>    transaction, but the request does not match that transaction (based
>    on the matching rules in Section 17.2.3), the UAS core SHOULD
>    generate a 482 (Loop Detected) response and pass it to the server
>    transaction."
> 
> Suppose if there are 2 UAS application sharing the same SIP stack
> (same process). They are listening for incoming calls for the same
> user. One application is handling the fax machine for user_a and the
> other the phone for user_a.
> 
> user_a's fax machine can be reached at  fax_user_a@test.com
> And user_a's phone can be reached at    phone_user_a@test.com
> 
> 
> A call made to user_a@test.com gets forked at a proxy. The
> 2 forked request arrives at the UAS with the respective request
> URI's
> 
> INVITE sip:fax_user_a@test.com
> ...
> 
> INVITE sip:phone_user_a@test.com
> ...
> 
> Now, according to the RFC, the first INVITE would be processed and the
> second would be responded with 482 "loop detected". Shouldn't the
> second INVITE also be processed? Logically, these are 2 different
> application handling 2 different devices for the same user. The fact that
> they are sharing the same process should not interfere each other.
> 
> I think merged request matching should also take the Request-Uri
> into consideration. Request-Uri  identifies the end point at UAS. If
> a UAS gets 2 INVITES  for the same request-uri with the same Call-ID,
> Cseq and From-Tag, and with no To-Tag then it should be considered
> as merged request.
> 
> Thanks
> Sachin
> 
> _______________________________________________
> Sip mailing list  https://www1.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.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Padma asked:

> "4) Header transparency: The B2BUA doesn't change headers that transit
it."
>
> This mail seems to suggest that a B2BUA forwards the headers to the
other endpoint (in other words does not change the headers that transit
it) .
>
> Am I wrong in this assumption?
>
> The content of this mail would definitely be pretty useful in a draft on
the B2BUA.

A B2BUA that has the propoerty of header transparency would not change
headers, as said. However, not all B2BUAs have such a property. For
example, the 3GPP Topology Hiding Internetwork Gateway,  or THIG,
transforms headers that reveal topology information about the operator
network. The DCS/PacketCable "Anonymizer" transforms headers that reveal
the identity or location of a caller. Even a basic proxy actually changes
some headers, for example Via and possibly Route and Record-Route.

--
Dean

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Subject: Re: [Sip] B2BUA & Max-Forwards
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: Adam Roach <adam@dynamicsoft.com>,
        "'Phil Reynolds'" <preynolds@ridgewaysystems.com>, sip@ietf.org
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Hello,
Fully agree with you.
Indeed, that's what should be explicitly added to the B2BUA definition in
3261, to avoid confusion:
"Anything that deviates at all from proxy rules will need to be a b2bua and
follow b2bua rules"
"...a b2bua depends entirely on the application logic"
All the debate is around these two notions.
Best regards
Juan Carlos





Jonathan Rosenberg <jdrosen@dynamicsoft.com>@ietf.org on 05/12/2002
07:37:05

Sent by:  sip-admin@ietf.org


To:   Adam Roach <adam@dynamicsoft.com>
cc:   "'Phil Reynolds'" <preynolds@ridgewaysystems.com>, sip@ietf.org
Subject:  Re: [Sip] B2BUA & Max-Forwards


inline.

Adam Roach wrote:
> -----Original Message-----
> From: Phil Reynolds [mailto:preynolds@ridgewaysystems.com]
>
>
>>Now your back to restricting what I can and can't do in an
>>intermediary. It can't be a B2BUA 'cos it doesn't need to
>>honour require headers (the UA at the other end does that,
>>if they were meant for my intermediary then they would be
>>proxy-require?), and it can't be a proxy 'cos of the rules
>>about not changing the body so what is it?
>
>
> Broken.

Adam is correct.

>
> The restrictions placed on nodes by 3261 (which I'm merely
> reiterating, not imposing) aren't an authoritarian attempt
> at unnecessary proscription. They serve a purpose.
>
> I'll describe one of the (several) purposes that applies
> to the situation you propose.
>
> One of the things that UAs are required to be able to do is
> sign and/or encrypt the bodies and/or some subset of the
> headers to ensure confidentiality and/or integrity. The
> protocol places restrictions on various nodes to ensure
> that this can be done. Your box is broken precisely because
> it interferes with these abilities (or, more precisely,
> these abilities interfere with it).

Let me add some more here.

Anything that deviates at all from proxy rules will need to be a b2bua
and follow b2bua rules, including processing of Require. There is no
black-or-white here. Let me give some examples.

Lets say my b2bua is a proxy in all ways, BUT it happens to muck with
SDP to enable firewall traversal. Surely then it can ignore Require,
right? WRONG. Lets say I define extension foo, which means "the content
of this SDP is not what it seems - look at this header to instead find
the real port/address for media". This extension needs a Require
(because only a UA ever looks at bodies), but not a Proxy-Require,
because proxies don't. Thus, your almost-a-proxy will fail in odd ways
because it has ignored Require when it shouldn't have.

Indeed, for any operation prohibited to a proxy, but allowed for a UA, I
can come up with an extension, indicated in a Require header, that would
need to be understood by a b2bua which breaks the proxy rules and
performs this operation. Indeed, the rules on what is a UA, and what is
a proxy, are meant precisely to help define what goes in Require, and
what goes in Proxy-Require.

Whether or not Require gets passed on by such a b2bua depends entirely
on the application logic.

-Jonathan R.

--
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Subject: RE: [Sip] B2BUA & Max-Forwards
To: Adam Roach <adam@dynamicsoft.com>
Cc: "'Padma Suresh'" <padma_suresh@hotmail.com>, vkg@lucent.com,
        pete@tech-know-ware.com, sip@ietf.org
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Adam,

I don't think there is any disagreement.

I said that "However, you can say that the application "x" behaves as a
B2BUA, and that the application "x" "needs to propagate" the a-header or
the b-header etc. for the needs of the application. This is the choice of
the application"
So, I wanted to stress the fact that once again the behaviour of the B2BUA
is depending on the application logic.

Strictly speaking, I should have not said that the "application "x" needs
to propagate" the a-header, etc.
Instead, one should say that the application "x" choice is to generate the
a-header on the UAC side with the same content of the a-header received in
the UAS side.
This means that the SIP dialog on the UAS side is terminated and that the
B2BUA is generating another dialog on the UAC side.
The criteria used by the application "x" to fill the a-header are strictly
depending on the application logic, and this can be applied to any SIP
header.

Best regards
Juan Carlos





Adam Roach <adam@dynamicsoft.com> on 04/12/2002 21:55:15

To:   Juan-Carlos ROJAS/FR/ALCATEL@ALCATEL, Adam Roach
      <adam@dynamicsoft.com>
cc:   "'Padma Suresh'" <padma_suresh@hotmail.com>, vkg@lucent.com,
      pete@tech-know-ware.com, sip@ietf.org
Subject:  RE: [Sip] B2BUA & Max-Forwards


Unless you were merely trying to agree at length with the
final paragraph of my previous message, I'm afraid that
you missed my point rather broadly.

I'll reiterate the main point concisely: a B2BUA is a UAS
attached to a UAC by application logic.

Read that carefully: Part of a B2BUA is a UAS.

That UAS is bound by RFC 3261 to certain behaviors.

One of those behaviors is honoring "Require" headers.




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From: "Ravi Shiroor" <rshiroor@aplion.stpn.soft.net>
To: "Adam Roach" <adam@dynamicsoft.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] B2BUA & Max-Forwards
Date: Thu, 5 Dec 2002 15:13:04 +0530
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> Take, for example, the situation in which User A (at UAC(a))
> calls User B (at UAS(b)). They have a brief discussion, and
> user B decides that he wants to add User C to the call;
> using whatever interface is available, User B instructs his
> client do so: User B (at UAC(b)) sends an INVITE request to
> User C (at UAS(c)). User B decides to drop out of the call,
> but instructs his device to keep the call up until one of
> the parties disconnects.
>
> So User B's device, which was obviously the called party in
> the initial phase of the call (by either the 3261 or any
> reasonable and most unreasonable colloquial definitions), is
> rather clearly acting as a B2BUA.
>
> /a
>

Looking at the above example and reading the discussions on
this thread I am wondering why SIP RFC should even try to name
or give definitions to such a device.

I feel that SIP should only define behaviour of UACs,
UASs and proxies; other intelliginet(?) devices are built
using these as its communication interfaces should be out of
scope of SIP. SIP RFC does say that the intelligence built
in these devices are not in its scope, but it still tries
to give a broad definition of the term B2BUA. I feel that
also should not be done. It should restrict itself to
defining terms which are used in the RFC to define the
protocol itself.

regards,
ravi.



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From: Phil Reynolds <preynolds@ridgewaysystems.com>
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Subject: RE: [Sip] B2BUA & Max-Forwards
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Our system isn't broken. I would say that this shows that the spec is broken
not our system.

Our system provides facilities for some (and we hope "most") users. We don't
provide these facilities for all users because: A) We don't think they are
important for most users, B) The spec precludes us from doing so (e.g.
end-to-end security)

But because the spec says we MUST do end-to-end security we can't be
sanctioned as compliant, despite the fact that "most" of our users don't
want that. (BTW I have yet to see a endpoint that does end-to-end security
so it appears that these are all broken as well)

-----Original Message-----
From: Adam Roach [mailto:adam@dynamicsoft.com] 
Sent: 04 December 2002 20:45
To: 'Phil Reynolds'; sip@ietf.org
Subject: RE: [Sip] B2BUA & Max-Forwards


-----Original Message-----
From: Phil Reynolds [mailto:preynolds@ridgewaysystems.com]

> Now your back to restricting what I can and can't do in an 
> intermediary. It can't be a B2BUA 'cos it doesn't need to honour 
> require headers (the UA at the other end does that, if they were meant 
> for my intermediary then they would be proxy-require?), and it can't 
> be a proxy 'cos of the rules about not changing the body so what is 
> it?

Broken.

The restrictions placed on nodes by 3261 (which I'm merely reiterating, not
imposing) aren't an authoritarian attempt at unnecessary proscription. They
serve a purpose.

I'll describe one of the (several) purposes that applies
to the situation you propose.

One of the things that UAs are required to be able to do is sign and/or
encrypt the bodies and/or some subset of the headers to ensure
confidentiality and/or integrity. The protocol places restrictions on
various nodes to ensure that this can be done. Your box is broken precisely
because it interferes with these abilities (or, more precisely, these
abilities interfere with it).

/a

----=_NextPart_ST_09_50_04_Thursday_December_05_2002_1982
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
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>
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version 5.5.2653.12"=
>
<TITLE>RE: [Sip] B2BUA &amp; Max-Forwards</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Our system isn't broken. I would say that this shows that=
 the spec is broken not our system.</FONT>
</P>

<P><FONT SIZE=3D2>Our system provides facilities for some (and we hope &quo=
t;most&quot;) users. We don't provide these facilities for all users becaus=
e: A) We don't think they are important for most users, B) The spec preclud=
es us from doing so (e.g. end-to-end security)</FONT></P>

<P><FONT SIZE=3D2>But because the spec says we MUST do end-to-end security =
we can't be sanctioned as compliant, despite the fact that &quot;most&quot;=
 of our users don't want that. (BTW I have yet to see a endpoint that does =
end-to-end security so it appears that these are all broken as well)</FONT>=
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Adam Roach [<A HREF=3D"mailto:adam@dynamicsoft.com=
">mailto:adam@dynamicsoft.com</A>] </FONT>
<BR><FONT SIZE=3D2>Sent: 04 December 2002 20:45</FONT>
<BR><FONT SIZE=3D2>To: 'Phil Reynolds'; sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: RE: [Sip] B2BUA &amp; Max-Forwards</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Phil Reynolds [<A HREF=3D"mailto:preynolds@ridgewa=
ysystems.com">mailto:preynolds@ridgewaysystems.com</A>]</FONT>
</P>

<P><FONT SIZE=3D2>&gt; Now your back to restricting what I can and can't do=
 in an </FONT>
<BR><FONT SIZE=3D2>&gt; intermediary. It can't be a B2BUA 'cos it doesn't n=
eed to honour </FONT>
<BR><FONT SIZE=3D2>&gt; require headers (the UA at the other end does that,=
 if they were meant </FONT>
<BR><FONT SIZE=3D2>&gt; for my intermediary then they would be proxy-requir=
e?), and it can't </FONT>
<BR><FONT SIZE=3D2>&gt; be a proxy 'cos of the rules about not changing the=
 body so what is </FONT>
<BR><FONT SIZE=3D2>&gt; it?</FONT>
</P>

<P><FONT SIZE=3D2>Broken.</FONT>
</P>

<P><FONT SIZE=3D2>The restrictions placed on nodes by 3261 (which I'm merel=
y reiterating, not imposing) aren't an authoritarian attempt at unnecessary=
 proscription. They serve a purpose.</FONT></P>

<P><FONT SIZE=3D2>I'll describe one of the (several) purposes that applies<=
/FONT>
<BR><FONT SIZE=3D2>to the situation you propose.</FONT>
</P>

<P><FONT SIZE=3D2>One of the things that UAs are required to be able to do =
is sign and/or encrypt the bodies and/or some subset of the headers to ensu=
re confidentiality and/or integrity. The protocol places restrictions on va=
rious nodes to ensure that this can be done. Your box is broken precisely b=
ecause it interferes with these abilities (or, more precisely, these abilit=
ies interfere with it).</FONT></P>

<P><FONT SIZE=3D2>/a</FONT>
</P>

</BODY>
</HTML>
----=_NextPart_ST_09_50_04_Thursday_December_05_2002_1982--

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X-Sybari-Trust: d2d12471 ca231590 0ab44ba3 00000138
From: Christer Holmberg <christer.holmberg@lmf.ericsson.se>
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Subject: [Sip] ABNF issue
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Hi,

The ABNF syntax in RFC3261 says the following things:

LWS              =  [*WSP CRLF] 1*WSP ; linear whitespace

Request          = Request-Line
                          *( message-header )
                          CRLF
                          [ message-body ]

message-body  =  *OCTET

Also, the syntax says that every message-header ends with CTRL.

Now, the spec says that the SIP part of a message ends with CTRL CTRL.

However, if the message-body part starts with one or more WSP, won't the

CRLF between the message-body be interpreted as LWS instead? Ie, there
will NOT be CRLF CRLF at the end of the SIP part of message, but instead

CRLF LWS.

I think this is a bug in the ABNF, since the message-body from a syntax
point
of view is part of the SIP messages, so they can't be "separated".

Now, to solve this the ABNF needs to be changed, so that the definition
for LWS is not valid at the beginning of a new line (when a
message-header has been successfully parsed), ie LWS is only valid
within headers. Then, if a new line starts with CRLF, is is CRLF no
matter what comes after.

Regards,

Christer Holmberg
Ericsson Finland


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From: "Pete Cordell" <pete@tech-know-ware.com>
To: "Henry Sinnreich" <Henry.Sinnreich@wcom.com>
Cc: <sip@ietf.org>
References: <000901c29bc7$b762fc30$248823a6@hsinnreich2>
Subject: Re: [Sip] B2BUA
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Henry,

The Internet has had a few famous outages as well.  Besides, if all the
telco traffic was put over IP, with today's level of QoS on the Internet, it
would have been brought to its knees also.

To say that one is all goodness and light and the other all evil doesn't do
us as a community any good as when customers find out there are problems
they tend not to believe anything else we say!

Only vapourware is flawless.  Most people know that.  Respect the
intelligence of your customer, or they will find somebody that does!!!

Pete.

----- Original Message -----
From: "Henry Sinnreich" <Henry.Sinnreich@wcom.com>
To: "'Phil Reynolds'" <preynolds@ridgewaysystems.com>; <sip@ietf.org>
Sent: 04 December 2002 19:02
Subject: RE: [Sip] B2BUA


> >The telecoms people are clever enough to have thought
> >about that and build redundancy, fault tolerance and failover into
> their switches.
>
> This has proven unfortunately not to be the case on September 11 when
> telecom service failed for days, while Internet traffic continued
> smoothly. I suggest you to look at the central control model of the
> switch network versus the e2e architecture of the Internet. SIP enabled
> networks are also fully distributed and the e2e control principle
> maintained - that's why state in placed in SIP endpoints.
>
> I am not sure if a few words here can give full justice to the superior
> resilience of the Internet and to SIP enabled services,...
>
> Thanks, Henry
>
> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Phil
> Reynolds
> Sent: Wednesday, December 04, 2002 7:37 AM
> To: sip@ietf.org
> Subject: RE: [Sip] B2BUA
>
>
> If a telephone switch crashes you loose all calls...  The telecoms
> people are clever enough to have thought about that and build
> redundancy, fault tolerance and failover into their switches.
> Maybe some of us IP-telecoms people have thought of similar techniques
> for SIP intermediaries?
> -----Original Message-----
> From: Jiri Kuthan [mailto:jiri.kuthan@fokus.fraunhofer.de]
> Sent: 04 December 2002 12:32
> To: Christian Huitema; Roy, Radhika R, ALASO; mbarnes@nortelnetworks.com
>
> Cc: dean.willis@softarmor.com; Juan-Carlos.Rojas@alcatel.fr;
> sip@ietf.org
> Subject: RE: [Sip] B2BUA
> [...snip...]
> I think there are more issues with B2BUA. A B2BUA's failure affects all
> current calls :-(
> -Jiri
> _______________________________________________
> Sip mailing list  https://www1.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://www1.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: "Pete Cordell" <pete@tech-know-ware.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
        "Adam Roach" <adam@dynamicsoft.com>
Cc: "'Phil Reynolds'" <preynolds@ridgewaysystems.com>, <sip@ietf.org>
References: <9BF66EBF6BEFD942915B4D4D45C051F3A64377@DYN-TX-EXCH-001.dynamicsoft.com> <3DEEF411.2020408@dynamicsoft.com>
Subject: Re: [Sip] B2BUA & Max-Forwards
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This is correct.  A B2BUA can't blindly forward a request containing a
Require header.

The saving grace for a B2BUA is that it does not need to understand the
extension in detail.  It just needs to know whether it can forward a request
that contains a given Require header.  This can be easily provisioned using
some sort of ASCII configuration file, and doesn't require any new code etc.
A B2BUA could even routinely download the list of acceptable Require headers
from some sort of server.  (A smart B2BUA could even post back a set of
unknown Require headers that it has seen recently.)

Added to that, ideally extensions will avoid the need for specifying Require
as much as possible.

Thus the impact of Require headers is likely to be fairly minimal for most
B2BUAs.  As it is so easy to support the majority of Require extensions in a
B2BUA, B2BUAs will likely support the extensions before the majority of
actual UAs!

(And a firewall related B2BUA really ought to be blocking what it doesn't
understand anyway, so there is little conflict here also.)

Pete.

----- Original Message -----
From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
To: "Adam Roach" <adam@dynamicsoft.com>
Cc: "'Phil Reynolds'" <preynolds@ridgewaysystems.com>; <sip@ietf.org>
Sent: 05 December 2002 06:37
Subject: Re: [Sip] B2BUA & Max-Forwards


> inline.
>
> Adam Roach wrote:
> > -----Original Message-----
> > From: Phil Reynolds [mailto:preynolds@ridgewaysystems.com]
> >
> >
> >>Now your back to restricting what I can and can't do in an
> >>intermediary. It can't be a B2BUA 'cos it doesn't need to
> >>honour require headers (the UA at the other end does that,
> >>if they were meant for my intermediary then they would be
> >>proxy-require?), and it can't be a proxy 'cos of the rules
> >>about not changing the body so what is it?
> >
> >
> > Broken.
>
> Adam is correct.
>
> >
> > The restrictions placed on nodes by 3261 (which I'm merely
> > reiterating, not imposing) aren't an authoritarian attempt
> > at unnecessary proscription. They serve a purpose.
> >
> > I'll describe one of the (several) purposes that applies
> > to the situation you propose.
> >
> > One of the things that UAs are required to be able to do is
> > sign and/or encrypt the bodies and/or some subset of the
> > headers to ensure confidentiality and/or integrity. The
> > protocol places restrictions on various nodes to ensure
> > that this can be done. Your box is broken precisely because
> > it interferes with these abilities (or, more precisely,
> > these abilities interfere with it).
>
> Let me add some more here.
>
> Anything that deviates at all from proxy rules will need to be a b2bua
> and follow b2bua rules, including processing of Require. There is no
> black-or-white here. Let me give some examples.
>
> Lets say my b2bua is a proxy in all ways, BUT it happens to muck with
> SDP to enable firewall traversal. Surely then it can ignore Require,
> right? WRONG. Lets say I define extension foo, which means "the content
> of this SDP is not what it seems - look at this header to instead find
> the real port/address for media". This extension needs a Require
> (because only a UA ever looks at bodies), but not a Proxy-Require,
> because proxies don't. Thus, your almost-a-proxy will fail in odd ways
> because it has ignored Require when it shouldn't have.
>
> Indeed, for any operation prohibited to a proxy, but allowed for a UA, I
> can come up with an extension, indicated in a Require header, that would
> need to be understood by a b2bua which breaks the proxy rules and
> performs this operation. Indeed, the rules on what is a UA, and what is
> a proxy, are meant precisely to help define what goes in Require, and
> what goes in Proxy-Require.
>
> Whether or not Require gets passed on by such a b2bua depends entirely
> on the application logic.
>
> -Jonathan R.
>
> --
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>
> _______________________________________________
> Sip mailing list  https://www1.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: "Pete Cordell" <pete@tech-know-ware.com>
To: "Cullen Jennings" <fluffy@cisco.com>, <sip@ietf.org>
References: <IOELLHIFFNFPHNDEMKCPIEBPEMAA.fluffy@cisco.com>
Subject: Re: [Sip] B2BUA & Max-Forwards
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Cullen,

That's an interesting question.  But I don't think that SIP to SS7 being a
difficult problem should stop us recommending simple behaviour that can help
protect the integrity of SIP networks where B2BUAs are involved.

Pete.

----- Original Message -----
From: "Cullen Jennings" <fluffy@cisco.com>
To: "Pete Cordell" <pete@tech-know-ware.com>; <sip@ietf.org>
Sent: 04 December 2002 02:44
Subject: RE: [Sip] B2BUA & Max-Forwards


>
> A related question to this was how to stop routing loops that included
some
> PSTN hops. One solution to this was to make sure that as a call was passed
> from SIP to SS7 there was a mapping of the number of hops. A trivial 1:1
> mapping would be unlikely to work.
>
> I really don't really care that much if something like this is documented
or
> not but I imagine that some devices will do it to stop routing loops that
> eat infinite amounts of resources.
>
> Cullen
>
>
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Pete
> > Cordell
> > Sent: Tuesday, December 03, 2002 9:44 AM
> > To: sip@ietf.org
> > Subject: [Sip] B2BUA & Max-Forwards
> >
> >
> > One issue that the recent B2BUA discussion has raised in my mind is what
a
> > B2BUA (whatever its decided that should be) should do with Max-Forwards.
> >
> > This may have been discussed before, but while RFC 3261 says a B2BUA is
a
> > back to back UAS / UAC, and the coupling is not specified in the RFC,
> > perhaps it ought to say that the max-forwards parameter MUST be
propagated
> > from the incoming request to the outgoing request using the same
> > rules as a
> > regular proxy.  This would seem helpful for the avoidance of loops.
> >
> > Any comments?
> >
> > Pete.
> >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.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://www1.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
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From: "Pete Cordell" <pete@tech-know-ware.com>
To: "Dean Willis" <dean.willis@softarmor.com>, <sip@ietf.org>
References: <00cd01c29bfc$e7dd0240$fe0c0c42@txdwillis>
Subject: Re: [Sip] B2BUA & Max-Forwards
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Dean,

Firstly, your statement that "The B2BUA doesn't change headers that transit
it." seems inconsistent with the statement "It does the same thing as ...
ANY 500-pound gorilla does".  I don't understand.

Secondly, I think max-forwards is a special case.  It is important for the
protection of SIP systems as it prevents loops.

Indeed, there is a major difference between a UAS and a B2BUA and that is
that the former terminates a request, and the latter terminates a request
AND ALSO forwards an associated request (or more).  The impact of this
difference should be covered in the spec in the interests of protecting
other SIP systems.

Pete.

----- Original Message -----
From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Pete Cordell'" <pete@tech-know-ware.com>; <sip@ietf.org>
Sent: 05 December 2002 01:23
Subject: RE: [Sip] B2BUA & Max-Forwards


> > One issue that the recent B2BUA discussion has raised in my
> > mind is what a B2BUA (whatever its decided that should be)
> > should do with Max-Forwards.
>
> It does the same thing as it does with any other header or body part,
> which is the same thing that ANY 500-pound gorilla does to somebody's
> body parts -- whatever it wants.
>
> What you're asking is "what is the transparency of this node with
> respect to max-forwards", which is "#4 -- Header Transaprency" in my
> top-10 list. The specification of a back-to-back user agent doesn't
> mandate any level of transparency on this or any other header.
>
> --
> Dean
>
> _______________________________________________
> Sip mailing list  https://www1.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
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From: "Pete Cordell" <pete@tech-know-ware.com>
To: "Christian Huitema" <huitema@windows.microsoft.com>
Cc: <sip@ietf.org>
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Subject: Re: [Sip] B2BUA - Security
Date: Thu, 5 Dec 2002 12:55:18 -0000
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Christian,

I'm afraid I think that end-to-end security of SDP is a complete red
herring.

I would say that encrypting and / or integrity checking of SDP alone
achieves absolutely no security benefit.

SDP really does two things.  It specifies transport addresses and chosen
codecs.

Let's assume that you do encrypt the SDP and I can't read it.

I can have a good guess at the transport addresses in use from many of the
other fields in SIP such as via lines, To, From, uri, contact, route etc.

By analysing the RTP traffic, I can easily find out what codec is being
used, either because the packet tells me, or from other statistical
characteristics.

I have therefore discovered, with very little effort most of what I would
find useful in SDP.  (Ironically sending stuff to a media relay would make
this slightly harder!!!)

If you want security of the media, then you have to encrypt and integrity
check the RTP.

Encrypting SDP simply burns CPU cycles!!!

(Put another way, encrypting SDP is another form of security by obscurity.
And we all know what that means.)

(Note that it may make sense to encrypt other bodies types, such as those
that contain HTML, but this is a different case.)

Pete.

----- Original Message -----
From: "Christian Huitema" <huitema@windows.microsoft.com>
To: "Roy, Radhika R, ALASO" <rrroy@att.com>; <mbarnes@nortelnetworks.com>
Cc: <dean.willis@softarmor.com>; <Juan-Carlos.Rojas@alcatel.fr>;
<sip@ietf.org>
Sent: 03 December 2002 16:35
Subject: RE: [Sip] B2BUA


> > > [MB]: Some of this, of course, will lead us back to the
> > > list of the reasons why some folks consider these things "evil", ...
> >
> > I do not remember whether anyone has ever said that B2BUA is evil
> > things. What I remember is that the dumping of the B2BUA functionality
> > in the name of SIP Proxy is an evil thing.
>
> The real issue in all this is security. Many security folks believe
> that, in order to get proper security, the body of the signaling
> messages should be signed, and possibly encrypted. In that logic, the
> proxies only manipulate the information exposed in the SIP header, which
> can be as little as the pair of source and destination URL.
>
> Any intermediary that attempts to "rewrite the SDP" has to somehow
> insert itself in the end-to-end security exchange; many security folks
> will resent that as "breaking end-to-end security". Insofar as SDP
> rewriting attempts to provide a legitimate service, the challenge is
> thus to provide the service in such a way that it does not require
> breaking end-to-end encryption.
>
> -- Christian Huitema
> _______________________________________________
> Sip mailing list  https://www1.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
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Subject: Re: [Sip] B2BUA - (session policy fix)
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Jonathan,

I agree with this whole heartedly.

However, as I think SDP security is a red herring (see other mail), and the
real issue is RTP end-to-end security, I would first like to explore whether
all the necessary security issues can be tackled by using SIP as is rather
than developing new extensions.

That way we could get away with just an informational RFC, or perhaps some
guidelines in 3261 bis.

Pete.

----- Original Message -----
From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>

...

>
> As an example, a very common offense is mucking with SDP, for example,
> to implement a firewall control proxy, ala midcom. Instead of defining a
> new element, called an SDP-munging proxy, and dealing with all the
> horrible security implications that will arise from it, I would propose
> to simply FIX THE PROBLEM and define a means by which a proxy can impose
> session policies without breaking SIP. Indeed, I wrote a draft on just
> such a topic:
>
>
http://www.jdrosen.net/papers/draft-rosenberg-sipping-session-policy-00.txt
>


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Subject: RE: [Sip] B2BUA & Max-Forwards
Date: Thu, 5 Dec 2002 13:21:36 -0000
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Thread-Topic: [Sip] B2BUA & Max-Forwards
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From: "James Undery" <jundery@ubiquity.net>
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Hi,

I think your confusion comes from mixing two email threads; the first statement in context says if a specific B2BUA demonstrates header transparency it doesn't change headers; the second statement is about B2BUAs in general and says they can do whatever they want (without commenting on the desirability of their actions).

Max-Forwards probably are an important case for your particular application, in general they may not be required (e.g. routing to an element you can guarantee not to proxy).

Your comment on Max-Forwards highlights well the difficulties faced in defining a B2BUA more specifically than the definition in 3261.

James 

> -----Original Message-----
> From: Pete Cordell [mailto:pete@tech-know-ware.com]
> Sent: 05 December 2002 12:10
> To: Dean Willis; sip@ietf.org
> Subject: Re: [Sip] B2BUA & Max-Forwards
> 
> 
> Dean,
> 
> Firstly, your statement that "The B2BUA doesn't change 
> headers that transit
> it." seems inconsistent with the statement "It does the same 
> thing as ...
> ANY 500-pound gorilla does".  I don't understand.
> 
> Secondly, I think max-forwards is a special case.  It is 
> important for the
> protection of SIP systems as it prevents loops.
> 
> Indeed, there is a major difference between a UAS and a B2BUA 
> and that is
> that the former terminates a request, and the latter 
> terminates a request
> AND ALSO forwards an associated request (or more).  The impact of this
> difference should be covered in the spec in the interests of 
> protecting
> other SIP systems.
> 
> Pete.
> 
> ----- Original Message -----
> From: "Dean Willis" <dean.willis@softarmor.com>
> To: "'Pete Cordell'" <pete@tech-know-ware.com>; <sip@ietf.org>
> Sent: 05 December 2002 01:23
> Subject: RE: [Sip] B2BUA & Max-Forwards
> 
> 
> > > One issue that the recent B2BUA discussion has raised in my
> > > mind is what a B2BUA (whatever its decided that should be)
> > > should do with Max-Forwards.
> >
> > It does the same thing as it does with any other header or 
> body part,
> > which is the same thing that ANY 500-pound gorilla does to 
> somebody's
> > body parts -- whatever it wants.
> >
> > What you're asking is "what is the transparency of this node with
> > respect to max-forwards", which is "#4 -- Header Transaprency" in my
> > top-10 list. The specification of a back-to-back user agent doesn't
> > mandate any level of transparency on this or any other header.
> >
> > --
> > Dean
> >
> > _______________________________________________
> > Sip mailing list  https://www1.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://www1.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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This list is for NEW development of the core SIP Protocol
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Subject: AW: [Sip] B2BUA
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Henry,

unfortunately you are comparing apples and pears when stating that the telecom service failed for days after Sept 11, while Internet traffic continued smoothly. The mentioned "telefon service" consists of thousands of (real-time) calls to emergency centres while the Internet traffic consists mainly of requests to the Web pages of CNN and others (with bad response times) to retrieve *information* on the status of the catastrophe.

Rainer


-----Ursprüngliche Nachricht-----
Von: Pete Cordell [mailto:pete@tech-know-ware.com]
Gesendet: Donnerstag, 5. Dezember 2002 13:25
An: Henry Sinnreich
Cc: sip@ietf.org
Betreff: Re: [Sip] B2BUA


Henry,

The Internet has had a few famous outages as well.  Besides, if all the
telco traffic was put over IP, with today's level of QoS on the Internet, it
would have been brought to its knees also.

To say that one is all goodness and light and the other all evil doesn't do
us as a community any good as when customers find out there are problems
they tend not to believe anything else we say!

Only vapourware is flawless.  Most people know that.  Respect the
intelligence of your customer, or they will find somebody that does!!!

Pete.

----- Original Message -----
From: "Henry Sinnreich" <Henry.Sinnreich@wcom.com>
To: "'Phil Reynolds'" <preynolds@ridgewaysystems.com>; <sip@ietf.org>
Sent: 04 December 2002 19:02
Subject: RE: [Sip] B2BUA


> >The telecoms people are clever enough to have thought
> >about that and build redundancy, fault tolerance and failover into
> their switches.
>
> This has proven unfortunately not to be the case on September 11 when
> telecom service failed for days, while Internet traffic continued
> smoothly. I suggest you to look at the central control model of the
> switch network versus the e2e architecture of the Internet. SIP enabled
> networks are also fully distributed and the e2e control principle
> maintained - that's why state in placed in SIP endpoints.
>
> I am not sure if a few words here can give full justice to the superior
> resilience of the Internet and to SIP enabled services,...
>
> Thanks, Henry
>
> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Phil
> Reynolds
> Sent: Wednesday, December 04, 2002 7:37 AM
> To: sip@ietf.org
> Subject: RE: [Sip] B2BUA
>
>
> If a telephone switch crashes you loose all calls...  The telecoms
> people are clever enough to have thought about that and build
> redundancy, fault tolerance and failover into their switches.
> Maybe some of us IP-telecoms people have thought of similar techniques
> for SIP intermediaries?
> -----Original Message-----
> From: Jiri Kuthan [mailto:jiri.kuthan@fokus.fraunhofer.de]
> Sent: 04 December 2002 12:32
> To: Christian Huitema; Roy, Radhika R, ALASO; mbarnes@nortelnetworks.com
>
> Cc: dean.willis@softarmor.com; Juan-Carlos.Rojas@alcatel.fr;
> sip@ietf.org
> Subject: RE: [Sip] B2BUA
> [...snip...]
> I think there are more issues with B2BUA. A B2BUA's failure affects all
> current calls :-(
> -Jiri
> _______________________________________________
> Sip mailing list  https://www1.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://www1.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: Thu, 05 Dec 2002 08:48:57 -0500
From: "David R. Oran" <oran@cisco.com>
To: Padma Suresh <padma_suresh@hotmail.com>, vkg@lucent.com
cc: pete@tech-know-ware.com, sip@ietf.org
Subject: Re: [Sip] B2BUA & Max-Forwards
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--On Wednesday, December 04, 2002 4:50 PM +0000 Padma Suresh 
<padma_suresh@hotmail.com> wrote:

>
>
>
> A B2BUA is an intermediate network element
no it isn't. It's one or more UA's in the same box communicating with each 
other using some private machinery, such as an API+application code.

Calling a B2BUA an intermediate network element is probably where most of 
the nonsense on this thread originates. Perhaps that's the basic 
clarification we need in the description to avoid these incorrect 
expectations?

>and MUST maintain the
> expectations of the endpoints involved in the session.
>
Also incorrect. Since the B2BUA is at least 2 UA's its job is to SEPARATELY 
maintain the expectations of EACH endpoint it is talking to. This does not 
infer ANY transitivity properties across a B2BUA at all; in fact, if you 
wanted well-defined transitivity you would have put in a proxy instead.

> Not forwarding the Require or Expires headers (and some more) would break
> the above which would not be quite acceptable.
>
Disagree vociferiously. As others have pointed out, it is crucial to NOT 
forward any headers. You MUST consume them and process them on receipt. You 
must GENERATE and send them on transmit. What happens in between is 
whatever algorithm or algorithms the application doing the B2BUA functions 
deems appropriate. If there is any relationship at all between the input 
and the output, that relationship is established by the the application 
logic. If the application logic performs the identity mapping, then that is 
what the application thinks is the appropriate thing to do. If the 
application logic (for example) decrements Max-Forwards by 42 because 
Douglas Adam's probate attorney wrote it, that's ok too (as long as the 
result is >0 of course...).

> However, yes, the B2BUA can choose to step out of the signalling path by
> setting the contact header to the appropriate endpoint.
>
It is not *in* the signaling path.

> Vijay I agree that the ambiguity of choosing which header to pass on and
> which not, could be well avoided.
>
> Regards
>
> Padma
>
If this level of misunderstanding of SIP persists by having the spec talk 
about B2BUAs, then perhaps the best this to do is the radical step of 
simply removing all references to them and banning the term from the 
lexicon.

Dave.

>
>
>> From: "Vijay K. Gurbani"
>> To: Padma Suresh
>> CC: pete@tech-know-ware.com, sip@ietf.org
>> Subject: Re: [Sip] B2BUA & Max-Forwards
>> Date: Wed, 04 Dec 2002 09:21:11 -0600
>>
>> Padma Suresh wrote:
>>> Vijay,
>>>
>>> You are right -
>>>
>>> A B2BUA needs to propagate some of the headers received in a
>>> request/response to the other endpoint.
>>
>> Choosing which headers should be passed by a B2BUA and which not is
>> exactly the ambiguity that we should AVOID, as I pointed out in my
>> my response to the original poster. As others have expressed, we
>> may
>> try to taxonomize different types of B2BUAs, but each will have a
>> different manner of handling the headers (depending on the
>> application
>> being realized by that particular B2BUA).
>>
>> An anonymizer B2BUA may change To, From, Contact, etc, for instance
>> (and
>> maintain internal state). A network-hiding B2BUA may scrap all Via
>> headers coming in and only expose one Via header going out. The
>> possibilities are endless.
>>
>>> Example: Require, Expires to name a few. This is because these
>>> headers actually are relevant end to end.
>>
>> A B2BUA may decide to consume the Require and change the Expires
>> header
>> (maybe decrease its value) before sending it downstream, thus
>> causing
>> havoc with the expectations of the sending UAC.
>>
>>> However, some of the other headers like Contact should be set
>>> by the B2BUA as itself.
>>
>> Likewise, I can imagine a B2BUA which rendezvous two users and steps
>> out of subsequent signaling. Thus the Contact it sends to each of
>> the users is that of the other peer.
>>
>> I would strongly *stay away* from prescribing which headers must be
>> passed by a B2BUA and which must not.
>>
>> Regards,
>>
>> - vijay
>> --
>> Vijay K. Gurbani vkg@{lucent.com,research.bell-labs.com,acm.org}
>> Wireless Networks Group/Internet Software and Services
>> Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm
>> 6G-440
>> Naperville, Illinois 60566 Voice: +1 630 224 0216
>>
>> _______________________________________________
>> Sip mailing list https://www1.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
>
>
> __________________________________________________
> Tired of spam? Get advanced junk mail protection with MSN
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> https://www1.ietf.org/mailman/listinfo/sip This list is for NEW
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> for questions on current sip Use sipping@ietf.org for new developments on
> the application of sip

------------------------
David R. Oran
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720
Office: +1 978 264 2048
VoIP: +1 408 571 4576
Email: oran@cisco.com
_______________________________________________
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Date: Thu, 05 Dec 2002 09:02:29 -0500
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To: Pete Cordell <pete@tech-know-ware.com>,
        Dean Willis <dean.willis@softarmor.com>, sip@ietf.org
Subject: Re: [Sip] B2BUA & Max-Forwards
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--On Thursday, December 05, 2002 12:09 PM +0000 Pete Cordell 
<pete@tech-know-ware.com> wrote:

> Dean,
>
> Firstly, your statement that "The B2BUA doesn't change headers that
> transit it." seems inconsistent with the statement "It does the same
> thing as ... ANY 500-pound gorilla does".  I don't understand.
>
> Secondly, I think max-forwards is a special case.  It is important for the
> protection of SIP systems as it prevents loops.
>
Once a dialog is terminated at a UAS it's up to the application to do loop 
prevention. SIP is not involved any more. The application could compute a 
value of Max-forwards for initiating outgoing calls based on incoming calls 
which preserves the "strictly decreasing" invariant, or alternatively it 
could just use the default MAx-Forwards in outgoing requests and execute a 
contant memory loop suppression algorithm such as Knuth's even-odd 
iterator. In either event the SIP spec(s) need say nothing about this.

> Indeed, there is a major difference between a UAS and a B2BUA and that is
> that the former terminates a request, and the latter terminates a request
> AND ALSO forwards an associated request (or more).
I disagree. A UAS is a UAS. It's behavior is completely specified by 
RFC3261 (unless there's a hole in the spec of course...). What it means to 
originate (NOT FORWARD!!!!) an "associated request" is completely 
application dependent. If I program my child's phone to initiate a SIP call 
to my surrepititious recording device every time it receives a call and 
give that device first chance to over-ride policy of the child's phone is 
the phone a B2BUA? Yes, no, maybe so, but who cares? Tell me what the 
applciation is and I can write some specs for that application and how it 
has to originate, terminate and process SIP transactions to do that 
application.

I think the botom line is that there is not a single, well defined 
appliation that one can point to and say "that's a B2BUA".

> The impact of this
> difference should be covered in the spec in the interests of protecting
> other SIP systems.
>
I disagree.

> Pete.
>
> ----- Original Message -----
> From: "Dean Willis" <dean.willis@softarmor.com>
> To: "'Pete Cordell'" <pete@tech-know-ware.com>; <sip@ietf.org>
> Sent: 05 December 2002 01:23
> Subject: RE: [Sip] B2BUA & Max-Forwards
>
>
>> > One issue that the recent B2BUA discussion has raised in my
>> > mind is what a B2BUA (whatever its decided that should be)
>> > should do with Max-Forwards.
>>
>> It does the same thing as it does with any other header or body part,
>> which is the same thing that ANY 500-pound gorilla does to somebody's
>> body parts -- whatever it wants.
>>
>> What you're asking is "what is the transparency of this node with
>> respect to max-forwards", which is "#4 -- Header Transaprency" in my
>> top-10 list. The specification of a back-to-back user agent doesn't
>> mandate any level of transparency on this or any other header.
>>
>> --
>> Dean
>>
>> _______________________________________________
>> Sip mailing list  https://www1.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://www1.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

------------------------
David R. Oran
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720
Office: +1 978 264 2048
VoIP: +1 408 571 4576
Email: oran@cisco.com
_______________________________________________
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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		: The SIP Refer Method
	Author(s)	: R. Sparks
	Filename	: draft-ietf-sip-refer-07.txt
	Pages		: 23
	Date		: 2002-12-4
	
This document defines the REFER method.  This SIP extension requests
that the recipient REFER to a resource provided in the request.  It
provides a mechanism allowing the party sending the REFER to be
notified of the outcome of the referenced request.  This can be used
to enable many applications, including call transfer.
In addition to the REFER method, this document defines the the refer
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Hi all,

we are two students at the university of applied sciences in Rapperswil
(Switzerland). Currently we're working on a student research project concerning
the use of SIP in a policy based network. Therefore we're looking for
information on this topic.

We've already read RFC 3312 (Integration of Resource Management and Session
Initiation Protocol) and the draft "SIP Extensions for QoS support" by L.
Veltri.

Does anybody know if there exist any other approaches/solutions for using SIP
signaling QoS?

We're looking forward to hearing from you.

Kind regards
C. Boesch
cboesch@hsr.ch
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From: "Mary Barnes" <mbarnes@nortelnetworks.com>
To: "'Liebhart Rainer'" <rainer.liebhart@siemens.com>,
        "'Pete Cordell'"
	 <pete@tech-know-ware.com>,
        Henry Sinnreich <Henry.Sinnreich@wcom.com>
Cc: sip@ietf.org
Subject: RE: [Sip] B2BUA
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Hi Rainer,

Henry was referring to was the fact that SIP calls were able to be made =
on
the Internet when the phone system in the NYC area was not available =
for use
by Columbia University, per the following thread:
http://www1.ietf.org/mail-archive/working-groups/sip/current/msg01026.ht=
ml

Mary.

-----Original Message-----
From: Liebhart Rainer [mailto:rainer.liebhart@siemens.com]
Sent: Thursday, December 05, 2002 7:46 AM
To: 'Pete Cordell'; Henry Sinnreich
Cc: sip@ietf.org
Subject: AW: [Sip] B2BUA


Henry,

unfortunately you are comparing apples and pears when stating that the
telecom service failed for days after Sept 11, while Internet traffic
continued smoothly. The mentioned "telefon service" consists of =
thousands of
(real-time) calls to emergency centres while the Internet traffic =
consists
mainly of requests to the Web pages of CNN and others (with bad =
response
times) to retrieve *information* on the status of the catastrophe.

Rainer


-----Urspr=FCngliche Nachricht-----
Von: Pete Cordell [mailto:pete@tech-know-ware.com]
Gesendet: Donnerstag, 5. Dezember 2002 13:25
An: Henry Sinnreich
Cc: sip@ietf.org
Betreff: Re: [Sip] B2BUA


Henry,

The Internet has had a few famous outages as well.  Besides, if all the
telco traffic was put over IP, with today's level of QoS on the =
Internet, it
would have been brought to its knees also.

To say that one is all goodness and light and the other all evil =
doesn't do
us as a community any good as when customers find out there are =
problems
they tend not to believe anything else we say!

Only vapourware is flawless.  Most people know that.  Respect the
intelligence of your customer, or they will find somebody that does!!!

Pete.

----- Original Message -----
From: "Henry Sinnreich" <Henry.Sinnreich@wcom.com>
To: "'Phil Reynolds'" <preynolds@ridgewaysystems.com>; <sip@ietf.org>
Sent: 04 December 2002 19:02
Subject: RE: [Sip] B2BUA


> >The telecoms people are clever enough to have thought
> >about that and build redundancy, fault tolerance and failover into
> their switches.
>
> This has proven unfortunately not to be the case on September 11 when
> telecom service failed for days, while Internet traffic continued
> smoothly. I suggest you to look at the central control model of the
> switch network versus the e2e architecture of the Internet. SIP =
enabled
> networks are also fully distributed and the e2e control principle
> maintained - that's why state in placed in SIP endpoints.
>
> I am not sure if a few words here can give full justice to the =
superior
> resilience of the Internet and to SIP enabled services,...
>
> Thanks, Henry
>
> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of =
Phil
> Reynolds
> Sent: Wednesday, December 04, 2002 7:37 AM
> To: sip@ietf.org
> Subject: RE: [Sip] B2BUA
>
>
> If a telephone switch crashes you loose all calls...  The telecoms
> people are clever enough to have thought about that and build
> redundancy, fault tolerance and failover into their switches.
> Maybe some of us IP-telecoms people have thought of similar =
techniques
> for SIP intermediaries?
> -----Original Message-----
> From: Jiri Kuthan [mailto:jiri.kuthan@fokus.fraunhofer.de]
> Sent: 04 December 2002 12:32
> To: Christian Huitema; Roy, Radhika R, ALASO; =
mbarnes@nortelnetworks.com
>
> Cc: dean.willis@softarmor.com; Juan-Carlos.Rojas@alcatel.fr;
> sip@ietf.org
> Subject: RE: [Sip] B2BUA
> [...snip...]
> I think there are more issues with B2BUA. A B2BUA's failure affects =
all
> current calls :-(
> -Jiri
> _______________________________________________
> Sip mailing list  https://www1.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://www1.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://www1.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://www1.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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<P><FONT SIZE=3D2>Hi Rainer,</FONT>
</P>

<P><FONT SIZE=3D2>Henry was referring to was the fact that SIP calls =
were able to be made on the Internet when the phone system in the NYC =
area was not available for use by Columbia University, per the =
following thread:</FONT></P>

<P><FONT SIZE=3D2><A =
HREF=3D"http://www1.ietf.org/mail-archive/working-groups/sip/current/msg=
01026.html" =
TARGET=3D"_blank">http://www1.ietf.org/mail-archive/working-groups/sip/c=
urrent/msg01026.html</A></FONT>
</P>

<P><FONT SIZE=3D2>Mary.</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Liebhart Rainer [<A =
HREF=3D"mailto:rainer.liebhart@siemens.com">mailto:rainer.liebhart@sieme=
ns.com</A>]</FONT>
<BR><FONT SIZE=3D2>Sent: Thursday, December 05, 2002 7:46 AM</FONT>
<BR><FONT SIZE=3D2>To: 'Pete Cordell'; Henry Sinnreich</FONT>
<BR><FONT SIZE=3D2>Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: AW: [Sip] B2BUA</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Henry,</FONT>
</P>

<P><FONT SIZE=3D2>unfortunately you are comparing apples and pears when =
stating that the telecom service failed for days after Sept 11, while =
Internet traffic continued smoothly. The mentioned &quot;telefon =
service&quot; consists of thousands of (real-time) calls to emergency =
centres while the Internet traffic consists mainly of requests to the =
Web pages of CNN and others (with bad response times) to retrieve =
*information* on the status of the catastrophe.</FONT></P>

<P><FONT SIZE=3D2>Rainer</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>-----Urspr=FCngliche Nachricht-----</FONT>
<BR><FONT SIZE=3D2>Von: Pete Cordell [<A =
HREF=3D"mailto:pete@tech-know-ware.com">mailto:pete@tech-know-ware.com</=
A>]</FONT>
<BR><FONT SIZE=3D2>Gesendet: Donnerstag, 5. Dezember 2002 13:25</FONT>
<BR><FONT SIZE=3D2>An: Henry Sinnreich</FONT>
<BR><FONT SIZE=3D2>Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Betreff: Re: [Sip] B2BUA</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Henry,</FONT>
</P>

<P><FONT SIZE=3D2>The Internet has had a few famous outages as =
well.&nbsp; Besides, if all the</FONT>
<BR><FONT SIZE=3D2>telco traffic was put over IP, with today's level of =
QoS on the Internet, it</FONT>
<BR><FONT SIZE=3D2>would have been brought to its knees also.</FONT>
</P>

<P><FONT SIZE=3D2>To say that one is all goodness and light and the =
other all evil doesn't do</FONT>
<BR><FONT SIZE=3D2>us as a community any good as when customers find =
out there are problems</FONT>
<BR><FONT SIZE=3D2>they tend not to believe anything else we =
say!</FONT>
</P>

<P><FONT SIZE=3D2>Only vapourware is flawless.&nbsp; Most people know =
that.&nbsp; Respect the</FONT>
<BR><FONT SIZE=3D2>intelligence of your customer, or they will find =
somebody that does!!!</FONT>
</P>

<P><FONT SIZE=3D2>Pete.</FONT>
</P>

<P><FONT SIZE=3D2>----- Original Message -----</FONT>
<BR><FONT SIZE=3D2>From: &quot;Henry Sinnreich&quot; =
&lt;Henry.Sinnreich@wcom.com&gt;</FONT>
<BR><FONT SIZE=3D2>To: &quot;'Phil Reynolds'&quot; =
&lt;preynolds@ridgewaysystems.com&gt;; &lt;sip@ietf.org&gt;</FONT>
<BR><FONT SIZE=3D2>Sent: 04 December 2002 19:02</FONT>
<BR><FONT SIZE=3D2>Subject: RE: [Sip] B2BUA</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>&gt; &gt;The telecoms people are clever enough to =
have thought</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;about that and build redundancy, fault =
tolerance and failover into</FONT>
<BR><FONT SIZE=3D2>&gt; their switches.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; This has proven unfortunately not to be the =
case on September 11 when</FONT>
<BR><FONT SIZE=3D2>&gt; telecom service failed for days, while Internet =
traffic continued</FONT>
<BR><FONT SIZE=3D2>&gt; smoothly. I suggest you to look at the central =
control model of the</FONT>
<BR><FONT SIZE=3D2>&gt; switch network versus the e2e architecture of =
the Internet. SIP enabled</FONT>
<BR><FONT SIZE=3D2>&gt; networks are also fully distributed and the e2e =
control principle</FONT>
<BR><FONT SIZE=3D2>&gt; maintained - that's why state in placed in SIP =
endpoints.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; I am not sure if a few words here can give full =
justice to the superior</FONT>
<BR><FONT SIZE=3D2>&gt; resilience of the Internet and to SIP enabled =
services,...</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; Thanks, Henry</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: sip-admin@ietf.org [<A =
HREF=3D"mailto:sip-admin@ietf.org">mailto:sip-admin@ietf.org</A>] On =
Behalf Of Phil</FONT>
<BR><FONT SIZE=3D2>&gt; Reynolds</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Wednesday, December 04, 2002 7:37 =
AM</FONT>
<BR><FONT SIZE=3D2>&gt; To: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: [Sip] B2BUA</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; If a telephone switch crashes you loose all =
calls...&nbsp; The telecoms</FONT>
<BR><FONT SIZE=3D2>&gt; people are clever enough to have thought about =
that and build</FONT>
<BR><FONT SIZE=3D2>&gt; redundancy, fault tolerance and failover into =
their switches.</FONT>
<BR><FONT SIZE=3D2>&gt; Maybe some of us IP-telecoms people have =
thought of similar techniques</FONT>
<BR><FONT SIZE=3D2>&gt; for SIP intermediaries?</FONT>
<BR><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Jiri Kuthan [<A =
HREF=3D"mailto:jiri.kuthan@fokus.fraunhofer.de">mailto:jiri.kuthan@fokus=
.fraunhofer.de</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: 04 December 2002 12:32</FONT>
<BR><FONT SIZE=3D2>&gt; To: Christian Huitema; Roy, Radhika R, ALASO; =
mbarnes@nortelnetworks.com</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: dean.willis@softarmor.com; =
Juan-Carlos.Rojas@alcatel.fr;</FONT>
<BR><FONT SIZE=3D2>&gt; sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: [Sip] B2BUA</FONT>
<BR><FONT SIZE=3D2>&gt; [...snip...]</FONT>
<BR><FONT SIZE=3D2>&gt; I think there are more issues with B2BUA. A =
B2BUA's failure affects all</FONT>
<BR><FONT SIZE=3D2>&gt; current calls :-(</FONT>
<BR><FONT SIZE=3D2>&gt; -Jiri</FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the core =
SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip Use</FONT>
<BR><FONT SIZE=3D2>&gt; sipping@ietf.org for new developments on the =
application of sip</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the core =
SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; Use sipping@ietf.org for new developments on =
the application of sip</FONT>
</P>

<P><FONT =
SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>This list is for NEW development of the core SIP =
Protocol</FONT>
<BR><FONT SIZE=3D2>Use sip-implementors@cs.columbia.edu for questions =
on current sip</FONT>
<BR><FONT SIZE=3D2>Use sipping@ietf.org for new developments on the =
application of sip</FONT>
<BR><FONT SIZE=3D2>_______________________________________________</FONT=
>
<BR><FONT SIZE=3D2>Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>This list is for NEW development of the core SIP =
Protocol</FONT>
<BR><FONT SIZE=3D2>Use sip-implementors@cs.columbia.edu for questions =
on current sip</FONT>
<BR><FONT SIZE=3D2>Use sipping@ietf.org for new developments on the =
application of sip</FONT>
</P>

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Use sip-implementors@cs.columbia.edu for questions on current sip
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Date: Thu, 05 Dec 2002 09:45:06 -0500
From: "David R. Oran" <oran@cisco.com>
To: Pete Cordell <pete@tech-know-ware.com>,
        Christian Huitema <huitema@windows.microsoft.com>
cc: sip@ietf.org
Subject: Re: [Sip] B2BUA - Security
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--On Thursday, December 05, 2002 12:55 PM +0000 Pete Cordell 
<pete@tech-know-ware.com> wrote:

> Christian,
>
> I'm afraid I think that end-to-end security of SDP is a complete red
> herring.
>
> I would say that encrypting and / or integrity checking of SDP alone
> achieves absolutely no security benefit.
>
> SDP really does two things.  It specifies transport addresses and chosen
> codecs.
>
...and carries keys for SRTP...hmmmmmmmm........

> Let's assume that you do encrypt the SDP and I can't read it.
>
> I can have a good guess at the transport addresses in use from many of the
> other fields in SIP such as via lines, To, From, uri, contact, route etc.
>
Uh huh, sure. My experience is that you can only guess with reasonable 
probability by knowing the implementation details of the UAs. I remember we 
changed the RTP port allocation algorithm on IOS and all of a sudden people 
started screaming because their ACL filters "stopped working". Trying to 
beat the devil seem a pretty poor direction.

> By analysing the RTP traffic, I can easily find out what codec is being
> used, either because the packet tells me, or from other statistical
> characteristics.
>
Really? Maybe for the standard codecs but why on earth bother?

> I have therefore discovered, with very little effort most of what I would
> find useful in SDP.  (Ironically sending stuff to a media relay would make
> this slightly harder!!!)
>
> If you want security of the media, then you have to encrypt and integrity
> check the RTP.
>
or put it through an IPSEC tunnel. What's the point?

> Encrypting SDP simply burns CPU cycles!!!
>
No, it prevents media hijacking by a MITM.

> (Put another way, encrypting SDP is another form of security by obscurity.
> And we all know what that means.)
>
Disagree vociferously.

> (Note that it may make sense to encrypt other bodies types, such as those
> that contain HTML, but this is a different case.)
>
> Pete.
>
Dave.


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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



From mailnull@www1.ietf.org  Thu Dec  5 11:19:39 2002
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Pete Cordell wrote:
> This is correct.  A B2BUA can't blindly forward a request containing a
> Require header.
> 
> The saving grace for a B2BUA is that it does not need to understand the
> extension in detail.  It just needs to know whether it can forward a request
> that contains a given Require header.  This can be easily provisioned using
> some sort of ASCII configuration file, and doesn't require any new code etc.
> A B2BUA could even routinely download the list of acceptable Require headers
> from some sort of server.  (A smart B2BUA could even post back a set of
> unknown Require headers that it has seen recently.)

This is a vast over-simplification.

As people have been saying over and over in this thread, there is no 
such thing as a "generic b2bua" as you seem to be implying. What is 
right in a b2bua depends entirely on what its doing. To be honest, I 
can't think of any b2bua for which the processing you describe above is 
sensible. As I mentioned in a previous note, a b2bua deviates from a 
proxy by doing some featurs that are normally associated with a UA. That 
b2bua will need to process extensions (i.e., implement them) that are 
associated with that feature.

I am at the point where I agree with the sentiment raised by others, 
that we should remove the term b2bua entirely from the specification. It 
has caused more confusion than anything else.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Pete Cordell wrote:

> I would say that encrypting and / or integrity checking of SDP alone
> achieves absolutely no security benefit.

Really?? Wow. I actually think it prevents the most devastating attacks 
against SIP. These include:

1. modifying SDP in transit to point media streams to an intermediate 
recording device, allowing attackers to establish wiretaps

2. downgrading any SRTP keying done in SDP, so that the media itself is 
not encrypted or integrity protected (this came up during the mmusic 
session in IETF as a driver for S/MIME)

3. launching a DoS attack by sending everyones voice traffic to the 
intended target. You get huge amplification gains here. Very appealing 
for hackers!




> 
> SDP really does two things.  It specifies transport addresses and chosen
> codecs.
> 
> Let's assume that you do encrypt the SDP and I can't read it.
> 
> I can have a good guess at the transport addresses in use from many of the
> other fields in SIP such as via lines, To, From, uri, contact, route etc.

Confidentiality is only one property of security. I am much more 
concerned about message integrity.

> (Put another way, encrypting SDP is another form of security by obscurity.
> And we all know what that means.)

I have no idea how you would arrive at such a conclusion. Security 
through obscurity means that information is not cryptographically 
protected, but encoded with a known process that is just a bit 
complicated. By definition, this is not encryption. Perhaps you have 
discovered some new flaws in encryption technology?

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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inline.

Christer Holmberg wrote:
> Hi,
> 
> The ABNF syntax in RFC3261 says the following things:
> 
> LWS              =  [*WSP CRLF] 1*WSP ; linear whitespace
> 
> Request          = Request-Line
>                           *( message-header )
>                           CRLF
>                           [ message-body ]
> 
> message-body  =  *OCTET
> 
> Also, the syntax says that every message-header ends with CTRL.
> 
> Now, the spec says that the SIP part of a message ends with CTRL CTRL.
> 
> However, if the message-body part starts with one or more WSP, won't the
> 
> CRLF between the message-body be interpreted as LWS instead? Ie, there
> will NOT be CRLF CRLF at the end of the SIP part of message, but instead
> 
> CRLF LWS.

I don't know how you come to this conclusion. If the body starts with 
LWS, you will see a CRLF after the last header field, another CRLF 
before the body, and then an LWS which is the first byte of the body. Thus:

CRLF CRLF LWS

which is fine.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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From: "Pete Cordell" <pete@tech-know-ware.com>
To: "James Undery" <jundery@ubiquity.net>, <sip@ietf.org>
References: <45730E094814E44488F789C1CDED27AE018C2553@GBNEWP0758M.eu.ubiquity.net>
Subject: Re: [Sip] B2BUA & Max-Forwards
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James,

Thanks for your comments.

I think in general you CAN'T guarantee that an element you route to won't
proxy (or be a B2BUA).  And therefore loop detection is a problem.

Propagating Max-Forwards even when you know the precise routing should not
cause you any problems.

Pete.

----- Original Message -----
From: "James Undery" <jundery@ubiquity.net>
To: "Pete Cordell" <pete@tech-know-ware.com>; "Dean Willis"
<dean.willis@softarmor.com>; <sip@ietf.org>
Sent: 05 December 2002 13:21
Subject: RE: [Sip] B2BUA & Max-Forwards


Hi,

I think your confusion comes from mixing two email threads; the first
statement in context says if a specific B2BUA demonstrates header
transparency it doesn't change headers; the second statement is about B2BUAs
in general and says they can do whatever they want (without commenting on
the desirability of their actions).

Max-Forwards probably are an important case for your particular application,
in general they may not be required (e.g. routing to an element you can
guarantee not to proxy).

Your comment on Max-Forwards highlights well the difficulties faced in
defining a B2BUA more specifically than the definition in 3261.

James

> -----Original Message-----
> From: Pete Cordell [mailto:pete@tech-know-ware.com]
> Sent: 05 December 2002 12:10
> To: Dean Willis; sip@ietf.org
> Subject: Re: [Sip] B2BUA & Max-Forwards
>
>
> Dean,
>
> Firstly, your statement that "The B2BUA doesn't change
> headers that transit
> it." seems inconsistent with the statement "It does the same
> thing as ...
> ANY 500-pound gorilla does".  I don't understand.
>
> Secondly, I think max-forwards is a special case.  It is
> important for the
> protection of SIP systems as it prevents loops.
>
> Indeed, there is a major difference between a UAS and a B2BUA
> and that is
> that the former terminates a request, and the latter
> terminates a request
> AND ALSO forwards an associated request (or more).  The impact of this
> difference should be covered in the spec in the interests of
> protecting
> other SIP systems.
>
> Pete.
>
> ----- Original Message -----
> From: "Dean Willis" <dean.willis@softarmor.com>
> To: "'Pete Cordell'" <pete@tech-know-ware.com>; <sip@ietf.org>
> Sent: 05 December 2002 01:23
> Subject: RE: [Sip] B2BUA & Max-Forwards
>
>
> > > One issue that the recent B2BUA discussion has raised in my
> > > mind is what a B2BUA (whatever its decided that should be)
> > > should do with Max-Forwards.
> >
> > It does the same thing as it does with any other header or
> body part,
> > which is the same thing that ANY 500-pound gorilla does to
> somebody's
> > body parts -- whatever it wants.
> >
> > What you're asking is "what is the transparency of this node with
> > respect to max-forwards", which is "#4 -- Header Transaprency" in my
> > top-10 list. The specification of a back-to-back user agent doesn't
> > mandate any level of transparency on this or any other header.
> >
> > --
> > Dean
> >
> > _______________________________________________
> > Sip mailing list  https://www1.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://www1.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
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>

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From: "Pete Cordell" <pete@tech-know-ware.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: "Adam Roach" <adam@dynamicsoft.com>,
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Subject: Re: [Sip] B2BUA & Max-Forwards
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Jonathan,

I have not said anything about generic B2BUAs here.  Each B2BUA
implementation will need to have its own record (and server) of what it is
transparent to or not.  It would not be a generic list.

Perhaps it is reasonable to remove the B2BUA definition, since there is no
definition of a gateway and they are similar beasts.  Just as long as such
an action does not imply that SIP is not suitable for a certain class of
problem.

Pete.

----- Original Message -----
From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
To: "Pete Cordell" <pete@tech-know-ware.com>
Cc: "Adam Roach" <adam@dynamicsoft.com>; "'Phil Reynolds'"
<preynolds@ridgewaysystems.com>; <sip@ietf.org>
Sent: 05 December 2002 16:16
Subject: Re: [Sip] B2BUA & Max-Forwards


>
>
> Pete Cordell wrote:
> > This is correct.  A B2BUA can't blindly forward a request containing a
> > Require header.
> >
> > The saving grace for a B2BUA is that it does not need to understand the
> > extension in detail.  It just needs to know whether it can forward a
request
> > that contains a given Require header.  This can be easily provisioned
using
> > some sort of ASCII configuration file, and doesn't require any new code
etc.
> > A B2BUA could even routinely download the list of acceptable Require
headers
> > from some sort of server.  (A smart B2BUA could even post back a set of
> > unknown Require headers that it has seen recently.)
>
> This is a vast over-simplification.
>
> As people have been saying over and over in this thread, there is no
> such thing as a "generic b2bua" as you seem to be implying. What is
> right in a b2bua depends entirely on what its doing. To be honest, I
> can't think of any b2bua for which the processing you describe above is
> sensible. As I mentioned in a previous note, a b2bua deviates from a
> proxy by doing some featurs that are normally associated with a UA. That
> b2bua will need to process extensions (i.e., implement them) that are
> associated with that feature.
>
> I am at the point where I agree with the sentiment raised by others,
> that we should remove the term b2bua entirely from the specification. It
> has caused more confusion than anything else.
>
> -Jonathan R.
> --
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>


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From: "Mahey, Sonit" <Sonit.Mahey@icn.siemens.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        Pete Cordell
	 <pete@tech-know-ware.com>
Cc: Christian Huitema <huitema@windows.microsoft.com>, sip@ietf.org
Subject: RE: [Sip] B2BUA - Security
Date: Thu, 5 Dec 2002 12:14:12 -0500 
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I agree with Jonathan.

That brings up the question:
Is NAT traversal for encrypted SIP traffic addressed anywhere? 

- sonit

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Thursday, December 05, 2002 11:28 AM
> To: Pete Cordell
> Cc: Christian Huitema; sip@ietf.org
> Subject: Re: [Sip] B2BUA - Security
> 
> 
> 
> 
> Pete Cordell wrote:
> 
> > I would say that encrypting and / or integrity checking of SDP alone
> > achieves absolutely no security benefit.
> 
> Really?? Wow. I actually think it prevents the most 
> devastating attacks 
> against SIP. These include:
> 
> 1. modifying SDP in transit to point media streams to an intermediate 
> recording device, allowing attackers to establish wiretaps
> 
> 2. downgrading any SRTP keying done in SDP, so that the media 
> itself is 
> not encrypted or integrity protected (this came up during the mmusic 
> session in IETF as a driver for S/MIME)
> 
> 3. launching a DoS attack by sending everyones voice traffic to the 
> intended target. You get huge amplification gains here. Very 
> appealing 
> for hackers!
> 
> 
> 
> 
> > 
> > SDP really does two things.  It specifies transport 
> addresses and chosen
> > codecs.
> > 
> > Let's assume that you do encrypt the SDP and I can't read it.
> > 
> > I can have a good guess at the transport addresses in use 
> from many of the
> > other fields in SIP such as via lines, To, From, uri, 
> contact, route etc.
> 
> Confidentiality is only one property of security. I am much more 
> concerned about message integrity.
> 
> > (Put another way, encrypting SDP is another form of 
> security by obscurity.
> > And we all know what that means.)
> 
> I have no idea how you would arrive at such a conclusion. Security 
> through obscurity means that information is not cryptographically 
> protected, but encoded with a known process that is just a bit 
> complicated. By definition, this is not encryption. Perhaps you have 
> discovered some new flaws in encryption technology?
> 
> -Jonathan R.
> -- 
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
> _______________________________________________
> Sip mailing list  https://www1.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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Jonathan Rosenberg wrote:
> 
> 
> Pete Cordell wrote:
> 
>> I would say that encrypting and / or integrity checking of SDP alone
>> achieves absolutely no security benefit.
> 
> 
> Really?? Wow. I actually think it prevents the most devastating attacks 
> against SIP. These include:
> 
> 1. modifying SDP in transit to point media streams to an intermediate 
> recording device, allowing attackers to establish wiretaps

True. But if you are worried about this, perhaps you should use
strong end-to-end media security instead of just relying on the
IP addresses to stay unchanged?

If the attacker is there to change the SDP, he can be there to intercept
the cleartext media streams as well.

> 2. downgrading any SRTP keying done in SDP, so that the media itself is 
> not encrypted or integrity protected (this came up during the mmusic 
> session in IETF as a driver for S/MIME)

I think all SRTP keying mechanisms should be written to be secure
against downgrade attacks. And as far as I know, they are. (With the
exception of downgrade between MIKEY and a hypothetical future key
management protocol that also runs within SDP. But that issue is being
addressed as we speak...)

> 3. launching a DoS attack by sending everyones voice traffic to the 
> intended target. You get huge amplification gains here. Very appealing 
> for hackers!

Yes. This is serious. Interesting attack!

Jari

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Subject: RE: [Sip] B2BUA & Max-Forwards
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Pete,

By sheer coincidence at the time I wrote that email I was writting some code to access a media server directly, via a B2BUA ;). Although adding Max-Forwards wouldn't hurt, I could absolutely guarantee the request wouldn't loop so just left the default value in from our "Message Factory". The fact that there is no general B2BUA is my whole point, you need the Max-Forwards to be strictly decreasing across your B2BUA, I don't and can guarantee I don't and still have both sides conform to rfc3261 without risk of looping.

Despite actually using B2BUAs I'll have to add my voice to the "remove all references to B2BUAs" camp. Personally I found the definition easy to comprehend (even the name tells you what it is), but it is obviously a source of confusion; also letting people build two UAs that conform to the spec and linking them together by whatever innovative applications they can think of doesn't strike me as an area suitable for standardisation. 

James

> -----Original Message-----
> From: Pete Cordell [mailto:pete@tech-know-ware.com]
> Sent: 05 December 2002 15:48
> To: James Undery; sip@ietf.org
> Subject: Re: [Sip] B2BUA & Max-Forwards
> 
> 
> James,
> 
> Thanks for your comments.
> 
> I think in general you CAN'T guarantee that an element you 
> route to won't
> proxy (or be a B2BUA).  And therefore loop detection is a problem.
> 
> Propagating Max-Forwards even when you know the precise 
> routing should not
> cause you any problems.
> 
> Pete.
> 
> ----- Original Message -----
> From: "James Undery" <jundery@ubiquity.net>
> To: "Pete Cordell" <pete@tech-know-ware.com>; "Dean Willis"
> <dean.willis@softarmor.com>; <sip@ietf.org>
> Sent: 05 December 2002 13:21
> Subject: RE: [Sip] B2BUA & Max-Forwards
> 
> 
> Hi,
> 
> I think your confusion comes from mixing two email threads; the first
> statement in context says if a specific B2BUA demonstrates header
> transparency it doesn't change headers; the second statement 
> is about B2BUAs
> in general and says they can do whatever they want (without 
> commenting on
> the desirability of their actions).
> 
> Max-Forwards probably are an important case for your 
> particular application,
> in general they may not be required (e.g. routing to an 
> element you can
> guarantee not to proxy).
> 
> Your comment on Max-Forwards highlights well the difficulties faced in
> defining a B2BUA more specifically than the definition in 3261.
> 
> James
> 
> > -----Original Message-----
> > From: Pete Cordell [mailto:pete@tech-know-ware.com]
> > Sent: 05 December 2002 12:10
> > To: Dean Willis; sip@ietf.org
> > Subject: Re: [Sip] B2BUA & Max-Forwards
> >
> >
> > Dean,
> >
> > Firstly, your statement that "The B2BUA doesn't change
> > headers that transit
> > it." seems inconsistent with the statement "It does the same
> > thing as ...
> > ANY 500-pound gorilla does".  I don't understand.
> >
> > Secondly, I think max-forwards is a special case.  It is
> > important for the
> > protection of SIP systems as it prevents loops.
> >
> > Indeed, there is a major difference between a UAS and a B2BUA
> > and that is
> > that the former terminates a request, and the latter
> > terminates a request
> > AND ALSO forwards an associated request (or more).  The 
> impact of this
> > difference should be covered in the spec in the interests of
> > protecting
> > other SIP systems.
> >
> > Pete.
> >
> > ----- Original Message -----
> > From: "Dean Willis" <dean.willis@softarmor.com>
> > To: "'Pete Cordell'" <pete@tech-know-ware.com>; <sip@ietf.org>
> > Sent: 05 December 2002 01:23
> > Subject: RE: [Sip] B2BUA & Max-Forwards
> >
> >
> > > > One issue that the recent B2BUA discussion has raised in my
> > > > mind is what a B2BUA (whatever its decided that should be)
> > > > should do with Max-Forwards.
> > >
> > > It does the same thing as it does with any other header or
> > body part,
> > > which is the same thing that ANY 500-pound gorilla does to
> > somebody's
> > > body parts -- whatever it wants.
> > >
> > > What you're asking is "what is the transparency of this node with
> > > respect to max-forwards", which is "#4 -- Header 
> Transaprency" in my
> > > top-10 list. The specification of a back-to-back user 
> agent doesn't
> > > mandate any level of transparency on this or any other header.
> > >
> > > --
> > > Dean
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.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://www1.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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Dave,

Please see inline...

Pete.

----- Original Message -----
From: "David R. Oran" <oran@cisco.com>
To: "Pete Cordell" <pete@tech-know-ware.com>; "Christian Huitema"
<huitema@windows.microsoft.com>
Cc: <sip@ietf.org>
Sent: 05 December 2002 14:45
Subject: Re: [Sip] B2BUA - Security


> --On Thursday, December 05, 2002 12:55 PM +0000 Pete Cordell
> <pete@tech-know-ware.com> wrote:
>
> > Christian,
> >
> > I'm afraid I think that end-to-end security of SDP is a complete red
> > herring.
> >
> > I would say that encrypting and / or integrity checking of SDP alone
> > achieves absolutely no security benefit.
> >
> > SDP really does two things.  It specifies transport addresses and chosen
> > codecs.
> >
> ...and carries keys for SRTP...hmmmmmmmm........

I assumed the keying used some sort of Diffee-Hellman type technique, and so
this wouldn't be an issue.  I will look into this.  Maybe that area of SDP
usage could be improved.

>
> > Let's assume that you do encrypt the SDP and I can't read it.
> >
> > I can have a good guess at the transport addresses in use from many of
the
> > other fields in SIP such as via lines, To, From, uri, contact, route
etc.
> >
> Uh huh, sure. My experience is that you can only guess with reasonable
> probability by knowing the implementation details of the UAs. I remember
we
> changed the RTP port allocation algorithm on IOS and all of a sudden
people
> started screaming because their ACL filters "stopped working". Trying to
> beat the devil seem a pretty poor direction.

Real time ACL control is very different issue to working out which RTP
stream goes with what SIP session at some point.

>
> > By analysing the RTP traffic, I can easily find out what codec is being
> > used, either because the packet tells me, or from other statistical
> > characteristics.
> >
> Really? Maybe for the standard codecs but why on earth bother?

Your point is?

>
> > I have therefore discovered, with very little effort most of what I
would
> > find useful in SDP.  (Ironically sending stuff to a media relay would
make
> > this slightly harder!!!)
> >
> > If you want security of the media, then you have to encrypt and
integrity
> > check the RTP.
> >
> or put it through an IPSEC tunnel. What's the point?

IPSEC would have trouble with NATs etc.  The point is to encrypt the media
some way.  I'm not seeing your point.

>
> > Encrypting SDP simply burns CPU cycles!!!
> >
> No, it prevents media hijacking by a MITM.

No - not really.  It helps in some MITM situations but it is not a complete
solution.

>
> > (Put another way, encrypting SDP is another form of security by
obscurity.
> > And we all know what that means.)
> >
> Disagree vociferously.
>
> > (Note that it may make sense to encrypt other bodies types, such as
those
> > that contain HTML, but this is a different case.)
> >
> > Pete.
> >
> Dave.
>
>

_______________________________________________
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To: "David R. Oran" <oran@cisco.com>,
        "Dean Willis" <dean.willis@softarmor.com>, <sip@ietf.org>
References: <00cd01c29bfc$e7dd0240$fe0c0c42@txdwillis> <013a01c29c5d$e1591b00$0200000a@tkw> <2986103.1039078949@ORANLT.cisco.com>
Subject: Re: [Sip] B2BUA & Max-Forwards
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Dave,

To save your disagreeableness, can we say something like:

"A B2BUA MUST perform loop prevention.  One way to do this is to propagate
the Max-Forwards header from the incoming request to the outgoing request in
the same way that a proxy does."

It's the loops that I'm worried about, not the Max-Forwards per se.

This addresses my concern, and I think James' and possibly yours.

Pete.

----- Original Message -----
From: "David R. Oran" <oran@cisco.com>
To: "Pete Cordell" <pete@tech-know-ware.com>; "Dean Willis"
<dean.willis@softarmor.com>; <sip@ietf.org>
Sent: 05 December 2002 14:02
Subject: Re: [Sip] B2BUA & Max-Forwards


> --On Thursday, December 05, 2002 12:09 PM +0000 Pete Cordell
> <pete@tech-know-ware.com> wrote:
>
> > Dean,
> >
> > Firstly, your statement that "The B2BUA doesn't change headers that
> > transit it." seems inconsistent with the statement "It does the same
> > thing as ... ANY 500-pound gorilla does".  I don't understand.
> >
> > Secondly, I think max-forwards is a special case.  It is important for
the
> > protection of SIP systems as it prevents loops.
> >
> Once a dialog is terminated at a UAS it's up to the application to do loop
> prevention. SIP is not involved any more. The application could compute a
> value of Max-forwards for initiating outgoing calls based on incoming
calls
> which preserves the "strictly decreasing" invariant, or alternatively it
> could just use the default MAx-Forwards in outgoing requests and execute a
> contant memory loop suppression algorithm such as Knuth's even-odd
> iterator. In either event the SIP spec(s) need say nothing about this.
>
> > Indeed, there is a major difference between a UAS and a B2BUA and that
is
> > that the former terminates a request, and the latter terminates a
request
> > AND ALSO forwards an associated request (or more).
> I disagree. A UAS is a UAS. It's behavior is completely specified by
> RFC3261 (unless there's a hole in the spec of course...). What it means to
> originate (NOT FORWARD!!!!) an "associated request" is completely
> application dependent. If I program my child's phone to initiate a SIP
call
> to my surrepititious recording device every time it receives a call and
> give that device first chance to over-ride policy of the child's phone is
> the phone a B2BUA? Yes, no, maybe so, but who cares? Tell me what the
> applciation is and I can write some specs for that application and how it
> has to originate, terminate and process SIP transactions to do that
> application.
>
> I think the botom line is that there is not a single, well defined
> appliation that one can point to and say "that's a B2BUA".
>
> > The impact of this
> > difference should be covered in the spec in the interests of protecting
> > other SIP systems.
> >
> I disagree.
>
> > Pete.
> >
> > ----- Original Message -----
> > From: "Dean Willis" <dean.willis@softarmor.com>
> > To: "'Pete Cordell'" <pete@tech-know-ware.com>; <sip@ietf.org>
> > Sent: 05 December 2002 01:23
> > Subject: RE: [Sip] B2BUA & Max-Forwards
> >
> >
> >> > One issue that the recent B2BUA discussion has raised in my
> >> > mind is what a B2BUA (whatever its decided that should be)
> >> > should do with Max-Forwards.
> >>
> >> It does the same thing as it does with any other header or body part,
> >> which is the same thing that ANY 500-pound gorilla does to somebody's
> >> body parts -- whatever it wants.
> >>
> >> What you're asking is "what is the transparency of this node with
> >> respect to max-forwards", which is "#4 -- Header Transaprency" in my
> >> top-10 list. The specification of a back-to-back user agent doesn't
> >> mandate any level of transparency on this or any other header.
> >>
> >> --
> >> Dean
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.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://www1.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
>
> ------------------------
> David R. Oran
> Cisco Systems
> 7 Ladyslipper Lane
> Acton, MA 01720
> Office: +1 978 264 2048
> VoIP: +1 408 571 4576
> Email: oran@cisco.com

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From: "Pete Cordell" <pete@tech-know-ware.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: "Christian Huitema" <huitema@windows.microsoft.com>, <sip@ietf.org>
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Subject: Re: [Sip] B2BUA - Security
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Jonathan,

Please see in-line...

Pete.

----- Original Message -----
From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
To: "Pete Cordell" <pete@tech-know-ware.com>
Cc: "Christian Huitema" <huitema@windows.microsoft.com>; <sip@ietf.org>
Sent: 05 December 2002 16:28
Subject: Re: [Sip] B2BUA - Security


>
>
> Pete Cordell wrote:
>
> > I would say that encrypting and / or integrity checking of SDP alone
> > achieves absolutely no security benefit.
>
> Really?? Wow. I actually think it prevents the most devastating attacks
> against SIP. These include:
>
> 1. modifying SDP in transit to point media streams to an intermediate
> recording device, allowing attackers to establish wiretaps

This is only partially successful.  Encryption of RTP is the only complete
solution.

>
> 2. downgrading any SRTP keying done in SDP, so that the media itself is
> not encrypted or integrity protected (this came up during the mmusic
> session in IETF as a driver for S/MIME)

I don't think you need to encrypt the entire SDP body to prevent key
downgrading.  In the absence of any other contraints that might be a simple
route to go, but if it causes problems (as it is) then we should look at
other ways.

As an aside - This is the sort of reason I would like to get this sort of
thing on the agenda.  It's not that its not possible to enable SIP to meet
these requirements, there's just not a desire to do among the core SIP
people.  I know why they don't want to do it, but I also think that SIP
fails to address a whole class of problem as a result.  In the long run I
think that will be detrimental to SIP.

> 3. launching a DoS attack by sending everyones voice traffic to the
> intended target. You get huge amplification gains here. Very appealing
> for hackers!

Don't need SDP to do that.  I could just rip out the entire SDP and put in a
new version.  I might not get the ports right, but I could probably cause
enough damage to get my kicks.

>
> >
> > SDP really does two things.  It specifies transport addresses and chosen
> > codecs.
> >
> > Let's assume that you do encrypt the SDP and I can't read it.
> >
> > I can have a good guess at the transport addresses in use from many of
the
> > other fields in SIP such as via lines, To, From, uri, contact, route
etc.
>
> Confidentiality is only one property of security. I am much more
> concerned about message integrity.

Integrity of which bits?  And are you sure that maintaining integrity buys
you anything except for in a few special cases?

>
> > (Put another way, encrypting SDP is another form of security by
obscurity.
> > And we all know what that means.)
>
> I have no idea how you would arrive at such a conclusion. Security
> through obscurity means that information is not cryptographically
> protected, but encoded with a known process that is just a bit
> complicated. By definition, this is not encryption.

That's what I meant - yes.  Hiding the SDP does not get you security.

> Perhaps you have
> discovered some new flaws in encryption technology?
>
> -Jonathan R.
> --
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>

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Subject: Re: [Sip] B2BUA & Max-Forwards
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Hello,

"I am at the point where I agree with the sentiment raised by others,
that we should remove the term b2bua entirely from the specification. It
has caused more confusion than anything else."

Yes, this is one possibility. An alternative (as I proposed in an earlier
mail) is to complete the current definition in 3261 saying explicitly that
:
1) A B2BUA handles two or more UAs, each one behaving as UAC/UAS, as
described in 3261 for UAs
2) The SIP protocol specification does not mandate for any behaviour of
such B2BUA, i.e. on how to "link" (or "concatenate") two or more UAs.

Best regards
Juan Carlos





Jonathan Rosenberg <jdrosen@dynamicsoft.com>@ietf.org on 05/12/2002
17:16:52

Sent by:  sip-admin@ietf.org


To:   Pete Cordell <pete@tech-know-ware.com>
cc:   Adam Roach <adam@dynamicsoft.com>, "'Phil Reynolds'"
      <preynolds@ridgewaysystems.com>, sip@ietf.org
Subject:  Re: [Sip] B2BUA & Max-Forwards




Pete Cordell wrote:
> This is correct.  A B2BUA can't blindly forward a request containing a
> Require header.
>
> The saving grace for a B2BUA is that it does not need to understand the
> extension in detail.  It just needs to know whether it can forward a
request
> that contains a given Require header.  This can be easily provisioned
using
> some sort of ASCII configuration file, and doesn't require any new code
etc.
> A B2BUA could even routinely download the list of acceptable Require
headers
> from some sort of server.  (A smart B2BUA could even post back a set of
> unknown Require headers that it has seen recently.)

This is a vast over-simplification.

As people have been saying over and over in this thread, there is no
such thing as a "generic b2bua" as you seem to be implying. What is
right in a b2bua depends entirely on what its doing. To be honest, I
can't think of any b2bua for which the processing you describe above is
sensible. As I mentioned in a previous note, a b2bua deviates from a
proxy by doing some featurs that are normally associated with a UA. That
b2bua will need to process extensions (i.e., implement them) that are
associated with that feature.

I am at the point where I agree with the sentiment raised by others,
that we should remove the term b2bua entirely from the specification. It
has caused more confusion than anything else.

-Jonathan R.
--
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Subject: RE: [Sip] B2BUA - Security
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From: "Christian Huitema" <huitema@windows.microsoft.com>
To: "Mahey, Sonit" <Sonit.Mahey@icn.siemens.com>,
        "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
        "Pete Cordell" <pete@tech-know-ware.com>
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> From: Mahey, Sonit [mailto:Sonit.Mahey@icn.siemens.com]
> Sent: Thursday, December 05, 2002 9:14 AM
>
> I agree with Jonathan.
> 
> That brings up the question:
> Is NAT traversal for encrypted SIP traffic addressed anywhere?

There are two possibilities. If your application only uses UDP, it is
possible to use IPv4 and STUN to find out the "outside ports" for your
UDP traffic (check draft-ietf-midcom-stun-03.txt); you may need to use
the "a:rtcp" convention to encode port numbers in SDP (check
draft-ietf-mmusic-sdp4nat-03.txt). If you need to also support TCP, or
use IPSEC, or generally do away with the complications of NAT, the best
solution is to just move to IPv6; see Teredo
(draft-ietf-ngtrans-shipworm-08.txt) for one possible way to carry IPv6
across NAT, and 6to4 (RFC 3056 & 3068) for a possible way to upgrade the
NAT and make it an IPv6 router.

-- Christian Huitema
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Cc: "David R. Oran" <oran@cisco.com>,
        "Dean Willis" <dean.willis@softarmor.com>, <sip@ietf.org>
Subject: Re: [Sip] B2BUA & Max-Forwards
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Pete Cordell writes:
 > Dave,
 > 
 > To save your disagreeableness, can we say something like:
 > 
 > "A B2BUA MUST perform loop prevention.  One way to do this is to propagate
 > the Max-Forwards header from the incoming request to the outgoing request in
 > the same way that a proxy does."
 > 
 > It's the loops that I'm worried about, not the Max-Forwards per se.

   And prevent me from building a SIP Cyclotron?

   Hell no.

	Mike
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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



From mailnull@www1.ietf.org  Thu Dec  5 13:00:54 2002
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From: "Mahey, Sonit" <Sonit.Mahey@icn.siemens.com>
To: "'Christian Huitema'" <huitema@windows.microsoft.com>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Pete Cordell
	 <pete@tech-know-ware.com>
Cc: sip@ietf.org
Subject: RE: [Sip] B2BUA - Security
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Thanks, Christian.

Assuming IPv4 addresses and without delving into the draft RFC on STUN:
The SDP contains the RTP/RTCP port numbers that are private and these are
also required to be NAT'd. Now, if the SDP, which forms part of the SIP
body, is encrypted, how will RTP/RTCP ports be translated?

regards,
- sonit

> -----Original Message-----
> From: Christian Huitema [mailto:huitema@windows.microsoft.com]
> Sent: Thursday, December 05, 2002 12:54 PM
> To: Mahey, Sonit; Jonathan Rosenberg; Pete Cordell
> Cc: sip@ietf.org
> Subject: RE: [Sip] B2BUA - Security
> 
> 
> > From: Mahey, Sonit [mailto:Sonit.Mahey@icn.siemens.com]
> > Sent: Thursday, December 05, 2002 9:14 AM
> >
> > I agree with Jonathan.
> > 
> > That brings up the question:
> > Is NAT traversal for encrypted SIP traffic addressed anywhere?
> 
> There are two possibilities. If your application only uses UDP, it is
> possible to use IPv4 and STUN to find out the "outside ports" for your
> UDP traffic (check draft-ietf-midcom-stun-03.txt); you may need to use
> the "a:rtcp" convention to encode port numbers in SDP (check
> draft-ietf-mmusic-sdp4nat-03.txt). If you need to also support TCP, or
> use IPSEC, or generally do away with the complications of 
> NAT, the best
> solution is to just move to IPv6; see Teredo
> (draft-ietf-ngtrans-shipworm-08.txt) for one possible way to 
> carry IPv6
> across NAT, and 6to4 (RFC 3056 & 3068) for a possible way to 
> upgrade the
> NAT and make it an IPv6 router.
> 
> -- Christian Huitema
> 
_______________________________________________
Sip mailing list  https://www1.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: Thu, 05 Dec 2002 11:56:40 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sip] B2BUA
In-reply-to: <013b01c29c5d$e22bad40$0200000a@tkw>
To: "'Pete Cordell'" <pete@tech-know-ware.com>
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Can you document any _outages_ of the _Internet_? I don't recall any.

Please provide references.

Thanks, Henry

> -----Original Message-----
> From: Pete Cordell [mailto:pete@tech-know-ware.com] 
> Sent: Thursday, December 05, 2002 6:25 AM
> To: Henry Sinnreich
> Cc: sip@ietf.org
> Subject: Re: [Sip] B2BUA
> 
> 
> Henry,
> 
> The Internet has had a few famous outages as well.  Besides, 
> if all the telco traffic was put over IP, with today's level 
> of QoS on the Internet, it would have been brought to its knees also.
> 
> To say that one is all goodness and light and the other all 
> evil doesn't do us as a community any good as when customers 
> find out there are problems they tend not to believe anything 
> else we say!
> 
> Only vapourware is flawless.  Most people know that.  Respect 
> the intelligence of your customer, or they will find somebody 
> that does!!!
> 
> Pete.
> 
> ----- Original Message -----
> From: "Henry Sinnreich" <Henry.Sinnreich@wcom.com>
> To: "'Phil Reynolds'" <preynolds@ridgewaysystems.com>; <sip@ietf.org>
> Sent: 04 December 2002 19:02
> Subject: RE: [Sip] B2BUA
> 
> 
> > >The telecoms people are clever enough to have thought
> > >about that and build redundancy, fault tolerance and failover into
> > their switches.
> >
> > This has proven unfortunately not to be the case on 
> September 11 when 
> > telecom service failed for days, while Internet traffic continued 
> > smoothly. I suggest you to look at the central control model of the 
> > switch network versus the e2e architecture of the Internet. SIP 
> > enabled networks are also fully distributed and the e2e control 
> > principle maintained - that's why state in placed in SIP endpoints.
> >
> > I am not sure if a few words here can give full justice to the 
> > superior resilience of the Internet and to SIP enabled services,...
> >
> > Thanks, Henry
> >
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Phil 
> > Reynolds
> > Sent: Wednesday, December 04, 2002 7:37 AM
> > To: sip@ietf.org
> > Subject: RE: [Sip] B2BUA
> >
> >
> > If a telephone switch crashes you loose all calls...  The telecoms 
> > people are clever enough to have thought about that and build 
> > redundancy, fault tolerance and failover into their switches. Maybe 
> > some of us IP-telecoms people have thought of similar 
> techniques for 
> > SIP intermediaries? -----Original Message-----
> > From: Jiri Kuthan [mailto:jiri.kuthan@fokus.fraunhofer.de]
> > Sent: 04 December 2002 12:32
> > To: Christian Huitema; Roy, Radhika R, ALASO; 
> mbarnes@nortelnetworks.com
> >
> > Cc: dean.willis@softarmor.com; Juan-Carlos.Rojas@alcatel.fr; 
> > sip@ietf.org
> > Subject: RE: [Sip] B2BUA
> > [...snip...]
> > I think there are more issues with B2BUA. A B2BUA's failure affects 
> > all current calls :-( -Jiri
> > _______________________________________________
> > Sip mailing list  https://www1.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://www1.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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This list is for NEW development of the core SIP Protocol
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From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sip] B2BUA
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Rainer, the fact is that people could communicate on 9/11/01 using IM
and SIP telephony from Columbia University, while the telecom phone
service, both landline and mobile was down for days.

There is very little or very much to comment on these facts.

Henry

> -----Original Message-----
> From: Liebhart Rainer [mailto:rainer.liebhart@siemens.com] 
> Sent: Thursday, December 05, 2002 7:46 AM
> To: 'Pete Cordell'; Henry Sinnreich
> Cc: sip@ietf.org
> Subject: AW: [Sip] B2BUA
> 
> 
> Henry,
> 
> unfortunately you are comparing apples and pears when stating 
> that the telecom service failed for days after Sept 11, while 
> Internet traffic continued smoothly. The mentioned "telefon 
> service" consists of thousands of (real-time) calls to 
> emergency centres while the Internet traffic consists mainly 
> of requests to the Web pages of CNN and others (with bad 
> response times) to retrieve *information* on the status of 
> the catastrophe.
> 
> Rainer
> 
> 
> -----Ursprüngliche Nachricht-----
> Von: Pete Cordell [mailto:pete@tech-know-ware.com]
> Gesendet: Donnerstag, 5. Dezember 2002 13:25
> An: Henry Sinnreich
> Cc: sip@ietf.org
> Betreff: Re: [Sip] B2BUA
> 
> 
> Henry,
> 
> The Internet has had a few famous outages as well.  Besides, 
> if all the telco traffic was put over IP, with today's level 
> of QoS on the Internet, it would have been brought to its knees also.
> 
> To say that one is all goodness and light and the other all 
> evil doesn't do us as a community any good as when customers 
> find out there are problems they tend not to believe anything 
> else we say!
> 
> Only vapourware is flawless.  Most people know that.  Respect 
> the intelligence of your customer, or they will find somebody 
> that does!!!
> 
> Pete.
> 
> ----- Original Message -----
> From: "Henry Sinnreich" <Henry.Sinnreich@wcom.com>
> To: "'Phil Reynolds'" <preynolds@ridgewaysystems.com>; <sip@ietf.org>
> Sent: 04 December 2002 19:02
> Subject: RE: [Sip] B2BUA
> 
> 
> > >The telecoms people are clever enough to have thought
> > >about that and build redundancy, fault tolerance and failover into
> > their switches.
> >
> > This has proven unfortunately not to be the case on 
> September 11 when 
> > telecom service failed for days, while Internet traffic continued 
> > smoothly. I suggest you to look at the central control model of the 
> > switch network versus the e2e architecture of the Internet. SIP 
> > enabled networks are also fully distributed and the e2e control 
> > principle maintained - that's why state in placed in SIP endpoints.
> >
> > I am not sure if a few words here can give full justice to the 
> > superior resilience of the Internet and to SIP enabled services,...
> >
> > Thanks, Henry
> >
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Phil 
> > Reynolds
> > Sent: Wednesday, December 04, 2002 7:37 AM
> > To: sip@ietf.org
> > Subject: RE: [Sip] B2BUA
> >
> >
> > If a telephone switch crashes you loose all calls...  The telecoms 
> > people are clever enough to have thought about that and build 
> > redundancy, fault tolerance and failover into their switches. Maybe 
> > some of us IP-telecoms people have thought of similar 
> techniques for 
> > SIP intermediaries? -----Original Message-----
> > From: Jiri Kuthan [mailto:jiri.kuthan@fokus.fraunhofer.de]
> > Sent: 04 December 2002 12:32
> > To: Christian Huitema; Roy, Radhika R, ALASO; 
> mbarnes@nortelnetworks.com
> >
> > Cc: dean.willis@softarmor.com; Juan-Carlos.Rojas@alcatel.fr; 
> > sip@ietf.org
> > Subject: RE: [Sip] B2BUA
> > [...snip...]
> > I think there are more issues with B2BUA. A B2BUA's failure affects 
> > all current calls :-( -Jiri
> > _______________________________________________
> > Sip mailing list  https://www1.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://www1.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://www1.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 mailnull@www1.ietf.org  Thu Dec  5 13:35:18 2002
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From: Adam Roach <adam@dynamicsoft.com>
To: "'Mahey, Sonit'" <Sonit.Mahey@icn.siemens.com>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>,
        Pete Cordell <pete@tech-know-ware.com>
Cc: Christian Huitema <huitema@windows.microsoft.com>, sip@ietf.org
Subject: RE: [Sip] B2BUA - Security
Date: Thu, 5 Dec 2002 12:33:45 -0600 
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Yes. See STUN.

/a

> -----Original Message-----
> From: Mahey, Sonit [mailto:Sonit.Mahey@icn.siemens.com]
> Sent: Thursday, December 05, 2002 11:14
> To: 'Jonathan Rosenberg'; Pete Cordell
> Cc: Christian Huitema; sip@ietf.org
> Subject: RE: [Sip] B2BUA - Security
> 
> 
> I agree with Jonathan.
> 
> That brings up the question:
> Is NAT traversal for encrypted SIP traffic addressed anywhere? 
> 
> - sonit
> 
> > -----Original Message-----
> > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > Sent: Thursday, December 05, 2002 11:28 AM
> > To: Pete Cordell
> > Cc: Christian Huitema; sip@ietf.org
> > Subject: Re: [Sip] B2BUA - Security
> > 
> > 
> > 
> > 
> > Pete Cordell wrote:
> > 
> > > I would say that encrypting and / or integrity checking 
> of SDP alone
> > > achieves absolutely no security benefit.
> > 
> > Really?? Wow. I actually think it prevents the most 
> > devastating attacks 
> > against SIP. These include:
> > 
> > 1. modifying SDP in transit to point media streams to an 
> intermediate 
> > recording device, allowing attackers to establish wiretaps
> > 
> > 2. downgrading any SRTP keying done in SDP, so that the media 
> > itself is 
> > not encrypted or integrity protected (this came up during 
> the mmusic 
> > session in IETF as a driver for S/MIME)
> > 
> > 3. launching a DoS attack by sending everyones voice traffic to the 
> > intended target. You get huge amplification gains here. Very 
> > appealing 
> > for hackers!
> > 
> > 
> > 
> > 
> > > 
> > > SDP really does two things.  It specifies transport 
> > addresses and chosen
> > > codecs.
> > > 
> > > Let's assume that you do encrypt the SDP and I can't read it.
> > > 
> > > I can have a good guess at the transport addresses in use 
> > from many of the
> > > other fields in SIP such as via lines, To, From, uri, 
> > contact, route etc.
> > 
> > Confidentiality is only one property of security. I am much more 
> > concerned about message integrity.
> > 
> > > (Put another way, encrypting SDP is another form of 
> > security by obscurity.
> > > And we all know what that means.)
> > 
> > I have no idea how you would arrive at such a conclusion. Security 
> > through obscurity means that information is not cryptographically 
> > protected, but encoded with a known process that is just a bit 
> > complicated. By definition, this is not encryption. Perhaps 
> you have 
> > discovered some new flaws in encryption technology?
> > 
> > -Jonathan R.
> > -- 
> > Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> > Chief Scientist                             First Floor
> > dynamicsoft                                 East Hanover, NJ 07936
> > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > http://www.jdrosen.net                      PHONE: (973) 952-5000
> > http://www.dynamicsoft.com
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.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://www1.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] B2BUA & Max-Forwards
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Hi, 

inline.

> "I am at the point where I agree with the sentiment raised by others,
> that we should remove the term b2bua entirely from the 
> specification. It
> has caused more confusion than anything else."
> 
> Yes, this is one possibility. An alternative (as I proposed 
> in an earlier
> mail) is to complete the current definition in 3261 saying 
> explicitly that
> :
> 1) A B2BUA handles two or more UAs, each one behaving as UAC/UAS, as
> described in 3261 for UAs

I don't get this.  Does two or more mean, for example, four?  And does that in turn make it a B2BB2BUA? ;)

The only minor clarification I think might be in order for RFC3261 is that it's not just a UAC/UAS concatenation, but a pair of UAC/UAS concatenations.

However, the fact that the traffic in this thread alone equals to a year's worth of emails on the SIMPLE list suggests that no clarification is enough clarification.

So I agree with the other people in that getting rid of the definition altogether is the best option.  After all, if it looks like a UA to the outside world, it must be a UA.

> 2) The SIP protocol specification does not mandate for any 
> behaviour of
> such B2BUA, i.e. on how to "link" (or "concatenate") two or more UAs.

Yes, probably the reason why the definition for a B2BUA is also useless.

Cheers,
Aki
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Date: Thu, 05 Dec 2002 14:10:34 -0500
From: David Oran <oran@cisco.com>
To: Pete Cordell <pete@tech-know-ware.com>,
        Dean Willis <dean.willis@softarmor.com>, sip@ietf.org
Subject: Re: [Sip] B2BUA & Max-Forwards
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--On Thursday, December 05, 2002 5:24 PM +0000 Pete Cordell 
<pete@tech-know-ware.com> wrote:

> Dave,
>
> To save your disagreeableness, can we say something like:
>
> "A B2BUA MUST perform loop prevention.  One way to do this is to propagate
> the Max-Forwards header from the incoming request to the outgoing request
> in the same way that a proxy does."
>
I don't understand what it means to have normative language for a beast 
which is fundamentally undefined. Suppose I write a piece of code which 
accepts SIP requests, issues tachyon commands to alpha centauri, and 
meanwhile receives routing instructions from a chronosynclastic 
infundibulum (for the unintiated, that is a device which effects reverse 
causality based on the message to alpha centauri whose response would not 
normally arrive for 7.8 years). It then uses those routing instructions to 
formulate an outbound SIP request to the destination received through the 
chronosynclastic infundibulum.

Question: is this application a B2BUA and hence required to have a 
Max-Forwards value in the request whose value is strictly less than the 
value received in the SIP request? Beats me.

> It's the loops that I'm worried about, not the Max-Forwards per se.
>
There are many ways to suppress loops. Using application semantics is 
actually safer than lower-layer protocol machinery because of the hazards 
introduced by protocol translation (e.g. the PSTN hop case mentioned 
earlier, or a SIP<->H.323 conversion somewhere else).

> This addresses my concern, and I think James' and possibly yours.
>
Sorry, not mine. From where I sit, the less said about B2BUAs the better, 
including asymptotically toward zero.

Cheers, Dave.

> Pete.
>
> ----- Original Message -----
> From: "David R. Oran" <oran@cisco.com>
> To: "Pete Cordell" <pete@tech-know-ware.com>; "Dean Willis"
> <dean.willis@softarmor.com>; <sip@ietf.org>
> Sent: 05 December 2002 14:02
> Subject: Re: [Sip] B2BUA & Max-Forwards
>
>
>> --On Thursday, December 05, 2002 12:09 PM +0000 Pete Cordell
>> <pete@tech-know-ware.com> wrote:
>>
>> > Dean,
>> >
>> > Firstly, your statement that "The B2BUA doesn't change headers that
>> > transit it." seems inconsistent with the statement "It does the same
>> > thing as ... ANY 500-pound gorilla does".  I don't understand.
>> >
>> > Secondly, I think max-forwards is a special case.  It is important for
> the
>> > protection of SIP systems as it prevents loops.
>> >
>> Once a dialog is terminated at a UAS it's up to the application to do
>> loop prevention. SIP is not involved any more. The application could
>> compute a value of Max-forwards for initiating outgoing calls based on
>> incoming
> calls
>> which preserves the "strictly decreasing" invariant, or alternatively it
>> could just use the default MAx-Forwards in outgoing requests and execute
>> a contant memory loop suppression algorithm such as Knuth's even-odd
>> iterator. In either event the SIP spec(s) need say nothing about this.
>>
>> > Indeed, there is a major difference between a UAS and a B2BUA and that
> is
>> > that the former terminates a request, and the latter terminates a
> request
>> > AND ALSO forwards an associated request (or more).
>> I disagree. A UAS is a UAS. It's behavior is completely specified by
>> RFC3261 (unless there's a hole in the spec of course...). What it means
>> to originate (NOT FORWARD!!!!) an "associated request" is completely
>> application dependent. If I program my child's phone to initiate a SIP
> call
>> to my surrepititious recording device every time it receives a call and
>> give that device first chance to over-ride policy of the child's phone is
>> the phone a B2BUA? Yes, no, maybe so, but who cares? Tell me what the
>> applciation is and I can write some specs for that application and how it
>> has to originate, terminate and process SIP transactions to do that
>> application.
>>
>> I think the botom line is that there is not a single, well defined
>> appliation that one can point to and say "that's a B2BUA".
>>
>> > The impact of this
>> > difference should be covered in the spec in the interests of protecting
>> > other SIP systems.
>> >
>> I disagree.
>>
>> > Pete.
>> >
>> > ----- Original Message -----
>> > From: "Dean Willis" <dean.willis@softarmor.com>
>> > To: "'Pete Cordell'" <pete@tech-know-ware.com>; <sip@ietf.org>
>> > Sent: 05 December 2002 01:23
>> > Subject: RE: [Sip] B2BUA & Max-Forwards
>> >
>> >
>> >> > One issue that the recent B2BUA discussion has raised in my
>> >> > mind is what a B2BUA (whatever its decided that should be)
>> >> > should do with Max-Forwards.
>> >>
>> >> It does the same thing as it does with any other header or body part,
>> >> which is the same thing that ANY 500-pound gorilla does to somebody's
>> >> body parts -- whatever it wants.
>> >>
>> >> What you're asking is "what is the transparency of this node with
>> >> respect to max-forwards", which is "#4 -- Header Transaprency" in my
>> >> top-10 list. The specification of a back-to-back user agent doesn't
>> >> mandate any level of transparency on this or any other header.
>> >>
>> >> --
>> >> Dean
>> >>
>> >> _______________________________________________
>> >> Sip mailing list  https://www1.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://www1.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
>>
>> ------------------------
>> David R. Oran
>> Cisco Systems
>> 7 Ladyslipper Lane
>> Acton, MA 01720
>> Office: +1 978 264 2048
>> VoIP: +1 408 571 4576
>> Email: oran@cisco.com
>


_______________________________________________
Sip mailing list  https://www1.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: Gordon Ledgard <gledgard@iperia.com>
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        Dean Willis <dean.willis@softarmor.com>, sip@ietf.org
Subject: RE: [Sip] B2BUA & Max-Forwards
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inline - 

> -----Original Message-----
> From: David Oran [mailto:oran@cisco.com]
> Sent: Thursday, December 05, 2002 2:11 PM
> To: Pete Cordell; Dean Willis; sip@ietf.org
> Subject: Re: [Sip] B2BUA & Max-Forwards
> 
> 
> --On Thursday, December 05, 2002 5:24 PM +0000 Pete Cordell 
> <pete@tech-know-ware.com> wrote:
> 
> > Dave,
> >
> > To save your disagreeableness, can we say something like:
> >
> > "A B2BUA MUST perform loop prevention.  One way to do this 
> is to propagate
> > the Max-Forwards header from the incoming request to the 
> outgoing request
> > in the same way that a proxy does."
> >
> I don't understand what it means to have normative language 
> for a beast 
> which is fundamentally undefined. Suppose I write a piece of 
> code which 
> accepts SIP requests, issues tachyon commands to alpha centauri, and 
> meanwhile receives routing instructions from a chronosynclastic 
> infundibulum (for the unintiated, that is a device which 
> effects reverse 
> causality based on the message to alpha centauri whose 
> response would not 
> normally arrive for 7.8 years). It then uses those routing 
> instructions to 
> formulate an outbound SIP request to the destination received 
> through the 
> chronosynclastic infundibulum.
> 
> Question: is this application a B2BUA and hence required to have a 
> Max-Forwards value in the request whose value is strictly 
> less than the 
> value received in the SIP request? Beats me.

Only if that's the way it wants to do it. You can defeat
the loops howsoever you want.

> 
> > It's the loops that I'm worried about, not the Max-Forwards per se.
> >
> There are many ways to suppress loops. Using application semantics is 
> actually safer than lower-layer protocol machinery because of 
> the hazards 
> introduced by protocol translation (e.g. the PSTN hop case mentioned 
> earlier, or a SIP<->H.323 conversion somewhere else).
> 
> > This addresses my concern, and I think James' and possibly yours.
> >
> Sorry, not mine. From where I sit, the less said about B2BUAs 
> the better, 
> including asymptotically toward zero.
> 
> Cheers, Dave.
> 
> > Pete.
> >
> > ----- Original Message -----
> > From: "David R. Oran" <oran@cisco.com>
> > To: "Pete Cordell" <pete@tech-know-ware.com>; "Dean Willis"
> > <dean.willis@softarmor.com>; <sip@ietf.org>
> > Sent: 05 December 2002 14:02
> > Subject: Re: [Sip] B2BUA & Max-Forwards
> >
> >
> >> --On Thursday, December 05, 2002 12:09 PM +0000 Pete Cordell
> >> <pete@tech-know-ware.com> wrote:
> >>
> >> > Dean,
> >> >
> >> > Firstly, your statement that "The B2BUA doesn't change 
> headers that
> >> > transit it." seems inconsistent with the statement "It 
> does the same
> >> > thing as ... ANY 500-pound gorilla does".  I don't understand.
> >> >
> >> > Secondly, I think max-forwards is a special case.  It is 
> important for
> > the
> >> > protection of SIP systems as it prevents loops.
> >> >
> >> Once a dialog is terminated at a UAS it's up to the 
> application to do
> >> loop prevention. SIP is not involved any more. The 
> application could
> >> compute a value of Max-forwards for initiating outgoing 
> calls based on
> >> incoming
> > calls
> >> which preserves the "strictly decreasing" invariant, or 
> alternatively it
> >> could just use the default MAx-Forwards in outgoing 
> requests and execute
> >> a contant memory loop suppression algorithm such as 
> Knuth's even-odd
> >> iterator. In either event the SIP spec(s) need say nothing 
> about this.
> >>
> >> > Indeed, there is a major difference between a UAS and a 
> B2BUA and that
> > is
> >> > that the former terminates a request, and the latter terminates a
> > request
> >> > AND ALSO forwards an associated request (or more).
> >> I disagree. A UAS is a UAS. It's behavior is completely 
> specified by
> >> RFC3261 (unless there's a hole in the spec of course...). 
> What it means
> >> to originate (NOT FORWARD!!!!) an "associated request" is 
> completely
> >> application dependent. If I program my child's phone to 
> initiate a SIP
> > call
> >> to my surrepititious recording device every time it 
> receives a call and
> >> give that device first chance to over-ride policy of the 
> child's phone is
> >> the phone a B2BUA? Yes, no, maybe so, but who cares? Tell 
> me what the
> >> applciation is and I can write some specs for that 
> application and how it
> >> has to originate, terminate and process SIP transactions to do that
> >> application.
> >>
> >> I think the botom line is that there is not a single, well defined
> >> appliation that one can point to and say "that's a B2BUA".
> >>
> >> > The impact of this
> >> > difference should be covered in the spec in the 
> interests of protecting
> >> > other SIP systems.
> >> >
> >> I disagree.
> >>
> >> > Pete.
> >> >
> >> > ----- Original Message -----
> >> > From: "Dean Willis" <dean.willis@softarmor.com>
> >> > To: "'Pete Cordell'" <pete@tech-know-ware.com>; <sip@ietf.org>
> >> > Sent: 05 December 2002 01:23
> >> > Subject: RE: [Sip] B2BUA & Max-Forwards
> >> >
> >> >
> >> >> > One issue that the recent B2BUA discussion has raised in my
> >> >> > mind is what a B2BUA (whatever its decided that should be)
> >> >> > should do with Max-Forwards.
> >> >>
> >> >> It does the same thing as it does with any other header 
> or body part,
> >> >> which is the same thing that ANY 500-pound gorilla does 
> to somebody's
> >> >> body parts -- whatever it wants.
> >> >>
> >> >> What you're asking is "what is the transparency of this 
> node with
> >> >> respect to max-forwards", which is "#4 -- Header 
> Transaprency" in my
> >> >> top-10 list. The specification of a back-to-back user 
> agent doesn't
> >> >> mandate any level of transparency on this or any other header.
> >> >>
> >> >> --
> >> >> Dean
> >> >>
> >> >> _______________________________________________
> >> >> Sip mailing list  https://www1.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://www1.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
> >>
> >> ------------------------
> >> David R. Oran
> >> Cisco Systems
> >> 7 Ladyslipper Lane
> >> Acton, MA 01720
> >> Office: +1 978 264 2048
> >> VoIP: +1 408 571 4576
> >> Email: oran@cisco.com
> >
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.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://www1.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] B2BUA
From: Wilhelm Wimmreuter <wilhelm@wimmreuter.de>
To: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Cc: "'Liebhart Rainer'" <rainer.liebhart@siemens.com>,
        "'Pete Cordell'"
	 <pete@tech-know-ware.com>, sip@ietf.org
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Hi Henry,
cant resist throwing my nonsense in as well

On Thu, 2002-12-05 at 19:16, Henry Sinnreich wrote:
> Rainer, the fact is that people could communicate on 9/11/01 using IM
> and SIP telephony from Columbia University, while the telecom phone
> service, both landline and mobile was down for days.
> 
> There is very little or very much to comment on these facts.
> 

Did you forget, that a qualified busy/congestion tone, even for
some days, does not reflect a system outage ;-)

 ... If required, call me on my SIP phone to sort that out. It
will work as usual :-)

Willi






_______________________________________________
Sip mailing list  https://www1.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: Thu, 05 Dec 2002 15:16:27 -0500
From: David Oran <oran@cisco.com>
To: Gordon Ledgard <gledgard@iperia.com>,
        Pete Cordell <pete@tech-know-ware.com>,
        Dean Willis <dean.willis@softarmor.com>, sip@ietf.org
Subject: RE: [Sip] B2BUA & Max-Forwards
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--On Thursday, December 05, 2002 3:03 PM -0500 Gordon Ledgard 
<gledgard@iperia.com> wrote:
> inline -
>
>> -----Original Message-----
>> From: David Oran [mailto:oran@cisco.com]
>> Sent: Thursday, December 05, 2002 2:11 PM
>> To: Pete Cordell; Dean Willis; sip@ietf.org
>> Subject: Re: [Sip] B2BUA & Max-Forwards
>>
>>
>> --On Thursday, December 05, 2002 5:24 PM +0000 Pete Cordell
>> <pete@tech-know-ware.com> wrote:
>>
>> > Dave,
>> >
>> > To save your disagreeableness, can we say something like:
>> >
>> > "A B2BUA MUST perform loop prevention.  One way to do this
>> is to propagate
>> > the Max-Forwards header from the incoming request to the
>> outgoing request
>> > in the same way that a proxy does."
>> >
>> I don't understand what it means to have normative language
>> for a beast
>> which is fundamentally undefined. Suppose I write a piece of
>> code which
>> accepts SIP requests, issues tachyon commands to alpha centauri, and
>> meanwhile receives routing instructions from a chronosynclastic
>> infundibulum (for the unintiated, that is a device which
>> effects reverse
>> causality based on the message to alpha centauri whose
>> response would not
>> normally arrive for 7.8 years). It then uses those routing
>> instructions to
>> formulate an outbound SIP request to the destination received
>> through the
>> chronosynclastic infundibulum.
>>
>> Question: is this application a B2BUA and hence required to have a
>> Max-Forwards value in the request whose value is strictly
>> less than the
>> value received in the SIP request? Beats me.
>
> Only if that's the way it wants to do it. You can defeat
> the loops howsoever you want.
>
Ok, then how about the following wording: "a B2BUA MUST prevent loops and 
it MUST NOT behave in such as way as to cause broadcast storms, or 
contribute to any exponential message explosion of SIP or any other 
messages. In addition, it MUST comply will all local and national fire 
codes, including the Underwriter's Laboratory flaming oil test*.

*Note: this is important guidance for implementers. I once had a device I 
helped design fail the flaming oil test for one of the six possible 
mounting orientations of the box (it passed in the other 5 orientations). 
The orientation it failed in was with all of the I/O sockets facing the 
wall (making it impossible to plug anything into the box, but that didn't 
matter to the testers).

Tiring of this subject, as should be apparent...Dave.

>>
>> > It's the loops that I'm worried about, not the Max-Forwards per se.
>> >
>> There are many ways to suppress loops. Using application semantics is
>> actually safer than lower-layer protocol machinery because of
>> the hazards
>> introduced by protocol translation (e.g. the PSTN hop case mentioned
>> earlier, or a SIP<->H.323 conversion somewhere else).
>>
>> > This addresses my concern, and I think James' and possibly yours.
>> >
>> Sorry, not mine. From where I sit, the less said about B2BUAs
>> the better,
>> including asymptotically toward zero.
>>
>> Cheers, Dave.
>>
>> > Pete.
>> >
>> > ----- Original Message -----
>> > From: "David R. Oran" <oran@cisco.com>
>> > To: "Pete Cordell" <pete@tech-know-ware.com>; "Dean Willis"
>> > <dean.willis@softarmor.com>; <sip@ietf.org>
>> > Sent: 05 December 2002 14:02
>> > Subject: Re: [Sip] B2BUA & Max-Forwards
>> >
>> >
>> >> --On Thursday, December 05, 2002 12:09 PM +0000 Pete Cordell
>> >> <pete@tech-know-ware.com> wrote:
>> >>
>> >> > Dean,
>> >> >
>> >> > Firstly, your statement that "The B2BUA doesn't change
>> headers that
>> >> > transit it." seems inconsistent with the statement "It
>> does the same
>> >> > thing as ... ANY 500-pound gorilla does".  I don't understand.
>> >> >
>> >> > Secondly, I think max-forwards is a special case.  It is
>> important for
>> > the
>> >> > protection of SIP systems as it prevents loops.
>> >> >
>> >> Once a dialog is terminated at a UAS it's up to the
>> application to do
>> >> loop prevention. SIP is not involved any more. The
>> application could
>> >> compute a value of Max-forwards for initiating outgoing
>> calls based on
>> >> incoming
>> > calls
>> >> which preserves the "strictly decreasing" invariant, or
>> alternatively it
>> >> could just use the default MAx-Forwards in outgoing
>> requests and execute
>> >> a contant memory loop suppression algorithm such as
>> Knuth's even-odd
>> >> iterator. In either event the SIP spec(s) need say nothing
>> about this.
>> >>
>> >> > Indeed, there is a major difference between a UAS and a
>> B2BUA and that
>> > is
>> >> > that the former terminates a request, and the latter terminates a
>> > request
>> >> > AND ALSO forwards an associated request (or more).
>> >> I disagree. A UAS is a UAS. It's behavior is completely
>> specified by
>> >> RFC3261 (unless there's a hole in the spec of course...).
>> What it means
>> >> to originate (NOT FORWARD!!!!) an "associated request" is
>> completely
>> >> application dependent. If I program my child's phone to
>> initiate a SIP
>> > call
>> >> to my surrepititious recording device every time it
>> receives a call and
>> >> give that device first chance to over-ride policy of the
>> child's phone is
>> >> the phone a B2BUA? Yes, no, maybe so, but who cares? Tell
>> me what the
>> >> applciation is and I can write some specs for that
>> application and how it
>> >> has to originate, terminate and process SIP transactions to do that
>> >> application.
>> >>
>> >> I think the botom line is that there is not a single, well defined
>> >> appliation that one can point to and say "that's a B2BUA".
>> >>
>> >> > The impact of this
>> >> > difference should be covered in the spec in the
>> interests of protecting
>> >> > other SIP systems.
>> >> >
>> >> I disagree.
>> >>
>> >> > Pete.
>> >> >
>> >> > ----- Original Message -----
>> >> > From: "Dean Willis" <dean.willis@softarmor.com>
>> >> > To: "'Pete Cordell'" <pete@tech-know-ware.com>; <sip@ietf.org>
>> >> > Sent: 05 December 2002 01:23
>> >> > Subject: RE: [Sip] B2BUA & Max-Forwards
>> >> >
>> >> >
>> >> >> > One issue that the recent B2BUA discussion has raised in my
>> >> >> > mind is what a B2BUA (whatever its decided that should be)
>> >> >> > should do with Max-Forwards.
>> >> >>
>> >> >> It does the same thing as it does with any other header
>> or body part,
>> >> >> which is the same thing that ANY 500-pound gorilla does
>> to somebody's
>> >> >> body parts -- whatever it wants.
>> >> >>
>> >> >> What you're asking is "what is the transparency of this
>> node with
>> >> >> respect to max-forwards", which is "#4 -- Header
>> Transaprency" in my
>> >> >> top-10 list. The specification of a back-to-back user
>> agent doesn't
>> >> >> mandate any level of transparency on this or any other header.
>> >> >>
>> >> >> --
>> >> >> Dean
>> >> >>
>> >> >> _______________________________________________
>> >> >> Sip mailing list  https://www1.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://www1.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
>> >>
>> >> ------------------------
>> >> David R. Oran
>> >> Cisco Systems
>> >> 7 Ladyslipper Lane
>> >> Acton, MA 01720
>> >> Office: +1 978 264 2048
>> >> VoIP: +1 408 571 4576
>> >> Email: oran@cisco.com
>> >
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.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: "Carr, Steve" <Steve.Carr@icn.siemens.com>
To: "'Henry Sinnreich'" <Henry.Sinnreich@wcom.com>,
        "'Pete Cordell'"
	 <pete@tech-know-ware.com>
Cc: sip@ietf.org
Subject: RE: [Sip] B2BUA
Date: Thu, 5 Dec 2002 12:31:36 -0800 
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http://www.lightreading.com/document.asp?doc_id=22549&site=lightreading

Some company called WorldCom, who are they??  :-{

-----Original Message-----
From: Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
Sent: Thursday, December 05, 2002 12:57 PM
To: 'Pete Cordell'
Cc: sip@ietf.org
Subject: RE: [Sip] B2BUA


Can you document any _outages_ of the _Internet_? I don't recall any.

Please provide references.

Thanks, Henry

> -----Original Message-----
> From: Pete Cordell [mailto:pete@tech-know-ware.com] 
> Sent: Thursday, December 05, 2002 6:25 AM
> To: Henry Sinnreich
> Cc: sip@ietf.org
> Subject: Re: [Sip] B2BUA
> 
> 
> Henry,
> 
> The Internet has had a few famous outages as well.  Besides, 
> if all the telco traffic was put over IP, with today's level 
> of QoS on the Internet, it would have been brought to its knees also.
> 
> To say that one is all goodness and light and the other all 
> evil doesn't do us as a community any good as when customers 
> find out there are problems they tend not to believe anything 
> else we say!
> 
> Only vapourware is flawless.  Most people know that.  Respect 
> the intelligence of your customer, or they will find somebody 
> that does!!!
> 
> Pete.
> 
> ----- Original Message -----
> From: "Henry Sinnreich" <Henry.Sinnreich@wcom.com>
> To: "'Phil Reynolds'" <preynolds@ridgewaysystems.com>; <sip@ietf.org>
> Sent: 04 December 2002 19:02
> Subject: RE: [Sip] B2BUA
> 
> 
> > >The telecoms people are clever enough to have thought
> > >about that and build redundancy, fault tolerance and failover into
> > their switches.
> >
> > This has proven unfortunately not to be the case on 
> September 11 when 
> > telecom service failed for days, while Internet traffic continued 
> > smoothly. I suggest you to look at the central control model of the 
> > switch network versus the e2e architecture of the Internet. SIP 
> > enabled networks are also fully distributed and the e2e control 
> > principle maintained - that's why state in placed in SIP endpoints.
> >
> > I am not sure if a few words here can give full justice to the 
> > superior resilience of the Internet and to SIP enabled services,...
> >
> > Thanks, Henry
> >
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Phil 
> > Reynolds
> > Sent: Wednesday, December 04, 2002 7:37 AM
> > To: sip@ietf.org
> > Subject: RE: [Sip] B2BUA
> >
> >
> > If a telephone switch crashes you loose all calls...  The telecoms 
> > people are clever enough to have thought about that and build 
> > redundancy, fault tolerance and failover into their switches. Maybe 
> > some of us IP-telecoms people have thought of similar 
> techniques for 
> > SIP intermediaries? -----Original Message-----
> > From: Jiri Kuthan [mailto:jiri.kuthan@fokus.fraunhofer.de]
> > Sent: 04 December 2002 12:32
> > To: Christian Huitema; Roy, Radhika R, ALASO; 
> mbarnes@nortelnetworks.com
> >
> > Cc: dean.willis@softarmor.com; Juan-Carlos.Rojas@alcatel.fr; 
> > sip@ietf.org
> > Subject: RE: [Sip] B2BUA
> > [...snip...]
> > I think there are more issues with B2BUA. A B2BUA's failure affects 
> > all current calls :-( -Jiri
> > _______________________________________________
> > Sip mailing list  https://www1.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://www1.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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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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Subject: Re: [Sip] ABNF issue
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Hi Jonathan,

> Christer Holmberg wrote:
> > Hi,
> >
> > The ABNF syntax in RFC3261 says the following things:
> >
> > LWS              =  [*WSP CRLF] 1*WSP ; linear whitespace
> >
> > Request          = Request-Line
> >                           *( message-header )
> >                           CRLF
> >                           [ message-body ]
> >
> > message-body  =  *OCTET
> >
> > Also, the syntax says that every message-header ends with CTRL.
> >
> > Now, the spec says that the SIP part of a message ends with CTRL CTRL.
> >
> > However, if the message-body part starts with one or more WSP, won't the
> >
> > CRLF between the message-body be interpreted as LWS instead? Ie, there
> > will NOT be CRLF CRLF at the end of the SIP part of message, but instead
> >
> > CRLF LWS.
>
> I don't know how you come to this conclusion. If the body starts with
> LWS, you will see a CRLF after the last header field, another CRLF
> before the body, and then an LWS which is the first byte of the body. Thus:
>
> CRLF CRLF LWS
>
> which is fine.

The body does not start with LWS. The body starts with WSP (RFC3261 doesn't
define what WSP is, but if I remember correct it is SP or TAB). So, in that
case you will have:

CRLF CRLF WSP

...which, according to the ABNF is:

CRLF LWS

Regards,

Christer Holmberg
Ericsson Finland


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Date: Thu, 05 Dec 2002 19:50:24 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sip] B2BUA
In-reply-to: <1039119246.1798.11.camel@mhhbwmbc>
To: "'Wilhelm Wimmreuter'" <wilhelm@wimmreuter.de>
Cc: "'Liebhart Rainer'" <rainer.liebhart@siemens.com>,
        "'Pete Cordell'" <pete@tech-know-ware.com>, sip@ietf.org
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No "qualified busy/congestion tone" was experienced by any IM, e-mail or
SIP users.

Henry

> -----Original Message-----
> From: Wilhelm Wimmreuter [mailto:wilhelm@wimmreuter.de] 
> Sent: Thursday, December 05, 2002 2:14 PM
> To: Henry Sinnreich
> Cc: 'Liebhart Rainer'; 'Pete Cordell'; sip@ietf.org
> Subject: RE: [Sip] B2BUA
> 
> 
> Hi Henry,
> cant resist throwing my nonsense in as well
> 
> On Thu, 2002-12-05 at 19:16, Henry Sinnreich wrote:
> > Rainer, the fact is that people could communicate on 
> 9/11/01 using IM 
> > and SIP telephony from Columbia University, while the telecom phone 
> > service, both landline and mobile was down for days.
> > 
> > There is very little or very much to comment on these facts.
> > 
> 
> Did you forget, that a qualified busy/congestion tone, even 
> for some days, does not reflect a system outage ;-)
> 
>  ... If required, call me on my SIP phone to sort that out. 
> It will work as usual :-)
> 
> Willi
> 
> 
> 
> 
> 
> 

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Subject: RE: [Sip] B2BUA - Security
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From: "Christian Huitema" <huitema@windows.microsoft.com>
To: "Mahey, Sonit" <Sonit.Mahey@icn.siemens.com>,
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The sending UA places the result of the STUN translation in the SDP, not
the private addresses and port numbers.

> -----Original Message-----
> From: Mahey, Sonit [mailto:Sonit.Mahey@icn.siemens.com]
> Sent: Thursday, December 05, 2002 10:00 AM
> To: Christian Huitema; Jonathan Rosenberg; Pete Cordell
> Cc: sip@ietf.org
> Subject: RE: [Sip] B2BUA - Security
> 
> Thanks, Christian.
> 
> Assuming IPv4 addresses and without delving into the draft RFC on
STUN:
> The SDP contains the RTP/RTCP port numbers that are private and these
are
> also required to be NAT'd. Now, if the SDP, which forms part of the
SIP
> body, is encrypted, how will RTP/RTCP ports be translated?
> 
> regards,
> - sonit
> 
> > -----Original Message-----
> > From: Christian Huitema [mailto:huitema@windows.microsoft.com]
> > Sent: Thursday, December 05, 2002 12:54 PM
> > To: Mahey, Sonit; Jonathan Rosenberg; Pete Cordell
> > Cc: sip@ietf.org
> > Subject: RE: [Sip] B2BUA - Security
> >
> >
> > > From: Mahey, Sonit [mailto:Sonit.Mahey@icn.siemens.com]
> > > Sent: Thursday, December 05, 2002 9:14 AM
> > >
> > > I agree with Jonathan.
> > >
> > > That brings up the question:
> > > Is NAT traversal for encrypted SIP traffic addressed anywhere?
> >
> > There are two possibilities. If your application only uses UDP, it
is
> > possible to use IPv4 and STUN to find out the "outside ports" for
your
> > UDP traffic (check draft-ietf-midcom-stun-03.txt); you may need to
use
> > the "a:rtcp" convention to encode port numbers in SDP (check
> > draft-ietf-mmusic-sdp4nat-03.txt). If you need to also support TCP,
or
> > use IPSEC, or generally do away with the complications of
> > NAT, the best
> > solution is to just move to IPv6; see Teredo
> > (draft-ietf-ngtrans-shipworm-08.txt) for one possible way to
> > carry IPv6
> > across NAT, and 6to4 (RFC 3056 & 3068) for a possible way to
> > upgrade the
> > NAT and make it an IPv6 router.
> >
> > -- Christian Huitema
> >
> _______________________________________________
> Sip mailing list  https://www1.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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inline.

Christer Holmberg wrote:
> The body does not start with LWS. The body starts with WSP (RFC3261 doesn't
> define what WSP is, but if I remember correct it is SP or TAB). So, in that
> case you will have:
> 
> CRLF CRLF WSP
> 
> ...which, according to the ABNF is:
> 
> CRLF LWS

You are taking this out of context. It is not the case that you would 
parse through the message replaceing all instances of CRLF WSP with LWS.

The BNF specifies that message-header ends with a CRLF. What follows 
that is one of two things - either another CRLF, marking the end of the 
headers, or another message header, which never begins with LWS. Thus, 
if a CRLF is followed by something OTHER than WSP, the next construction 
cannot be LWS.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From mailnull@www1.ietf.org  Fri Dec  6 01:14:57 2002
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        Pete Cordell <pete@tech-know-ware.com>, sip@ietf.org
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When STUN is used, there is no need to translate. THe client itself 
inserts the proper addresses.

If you are not using stun, and rather using an ALG or an FCP ala midcom, 
and the SIP body is encrypted, well, right now you are out of luck. I 
will once again point out that this is the reason I wrote the session 
policy draft:

http://www.jdrosen.net/papers/draft-rosenberg-sipping-session-policy-00.txt

-Jonathan R.

Mahey, Sonit wrote:
> Thanks, Christian.
> 
> Assuming IPv4 addresses and without delving into the draft RFC on STUN:
> The SDP contains the RTP/RTCP port numbers that are private and these are
> also required to be NAT'd. Now, if the SDP, which forms part of the SIP
> body, is encrypted, how will RTP/RTCP ports be translated?
> 
> regards,
> - sonit
> 
> 
>>-----Original Message-----
>>From: Christian Huitema [mailto:huitema@windows.microsoft.com]
>>Sent: Thursday, December 05, 2002 12:54 PM
>>To: Mahey, Sonit; Jonathan Rosenberg; Pete Cordell
>>Cc: sip@ietf.org
>>Subject: RE: [Sip] B2BUA - Security
>>
>>
>>
>>>From: Mahey, Sonit [mailto:Sonit.Mahey@icn.siemens.com]
>>>Sent: Thursday, December 05, 2002 9:14 AM
>>>
>>>I agree with Jonathan.
>>>
>>>That brings up the question:
>>>Is NAT traversal for encrypted SIP traffic addressed anywhere?
>>
>>There are two possibilities. If your application only uses UDP, it is
>>possible to use IPv4 and STUN to find out the "outside ports" for your
>>UDP traffic (check draft-ietf-midcom-stun-03.txt); you may need to use
>>the "a:rtcp" convention to encode port numbers in SDP (check
>>draft-ietf-mmusic-sdp4nat-03.txt). If you need to also support TCP, or
>>use IPSEC, or generally do away with the complications of 
>>NAT, the best
>>solution is to just move to IPv6; see Teredo
>>(draft-ietf-ngtrans-shipworm-08.txt) for one possible way to 
>>carry IPv6
>>across NAT, and 6to4 (RFC 3056 & 3068) for a possible way to 
>>upgrade the
>>NAT and make it an IPv6 router.
>>
>>-- Christian Huitema
>>
> 
> 

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
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Pete Cordell wrote:
> Jonathan,
> 
> I have not said anything about generic B2BUAs here.  Each B2BUA
> implementation will need to have its own record (and server) of what it is
> transparent to or not.  It would not be a generic list.
> 
> Perhaps it is reasonable to remove the B2BUA definition, since there is no
> definition of a gateway and they are similar beasts.  Just as long as such
> an action does not imply that SIP is not suitable for a certain class of
> problem.

Removal of the definition has nothing at all to do with whether or not 
such a SIP application can be built. Clearly it can.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
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inline as always. But, first, a question. Pete - what are you after? Are 
you actually trying to convince the working group to disallow 
encryption/authentication over SDP so that your b2bua-thingy can modify 
it?? As I have tried to argue in my session policy draft, there are a 
LOT of issues with intermediary modification of media information, many 
of them related to the kind of things OPES has been dealing with. You 
need to consider the problem more broadly.


Pete Cordell wrote:

>>1. modifying SDP in transit to point media streams to an intermediate
>>recording device, allowing attackers to establish wiretaps
> 
> 
> This is only partially successful.  Encryption of RTP is the only complete
> solution.

True.

> 
> 
>>2. downgrading any SRTP keying done in SDP, so that the media itself is
>>not encrypted or integrity protected (this came up during the mmusic
>>session in IETF as a driver for S/MIME)
> 
> 
> I don't think you need to encrypt the entire SDP body to prevent key
> downgrading.  In the absence of any other contraints that might be a simple
> route to go, but if it causes problems (as it is) then we should look at
> other ways.

You need to protect the key exchanges. I am trying to establish that a 
lot of the information in SDP needs to be protected e2e.

> 
> As an aside - This is the sort of reason I would like to get this sort of
> thing on the agenda.  It's not that its not possible to enable SIP to meet
> these requirements, there's just not a desire to do among the core SIP
> people.  I know why they don't want to do it, but I also think that SIP
> fails to address a whole class of problem as a result.  In the long run I
> think that will be detrimental to SIP.

I have no idea what "it" is in this aside.

> 
> 
>>3. launching a DoS attack by sending everyones voice traffic to the
>>intended target. You get huge amplification gains here. Very appealing
>>for hackers!
> 
> 
> Don't need SDP to do that.  I could just rip out the entire SDP and put in a
> new version.  I might not get the ports right, but I could probably cause
> enough damage to get my kicks.

If you rip it out and replace it, and the UA is requiring signed SDP, 
then this won't work.

Another interesting attack I thought of, is to modify the SDP to send 
the RTP to a relay, which does forward it to the right destination, but 
only after inserting 100ms of delay. The result is that the conversation 
takes place, but at lousy quality. Why do this? Maybe a way for one 
provider to convince customers of another provider to switch. Dropping 
the packets outright might be too overt of an attack, but a subtle 
increase in delays... just a thought.

>>Confidentiality is only one property of security. I am much more
>>concerned about message integrity.
> 
> 
> Integrity of which bits?  And are you sure that maintaining integrity buys
> you anything except for in a few special cases?

Definitely.

I don't want someone removing my video stream my session.
I don't want someone to modify my codecs, forcing me to use a lower 
quality one when a better one would actually work.
I don't want someone to receive my media when I intended it for the 
party I called.

And so on. The above concerns are real, and cover a good set of the 
information in SDP.

Let me pose the question in reverse. What information in SDP do you feel 
is NOT worth protecting?

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
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Pete Cordell wrote:
> Dave,
> 
> To save your disagreeableness, can we say something like:
> 
> "A B2BUA MUST perform loop prevention.  One way to do this is to propagate
> the Max-Forwards header from the incoming request to the outgoing request in
> the same way that a proxy does."

A statement like this is completely out of scope for rfc3261. It might 
be a reasonable statement to make in some kind of application document 
that describes a particular type of b2bua. However, it makes no more 
sense to say this in RFC 3261 as it would be to discuss the ways in 
which to build a focus or a 3pcc controller (both of which are also 
informational, by the way).

I can't resist pointing out that much of this is deja vu. Folks may 
recall heated discussions in Adelaide, where there was a bof (I think) 
that wanted to have a working group to specify SIP to H.323 conversion. 
THe same arguemtns that were made there are being made here; its an 
application level issue, there are many ways to do it, don't stifle 
creativity, etc. In fact I recall that Dave was, as always, quite 
vociferous on these points.

The eventual conclusion in the h323/sip conversion case was to write an 
INFORMATIONAL RFC that documents reasonable practices in building such a 
device. I believe the same logic and conclusions apply here. If folks 
have a particular type of b2bua application they are fond of, please 
feel free to write up your implementation advice and I am sure sipping 
would consider whether or not it would make a useful informational RFC.

However, it is completely inappropriate to ever make such application 
logic normative, and even more inappropraite for such wording to be in 
rfc3261.

Anyway, I too am tired of this thread. I think, in fact, there is 
"rough" consensus to remove the definition of b2bua entirely from 
rfc3261 in its next revision (which is quite some time in the future 
anyway). However, I'll let the chairs decide that.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Hello,

"I don't get this.  Does two or more mean, for example, four?"
Two or more means exactly that: two or more. The exact number depends on
what the application is actually providing, and you cannot fixe it in
advance.

"but a pair of UAC/UAS concatenations"
OK, I get your point, you are thinking on a specific class of application
for bi-party communications. But this is only one case.

At the end, I start to agree with some other people:
- either we are able to provide a definition of a B2BUA independently of
what the B2BUA is actually doing
- or we drop the definition

Best regards
Juan Carlos





<aki.niemi@nokia.com> on 05/12/2002 19:54:26

To:   Juan-Carlos ROJAS/FR/ALCATEL@ALCATEL
cc:   <sip@ietf.org>
Subject:  RE: [Sip] B2BUA & Max-Forwards


Hi,

inline.

> "I am at the point where I agree with the sentiment raised by others,
> that we should remove the term b2bua entirely from the
> specification. It
> has caused more confusion than anything else."
>
> Yes, this is one possibility. An alternative (as I proposed
> in an earlier
> mail) is to complete the current definition in 3261 saying
> explicitly that
> :
> 1) A B2BUA handles two or more UAs, each one behaving as UAC/UAS, as
> described in 3261 for UAs

I don't get this.  Does two or more mean, for example, four?  And does that
in turn make it a B2BB2BUA? ;)

The only minor clarification I think might be in order for RFC3261 is that
it's not just a UAC/UAS concatenation, but a pair of UAC/UAS
concatenations.

However, the fact that the traffic in this thread alone equals to a year's
worth of emails on the SIMPLE list suggests that no clarification is enough
clarification.

So I agree with the other people in that getting rid of the definition
altogether is the best option.  After all, if it looks like a UA to the
outside world, it must be a UA.

> 2) The SIP protocol specification does not mandate for any
> behaviour of
> such B2BUA, i.e. on how to "link" (or "concatenate") two or more UAs.

Yes, probably the reason why the definition for a B2BUA is also useless.

Cheers,
Aki




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Henry,
thank you for that info, Mary provided this already. But the question is: how many users at Columbia were using SIP as communication medium and was the communication restricted to the Columbia campus?

Rainer

-------------------------------------------------------------------------------
Rainer Liebhart
Siemens AG
ICM N PG SP ST N 1
Mch H; R. 1759/1048
Tel.: +49 89 722 50302
Fax:  +49 89 722 39793
mailto:rainer.liebhart@siemens.com
-------------------------------------------------------------------------------


-----Ursprüngliche Nachricht-----
Von: Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
Gesendet: Donnerstag, 5. Dezember 2002 19:17
An: 'Liebhart Rainer'; 'Pete Cordell'
Cc: sip@ietf.org
Betreff: RE: [Sip] B2BUA


Rainer, the fact is that people could communicate on 9/11/01 using IM
and SIP telephony from Columbia University, while the telecom phone
service, both landline and mobile was down for days.

There is very little or very much to comment on these facts.

Henry

> -----Original Message-----
> From: Liebhart Rainer [mailto:rainer.liebhart@siemens.com] 
> Sent: Thursday, December 05, 2002 7:46 AM
> To: 'Pete Cordell'; Henry Sinnreich
> Cc: sip@ietf.org
> Subject: AW: [Sip] B2BUA
> 
> 
> Henry,
> 
> unfortunately you are comparing apples and pears when stating 
> that the telecom service failed for days after Sept 11, while 
> Internet traffic continued smoothly. The mentioned "telefon 
> service" consists of thousands of (real-time) calls to 
> emergency centres while the Internet traffic consists mainly 
> of requests to the Web pages of CNN and others (with bad 
> response times) to retrieve *information* on the status of 
> the catastrophe.
> 
> Rainer
> 
> 
> -----Ursprüngliche Nachricht-----
> Von: Pete Cordell [mailto:pete@tech-know-ware.com]
> Gesendet: Donnerstag, 5. Dezember 2002 13:25
> An: Henry Sinnreich
> Cc: sip@ietf.org
> Betreff: Re: [Sip] B2BUA
> 
> 
> Henry,
> 
> The Internet has had a few famous outages as well.  Besides, 
> if all the telco traffic was put over IP, with today's level 
> of QoS on the Internet, it would have been brought to its knees also.
> 
> To say that one is all goodness and light and the other all 
> evil doesn't do us as a community any good as when customers 
> find out there are problems they tend not to believe anything 
> else we say!
> 
> Only vapourware is flawless.  Most people know that.  Respect 
> the intelligence of your customer, or they will find somebody 
> that does!!!
> 
> Pete.
> 
> ----- Original Message -----
> From: "Henry Sinnreich" <Henry.Sinnreich@wcom.com>
> To: "'Phil Reynolds'" <preynolds@ridgewaysystems.com>; <sip@ietf.org>
> Sent: 04 December 2002 19:02
> Subject: RE: [Sip] B2BUA
> 
> 
> > >The telecoms people are clever enough to have thought
> > >about that and build redundancy, fault tolerance and failover into
> > their switches.
> >
> > This has proven unfortunately not to be the case on 
> September 11 when 
> > telecom service failed for days, while Internet traffic continued 
> > smoothly. I suggest you to look at the central control model of the 
> > switch network versus the e2e architecture of the Internet. SIP 
> > enabled networks are also fully distributed and the e2e control 
> > principle maintained - that's why state in placed in SIP endpoints.
> >
> > I am not sure if a few words here can give full justice to the 
> > superior resilience of the Internet and to SIP enabled services,...
> >
> > Thanks, Henry
> >
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Phil 
> > Reynolds
> > Sent: Wednesday, December 04, 2002 7:37 AM
> > To: sip@ietf.org
> > Subject: RE: [Sip] B2BUA
> >
> >
> > If a telephone switch crashes you loose all calls...  The telecoms 
> > people are clever enough to have thought about that and build 
> > redundancy, fault tolerance and failover into their switches. Maybe 
> > some of us IP-telecoms people have thought of similar 
> techniques for 
> > SIP intermediaries? -----Original Message-----
> > From: Jiri Kuthan [mailto:jiri.kuthan@fokus.fraunhofer.de]
> > Sent: 04 December 2002 12:32
> > To: Christian Huitema; Roy, Radhika R, ALASO; 
> mbarnes@nortelnetworks.com
> >
> > Cc: dean.willis@softarmor.com; Juan-Carlos.Rojas@alcatel.fr; 
> > sip@ietf.org
> > Subject: RE: [Sip] B2BUA
> > [...snip...]
> > I think there are more issues with B2BUA. A B2BUA's failure affects 
> > all current calls :-( -Jiri
> > _______________________________________________
> > Sip mailing list  https://www1.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://www1.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://www1.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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Jonathan,

As principle author of the document you obviously have a major say on
what's in scope or not.

However, I would argue that RFC 3261 says much about what a proxy must do to
be a proxy.  However, it does not do it in complete detail.  For example, it
does not say that it must do a SQL query to find a user records, or ask the
Ferengi.  It says enough about the proxy to make it functional from a SIP
point of view.

I'm only arguing that the spec should have the same scope for B2BUAs.  There
are actually less SIP issues for a B2BUA than there are for a proxy.
However, I think prevention of loops is one thing it should say, as this has
a direct impact on the integrity of SIP systems, and this must surely be a
primary concern of the RFC.

If there is no need to say something about it for B2BUAs, then I see no need
to say anything about it for proxies!

---

As for removing the definition of B2BUA from the RFC, that may be a sensible
thing to do.  The upshot of doing that is that people might start calling
what is now a B2BUA, a proxy.  This wouldn't be unreasonable as the
functionality of a B2BUA is analogous to an FTP proxy.

How about the following as a definition of a B2BUA?  Or is it too loose?

"Back-to-Back User Agent: A back-to-back user agent (B2BUA) is a logical
entity that has one or more egress SIP dialogs associated with an ingress
SIP dialog, but does not conform to the definition of a proxy.  The manner
in which the ingress and egress dialogs are associated is purely an
implementation issue and outside the scope of this document.  When it
receives a request it appears from an external point of view to process the
request as if it were a user agent server (UAS).  On a transaction by
transaction basis, to determine how the request should be handled it may
generate and send SIP requests such that from an external point of view it
appears as one or more user agent clients (UAC), or it may respond to the
request locally."


Pete.

----- Original Message -----
From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
To: "Pete Cordell" <pete@tech-know-ware.com>
Cc: "David R. Oran" <oran@cisco.com>; "Dean Willis"
<dean.willis@softarmor.com>; <sip@ietf.org>
Sent: 06 December 2002 06:58
Subject: Re: [Sip] B2BUA & Max-Forwards


>
>
> Pete Cordell wrote:
> > Dave,
> >
> > To save your disagreeableness, can we say something like:
> >
> > "A B2BUA MUST perform loop prevention.  One way to do this is to
propagate
> > the Max-Forwards header from the incoming request to the outgoing
request in
> > the same way that a proxy does."
>
> A statement like this is completely out of scope for rfc3261. It might
> be a reasonable statement to make in some kind of application document
> that describes a particular type of b2bua. However, it makes no more
> sense to say this in RFC 3261 as it would be to discuss the ways in
> which to build a focus or a 3pcc controller (both of which are also
> informational, by the way).
>
> I can't resist pointing out that much of this is deja vu. Folks may
> recall heated discussions in Adelaide, where there was a bof (I think)
> that wanted to have a working group to specify SIP to H.323 conversion.
> THe same arguemtns that were made there are being made here; its an
> application level issue, there are many ways to do it, don't stifle
> creativity, etc. In fact I recall that Dave was, as always, quite
> vociferous on these points.
>
> The eventual conclusion in the h323/sip conversion case was to write an
> INFORMATIONAL RFC that documents reasonable practices in building such a
> device. I believe the same logic and conclusions apply here. If folks
> have a particular type of b2bua application they are fond of, please
> feel free to write up your implementation advice and I am sure sipping
> would consider whether or not it would make a useful informational RFC.
>
> However, it is completely inappropriate to ever make such application
> logic normative, and even more inappropraite for such wording to be in
> rfc3261.
>
> Anyway, I too am tired of this thread. I think, in fact, there is
> "rough" consensus to remove the definition of b2bua entirely from
> rfc3261 in its next revision (which is quite some time in the future
> anyway). However, I'll let the chairs decide that.
>
> -Jonathan R.
>
> --
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>


_______________________________________________
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From: "Pete Cordell" <pete@tech-know-ware.com>
To: "David Oran" <oran@cisco.com>, <sip@ietf.org>
References: <00cd01c29bfc$e7dd0240$fe0c0c42@txdwillis> <013a01c29c5d$e1591b00$0200000a@tkw> <2986103.1039078949@ORANLT.cisco.com> <009901c29c83$9a304a20$0200000a@tkw> <21471264.1039097434@ORANLT.cisco.com>
Subject: Re: [Sip] B2BUA & Max-Forwards
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Please see inline...

----- Original Message -----
From: "David Oran" <oran@cisco.com>


> --On Thursday, December 05, 2002 5:24 PM +0000 Pete Cordell
> <pete@tech-know-ware.com> wrote:
>
> > Dave,
> >
> > To save your disagreeableness, can we say something like:
> >
> > "A B2BUA MUST perform loop prevention.  One way to do this is to
propagate
> > the Max-Forwards header from the incoming request to the outgoing
request
> > in the same way that a proxy does."
> >
> I don't understand what it means to have normative language for a beast
> which is fundamentally undefined. Suppose I write a piece of code which
> accepts SIP requests, issues tachyon commands to alpha centauri, and
> meanwhile receives routing instructions from a chronosynclastic
> infundibulum (for the unintiated, that is a device which effects reverse
> causality based on the message to alpha centauri whose response would not
> normally arrive for 7.8 years). It then uses those routing instructions to
> formulate an outbound SIP request to the destination received through the
> chronosynclastic infundibulum.
>
> Question: is this application a B2BUA and hence required to have a
> Max-Forwards value in the request whose value is strictly less than the
> value received in the SIP request? Beats me.

If you read what I said, you would see that the cyborg (or whoever) didn't
actually have to know about Max-Fowards.  They could use the colonic drive
if they wished to.

>
> > It's the loops that I'm worried about, not the Max-Forwards per se.
> >
> There are many ways to suppress loops. Using application semantics is
> actually safer than lower-layer protocol machinery because of the hazards
> introduced by protocol translation (e.g. the PSTN hop case mentioned
> earlier, or a SIP<->H.323 conversion somewhere else).

Which is what I was agreeing to.

> > This addresses my concern, and I think James' and possibly yours.
> >
> Sorry, not mine. From where I sit, the less said about B2BUAs the better,
> including asymptotically toward zero.

But not quite zero then.  Mentioning taking steps to prevent loops doesn't
seem much.

>
> Cheers, Dave.
>
> > Pete.
> >
> > ----- Original Message -----
> > From: "David R. Oran" <oran@cisco.com>
> > To: "Pete Cordell" <pete@tech-know-ware.com>; "Dean Willis"
> > <dean.willis@softarmor.com>; <sip@ietf.org>
> > Sent: 05 December 2002 14:02
> > Subject: Re: [Sip] B2BUA & Max-Forwards
> >
> >
> >> --On Thursday, December 05, 2002 12:09 PM +0000 Pete Cordell
> >> <pete@tech-know-ware.com> wrote:
> >>
> >> > Dean,
> >> >
> >> > Firstly, your statement that "The B2BUA doesn't change headers that
> >> > transit it." seems inconsistent with the statement "It does the same
> >> > thing as ... ANY 500-pound gorilla does".  I don't understand.
> >> >
> >> > Secondly, I think max-forwards is a special case.  It is important
for
> > the
> >> > protection of SIP systems as it prevents loops.
> >> >
> >> Once a dialog is terminated at a UAS it's up to the application to do
> >> loop prevention. SIP is not involved any more. The application could
> >> compute a value of Max-forwards for initiating outgoing calls based on
> >> incoming
> > calls
> >> which preserves the "strictly decreasing" invariant, or alternatively
it
> >> could just use the default MAx-Forwards in outgoing requests and
execute
> >> a contant memory loop suppression algorithm such as Knuth's even-odd
> >> iterator. In either event the SIP spec(s) need say nothing about this.
> >>
> >> > Indeed, there is a major difference between a UAS and a B2BUA and
that
> > is
> >> > that the former terminates a request, and the latter terminates a
> > request
> >> > AND ALSO forwards an associated request (or more).
> >> I disagree. A UAS is a UAS. It's behavior is completely specified by
> >> RFC3261 (unless there's a hole in the spec of course...). What it means
> >> to originate (NOT FORWARD!!!!) an "associated request" is completely
> >> application dependent. If I program my child's phone to initiate a SIP
> > call
> >> to my surrepititious recording device every time it receives a call and
> >> give that device first chance to over-ride policy of the child's phone
is
> >> the phone a B2BUA? Yes, no, maybe so, but who cares? Tell me what the
> >> applciation is and I can write some specs for that application and how
it
> >> has to originate, terminate and process SIP transactions to do that
> >> application.
> >>
> >> I think the botom line is that there is not a single, well defined
> >> appliation that one can point to and say "that's a B2BUA".
> >>
> >> > The impact of this
> >> > difference should be covered in the spec in the interests of
protecting
> >> > other SIP systems.
> >> >
> >> I disagree.
> >>
> >> > Pete.
> >> >


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To: "Henry Sinnreich" <Henry.Sinnreich@wcom.com>
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I was thinking of this one.  Admittedly not as long, but a much wider area.

http://flix.flirble.org/demon.html

Pete.

----- Original Message -----
From: "Henry Sinnreich" <Henry.Sinnreich@wcom.com>
To: "'Pete Cordell'" <pete@tech-know-ware.com>
Cc: <sip@ietf.org>
Sent: 05 December 2002 17:56
Subject: RE: [Sip] B2BUA


> Can you document any _outages_ of the _Internet_? I don't recall any.
>
> Please provide references.
>
> Thanks, Henry
>
> > -----Original Message-----
> > From: Pete Cordell [mailto:pete@tech-know-ware.com]
> > Sent: Thursday, December 05, 2002 6:25 AM
> > To: Henry Sinnreich
> > Cc: sip@ietf.org
> > Subject: Re: [Sip] B2BUA
> >
> >
> > Henry,
> >
> > The Internet has had a few famous outages as well.  Besides,
> > if all the telco traffic was put over IP, with today's level
> > of QoS on the Internet, it would have been brought to its knees also.
> >
> > To say that one is all goodness and light and the other all
> > evil doesn't do us as a community any good as when customers
> > find out there are problems they tend not to believe anything
> > else we say!
> >
> > Only vapourware is flawless.  Most people know that.  Respect
> > the intelligence of your customer, or they will find somebody
> > that does!!!
> >
> > Pete.
> >
> > ----- Original Message -----
> > From: "Henry Sinnreich" <Henry.Sinnreich@wcom.com>
> > To: "'Phil Reynolds'" <preynolds@ridgewaysystems.com>; <sip@ietf.org>
> > Sent: 04 December 2002 19:02
> > Subject: RE: [Sip] B2BUA
> >
> >
> > > >The telecoms people are clever enough to have thought
> > > >about that and build redundancy, fault tolerance and failover into
> > > their switches.
> > >
> > > This has proven unfortunately not to be the case on
> > September 11 when
> > > telecom service failed for days, while Internet traffic continued
> > > smoothly. I suggest you to look at the central control model of the
> > > switch network versus the e2e architecture of the Internet. SIP
> > > enabled networks are also fully distributed and the e2e control
> > > principle maintained - that's why state in placed in SIP endpoints.
> > >
> > > I am not sure if a few words here can give full justice to the
> > > superior resilience of the Internet and to SIP enabled services,...
> > >
> > > Thanks, Henry
> > >
> > > -----Original Message-----
> > > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On
> > Behalf Of Phil
> > > Reynolds
> > > Sent: Wednesday, December 04, 2002 7:37 AM
> > > To: sip@ietf.org
> > > Subject: RE: [Sip] B2BUA
> > >
> > >
> > > If a telephone switch crashes you loose all calls...  The telecoms
> > > people are clever enough to have thought about that and build
> > > redundancy, fault tolerance and failover into their switches. Maybe
> > > some of us IP-telecoms people have thought of similar
> > techniques for
> > > SIP intermediaries? -----Original Message-----
> > > From: Jiri Kuthan [mailto:jiri.kuthan@fokus.fraunhofer.de]
> > > Sent: 04 December 2002 12:32
> > > To: Christian Huitema; Roy, Radhika R, ALASO;
> > mbarnes@nortelnetworks.com
> > >
> > > Cc: dean.willis@softarmor.com; Juan-Carlos.Rojas@alcatel.fr;
> > > sip@ietf.org
> > > Subject: RE: [Sip] B2BUA
> > > [...snip...]
> > > I think there are more issues with B2BUA. A B2BUA's failure affects
> > > all current calls :-( -Jiri
> > > _______________________________________________
> > > Sip mailing list  https://www1.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://www1.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://www1.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: "Pete Cordell" <pete@tech-know-ware.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: <sip@ietf.org>
References: <DAC3FCB50E31C54987CD10797DA511BA1D2A53@WIN-MSG-10.wingroup.windeploy.ntdev.microsoft.com> <013d01c29c5d$e4a7f7e0$0200000a@tkw> <3DEF7EAB.2070109@dynamicsoft.com> <009801c29c83$985129e0$0200000a@tkw> <3DF04799.7010208@dynamicsoft.com>
Subject: Re: [Sip] B2BUA - Security - (Midcom Reqts?)
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Jonathan,

Perhaps as you say, we should take a step back.

I want to address midcom scenarios shown in the time line shown in section
7.1. of RFC 3303 (Midcom framework) and 2.3.1 and 2.3.2 in
draft-ietf-midcom-scenarios-02.txt.

The current security situation in SIP makes this problematic, and I don't
think there is any chartered item in SIP or SIPPING to address this.

I would like there to be a secure mode of operation even for midcom systems
that allows encryption and authentication.  I would imagine that the IESG
would block midcom if it did not allow such security.

I think we know enough from a SIP point of view what a midcom solution will
need to do to a SIP system.  Therefore, I believe we should start work on
this sooner rather than later.

Perhaps the way to go is for the midcom group to produce a requirements
draft detailing what it needs from the SIP world (in its broadest sense)
that can be put into SIPPING and hopefully lead to a chartered item.

To me the successful conclusion of midcom seems dependent on this sort of
work being done.

Pete.

----- Original Message -----
From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
To: "Pete Cordell" <pete@tech-know-ware.com>
Cc: "Christian Huitema" <huitema@windows.microsoft.com>; <sip@ietf.org>
Sent: 06 December 2002 06:45
Subject: Re: [Sip] B2BUA - Security


> inline as always. But, first, a question. Pete - what are you after? Are
> you actually trying to convince the working group to disallow
> encryption/authentication over SDP so that your b2bua-thingy can modify
> it?? As I have tried to argue in my session policy draft, there are a
> LOT of issues with intermediary modification of media information, many
> of them related to the kind of things OPES has been dealing with. You
> need to consider the problem more broadly.
>
>
> Pete Cordell wrote:
>
> >>1. modifying SDP in transit to point media streams to an intermediate
> >>recording device, allowing attackers to establish wiretaps
> >
> >
> > This is only partially successful.  Encryption of RTP is the only
complete
> > solution.
>
> True.
>
> >
> >
> >>2. downgrading any SRTP keying done in SDP, so that the media itself is
> >>not encrypted or integrity protected (this came up during the mmusic
> >>session in IETF as a driver for S/MIME)
> >
> >
> > I don't think you need to encrypt the entire SDP body to prevent key
> > downgrading.  In the absence of any other contraints that might be a
simple
> > route to go, but if it causes problems (as it is) then we should look at
> > other ways.
>
> You need to protect the key exchanges. I am trying to establish that a
> lot of the information in SDP needs to be protected e2e.
>
> >
> > As an aside - This is the sort of reason I would like to get this sort
of
> > thing on the agenda.  It's not that its not possible to enable SIP to
meet
> > these requirements, there's just not a desire to do among the core SIP
> > people.  I know why they don't want to do it, but I also think that SIP
> > fails to address a whole class of problem as a result.  In the long run
I
> > think that will be detrimental to SIP.
>
> I have no idea what "it" is in this aside.
>
> >
> >
> >>3. launching a DoS attack by sending everyones voice traffic to the
> >>intended target. You get huge amplification gains here. Very appealing
> >>for hackers!
> >
> >
> > Don't need SDP to do that.  I could just rip out the entire SDP and put
in a
> > new version.  I might not get the ports right, but I could probably
cause
> > enough damage to get my kicks.
>
> If you rip it out and replace it, and the UA is requiring signed SDP,
> then this won't work.
>
> Another interesting attack I thought of, is to modify the SDP to send
> the RTP to a relay, which does forward it to the right destination, but
> only after inserting 100ms of delay. The result is that the conversation
> takes place, but at lousy quality. Why do this? Maybe a way for one
> provider to convince customers of another provider to switch. Dropping
> the packets outright might be too overt of an attack, but a subtle
> increase in delays... just a thought.
>
> >>Confidentiality is only one property of security. I am much more
> >>concerned about message integrity.
> >
> >
> > Integrity of which bits?  And are you sure that maintaining integrity
buys
> > you anything except for in a few special cases?
>
> Definitely.
>
> I don't want someone removing my video stream my session.
> I don't want someone to modify my codecs, forcing me to use a lower
> quality one when a better one would actually work.
> I don't want someone to receive my media when I intended it for the
> party I called.
>
> And so on. The above concerns are real, and cover a good set of the
> information in SDP.
>
> Let me pose the question in reverse. What information in SDP do you feel
> is NOT worth protecting?
>
> -Jonathan R.
> --
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>



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To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
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	 <pete@tech-know-ware.com>
Cc: "David R. Oran" <oran@cisco.com>,
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Subject: RE: [Sip] B2BUA & Max-Forwards
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I agree with Jonathan.  As mentioned earlier this week apparently, there is
already work underway by some individuals to write an informational draft
documenting BBUA scenarios/call flows. 

I'll keep my fingers crossed that the existence of such a document can save
us from another round of B2BUA discussions again in 3 months. 

Regards,
Mary.

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Friday, December 06, 2002 12:59 AM
To: Pete Cordell
Cc: David R. Oran; Dean Willis; sip@ietf.org
Subject: Re: [Sip] B2BUA & Max-Forwards




Pete Cordell wrote:
> Dave,
> 
> To save your disagreeableness, can we say something like:
> 
> "A B2BUA MUST perform loop prevention.  One way to do this is to propagate
> the Max-Forwards header from the incoming request to the outgoing request
in
> the same way that a proxy does."

A statement like this is completely out of scope for rfc3261. It might 
be a reasonable statement to make in some kind of application document 
that describes a particular type of b2bua. However, it makes no more 
sense to say this in RFC 3261 as it would be to discuss the ways in 
which to build a focus or a 3pcc controller (both of which are also 
informational, by the way).

I can't resist pointing out that much of this is deja vu. Folks may 
recall heated discussions in Adelaide, where there was a bof (I think) 
that wanted to have a working group to specify SIP to H.323 conversion. 
THe same arguemtns that were made there are being made here; its an 
application level issue, there are many ways to do it, don't stifle 
creativity, etc. In fact I recall that Dave was, as always, quite 
vociferous on these points.

The eventual conclusion in the h323/sip conversion case was to write an 
INFORMATIONAL RFC that documents reasonable practices in building such a 
device. I believe the same logic and conclusions apply here. If folks 
have a particular type of b2bua application they are fond of, please 
feel free to write up your implementation advice and I am sure sipping 
would consider whether or not it would make a useful informational RFC.

However, it is completely inappropriate to ever make such application 
logic normative, and even more inappropraite for such wording to be in 
rfc3261.

Anyway, I too am tired of this thread. I think, in fact, there is 
"rough" consensus to remove the definition of b2bua entirely from 
rfc3261 in its next revision (which is quite some time in the future 
anyway). However, I'll let the chairs decide that.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
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Use sip-implementors@cs.columbia.edu for questions on current sip
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----- Original Message -----
From: "David Oran" <oran@cisco.com>

... much deleted ...

> >> Question: is this application a B2BUA and hence required to have a
> >> Max-Forwards value in the request whose value is strictly
> >> less than the
> >> value received in the SIP request? Beats me.
> >
> > Only if that's the way it wants to do it. You can defeat
> > the loops howsoever you want.
> >
> Ok, then how about the following wording: "a B2BUA MUST prevent loops and
> it MUST NOT behave in such as way as to cause broadcast storms, or
> contribute to any exponential message explosion of SIP or any other
> messages.

Preventing loops will help with this.  But more helpfully, the text used to
describe proxies in this respect might be useful guidance on how far to go.

> In addition, it MUST comply will all local and national fire
> codes, including the Underwriter's Laboratory flaming oil test*.

Doesn't seem a SIP issue, but if you think its helpful from a SIP point of
view...

Pete

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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3428
                                
        Title:      Session Initiation Protocol (SIP) Extension for
                    Instant Messaging
        Author(s):  B. Campbell, Ed., J. Rosenberg, H. Schulzrinne,
                    C. Huitema, D. Gurle
        Status:     Standards Track
        Date:       December 2002
        Mailbox:    bcampbell@dynamicsoft.com,
                    jdrosen@dynamicsoft.com,
                    schulzrinne@cs.columbia.edu,
                    huitema@microsoft.com, dgurle@microsoft.com
        Pages:      18
        Characters: 41475
        Updates/Obsoletes:SeeAlso:    None
                                
        I-D Tag:    draft-ietf-sip-message-07.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3428.txt


Instant Messaging (IM) refers to the transfer of messages between
users in near real-time.  These messages are usually, but not
required to be, short.  IMs are often used in a conversational mode,
that is, the transfer of messages back and forth is fast enough for
participants to maintain an interactive conversation.

This document proposes the MESSAGE method, an extension to the
Session Initiation Protocol (SIP) that allows the transfer of Instant
Messages.  Since the MESSAGE request is an extension to SIP, it
inherits all the request routing and security features of that
protocol.  MESSAGE requests carry the content in the form of MIME
body parts.  MESSAGE requests do not themselves initiate a SIP
dialog; under normal usage each Instant Message stands alone, much
like pager messages.  MESSAGE requests may be sent in the context of
a dialog initiated by some other SIP request.

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

This is now a Proposed Standard Protocol.

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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Submissions for Requests for Comments should be sent to
RFC-EDITOR@RFC-EDITOR.ORG.  Please consult RFC 2223, Instructions to RFC
Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

...

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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3326
                                
        Title:      The Reason Header Field for the Session Initiation
                    Protocol (SIP)
        Author(s):  H. Schulzrinne, D. Oran, G. Camarillo
        Status:     Standards Track
        Date:       December 2002
        Mailbox:    schulzrinne@cs.columbia.edu, oran@cisco.com,
                    Gonzalo.Camarillo@ericsson.com
        Pages:      8
        Characters: 15695
        Updates/Obsoletes/SeeAlso:   None
                                
        I-D Tag:    draft-ietf-sip-reason-01.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3326.txt


For creating services, it is often useful to know why a Session
Initiation Protocol (SIP) request was issued.  This document defines
a header field, Reason, that provides this information.  The Reason
header field is also intended to be used to encapsulate a final
status code in a provisional response.  This functionality is needed
to resolve the "Heterogeneous Error Response Forking Problem", or
HERFP.  

This document is a product of the Session Initiation Protocol and the
Session Initiation Proposal Investigation Working Groups of the IETF.

This is now a Proposed Standard Protocol.

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.

This announcement is sent to the IETF list and the RFC-DIST list.
Requests to be added to or deleted from the IETF distribution list
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be sent to RFC-DIST-REQUEST@RFC-EDITOR.ORG.

Details on obtaining RFCs via FTP or EMAIL may be obtained by sending
an EMAIL message to rfc-info@RFC-EDITOR.ORG with the message body 
help: ways_to_get_rfcs.  For example:

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        help: ways_to_get_rfcs

Requests for special distribution should be addressed to either the
author of the RFC in question, or to RFC-Manager@RFC-EDITOR.ORG.  Unless
specifically noted otherwise on the RFC itself, all RFCs are for
unlimited distribution.echo 
Submissions for Requests for Comments should be sent to
RFC-EDITOR@RFC-EDITOR.ORG.  Please consult RFC 2223, Instructions to RFC
Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Pete Cordell'" <pete@tech-know-ware.com>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>
Cc: sip@ietf.org
Subject: RE: [Sip] B2BUA - Security - (Midcom Reqts?)
Date: Fri, 6 Dec 2002 18:27:59 -0500 
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My favorite property of S/MIME protection for SDP is middlebox detection.
I'd like to know if some putatively benevolent middlebox is attempting to
'help' my session media get where it needs to go.

Although the midcom working group may favor middlebox transparency, I do not
believe that should be a goal of this working group. SIP is a peer-to-peer
protocol. While applications of SIP are free not to apply security
properties to SDP if they wish, I think it is unarguably valid for a SIP
user agent to secure SDP, even if that runs the risk of troubling
middleboxes.

In any event, there already is a secure way to allow encryption and
authentication via middleboxes - you can apply security properties to the
SDP using a certificate held by the middlebox (or its controller as the case
may be). If you actually intend for the middlebox to be in the path of your
media, this seems like a reasonable solution. I don't think we need anything
more.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Pete Cordell [mailto:pete@tech-know-ware.com]
> Sent: Friday, December 06, 2002 7:35 AM
> To: Jonathan Rosenberg
> Cc: sip@ietf.org
> Subject: Re: [Sip] B2BUA - Security - (Midcom Reqts?)
> 
> 
> Jonathan,
> 
> Perhaps as you say, we should take a step back.
> 
> I want to address midcom scenarios shown in the time line shown in section
> 7.1. of RFC 3303 (Midcom framework) and 2.3.1 and 2.3.2 in
> draft-ietf-midcom-scenarios-02.txt.
> 
> The current security situation in SIP makes this problematic, and I don't
> think there is any chartered item in SIP or SIPPING to address this.
> 
> I would like there to be a secure mode of operation even for midcom
systems
> that allows encryption and authentication.  I would imagine that the IESG
> would block midcom if it did not allow such security.
> 
> I think we know enough from a SIP point of view what a midcom solution
will
> need to do to a SIP system.  Therefore, I believe we should start work on
> this sooner rather than later.
> 
> Perhaps the way to go is for the midcom group to produce a requirements
> draft detailing what it needs from the SIP world (in its broadest sense)
> that can be put into SIPPING and hopefully lead to a chartered item.
> 
> To me the successful conclusion of midcom seems dependent on this sort of
> work being done.
> 
> Pete.
> 
> ----- Original Message -----
> From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
> To: "Pete Cordell" <pete@tech-know-ware.com>
> Cc: "Christian Huitema" <huitema@windows.microsoft.com>; 
> <sip@ietf.org>
> Sent: 06 December 2002 06:45
> Subject: Re: [Sip] B2BUA - Security
> 
> 
> > inline as always. But, first, a question. Pete - what are 
> you after? Are
> > you actually trying to convince the working group to disallow
> > encryption/authentication over SDP so that your 
> b2bua-thingy can modify
> > it?? As I have tried to argue in my session policy draft, 
> there are a
> > LOT of issues with intermediary modification of media 
> information, many
> > of them related to the kind of things OPES has been dealing 
> with. You
> > need to consider the problem more broadly.
> >
> >
> > Pete Cordell wrote:
> >
> > >>1. modifying SDP in transit to point media streams to an 
> intermediate
> > >>recording device, allowing attackers to establish wiretaps
> > >
> > >
> > > This is only partially successful.  Encryption of RTP is the only
> complete
> > > solution.
> >
> > True.
> >
> > >
> > >
> > >>2. downgrading any SRTP keying done in SDP, so that the 
> media itself is
> > >>not encrypted or integrity protected (this came up during 
> the mmusic
> > >>session in IETF as a driver for S/MIME)
> > >
> > >
> > > I don't think you need to encrypt the entire SDP body to 
> prevent key
> > > downgrading.  In the absence of any other contraints that 
> might be a
> simple
> > > route to go, but if it causes problems (as it is) then we 
> should look at
> > > other ways.
> >
> > You need to protect the key exchanges. I am trying to 
> establish that a
> > lot of the information in SDP needs to be protected e2e.
> >
> > >
> > > As an aside - This is the sort of reason I would like to 
> get this sort
> of
> > > thing on the agenda.  It's not that its not possible to 
> enable SIP to
> meet
> > > these requirements, there's just not a desire to do among 
> the core SIP
> > > people.  I know why they don't want to do it, but I also 
> think that SIP
> > > fails to address a whole class of problem as a result.  
> In the long run
> I
> > > think that will be detrimental to SIP.
> >
> > I have no idea what "it" is in this aside.
> >
> > >
> > >
> > >>3. launching a DoS attack by sending everyones voice 
> traffic to the
> > >>intended target. You get huge amplification gains here. 
> Very appealing
> > >>for hackers!
> > >
> > >
> > > Don't need SDP to do that.  I could just rip out the 
> entire SDP and put
> in a
> > > new version.  I might not get the ports right, but I 
> could probably
> cause
> > > enough damage to get my kicks.
> >
> > If you rip it out and replace it, and the UA is requiring 
> signed SDP,
> > then this won't work.
> >
> > Another interesting attack I thought of, is to modify the 
> SDP to send
> > the RTP to a relay, which does forward it to the right 
> destination, but
> > only after inserting 100ms of delay. The result is that the 
> conversation
> > takes place, but at lousy quality. Why do this? Maybe a way for one
> > provider to convince customers of another provider to 
> switch. Dropping
> > the packets outright might be too overt of an attack, but a subtle
> > increase in delays... just a thought.
> >
> > >>Confidentiality is only one property of security. I am much more
> > >>concerned about message integrity.
> > >
> > >
> > > Integrity of which bits?  And are you sure that 
> maintaining integrity
> buys
> > > you anything except for in a few special cases?
> >
> > Definitely.
> >
> > I don't want someone removing my video stream my session.
> > I don't want someone to modify my codecs, forcing me to use a lower
> > quality one when a better one would actually work.
> > I don't want someone to receive my media when I intended it for the
> > party I called.
> >
> > And so on. The above concerns are real, and cover a good set of the
> > information in SDP.
> >
> > Let me pose the question in reverse. What information in 
> SDP do you feel
> > is NOT worth protecting?
> >
> > -Jonathan R.
> > --
> > Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> > Chief Scientist                             First Floor
> > dynamicsoft                                 East Hanover, NJ 07936
> > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > http://www.jdrosen.net                      PHONE: (973) 952-5000
> > http://www.dynamicsoft.com
> >
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.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] B2BUA & Max-Forwards
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On Thursday, December 5, 2002, at 01:49 AM, Phil Reynolds wrote:
> Our system isn't broken. I would say that this shows that the spec is 
> broken not our system.

That remains to be seen.  If I demonstrated how to satisfy the business 
goal you are trying to solve without violating the spec, then I suspect 
the spec is just fine.

> Our system provides facilities for some (and we hope "most") users. We 
> don't provide these facilities for all users because: A) We don't 
> think they are important for most users, B) The spec precludes us from 
> doing so (e.g. end-to-end security)

... or that you preclude secure customers from using your product.

> But because the spec says we MUST do end-to-end security we can't be 
> sanctioned as compliant, despite the fact that "most" of our users 
> don't want that. (BTW I have yet to see a endpoint that does 
> end-to-end security so it appears that these are all broken as well)

re: end-to-end, check out the datasheet for:

	http://www.marconi.com/html/products/virtualpresencesystem.htm

likewise I believe Pingtel has an implementation.

thanks,
-rohan

> -----Original Message-----
> From: Adam Roach [mailto:adam@dynamicsoft.com]
> Sent: 04 December 2002 20:45
> To: 'Phil Reynolds'; sip@ietf.org
> Subject: RE: [Sip] B2BUA & Max-Forwards
>
>
> -----Original Message-----
> From: Phil Reynolds [mailto:preynolds@ridgewaysystems.com]
>
> > Now your back to restricting what I can and can't do in an
> > intermediary. It can't be a B2BUA 'cos it doesn't need to honour
> > require headers (the UA at the other end does that, if they were 
> meant
> > for my intermediary then they would be proxy-require?), and it can't
> > be a proxy 'cos of the rules about not changing the body so what is
> > it?
>
> Broken.
>
> The restrictions placed on nodes by 3261 (which I'm merely 
> reiterating, not imposing) aren't an authoritarian attempt at 
> unnecessary proscription. They serve a purpose.
>
> I'll describe one of the (several) purposes that applies
> to the situation you propose.
>
> One of the things that UAs are required to be able to do is sign 
> and/or encrypt the bodies and/or some subset of the headers to ensure 
> confidentiality and/or integrity. The protocol places restrictions on 
> various nodes to ensure that this can be done. Your box is broken 
> precisely because it interferes with these abilities (or, more 
> precisely, these abilities interfere with it).
>
> /a
>

--Apple-Mail-41-781012764
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On Thursday, December 5, 2002, at 01:49 AM, Phil Reynolds wrote:

<excerpt><smaller>Our system isn't broken. I would say that this shows
that the spec is broken not our system.</smaller>

</excerpt>

That remains to be seen.  If I demonstrated how to satisfy the
business goal you are trying to solve without violating the spec, then
I suspect the spec is just fine.


<excerpt><smaller>Our system provides facilities for some (and we hope
"most") users. We don't provide these facilities for all users
because: A) We don't think they are important for most users, B) The
spec precludes us from doing so (e.g. end-to-end security)</smaller>

</excerpt>

... or that you preclude secure customers from using your product.


<excerpt><smaller>But because the spec says we MUST do end-to-end
security we can't be sanctioned as compliant, despite the fact that
"most" of our users don't want that. (BTW I have yet to see a endpoint
that does end-to-end security so it appears that these are all broken
as well)</smaller>

</excerpt>

re: end-to-end, check out the datasheet for:


	http://www.marconi.com/html/products/virtualpresencesystem.htm


likewise I believe Pingtel has an implementation.


thanks,

-rohan


<excerpt><smaller>-----Original Message-----

From: Adam Roach
[<underline><color><param>1999,1999,FFFF</param>mailto:adam@dynamicsoft.com</color></underline>]

Sent: 04 December 2002 20:45

To: 'Phil Reynolds'; sip@ietf.org

Subject: RE: [Sip] B2BUA & Max-Forwards</smaller>



<smaller>-----Original Message-----

From: Phil Reynolds
[<underline><color><param>1999,1999,FFFF</param>mailto:preynolds@ridgewaysystems.com</color></underline>]</smaller>


<smaller>> Now your back to restricting what I can and can't do in an

> intermediary. It can't be a B2BUA 'cos it doesn't need to honour

> require headers (the UA at the other end does that, if they were
meant

> for my intermediary then they would be proxy-require?), and it can't

> be a proxy 'cos of the rules about not changing the body so what is

> it?</smaller>


<smaller>Broken.</smaller>


<smaller>The restrictions placed on nodes by 3261 (which I'm merely
reiterating, not imposing) aren't an authoritarian attempt at
unnecessary proscription. They serve a purpose.</smaller>


<smaller>I'll describe one of the (several) purposes that applies

to the situation you propose.</smaller>


<smaller>One of the things that UAs are required to be able to do is
sign and/or encrypt the bodies and/or some subset of the headers to
ensure confidentiality and/or integrity. The protocol places
restrictions on various nodes to ensure that this can be done. Your
box is broken precisely because it interferes with these abilities
(or, more precisely, these abilities interfere with it).</smaller>


<smaller>/a</smaller>


</excerpt>
--Apple-Mail-41-781012764--

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On Thursday, December 5, 2002, at 11:35 PM, 
Juan-Carlos.Rojas@alcatel.fr wrote:

>
> Hello,
>
> "I don't get this.  Does two or more mean, for example, four?"
> Two or more means exactly that: two or more. The exact number depends 
> on
> what the application is actually providing, and you cannot fixe it in
> advance.

think "starfish"

thx,
-r


> "but a pair of UAC/UAS concatenations"
> OK, I get your point, you are thinking on a specific class of 
> application
> for bi-party communications. But this is only one case.
>
> At the end, I start to agree with some other people:
> - either we are able to provide a definition of a B2BUA independently 
> of
> what the B2BUA is actually doing
> - or we drop the definition
>
> Best regards
> Juan Carlos
>
>
>
>
>
> <aki.niemi@nokia.com> on 05/12/2002 19:54:26
>
> To:   Juan-Carlos ROJAS/FR/ALCATEL@ALCATEL
> cc:   <sip@ietf.org>
> Subject:  RE: [Sip] B2BUA & Max-Forwards
>
>
> Hi,
>
> inline.
>
>> "I am at the point where I agree with the sentiment raised by others,
>> that we should remove the term b2bua entirely from the
>> specification. It
>> has caused more confusion than anything else."
>>
>> Yes, this is one possibility. An alternative (as I proposed
>> in an earlier
>> mail) is to complete the current definition in 3261 saying
>> explicitly that
>> :
>> 1) A B2BUA handles two or more UAs, each one behaving as UAC/UAS, as
>> described in 3261 for UAs
>
> I don't get this.  Does two or more mean, for example, four?  And does 
> that
> in turn make it a B2BB2BUA? ;)
>
> The only minor clarification I think might be in order for RFC3261 is 
> that
> it's not just a UAC/UAS concatenation, but a pair of UAC/UAS
> concatenations.
>
> However, the fact that the traffic in this thread alone equals to a 
> year's
> worth of emails on the SIMPLE list suggests that no clarification is 
> enough
> clarification.
>
> So I agree with the other people in that getting rid of the definition
> altogether is the best option.  After all, if it looks like a UA to the
> outside world, it must be a UA.
>
>> 2) The SIP protocol specification does not mandate for any
>> behaviour of
>> such B2BUA, i.e. on how to "link" (or "concatenate") two or more UAs.
>
> Yes, probably the reason why the definition for a B2BUA is also 
> useless.
>
> Cheers,
> Aki
>
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.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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Hi Folks,

Let's please leave this flame war for the IEPREP mailing list, ok ;-)

thanks,
-rohan


On Thursday, December 5, 2002, at 05:45 AM, Liebhart Rainer wrote:

> Henry,
>
> unfortunately you are comparing apples and pears when stating that the 
> telecom service failed for days after Sept 11, while Internet traffic 
> continued smoothly. The mentioned "telefon service" consists of 
> thousands of (real-time) calls to emergency centres while the Internet 
> traffic consists mainly of requests to the Web pages of CNN and others 
> (with bad response times) to retrieve *information* on the status of 
> the catastrophe.
>
> Rainer
>
>
> -----Ursprüngliche Nachricht-----
> Von: Pete Cordell [mailto:pete@tech-know-ware.com]
> Gesendet: Donnerstag, 5. Dezember 2002 13:25
> An: Henry Sinnreich
> Cc: sip@ietf.org
> Betreff: Re: [Sip] B2BUA
>
>
> Henry,
>
> The Internet has had a few famous outages as well.  Besides, if all the
> telco traffic was put over IP, with today's level of QoS on the 
> Internet, it
> would have been brought to its knees also.
>
> To say that one is all goodness and light and the other all evil 
> doesn't do
> us as a community any good as when customers find out there are 
> problems
> they tend not to believe anything else we say!
>
> Only vapourware is flawless.  Most people know that.  Respect the
> intelligence of your customer, or they will find somebody that does!!!
>
> Pete.
>
> ----- Original Message -----
> From: "Henry Sinnreich" <Henry.Sinnreich@wcom.com>
> To: "'Phil Reynolds'" <preynolds@ridgewaysystems.com>; <sip@ietf.org>
> Sent: 04 December 2002 19:02
> Subject: RE: [Sip] B2BUA
>
>
>>> The telecoms people are clever enough to have thought
>>> about that and build redundancy, fault tolerance and failover into
>> their switches.
>>
>> This has proven unfortunately not to be the case on September 11 when
>> telecom service failed for days, while Internet traffic continued
>> smoothly. I suggest you to look at the central control model of the
>> switch network versus the e2e architecture of the Internet. SIP 
>> enabled
>> networks are also fully distributed and the e2e control principle
>> maintained - that's why state in placed in SIP endpoints.
>>
>> I am not sure if a few words here can give full justice to the 
>> superior
>> resilience of the Internet and to SIP enabled services,...
>>
>> Thanks, Henry
>>
>> -----Original Message-----
>> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Phil
>> Reynolds
>> Sent: Wednesday, December 04, 2002 7:37 AM
>> To: sip@ietf.org
>> Subject: RE: [Sip] B2BUA
>>
>>
>> If a telephone switch crashes you loose all calls...  The telecoms
>> people are clever enough to have thought about that and build
>> redundancy, fault tolerance and failover into their switches.
>> Maybe some of us IP-telecoms people have thought of similar techniques
>> for SIP intermediaries?
>> -----Original Message-----
>> From: Jiri Kuthan [mailto:jiri.kuthan@fokus.fraunhofer.de]
>> Sent: 04 December 2002 12:32
>> To: Christian Huitema; Roy, Radhika R, ALASO; 
>> mbarnes@nortelnetworks.com
>>
>> Cc: dean.willis@softarmor.com; Juan-Carlos.Rojas@alcatel.fr;
>> sip@ietf.org
>> Subject: RE: [Sip] B2BUA
>> [...snip...]
>> I think there are more issues with B2BUA. A B2BUA's failure affects 
>> all
>> current calls :-(
>> -Jiri
>> _______________________________________________
>> Sip mailing list  https://www1.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://www1.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://www1.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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> 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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In addition to Jonathan's approach to session policy, there is no 
reason a middlebox controller can't be a proxy if it:

1. opens pinholes referenced in a clear text or integrity protected SDP
2. rejects requests with no readable SDP (the UA could instead include 
a full SDP, with keying material, encrypted with SMIME, and an 
abbreviated copy which is only integrity protected).
3. rejects requests which use private addresses, and include in its 
response an offer with corresponding public addresses.

hope this helps.
thanks,
-rohan



On Thursday, December 5, 2002, at 09:14 AM, Mahey, Sonit wrote:

> I agree with Jonathan.
>
> That brings up the question:
> Is NAT traversal for encrypted SIP traffic addressed anywhere?
>
> - sonit
>
>> -----Original Message-----
>> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
>> Sent: Thursday, December 05, 2002 11:28 AM
>> To: Pete Cordell
>> Cc: Christian Huitema; sip@ietf.org
>> Subject: Re: [Sip] B2BUA - Security
>>
>>
>>
>>
>> Pete Cordell wrote:
>>
>>> I would say that encrypting and / or integrity checking of SDP alone
>>> achieves absolutely no security benefit.
>>
>> Really?? Wow. I actually think it prevents the most
>> devastating attacks
>> against SIP. These include:
>>
>> 1. modifying SDP in transit to point media streams to an intermediate
>> recording device, allowing attackers to establish wiretaps
>>
>> 2. downgrading any SRTP keying done in SDP, so that the media
>> itself is
>> not encrypted or integrity protected (this came up during the mmusic
>> session in IETF as a driver for S/MIME)
>>
>> 3. launching a DoS attack by sending everyones voice traffic to the
>> intended target. You get huge amplification gains here. Very
>> appealing
>> for hackers!
>>
>>
>>
>>
>>>
>>> SDP really does two things.  It specifies transport
>> addresses and chosen
>>> codecs.
>>>
>>> Let's assume that you do encrypt the SDP and I can't read it.
>>>
>>> I can have a good guess at the transport addresses in use
>> from many of the
>>> other fields in SIP such as via lines, To, From, uri,
>> contact, route etc.
>>
>> Confidentiality is only one property of security. I am much more
>> concerned about message integrity.
>>
>>> (Put another way, encrypting SDP is another form of
>> security by obscurity.
>>> And we all know what that means.)
>>
>> I have no idea how you would arrive at such a conclusion. Security
>> through obscurity means that information is not cryptographically
>> protected, but encoded with a known process that is just a bit
>> complicated. By definition, this is not encryption. Perhaps you have
>> discovered some new flaws in encryption technology?
>>
>> -Jonathan R.
>> -- 
>> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
>> Chief Scientist                             First Floor
>> dynamicsoft                                 East Hanover, NJ 07936
>> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>> http://www.jdrosen.net                      PHONE: (973) 952-5000
>> http://www.dynamicsoft.com
>>
>> _______________________________________________
>> Sip mailing list  https://www1.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://www1.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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Use sipping@ietf.org for new developments on the application of sip



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On Thursday, December 5, 2002, at 09:05 AM, Pete Cordell wrote:

> Dave,
>
> Please see inline...
>
> Pete.
>
> ----- Original Message -----
> From: "David R. Oran" <oran@cisco.com>
> To: "Pete Cordell" <pete@tech-know-ware.com>; "Christian Huitema"
> <huitema@windows.microsoft.com>
> Cc: <sip@ietf.org>
> Sent: 05 December 2002 14:45
> Subject: Re: [Sip] B2BUA - Security
>
>
>> --On Thursday, December 05, 2002 12:55 PM +0000 Pete Cordell
>> <pete@tech-know-ware.com> wrote:
>>
>>> Christian,
>>>
>>> I'm afraid I think that end-to-end security of SDP is a complete red
>>> herring.
>>>
>>> I would say that encrypting and / or integrity checking of SDP alone
>>> achieves absolutely no security benefit.
>>>
>>> SDP really does two things.  It specifies transport addresses and 
>>> chosen
>>> codecs.
>>>
>> ...and carries keys for SRTP...hmmmmmmmm........
>
> I assumed the keying used some sort of Diffee-Hellman type technique, 
> and so
> this wouldn't be an issue.  I will look into this.  Maybe that area of 
> SDP
> usage could be improved.

see draft-baugher-mmusic-sdpmediasec-00.txt

thx,
-r

>>
>>> Let's assume that you do encrypt the SDP and I can't read it.
>>>
>>> I can have a good guess at the transport addresses in use from many 
>>> of
> the
>>> other fields in SIP such as via lines, To, From, uri, contact, route
> etc.
>>>
>> Uh huh, sure. My experience is that you can only guess with reasonable
>> probability by knowing the implementation details of the UAs. I 
>> remember
> we
>> changed the RTP port allocation algorithm on IOS and all of a sudden
> people
>> started screaming because their ACL filters "stopped working". Trying 
>> to
>> beat the devil seem a pretty poor direction.
>
> Real time ACL control is very different issue to working out which RTP
> stream goes with what SIP session at some point.
>
>>
>>> By analysing the RTP traffic, I can easily find out what codec is 
>>> being
>>> used, either because the packet tells me, or from other statistical
>>> characteristics.
>>>
>> Really? Maybe for the standard codecs but why on earth bother?
>
> Your point is?
>
>>
>>> I have therefore discovered, with very little effort most of what I
> would
>>> find useful in SDP.  (Ironically sending stuff to a media relay would
> make
>>> this slightly harder!!!)
>>>
>>> If you want security of the media, then you have to encrypt and
> integrity
>>> check the RTP.
>>>
>> or put it through an IPSEC tunnel. What's the point?
>
> IPSEC would have trouble with NATs etc.  The point is to encrypt the 
> media
> some way.  I'm not seeing your point.
>
>>
>>> Encrypting SDP simply burns CPU cycles!!!
>>>
>> No, it prevents media hijacking by a MITM.
>
> No - not really.  It helps in some MITM situations but it is not a 
> complete
> solution.
>
>>
>>> (Put another way, encrypting SDP is another form of security by
> obscurity.
>>> And we all know what that means.)
>>>
>> Disagree vociferously.
>>
>>> (Note that it may make sense to encrypt other bodies types, such as
> those
>>> that contain HTML, but this is a different case.)
>>>
>>> Pete.
>>>
>> Dave.
>>
>>
>
> _______________________________________________
> Sip mailing list  https://www1.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: "Dean Willis" <dean.willis@softarmor.com>
To: "'Rohan Mahy'" <rohan@cisco.com>,
        "'Phil Reynolds'" <preynolds@ridgewaysystems.com>
Cc: <sip@ietf.org>
Subject: STOP! RE: [Sip] B2BUA & Max-Forwards
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Pardon me while I assert.
 
This thread is non-productive and downloading it is taking forever. I
think we have enough input to make any clarifications on B2BUA in bis
that I think we're going to be able to make.
 
The IESG has decreed that our specs WILL have provisions for security
functions, etc. and that anything that isn't compliant with these
provisions is NOT COMPLIANT with the specification. Enough said. If you
don't like it, talk to the non-com about changing the IESG or take it up
with the iESG.
 
Your can build ANY NON-COMPLIANT SYSTEM YOU WANT TO BUILD. Many such
systems will be USEFUL and WIDELY DEPLOYED. But it would be an error to
say that any such system is "compliant  o RFC ABCD" when that system
does not implement something that RFC ABCD lists as MUST IMPLEMENT. 
 
So if you have a thing you call a SIP proxy that does not implement MUST
IMPLEMENT functions of a SIP proxy as specified in RFC 3261, then it IS
NOT, by definition, a fully compliant SIP proxy. Odds are, it's some
kind of B2BUA that acts mostly but not quite like a proxy, and that's
OK. But DO be advised that anything that is "not a compliant proxy" is
likely to have limitations with some kind of service, extension or usage
of SIP, and that we're NOT going to worry about that a whole lot in the
protocol design.  So be prepared to update the widget occasionally if it
needs it to continue being useful.
 
Thanks.
 
--
Dean Willis
Annoyed Co-Chair, SIP Working Group.
 

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	charset="us-ascii"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 6.00.2800.1126" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D665093500-07122002>Pardon=20
me while I assert.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D665093500-07122002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D665093500-07122002>This=20
thread is non-productive and downloading it is taking forever. I think =
we have=20
enough input to make any clarifications on B2BUA in bis that I think =
we're going=20
to be able to make.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D665093500-07122002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D665093500-07122002>The=20
IESG has decreed that our specs WILL have provisions for security =
functions,=20
etc. and that anything that isn't compliant with these provisions is NOT =

COMPLIANT with the specification. Enough said. If you don't like it, =
talk to the=20
non-com about changing the IESG or take it up with the =
iESG.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D665093500-07122002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D665093500-07122002>Your=20
can build ANY NON-COMPLIANT SYSTEM YOU WANT TO BUILD. Many such systems =
will be=20
USEFUL and WIDELY DEPLOYED. But it would be an error to say that any =
such system=20
is "compliant&nbsp; o RFC ABCD" when that system does not implement =
something=20
that RFC ABCD lists as MUST IMPLEMENT. </SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D665093500-07122002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D665093500-07122002>So if=20
you have a thing you call a SIP proxy that does not implement MUST =
IMPLEMENT=20
functions of a SIP proxy as specified in RFC 3261, then it IS NOT, by=20
definition,&nbsp;a fully compliant SIP proxy. Odds are, it's some kind =
of B2BUA=20
that acts mostly but not quite like a proxy, and that's OK. But DO be =
advised=20
that anything that is "not a compliant proxy" is likely to have =
limitations with=20
some kind of service, extension or usage of SIP, and that we're NOT =
going to=20
worry about that a whole lot in the protocol design.&nbsp; So be =
prepared to=20
update the widget occasionally if it needs it to continue being=20
useful.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D665093500-07122002></SPAN></FONT><FONT face=3DArial =
color=3D#0000ff=20
size=3D2><SPAN class=3D665093500-07122002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D665093500-07122002>Thanks.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D665093500-07122002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D665093500-07122002>--</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D665093500-07122002>Dean=20
Willis</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D665093500-07122002>Annoyed Co-Chair, SIP Working=20
Group.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff =
size=3D2></FONT>&nbsp;</DIV></BODY></HTML>

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Thread-Topic: Refer subscription clarification
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From: "Hearty, John" <John.Hearty@Level3.com>
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Subject: [Sip] Refer subscription clarification
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The refer-07 draft states:

"The lifetime of the state being subscribed to is determined by the progress of the referenced request."

If the original dialog that was refered is dropped after the transferee already sent a 202 to the transferor for the Refer, is the subscription still considered alive by the Transferor?  Example flow to illustrate what I'm getting at:

              Transferor           Transferee             Transfer
                   |                    |                  Target
                   |            INVITE  |                    |
         Call-ID:1 |<-------------------|                    |
                   |            200 OK  |                    |
         Call-ID:1 |------------------->|                    |
                   |            ACK     |                    |
         Call-ID:1 |<-------------------|                    |
                   |  INVITE (hold)     |                    |
         Call-ID:1 |------------------->|                    |
                   |  200 OK            |                    |
         Call-ID:1 |<-------------------|                    |
                   |  ACK               |                    |
         Call-ID:1 |------------------->|                    |
                   |  REFER             |                    |
         Call-ID:1 |------------------->|                    |
                   |  202 Accepted      |                    |
         Call-ID:1 |<-------------------|                    |
                   | NOTIFY (100 Trying)|                    |
         Call-ID:1 |<-------------------|                    |
                   |            200 OK  |                    |
         Call-ID:1 |------------------->|                    |
                   |                    |  INVITE            |
         Call-ID:2 |                    |------------------->|
                   |                    |  180               |
         Call-ID:2 |                    |<-------------------|
                   |  BYE               |                    |
         Call-ID:1 |<-------------------|                    |
                   |  200 OK            |                    |
         Call-ID:1 |------------------->|                    |
                   |                    |  CANCEL            |
         Call-ID:2 |                    |------------------->|
                   |                    |  200 OK            |
         Call-ID:2 |                    |<-------------------|
                   |                    |             487    |
         Call-ID:2 |                    |<-------------------|
                   |                    |             ACK    |
         Call-ID:2 |                    |------------------->|

 Is the Transferee expected to still send a Notify at this point?  I presume he is based on the text above, and that the subscription is not over when the Transferor gets the BYE for Call-ID 1.  Please confirm.  I realize the CANCEL of Call-ID:2 should probabally happen before the BYE on Call-ID:1, I have reasons I'd rather not go into why it is happening in the order shown.  Posted to the sip list because I could not find any clear reference to this, and clarification may be needed in the draft.


John Hearty
Level 3
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Subject: Re: [Sip] Refer subscription clarification
From: Robert Sparks <rsparks@dynamicsoft.com>
To: "Hearty, John" <John.Hearty@level3.com>
Cc: sip@ietf.org
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	<3DB099919328BA41905F651A6EF83976083F88@idc1exc0004.corp.global.level3.com>
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inline

On Fri, 2002-12-06 at 21:58, Hearty, John wrote:
> The refer-07 draft states:
> 
> "The lifetime of the state being subscribed to is determined by the progress of the referenced request."
> 
> If the original dialog that was refered is dropped after the transferee already sent a 202 to the transferor for the Refer, is the subscription still considered alive by the Transferor?  Example flow to illustrate what I'm getting at:
I dont know what "the original dialog that was refered" means.
I am assuming from the flow that you are asking this question.
  
   If I send a REFER inside a dialog created by an INVITE, does
   a BYE terminate the subscription?

which is exactly the same question as
  
   If I send a SUBSCRIBE inside a dialog created by an INVITE, does
   a BYE terminate the subscription?

If that's not your question ask again.

The answer to the above question is no - and the answer to your below
question is yes, a NOTIFY is still expected.

If that was your question, then I don't think _this_ draft needs
further clarification. That's a basic use of sip-events question, as
we've discussed on the list many times, and should be clarified there.

RjS

> 
>               Transferor           Transferee             Transfer
>                    |                    |                  Target
>                    |            INVITE  |                    |
>          Call-ID:1 |<-------------------|                    |
>                    |            200 OK  |                    |
>          Call-ID:1 |------------------->|                    |
>                    |            ACK     |                    |
>          Call-ID:1 |<-------------------|                    |
>                    |  INVITE (hold)     |                    |
>          Call-ID:1 |------------------->|                    |
>                    |  200 OK            |                    |
>          Call-ID:1 |<-------------------|                    |
>                    |  ACK               |                    |
>          Call-ID:1 |------------------->|                    |
>                    |  REFER             |                    |
>          Call-ID:1 |------------------->|                    |
>                    |  202 Accepted      |                    |
>          Call-ID:1 |<-------------------|                    |
>                    | NOTIFY (100 Trying)|                    |
>          Call-ID:1 |<-------------------|                    |
>                    |            200 OK  |                    |
>          Call-ID:1 |------------------->|                    |
>                    |                    |  INVITE            |
>          Call-ID:2 |                    |------------------->|
>                    |                    |  180               |
>          Call-ID:2 |                    |<-------------------|
>                    |  BYE               |                    |
>          Call-ID:1 |<-------------------|                    |
>                    |  200 OK            |                    |
>          Call-ID:1 |------------------->|                    |
>                    |                    |  CANCEL            |
>          Call-ID:2 |                    |------------------->|
>                    |                    |  200 OK            |
>          Call-ID:2 |                    |<-------------------|
>                    |                    |             487    |
>          Call-ID:2 |                    |<-------------------|
>                    |                    |             ACK    |
>          Call-ID:2 |                    |------------------->|
> 
>  Is the Transferee expected to still send a Notify at this point?  I presume he is based on the text above, and that the subscription is not over when the Transferor gets the BYE for Call-ID 1.  Please confirm.  I realize the CANCEL of Call-ID:2 should probabally happen before the BYE on Call-ID:1, I have reasons I'd rather not go into why it is happening in the order shown.  Posted to the sip list because I could not find any clear reference to this, and clarification may be needed in the draft.
> 
> 
> John Hearty
> Level 3
> _______________________________________________
> Sip mailing list  https://www1.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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Use sipping@ietf.org for new developments on the application of sip



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Hi,

> >The body does not start with LWS. The body starts with WSP (RFC3261 doesn't
> >define what WSP is, but if I remember correct it is SP or TAB). So, in that
> >case you will have:
> >
> >CRLF CRLF WSP
> >
> >...which, according to the ABNF is:
> >
> >CRLF LWS
>
> You are taking this out of context. It is not the case that you would
> parse through the message replaceing all instances of CRLF WSP with LWS.

So, where in the rules is it said when I should do that, and when I should not?
To my understanding the LWS rule is defined as a "general rule".

> The BNF specifies that message-header ends with a CRLF.

Yes.

> What follows that is one of two things - either another CRLF, marking the end
> of the
> headers, or another message header, which never begins with LWS.

From a SIP grammar point of view that is correct. However, I don't see it said
anywhere that the definition of LWS isn't valid at the beginning of a new line.

Also, just for the record, where is it said that a sip-header can't start with
LWS? Where is it said that line folding can't be used at the beginning of a line?
The text only says that the sip-header field values can be folded onto multiple
lines.

> Thus, if a CRLF is followed by something OTHER than WSP, the next construction

> cannot be LWS.

Maybe I missunderstand you, but the case I try to describe is when the CRLF IS
followed by WSP.

Regards,

Christer Holmberg
Ericsson Finland


_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From mailnull@www1.ietf.org  Mon Dec  9 04:14:59 2002
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From: Loh Kok Jeng <kokjeng.loh@siemens.com>
To: sip@ietf.org
Date: Mon, 9 Dec 2002 17:03:53 +0800 
Subject: [Sip] Handling of REFER/NOTIFY by Stateful Proxy
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Dear all,

I have a question regarding the handling of REFER/NOTIFY that is tied to 
a session initiated with INVITE, i.e. REFER/NOTIFY belongs to the same 
dialog as INVITE.  What should a stateful proxy do when it receives a 
NOTIFY whose dialog has been terminated?  I have 2 specific scenarios in 
mind, and both of them are illustrated in the diagrams below.

Case 1:
The Referrer terminates the session with BYE.  The stateful proxy receives
NOTIFY after 200 OK for BYE.  At that particular instance, the stateful
proxy has already removed the dialog.

Case 2:
The session expires and the stateful proxy removed the session.  It receives
NOTIFY from UAS.

For these two cases, should the stateful proxy rejects NOTIFY with 481?  Or
should it proxy NOTIFY onwards?

Thanks.

regards,
KJ Loh



  +--------+                +--------+                 +--------+
  |Referrer|                |Stateful|                 |  UAS   |
  |        |                | Proxy  |                 |        |
  +--------+    1 INVITE    +--------+                 +--------+
      |------------------------->|         2 INVITE         |
      |                          |------------------------->|
      |                          |         3 200 OK         |
      |                          |<-------------------------|
      |         4 200 OK         |                          |
      |<-------------------------|                          |
      |         5 REFER          |                          |
      |------------------------->|         6 REFER          |
      |                          |------------------------->|
      |                          |         7 200 OK         |
      |                          |<-------------------------|
      |         8 200 OK         |                          |
      |<-------------------------|                          |
      |                          |         9 NOTIFY         |
      |                          |<-------------------------|
      |         10 NOTIFY        |                          |
      |<-------------------------|                          |
      |         11 200 OK        |                          |
      |------------------------->|         12 200 OK        |
      |                          |------------------------->|
      |        .......           |         ......           |
      |                          |                          |
      |        13 REFER          |                          |
      |------------------------->|        14 REFER          |
      |                          |------------------------->|
      |                          |        15 200 OK         |
      |                          |<-------------------------|
      |        16 200 OK         |                          |
      |<-------------------------|                          |
      |        17 BYE            |                          |
      |------------------------->|        18 BYE            |
      |                          |------------------------->|
      |                          |        19 200 OK         |
      |                          |<-------------------------|
      |        21 200 OK         |        20 NOTIFY         |
      |<-------------------------|<-------------------------|



  +--------+                +--------+                 +--------+
  |Referrer|                |Stateful|                 |  UAS   |
  |        |                | Proxy  |                 |        |
  +--------+    1 INVITE    +--------+                 +--------+
      |------------------------->|         2 INVITE         |
      |                          |------------------------->|
      |                          |         3 200 OK         |
      |                          |<-------------------------|
      |         4 200 OK         |                          |
      |<-------------------------|                          |
      |         5 REFER          |                          |
      |------------------------->|         6 REFER          |
      |                          |------------------------->|
      |                          |         7 200 OK         |
      |                          |<-------------------------|
      |         8 200 OK         |                          |
      |<-------------------------|                          |
      |                          |         9 NOTIFY         |
      |                          |<-------------------------|
      |         10 NOTIFY        |                          |
      |<-------------------------|                          |
      |         11 200 OK        |                          |
      |------------------------->|         12 200 OK        |
      |                          |------------------------->|
      |        .......           |         ......           |
      |                          |                          |
      |        13 REFER          |                          |
      |------------------------->|        14 REFER          |
      |                          |------------------------->|
      |                          |        15 200 OK         |
      |                          |<-------------------------|
      |        16 200 OK         |                          |
      |<-------------------------|                          |
      |                +-----------------+                  |
      |                |Session Timeout  |                  |
      |                |Session torn down|                  |
      |                +-----------------+                  |
      |                          |        17 NOTIFY         |
      |                          |<-------------------------|
_______________________________________________
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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
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From: "Mark Watson" <mwatson@nortelnetworks.com>
To: "'Rohan Mahy'" <rohan@cisco.com>,
        "Mahey, Sonit"
	 <Sonit.Mahey@icn.siemens.com>
Cc: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        Pete Cordell
	 <pete@tech-know-ware.com>,
        Christian Huitema
	 <huitema@windows.microsoft.com>, sip@ietf.org
Subject: RE: [Sip] B2BUA - Security
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Rohan wrote:

> In addition to Jonathan's approach to session policy, there is no 
> reason a middlebox controller can't be a proxy if it:
> 
> 1. opens pinholes referenced in a clear text or integrity 
> protected SDP
> 2. rejects requests with no readable SDP (the UA could 
> instead include 
> a full SDP, with keying material, encrypted with SMIME, ****and an 
> abbreviated copy which is only integrity protected****).


I was about to write a draft suggesting exactly this.

As per Jonathan's draft there is a need for the UA to be able to 'declare'
to network entities those parts of the SDP information which are needed for
the network entity to perform whatever service it performs (e.g. NAT or
Firewall control). But the UA should not have to declare the whole SDP, and
the terminating UA should be able to tell if a network entity has modified
these parameters.

Rather than putting this information into new SIP headers, we could include
two copies of the SDP. One copy is unprotected and contains only the
information that the UE wishes to 'declare' to the network entities. The
other copy is protected and contains the full information.

This approach would work through existing devices which inspect/modify SDP.

I'm not sure how an existing S/MIME compliant client would behave on
receiving these two copies. If it uses the unprotected SDP, then things are
no worse than the present situation - the call suceeds, but without the
advantages of S/MIME. If it uses the protected SDP (which has not been
modified by the NAT controlling device), then the call fails, but this is
still no worse than the present situation.

I guess if people liked this approach we would need to document somewhere
the UAS behaviour, namely: compare protected & unprotected copies & take
'appropriate' action on discrepencies. etc. Also to recommend that the UAC
include both copies when it uses S/MIME encryption.

Regards...Mark


> 3. rejects requests which use private addresses, and include in its 
> response an offer with corresponding public addresses.
> 
> hope this helps.
> thanks,
> -rohan
> 
> 
> 
> On Thursday, December 5, 2002, at 09:14 AM, Mahey, Sonit wrote:
> 
> > I agree with Jonathan.
> >
> > That brings up the question:
> > Is NAT traversal for encrypted SIP traffic addressed anywhere?
> >
> > - sonit
> >
> >> -----Original Message-----
> >> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> >> Sent: Thursday, December 05, 2002 11:28 AM
> >> To: Pete Cordell
> >> Cc: Christian Huitema; sip@ietf.org
> >> Subject: Re: [Sip] B2BUA - Security
> >>
> >>
> >>
> >>
> >> Pete Cordell wrote:
> >>
> >>> I would say that encrypting and / or integrity checking 
> of SDP alone
> >>> achieves absolutely no security benefit.
> >>
> >> Really?? Wow. I actually think it prevents the most
> >> devastating attacks
> >> against SIP. These include:
> >>
> >> 1. modifying SDP in transit to point media streams to an 
> intermediate
> >> recording device, allowing attackers to establish wiretaps
> >>
> >> 2. downgrading any SRTP keying done in SDP, so that the media
> >> itself is
> >> not encrypted or integrity protected (this came up during 
> the mmusic
> >> session in IETF as a driver for S/MIME)
> >>
> >> 3. launching a DoS attack by sending everyones voice traffic to the
> >> intended target. You get huge amplification gains here. Very
> >> appealing
> >> for hackers!
> >>
> >>
> >>
> >>
> >>>
> >>> SDP really does two things.  It specifies transport
> >> addresses and chosen
> >>> codecs.
> >>>
> >>> Let's assume that you do encrypt the SDP and I can't read it.
> >>>
> >>> I can have a good guess at the transport addresses in use
> >> from many of the
> >>> other fields in SIP such as via lines, To, From, uri,
> >> contact, route etc.
> >>
> >> Confidentiality is only one property of security. I am much more
> >> concerned about message integrity.
> >>
> >>> (Put another way, encrypting SDP is another form of
> >> security by obscurity.
> >>> And we all know what that means.)
> >>
> >> I have no idea how you would arrive at such a conclusion. Security
> >> through obscurity means that information is not cryptographically
> >> protected, but encoded with a known process that is just a bit
> >> complicated. By definition, this is not encryption. 
> Perhaps you have
> >> discovered some new flaws in encryption technology?
> >>
> >> -Jonathan R.
> >> -- 
> >> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> >> Chief Scientist                             First Floor
> >> dynamicsoft                                 East Hanover, NJ 07936
> >> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> >> http://www.jdrosen.net                      PHONE: (973) 952-5000
> >> http://www.dynamicsoft.com
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.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://www1.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://www1.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
> 

------_=_NextPart_001_01C29F6B.7129FF50
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<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2655.35">
<TITLE>RE: [Sip] B2BUA - Security</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Rohan wrote:</FONT>
</P>

<P><FONT SIZE=3D2>&gt; In addition to Jonathan's approach to session =
policy, there is no </FONT>
<BR><FONT SIZE=3D2>&gt; reason a middlebox controller can't be a proxy =
if it:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; 1. opens pinholes referenced in a clear text or =
integrity </FONT>
<BR><FONT SIZE=3D2>&gt; protected SDP</FONT>
<BR><FONT SIZE=3D2>&gt; 2. rejects requests with no readable SDP (the =
UA could </FONT>
<BR><FONT SIZE=3D2>&gt; instead include </FONT>
<BR><FONT SIZE=3D2>&gt; a full SDP, with keying material, encrypted =
with SMIME, ****and an </FONT>
<BR><FONT SIZE=3D2>&gt; abbreviated copy which is only integrity =
protected****).</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>I was about to write a draft suggesting exactly =
this.</FONT>
</P>

<P><FONT SIZE=3D2>As per Jonathan's draft there is a need for the UA to =
be able to 'declare' to network entities those parts of the SDP =
information which are needed for the network entity to perform whatever =
service it performs (e.g. NAT or Firewall control). But the UA should =
not have to declare the whole SDP, and the terminating UA should be =
able to tell if a network entity has modified these =
parameters.</FONT></P>

<P><FONT SIZE=3D2>Rather than putting this information into new SIP =
headers, we could include two copies of the SDP. One copy is =
unprotected and contains only the information that the UE wishes to =
'declare' to the network entities. The other copy is protected and =
contains the full information.</FONT></P>

<P><FONT SIZE=3D2>This approach would work through existing devices =
which inspect/modify SDP.</FONT>
</P>

<P><FONT SIZE=3D2>I'm not sure how an existing S/MIME compliant client =
would behave on receiving these two copies. If it uses the unprotected =
SDP, then things are no worse than the present situation - the call =
suceeds, but without the advantages of S/MIME. If it uses the protected =
SDP (which has not been modified by the NAT controlling device), then =
the call fails, but this is still no worse than the present =
situation.</FONT></P>

<P><FONT SIZE=3D2>I guess if people liked this approach we would need =
to document somewhere the UAS behaviour, namely: compare protected =
&amp; unprotected copies &amp; take 'appropriate' action on =
discrepencies. etc. Also to recommend that the UAC include both copies =
when it uses S/MIME encryption.</FONT></P>

<P><FONT SIZE=3D2>Regards...Mark</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>&gt; 3. rejects requests which use private addresses, =
and include in its </FONT>
<BR><FONT SIZE=3D2>&gt; response an offer with corresponding public =
addresses.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; hope this helps.</FONT>
<BR><FONT SIZE=3D2>&gt; thanks,</FONT>
<BR><FONT SIZE=3D2>&gt; -rohan</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; On Thursday, December 5, 2002, at 09:14 AM, =
Mahey, Sonit wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; I agree with Jonathan.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; That brings up the question:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Is NAT traversal for encrypted SIP traffic =
addressed anywhere?</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; - sonit</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; From: Jonathan Rosenberg [<A =
HREF=3D"mailto:jdrosen@dynamicsoft.com">mailto:jdrosen@dynamicsoft.com</=
A>]</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; Sent: Thursday, December 05, 2002 =
11:28 AM</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; To: Pete Cordell</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; Cc: Christian Huitema; =
sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; Subject: Re: [Sip] B2BUA - =
Security</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; Pete Cordell wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;&gt; I would say that encrypting and / =
or integrity checking </FONT>
<BR><FONT SIZE=3D2>&gt; of SDP alone</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;&gt; achieves absolutely no security =
benefit.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; Really?? Wow. I actually think it =
prevents the most</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; devastating attacks</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; against SIP. These include:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; 1. modifying SDP in transit to point =
media streams to an </FONT>
<BR><FONT SIZE=3D2>&gt; intermediate</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; recording device, allowing attackers =
to establish wiretaps</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; 2. downgrading any SRTP keying done in =
SDP, so that the media</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; itself is</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; not encrypted or integrity protected =
(this came up during </FONT>
<BR><FONT SIZE=3D2>&gt; the mmusic</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; session in IETF as a driver for =
S/MIME)</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; 3. launching a DoS attack by sending =
everyones voice traffic to the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; intended target. You get huge =
amplification gains here. Very</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; appealing</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; for hackers!</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;&gt; SDP really does two things.&nbsp; =
It specifies transport</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; addresses and chosen</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;&gt; codecs.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;&gt; Let's assume that you do encrypt =
the SDP and I can't read it.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;&gt; I can have a good guess at the =
transport addresses in use</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; from many of the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;&gt; other fields in SIP such as via =
lines, To, From, uri,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; contact, route etc.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; Confidentiality is only one property =
of security. I am much more</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; concerned about message =
integrity.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;&gt; (Put another way, encrypting SDP =
is another form of</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; security by obscurity.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;&gt; And we all know what that =
means.)</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; I have no idea how you would arrive at =
such a conclusion. Security</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; through obscurity means that =
information is not cryptographically</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; protected, but encoded with a known =
process that is just a bit</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; complicated. By definition, this is =
not encryption. </FONT>
<BR><FONT SIZE=3D2>&gt; Perhaps you have</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; discovered some new flaws in =
encryption technology?</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; -Jonathan R.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; -- </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; Jonathan D. Rosenberg, =
Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; 72 Eagle Rock Ave.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; Chief =
Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; First Floor</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; =
dynamicsoft&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; East =
Hanover, NJ 07936</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; =
jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
FAX:&nbsp;&nbsp; (973) 952-5050</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; <A HREF=3D"http://www.jdrosen.net" =
TARGET=3D"_blank">http://www.jdrosen.net</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp; PHONE: (973) 952-5000</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; <A HREF=3D"http://www.dynamicsoft.com" =
TARGET=3D"_blank">http://www.dynamicsoft.com</A></FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; This list is for NEW development of =
the core SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; Use sip-implementors@cs.columbia.edu =
for questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt; Use sipping@ietf.org for new =
developments on the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; &gt; This list is for NEW development of the =
core SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Use sipping@ietf.org for new developments =
on the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the core =
SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; Use sipping@ietf.org for new developments on =
the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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Subject: RE: [Sip] B2BUA & Max-Forwards
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> -----Original Message-----
> From: ext Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Thursday, December 05, 2002 6:17 PM
> To: Pete Cordell
> Cc: Adam Roach; 'Phil Reynolds'; sip@ietf.org
> Subject: Re: [Sip] B2BUA & Max-Forwards
> 
> 
> I am at the point where I agree with the sentiment raised by others, 
> that we should remove the term b2bua entirely from the 
> specification. It 
> has caused more confusion than anything else.
> 

I agree. Just forget about this B2BUA thing. There are things called "Application Servers (AS)" that perform whatever service you wish to invent. And most importantly, they are UASs.

Think presence server. It is one type of AS. It receives SUBSCRIBE, on its UAS side, that is destined to a URI who's principle normally uses a terminal, and responds to them. Therefore needs to handle require-header.

This Presence Server may initiate a SUBSCRIBE of its own, from its UAC side, that is destined to the same user. Here we have a B2BUA, but I hate that name, so I say here we have an Application Server.

Think also presence list server. This is an example of one UAS to many UACs or one to many B2BUA. I like to call it Application Server.

So lets drop this term B2BUA and use Application Server instead. I think people understand this term better.

If 3GPP want every intermediary in the network to do something that is not SIP proxy compliant, then they better call it Application Server.

Regards,
Hisham
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Subject: RE: [Sip] Handling of REFER/NOTIFY by Stateful Proxy
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You are talking about Dialog Stateful proxy. In this case, the dialog at the proxy is not removed until the NOTIFY of a REFER terminates the dialog (or an unSUBSCRIBE).

Remember, from recent discussions on the mailing list and from RFC3265, a dialog exists until all "associations" with that dialog are terminated.

regards,
Hisham

> -----Original Message-----
> From: ext Loh Kok Jeng [mailto:kokjeng.loh@siemens.com]
> Sent: Monday, December 09, 2002 11:04 AM
> To: sip@ietf.org
> Subject: [Sip] Handling of REFER/NOTIFY by Stateful Proxy
> 
> 
> Dear all,
> 
> I have a question regarding the handling of REFER/NOTIFY that 
> is tied to 
> a session initiated with INVITE, i.e. REFER/NOTIFY belongs to 
> the same 
> dialog as INVITE.  What should a stateful proxy do when it receives a 
> NOTIFY whose dialog has been terminated?  I have 2 specific 
> scenarios in 
> mind, and both of them are illustrated in the diagrams below.
> 
> Case 1:
> The Referrer terminates the session with BYE.  The stateful 
> proxy receives
> NOTIFY after 200 OK for BYE.  At that particular instance, 
> the stateful
> proxy has already removed the dialog.
> 
> Case 2:
> The session expires and the stateful proxy removed the 
> session.  It receives
> NOTIFY from UAS.
> 
> For these two cases, should the stateful proxy rejects NOTIFY 
> with 481?  Or
> should it proxy NOTIFY onwards?
> 
> Thanks.
> 
> regards,
> KJ Loh
> 
> 
> 
>   +--------+                +--------+                 +--------+
>   |Referrer|                |Stateful|                 |  UAS   |
>   |        |                | Proxy  |                 |        |
>   +--------+    1 INVITE    +--------+                 +--------+
>       |------------------------->|         2 INVITE         |
>       |                          |------------------------->|
>       |                          |         3 200 OK         |
>       |                          |<-------------------------|
>       |         4 200 OK         |                          |
>       |<-------------------------|                          |
>       |         5 REFER          |                          |
>       |------------------------->|         6 REFER          |
>       |                          |------------------------->|
>       |                          |         7 200 OK         |
>       |                          |<-------------------------|
>       |         8 200 OK         |                          |
>       |<-------------------------|                          |
>       |                          |         9 NOTIFY         |
>       |                          |<-------------------------|
>       |         10 NOTIFY        |                          |
>       |<-------------------------|                          |
>       |         11 200 OK        |                          |
>       |------------------------->|         12 200 OK        |
>       |                          |------------------------->|
>       |        .......           |         ......           |
>       |                          |                          |
>       |        13 REFER          |                          |
>       |------------------------->|        14 REFER          |
>       |                          |------------------------->|
>       |                          |        15 200 OK         |
>       |                          |<-------------------------|
>       |        16 200 OK         |                          |
>       |<-------------------------|                          |
>       |        17 BYE            |                          |
>       |------------------------->|        18 BYE            |
>       |                          |------------------------->|
>       |                          |        19 200 OK         |
>       |                          |<-------------------------|
>       |        21 200 OK         |        20 NOTIFY         |
>       |<-------------------------|<-------------------------|
> 
> 
> 
>   +--------+                +--------+                 +--------+
>   |Referrer|                |Stateful|                 |  UAS   |
>   |        |                | Proxy  |                 |        |
>   +--------+    1 INVITE    +--------+                 +--------+
>       |------------------------->|         2 INVITE         |
>       |                          |------------------------->|
>       |                          |         3 200 OK         |
>       |                          |<-------------------------|
>       |         4 200 OK         |                          |
>       |<-------------------------|                          |
>       |         5 REFER          |                          |
>       |------------------------->|         6 REFER          |
>       |                          |------------------------->|
>       |                          |         7 200 OK         |
>       |                          |<-------------------------|
>       |         8 200 OK         |                          |
>       |<-------------------------|                          |
>       |                          |         9 NOTIFY         |
>       |                          |<-------------------------|
>       |         10 NOTIFY        |                          |
>       |<-------------------------|                          |
>       |         11 200 OK        |                          |
>       |------------------------->|         12 200 OK        |
>       |                          |------------------------->|
>       |        .......           |         ......           |
>       |                          |                          |
>       |        13 REFER          |                          |
>       |------------------------->|        14 REFER          |
>       |                          |------------------------->|
>       |                          |        15 200 OK         |
>       |                          |<-------------------------|
>       |        16 200 OK         |                          |
>       |<-------------------------|                          |
>       |                +-----------------+                  |
>       |                |Session Timeout  |                  |
>       |                |Session torn down|                  |
>       |                +-----------------+                  |
>       |                          |        17 NOTIFY         |
>       |                          |<-------------------------|
> _______________________________________________
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Subject: Re: [Sip] ABNF issue
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Christer Holmberg <christer.holmberg@lmf.ericsson.se>
Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@ietf.org
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inline

On Mon, 2002-12-09 at 01:16, Christer Holmberg wrote:
> 
> Hi,
> 
> > >The body does not start with LWS. The body starts with WSP (RFC3261 doesn't
> > >define what WSP is, but if I remember correct it is SP or TAB). So, in that
> > >case you will have:
> > >
> > >CRLF CRLF WSP
> > >
> > >...which, according to the ABNF is:
> > >
> > >CRLF LWS
> >
> > You are taking this out of context. It is not the case that you would
> > parse through the message replaceing all instances of CRLF WSP with LWS.
> 
> So, where in the rules is it said when I should do that, and when I should not?
> To my understanding the LWS rule is defined as a "general rule".

You need to be very careful here.

Just because there is a production rule that says A -> BC
does not mean all occurrences of BC in the language resulted
from the expansion of A. 

Blindly doing a reverse token substitution will produce an incorrect
parse. You can only apply it when you have proven the left side of
the rule fits in a valid parse of the message.

So from your original example. LWS is not defined inside a body - the
body is just a sequence of octets from the perspective of the SIP
grammar. Your SIP parser _cannot_ apply a match against LWS inside it -
there is no valid expansion of the SIP grammar that has LWS appearing
after CRLF CRLF
> 
> > The BNF specifies that message-header ends with a CRLF.
> 
> Yes.
> 
> > What follows that is one of two things - either another CRLF, marking the end
> > of the
> > headers, or another message header, which never begins with LWS.
> 
> >From a SIP grammar point of view that is correct. However, I don't see it said
> anywhere that the definition of LWS isn't valid at the beginning of a new line.

It _is_ valid after CRLF (which is what I think you mean by a new line)
if (and only if) you are still in the expansion of message-header.

> 
> Also, just for the record, where is it said that a sip-header can't start with
> LWS? Where is it said that line folding can't be used at the beginning of a line?
> The text only says that the sip-header field values can be folded onto multiple
> lines.
I don't know what you mean by beginning of a line here. I assume you are
asking why we cant have a header that looks like (\b for blank):
\b\bNewHeader: foo
which is a "\b\bNewheader"?

The grammar answers that. 
extension-header starts with a token.
> 
> > Thus, if a CRLF is followed by something OTHER than WSP, the next construction
> 
> > cannot be LWS.
> 
> Maybe I missunderstand you, but the case I try to describe is when the CRLF IS
> followed by WSP.
If its in a valid expansion of message-header (which will not happen
across a CRLFCRLF string), then you are looking at LWS. The _only_
place that second CRLF matches in the grammar is the literal in
the Request and Response rules. So for your example, any whitespace
following that second CRLF matches message-body = *OCTET, NOT LWS.

> 
> Regards,
> 
> Christer Holmberg
> Ericsson Finland
> 
> 
> _______________________________________________
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "Rosen, Brian" <Brian.Rosen@marconi.com>,
        "'aki.niemi@nokia.com'"
	 <aki.niemi@nokia.com>
Cc: sip@ietf.org
Subject: RE: [Sip] B2BUA
Date: Mon, 9 Dec 2002 15:41:46 -0000 
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I would suggest that in addition to this, we list in this proposed informational document Deans 10 rules of transparency (suitably enhanced if necessary). B2BUA providers can then use this as a tick list to indicate some of the functionality of the B2BUA, and receivers of such a list will get an idea of how significantly the entity breaks the end-to-end SIP transparency, which is after all the main thing of interest to the end UA.

Doing this may actually provide a document of some use, and allow IETF to put forward that even if it is a B2BUA, designing as much transparency into it, while still performing the un-proxy-like function that is intended, is a good thing.

Keith

Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com 

> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> Sent: 03 December 2002 16:29
> To: 'aki.niemi@nokia.com'
> Cc: sip@ietf.org
> Subject: RE: [Sip] B2BUA
> 
> 
>  
> > So the B2BUA is a concatenation of two User Agents, each of 
> > which acts both as UAC and UAS. A conference server will 
> > probably look like a UA, definitely not a B2BUA, IMHO.
> >    
> This thread is, hopefully reaching an end (with an agreement to
> do some examples with call flows).  However, the above statement
> is yet one more example of the confusion people have with B2BUA.
> 
> The key thing about a B2BUA is that it looks like a UA to whatever
> it is connected to (on both sides).  So, a conference server
> looks like a UA to each of it's ports, and it has at least two of
> them, and there is some "magic" between them.  If you look at the
> concept of the "focus" from the conferencing work, it is a
> thing that has a bunch of two ended SIP calls, one with each
> participant.  Each participant thinks it is talking to a UA.
> While some messages sent from a participant to a UA might be
> consumed by the UA (state changes), often, a message will trigger
> messages from the focus to other participants.
> 
> This seems, to me, to meet the definition of B2BUA in 3261,
> and more strongly meets proposed changes in the definition.
> Of course, it really doesn't matter if you consider the focus to
> be a B2BUA or not.
> 
> Brian 
> _______________________________________________
> Sip mailing list  https://www1.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: "Drage, Keith (Keith)" <drage@lucent.com>
To: aki.niemi@nokia.com, Brian.Rosen@marconi.com
Cc: sip@ietf.org
Subject: RE: [Sip] B2BUA
Date: Mon, 9 Dec 2002 15:41:47 -0000 
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I would assume there are some instances where we want transparency of SIP contents and bodies between UAs "connected" to a focus, and therefore defining it as a B2BUA with some requirements on transparency, is actually better than saying there is no relationship between the UAs that make up the focus.

Keith

Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com 

> -----Original Message-----
> From: aki.niemi@nokia.com [mailto:aki.niemi@nokia.com]
> Sent: 03 December 2002 18:00
> To: Brian.Rosen@marconi.com
> Cc: sip@ietf.org
> Subject: RE: [Sip] B2BUA
> 
> 
> Brian,
> 
> inline.
> 
> > Of course these things are never so simple.
> > On an "end system mixed" conference, an INVITE
> > from a new participant triggers Re-INVITES to
> > existing participants.  Depending on details,
> > a focus could hold the response from the INVITE
> > of the new participant until it receives responses
> > from all existing participants (new CODEC negotiation
> > for example).  That meets your definition of B2BUA.
> > 
> > On the other hand, a centrally mixed conference
> > would not normally trigger Re-INVITES.  Does that
> > mean that an end-system mixed focus is a B2BUA
> > but a centrally mixed focus is not?  Hard to
> > differentiate.  With both of them, a state
> > change, signaled by, say a NOTIFY from one to
> > the focus triggers NOTIFIES to (possibly a
> > subset of) the other participants, so that
> > would meet the definition.  Would you call
> > that a change in roles?
> 
> I agree, a conference server could assume both UA and B2BUA 
> roles, as well as proxy and redirect server roles depending 
> on the system, application, request, and so on.  But whether 
> a focus is a UA, a B2BUA, or both is somewhat besides the 
> point.  The roles need to be defined so that B2BUA behavior 
> is distinct from UA behavior.  The end-point can then use one 
> that best fits its purpose.
> 
> Cheers,
> Aki
> 
> > 
> > Brian
> > 
> > > -----Original Message-----
> > > From: aki.niemi@nokia.com [mailto:aki.niemi@nokia.com]
> > > Sent: Tuesday, December 03, 2002 12:13 PM
> > > To: Brian.Rosen@marconi.com
> > > Cc: sip@ietf.org
> > > Subject: RE: [Sip] B2BUA
> > > 
> > > 
> > > Inline.
> > > 
> > > > The key thing about a B2BUA is that it looks like a UA 
> to whatever
> > > > it is connected to (on both sides).  So, a conference server
> > > > looks like a UA to each of it's ports, and it has at 
> least two of
> > > > them, and there is some "magic" between them.  If you 
> look at the
> > > > concept of the "focus" from the conferencing work, it is a
> > > > thing that has a bunch of two ended SIP calls, one with each
> > > > participant.  Each participant thinks it is talking to a UA.
> > > > While some messages sent from a participant to a UA might be
> > > > consumed by the UA (state changes), often, a message 
> will trigger
> > > > messages from the focus to other participants.
> > > 
> > > This just generalized the B2BUA to cover any normal UA 
> > > behavior. Example, I get a MESSAGE from a friend to my UA, 
> > > based on which decide to call another friend by sending an 
> > > INVITE from that UA. This I think is quite normal UA 
> > > operation, the "magic" being me via my UI.
> > > 
> > > > This seems, to me, to meet the definition of B2BUA in 3261,
> > > > and more strongly meets proposed changes in the definition.
> > > > Of course, it really doesn't matter if you consider the focus to
> > > > be a B2BUA or not.
> > > 
> > > Hmm... Let's think of the B2BUA as 'in-UA' concatenated with 
> > > 'out-UA'. By definition:
> > > 
> > > 1) in-UAS receives request,
> > >    <Magic happens>
> > > 2) out-UAC sends request'
> > > 3) out-UAC receives response'
> > >    <Magic happens>
> > > 4) in-UAS sends response
> > > 
> > > (The same process happens with requests going from 'out' to 'in')
> > > 
> > > Within this B2BUA, 'in-UAS' shares magic with 'out-UAC', and 
> > > 'in-UAC' shares magic with 'out-UAS'. However, if (as you 
> > > state above), 4) follows immediately after 1), it's no longer 
> > > a B2BUA, but a plain old UA, IMHO. 
> > > 
> > > Cheers,
> > > Aki
> > > 
> > 
> _______________________________________________
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: Mary Barnes <mbarnes@nortelnetworks.com>,
        "'Jonathan Rosenberg'"
	 <jdrosen@dynamicsoft.com>,
        Pete Cordell <pete@tech-know-ware.com>
Cc: "David R. Oran" <oran@cisco.com>,
        Dean Willis
	 <dean.willis@softarmor.com>, sip@ietf.org
Subject: RE: [Sip] B2BUA & Max-Forwards
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This thread is definitely going way beyond what it should do (both in scope and duration) - it did however start (about a 100 messages ago) with some reasonable modifications to the B2BUA definition, and save those relatively minor modifications, I do not want to see any other modifications in RFC 3261 or its next revision.

I certainly will not agree to removal of any existing definition of B2BUA.

Keith

Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com 

> -----Original Message-----
> From: Mary Barnes [mailto:mbarnes@nortelnetworks.com]
> Sent: 06 December 2002 15:40
> To: 'Jonathan Rosenberg'; Pete Cordell
> Cc: David R. Oran; Dean Willis; sip@ietf.org
> Subject: RE: [Sip] B2BUA & Max-Forwards
> 
> 
> 
> I agree with Jonathan.  As mentioned earlier this week 
> apparently, there is
> already work underway by some individuals to write an 
> informational draft
> documenting BBUA scenarios/call flows. 
> 
> I'll keep my fingers crossed that the existence of such a 
> document can save
> us from another round of B2BUA discussions again in 3 months. 
> 
> Regards,
> Mary.
> 
> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Friday, December 06, 2002 12:59 AM
> To: Pete Cordell
> Cc: David R. Oran; Dean Willis; sip@ietf.org
> Subject: Re: [Sip] B2BUA & Max-Forwards
> 
> 
> 
> 
> Pete Cordell wrote:
> > Dave,
> > 
> > To save your disagreeableness, can we say something like:
> > 
> > "A B2BUA MUST perform loop prevention.  One way to do this 
> is to propagate
> > the Max-Forwards header from the incoming request to the 
> outgoing request
> in
> > the same way that a proxy does."
> 
> A statement like this is completely out of scope for rfc3261. 
> It might 
> be a reasonable statement to make in some kind of application 
> document 
> that describes a particular type of b2bua. However, it makes no more 
> sense to say this in RFC 3261 as it would be to discuss the ways in 
> which to build a focus or a 3pcc controller (both of which are also 
> informational, by the way).
> 
> I can't resist pointing out that much of this is deja vu. Folks may 
> recall heated discussions in Adelaide, where there was a bof 
> (I think) 
> that wanted to have a working group to specify SIP to H.323 
> conversion. 
> THe same arguemtns that were made there are being made here; its an 
> application level issue, there are many ways to do it, don't stifle 
> creativity, etc. In fact I recall that Dave was, as always, quite 
> vociferous on these points.
> 
> The eventual conclusion in the h323/sip conversion case was 
> to write an 
> INFORMATIONAL RFC that documents reasonable practices in 
> building such a 
> device. I believe the same logic and conclusions apply here. If folks 
> have a particular type of b2bua application they are fond of, please 
> feel free to write up your implementation advice and I am 
> sure sipping 
> would consider whether or not it would make a useful 
> informational RFC.
> 
> However, it is completely inappropriate to ever make such application 
> logic normative, and even more inappropraite for such wording 
> to be in 
> rfc3261.
> 
> Anyway, I too am tired of this thread. I think, in fact, there is 
> "rough" consensus to remove the definition of b2bua entirely from 
> rfc3261 in its next revision (which is quite some time in the future 
> anyway). However, I'll let the chairs decide that.
> 
> -Jonathan R.
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
> _______________________________________________
> Sip mailing list  https://www1.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://www1.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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        "David R. Oran" <oran@cisco.com>,
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Drage, Keith (Keith) wrote:
 > This thread is definitely going way beyond what it should do (both in
 > scope and duration) - it did however start (about a 100 messages ago)
 > with some reasonable modifications to the B2BUA definition, and save
 > those relatively minor modifications, I do not want to see any other
 > modifications in RFC 3261 or its next revision.
 >
 > I certainly will not agree to removal of any existing definition of
 > B2BUA.

Well, we have some time to conclude this (since we are not revving 
rfc3261 for some time), but it was my sense that there was ROUGH 
consensus to remove it.

Rough doesn't mean unanimous of course...

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From mailnull@www1.ietf.org  Mon Dec  9 13:40:44 2002
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Date: Mon, 09 Dec 2002 13:37:52 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Mark Watson <mwatson@nortelnetworks.com>
CC: "'Rohan Mahy'" <rohan@cisco.com>,
        "Mahey, Sonit" <Sonit.Mahey@icn.siemens.com>,
        Pete Cordell <pete@tech-know-ware.com>,
        Christian Huitema <huitema@windows.microsoft.com>, sip@ietf.org
Subject: Re: [Sip] B2BUA - Security
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inline.

Mark Watson wrote:
> Rohan wrote:
> 
>  > In addition to Jonathan's approach to session policy, there is no
>  > reason a middlebox controller can't be a proxy if it:
>  >
>  > 1. opens pinholes referenced in a clear text or integrity
>  > protected SDP
>  > 2. rejects requests with no readable SDP (the UA could
>  > instead include
>  > a full SDP, with keying material, encrypted with SMIME, ****and an
>  > abbreviated copy which is only integrity protected****).
> 
> 
> I was about to write a draft suggesting exactly this.

THis is simply not sufficient.

It will suffice for firewalls, but not for NAT, where an actual 
modification needs to occur.

I think that, per Pete's suggestion, the correct path forward is a 
requirements document that describes the problem and the requirements 
for the solution. I believe there are many interesting requirements here 
about disclosure of what is being done, that will strongly impact the 
solution.

> 
> As per Jonathan's draft there is a need for the UA to be able to 
> 'declare' to network entities those parts of the SDP information which 
> are needed for the network entity to perform whatever service it 
> performs (e.g. NAT or Firewall control). But the UA should not have to 
> declare the whole SDP, and the terminating UA should be able to tell if 
> a network entity has modified these parameters.

Sounds like we are beginning the requirements work here...

> 
> Rather than putting this information into new SIP headers, we could 
> include two copies of the SDP. One copy is unprotected and contains only 
> the information that the UE wishes to 'declare' to the network entities. 
> The other copy is protected and contains the full information.

I think we can leave the header/body debate for down the road. I would 
be happy if we can get agreement on this general problem as one that we 
need to solve.

> 
> This approach would work through existing devices which inspect/modify SDP.
> 
> I'm not sure how an existing S/MIME compliant client would behave on 
> receiving these two copies. If it uses the unprotected SDP, then things 
> are no worse than the present situation - the call suceeds, but without 
> the advantages of S/MIME. If it uses the protected SDP (which has not 
> been modified by the NAT controlling device), then the call fails, but 
> this is still no worse than the present situation.

It comes down to disclosure. The message needs to convey information 
about the meaning of the multiple session descriptions - who set which, 
and why.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From mailnull@www1.ietf.org  Mon Dec  9 13:43:35 2002
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From: "Mahey, Sonit" <Sonit.Mahey@icn.siemens.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        Mark Watson
	 <mwatson@nortelnetworks.com>
Cc: "'Rohan Mahy'" <rohan@cisco.com>, Pete Cordell <pete@tech-know-ware.com>,
        Christian Huitema <huitema@windows.microsoft.com>, sip@ietf.org
Subject: RE: [Sip] B2BUA - Security
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Folks,

I agree with Jonathan. We need some definition around the work.

- sonit

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Monday, December 09, 2002 1:38 PM
> To: Mark Watson
> Cc: 'Rohan Mahy'; Mahey, Sonit; Pete Cordell; Christian Huitema;
> sip@ietf.org
> Subject: Re: [Sip] B2BUA - Security
> 
> 
> inline.
> 
> Mark Watson wrote:
> > Rohan wrote:
> > 
> >  > In addition to Jonathan's approach to session policy, there is no
> >  > reason a middlebox controller can't be a proxy if it:
> >  >
> >  > 1. opens pinholes referenced in a clear text or integrity
> >  > protected SDP
> >  > 2. rejects requests with no readable SDP (the UA could
> >  > instead include
> >  > a full SDP, with keying material, encrypted with SMIME, 
> ****and an
> >  > abbreviated copy which is only integrity protected****).
> > 
> > 
> > I was about to write a draft suggesting exactly this.
> 
> THis is simply not sufficient.
> 
> It will suffice for firewalls, but not for NAT, where an actual 
> modification needs to occur.
> 
> I think that, per Pete's suggestion, the correct path forward is a 
> requirements document that describes the problem and the requirements 
> for the solution. I believe there are many interesting 
> requirements here 
> about disclosure of what is being done, that will strongly impact the 
> solution.
> 
> > 
> > As per Jonathan's draft there is a need for the UA to be able to 
> > 'declare' to network entities those parts of the SDP 
> information which 
> > are needed for the network entity to perform whatever service it 
> > performs (e.g. NAT or Firewall control). But the UA should 
> not have to 
> > declare the whole SDP, and the terminating UA should be 
> able to tell if 
> > a network entity has modified these parameters.
> 
> Sounds like we are beginning the requirements work here...
> 
> > 
> > Rather than putting this information into new SIP headers, we could 
> > include two copies of the SDP. One copy is unprotected and 
> contains only 
> > the information that the UE wishes to 'declare' to the 
> network entities. 
> > The other copy is protected and contains the full information.
> 
> I think we can leave the header/body debate for down the 
> road. I would 
> be happy if we can get agreement on this general problem as 
> one that we 
> need to solve.
> 
> > 
> > This approach would work through existing devices which 
> inspect/modify SDP.
> > 
> > I'm not sure how an existing S/MIME compliant client would 
> behave on 
> > receiving these two copies. If it uses the unprotected SDP, 
> then things 
> > are no worse than the present situation - the call suceeds, 
> but without 
> > the advantages of S/MIME. If it uses the protected SDP 
> (which has not 
> > been modified by the NAT controlling device), then the call 
> fails, but 
> > this is still no worse than the present situation.
> 
> It comes down to disclosure. The message needs to convey information 
> about the meaning of the multiple session descriptions - who 
> set which, 
> and why.
> 
> -Jonathan R.
> -- 
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
_______________________________________________
Sip mailing list  https://www1.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 mailnull@www1.ietf.org  Mon Dec  9 15:00:55 2002
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From: Adam Roach <adam@dynamicsoft.com>
To: "'hisham.khartabil@nokia.com'" <hisham.khartabil@nokia.com>,
        kokjeng.loh@siemens.com, sip@ietf.org
Subject: RE: [Sip] Handling of REFER/NOTIFY by Stateful Proxy
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Hisham Khartabil wrote:

> You are talking about Dialog Stateful proxy. In this case, 
> the dialog at the proxy is not removed until the NOTIFY of a 
> REFER terminates the dialog (or an unSUBSCRIBE).

The issue, I beleive, is that the proxy in question doesn't
necessarily know anything about the behavior associated with
SUBSCRIBE and NOTIFY. So, from its perspective, the dialog
*is* terminated with the BYE.

So, that brings us back to Loh Kok Jeng's question:

> What should a stateful proxy do when it receives
> a NOTIFY whose dialog has been terminated?
> ...[S]hould the stateful proxy rejects NOTIFY 
> with 481?  Or should it proxy NOTIFY onwards?

As Hisham points out, your question appears to apply
to a *dialog* stateful proxy, which is a rather different
beast than a stateful proxy.

The simple answer is: it should proxy it on.

However, this behavior should naturally fall out of good
design on the part of the proxy. Ask yourself: in general,
if your proxy receives a request with a "Route" header for
which you have no dialog, what is the correct behavior? 

Well, if you know *how* to proxy it, you *should* proxy it.
In this case, the presence of a "Route" header gives you
sufficient information about where to send it. (This is one
of the important SIP protocol behaviors, for example,
behind the "Service Route Discovery" draft).

/a
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From mailnull@www1.ietf.org  Mon Dec  9 16:01:03 2002
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From: "Jitender Arora" <jitender_arora@lycos.com>
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Hi ,

       Can we use REFER call transfer mechanism to , transfer the call from one PSTN subscriber to the other PSTN subscriber ( in the Net-2-Phone calls ).......

       I am talking of the Net-2-Phone Scenario ( say one guy sitting on PC, Using SIP Phone ) , is making a call to PSTN Phone .

       So we will be using the SIP on one side and ISUP ( just for example) on the PSTN side .........
 
       So will the following call flow valid , and can be used as a REFER call scenario :

       SIP PHONE/Proxy                  Switch                          PSTN
       INVITE F1 
        -------------------------------->
                                            IAM S1
                                           -------------------------------> ( PSTN Phone 1)
                 100 Trying F2
        <---------------------------------                                 
                                            ACM S2
                                           <-------------------------------- ( PSTN Phone 1)    
                 183 F3
         <--------------------------------
                                           ANM S3
                                           <-------------------------------- ( PSTN Phone 1)
                 200 O.K  F4       
         <-------------------------------- 

                  ACK F5
         --------------------------------->
                  INVITE ( Hold )F6
         ---------------------------------->
                   200 O.K F7
         <----------------------------------
                   ACK F8
          --------------------------------->
                REFER ( New PSTN URI )F9
          ---------------------------------->
                  202 ACCEPTED F10
          <---------------------------------
                                              IAM S4
                                            ------------------------------------> (PSTN Phone 2)
                                              ACM S5
                                            <------------------------------------  ( PSTN Phone2 )
                                              ANM S6
                                             <-----------------------------------  ( PSTN Phone 2)

                   NOTIFY F11
           <---------------------------------
                    200 OK  F12
           ---------------------------------->
                                              REL for first PSTN Subscriber S7
                                             --------------------------------------->(PSTN Phone 1)
                                              REL Confirm  S8
                                             <----------------------------------------- (PSTN Phone 1)


         Plese let me know if it is valid call flow ......

Regards,
jitender .
            


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Subject: Re: [Sip] Buggy example in draft-ietf-sipping-reg-event-00
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Yes, this is a bug in the example. Thanks for pointing it out.

-Jonathan R.

Peter PÃ¤ppinghaus wrote:
> Hi,
> 
> I believe to have stumbled on a buggy example in 
> draft-ietf-sipping-reg-event-00.txt.
> 
> Quotation from top of p. 19:
> 
>   "Later on, the user registers (5):
> 
>    REGISTER sip:joe@bar.com SIP/2.0
>    Via: SIP/2.0/UDP pc34.bar.com;branch=z9hG4bKnaaff
>    From: sip:joe@bar.com;tag=99a8s
>    To: sip:joe@bar.com
>    Call-ID: 88askjda9@pc34.bar.com
>    CSeq: 9976 REGISTER
>    Contact: sip:joe@pc34.bar.com"
> 
>    The Request-URI in this example violates RFC 3261, sec. 10.2:
>    "The userinfo and "@" components of the SIP URI MUST NOT be present."
> 
> Regards,
> 
> Peter Paeppinghaus
> 

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Subject: Re: [Sip] B2BUA - Security
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On Monday, December 9, 2002, at 10:37 AM, Jonathan Rosenberg wrote:

> inline.
>
> Mark Watson wrote:
>> Rohan wrote:
>>  > In addition to Jonathan's approach to session policy, there is no
>>  > reason a middlebox controller can't be a proxy if it:
>>  >
>>  > 1. opens pinholes referenced in a clear text or integrity
>>  > protected SDP
>>  > 2. rejects requests with no readable SDP (the UA could
>>  > instead include
>>  > a full SDP, with keying material, encrypted with SMIME, ****and an
>>  > abbreviated copy which is only integrity protected****).
>> I was about to write a draft suggesting exactly this.
>
> THis is simply not sufficient.
>
> It will suffice for firewalls, but not for NAT, where an actual 
> modification needs to occur.

The midbox controller could propose a new SDP with public 
addresses/ports, which the UAC could actually sign.  I believe this 
will work, and requires minimal changes to SIP (at most a new response 
code).

Also if we separate the NAT problem into address/port discovery, and 
refresh, the UAC could use STUN to discover addresses, and just send a 
copy of the SDP with these recently discovered public addresses to the 
midbox controller, which just needs to *a* copy of part of the SDP to 
keep ports open.  It can even use session timer to avoid keeping state.

thanks,
-rohan

> I think that, per Pete's suggestion, the correct path forward is a 
> requirements document that describes the problem and the requirements 
> for the solution. I believe there are many interesting requirements 
> here about disclosure of what is being done, that will strongly impact 
> the solution.
>
>> As per Jonathan's draft there is a need for the UA to be able to 
>> 'declare' to network entities those parts of the SDP information 
>> which are needed for the network entity to perform whatever service 
>> it performs (e.g. NAT or Firewall control). But the UA should not 
>> have to declare the whole SDP, and the terminating UA should be able 
>> to tell if a network entity has modified these parameters.
>
> Sounds like we are beginning the requirements work here...
>
>> Rather than putting this information into new SIP headers, we could 
>> include two copies of the SDP. One copy is unprotected and contains 
>> only the information that the UE wishes to 'declare' to the network 
>> entities. The other copy is protected and contains the full 
>> information.
>
> I think we can leave the header/body debate for down the road. I would 
> be happy if we can get agreement on this general problem as one that 
> we need to solve.
>
>> This approach would work through existing devices which 
>> inspect/modify SDP.
>> I'm not sure how an existing S/MIME compliant client would behave on 
>> receiving these two copies. If it uses the unprotected SDP, then 
>> things are no worse than the present situation - the call suceeds, 
>> but without the advantages of S/MIME. If it uses the protected SDP 
>> (which has not been modified by the NAT controlling device), then the 
>> call fails, but this is still no worse than the present situation.
>
> It comes down to disclosure. The message needs to convey information 
> about the meaning of the multiple session descriptions - who set 
> which, and why.
>
> -Jonathan R.
> -- 
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>

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	 <padma_suresh@hotmail.com>, vkg@lucent.com,
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Subject: RE: [Sip] B2BUA & Max-Forwards
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This discussion raises one more doubt for me, pls. clarify.

UA1------->BtoBUA---------->UA2

Is the BtoBUA a SIP endpoint from UA1's point of view? In other
words, are the the "Require:" etc end-to-end headers sent by UA1,
meant for B2BUA or UA2?

Is the UAS part in in BtoBUA is bound by the definition and rules
in 3261?

thanks,
= Kedar =

> -----Original Message-----
> From: Adam Roach [mailto:adam@dynamicsoft.com]
> Sent: Tuesday, December 03, 2002 6:37 PM
> To: 'Padma Suresh'; vkg@lucent.com; pete@tech-know-ware.com
> Cc: sip@ietf.org
> Subject: RE: [Sip] B2BUA & Max-Forwards
> 
> 
> -----Original Message-----
> From: Padma Suresh [mailto:padma_suresh@hotmail.com]
>  
> > A B2BUA needs to propagate some of the headers received in a
> > request/response to the other endpoint. Example: Require...
> 
> No. As emphatically as possible: NO. This is completely wrong,
> and it appears to be a growing -- and dangerous -- misconception.
> 
> A B2BUA cannot transparently pass Require through. It must
> understand and invoke the extension indicated.
> 
>   "The Require header field contains a list of option tags, 
> described in
>    Section 19.2.  Each option tag defines a SIP extension that MUST be
>    understood to process the request."
> 
> A back to back user agent *is* a UAS, and *is* bound by this
> provision.
> 
> Whether the UAC portion then turns around and imposes the same
> requirement on the UAS it contacts is a local decision.
> 
> /a
 

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<TITLE>RE: [Sip] B2BUA &amp; Max-Forwards</TITLE>
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<P><FONT SIZE=2>This discussion raises one more doubt for me, pls. clarify.</FONT>
</P>

<P><FONT SIZE=2>UA1-------&gt;BtoBUA----------&gt;UA2</FONT>
</P>

<P><FONT SIZE=2>Is the BtoBUA a SIP endpoint from UA1's point of view? In other</FONT>
<BR><FONT SIZE=2>words, are the the &quot;Require:&quot; etc end-to-end headers sent by UA1,</FONT>
<BR><FONT SIZE=2>meant for B2BUA or UA2?</FONT>
</P>

<P><FONT SIZE=2>Is the UAS part in in BtoBUA is bound by the definition and rules</FONT>
<BR><FONT SIZE=2>in 3261?</FONT>
</P>

<P><FONT SIZE=2>thanks,</FONT>
<BR><FONT SIZE=2>= Kedar =</FONT>
</P>

<P><FONT SIZE=2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=2>&gt; From: Adam Roach [<A HREF="mailto:adam@dynamicsoft.com">mailto:adam@dynamicsoft.com</A>]</FONT>
<BR><FONT SIZE=2>&gt; Sent: Tuesday, December 03, 2002 6:37 PM</FONT>
<BR><FONT SIZE=2>&gt; To: 'Padma Suresh'; vkg@lucent.com; pete@tech-know-ware.com</FONT>
<BR><FONT SIZE=2>&gt; Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=2>&gt; Subject: RE: [Sip] B2BUA &amp; Max-Forwards</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=2>&gt; From: Padma Suresh [<A HREF="mailto:padma_suresh@hotmail.com">mailto:padma_suresh@hotmail.com</A>]</FONT>
<BR><FONT SIZE=2>&gt;&nbsp; </FONT>
<BR><FONT SIZE=2>&gt; &gt; A B2BUA needs to propagate some of the headers received in a</FONT>
<BR><FONT SIZE=2>&gt; &gt; request/response to the other endpoint. Example: Require...</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; No. As emphatically as possible: NO. This is completely wrong,</FONT>
<BR><FONT SIZE=2>&gt; and it appears to be a growing -- and dangerous -- misconception.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; A B2BUA cannot transparently pass Require through. It must</FONT>
<BR><FONT SIZE=2>&gt; understand and invoke the extension indicated.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt;&nbsp;&nbsp; &quot;The Require header field contains a list of option tags, </FONT>
<BR><FONT SIZE=2>&gt; described in</FONT>
<BR><FONT SIZE=2>&gt;&nbsp;&nbsp;&nbsp; Section 19.2.&nbsp; Each option tag defines a SIP extension that MUST be</FONT>
<BR><FONT SIZE=2>&gt;&nbsp;&nbsp;&nbsp; understood to process the request.&quot;</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; A back to back user agent *is* a UAS, and *is* bound by this</FONT>
<BR><FONT SIZE=2>&gt; provision.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Whether the UAC portion then turns around and imposes the same</FONT>
<BR><FONT SIZE=2>&gt; requirement on the UAS it contacts is a local decision.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; /a</FONT>
<BR><FONT SIZE=2>&nbsp;</FONT>
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To: "'Roy, Radhika R, ALASO'" <rrroy@att.com>
Cc: sip@ietf.org, "'Adam Roach'" <adam@dynamicsoft.com>
Subject: RE: [Sip] B2BUA & Max-Forwards
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> -----Original Message-----
> From: Adam Roach [mailto:adam@dynamicsoft.com]
> Sent: Wednesday, December 04, 2002 4:58 PM
> To: 'Roy, Radhika R, ALASO'; Adam Roach
> Cc: sip@ietf.org
> Subject: RE: [Sip] B2BUA & Max-Forwards
> 
> 
> > UAC(a)--->UAS(b)===[application logic here]===UAC(b)--->UAS(c)
> > 
> > [RRR] I like to know who has been the called party when 
> > UAC(a)) sent the INVITE message? Has it been UAS(b) or UAS(c)?
> 
> I'm going to first need to clarify your question.
> 
> The question I will answer is: "If an INVITE is sent through the
> network described in this diagram, what is the value of the
> Request-URI when UAC(a) sends it?"
> 
> The answer is: it depends. In general, yes, it will probably be
> either of UAS(b) or UAS(c).
> 
> > [RRR] If the answer is UAS(c), the question is: Why the call 
> > has been routed to UAS(b) by UAC(a) in the first place?
> 
> The most obvious of several plausible answers is: UAC(a)
> has UAS(b) configured as a local outbound proxy.

Or most likely, when the INVITE request arrived at some outbound
proxy of UAC(a), it decided to route the request to UAS(b) as
a local polocy.

This is possible irrespective of Request-URI and To: field.

My 2-cents.

thanks,
= Kedar =

> 
> > [RRR] If the answer is UAS(b), probably UAC(a) does not care.
> 
> About what?
> 
> /a
> _______________________________________________
> Sip mailing list  https://www1.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
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<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Adam Roach [<A =
HREF=3D"mailto:adam@dynamicsoft.com">mailto:adam@dynamicsoft.com</A>]</F=
ONT>
<BR><FONT SIZE=3D2>&gt; Sent: Wednesday, December 04, 2002 4:58 =
PM</FONT>
<BR><FONT SIZE=3D2>&gt; To: 'Roy, Radhika R, ALASO'; Adam Roach</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: [Sip] B2BUA &amp; =
Max-Forwards</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; UAC(a)---&gt;UAS(b)=3D=3D=3D[application =
logic here]=3D=3D=3DUAC(b)---&gt;UAS(c)</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; [RRR] I like to know who has been the =
called party when </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; UAC(a)) sent the INVITE message? Has it =
been UAS(b) or UAS(c)?</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I'm going to first need to clarify your =
question.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; The question I will answer is: &quot;If an =
INVITE is sent through the</FONT>
<BR><FONT SIZE=3D2>&gt; network described in this diagram, what is the =
value of the</FONT>
<BR><FONT SIZE=3D2>&gt; Request-URI when UAC(a) sends it?&quot;</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; The answer is: it depends. In general, yes, it =
will probably be</FONT>
<BR><FONT SIZE=3D2>&gt; either of UAS(b) or UAS(c).</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; [RRR] If the answer is UAS(c), the =
question is: Why the call </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; has been routed to UAS(b) by UAC(a) in the =
first place?</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; The most obvious of several plausible answers =
is: UAC(a)</FONT>
<BR><FONT SIZE=3D2>&gt; has UAS(b) configured as a local outbound =
proxy.</FONT>
</P>

<P><FONT SIZE=3D2>Or most likely, when the INVITE request arrived at =
some outbound</FONT>
<BR><FONT SIZE=3D2>proxy of UAC(a), it decided to route the request to =
UAS(b) as</FONT>
<BR><FONT SIZE=3D2>a local polocy.</FONT>
</P>

<P><FONT SIZE=3D2>This is possible irrespective of Request-URI and To: =
field.</FONT>
</P>

<P><FONT SIZE=3D2>My 2-cents.</FONT>
</P>

<P><FONT SIZE=3D2>thanks,</FONT>
<BR><FONT SIZE=3D2>=3D Kedar =3D</FONT>
</P>

<P><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; [RRR] If the answer is UAS(b), probably =
UAC(a) does not care.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; About what?</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; /a</FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the core =
SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; Use sipping@ietf.org for new developments on =
the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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Cc: "David R. Oran" <oran@cisco.com>,
        Dean Willis
	 <dean.willis@softarmor.com>, sip@ietf.org
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Interesting.

What if a BtoBUA(UAS side) received a request with Max-forward=0?
A proxy would have responded with 483. I believe a B2BUA may not
do the same.

thanks,
= Kedar =


> -----Original Message-----
> From: Pete Cordell [mailto:pete@tech-know-ware.com]
> Sent: Thursday, December 05, 2002 11:24 AM
> To: David R. Oran; Dean Willis; sip@ietf.org
> Subject: Re: [Sip] B2BUA & Max-Forwards
> 
> 
> Dave,
> 
> To save your disagreeableness, can we say something like:
> 
> "A B2BUA MUST perform loop prevention.  One way to do this is 
> to propagate
> the Max-Forwards header from the incoming request to the 
> outgoing request in
> the same way that a proxy does."
> 
> It's the loops that I'm worried about, not the Max-Forwards per se.
> 
> This addresses my concern, and I think James' and possibly yours.
> 
> Pete.
> 
> 

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<P><FONT SIZE=2>Interesting.</FONT>
</P>

<P><FONT SIZE=2>What if a BtoBUA(UAS side) received a request with Max-forward=0?</FONT>
<BR><FONT SIZE=2>A proxy would have responded with 483. I believe a B2BUA may not</FONT>
<BR><FONT SIZE=2>do the same.</FONT>
</P>

<P><FONT SIZE=2>thanks,</FONT>
<BR><FONT SIZE=2>= Kedar =</FONT>
</P>
<BR>

<P><FONT SIZE=2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=2>&gt; From: Pete Cordell [<A HREF="mailto:pete@tech-know-ware.com">mailto:pete@tech-know-ware.com</A>]</FONT>
<BR><FONT SIZE=2>&gt; Sent: Thursday, December 05, 2002 11:24 AM</FONT>
<BR><FONT SIZE=2>&gt; To: David R. Oran; Dean Willis; sip@ietf.org</FONT>
<BR><FONT SIZE=2>&gt; Subject: Re: [Sip] B2BUA &amp; Max-Forwards</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Dave,</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; To save your disagreeableness, can we say something like:</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; &quot;A B2BUA MUST perform loop prevention.&nbsp; One way to do this is </FONT>
<BR><FONT SIZE=2>&gt; to propagate</FONT>
<BR><FONT SIZE=2>&gt; the Max-Forwards header from the incoming request to the </FONT>
<BR><FONT SIZE=2>&gt; outgoing request in</FONT>
<BR><FONT SIZE=2>&gt; the same way that a proxy does.&quot;</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; It's the loops that I'm worried about, not the Max-Forwards per se.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; This addresses my concern, and I think James' and possibly yours.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Pete.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
</P>

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To: "'Jiri Kuthan'" <jiri@iptel.org>,
        Phil Reynolds
	 <preynolds@ridgewaysystems.com>
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BtoBUA need not always be call stateful.
e.g. a SIP REGISTER request reaches a BtoBUA,
it sends a new request to appropriate Registrar
based on some local application specific
configuration.

There isn't any network state here (except the
transaction state, like sip proxy).

Pls. let me know, if you foresee any problem here.

thanks,
= Kedar =


> -----Original Message-----
> From: Jiri Kuthan [mailto:jiri@iptel.org]
> Sent: Wednesday, December 04, 2002 8:16 AM
> To: Phil Reynolds; sip@ietf.org
> Subject: RE: [Sip] B2BUA
> 
> 
> I think a better strategy to design reliable IP networks is not to
> have stateful elements in the middle. It is sometimes not
> easy to achieve, but you don't need to solve the problem what
> happens when they fail. 
> 
> Besides that, all sort of middleboxes (B2BUA can IMHO be labeled
> as app-layer middlebox) have e2e security issues as pointed out
> by Christian, and are not free of scalability issues.
> 
> I suggest you read RFC3234 -- it covers various issues middleboxes
> have, including failure handling.
> 
> -Jiri
> 

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<P><FONT SIZE=2>BtoBUA need not always be call stateful.</FONT>
<BR><FONT SIZE=2>e.g. a SIP REGISTER request reaches a BtoBUA,</FONT>
<BR><FONT SIZE=2>it sends a new request to appropriate Registrar</FONT>
<BR><FONT SIZE=2>based on some local application specific</FONT>
<BR><FONT SIZE=2>configuration.</FONT>
</P>

<P><FONT SIZE=2>There isn't any network state here (except the</FONT>
<BR><FONT SIZE=2>transaction state, like sip proxy).</FONT>
</P>

<P><FONT SIZE=2>Pls. let me know, if you foresee any problem here.</FONT>
</P>

<P><FONT SIZE=2>thanks,</FONT>
<BR><FONT SIZE=2>= Kedar =</FONT>
</P>
<BR>

<P><FONT SIZE=2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=2>&gt; From: Jiri Kuthan [<A HREF="mailto:jiri@iptel.org">mailto:jiri@iptel.org</A>]</FONT>
<BR><FONT SIZE=2>&gt; Sent: Wednesday, December 04, 2002 8:16 AM</FONT>
<BR><FONT SIZE=2>&gt; To: Phil Reynolds; sip@ietf.org</FONT>
<BR><FONT SIZE=2>&gt; Subject: RE: [Sip] B2BUA</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; I think a better strategy to design reliable IP networks is not to</FONT>
<BR><FONT SIZE=2>&gt; have stateful elements in the middle. It is sometimes not</FONT>
<BR><FONT SIZE=2>&gt; easy to achieve, but you don't need to solve the problem what</FONT>
<BR><FONT SIZE=2>&gt; happens when they fail. </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Besides that, all sort of middleboxes (B2BUA can IMHO be labeled</FONT>
<BR><FONT SIZE=2>&gt; as app-layer middlebox) have e2e security issues as pointed out</FONT>
<BR><FONT SIZE=2>&gt; by Christian, and are not free of scalability issues.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; I suggest you read RFC3234 -- it covers various issues middleboxes</FONT>
<BR><FONT SIZE=2>&gt; have, including failure handling.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; -Jiri</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
</P>

</BODY>
</HTML>
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Date: Tue, 10 Dec 2002 07:36:47 -0500
From: "David R. Oran" <oran@cisco.com>
To: Kedar Patil <kpatil@sta.samsung.com>
cc: sip@ietf.org
Subject: RE: [Sip] B2BUA & Max-Forwards
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--On Wednesday, December 04, 2002 10:54 AM -0600 Kedar Patil 
<kpatil@sta.samsung.com> wrote:

>
> This discussion raises one more doubt for me, pls. clarify.
>
> UA1------->BtoBUA---------->UA2
>
> Is the BtoBUA a SIP endpoint from UA1's point of view?
Yes. Definintely.

> In other
> words, are the the "Require:" etc end-to-end headers sent by UA1,
> meant for B2BUA or UA2?
>
It doesn't matter who they are "meant for". They MUST be processed and 
understood by B2BUA. Whatever UA2 sees is at the whim of the application on 
top of the UAS/UAC performing as the B2BUA.

> Is the UAS part in in BtoBUA is bound by the definition and rules
> in 3261?
>
Yes. Emphatically.

> thanks,
> = Kedar =
>
>> -----Original Message-----
>> From: Adam Roach [mailto:adam@dynamicsoft.com]
>> Sent: Tuesday, December 03, 2002 6:37 PM
>> To: 'Padma Suresh'; vkg@lucent.com; pete@tech-know-ware.com
>> Cc: sip@ietf.org
>> Subject: RE: [Sip] B2BUA & Max-Forwards
>>
>>
>> -----Original Message-----
>> From: Padma Suresh [mailto:padma_suresh@hotmail.com]
>>
>> > A B2BUA needs to propagate some of the headers received in a
>> > request/response to the other endpoint. Example: Require...
>>
>> No. As emphatically as possible: NO. This is completely wrong,
>> and it appears to be a growing -- and dangerous -- misconception.
>>
>> A B2BUA cannot transparently pass Require through. It must
>> understand and invoke the extension indicated.
>>
>>   "The Require header field contains a list of option tags,
>> described in
>>    Section 19.2.  Each option tag defines a SIP extension that MUST be
>>    understood to process the request."
>>
>> A back to back user agent *is* a UAS, and *is* bound by this
>> provision.
>>
>> Whether the UAC portion then turns around and imposes the same
>> requirement on the UAS it contacts is a local decision.
>>
>> /a
>

------------------------
David R. Oran
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720
Office: +1 978 264 2048
VoIP: +1 408 571 4576
Email: oran@cisco.com
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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From: "David R. Oran" <oran@cisco.com>
To: Kedar Patil <kpatil@sta.samsung.com>
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Subject: RE: [Sip] B2BUA
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--On Wednesday, December 04, 2002 11:14 AM -0600 Kedar Patil 
<kpatil@sta.samsung.com> wrote:

>
> BtoBUA need not always be call stateful.
Uh, on the contrary, since it has a UAS and a UAC, it most certainly has 
call (properly: dialog) state.

> e.g. a SIP REGISTER request reaches a BtoBUA,
> it sends a new request to appropriate Registrar
> based on some local application specific
> configuration.
>
> There isn't any network state here (except the
> transaction state, like sip proxy).
>
Well, that's because REGISTER doesn't establish a dialog.

> Pls. let me know, if you foresee any problem here.
>
Possibly, mostly having to do with security. As long as 3rd party REGISTERs 
work (i.e. somebody other than the contact being registered for the AOR 
doing the transaction), then this scenario should be no different. The 
operative condition though, is that the register comes from the B2BUA. From 
the point of view of the registrar it could just as we'll have been a 
mystical experience that caused the register to be sent by the B2BUA as 
some incoming register transaction.

> thanks,
> = Kedar =
>
>> -----Original Message-----
>> From: Jiri Kuthan [mailto:jiri@iptel.org]
>> Sent: Wednesday, December 04, 2002 8:16 AM
>> To: Phil Reynolds; sip@ietf.org
>> Subject: RE: [Sip] B2BUA
>>
>>
>> I think a better strategy to design reliable IP networks is not to
>> have stateful elements in the middle. It is sometimes not
>> easy to achieve, but you don't need to solve the problem what
>> happens when they fail.
>>
>> Besides that, all sort of middleboxes (B2BUA can IMHO be labeled
>> as app-layer middlebox) have e2e security issues as pointed out
>> by Christian, and are not free of scalability issues.
>>
>> I suggest you read RFC3234 -- it covers various issues middleboxes
>> have, including failure handling.
>>
>> -Jiri

------------------------
David R. Oran
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720
Office: +1 978 264 2048
VoIP: +1 408 571 4576
Email: oran@cisco.com
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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 mailnull@www1.ietf.org  Tue Dec 10 08:19:16 2002
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Dave Oran wrote:
> <kpatil@sta.samsung.com> wrote:
> >
> > BtoBUA need not always be call stateful.
> Uh, on the contrary, since it has a UAS and a UAC, it most certainly has 
> call (properly: dialog) state.

I think you're agruing details of semantics here.  If the B2BUA has 
nothing stored itself that it needs to clean up when the call/dialog/
transaction completes, I'd call is stateless.  However, it may have
used a mechanism like what was defined in draft-ietf-sip-state to
have that information stored somewhere else, or could have it
encoded in the Contact header of the outgoing request, or numerous
other valid hiding places.  End result is a "stateful" box that
is "stateless".  The term looses meaning.

Bill Marshall
wtm@research.att.com

-----original message-----
Date: Tue, 10 Dec 2002 07:44:12 -0500
From: "David R. Oran" <oran@cisco.com>
To: Kedar Patil <kpatil@sta.samsung.com>
Cc: sip@ietf.org
Subject: RE: [Sip] B2BUA

--On Wednesday, December 04, 2002 11:14 AM -0600 Kedar Patil 

<kpatil@sta.samsung.com> wrote:
>
> BtoBUA need not always be call stateful.
Uh, on the contrary, since it has a UAS and a UAC, it most certainly has 
call (properly: dialog) state.

> e.g. a SIP REGISTER request reaches a BtoBUA,
> it sends a new request to appropriate Registrar
> based on some local application specific
> configuration.
>
> There isn't any network state here (except the
> transaction state, like sip proxy).
>
Well, that's because REGISTER doesn't establish a dialog.

> Pls. let me know, if you foresee any problem here.
>
Possibly, mostly having to do with security. As long as 3rd party REGISTERs 
work (i.e. somebody other than the contact being registered for the AOR 
doing the transaction), then this scenario should be no different. The 
operative condition though, is that the register comes from the B2BUA. From 
the point of view of the registrar it could just as we'll have been a 
mystical experience that caused the register to be sent by the B2BUA as 
some incoming register transaction.

> thanks,
> = Kedar =
>
>> -----Original Message-----
>> From: Jiri Kuthan [mailto:jiri@iptel.org]
>> Sent: Wednesday, December 04, 2002 8:16 AM
>> To: Phil Reynolds; sip@ietf.org
>> Subject: RE: [Sip] B2BUA
>>
>>
>> I think a better strategy to design reliable IP networks is not to
>> have stateful elements in the middle. It is sometimes not
>> easy to achieve, but you don't need to solve the problem what
>> happens when they fail.
>>
>> Besides that, all sort of middleboxes (B2BUA can IMHO be labeled
>> as app-layer middlebox) have e2e security issues as pointed out
>> by Christian, and are not free of scalability issues.
>>
>> I suggest you read RFC3234 -- it covers various issues middleboxes
>> have, including failure handling.
>>
>> -Jiri

------------------------
David R. Oran
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720
Office: +1 978 264 2048
VoIP: +1 408 571 4576
Email: oran@cisco.com
_______________________________________________
Sip mailing list  https://www1.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 12th SIPit will be hosted by Hotsip in Stockholm
February 24-28, 2003.

Please see
http://www.hotsip.com/sipit12/
for details and registration.

Registration closes January 20.

RjS



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To: Robert Sparks <rsparks@dynamicsoft.com>
CC: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: Re: [Sip] ABNF issue
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Hi,

> Blindly doing a reverse token substitution will produce an incorrect
> parse. You can only apply it when you have proven the left side of
> the rule fits in a valid parse of the message.

[CHH] Correct. And, I "shift" a message-header when I have found one CRLF (the end of
the header). After that I'm done with that header. After that I continue parsing, and
if the previous message-header was the last one I expect to parse a second CRLF, to
indentify the end of the SIP header part of the message. So far I agree. What we
disagree on is the fact that you say that now the definition of LWS is not valid
anymore (even if we are still parsing the SIP part of the message), and that in this
case a CRLF is a CRLF - no matter what comes after it.

> So from your original example. LWS is not defined inside a body - the
> body is just a sequence of octets from the perspective of the SIP
> grammar. Your SIP parser _cannot_ apply a match against LWS inside it -
> there is no valid expansion of the SIP grammar that has LWS appearing
> after CRLF CRLF

[CHH] I don't think there is such thing as CRLF CRLF in the current SIP grammar (it
used to be before, though, I think). Again, the message-header is shifted once I found
one CRLF, so there will be no CRLF CRLF shift.

> >>The BNF specifies that message-header ends with a CRLF.
> >
> >Yes.
> >
> >>What follows that is one of two things - either another CRLF, marking the end
> >>of the headers, or another message header, which never begins with LWS.
> >
> >>From a SIP grammar point of view that is correct. However, I don't see it said
> >>anywhere that the definition of LWS isn't valid at the beginning of a new line.
>
> It _is_ valid after CRLF (which is what I think you mean by a new line)
> if (and only if) you are still in the expansion of message-header.

[CHH] That is what I am asking: where is it said that LWS is only valid within the
context of a message-header?

I do agree that LWS is not valid within the message-body part of the message, once we
get there, but I see it nowhere said that I can't parse a LWS AFTER I've shifted a
message-header, when I'm still in the SIP part of the message. No, the SIP grammar may
not allow it, and we may return a parser error, but that is a separate issue.

> >Also, just for the record, where is it said that a sip-header can't start with
> >LWS? Where is it said that line folding can't be used at the beginning of a line?
> >The text only says that the sip-header field values can be folded onto multiple
> >lines.
> I don't know what you mean by beginning of a line here. I assume you are
> asking why we cant have a header that looks like (\b for blank):
> \b\bNewHeader: foo which is a "\b\bNewheader"?
>
> The grammar answers that. extension-header starts with a token.

[CHH] Of course the LWS would not be part of the message-header grammar itself. LWS is
not even part of the message-header grammar within a header (ie the SIP grammar doesn't
require any LWS to be parsed). The grammar only says that LWS can be used
message-header parts for line folding, but it's not part of the grammar.

Let's take an example:

INVITE sip:12345@sip.com SIP/2.0 <CRLF_1>
Call-ID: 67890@sip.com <CRLF_2>
<CRLF_3>
<SP_1>CSeq: 100 INVITE <CRLF_4>

Now, the parser shifts the Call-ID header when it has found CRLF_2.

Then, it finds CRLF_3 and SP_1, and returns LWS. So, there is a line-folding before the
beginning of the CSeq header. Again, the LWS is NOT part of the CSeq header itself, but
my question is: where is it said that this LWS is NOT allowed at the beginning of a
line, before the actual message-header, but only WITHIN a message-header? Have I missed
it in the text?

> >>Thus, if a CRLF is followed by something OTHER than WSP, the next construction
> >>cannot be LWS.
> >
> >Maybe I missunderstand you, but the case I try to describe is when the CRLF IS
> >followed by WSP.
> If its in a valid expansion of message-header (which will not happen
> across a CRLFCRLF string), then you are looking at LWS.

[CHH] I may be wrong, but again I don't think there is such a thing as CRLFCRLF
(DOUBLE-CRLF) in the current SIP grammar.

Each CRLF is treated separately, and if there is another CRLF after a message-header
has been shifted we have found the end of the SIP part of the message. I don't argue on
that point.

My issue is, that once we've shifted a message-header, why isn't the definition of LWS
valid anymore, since we still are parsing the SIP part of the message?

> The _only_ place that second CRLF matches in the grammar is the literal in
> the Request and Response rules.

[CHH] Again, the point is not where it matches in the grammar. I am not talking about
the SIP grammar, but of the definition of the parts/tokens building the grammar.

> So for your example, any whitespace following that second CRLF matches message-body =
> *OCTET, NOT LWS.

[CHH] Again, I don't think we have any token called the "second CRLF" in the SIP
grammar.

I also think it's important to have a look at the definition of OCTET. RFC2234 says:

OCTET          =  %x00-FF

SP                  =  %x20

Now, if we say that CRLF/SP is LWS, but that CRLF/OCTET is not LWS, don't we have a
conflict here, since %x20 is part of both SP and OCTET?

Anyway, I don't want to go on with this argument forever. I am not an ABNF expert, and
it is possible to fix this on the implementation level, so...

Regards,

Christer Holmberg
Ericsson Finland



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From: "Pete Cordell" <pete@tech-know-ware.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
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Subject: Re: [Sip] B2BUA - Security
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OK - I'll go through some scenarios (host is agent, 'proxy' is agent, NAT
only, firewall only, NAT & firewall) and derive some requirements that
Midcom desires of SIP.

I assume these are application specific requirements which will be
generalised in a future round of development.  Is that the correct way to do
it?

Pete.

(to: and cc: list pruned in case it looks like spam!)

----- Original Message -----
From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
To: "Mark Watson" <mwatson@nortelnetworks.com>
Cc: "'Rohan Mahy'" <rohan@cisco.com>; "Mahey, Sonit"
<Sonit.Mahey@icn.siemens.com>; "Pete Cordell" <pete@tech-know-ware.com>;
"Christian Huitema" <huitema@windows.microsoft.com>; <sip@ietf.org>
Sent: 09 December 2002 18:37
Subject: Re: [Sip] B2BUA - Security


> inline.
>
> Mark Watson wrote:
> > Rohan wrote:
> >
> >  > In addition to Jonathan's approach to session policy, there is no
> >  > reason a middlebox controller can't be a proxy if it:
> >  >
> >  > 1. opens pinholes referenced in a clear text or integrity
> >  > protected SDP
> >  > 2. rejects requests with no readable SDP (the UA could
> >  > instead include
> >  > a full SDP, with keying material, encrypted with SMIME, ****and an
> >  > abbreviated copy which is only integrity protected****).
> >
> >
> > I was about to write a draft suggesting exactly this.
>
> THis is simply not sufficient.
>
> It will suffice for firewalls, but not for NAT, where an actual
> modification needs to occur.
>
> I think that, per Pete's suggestion, the correct path forward is a
> requirements document that describes the problem and the requirements
> for the solution. I believe there are many interesting requirements here
> about disclosure of what is being done, that will strongly impact the
> solution.
>
> >
> > As per Jonathan's draft there is a need for the UA to be able to
> > 'declare' to network entities those parts of the SDP information which
> > are needed for the network entity to perform whatever service it
> > performs (e.g. NAT or Firewall control). But the UA should not have to
> > declare the whole SDP, and the terminating UA should be able to tell if
> > a network entity has modified these parameters.
>
> Sounds like we are beginning the requirements work here...
>
> >
> > Rather than putting this information into new SIP headers, we could
> > include two copies of the SDP. One copy is unprotected and contains only
> > the information that the UE wishes to 'declare' to the network entities.
> > The other copy is protected and contains the full information.
>
> I think we can leave the header/body debate for down the road. I would
> be happy if we can get agreement on this general problem as one that we
> need to solve.
>
> >
> > This approach would work through existing devices which inspect/modify
SDP.
> >
> > I'm not sure how an existing S/MIME compliant client would behave on
> > receiving these two copies. If it uses the unprotected SDP, then things
> > are no worse than the present situation - the call suceeds, but without
> > the advantages of S/MIME. If it uses the protected SDP (which has not
> > been modified by the NAT controlling device), then the call fails, but
> > this is still no worse than the present situation.
>
> It comes down to disclosure. The message needs to convey information
> about the meaning of the multiple session descriptions - who set which,
> and why.
>
> -Jonathan R.
> --
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>
> _______________________________________________
> Sip mailing list  https://www1.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] ABNF issue
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CRLF/WSP is definitely a LWS, but you have to realise that the second CRLF signals the end of the SIP headers. The WSP that follows is part of the body so you can't combine the CRLF (that is part of the headers) and the WSP (which is part of the body) as a LWS

sipheader:value/CRLF/CRLF/WSP
                      ^
                      |
                      |
            sip headers end. WSP is sent to the body parser (which is presumably a separate one from the sip parser)

Regards,
Hisham

> -----Original Message-----
> From: ext Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
> Sent: Wednesday, December 11, 2002 10:15 AM
> To: Robert Sparks
> Cc: Jonathan Rosenberg; sip@ietf.org
> Subject: Re: [Sip] ABNF issue
> 
> 
> 
> Hi,
> 
> > Blindly doing a reverse token substitution will produce an incorrect
> > parse. You can only apply it when you have proven the left side of
> > the rule fits in a valid parse of the message.
> 
> [CHH] Correct. And, I "shift" a message-header when I have 
> found one CRLF (the end of
> the header). After that I'm done with that header. After that 
> I continue parsing, and
> if the previous message-header was the last one I expect to 
> parse a second CRLF, to
> indentify the end of the SIP header part of the message. So 
> far I agree. What we
> disagree on is the fact that you say that now the definition 
> of LWS is not valid
> anymore (even if we are still parsing the SIP part of the 
> message), and that in this
> case a CRLF is a CRLF - no matter what comes after it.
> 
> > So from your original example. LWS is not defined inside a 
> body - the
> > body is just a sequence of octets from the perspective of the SIP
> > grammar. Your SIP parser _cannot_ apply a match against LWS 
> inside it -
> > there is no valid expansion of the SIP grammar that has LWS 
> appearing
> > after CRLF CRLF
> 
> [CHH] I don't think there is such thing as CRLF CRLF in the 
> current SIP grammar (it
> used to be before, though, I think). Again, the 
> message-header is shifted once I found
> one CRLF, so there will be no CRLF CRLF shift.
> 
> > >>The BNF specifies that message-header ends with a CRLF.
> > >
> > >Yes.
> > >
> > >>What follows that is one of two things - either another 
> CRLF, marking the end
> > >>of the headers, or another message header, which never 
> begins with LWS.
> > >
> > >>From a SIP grammar point of view that is correct. 
> However, I don't see it said
> > >>anywhere that the definition of LWS isn't valid at the 
> beginning of a new line.
> >
> > It _is_ valid after CRLF (which is what I think you mean by 
> a new line)
> > if (and only if) you are still in the expansion of message-header.
> 
> [CHH] That is what I am asking: where is it said that LWS is 
> only valid within the
> context of a message-header?
> 
> I do agree that LWS is not valid within the message-body part 
> of the message, once we
> get there, but I see it nowhere said that I can't parse a LWS 
> AFTER I've shifted a
> message-header, when I'm still in the SIP part of the 
> message. No, the SIP grammar may
> not allow it, and we may return a parser error, but that is a 
> separate issue.
> 
> > >Also, just for the record, where is it said that a 
> sip-header can't start with
> > >LWS? Where is it said that line folding can't be used at 
> the beginning of a line?
> > >The text only says that the sip-header field values can be 
> folded onto multiple
> > >lines.
> > I don't know what you mean by beginning of a line here. I 
> assume you are
> > asking why we cant have a header that looks like (\b for blank):
> > \b\bNewHeader: foo which is a "\b\bNewheader"?
> >
> > The grammar answers that. extension-header starts with a token.
> 
> [CHH] Of course the LWS would not be part of the 
> message-header grammar itself. LWS is
> not even part of the message-header grammar within a header 
> (ie the SIP grammar doesn't
> require any LWS to be parsed). The grammar only says that LWS 
> can be used
> message-header parts for line folding, but it's not part of 
> the grammar.
> 
> Let's take an example:
> 
> INVITE sip:12345@sip.com SIP/2.0 <CRLF_1>
> Call-ID: 67890@sip.com <CRLF_2>
> <CRLF_3>
> <SP_1>CSeq: 100 INVITE <CRLF_4>
> 
> Now, the parser shifts the Call-ID header when it has found CRLF_2.
> 
> Then, it finds CRLF_3 and SP_1, and returns LWS. So, there is 
> a line-folding before the
> beginning of the CSeq header. Again, the LWS is NOT part of 
> the CSeq header itself, but
> my question is: where is it said that this LWS is NOT allowed 
> at the beginning of a
> line, before the actual message-header, but only WITHIN a 
> message-header? Have I missed
> it in the text?
> 
> > >>Thus, if a CRLF is followed by something OTHER than WSP, 
> the next construction
> > >>cannot be LWS.
> > >
> > >Maybe I missunderstand you, but the case I try to describe 
> is when the CRLF IS
> > >followed by WSP.
> > If its in a valid expansion of message-header (which will not happen
> > across a CRLFCRLF string), then you are looking at LWS.
> 
> [CHH] I may be wrong, but again I don't think there is such a 
> thing as CRLFCRLF
> (DOUBLE-CRLF) in the current SIP grammar.
> 
> Each CRLF is treated separately, and if there is another CRLF 
> after a message-header
> has been shifted we have found the end of the SIP part of the 
> message. I don't argue on
> that point.
> 
> My issue is, that once we've shifted a message-header, why 
> isn't the definition of LWS
> valid anymore, since we still are parsing the SIP part of the message?
> 
> > The _only_ place that second CRLF matches in the grammar is 
> the literal in
> > the Request and Response rules.
> 
> [CHH] Again, the point is not where it matches in the 
> grammar. I am not talking about
> the SIP grammar, but of the definition of the parts/tokens 
> building the grammar.
> 
> > So for your example, any whitespace following that second 
> CRLF matches message-body =
> > *OCTET, NOT LWS.
> 
> [CHH] Again, I don't think we have any token called the 
> "second CRLF" in the SIP
> grammar.
> 
> I also think it's important to have a look at the definition 
> of OCTET. RFC2234 says:
> 
> OCTET          =  %x00-FF
> 
> SP                  =  %x20
> 
> Now, if we say that CRLF/SP is LWS, but that CRLF/OCTET is 
> not LWS, don't we have a
> conflict here, since %x20 is part of both SP and OCTET?
> 
> Anyway, I don't want to go on with this argument forever. I 
> am not an ABNF expert, and
> it is possible to fix this on the implementation level, so...
> 
> Regards,
> 
> Christer Holmberg
> Ericsson Finland
> 
> 
> 
> _______________________________________________
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Hi,

> CRLF/WSP is definitely a LWS, but you have to realise that the second CRLF signals the end of the SIP headers. The WSP that follows is part of the body so you can't combine the CRLF (that is part of the headers) and the WSP (which is part of the body) as a LWS
>
> sipheader:value/CRLF/CRLF/WSP
>                       ^
>                       |
>                       |
>             sip headers end. WSP is sent to the body parser (which is presumably a separate one from the sip parser)

What you say would work very fine, but that is not how the grammar is defined today. Please correct me if I'm wrong, but there is no CRLF/CRLF/WSP token defined in the grammar today. YES, you can do this in your implementation, so that is not the problem. My
original question was if there is a bug in the spec or not.

Regards,

Christer Holmberg
Ericsson Finland


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I would argue that the grammar is not ambiguous when taken in context.  Refer the ABNF and text in section 7:

         generic-message  =  start-line
                             *message-header
                             CRLF
                             [ message-body ]
         start-line       =  Request-Line / Status-Line

   The start-line, each message-header line, and the empty line MUST be
   terminated by a carriage-return line-feed sequence (CRLF).  Note that
   the empty line MUST be present even if the message-body is not.

As has been shown before, you can't always take the ABNF in a vacuum - the context found in the rest of the RFC can often narrow what's allowed by the protocol, even when the ABNF says it's "legal".

- rob

--
Rob Phillips, Sr. Software Engineer                 Netrake Corporation
mailto: rob@netrake.com                           3000 Technology Drive
voice: (214) 291-1096                                         Suite 100
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Hi Rob,

> I would argue that the grammar is not ambiguous when taken in context.  Refer the ABNF and text in section 7:
>
>          generic-message  =  start-line
>                              *message-header
>                              CRLF
>                              [ message-body ]
>          start-line       =  Request-Line / Status-Line
>
>    The start-line, each message-header line, and the empty line MUST be
>    terminated by a carriage-return line-feed sequence (CRLF).  Note that
>    the empty line MUST be present even if the message-body is not.

Yes.

> As has been shown before, you can't always take the ABNF in a vacuum - the context found in the rest of the RFC can often narrow what's allowed by the protocol, even when the ABNF says it's "legal".

I FULLY agree. But, in those cases you have to specify the things in the spec. Now the spec only says that the SIP part of the message ends with CRLF, and that is not the issue. IF the spec would
"narrow" the ABNF rules by saying that the definition of LWS is not valid in the beginning of a new line, OR if the spec would define the CRLFCRLF token, we would not have this problem.

Regards,

Christer Holmberg
Ericsson Finland








>
>
> - rob
>
> --
> Rob Phillips, Sr. Software Engineer                 Netrake Corporation
> mailto: rob@netrake.com                           3000 Technology Drive
> voice: (214) 291-1096                                         Suite 100
> fax:   (214) 291-1010     http://www.netrake.com     Plano, Texas 75074

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Christer Holmberg writes:

> IF the spec would "narrow" the ABNF rules by saying that the definition 
> of LWS is not valid in the beginning of a new line, OR if the spec 
> would define the CRLFCRLF token, we would not have this problem.

I guess the way I read

          generic-message  =  start-line
                              *message-header
                              CRLF
                              [ message-body ]

the spec essentially DOES define a defacto CRLF-CRLF token (if not in ABNF) by saying that:
1. All message headers end with CRLF, and
2. When another CRLF follows directly thereafter, that marks the end of the message headers and the beginning of the message body (if any).

So that in taking your example:

INVITE sip:12345@sip.com SIP/2.0 <CRLF_1>
Call-ID: 67890@sip.com <CRLF_2>
<CRLF_3>
<SP_1>CSeq: 100 INVITE <CRLF_4>

the precedence rules for interpreting CRLF_2-CRLF_3-SP_1 have to be treated as 
	(CRLF_2-CRLF_3) SP_1	"end of headers" 
instead of 
	CRLF_2 (CRLF_3-SP_1)	"CRLF LWS"

- rob
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Hi Rob,

> > IF the spec would "narrow" the ABNF rules by saying that the definition
> > of LWS is not valid in the beginning of a new line, OR if the spec
> > would define the CRLFCRLF token, we would not have this problem.
>
> I guess the way I read
>
>           generic-message  =  start-line
>                               *message-header
>                               CRLF
>                               [ message-body ]
>
> the spec essentially DOES define a defacto CRLF-CRLF token (if not in ABNF) by saying that:
> 1. All message headers end with CRLF, and
> 2. When another CRLF follows directly thereafter, that marks the end of the message headers and the beginning of the message body (if any).

This "directly after" is part of what confuses me. To me it still sounds like a separate token, ie not a single CRLF-CRLF token. Anyway, if I
am the onlyone who understands in a different way there is no idea to continue arguing about it. Also, if everybody also agree on what is said
below, ie that a SIP header always must start at the beginning of a new line, there are no conflicts.

> So that in taking your example:
>
> INVITE sip:12345@sip.com SIP/2.0 <CRLF_1>
> Call-ID: 67890@sip.com <CRLF_2>
> <CRLF_3>
> <SP_1>CSeq: 100 INVITE <CRLF_4>
>
> the precedence rules for interpreting CRLF_2-CRLF_3-SP_1 have to be treated as
>         (CRLF_2-CRLF_3) SP_1    "end of headers"
> instead of
>         CRLF_2 (CRLF_3-SP_1)    "CRLF LWS"

So, this means that is not possible to use line folding before the first token (the header name) of a header, since assuming that the "directly
after" rule really means a CRLF-CRLF token the parser will assume it has found the end of the SIP part (due to the CRLF_2-CRLF_3) token, and
<SP_1> and the CSeq header will be treated as part of the message-body.

In this case, I think it is important to note that a SIP header MUST start at the BEGINNING of a new line. I think is said in the SDP spec, for
the SDP parameters, but not in the SIP spec for the SIP headers, is it? Also, H.248 explicitly says when something must be at the beginning of
a new line, and when SP eg is allowed.

Maybe this is something that could be tested at SIPit, or at least added to the torture test document?

Regards,

Christer Holmberg
Ericsson Finland

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> -----Original Message-----
> From: ext Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
> Sent: Wednesday, December 11, 2002 2:42 PM
> To: Rob Phillips
> Cc: Khartabil Hisham (NMP/Helsinki); rsparks@dynamicsoft.com;
> jdrosen@dynamicsoft.com; sip@ietf.org
> Subject: Re: [Sip] ABNF issue

> 
> In this case, I think it is important to note that a SIP 
> header MUST start at the BEGINNING of a new line. 

Header fields can be extended over multiple lines by preceding each extra line with at least one SP or horizontal tab (HT).  

Also, in RFC2822:

"Messages are divided into lines of characters.  A line is a series of
   characters that is delimited with the two characters carriage-return
   and line-feed; that is, the carriage return (CR) character (ASCII
   value 13) followed immediately by the line feed (LF) character (ASCII
   value 10).  (The carriage-return/line-feed pair is usually written in
   this document as "CRLF".)

A message consists of header fields (collectively called "the header
   of the message") followed, optionally, by a body.  

...

Header fields are lines composed of a field name, followed by a colon
   (":"), followed by a field body, and terminated by CRLF.  A field
   name MUST be composed of printable US-ASCII characters (i.e.,
   characters that have values between 33 and 126, inclusive), except
   colon.  "

No header name is allowed a CR or LF in it,and certainly no sip header has CRLF included in its name.

Regards,
Hisham

> 
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Hi,

> >In this case, I think it is important to note that a SIP header MUST start at the BEGINNING of a new line.
>
> Header fields can be extended over multiple lines by preceding each extra line with at least one SP or horizontal tab (HT).

That's line folding, which is a separate issue. What I said was that the START of the header MUST be at the BEGINNING of a new
line, no matter if the rest of the header is then folded/extended over multiple lines.

> Also, in RFC2822:
>
> "Messages are divided into lines of characters.  A line is a series of
>    characters that is delimited with the two characters carriage-return
>    and line-feed; that is, the carriage return (CR) character (ASCII
>    value 13) followed immediately by the line feed (LF) character (ASCII
>    value 10).  (The carriage-return/line-feed pair is usually written in
>    this document as "CRLF".)
>
> A message consists of header fields (collectively called "the header
>    of the message") followed, optionally, by a body.
>
> ...
>
> Header fields are lines composed of a field name, followed by a colon
>    (":"), followed by a field body, and terminated by CRLF.  A field
>    name MUST be composed of printable US-ASCII characters (i.e.,
>    characters that have values between 33 and 126, inclusive), except
>    colon.  "
>
> No header name is allowed a CR or LF in it,and certainly no sip header has CRLF included in its name.

I never said that the CRLF is part of the header name. My question was if the header name token could be proceeded by LWS, or
the header name token always be at the beginning of a new line.

Regards,

Christer Holmberg
Ericsson Finland


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The answer to this question:

> My question was if the header name token could be proceeded by LWS, or
> the header name token always be at the beginning of a new line.

is:

	RFC-3261 does not specifically ALLOW the name token to be proceeded by LWS, i.e. the definition of message-header per ABNF is:

	message-header = (Accept / .... / extension-header) CRLF

Therefore, the header name token MUST always be at the beginning of a new line.

- rob
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Subject: RE: [Sip] B2BUA
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Hello,

The last mails on this thread are once again confirming that the SIP
protocol specification does not mandate for any B2BUA behaviour.
This should be clearly written in the RFC 3261.

This does not preclude in anyway to write separate drafts defining specific
SIP applications or classes of SIP applications behaving as B2BUA with
given rules (transparency, etc.). Interested people can start specific
discussion threads for those cases.

Comments ?

Best regards
Juan Carlos



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Subject: Re: [Sip] B2BUA - Security
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On Wednesday, December 11, 2002, at 01:03 AM, Pete Cordell wrote:

> OK - I'll go through some scenarios (host is agent, 'proxy' is agent, 
> NAT
> only, firewall only, NAT & firewall) and derive some requirements that
> Midcom desires of SIP.

good.

> I assume these are application specific requirements which will be
> generalised in a future round of development.  Is that the correct way 
> to do
> it?

yes, thanks.  whether it is a draft or just some text sent to the 
mailing list, lets discuss this in SIPPING, since it is a requirements 
discussion.

thx,
-rohan


> Pete.
>
> (to: and cc: list pruned in case it looks like spam!)
>
> ----- Original Message -----
> From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
> To: "Mark Watson" <mwatson@nortelnetworks.com>
> Cc: "'Rohan Mahy'" <rohan@cisco.com>; "Mahey, Sonit"
> <Sonit.Mahey@icn.siemens.com>; "Pete Cordell" 
> <pete@tech-know-ware.com>;
> "Christian Huitema" <huitema@windows.microsoft.com>; <sip@ietf.org>
> Sent: 09 December 2002 18:37
> Subject: Re: [Sip] B2BUA - Security
>
>
>> inline.
>>
>> Mark Watson wrote:
>>> Rohan wrote:
>>>
>>>> In addition to Jonathan's approach to session policy, there is no
>>>> reason a middlebox controller can't be a proxy if it:
>>>>
>>>> 1. opens pinholes referenced in a clear text or integrity
>>>> protected SDP
>>>> 2. rejects requests with no readable SDP (the UA could
>>>> instead include
>>>> a full SDP, with keying material, encrypted with SMIME, ****and an
>>>> abbreviated copy which is only integrity protected****).
>>>
>>>
>>> I was about to write a draft suggesting exactly this.
>>
>> THis is simply not sufficient.
>>
>> It will suffice for firewalls, but not for NAT, where an actual
>> modification needs to occur.
>>
>> I think that, per Pete's suggestion, the correct path forward is a
>> requirements document that describes the problem and the requirements
>> for the solution. I believe there are many interesting requirements 
>> here
>> about disclosure of what is being done, that will strongly impact the
>> solution.
>>
>>>
>>> As per Jonathan's draft there is a need for the UA to be able to
>>> 'declare' to network entities those parts of the SDP information 
>>> which
>>> are needed for the network entity to perform whatever service it
>>> performs (e.g. NAT or Firewall control). But the UA should not have 
>>> to
>>> declare the whole SDP, and the terminating UA should be able to tell 
>>> if
>>> a network entity has modified these parameters.
>>
>> Sounds like we are beginning the requirements work here...
>>
>>>
>>> Rather than putting this information into new SIP headers, we could
>>> include two copies of the SDP. One copy is unprotected and contains 
>>> only
>>> the information that the UE wishes to 'declare' to the network 
>>> entities.
>>> The other copy is protected and contains the full information.
>>
>> I think we can leave the header/body debate for down the road. I would
>> be happy if we can get agreement on this general problem as one that 
>> we
>> need to solve.
>>
>>>
>>> This approach would work through existing devices which 
>>> inspect/modify
> SDP.
>>>
>>> I'm not sure how an existing S/MIME compliant client would behave on
>>> receiving these two copies. If it uses the unprotected SDP, then 
>>> things
>>> are no worse than the present situation - the call suceeds, but 
>>> without
>>> the advantages of S/MIME. If it uses the protected SDP (which has not
>>> been modified by the NAT controlling device), then the call fails, 
>>> but
>>> this is still no worse than the present situation.
>>
>> It comes down to disclosure. The message needs to convey information
>> about the meaning of the multiple session descriptions - who set 
>> which,
>> and why.
>>
>> -Jonathan R.
>> --
>> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
>> Chief Scientist                             First Floor
>> dynamicsoft                                 East Hanover, NJ 07936
>> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>> http://www.jdrosen.net                      PHONE: (973) 952-5000
>> http://www.dynamicsoft.com
>>
>> _______________________________________________
>> Sip mailing list  https://www1.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://www1.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] ABNF issue
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To: Christer Holmberg <christer.holmberg@lmf.ericsson.se>
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Christer,

You specific question has been answered twice already, and I will 
answer it a third time.  If you still do not understand, please email 
one of the chairs privately and we will arrange for someone to explain 
it to you.



On Wednesday, December 11, 2002, at 06:21 AM, Christer Holmberg wrote:
> I never said that the CRLF is part of the header name. My question was 
> if the header name token could be proceeded by LWS, or
> the header name token always be at the beginning of a new line.

as rob@netrake said:
> 	RFC-3261 does not specifically ALLOW the name token to be proceeded 
> by LWS, i.e. the definition of message-header per ABNF is:
>
> 	message-header = (Accept / .... / extension-header) CRLF
>
> Therefore, the header name token MUST always be at the beginning of a 
> new line.

and as robert sparks said:
> The grammar answers that.
> extension-header starts with a token.

in other words, a SIP header ALWAYS starts with a token (all of the 
existing headers are tokens), which can NEVER contain leading 
whitespace.

thanks,
-rohan

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Hi,

> You specific question has been answered twice already, and I will
> answer it a third time.  If you still do not understand, please email
> one of the chairs privately and we will arrange for someone to explain
> it to you.
>
> On Wednesday, December 11, 2002, at 06:21 AM, Christer Holmberg wrote:
> > I never said that the CRLF is part of the header name. My question was
> > if the header name token could be proceeded by LWS, or
> > the header name token always be at the beginning of a new line.
>
> as rob@netrake said:
> >       RFC-3261 does not specifically ALLOW the name token to be proceeded
> > by LWS, i.e. the definition of message-header per ABNF is:
> >
> >       message-header = (Accept / .... / extension-header) CRLF
> >
> > Therefore, the header name token MUST always be at the beginning of a
> > new line.
>
> and as robert sparks said:
> > The grammar answers that.
> > extension-header starts with a token.
>
> in other words, a SIP header ALWAYS starts with a token (all of the
> existing headers are tokens), which can NEVER contain leading
> whitespace.

And I don't know how many times I've said this: I've NEVER claimed that a
header would start with anything else than a token. What I've been talking
about are general ABNF rules.

Anway, I have had a private discussion with Brian about this, so there is no
idea to continue it here. Rob also presented some assumptions we make in SIP,
and if everybody agrees on those I am happy.

Regards,

Christer Holmberg
Ericsson Finland


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Hi all,

     When I'm reading RFC3261,  I met "no-op" in the document.  But I 
can not find where it is defined, and I want to
  know what's the meaning of "no-op"?  no-op ?= no-operation

     Thanks  a lot!

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I'm gathering SIP network deployment information at the service provider level.  Anyone that has experience deploying a SIP network at this level please reply with troubleshooting tips and significant challenges that were dealt with.  Thank you.

Brian
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From: "Cullen Jennings" <fluffy@cisco.com>
To: <sip@ietf.org>
Subject: RE: [Sip] B2BUA & Max-Forwards
Date: Thu, 12 Dec 2002 11:23:10 -0800
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I can't really see how it matters if the term B2BUA is defined in 3261bis or
not since it would make no normative changes to the specification either
way.

It does seem very obvious to me that B2BUA is a useful description of a
certain class of valid SIP applications. I'm sure the term will continue to
be widely used and one the most commonly occurring works in the sip mailing
lists. I can't imagine how we could wish the term away or even why we would
want too. Deans 10 rules of transparency will no doubt be used to describe
the characteristics of certain B2BUA. Thanks Dean.

So out of this long and dreary thread, I see only a few real proposals of

Do a INFORMATIONAL draft(s) with one or more of the following:

1) describing various transparencies of B2BUA
2) attempt some level of definition of what we mean by a B2BUA
3) show some call flows (like the service examples) that show possible call
flows for solving some of the common problems people are trying to solve
with B2BUA

My view is I don't see any particular harm in doing any of the above but I
would not have much time to help on them because I think there are more
important things. If other people do have time (as several have suggested
they do) lets say thanks and get on with it.

Cullen



> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Drage,
> Keith (Keith)
> Sent: Monday, December 09, 2002 7:42 AM
> To: Mary Barnes; 'Jonathan Rosenberg'; Pete Cordell
> Cc: David R. Oran; Dean Willis; sip@ietf.org
> Subject: RE: [Sip] B2BUA & Max-Forwards
>
>
> This thread is definitely going way beyond what it should do
> (both in scope and duration) - it did however start (about a 100
> messages ago) with some reasonable modifications to the B2BUA
> definition, and save those relatively minor modifications, I do
> not want to see any other modifications in RFC 3261 or its next revision.
>
> I certainly will not agree to removal of any existing definition of B2BUA.
>
> Keith
>
> Keith Drage
> Lucent Technologies
> Tel: +44 1793 776249
> Email: drage@lucent.com
>
> > -----Original Message-----
> > From: Mary Barnes [mailto:mbarnes@nortelnetworks.com]
> > Sent: 06 December 2002 15:40
> > To: 'Jonathan Rosenberg'; Pete Cordell
> > Cc: David R. Oran; Dean Willis; sip@ietf.org
> > Subject: RE: [Sip] B2BUA & Max-Forwards
> >
> >
> >
> > I agree with Jonathan.  As mentioned earlier this week
> > apparently, there is
> > already work underway by some individuals to write an
> > informational draft
> > documenting BBUA scenarios/call flows.
> >
> > I'll keep my fingers crossed that the existence of such a
> > document can save
> > us from another round of B2BUA discussions again in 3 months.
> >
> > Regards,
> > Mary.
> >
> > -----Original Message-----
> > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > Sent: Friday, December 06, 2002 12:59 AM
> > To: Pete Cordell
> > Cc: David R. Oran; Dean Willis; sip@ietf.org
> > Subject: Re: [Sip] B2BUA & Max-Forwards
> >
> >
> >
> >
> > Pete Cordell wrote:
> > > Dave,
> > >
> > > To save your disagreeableness, can we say something like:
> > >
> > > "A B2BUA MUST perform loop prevention.  One way to do this
> > is to propagate
> > > the Max-Forwards header from the incoming request to the
> > outgoing request
> > in
> > > the same way that a proxy does."
> >
> > A statement like this is completely out of scope for rfc3261.
> > It might
> > be a reasonable statement to make in some kind of application
> > document
> > that describes a particular type of b2bua. However, it makes no more
> > sense to say this in RFC 3261 as it would be to discuss the ways in
> > which to build a focus or a 3pcc controller (both of which are also
> > informational, by the way).
> >
> > I can't resist pointing out that much of this is deja vu. Folks may
> > recall heated discussions in Adelaide, where there was a bof
> > (I think)
> > that wanted to have a working group to specify SIP to H.323
> > conversion.
> > THe same arguemtns that were made there are being made here; its an
> > application level issue, there are many ways to do it, don't stifle
> > creativity, etc. In fact I recall that Dave was, as always, quite
> > vociferous on these points.
> >
> > The eventual conclusion in the h323/sip conversion case was
> > to write an
> > INFORMATIONAL RFC that documents reasonable practices in
> > building such a
> > device. I believe the same logic and conclusions apply here. If folks
> > have a particular type of b2bua application they are fond of, please
> > feel free to write up your implementation advice and I am
> > sure sipping
> > would consider whether or not it would make a useful
> > informational RFC.
> >
> > However, it is completely inappropriate to ever make such application
> > logic normative, and even more inappropraite for such wording
> > to be in
> > rfc3261.
> >
> > Anyway, I too am tired of this thread. I think, in fact, there is
> > "rough" consensus to remove the definition of b2bua entirely from
> > rfc3261 in its next revision (which is quite some time in the future
> > anyway). However, I'll let the chairs decide that.
> >
> > -Jonathan R.
> >
> > --
> > Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> > Chief Scientist                             First Floor
> > dynamicsoft                                 East Hanover, NJ 07936
> > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > http://www.jdrosen.net                      PHONE: (973) 952-5000
> > http://www.dynamicsoft.com
> >
> > _______________________________________________
> > Sip mailing list  https://www1.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://www1.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://www1.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: "Dean Willis" <dwillis@dynamicsoft.com>
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Date: Thu, 12 Dec 2002 23:10:46 -0600
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I have posted draft minutes at:

http://www.softarmor.com/sipwg/meets/ietf55/minutes/minutes.html

Please review ASAP.

Thanks,

--
Dean

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Subject: [Sip] Challenges when interworking QSIG and SIP Call Transfer
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When trying to put together some interworking scenarios between SIP call
transfer and QSIG call transfer, some problems were identified. The
purpose of this E-mail message is to solicit advice on how these
problems could best be tackled. In the below text, SIP terminology is
used, rather than telecoms terminology. Therefore, parties that would
normally be referred to as "Party X or X-Party" are being referred to
as Transferor, Transferee and Transfer Target. Hopefully, this makes the
text more readable. 

Currently there is no specific SIP mechanism to notify parties involved
in SIP call transfer that a call transfer is taking place or has taken
place, without establishing a new session. I.e. the notification of
transfer and the communication of identities is solved (for attended
transfer only) by including a Replaces "header" in a SIP URI in the
Refer-to header in a REFER (to the Transferee) and/or by including a
Replaces header in an INVITE (to the Transfer Target). 

The current mechanisms are sufficient to effect the call transfer, but
in some cases the Transferee and/or the Transfer Target will not be
aware of what is going on. In particular in interworking scenarios with
QSIG, some difficulties have been identified. 

Below, first an attempt to analyse the situation for a pure SIP
situation is presented. The purpose of this is to determine if the
problems that were identified for QSIG interworking scenarios have a
broader scope. 

The "All" SIP situation 

The below examples are based on the situation that the Transferor has a
session established with both the Transferee and the Transfer Target.
For the purpose of this analysis, this will be referred to as "attended
transfer". 

1) When the Transferor does not support the use of the
Replaces header in the Refer-to URI, neither the Transferee, nor the
Transfer Target can be explicitely aware of a transfer. This knowledge,
however, could be inferred from the fact that the calls between the
Transferor and the Transferee and between the Transferor and the
Transfer Target are terminated shortly after the new call between the
Transferee and the Transfer Target has been established. Both the
Transferee and the Transfer Target are aware of each other's identity
because the Tranferee establishes the new session with the Transfer
Target. 

2) When the Transferor does support the use of the Replaces
header in the Refer-to URI, but the Transferee does not, the situation,
could be identical to the above one, provided that the Transferee simply
ignores this header. An alternative could be that the Transferee does
not interpret this header and simply copies it into the INVITE to the
Transfer Target. The Transfer Target, depending on its capabilities, can
then be aware of the fact that a call transfer is taking place. The
Transferee will not be aware of this. The Transferee and the Transfer
Target are aware of each other's identity. 

3) When the Transferor and the Transferee support the use of
Replaces header in a Refer-to URI and the Transfer Target supports
the use of the Replaces header in an INVITE message, both the Transferee
and the Transfer Target are aware of the transfer and the identities of
the new parties in the call. 

Summarising the above situations, it can be stated that in all the above
cases the Transferee and the Transfer Target are aware of each other's
identity, but may not be aware of the fact that a transfer has taken
place. When all involved parties know about support the use of the
Replaces header (in a Refer-to URI and/or an INVITE message), the
Transferee and the Transfer Target are both aware of transfer taking
place and the identities of each other. The means of "signalling" that a
transfer is taking place is the Replaces header (either in a Refer-to
URI or in an INVITE message). 

For situations in which the Transferor does have a session established
with the Transferee, but not with the Transfer Target (prior to sending
a REFER message to the Transferee) the situation is somewhat simpler.
For the purpose of this analysis, this will be referred to as "unattended
transfer". It is not possible to use the Replaces header (either in a
Refer-to URI or in an INVITE message) because there is no session to
replace. Therefore, neither the Transferee nor the Transfer Target are
explicitely aware of the fact that a transfer is taking place. The
Transferee and the Transfer Target are aware of each other's identity. 

Interworking scenarios with QSIG 

When interworking with QSIG call transfer (attended transfer) or QSIG
single step call transfer ("unattended transfer"), the situation is
different. Two examples (one for attended transfer and one for
unattended transfer) are given below. 

1) Attended transfer. 

The Transferor is in QSIG and the Transferee and the Transfer Target are
in SIP. When the Transferor performs a Call Transfer by join, it will
send a notification to both the Transferee and the Transfer Target and
then connect the B-channels of the respective calls. 
Each notification will arrive at a SIP gateway, which has an established
session with either Transferee or the Transfer Target. No SIP signalling
is required to "connect" these two sessions at the Transferor. Although
it may be desirable (from a SIP point of view or for QoS reasons) to
have the Transferee establish a session with the Transfer Target, this
would complicate matters from the QSIG point of view: The (QSIG)
Transferor assumes that it is in control over the two calls and the
associated B-channels. Relinquishing control to the Transferee would cut
the Transferor out of the picture. Furthermore, the Transferor in this
situation is a node in the QSIG network and we cannot assume that this
node will be upgraded in order to support calls via gateways to/from
SIP. In any case, such a node is unlikely to be aware that the
transferee and transfer target are in a SIP network, and therefore will
not be in a position to treat such a scenario differently from a normal
QSIG-only scenario. 

To be able to inform the Transferee and the Transfer Target about the
transfer that took place and about the identities of the new parties in
the call, the options that are available today would seem to be the
following: 

- send a new INVITE with a Replaces header to both the Transferee and
  the Transfer Target. The destination of the INVITE, however, has to
  support the Replaces header to avoid undesirable side effects.

- send a re-INVITE, optionally with a Referred-By header, to both the 
  Transferee and the Transfer Target. Although the meaning of 
  "Referred-By" is not that a call transfer took place, subsequent 
  release of the already established sessions (in combination with 
  Referred-by) could be used as a strong hint that call transfer did 
  take place.

- send a REFER to the Transferee (with the address of the GW in the
  Refer-to URI) and an INVITE with a Replaces header to the Transfer
  Target. 

These methods seem to generate a lot of messages and processing, while 
the only purpose of the establishment of the new sessions is to inform 
both parties about the transfer and/or the identities of the new 
parties. 

2) Unattended transfer. 

The Transferor is in QSIG and the Transferee and the Transfer Target are
in SIP. When the Transferor performs an unattended transfer, it can
establish a call with the Transfer Target (which also notifies the
Transfer Target that a transfer is taking place) and notify the
Transferee about the fact that a transfer is taking place and about the
progress of this new call. When the new call is established, the
B-channels of the two calls can be connected at the Transferor. There is
no way to inform the Transfer Target that this call is the result of a
transfer. There is no way to inform the Transferree that transfer is
taking place. The only way to "repair" this situation would seem to be
the use of the options mentioned in the previous example. Another 
possibility, is that the INVITE to the Transfer Target contains
history-info with a new reason code to indicate transfer.

Several interworking scenarios between SIP and QSIG for call transfer 
present this sort of problem. In most cases it would be sufficient if it 
were possible to simply send a notification message (i.e. a message that 
does neither modify the characteristics of the session nor its state) to 
the Transferee and/or the Transfer Target. (An example this could 
be a be a SIP INFO message with a P-Asserted-Identity header)

Any comments and/or proposals on the above are welcomed. 


Regards,

Frank
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Subject: [Sip] RFC 3327 on Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts
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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3327
                                    
        Title:      Session Initiation Protocol (SIP) Extension Header
                    Field for Registering Non-Adjacent Contacts
        Author(s):  D. Willis, B. Hoeneisen
        Status:     Standards Track
        Date:       December 2002
        Mailbox:    dean.willis@softarmor.com,
                    bernhard.honeisen@nokia.com, b.hoeneisen@ieee.org
        Pages:      17
        Characters: 36493
        Updates/Obsoletes/SeeAlso:   None
                                    
        I-D Tag:    draft-willis-sip-path-08.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3327.txt


The REGISTER function is used in a Session Initiation Protocol (SIP)
system primarily to associate a temporary contact address with an
address-of-record.  This contact is generally in the form of a Uniform
Resource Identifier (URI), such as Contact:
<sip:alice@pc33.atlanta.com> and is generally dynamic and associated
with the IP address or hostname of the SIP User Agent (UA).  The
problem is that network topology may have one or more SIP proxies
between the UA and the registrar, such that any request traveling from
the user's home network to the registered UA must traverse these
proxies.  The REGISTER method does not give us a mechanism to discover
and record this sequence of proxies in the registrar for future use.
This document defines an extension header field, "Path" which provides
such a mechanism.

This document is a product of the Session Initiation Protocol and the
Session Initiation Proposal Investigation Working Groups of the IETF.

This is now a Proposed Standard Protocol.

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
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Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

...

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Hello all,

  Can the authors give the detailed Changelog between refer-07 & refer-06?

  Thanks a lot!

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From mailnull@www1.ietf.org  Sat Dec 14 08:04:00 2002
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Hi Folks,
  (potentially dumb question warning):

Section 8.1.1.2 (To Header) of RFC3261 - Use of tel URL.

I imply from this that the tel URL in a To header can be evaluated at 
"each domain through which the request passes". Example given there - 
dialing "411" at an Airport would be evaluated at the outgoing proxy 
at the Airport, not in the user's home domain.

Q:: What is the meaning of the "user's home domain" when dialing out 
to a tel URL?

Follow-up:
When originating a call, one assumes that the INVITE request could go 
via an outgoing proxy. Is this proxy required to route a call onwards 
to a "home server", and if so how is that "home server" indicated in 
the INVITE sent to the outgoing proxy?

The From header and Contact header indicates the (potential) AoR and 
current addressability of the caller.
Is the From header used to indicate the home domain for the caller?

all the best from a confused,
   Lawrence
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Subject: Re: [Sip] Challenges when interworking QSIG and SIP Call Transfer
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Hi,

I appreciate the thorough analysis in your mail.  I plan to reply to 
the content of your message shortly. However, I wanted to reply to your 
mail right away with an important list management issue.

In the spirit of reducing cross postings between SIP and SIPPING, I am 
removing SIP from this reply.  This is purely a SIPPING issue, so 
everyone please do not post/reply to both lists. I think this is 
especially important when we are also cross posting to another 
standards group mailing list.

many thanks,
-rohan



On Friday, December 13, 2002, at 05:57 AM, frank.derks@philips.com 
wrote:

> When trying to put together some interworking scenarios between SIP 
> call
> transfer and QSIG call transfer, some problems were identified. The
> purpose of this E-mail message is to solicit advice on how these
> problems could best be tackled. In the below text, SIP terminology is
> used, rather than telecoms terminology. Therefore, parties that would
> normally be referred to as "Party X or X-Party" are being referred to
> as Transferor, Transferee and Transfer Target. Hopefully, this makes 
> the
> text more readable.
>
> Currently there is no specific SIP mechanism to notify parties involved
> in SIP call transfer that a call transfer is taking place or has taken
> place, without establishing a new session. I.e. the notification of
> transfer and the communication of identities is solved (for attended
> transfer only) by including a Replaces "header" in a SIP URI in the
> Refer-to header in a REFER (to the Transferee) and/or by including a
> Replaces header in an INVITE (to the Transfer Target).
>
> The current mechanisms are sufficient to effect the call transfer, but
> in some cases the Transferee and/or the Transfer Target will not be
> aware of what is going on. In particular in interworking scenarios with
> QSIG, some difficulties have been identified.
>
> Below, first an attempt to analyse the situation for a pure SIP
> situation is presented. The purpose of this is to determine if the
> problems that were identified for QSIG interworking scenarios have a
> broader scope.
>
> The "All" SIP situation
>
> The below examples are based on the situation that the Transferor has a
> session established with both the Transferee and the Transfer Target.
> For the purpose of this analysis, this will be referred to as "attended
> transfer".
>
> 1) When the Transferor does not support the use of the
> Replaces header in the Refer-to URI, neither the Transferee, nor the
> Transfer Target can be explicitely aware of a transfer. This knowledge,
> however, could be inferred from the fact that the calls between the
> Transferor and the Transferee and between the Transferor and the
> Transfer Target are terminated shortly after the new call between the
> Transferee and the Transfer Target has been established. Both the
> Transferee and the Transfer Target are aware of each other's identity
> because the Tranferee establishes the new session with the Transfer
> Target.
>
> 2) When the Transferor does support the use of the Replaces
> header in the Refer-to URI, but the Transferee does not, the situation,
> could be identical to the above one, provided that the Transferee 
> simply
> ignores this header. An alternative could be that the Transferee does
> not interpret this header and simply copies it into the INVITE to the
> Transfer Target. The Transfer Target, depending on its capabilities, 
> can
> then be aware of the fact that a call transfer is taking place. The
> Transferee will not be aware of this. The Transferee and the Transfer
> Target are aware of each other's identity.
>
> 3) When the Transferor and the Transferee support the use of
> Replaces header in a Refer-to URI and the Transfer Target supports
> the use of the Replaces header in an INVITE message, both the 
> Transferee
> and the Transfer Target are aware of the transfer and the identities of
> the new parties in the call.
>
> Summarising the above situations, it can be stated that in all the 
> above
> cases the Transferee and the Transfer Target are aware of each other's
> identity, but may not be aware of the fact that a transfer has taken
> place. When all involved parties know about support the use of the
> Replaces header (in a Refer-to URI and/or an INVITE message), the
> Transferee and the Transfer Target are both aware of transfer taking
> place and the identities of each other. The means of "signalling" that 
> a
> transfer is taking place is the Replaces header (either in a Refer-to
> URI or in an INVITE message).
>
> For situations in which the Transferor does have a session established
> with the Transferee, but not with the Transfer Target (prior to sending
> a REFER message to the Transferee) the situation is somewhat simpler.
> For the purpose of this analysis, this will be referred to as 
> "unattended
> transfer". It is not possible to use the Replaces header (either in a
> Refer-to URI or in an INVITE message) because there is no session to
> replace. Therefore, neither the Transferee nor the Transfer Target are
> explicitely aware of the fact that a transfer is taking place. The
> Transferee and the Transfer Target are aware of each other's identity.
>
> Interworking scenarios with QSIG
>
> When interworking with QSIG call transfer (attended transfer) or QSIG
> single step call transfer ("unattended transfer"), the situation is
> different. Two examples (one for attended transfer and one for
> unattended transfer) are given below.
>
> 1) Attended transfer.
>
> The Transferor is in QSIG and the Transferee and the Transfer Target 
> are
> in SIP. When the Transferor performs a Call Transfer by join, it will
> send a notification to both the Transferee and the Transfer Target and
> then connect the B-channels of the respective calls.
> Each notification will arrive at a SIP gateway, which has an 
> established
> session with either Transferee or the Transfer Target. No SIP 
> signalling
> is required to "connect" these two sessions at the Transferor. Although
> it may be desirable (from a SIP point of view or for QoS reasons) to
> have the Transferee establish a session with the Transfer Target, this
> would complicate matters from the QSIG point of view: The (QSIG)
> Transferor assumes that it is in control over the two calls and the
> associated B-channels. Relinquishing control to the Transferee would 
> cut
> the Transferor out of the picture. Furthermore, the Transferor in this
> situation is a node in the QSIG network and we cannot assume that this
> node will be upgraded in order to support calls via gateways to/from
> SIP. In any case, such a node is unlikely to be aware that the
> transferee and transfer target are in a SIP network, and therefore will
> not be in a position to treat such a scenario differently from a normal
> QSIG-only scenario.
>
> To be able to inform the Transferee and the Transfer Target about the
> transfer that took place and about the identities of the new parties in
> the call, the options that are available today would seem to be the
> following:
>
> - send a new INVITE with a Replaces header to both the Transferee and
>   the Transfer Target. The destination of the INVITE, however, has to
>   support the Replaces header to avoid undesirable side effects.
>
> - send a re-INVITE, optionally with a Referred-By header, to both the
>   Transferee and the Transfer Target. Although the meaning of
>   "Referred-By" is not that a call transfer took place, subsequent
>   release of the already established sessions (in combination with
>   Referred-by) could be used as a strong hint that call transfer did
>   take place.
>
> - send a REFER to the Transferee (with the address of the GW in the
>   Refer-to URI) and an INVITE with a Replaces header to the Transfer
>   Target.
>
> These methods seem to generate a lot of messages and processing, while
> the only purpose of the establishment of the new sessions is to inform
> both parties about the transfer and/or the identities of the new
> parties.
>
> 2) Unattended transfer.
>
> The Transferor is in QSIG and the Transferee and the Transfer Target 
> are
> in SIP. When the Transferor performs an unattended transfer, it can
> establish a call with the Transfer Target (which also notifies the
> Transfer Target that a transfer is taking place) and notify the
> Transferee about the fact that a transfer is taking place and about the
> progress of this new call. When the new call is established, the
> B-channels of the two calls can be connected at the Transferor. There 
> is
> no way to inform the Transfer Target that this call is the result of a
> transfer. There is no way to inform the Transferree that transfer is
> taking place. The only way to "repair" this situation would seem to be
> the use of the options mentioned in the previous example. Another
> possibility, is that the INVITE to the Transfer Target contains
> history-info with a new reason code to indicate transfer.
>
> Several interworking scenarios between SIP and QSIG for call transfer
> present this sort of problem. In most cases it would be sufficient if 
> it
> were possible to simply send a notification message (i.e. a message 
> that
> does neither modify the characteristics of the session nor its state) 
> to
> the Transferee and/or the Transfer Target. (An example this could
> be a be a SIP INFO message with a P-Asserted-Identity header)
>
> Any comments and/or proposals on the above are welcomed.
>
>
> Regards,
>
> Frank
> _______________________________________________
> Sip mailing list  https://www1.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: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
CC: Robert Sparks <rsparks@dynamicsoft.com>,
        Anders Kristensen <akristensen@dynamicsoft.com>, sip@ietf.org
Subject: Re: [Sip] Constructing the branch parameter at stateless proxies
References: <1038933302.911.62.camel@RjS.localdomain>		<3DEE251A.5080306@dynamicsoft.com>	<1039017937.910.76.camel@RjS.localdomain> 	<3DEE294A.9080806@dynamicsoft.com> <1039019630.910.132.camel@RjS.localdomain> <3DEEFF6F.3000605@dynamicsoft.com>
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Hi,

I was thiking of an scenario where we may have problems:

        +----->S1-----+
        |             v
UAC--->P1             P2--->UAS
        |             ^
        +----->S2-----+

The UAC sends an INVITE. P1 forks to S1 and S2 (both stateless). Both S1
and S2 will route the INVITE to P2 (stateful).

From P2 perspective, both INVITEs will have the same branch ID.
Therefore, P2 will treat the INVITE that arrives later as a
retransmission of the first INVITE.

P2 could be able to differentiate both INVITEs using the received Via
parameter or the port number in the Via header. But P2 is not supposed
to need to include these paramteres when calculating the transaction ID.

In this situation, everything will recover fine (P1 will CANCEL the
branch that does not return a final response), but it is, nevertheless,
an error situation.

Thoughts?

Gonzalo




Jonathan Rosenberg wrote:
> 
> inline.
> 
> Robert Sparks wrote:
> > On Wed, 2002-12-04 at 10:11, Anders Kristensen wrote:
> >
> >>
> >>Robert Sparks wrote:
> >>
> >>>On Wed, 2002-12-04 at 09:54, Anders Kristensen wrote:
> >>>
> >>>
> >>>>Hi Robert,
> >>>>
> >>>>Sounds like a very good idea. One special case to consider is what
> >>>>happens if the request spirals back to the sending proxy (which
> >>>>presumably is stateful) -- either directly from the stateless proxy or
> >>>>via a number of other stateless proxies.  The stateful proxy in this
> >>>>case will have a server transaction and a client transaction with the
> >>>>same id, which might confuse some implementations.
> >>>
> >>>
> >>>This is no different from the same proxy spiraling directly to itself
> >>>is it?
> >>
> >>Well, it's different in that it is NOT spiraling directly to itself and
> >>so is currently guaranteed to see a different branch ID in a spiralled
> >>request. It may be the case that noone depends on this but who knows?
> >
> >
> > Ok - so an implementation _may_ have gone to some effort to ensure
> > that it wasn't sending a message to itself and is counting on not
> > seeing the same transaction identifier come back to it. Not a choice
> > I would recommend, but I can see how someone might have made it.
> >
> > We could avoid this with the suffix mechanism you suggest, or we
> > could add an admonishment to the spec to tell systems to be prepared
> > to have their transaction ID come back. I'd rather do the second.
> 
> I agree with Robert. Over time, our view of stateless proxies has
> changed a lot. Initially, we tried to treat them as the same as stateful
> ones, but it has become clear that there are important differences (for
> example, there are restrictions on the types of routing logic that are
> permitted).
> 
> One of the big changes in bis was the formal definition of a transaction
>   as being between a client transaction in one element and a server
> transaction in another (figure 4). Whats important is that the
> transaction is between transaction-stateful elements (by definition). It
> would seem very logical, then, that the transaction identifier (the
> branch ID) would be the same for all messages in this transaction. Since
> a stateless proxy doesn't muck with the transaction, just forwards
> messages, I would argue that the transaction ID should stay the same.
> 
> Indeed, the analogy is similar to the dialog identifiers in a stateful
> proxy; they too are not modified, since the proxy doesn't act as a
> termination point of the dialog. If it did (b2bua), it would need to
> modify those identifiers.
> 
> >
> > Well, nothing earth-shattering. Right now, just an increased probability
> > that implementation error will lead to protocol failure. Its hard to get
> > this right, and we can greatly simplify it.
> >
> > My growing distaste for what the stateless proxies are currently doing
> > comes from seeing it somewhat like a NAT. The transaction is really
> > between the two stateful elements reguardless of how many stateless
> > elements lie between. Right now, the stateless elements are
> > "translating" the transaction identifier as the message goes through.
> > We haven't identified anything in the protocol that breaks as long
> > as they do this consistently, but I haven't been able to identify any
> > benefit to the behavior.
> 
> I agree. I think its really legacy at this point.
> 
> -Jonathan R.
> --
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
> _______________________________________________
> Sip mailing list  https://www1.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

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo
_______________________________________________
Sip mailing list  https://www1.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: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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Subject: Re: [Sip] Use of Tel URL in To header
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Lawrence,

inline:

"Conroy, Lawrence (SMTP)" wrote:
> 
> Hi Folks,
>   (potentially dumb question warning):
> 
> Section 8.1.1.2 (To Header) of RFC3261 - Use of tel URL.
> 
> I imply from this that the tel URL in a To header can be evaluated at
> "each domain through which the request passes". Example given there -
> dialing "411" at an Airport would be evaluated at the outgoing proxy
> at the Airport, not in the user's home domain.
> 
> Q:: What is the meaning of the "user's home domain" when dialing out
> to a tel URL?

That is what the text in 8.1.1.2 is trying to say. Since the caller does
not have any specific domain in mind, it uses a tel URL.

Alternatively, the caller could have dialed
sip:411@ericsson.com;user=phone

In this situation, 411 would have been evaluated in the domain
ericsson.com

 
> Follow-up:
> When originating a call, one assumes that the INVITE request could go
> via an outgoing proxy.

Yes, an outbound proxy may or may not be used.

> Is this proxy required to route a call onwards
> to a "home server"

No. The outbound proxy, like any other proxy, will make routing
decisions based on the Route header field, the Request-URI, and any
other information the proxy wants (e.g., time, day of the week, caller's
name...).

>, and if so how is that "home server" indicated in
> the INVITE sent to the outgoing proxy?
>

If you want your INVITE to traverse an outbound proxy and your home
proxy you should indicate so in a Route header. The SIP loose routing
mechanisms allow you to do that.
 
> The From header and Contact header indicates the (potential) AoR and
> current addressability of the caller.
> Is the From header used to indicate the home domain for the caller?

The From header is mostly used to identify the caller from a human
perspective. In order to send requests back within the same dialog you
would use the Contact header field. To send requests to the caller
*outside* the dialog you would typically use the Reply-To header field.

Regards,

Gonzalo
_______________________________________________
Sip mailing list  https://www1.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: "Kowsalya Subramanian" <kowsalya@npd.hcltech.com>
To: "Gonzalo Camarillo" <Gonzalo.Camarillo@lmf.ericsson.se>,
        "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: "Robert Sparks" <rsparks@dynamicsoft.com>,
        "Anders Kristensen" <akristensen@dynamicsoft.com>, <sip@ietf.org>
References: <1038933302.911.62.camel@RjS.localdomain>		<3DEE251A.5080306@dynamicsoft.com>	<1039017937.910.76.camel@RjS.localdomain> 	<3DEE294A.9080806@dynamicsoft.com> <1039019630.910.132.camel@RjS.localdomain> <3DEEFF6F.3000605@dynamicsoft.com> <3DFDB100.3476D31C@lmf.ericsson.se>
Subject: Re: [Sip] Constructing the branch parameter at stateless proxies
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Hi Gonzalo,

From my understanding of 16.5, forking will be done by P1 only if its the
edge proxy for the req-uri. The only way the forked request can reach other
proxies is through route headers. so the scenario under question can never
happen or am i missing something?.

Thanks
Kowsalya

----- Original Message -----
From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: Robert Sparks <rsparks@dynamicsoft.com>; Anders Kristensen
<akristensen@dynamicsoft.com>; <sip@ietf.org>
Sent: Monday, December 16, 2002 4:24 PM
Subject: Re: [Sip] Constructing the branch parameter at stateless proxies


> Hi,
>
> I was thiking of an scenario where we may have problems:
>
>         +----->S1-----+
>         |             v
> UAC--->P1             P2--->UAS
>         |             ^
>         +----->S2-----+
>
> The UAC sends an INVITE. P1 forks to S1 and S2 (both stateless). Both S1
> and S2 will route the INVITE to P2 (stateful).
>
> >From P2 perspective, both INVITEs will have the same branch ID.
> Therefore, P2 will treat the INVITE that arrives later as a
> retransmission of the first INVITE.
>
> P2 could be able to differentiate both INVITEs using the received Via
> parameter or the port number in the Via header. But P2 is not supposed
> to need to include these paramteres when calculating the transaction ID.
>
> In this situation, everything will recover fine (P1 will CANCEL the
> branch that does not return a final response), but it is, nevertheless,
> an error situation.
>
> Thoughts?
>
> Gonzalo
>
>
>
>
> Jonathan Rosenberg wrote:
> >
> > inline.
> >
> > Robert Sparks wrote:
> > > On Wed, 2002-12-04 at 10:11, Anders Kristensen wrote:
> > >
> > >>
> > >>Robert Sparks wrote:
> > >>
> > >>>On Wed, 2002-12-04 at 09:54, Anders Kristensen wrote:
> > >>>
> > >>>
> > >>>>Hi Robert,
> > >>>>
> > >>>>Sounds like a very good idea. One special case to consider is what
> > >>>>happens if the request spirals back to the sending proxy (which
> > >>>>presumably is stateful) -- either directly from the stateless proxy
or
> > >>>>via a number of other stateless proxies.  The stateful proxy in this
> > >>>>case will have a server transaction and a client transaction with
the
> > >>>>same id, which might confuse some implementations.
> > >>>
> > >>>
> > >>>This is no different from the same proxy spiraling directly to itself
> > >>>is it?
> > >>
> > >>Well, it's different in that it is NOT spiraling directly to itself
and
> > >>so is currently guaranteed to see a different branch ID in a spiralled
> > >>request. It may be the case that noone depends on this but who knows?
> > >
> > >
> > > Ok - so an implementation _may_ have gone to some effort to ensure
> > > that it wasn't sending a message to itself and is counting on not
> > > seeing the same transaction identifier come back to it. Not a choice
> > > I would recommend, but I can see how someone might have made it.
> > >
> > > We could avoid this with the suffix mechanism you suggest, or we
> > > could add an admonishment to the spec to tell systems to be prepared
> > > to have their transaction ID come back. I'd rather do the second.
> >
> > I agree with Robert. Over time, our view of stateless proxies has
> > changed a lot. Initially, we tried to treat them as the same as stateful
> > ones, but it has become clear that there are important differences (for
> > example, there are restrictions on the types of routing logic that are
> > permitted).
> >
> > One of the big changes in bis was the formal definition of a transaction
> >   as being between a client transaction in one element and a server
> > transaction in another (figure 4). Whats important is that the
> > transaction is between transaction-stateful elements (by definition). It
> > would seem very logical, then, that the transaction identifier (the
> > branch ID) would be the same for all messages in this transaction. Since
> > a stateless proxy doesn't muck with the transaction, just forwards
> > messages, I would argue that the transaction ID should stay the same.
> >
> > Indeed, the analogy is similar to the dialog identifiers in a stateful
> > proxy; they too are not modified, since the proxy doesn't act as a
> > termination point of the dialog. If it did (b2bua), it would need to
> > modify those identifiers.
> >
> > >
> > > Well, nothing earth-shattering. Right now, just an increased
probability
> > > that implementation error will lead to protocol failure. Its hard to
get
> > > this right, and we can greatly simplify it.
> > >
> > > My growing distaste for what the stateless proxies are currently doing
> > > comes from seeing it somewhat like a NAT. The transaction is really
> > > between the two stateful elements reguardless of how many stateless
> > > elements lie between. Right now, the stateless elements are
> > > "translating" the transaction identifier as the message goes through.
> > > We haven't identified anything in the protocol that breaks as long
> > > as they do this consistently, but I haven't been able to identify any
> > > benefit to the behavior.
> >
> > I agree. I think its really legacy at this point.
> >
> > -Jonathan R.
> > --
> > Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> > Chief Scientist                             First Floor
> > dynamicsoft                                 East Hanover, NJ 07936
> > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > http://www.jdrosen.net                      PHONE: (973) 952-5000
> > http://www.dynamicsoft.com
> >
> > _______________________________________________
> > Sip mailing list  https://www1.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
>
> --
> Gonzalo Camarillo         Phone :  +358  9 299 33 71
> Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
> Telecom R&D               Fax   :  +358  9 299 30 52
> FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
> Finland                   http://www.hut.fi/~gonzalo
> _______________________________________________
> Sip mailing list  https://www1.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://www1.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: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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To: Kowsalya Subramanian <kowsalya@npd.hcltech.com>
CC: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Robert Sparks <rsparks@dynamicsoft.com>,
        Anders Kristensen <akristensen@dynamicsoft.com>, sip@ietf.org
Subject: Re: [Sip] Constructing the branch parameter at stateless proxies
References: <1038933302.911.62.camel@RjS.localdomain>		<3DEE251A.5080306@dynamicsoft.com>	<1039017937.910.76.camel@RjS.localdomain> 	<3DEE294A.9080806@dynamicsoft.com> <1039019630.910.132.camel@RjS.localdomain> <3DEEFF6F.3000605@dynamicsoft.com> <3DFDB100.3476D31C@lmf.ericsson.se> <00f601c2a4f6$f9386d10$6f13a8c0@hcltntl.co.in>
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Hi,

to give you a more concrete example:

P1: SIP proxy at ericsson.com
S1: Stateless proxy at cs.columbia.edu
S2: Stateless proxy at lmf.ericsson.se
P2: proxy at hut.fi, where I am giving a speech today.

You send an INVITE to gonzalo@ericsson.com. That proxy will fork the
INVITE to try to reach me at the university (cs.columbia.edu) or at my
office (lmf.ericsson.se). In both places, I have registered myself as
being in hut.fi. There request will reach me at hut.fi

Regards,

Gonzalo

Kowsalya Subramanian wrote:
> 
> Hi Gonzalo,
> 
> From my understanding of 16.5, forking will be done by P1 only if its the
> edge proxy for the req-uri. The only way the forked request can reach other
> proxies is through route headers. so the scenario under question can never
> happen or am i missing something?.
> 
> Thanks
> Kowsalya
> 
> ----- Original Message -----
> From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
> To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
> Cc: Robert Sparks <rsparks@dynamicsoft.com>; Anders Kristensen
> <akristensen@dynamicsoft.com>; <sip@ietf.org>
> Sent: Monday, December 16, 2002 4:24 PM
> Subject: Re: [Sip] Constructing the branch parameter at stateless proxies
> 
> > Hi,
> >
> > I was thiking of an scenario where we may have problems:
> >
> >         +----->S1-----+
> >         |             v
> > UAC--->P1             P2--->UAS
> >         |             ^
> >         +----->S2-----+
> >
> > The UAC sends an INVITE. P1 forks to S1 and S2 (both stateless). Both S1
> > and S2 will route the INVITE to P2 (stateful).
> >
> > >From P2 perspective, both INVITEs will have the same branch ID.
> > Therefore, P2 will treat the INVITE that arrives later as a
> > retransmission of the first INVITE.
> >
> > P2 could be able to differentiate both INVITEs using the received Via
> > parameter or the port number in the Via header. But P2 is not supposed
> > to need to include these paramteres when calculating the transaction ID.
> >
> > In this situation, everything will recover fine (P1 will CANCEL the
> > branch that does not return a final response), but it is, nevertheless,
> > an error situation.
> >
> > Thoughts?
> >
> > Gonzalo
> >
> >
> >
> >
> > Jonathan Rosenberg wrote:
> > >
> > > inline.
> > >
> > > Robert Sparks wrote:
> > > > On Wed, 2002-12-04 at 10:11, Anders Kristensen wrote:
> > > >
> > > >>
> > > >>Robert Sparks wrote:
> > > >>
> > > >>>On Wed, 2002-12-04 at 09:54, Anders Kristensen wrote:
> > > >>>
> > > >>>
> > > >>>>Hi Robert,
> > > >>>>
> > > >>>>Sounds like a very good idea. One special case to consider is what
> > > >>>>happens if the request spirals back to the sending proxy (which
> > > >>>>presumably is stateful) -- either directly from the stateless proxy
> or
> > > >>>>via a number of other stateless proxies.  The stateful proxy in this
> > > >>>>case will have a server transaction and a client transaction with
> the
> > > >>>>same id, which might confuse some implementations.
> > > >>>
> > > >>>
> > > >>>This is no different from the same proxy spiraling directly to itself
> > > >>>is it?
> > > >>
> > > >>Well, it's different in that it is NOT spiraling directly to itself
> and
> > > >>so is currently guaranteed to see a different branch ID in a spiralled
> > > >>request. It may be the case that noone depends on this but who knows?
> > > >
> > > >
> > > > Ok - so an implementation _may_ have gone to some effort to ensure
> > > > that it wasn't sending a message to itself and is counting on not
> > > > seeing the same transaction identifier come back to it. Not a choice
> > > > I would recommend, but I can see how someone might have made it.
> > > >
> > > > We could avoid this with the suffix mechanism you suggest, or we
> > > > could add an admonishment to the spec to tell systems to be prepared
> > > > to have their transaction ID come back. I'd rather do the second.
> > >
> > > I agree with Robert. Over time, our view of stateless proxies has
> > > changed a lot. Initially, we tried to treat them as the same as stateful
> > > ones, but it has become clear that there are important differences (for
> > > example, there are restrictions on the types of routing logic that are
> > > permitted).
> > >
> > > One of the big changes in bis was the formal definition of a transaction
> > >   as being between a client transaction in one element and a server
> > > transaction in another (figure 4). Whats important is that the
> > > transaction is between transaction-stateful elements (by definition). It
> > > would seem very logical, then, that the transaction identifier (the
> > > branch ID) would be the same for all messages in this transaction. Since
> > > a stateless proxy doesn't muck with the transaction, just forwards
> > > messages, I would argue that the transaction ID should stay the same.
> > >
> > > Indeed, the analogy is similar to the dialog identifiers in a stateful
> > > proxy; they too are not modified, since the proxy doesn't act as a
> > > termination point of the dialog. If it did (b2bua), it would need to
> > > modify those identifiers.
> > >
> > > >
> > > > Well, nothing earth-shattering. Right now, just an increased
> probability
> > > > that implementation error will lead to protocol failure. Its hard to
> get
> > > > this right, and we can greatly simplify it.
> > > >
> > > > My growing distaste for what the stateless proxies are currently doing
> > > > comes from seeing it somewhat like a NAT. The transaction is really
> > > > between the two stateful elements reguardless of how many stateless
> > > > elements lie between. Right now, the stateless elements are
> > > > "translating" the transaction identifier as the message goes through.
> > > > We haven't identified anything in the protocol that breaks as long
> > > > as they do this consistently, but I haven't been able to identify any
> > > > benefit to the behavior.
> > >
> > > I agree. I think its really legacy at this point.
> > >
> > > -Jonathan R.
> > > --
> > > Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> > > Chief Scientist                             First Floor
> > > dynamicsoft                                 East Hanover, NJ 07936
> > > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > > http://www.jdrosen.net                      PHONE: (973) 952-5000
> > > http://www.dynamicsoft.com
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.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
> >
> > --
> > Gonzalo Camarillo         Phone :  +358  9 299 33 71
> > Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
> > Telecom R&D               Fax   :  +358  9 299 30 52
> > FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
> > Finland                   http://www.hut.fi/~gonzalo
> > _______________________________________________
> > Sip mailing list  https://www1.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
> >

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo
_______________________________________________
Sip mailing list  https://www1.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
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Subject: Re: [Sip] Constructing the branch parameter at stateless proxies
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Anders Kristensen
	 <akristensen@dynamicsoft.com>, sip@ietf.org
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inline
On Mon, 2002-12-16 at 04:54, Gonzalo Camarillo wrote:
> Hi,
> 
> I was thiking of an scenario where we may have problems:
> 
>         +----->S1-----+
>         |             v
> UAC--->P1             P2--->UAS
>         |             ^
>         +----->S2-----+
> 
> The UAC sends an INVITE. P1 forks to S1 and S2 (both stateless). Both S1
> and S2 will route the INVITE to P2 (stateful).
> 
> >From P2 perspective, both INVITEs will have the same branch ID.
> Therefore, P2 will treat the INVITE that arrives later as a
> retransmission of the first INVITE.

P1 spawned two branches - why would it use the same branch ID
on each of them?

> 
> P2 could be able to differentiate both INVITEs using the received Via
> parameter or the port number in the Via header. But P2 is not supposed
> to need to include these paramteres when calculating the transaction ID.
> 
> In this situation, everything will recover fine (P1 will CANCEL the
> branch that does not return a final response), but it is, nevertheless,
> an error situation.
> 
> Thoughts?
> 
> Gonzalo
> 
> 
> 
> 
> Jonathan Rosenberg wrote:
> > 
> > inline.
> > 
> > Robert Sparks wrote:
> > > On Wed, 2002-12-04 at 10:11, Anders Kristensen wrote:
> > >
> > >>
> > >>Robert Sparks wrote:
> > >>
> > >>>On Wed, 2002-12-04 at 09:54, Anders Kristensen wrote:
> > >>>
> > >>>
> > >>>>Hi Robert,
> > >>>>
> > >>>>Sounds like a very good idea. One special case to consider is what
> > >>>>happens if the request spirals back to the sending proxy (which
> > >>>>presumably is stateful) -- either directly from the stateless proxy or
> > >>>>via a number of other stateless proxies.  The stateful proxy in this
> > >>>>case will have a server transaction and a client transaction with the
> > >>>>same id, which might confuse some implementations.
> > >>>
> > >>>
> > >>>This is no different from the same proxy spiraling directly to itself
> > >>>is it?
> > >>
> > >>Well, it's different in that it is NOT spiraling directly to itself and
> > >>so is currently guaranteed to see a different branch ID in a spiralled
> > >>request. It may be the case that noone depends on this but who knows?
> > >
> > >
> > > Ok - so an implementation _may_ have gone to some effort to ensure
> > > that it wasn't sending a message to itself and is counting on not
> > > seeing the same transaction identifier come back to it. Not a choice
> > > I would recommend, but I can see how someone might have made it.
> > >
> > > We could avoid this with the suffix mechanism you suggest, or we
> > > could add an admonishment to the spec to tell systems to be prepared
> > > to have their transaction ID come back. I'd rather do the second.
> > 
> > I agree with Robert. Over time, our view of stateless proxies has
> > changed a lot. Initially, we tried to treat them as the same as stateful
> > ones, but it has become clear that there are important differences (for
> > example, there are restrictions on the types of routing logic that are
> > permitted).
> > 
> > One of the big changes in bis was the formal definition of a transaction
> >   as being between a client transaction in one element and a server
> > transaction in another (figure 4). Whats important is that the
> > transaction is between transaction-stateful elements (by definition). It
> > would seem very logical, then, that the transaction identifier (the
> > branch ID) would be the same for all messages in this transaction. Since
> > a stateless proxy doesn't muck with the transaction, just forwards
> > messages, I would argue that the transaction ID should stay the same.
> > 
> > Indeed, the analogy is similar to the dialog identifiers in a stateful
> > proxy; they too are not modified, since the proxy doesn't act as a
> > termination point of the dialog. If it did (b2bua), it would need to
> > modify those identifiers.
> > 
> > >
> > > Well, nothing earth-shattering. Right now, just an increased probability
> > > that implementation error will lead to protocol failure. Its hard to get
> > > this right, and we can greatly simplify it.
> > >
> > > My growing distaste for what the stateless proxies are currently doing
> > > comes from seeing it somewhat like a NAT. The transaction is really
> > > between the two stateful elements reguardless of how many stateless
> > > elements lie between. Right now, the stateless elements are
> > > "translating" the transaction identifier as the message goes through.
> > > We haven't identified anything in the protocol that breaks as long
> > > as they do this consistently, but I haven't been able to identify any
> > > benefit to the behavior.
> > 
> > I agree. I think its really legacy at this point.
> > 
> > -Jonathan R.
> > --
> > Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> > Chief Scientist                             First Floor
> > dynamicsoft                                 East Hanover, NJ 07936
> > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > http://www.jdrosen.net                      PHONE: (973) 952-5000
> > http://www.dynamicsoft.com
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.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
> 
> -- 
> Gonzalo Camarillo         Phone :  +358  9 299 33 71
> Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
> Telecom R&D               Fax   :  +358  9 299 30 52
> FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
> Finland                   http://www.hut.fi/~gonzalo


_______________________________________________
Sip mailing list  https://www1.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: The IESG <iesg-secretary@ietf.org>
Date: Mon, 16 Dec 2002 12:58:36 -0500
Subject: [Sip] Protocol Action: The Refer Method to Proposed Standard
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The IESG has approved publication of The Refer Method 
<draft-ietf-sip-refer-07.txt> as a Proposed Standard.

This document is the product of the Session Initiation Protocol Working
Group. The IESG contact persons are Allison Mankin and Scott Bradner.

 
Technical Summary
 
      This document defines the REFER method. This SIP extension requests
      that the recipient REFER to a resource provided in the request. 
	It provides a mechanism allowing the party sending the REFER to be
      notified of the outcome of the referenced request. This can be used
      to enable many applications, including call transfer.

      In addition to the REFER method, this document defines the refer
      event package and the Refer-To request header.

 
Working Group Summary
 
    The working group reached good consensus on this document.
    A header for Referred-by was split out so that it could await
    authenticated identity.
 
Protocol Quality
 
    The document was reviewed for the IESG by Eric Rescorla, Steve
    Bellovin, and Allison Mankin. Implementations of this method
    have been tested in interoperability events by a number of vendors.


RFC Editor

         Please replace each instance of:

                 "atlanta.com" with "atlanta.example.com"
                 "biloxi.com" with "biloxi.example.net"
                 "cleveland.com" with "cleveland.example.org"
                 "denver.com" with "denver.example.org"

         Please add to the end of the document before the Copyright 
	   Statement a copy of the RFC 2026 10.4(A) and 10.4(B) notices 
	   under the heading Intellectual Property Statement.

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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        Anders Kristensen <akristensen@dynamicsoft.com>, sip@ietf.org
Subject: Re: [Sip] Constructing the branch parameter at stateless proxies
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inline.

Robert Sparks wrote:
> inline
> On Mon, 2002-12-16 at 04:54, Gonzalo Camarillo wrote:
> 
>>Hi,
>>
>>I was thiking of an scenario where we may have problems:
>>
>>        +----->S1-----+
>>        |             v
>>UAC--->P1             P2--->UAS
>>        |             ^
>>        +----->S2-----+
>>
>>The UAC sends an INVITE. P1 forks to S1 and S2 (both stateless). Both S1
>>and S2 will route the INVITE to P2 (stateful).
>>
>>>From P2 perspective, both INVITEs will have the same branch ID.
>>Therefore, P2 will treat the INVITE that arrives later as a
>>retransmission of the first INVITE.
> 
> 
> P1 spawned two branches - why would it use the same branch ID
> on each of them?

Yes, I was going to ask that too. It should have used two different 
branch IDs.

Indeed, this is a good example of why not touching the branch ID works 
better. In order for P2 to properly disambiguate the transactions, it 
needs to have S1 and S2 BOTH correctly set them. GIven the ease with 
which this is done improperly, having S1 and S2 not change them reduces 
the chances of a problem.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
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From: Simone Leggio <simone.leggio@cs.helsinki.fi>
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Subject: [Sip] Scalability of the Q-SIP solution
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Hi Luca and all,

in the draft-veltri-sip-qsip-01 "SIP extension for QoS support" it is 
stated as requirement that the resulting architecture should be as 
scalable as possible.
I wonder whether it is possible to reconcile this requirement with the 
functionalities of a Q-SIP stateful proxy server: is such a server 
scalable when it has to manage state information pertaining to many flows?

On the other hand, in case of stateless configuration, what could be the 
increase in size of the SIP message? If it is relevant, problems of UDP 
fragmentation could arise as evidenced in:
draft-willis-sip-congestsafe-00

Regards,

	Simone.

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From: Frank Strauss <strauss@ibr.cs.tu-bs.de>
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Subject: [Sip] Re: Protocol Action: The Refer Method to Proposed Standard
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Hi!

IESG> The IESG has approved publication of The Refer Method 
IESG> <draft-ietf-sip-refer-07.txt> as a Proposed Standard.

Sorry for being late. -- Really late. :-(

I didn't read anything but the abstract. But, wouldn't it be good to
express the relation to SIP in the document title, e.g. simply
"The SIP Refer Method"?

Just a thought. I don't have a strong opinion about this.
If it's too late, just let it move on.

 -frank
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Date: Tue, 17 Dec 2002 11:04:38 +0100
To: Robert Sparks <rsparks@dynamicsoft.com>, sip@ietf.org
From: Jiri Kuthan <jiri.kuthan@fokus.fraunhofer.de>
Subject: Re: [Sip] Constructing the branch parameter at stateless
  proxies
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Do we really need to specify algorithms for branch computing?
Isn't it sufficient to mandate uniquiness within address in Via,
so that downstream elements can use it for transaction matching?
Implementers may decide whether their branch construction features
additional properties such as loop detection, transaction matching
ability and whatsoever.

If you still like to do that, I see a possible problem with
your specific proposal. The spec now promises the branch parameter
to be unique within space of address stated in Via. If a proxy
only reuses branch, it effectively merges two spaces, which may
result in a collision. That would prevent a downstream server
from recognizing two such requests as parts of distinct transactions.

-Jiri

ps -- regarding your motivation to have a globaly unique tid for
the purpose of diagnostics -- can't bottommost via make the job?

At 05:35 PM 12/3/2002, Robert Sparks wrote:
>This list has seen a couple of requests to clarify the language in
>RFC 3261 concerning how a stateless proxy creates a branch identifier.
>
>The current text requires uniqueness, dependency on route decision
>affecting parameters and care to be sure that those parameters are
>retransmission invariant. It goes on to recommend a hash over a
>set of fields (this set has been the subject of the requests for
>clarification).
>
>I think there's a much simpler solution. The purpose of this 
>branch parameter is to act as a transaction identifier. Who
>cares about this identifier? Only the stateful elements at
>each end of the transaction. Any stateless proxies between
>these elements don't use that identifier in any of their
>decisions, so I assert that it is not useful to change it
>as it crosses the stateless proxy.
>
>I propose, then, that a stateless proxy simply copy
>the branch parameter from the topmost Via header field
>value it receives into the Via header field value it adds
>to the request. 
>
>I don't believe it matters that this parameter contain the
>magic cookie. If it doesn't, the next stateful hop will use
>the 2543-style transaction identifiers provided by the sending
>stateful element. The stateless elements are by definition
>transaction transparent.
>
>So - can anyone construct a scenario where this doesn't 
>function correctly? If not, we can remove several convoluted
>paragraphs from the spec and replace them with a very simple
>couple of sentences.
>
>RjS
>
>
>
>_______________________________________________
>Sip mailing list  https://www1.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: Tue, 17 Dec 2002 20:45:00 +0900
From: OKUMURA Shinji <shin@softfront.co.jp>
To: sip@ietf.org
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Subject: [Sip] Query on 'new offer' during INVITE-PRACK transaction
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Hi Folks,

I have one question concerning a new offer inclusion during a transaction of 
INVITE-PRACK at stated in the following figure.

Here my question goes, is it possible to include a new 'offer 2' within 200 
INVITE at line (5) and the corresponding 'answer 2' in ACK at line (6)? 

Thanks in advance for your kind response.

Caller                        Callee
   |                             |
   |                             |
   |(1) INVITE with offer 1      |
   |---------------------------->|
   |                             |
   |                             |
   |(2) 180 INVITE with answer 1 |
   |<----------------------------|
   |    Require: 100rel          |
   |                             |
   |                             |
   |(3) PRACK                    |
   |---------------------------->|
   |                             |
   |                             |
   |(4) 200 PRACK                |
   |<----------------------------|
   |                             |
   |                             |
   |(5) 200 INVITE with offer 2  |
   |<----------------------------|
   |                             |
   |                             |
   |(6) ACK with answer 3        |
   |---------------------------->|
   |                             |
   |                             |

Best regards,
_________________________________________________________
OKUMURA Shinji               E-mail:shin@softfront.co.jp
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Subject: Re: [Sip] Query on 'new offer' during INVITE-PRACK transaction
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Hi,
     " If the initial offer is in an INVITE, the answer MUST be in a
     reliable non-failure message from UAS back to UAC which is
     correlated to that INVITE. For this specification, that is
     only the final 2xx response to that INVITE. That same exact
     answer MAY also be placed in any provisional responses sent
     prior to the answer. * The UAC MUST treat the first session
     description it receives as the answer, and MUST ignore any
     session descriptions in subsequent responses to the initial
     INVITE.* ",

       According to the above excerpt from RFC 3261, you cant send a
*newoffer* in 200 OK.
If an *answer* is sent in any Reliable Provisional Response,then any
session description in subsequent
responses(including 200OK for INVITE) should be ignored.So,though you send
a *newoffer*, your session
description in 200OK will be ignored.

If at all you want to propose a *newoffer*,you should use *Update* for the
same.

regds,
srikanth sundaragopalan





OKUMURA Shinji <shin@softfront.co.jp> on 12/17/2002 05:15:00 PM

To:   sip@ietf.org
cc:    (bcc: Srikanth Sundaragopalan/HSSBLR)

Subject:  [Sip] Query on 'new offer' during INVITE-PRACK transaction




Hi Folks,

I have one question concerning a new offer inclusion during a transaction
of
INVITE-PRACK at stated in the following figure.

Here my question goes, is it possible to include a new 'offer 2' within 200
INVITE at line (5) and the corresponding 'answer 2' in ACK at line (6)?

Thanks in advance for your kind response.

Caller                        Callee
   |                             |
   |                             |
   |(1) INVITE with offer 1      |
   |---------------------------->|
   |                             |
   |                             |
   |(2) 180 INVITE with answer 1 |
   |<----------------------------|
   |    Require: 100rel          |
   |                             |
   |                             |
   |(3) PRACK                    |
   |---------------------------->|
   |                             |
   |                             |
   |(4) 200 PRACK                |
   |<----------------------------|
   |                             |
   |                             |
   |(5) 200 INVITE with offer 2  |
   |<----------------------------|
   |                             |
   |                             |
   |(6) ACK with answer 3        |
   |---------------------------->|
   |                             |
   |                             |

Best regards,
_________________________________________________________
OKUMURA Shinji               E-mail:shin@softfront.co.jp
_______________________________________________
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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




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Use sipping@ietf.org for new developments on the application of sip



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I would like to verify with the mailing list whether negative replies to
ACKs are legal or not. 

Clearly, it makes no use to solicit positive reply for a "good" ACK and 
it is explicitely prohibited to challenge ACKs. However, what about "bad" 
ACKs? Let's say an ACK misses 'CallID' or has a broken r-uri -- should it 
be replied the same way as say BYE/INFO/FOOBAR requests? 

Some implementations do that and it helps upstream clients to understand 
what has gone wrong. Some implementations don't do that.

Can a spec-interpreter share with my an opinion on this?

Thanks,

-Jiri


--
Jiri Kuthan            http://iptel.org/~jiri/ 

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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Jiri Kuthan'" <jiri@iptel.org>, sip@ietf.org
Subject: RE: [Sip] Replies to ACK
Date: Tue, 17 Dec 2002 14:54:03 -0000
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I would have expected this to occur only if the request was so broken that it could not be recognised as an ACK and I am not sure whether this is possible.

I note that the security clauses of RFC 3261 explicitly preclude any challenge (401 or 407) to an ACK.

My understanding of section 17 4th paragraph [The client transaction is also responsible for receiving responses and delivering them to the TU, filtering out any response retransmissions or disallowed responses (such as a response to ACK)].is that if a proxy receives such a response, it would kill it at the client transaction, so the sender would probably not see the response anyway. This would imply to me that there is no advantage to sending a response to ACK even for fault detection purposes.

Keith


Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com 

> -----Original Message-----
> From: Jiri Kuthan [mailto:jiri@iptel.org]
> Sent: 17 December 2002 13:58
> To: sip@ietf.org
> Subject: [Sip] Replies to ACK
> 
> 
> I would like to verify with the mailing list whether negative 
> replies to
> ACKs are legal or not. 
> 
> Clearly, it makes no use to solicit positive reply for a 
> "good" ACK and 
> it is explicitely prohibited to challenge ACKs. However, what 
> about "bad" 
> ACKs? Let's say an ACK misses 'CallID' or has a broken r-uri 
> -- should it 
> be replied the same way as say BYE/INFO/FOOBAR requests? 
> 
> Some implementations do that and it helps upstream clients to 
> understand 
> what has gone wrong. Some implementations don't do that.
> 
> Can a spec-interpreter share with my an opinion on this?
> 
> Thanks,
> 
> -Jiri
> 
> 
> --
> Jiri Kuthan            http://iptel.org/~jiri/ 
> 
> _______________________________________________
> Sip mailing list  https://www1.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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This list is for NEW development of the core SIP Protocol
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From: "Alexeitsev, D" <D.Alexeitsev@telekom.de>
To: Gonzalo.Camarillo@lmf.ericsson.se
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Subject: RE: [Sip] Use of Tel URL in To header
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Hello All

Sorry for the jumping onto the asked question, but I'm really interested why could not it be just said that in case that the caller did not have any preferences the address of the outbound proxy shall be used as hostport? And then it is as in the tel: URI case the issue of the proxy to route it wherever it wants.

like: 

sip:411@outbound.airport.com;user=phone

Greetings,
Denis Alexeitsev

>That is what the text in 8.1.1.2 is trying to say. Since the caller does
>not have any specific domain in mind, it uses a tel URL.
>
>Alternatively, the caller could have dialed
>sip:411@ericsson.com;user=phone
>
>In this situation, 411 would have been evaluated in the domain
>ericsson.com
_______________________________________________
Sip mailing list  https://www1.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
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Hi,

If I understand your suggestion, you are suggesting that we re-phrase
the sentence, right? If this is the case, it is far too late for that
type of change.

Thanks,

Gonzalo

"Alexeitsev, D" wrote:
> 
> Hello All
> 
> Sorry for the jumping onto the asked question, but I'm really interested why could not it be just said that in case that the caller did not have any preferences the address of the outbound proxy shall be used as hostport? And then it is as in the tel: URI case the issue of the proxy to route it wherever it wants.
> 
> like:
> 
> sip:411@outbound.airport.com;user=phone
> 
> Greetings,
> Denis Alexeitsev
> 
> >That is what the text in 8.1.1.2 is trying to say. Since the caller does
> >not have any specific domain in mind, it uses a tel URL.
> >
> >Alternatively, the caller could have dialed
> >sip:411@ericsson.com;user=phone
> >
> >In this situation, 411 would have been evaluated in the domain
> >ericsson.com

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo
_______________________________________________
Sip mailing list  https://www1.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 mailnull@www1.ietf.org  Tue Dec 17 11:09:50 2002
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From: "Alexeitsev, D" <D.Alexeitsev@telekom.de>
To: Gonzalo.Camarillo@lmf.ericsson.se
Cc: sip@ietf.org
Subject: RE: [Sip] Use of Tel URL in To header
Date: Tue, 17 Dec 2002 17:02:04 +0100
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Hello Gonzalo,

No I was not proposing any changes to the baseline, I was just trying to clarify how tel URI is used in the SIP environment.
As the tel URI spec itself does not specify semantics for tel URI it is important to have a clear definition about it's application in SIP environment. So far the only use I could recognise in SIP was to transfer the processing of the userpart of Request-URI to local proxy and this could be achieved by native SIP procedures. 
If this is correct observation then probably some additional clarifications are required concerning the use of tel URI in SIP, that could be realised outside of the RFC 3261

Greetings,
Denis Alexeitsev

-----Original Message-----
From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@lmf.ericsson.se]
Sent: Tuesday, December 17, 2002 4:38 PM
To: Alexeitsev, D
Cc: sip@ietf.org
Subject: Re: [Sip] Use of Tel URL in To header


Hi,

If I understand your suggestion, you are suggesting that we re-phrase
the sentence, right? If this is the case, it is far too late for that
type of change.

Thanks,

Gonzalo

"Alexeitsev, D" wrote:
> 
> Hello All
> 
> Sorry for the jumping onto the asked question, but I'm really interested why could not it be just said that in case that the caller did not have any preferences the address of the outbound proxy shall be used as hostport? And then it is as in the tel: URI case the issue of the proxy to route it wherever it wants.
> 
> like:
> 
> sip:411@outbound.airport.com;user=phone
> 
> Greetings,
> Denis Alexeitsev
> 
> >That is what the text in 8.1.1.2 is trying to say. Since the caller does
> >not have any specific domain in mind, it uses a tel URL.
> >
> >Alternatively, the caller could have dialed
> >sip:411@ericsson.com;user=phone
> >
> >In this situation, 411 would have been evaluated in the domain
> >ericsson.com

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo
_______________________________________________
Sip mailing list  https://www1.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: Christer Holmberg <christer.holmberg@lmf.ericsson.se>
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CC: Gonzalo.Camarillo@lmf.ericsson.se, sip@ietf.org
Subject: Re: [Sip] Use of Tel URL in To header
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Hi Denis,

Please remember that we are allowed to use basically ANY kind of URI in the Request-URI, so even if we would add some clarifying text for TEL-URLs we would have the same routing problem for other URIs.

Regards,

Christer Holmberg
Ericsson Finland


"Alexeitsev, D" wrote:

> Hello Gonzalo,
>
> No I was not proposing any changes to the baseline, I was just trying to clarify how tel URI is used in the SIP environment.
> As the tel URI spec itself does not specify semantics for tel URI it is important to have a clear definition about it's application in SIP environment. So far the only use I could recognise in SIP was to transfer the processing of the userpart of Request-URI to local proxy and this could be achieved by native SIP procedures.
> If this is correct observation then probably some additional clarifications are required concerning the use of tel URI in SIP, that could be realised outside of the RFC 3261
>
> Greetings,
> Denis Alexeitsev
>
> -----Original Message-----
> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@lmf.ericsson.se]
> Sent: Tuesday, December 17, 2002 4:38 PM
> To: Alexeitsev, D
> Cc: sip@ietf.org
> Subject: Re: [Sip] Use of Tel URL in To header
>
> Hi,
>
> If I understand your suggestion, you are suggesting that we re-phrase
> the sentence, right? If this is the case, it is far too late for that
> type of change.
>
> Thanks,
>
> Gonzalo
>
> "Alexeitsev, D" wrote:
> >
> > Hello All
> >
> > Sorry for the jumping onto the asked question, but I'm really interested why could not it be just said that in case that the caller did not have any preferences the address of the outbound proxy shall be used as hostport? And then it is as in the tel: URI case the issue of the proxy to route it wherever it wants.
> >
> > like:
> >
> > sip:411@outbound.airport.com;user=phone
> >
> > Greetings,
> > Denis Alexeitsev
> >
> > >That is what the text in 8.1.1.2 is trying to say. Since the caller does
> > >not have any specific domain in mind, it uses a tel URL.
> > >
> > >Alternatively, the caller could have dialed
> > >sip:411@ericsson.com;user=phone
> > >
> > >In this situation, 411 would have been evaluated in the domain
> > >ericsson.com
>
> --
> Gonzalo Camarillo         Phone :  +358  9 299 33 71
> Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
> Telecom R&D               Fax   :  +358  9 299 30 52
> FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
> Finland                   http://www.hut.fi/~gonzalo
> _______________________________________________
> Sip mailing list  https://www1.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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Hi,

Since the ACK will stop the re-transmission of the final response, if the re-transmissions of the final responses keep coming it does indicate to the UAC that something (even if not exactly what) is wrong.

Regards,

Christer Holmberg
Ericsson Finland

"Drage, Keith (Keith)" wrote:

> I would have expected this to occur only if the request was so broken that it could not be recognised as an ACK and I am not sure whether this is possible.
>
> I note that the security clauses of RFC 3261 explicitly preclude any challenge (401 or 407) to an ACK.
>
> My understanding of section 17 4th paragraph [The client transaction is also responsible for receiving responses and delivering them to the TU, filtering out any response retransmissions or disallowed responses (such as a response to ACK)].is that if a proxy receives such a response, it would kill it at the client transaction, so the sender would probably not see the response anyway. This would imply to me that there is no advantage to sending a response to ACK even for fault detection purposes.
>
> Keith
>
> Keith Drage
> Lucent Technologies
> Tel: +44 1793 776249
> Email: drage@lucent.com
>
> > -----Original Message-----
> > From: Jiri Kuthan [mailto:jiri@iptel.org]
> > Sent: 17 December 2002 13:58
> > To: sip@ietf.org
> > Subject: [Sip] Replies to ACK
> >
> >
> > I would like to verify with the mailing list whether negative
> > replies to
> > ACKs are legal or not.
> >
> > Clearly, it makes no use to solicit positive reply for a
> > "good" ACK and
> > it is explicitely prohibited to challenge ACKs. However, what
> > about "bad"
> > ACKs? Let's say an ACK misses 'CallID' or has a broken r-uri
> > -- should it
> > be replied the same way as say BYE/INFO/FOOBAR requests?
> >
> > Some implementations do that and it helps upstream clients to
> > understand
> > what has gone wrong. Some implementations don't do that.
> >
> > Can a spec-interpreter share with my an opinion on this?
> >
> > Thanks,
> >
> > -Jiri
> >
> >
> > --
> > Jiri Kuthan            http://iptel.org/~jiri/
> >
> > _______________________________________________
> > Sip mailing list  https://www1.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://www1.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
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At 05:56 PM 12/17/2002, Christer Holmberg wrote:

>Hi,
>
>Since the ACK will stop the re-transmission of the final response, if the re-transmissions of the final responses keep coming it does indicate to the UAC that something (even if not exactly what) is wrong.
                                                                   ^^^^^^^^^^^^^^^^^^^^^^^^^
That's the primary argument, I think, some implementers have decided to send replies
to ACKs. (However surprising this technique may sound [to me too], I don't see it 
breaks the spec, except challenges are prohibitted.) An assertation "the r-uri which 
made it to me is bad" is certainly much more useful than one-bit diagnostic assertation 
"something is wrong".

-Jiri 

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Date: Tue, 17 Dec 2002 19:16:57 -0500
From: Bert Culpepper <bertculpepper@netscape.net>
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Standardization of mechanisms on sending DTMF in INFO has not occurred. 
    For SIP, the use of DTMF is seen as a means for user interaction 
with network-based applications.  To this end, work is ongoing on to 
define a framework and mechanisms for users to interact with 
applications in SIP.  See the I-Ds below for how this is expected to be 
supported.

http://www.ietf.org/internet-drafts/draft-culpepper-sipping-app-interact-reqs-02.txt
http://www.jdrosen.net/papers/draft-rosenberg-sipping-app-interaction-framework-00.txt
http://www.ietf.org/internet-drafts/draft-burger-sipping-kpml-00.txt
http://www.ietf.org/internet-drafts/draft-jennings-sip-app-info-00.txt

Regards,
Bert

sathish kumar wrote:

> Hi,
> 
> Pertaining to the SIP INFO method defined in RFC 2976 used for sending 
> mid call signaling information.
> 
> After establishment of Voice path between two user agents i wish to send 
> the DTMF digits from UAC to UAS.
> 
> 1. Can anyone tell me the format of sending the information (Header, 
> Message Body if any).
> 
> thanx and regards
> 
> sathises.
> 
> 
> ------------------------------------------------------------------------
> Do you Yahoo!?
> Yahoo! Mail Plus 
> <http://rd.yahoo.com/mail/mailsig/*http://mailplus.yahoo.com> - 
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Subject: [Sip] Contact in REGISTER
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Dear all,

Suppose there is no binding at all for an Address-of-Record, and
the Registrar receives a REGISTER message with "Contact: *" for 
that particular address-of-record, how should the Registrar handle 
it?  Likewise, if the Contact header is missing, what should the 
Registrar do?  

In both cases, there is no additional contact information carried
in the REGISTER message.  So effectively the Registrar still doesn't
know the contact binding of the Address-of-Record.

Thanks.


regards,
KJ Loh
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From: "Ravi Shiroor" <rshiroor@aplion.stpn.soft.net>
To: "Jiri Kuthan" <jiri@iptel.org>, <sip@ietf.org>
Subject: RE: [Sip] Replies to ACK
Date: Wed, 18 Dec 2002 09:15:25 +0530
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ACK is a part of the three way handshake unlike other methods.
So I think it would not need a negetive reply to it.

Implementing a negetive reply to help an upstream client can
only be a debugging mechanism, but what then if the negetive
reply gets broken? Should we have a negetive reply for a broken
negetive reply?

regards,
ravi.

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Jiri
> Kuthan
> Sent: Tuesday, December 17, 2002 7:28 PM
> To: sip@ietf.org
> Subject: [Sip] Replies to ACK
>
>
> I would like to verify with the mailing list whether negative replies to
> ACKs are legal or not.
>
> Clearly, it makes no use to solicit positive reply for a "good" ACK and
> it is explicitely prohibited to challenge ACKs. However, what about "bad"
> ACKs? Let's say an ACK misses 'CallID' or has a broken r-uri -- should it
> be replied the same way as say BYE/INFO/FOOBAR requests?
>
> Some implementations do that and it helps upstream clients to understand
> what has gone wrong. Some implementations don't do that.
>
> Can a spec-interpreter share with my an opinion on this?
>
> Thanks,
>
> -Jiri
>
>
> --
> Jiri Kuthan            http://iptel.org/~jiri/
>
> _______________________________________________
> Sip mailing list  https://www1.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] Contact in REGISTER
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Hi Loh,
     In both cases, the registrar should respond
with a 200 OK containing no Contacts. Whenever a
registrar responds to a REGISTER request it includes
the bindings that are currently known to it in the
response. (The second case that you refer to, is
equivalent to a query to the registrar to fetch the
current bindings).

Regards,
Subhash.
P.S.> Implementation related questions can be addressed
to sip-implementors@cs.columbia.edu





Loh Kok Jeng <kokjeng.loh@siemens.com> on 12/18/2002 07:10:49 AM

To:   sip@ietf.org
cc:    (bcc: Subhash Ullal Nayak/HSSBLR)

Subject:  [Sip] Contact in REGISTER




Dear all,

Suppose there is no binding at all for an Address-of-Record, and
the Registrar receives a REGISTER message with "Contact: *" for
that particular address-of-record, how should the Registrar handle
it?  Likewise, if the Contact header is missing, what should the
Registrar do?

In both cases, there is no additional contact information carried
in the REGISTER message.  So effectively the Registrar still doesn't
know the contact binding of the Address-of-Record.

Thanks.


regards,
KJ Loh
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This list is for NEW development of the core SIP Protocol
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To: "'Jiri Kuthan'" <jiri@iptel.org>,
        Christer Holmberg
	 <christer.holmberg@lmf.ericsson.se>,
        "Drage, Keith (Keith)"
	 <drage@lucent.com>
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Subject: RE: [Sip] Replies to ACK
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Remember we do not tend to write standards stating what is precluded.

There is an implicit assumption that what is not allowed is precluded. Certainly any equipment claiming conformance to RFC 3261 should list this as an exception, like any other extensions, as it is not explicitly required by RFC 3261.

I would also still question the use. There seems to be little point in responding when as far as I read RFC 3261, any intermediate proxy would kill the incoming response message at the client process, and not pass it up to the transaction layer for further processing. The sort of error detection you are talking about would need to be done by the sender of the ACK, which would be this proxy only for ACK following non 2xx final responses.

Keith

Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com 

> -----Original Message-----
> From: Jiri Kuthan [mailto:jiri@iptel.org]
> Sent: 17 December 2002 17:37
> To: Christer Holmberg; Drage, Keith (Keith)
> Cc: sip@ietf.org
> Subject: Re: [Sip] Replies to ACK
> 
> 
> At 05:56 PM 12/17/2002, Christer Holmberg wrote:
> 
> >Hi,
> >
> >Since the ACK will stop the re-transmission of the final 
> response, if the re-transmissions of the final responses keep 
> coming it does indicate to the UAC that something (even if 
> not exactly what) is wrong.
>                                                               
>      ^^^^^^^^^^^^^^^^^^^^^^^^^
> That's the primary argument, I think, some implementers have 
> decided to send replies
> to ACKs. (However surprising this technique may sound [to me 
> too], I don't see it 
> breaks the spec, except challenges are prohibitted.) An 
> assertation "the r-uri which 
> made it to me is bad" is certainly much more useful than 
> one-bit diagnostic assertation 
> "something is wrong".
> 
> -Jiri 
> 
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Date: Wed, 18 Dec 2002 20:10:17 +0900
From: OKUMURA Shinji <shin@softfront.co.jp>
Subject: Re: [Sip] Query on 'new offer' during INVITE-PRACK transaction
To: ssundaragopalan@hss.hns.com, Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Hi Srikanth,

Thank you very much for your response. I have gotten the answer but I wish 
to understand the reason behind for adopting such regulation. For example, 
consider the following two transaction scenarios side by side.


Caller                        Callee    Caller                        Callee
   |                             |         |                             |
   |                             |         |                             |
   |(1) INVITE with offer 1      |         |(1) INVITE with offer 1      |
   |---------------------------->|         |---------------------------->|
   |                             |         |                             |
   |                             |         |                             |
   |(2) 180 INVITE with answer 1 |         |(2) 180 INVITE with answer 1 |
   |<----------------------------|         |<----------------------------|
   |    Require: 100rel          |         |    Require: 100rel          |
   |                             |         |                             |
   |                             |         |                             |
   |(3) PRACK                    |         |(3) PRACK                    |
   |---------------------------->|         |---------------------------->|
   |                             |         |                             |
   |                             |         |                             |
   |(4) 200 PRACK                |         |(4) 200 PRACK                |
   |<----------------------------|         |<----------------------------|
   |                             |         |                             |
   |                             |         |                             |
   |(5) UPDATE with offer 2      |         |                             |
   |<----------------------------|         |                             |
   |                             |         |                             |
   |                             |         |                             |
   |(6) 200 UPDATE with answer 2 |         |                             |
   |---------------------------->|         |                             |
   |                             |         |                             |
   |                             |         |                             |
   |(7) 200 INVITE               |         |(5) 200 INVITE with offer 2  |
   |<----------------------------|         |<----------------------------|
   |                             |         |                             |
   |                             |         |                             |
   |(8) ACK                      |         |(6) ACK with answer 2        |
   |---------------------------->|         |---------------------------->|
   |                             |         |                             |

             Figure 1-A                             Figure 1-B


IMHU, both of these two processes can solve the same problem whereas in the
process of Figure 1-B, it looks more optimized. Then why this second process
could not be accepted? What possible problem does it cause?

Looking for any comment.

Best regards.

ssundaragopalan@hss.hns.com wrote in <65256C92.0042D70D.00@sampark.hss.hns.com>
>Hi,
>     " If the initial offer is in an INVITE, the answer MUST be in a
>     reliable non-failure message from UAS back to UAC which is
>     correlated to that INVITE. For this specification, that is
>     only the final 2xx response to that INVITE. That same exact
>     answer MAY also be placed in any provisional responses sent
>     prior to the answer. * The UAC MUST treat the first session
>     description it receives as the answer, and MUST ignore any
>     session descriptions in subsequent responses to the initial
>     INVITE.* ",
>
>       According to the above excerpt from RFC 3261, you cant send a
>*newoffer* in 200 OK.
>If an *answer* is sent in any Reliable Provisional Response,then any
>session description in subsequent
>responses(including 200OK for INVITE) should be ignored.So,though you send
>a *newoffer*, your session
>description in 200OK will be ignored.
>
>If at all you want to propose a *newoffer*,you should use *Update* for the same.
>
>regds,
>srikanth sundaragopalan
>
>OKUMURA Shinji <shin@softfront.co.jp> on 12/17/2002 05:15:00 PM
>To:   sip@ietf.org
>cc:    (bcc: Srikanth Sundaragopalan/HSSBLR)
>Subject:  [Sip] Query on 'new offer' during INVITE-PRACK transaction
>
>Hi Folks,
>
>I have one question concerning a new offer inclusion during a transaction of
>INVITE-PRACK at stated in the following figure.
>
>Here my question goes, is it possible to include a new 'offer 2' within 200
>INVITE at line (5) and the corresponding 'answer 2' in ACK at line (6)?
>
>Thanks in advance for your kind response.
>
>Caller                        Callee
>   |                             |
>   |                             |
>   |(1) INVITE with offer 1      |
>   |---------------------------->|
>   |                             |
>   |                             |
>   |(2) 180 INVITE with answer 1 |
>   |<----------------------------|
>   |    Require: 100rel          |
>   |                             |
>   |                             |
>   |(3) PRACK                    |
>   |---------------------------->|
>   |                             |
>   |                             |
>   |(4) 200 PRACK                |
>   |<----------------------------|
>   |                             |
>   |                             |
>   |(5) 200 INVITE with offer 2  |
>   |<----------------------------|
>   |                             |
>   |                             |
>   |(6) ACK with answer 3        |
>   |---------------------------->|
>   |                             |
>   |                             |

_________________________________________________________
OKUMURA Shinji               E-mail:shin@softfront.co.jp
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Just to be safe -- I'm not advocating ACK/replies, I'm trying to
understand a potential gap betweeen running code I sighted, 
specification and its interpretations. I simply wish to know
whether it is a bug or feature, which is not apparent to me
from the spec.

At 12:09 PM 12/18/2002, Drage, Keith (Keith) wrote:
>Remember we do not tend to write standards stating what is precluded.
>
>There is an implicit assumption that what is not allowed is precluded. 
>Certainly any equipment claiming conformance to RFC 3261 should list this as an exception, like any other extensions, as it is not explicitly required by RFC 3261.

What implementers of the device in question might have seen in RFC3261
is "...each transaction consists of a reuqest...and at least one response..." 
(Section 4) and "...ACK...considered its own transaction" (Section 17).

>I would also still question the use. There seems to be little point in responding when as far as I read RFC 3261, any intermediate proxy would kill the incoming response message at the client process, and not pass it up to the transaction layer for further processing. 

I read the spec slightly differently. Reply matching succeeds if both branch 
and CSeq method matches (you would not be able to distinguish replies to CANCEL 
and INVITEs otherwise). If a device sends a reply to an ACK, the reply will be 
forwarded upstream statelessly. Besides that, such a reply does not have to visit 
a proxy at all, as it follows ACK path, which may be direct.

-Jiri 

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        Christer Holmberg <christer.holmberg@lmf.ericsson.se>, sip@ietf.org
Subject: RE: [Sip] Replies to ACK
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Jiri Kuthan writes:

 > Just to be safe -- I'm not advocating ACK/replies, I'm trying to
 > understand a potential gap betweeen running code I sighted, 
 > specification and its interpretations. I simply wish to know
 > whether it is a bug or feature, which is not apparent to me
 > from the spec.

i looked every place where rfc3261 mentions ack and didn't mention
anywhere if a proxy is allowed to respond to an ack it doesn't like.  my
interpretation of rfc3261 is that an ack to 2xx response is routed
exactly like any other request, which means, for example, that if a
proxy gets such an ack that contains a request uri it doesn't for some
reason like, it is allowed to respond with, for example, 403 forbidden.

since this clearly is unclear, i would suggest that whoever keeps an bug
list on rfc3261, adds this issue there to be clarified one way or
another in the next revision.

-- juha

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From mailnull@www1.ietf.org  Wed Dec 18 17:15:37 2002
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Subject: RE: [Sip] Replies to ACK
From: Robert Sparks <rsparks@dynamicsoft.com>
To: jh@lohi.eng.song.fi
Cc: Jiri Kuthan <jiri@iptel.org>, "Drage, Keith (Keith)" <drage@lucent.com>,
        Christer Holmberg <christer.holmberg@lmf.ericsson.se>, sip@ietf.org
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I've added this to the list of bugs being tracked as bug #684.

It has always been the intent that no element ever generate
a response to an ACK. 

Here is some evidence of that intent:

From the first paragraph of page 124  (this alone addresses
what stateful proxies are going to do with responses to ACKs):

  The client transaction is also responsible for receiving responses
  and delivering them to the TU, filtering out any response
  retransmissions or disallowed responses (such as a response to ACK).

From page 130:

  Although any request MAY contain a body, a body in an ACK is special
  since the request cannot be rejected if the body is not understood.

From page 51 
 o  A stateless UAS MUST ignore ACK requests.

There are probably more, but I think those make the point

RjS


On Wed, 2002-12-18 at 07:29, jh@lohi.eng.song.fi wrote:
> Jiri Kuthan writes:
> 
>  > Just to be safe -- I'm not advocating ACK/replies, I'm trying to
>  > understand a potential gap betweeen running code I sighted, 
>  > specification and its interpretations. I simply wish to know
>  > whether it is a bug or feature, which is not apparent to me
>  > from the spec.
> 
> i looked every place where rfc3261 mentions ack and didn't mention
> anywhere if a proxy is allowed to respond to an ack it doesn't like.  my
> interpretation of rfc3261 is that an ack to 2xx response is routed
> exactly like any other request, which means, for example, that if a
> proxy gets such an ack that contains a request uri it doesn't for some
> reason like, it is allowed to respond with, for example, 403 forbidden.
> 
> since this clearly is unclear, i would suggest that whoever keeps an bug
> list on rfc3261, adds this issue there to be clarified one way or
> another in the next revision.
> 
> -- juha
> 
> _______________________________________________
> Sip mailing list  https://www1.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 mailnull@www1.ietf.org  Thu Dec 19 03:45:52 2002
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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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Folks,

I have just submitted the draft below:

http://standards.ericsson.net/gonzalo/papers/draft-camarillo-sip-parameter-registry-00.txt

It defines a new IANA registry for SIP parameters. If you know of any
RFC or Internet draft that defines header field parameters or SIP/SIPS
URI parameters, please let me know. Answer directly to me, do NOT reply
to all the mailing lists.

The specs we have taken into account are RFC 3261, comp=sigcomp, rport,
3GPP P-headers and caller preferences... but we might have missed
something.

Thanks,

Gonzalo
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Subject: Re: [Sip] Replies to ACK
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Robert Sparks wrote:
> I've added this to the list of bugs being tracked as bug #684.
> 
> It has always been the intent that no element ever generate
> a response to an ACK. 

Robert is absolutely correct.

I believe there were explicit statements to this regard in bis 
revisions, but this is something that got lost. I suspect it got lost 
because it was always obvious in the minds of the editors and thus 
somehow never got stated...

Jiri has claimed that they are "harmless". However, in practice, we 
found that they confused implementations. Indeed, according to the 
transaction matching rules, an ACK for non-2xx has the same branch ID as 
the INVITE. As such, the response to ACK also has this same branch ID. 
This will cause the INVITE state machine to consider that response as a 
response to the original INVITE (CSeq method is not needed in the 
matching process!). This will trigger an ACK retransmit, which triggers 
an ACK response, which triggers an ACK retransmit, etc. The result is a 
message storm.

Thus, sending an ACK response is a BAD THING.

Robert has already logged a bug against it.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From: FERRIERI Antonio <Antonio.FERRIERI@vodafoneomnitel.it>
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Subject: [Sip] INFO Contact
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Folks,
I'm a new entry in SIP world and I've a question on Contact Header:

 Does a Registar bind the Request-URI, in a Register message, to the URI in
Contact?
So, is not a Request-URI binded to the IP address in VIA field?
and If the contact Header is absent, what happens?


Thanks,
Antonio


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From: Jiri Kuthan <jiri@iptel.org>
Subject: Re: [Sip] Replies to ACK
Cc: jh@lohi.eng.song.fi, "Drage, Keith (Keith)" <drage@lucent.com>,
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At 04:07 PM 12/19/2002, Jonathan Rosenberg wrote:
>Jiri has claimed that they are "harmless". 

I'm not sure I ever did. Never mind.

[...]

>This will cause the INVITE state machine to consider that response as 
>a response to the original INVITE (CSeq method is not needed in the matching process!). 

Is that some implicit assumption too? I read the following in spec,
section 17.1.3: "A response matches ... If the method parameter in the 
CSeq header field matches the method of the request that created the 
transaction." It does not mention any special case, in which method 
matching is not used. Which is IMHO correct.

>This will trigger an ACK retransmit, which triggers an ACK response, which triggers an ACK retransmit, etc. The result is a message storm.
>
>Thus, sending an ACK response is a BAD THING.
>
>Robert has already logged a bug against it.

Thanks -- I think explicitely declaring replies to ACKs as illegal
would add clarity. 

-Jiri  

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Session Initiation Protocol Extension for Response 
                          Integrity Check using Validation Cookie
	Author(s)	: S. Shenoy
	Filename	: draft-shenoy-sip-via-validation-00.txt
	Pages		: 7
	Date		: 2002-12-18
	
This document defines an extension to the Session Initiation Protocol
(SIP). This extension allows detection of stale responses and 
responses whose Via header (SIP [1]) have been modified. This memo 
suggests new processing rules at the proxy servers while forwarding
requests and responses. The extension works by defining a new 
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From mailnull@www1.ietf.org  Fri Dec 20 14:42:18 2002
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From: "Cullen Jennings" <fluffy@cisco.com>
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Subject: [Sip] Enrollment in SIP services (and should we undeprecate basic auth)
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I have been thinking about how devices can easily enroll to some sip
service such as a registrar. In some cases it will be complex and
require credit cards and web pages but in some cases it could be very
simple. Imagine someone was offering a basic service and not directly
charging for it. It would be nice if I could send it a message and it
would create an account for me. 

It seems that a I could send a REGISTER to a registrar. If the registrar
had never heard of my user name before, it could just create it as a new
account. However, I would need to somehow provide it with the password I
wish to use. Using basic authentication over a TLS connection to the
registrar would be a nice way to do this. 

I'm wondering if we should undeprecate basic authentication. Is there
better ways to meet this requirement? Are there other uses for basic
that would support it coming back? 

Cullen

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From mailnull@www1.ietf.org  Fri Dec 20 15:00:17 2002
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Date: Fri, 20 Dec 2002 14:56:23 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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CC: sip@ietf.org
Subject: Re: [Sip] Enrollment in SIP services (and should we undeprecate basic
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User provisioning (aka enrollment) is not the role of the SIP REGISTER 
request. It usually involves far more than just providing a 
username/password, and for many reasons REGISTER is a poor fit (and 
ultimately, was a poor choice of names because of its (ab)use for things 
that seemed like "registrations"). Genreally I like web-based enrollment 
in applications.

I have been thinking about undeprecating basic, but for a totally 
different reason (which was suggested by Christian some time back). When 
you do authentication over TLS, there is no security benefits to digest 
as opposed to basic. However, with basic, it is easier to integrate with 
back-end AAA systems, because you don't need to assume that they've 
stored the password in the form described in rfc 2617.

Of course, if the server doing the authentication is several hops beyond 
the one you've terminated your TLS connection to, you've got problems. 
The combination of sips and basic would always work, however.

-Jonathan R.

Cullen Jennings wrote:
> I have been thinking about how devices can easily enroll to some sip
> service such as a registrar. In some cases it will be complex and
> require credit cards and web pages but in some cases it could be very
> simple. Imagine someone was offering a basic service and not directly
> charging for it. It would be nice if I could send it a message and it
> would create an account for me. 
> 
> It seems that a I could send a REGISTER to a registrar. If the registrar
> had never heard of my user name before, it could just create it as a new
> account. However, I would need to somehow provide it with the password I
> wish to use. Using basic authentication over a TLS connection to the
> registrar would be a nice way to do this. 
> 
> I'm wondering if we should undeprecate basic authentication. Is there
> better ways to meet this requirement? Are there other uses for basic
> that would support it coming back? 
> 
> Cullen
> 
> _______________________________________________
> Sip mailing list  https://www1.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.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
Sip mailing list  https://www1.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 mailnull@www1.ietf.org  Fri Dec 20 15:25:09 2002
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Subject: Re: [Sip] Enrollment in SIP services (and should we undeprecate
	basic auth)
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: Cullen Jennings <fluffy@cisco.com>, sip@ietf.org
In-Reply-To: <3E0375E7.1080608@dynamicsoft.com>
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	<3E0375E7.1080608@dynamicsoft.com>
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I see no value at all to resurrecting BASIC.

If you are wanting to provide a shared secret and punt all of the
security responsibilities to the transport layer, then express that
secret in its native form (plaintext for a password). Anything else
you do wastes cycles and encourages people to make bad assumptions
about the level of protection they are getting (increasing the risk
that it would get foolishly used outside the environment you intended
to punt to).


More inline:

On Fri, 2002-12-20 at 13:56, Jonathan Rosenberg wrote:
> User provisioning (aka enrollment) is not the role of the SIP REGISTER 
> request. It usually involves far more than just providing a 
> username/password, and for many reasons REGISTER is a poor fit (and 
> ultimately, was a poor choice of names because of its (ab)use for things 
> that seemed like "registrations"). Genreally I like web-based enrollment 
> in applications.
> 
> I have been thinking about undeprecating basic, but for a totally 
> different reason (which was suggested by Christian some time back). When 
> you do authentication over TLS, there is no security benefits to digest 
> as opposed to basic. However, with basic, it is easier to integrate with 
> back-end AAA systems, because you don't need to assume that they've 
> stored the password in the form described in rfc 2617.
I don't understand the last statement. A system that accepts digest
credentials is not required to keep its local knowledge of the password
in the hash recommended in 2617.
> 
> Of course, if the server doing the authentication is several hops beyond 
> the one you've terminated your TLS connection to, you've got problems. 
> The combination of sips and basic would always work, however.
> 
> -Jonathan R.
> 
> Cullen Jennings wrote:
> > I have been thinking about how devices can easily enroll to some sip
> > service such as a registrar. In some cases it will be complex and
> > require credit cards and web pages but in some cases it could be very
> > simple. Imagine someone was offering a basic service and not directly
> > charging for it. It would be nice if I could send it a message and it
> > would create an account for me. 
> > 
> > It seems that a I could send a REGISTER to a registrar. If the registrar
> > had never heard of my user name before, it could just create it as a new
> > account. However, I would need to somehow provide it with the password I
> > wish to use. Using basic authentication over a TLS connection to the
> > registrar would be a nice way to do this. 
> > 
> > I'm wondering if we should undeprecate basic authentication. Is there
> > better ways to meet this requirement? Are there other uses for basic
> > that would support it coming back? 
> > 
> > Cullen
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.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.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
> _______________________________________________
> Sip mailing list  https://www1.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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We don't seem to have all of the slides that were used at IETF 55 posted at:

http://www.softarmor.com/sipwg/meets/ietf55/slides/

Please check and see if yours are there, if you used any.

If not, get them to me ASAP.

Thanks,

--
Dean

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Also posted at:
http://www.softarmor.com/sipwg/meets/ietf55/minutes/minutes.html

Edited by Dean Willis dean.willis@softarmor.com 
Notes recorded by Andrew Bender and Mary Barnes and Vijay Gurbani

Meeting held Tuesday November 19, 1930-2200 in Atlanta, GA, USA

Agenda Bash
--------------

No objections raised

Status and Charter, Chairs
---------------------------

New co-chairs Rohan Mahy and Jon Peterson introduced.  Status of work items
reviewed by Rohan Mahy.  11 of 19 charter items reported complete. New RFCs
issued include 3310, 3311, 3312, 3361. 

Identity and Privacy and AuthID Open Issues, Jon Peterson
------------------------------------------------------------

Draft-peterson-sip-identity-00 split into 2 WG documents, which were
discussed seperately.
 
Document draft-ietf-sip-authid-body-00.txt
--------------------------------------------

Status:  A specification of a MIME body that could be used as an
authentication token. Now relies on either message/sip or message/sigfrag 

Discussion: There was discussion around whether Call-id should always be a
must as there are scenarios (Referred-by and 3pcc) that need a flexible
mechanism for correlation. There was a proposal that it  should be Call-id
unless you provide another mechanism for replay protection. There was
discussion and agreement that it is always deterministic and clear when you
don't want to include call-id.  There was also discussion around the
vulnerabilities of NOT sending the Contact header, with the suggestion that
if the AIB is intended for use inside a Dialog, then a Contact should be a
MUST. 

Conclusion:  Jon will address this in the draft. 

Document draft-ietf-sip-identity-00.txt
---------------------------------------
Status:  Now defines status codes and practices for providing an
authentication token (which might be auth-id body, or could be something
else).  An AuthService authenticates user agents, creates some sort of
authentication token (as a MIME body), adds token to messages.  Document
describes 3 ways that body can be added to requests:  1) (Essentially)
Redirection (push body back to UA), 2) Auth Service acts as B2BUA, and the
optimal mechanism 3) Content-Indirection  

Discussion:

Point 1: Why a header can't be used for the mime body.  Given the S/MIME
direction it really has to be a body. 

 Point 2: Long term identity versus short term identity solutions (as
described in the past). It was questioned as to how the multiple identities
in the "short term" document would be supported.  It was discussed that the
support of this has to do  with how the token is structured.  There are 6
headers, 3 musts, 3 shoulds and optionals. It wasn't believed that it's
necessary to define the Multiple identities explicitly in this draft.  But,
this discussion is a reasonable one to have.  The concern expressed was that
although there are multiple identity headers, there aren't clearly defined
semantics.   Proposal: Discussion of this topic should continue on the list.


 Point 3: Normative support for AIB.  It was agreed that a normative
statement is needed for interoperability.  You need to define the scope if
it's negotiable, etc.  This gets into the downgrade attack issues, etc.
There was concern that this might limit extensibility of the mechanism,
however, Allison highlighted that the minimum mandatory to implement should
be specified and that should not limit SIP's future.  

 
Document draft-peterson-sip-smime-aes-00.txt
-------------------------------------------------

Status:  Recommends changing required encryption for the SIP profile of
S/MIME from 3DES to AES.  Why deal with this now? Because S/MIME for SIP has
yet to catch on, better to fix now.  Lower footprint of AES might also help
foster S/MIME adoption.  Additonal advantage: Same encryption algorithm as
TLS (AES). 
 
Discussion:  There was a proposal to add this as a 3261 bis bug, but it was
highlighted that this is of much greater scope than a typical 3261 bug.
There was also a concern raised over obsoleting parts of  an RFC, however,
Allison highlighted that it's an accepted practice to update parts of RFCs
with another RFC.  It was suggested that occasionally writing a roadmap
makes navigating the RFCs easier.  

Conclusion: Seemed to be general WG agreement for this to be a WG document. 

Content Indirection Open Issues, Sean Olson
----------------------------------------------

Issue: use of size parameter in content-type - this can be used instead of
content length header richer negotiation of indirection per mime type
security issues- sips/smime, access control, revocation / invalidation
Proposal(s)-
 * use standard size parameter*
 * recommend s/mime and or tls
 * acl and negotiation are out of scope for this document... should be
   solved for sip in general
 * suggest wglc

Discussion: why do we need size? Should it be madatory or optional?
Conclusion: should make optional, as it may not always be possible to know
the real size at the time of the indirection. Issue WGLC 2 weeks after
changes published if there are no objections on list.


RFC3261 Open Issues Jonathan Rosenberg
----------------------------------------------------------------

Group decided not to discuss at this time as the author indicated all open
issues had been adeuqately discussed on-list.


Caller Preferences Open Issues,  Jonathan Rosenberg
------------------------------------------------------

Slides presented on open issues. Briefly reviewed Changes since last
revision with the biggest change being the  Require-Contact. A draft on
related scenarios should be published shortly.

Issue#1: Forced Parameter
If a UA doesn't say anything about a parameter, it still matches a rule with
that parameter.  
Proposal: Add a parm to Require Contact rule that explicitly says whether it
MUST match.  No arguments against proposal raised.

Issue #2 AND within single feature tag
In some cases, you want to specifiy that a UA has to support multiple values
for the same feature tag. Current mechanism is to use multiple values of the
Require-Contact header field
Proposal: keep as is. No arguments raised against proposal. Some discussion
held about differences between multiple occurences of the same value vs.
multiple values.
 

Issue #3: Weight q-values based on amount of overlap 
Proposal: develop q-value algorithm which weighs based on amount of overlap;
can be done in RFC 2533 framework. 
Discussion indicated that we need to develop a metric for the amount of
overlap, and that in order to prevent spiral problems where each parameter
has a differnet q-value, we must compute q-values automatically based on the
amount of overlap. There was also discussion on exact match as a limiting
case. The argument was made that whereas RFC2533 implies that partial
matching is impractical in a general case, we believe we have enough
constraints to be able to solve the problem within the given scenarios.

Issue #4: Merge Require, Accept Contacts
Require and Accept Contact are similar 
Proposal: Merge back into single Accept-Contact header field and define
should-match parm which if contact predicate doesn't match, causes it to
drop. No arguments raised against proposal.
Discussion: Similar to Issue #1. 

Issue #5: No one cares
There has been little feedback on this draft given that it is widely
referencec.
Proposal: Rewrite intro to give it a bit more spark, explain better the
power of the spec and that it is truly providing "one number" service. 
Discussion: lack of comments does not equate to lack of interest. It may be
that this is just not contentious anymore.

Issue #6: Priority semantics
Proposal:  work through use cases and express in draft. 
Discussion:  priority semantics may be out of scope, since this is really a
callee rather than caller issue. Proposed that this should handled through
CPL.

 
Issue #7: Implicit compatibility mechanism
Skipped.

Issue #8: Escaping
Sadly, RFC 2533 syntax for feature tags uses characters not permitted in
token (slash and colon). These characters are in use. 
Proposal: Use characters allowed in token, but not in feature tag, to
represent those characters; bang gets mapped to colon, single quote gets
mapped to slash. 
Conclusion: Although, there was lots of discussion on alternatives, none
were deemed reasonable, so will go forward with proposal.
Discussion: Could we have only one escaping mechanism? Point: There will be
not translation that occurs. we are just renaming feature tags in the token
codespace, and there is never a decode.
 

Path Forward:
Proposal: One more rev with these issues. 
Question: Ready for WGLC?  Poll taken, general consensus recorded. 
Question: What about use cases?  Should it find a home somewhere? 
There was some discussion on what to do with the use cases, with the general
consensus expressed that the use cases are useful; whether they should
remain in a separate doc, be put in an appendix of this doc or merged with
other call flows is still undecided. 
Conclusion: 2 week WGLC after proposed changes.  


SIP Guidelines Open Issues,  Rohan Mahy
-------------------------------------------

Issues: Notion of CANCELs and provisional responses for non-INVITES
currently conflicts with RFC 3261, that you must specify processing in new
drafts.  Does this causes more harm than good?

Discussion: There was discussion around exactly what's currently in RFC
3261, with Rohan suggesting that there's never a good reason to receive
Cancel  for a non-INVITE. However,  Robert S. highlighted that there is one.
If a non-INVITE times out with no responses, the SRV draft is written that
the endpoint fails over. Immediate provisionals are BAD for non-INVITE.
Sending a provisional before a transaction times out can prevent some
problems.  There are really two different problems here: provisional
responses, and CANCELs for non-invite messages. The discussion continued
around TCP introducing additional problems and the importance of separating
the 100 Trying from the discussion of provisionals. 

Conclusion: Robert to start list discussion on this particular topic once
draft is available. 


SIP Session Timer -- Jonathan Rosenberg
------------------------------------------
Changes from last rev: 
* Rewrote overview of operation 
* 422 response causes you to retry with same call-ID and bumped CSEQ like
401 
* Clarified that mid-dialog re-INVITE w/o session timer cancels session
timer 
 
Issues:

* Many complaints and limited interest 
* No defined requirements, not entirely clear the problem that is being
solved. 

Proposition: only useful application is cleanup of state in proxies 
* Is it too complex for that usage? 

Proposals: 
* Redefine the scope 
* Keep the scope, but clarify the wording 
* Keep the scope, but pursue a completely a new design 
* For example, just use the dialog package! 
 
Conclusion: Concensus appears to be option 2 - keep the scope and clarify
the wording. 
 
Discussion: There was some discussion around the lack of requirements, which
is likely what leads to many of the problems in understanding the draft and
finding issues therewith.  

Proposal: Draft should have a reasonable statement of requirements in the
draft, including some which are not already there. 
 

Discussion of previously discussed alternatives, including the suggestion
that this solution could align with RTSP and use the PING method.  
 

There was also discussion around support of this in other methods, however,
it was highlighted that that introduces the need to maintain state. 
 
Question:   Who would support this draft? Medium hum vote support
Question:   Who would support if it were enhanced (that wouldn't support
otherwise)? Even smaller hum vote. 
Question:   How many people violently object to moving this draft forward
as-is? No objections.
o         

Final Conclusion:  Jonathan will clarify the current version and submit for
going forward and WGLC. 


SIPS URIs, Jonathan Rosenberg
---------------------------------
Basic problem (SIPS URI):  Text is a bit confused on whether its e2e or
hop-by-hop 
Additional problems: Mixed cases - SIPS URI in Contact 200 ok if it ws
nowhere else? 
Solution Need to come to agreement on the high level behavior, from there
the detailed behaviors will flow 
 

Discussion:  There was some discussion around the requirement for using
SIPS.  The general problem is with the double Record Routing. 
It was highlighted that there needs to be one place that properly  describes
record routing that can be re-used by compression, SIPS and transport.  It
was highlighted that (for compression) the  double record routing  works
okay because there is not currently a scenario that needs to change
something in the response. Robert proposed that you can use this abstraction
to highlight that the link characteristics on either side are different. And
that, SIPS may introduce additional constraints. 
 
Proposal: Robert will specify this general mechanism for multiple record
routing. The  specific cases where the issues have been identified in
RFC3261 with different characteristics should reference the general
mechanism. 

Conclusion: Go forward with Robert's proposal and update the text in RFC3261
appropriately to make use of this mechanism.  

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Subject: RE: [SIP] INFO Contact
Date: Sun, 22 Dec 2002 15:55:08 +0100
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Hi Antonio,

in a REGISTER message,

the "To" Header Field contains the address of Record whose registration it to be created.

The "Contact" Field MAY be present in REGISTER REQUEST and it may contain "zero
or more values" containg address bindings.


If Contact Header is absent, the only contact address for the user is the Address present
in the "To" Header.


What do you mean about the binding between IP address in VIA field and the Request-URI?
I'm not understand the question and what do you mean.


ciao
Salvatore

personal mail: loretosa@vizzavi.it
work mail: salvatore.loreto@eri.ericsson.se
                loreto@coritel.it
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CC: "Pramodh.N" <pramodhn@cdotb.ernet.in>,
        Gade Ramesh
 <gramesh@cdotb.ernet.in>
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Subject: [Sip] SIP backward compatibility issue
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1) Assuming we have an application built using SIP stack developed to 
comply with RFC 3261, does it have to interwork with RFC 2543 compliant 
stack? If not RFC2543, then what about the various drafts (draft 5, 
draft 6, ..., draft 9)?

2) I was also wondering about the theoretical possibility of backward 
compatibility? Any provisions in the protocol for an implementation to 
detect and manage this issue?

3) Is it feasible for applications to have dual SIP Stacks? (Both 2543 
and 3261)?

4) Does it make sense now to still deal with 2543?

Thanks in advance.

Regards,
-- 
Prakash R
Senior Program Manager, Internet Telephony Group
=====================================================================
C-DOT (Centre for Development of Telematics)
71/1, Sneha Complex, Miller Road, Bangalore - 560052, India
email: rprakash@cdotb.ernet.in, prakash_r@hotmail.com
Phone: +91-80-2263399, +91-80-2383951, Fax: +91-80-2263256
=====================================================================

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	Title		: The Internet Assigned Number Authority Header Field
                          and Universal Resource Identifier Parameter Registry
                          for the Session Initiation Protocol
	Author(s)	: G. Camarillo
	Filename	: draft-camarillo-sip-parameter-registry-00.txt
	Pages		: 6
	Date		: 2002-12-20
	
This document creates an IANA registry for SIP header field, SIP URI
and SIPS URI parameters. It also lists the already existing
parameters to be used as initial values for that registry.

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	Title		: The Internet Assigned Number Authority Header Field
                          and Universal Resource Identifier Parameter Registry
                          for the Session Initiation Protocol
	Author(s)	: G. Camarillo
	Filename	: draft-camarillo-sip-parameter-registry-00.txt
	Pages		: 6
	Date		: 2002-12-20
	
This document creates an IANA registry for SIP header field, SIP URI
and SIPS URI parameters. It also lists the already existing
parameters to be used as initial values for that registry.

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_______________________________________________
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From mailnull@www1.ietf.org  Mon Dec 23 09:53:25 2002
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Subject: [Sip] Contact header for PSTN 2 IP call
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Hi,
I have a question ! What will be the contact header in the sip message
when an PSTN to IP call is made via a media gateway ? Since PSTN phone
is not a SIP entity and cannot have a SIP URI.
Should it be sip:phone_no@media_gateway_domain.com? or something else?
pls help


Rinchen



--
Rinchen Tundup



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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
&nbsp;
<br>Hi,
<br>I have a question ! What will be the contact header in the sip message
<br>when an PSTN to IP call is made via a media gateway ? Since PSTN phone
<br>is not a SIP entity and cannot have a SIP URI.
<br>Should it be sip:phone_no@media_gateway_domain.com? or something else?
<br>pls help
<br>&nbsp;
<p>Rinchen
<br>&nbsp;
<br>&nbsp;
<pre>--&nbsp;
<b>Rinchen Tundup&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</b>
</pre>
&nbsp;</html>

--------------695C5364C8528D56E7A4EC24--


_______________________________________________
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Use sipping@ietf.org for new developments on the application of sip



From mailnull@www1.ietf.org  Mon Dec 23 10:39:10 2002
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To: Rinchen Tundup <rinchen@cdotd.ernet.in>
CC: sip@ietf.org, sip@ietf.com
Subject: Re: [Sip] Contact header for PSTN 2 IP call
References: <3E074FE4.6C05CC10@cdotd.ernet.in>
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Hi,

The PSTN phone is not a SIP entity, but the media gateway is. In a
normal case I would assume the values of the userpart and host are as
you describe in your example.

Regards,

Christer Holmberg
Ericsson Finland

Rinchen Tundup wrote:

>
> Hi,
> I have a question ! What will be the contact header in the sip message
>
> when an PSTN to IP call is made via a media gateway ? Since PSTN phone
>
> is not a SIP entity and cannot have a SIP URI.
> Should it be sip:phone_no@media_gateway_domain.com? or something else?
>
> pls help
>
>
> Rinchen
>
>
>
> --
> Rinchen Tundup
>
>

_______________________________________________
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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



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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Rinchen Tundup'" <rinchen@cdotd.ernet.in>, sip@ietf.org
Subject: RE: [Sip] Contact header for PSTN 2 IP call
Date: Mon, 23 Dec 2002 10:47:40 -0500
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Usually it is phonenumber@gateway.com
There is some work to update the "tel" URI, which would
probably be used, as in
tel:+17247426826@gateway.marconi.com

Some gateways need to get a "port" number in there,
and you occasionally see:
"7247426826" <port5@gateway.example.com>
although that is more often found in a From or To rather
than a Contact.

Brian

-----Original Message-----
From: Rinchen Tundup [mailto:rinchen@cdotd.ernet.in]
Sent: Monday, December 23, 2002 1:03 PM
To: sip@ietf.org; sip@ietf.com
Subject: [Sip] Contact header for PSTN 2 IP call


  
Hi, 
I have a question ! What will be the contact header in the sip message 
when an PSTN to IP call is made via a media gateway ? Since PSTN phone 
is not a SIP entity and cannot have a SIP URI. 
Should it be sip:phone_no@media_gateway_domain.com? or something else? 
pls help 
  
Rinchen 
  
  
-- 
Rinchen Tundup      

  
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From mailnull@www1.ietf.org  Mon Dec 23 11:28:26 2002
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To: sip@ietf.org, sip@ietf.com, "Rinchen Tundup" <rinchen@cdotd.ernet.in>
Date: Mon, 23 Dec 2002 11:26:26 -0500
From: "Jitender Arora" <jitender_arora@lycos.com>
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Hi ,

          Assume that u are using the ISUP on the PSTN side , so on recieving the IAM ,at the Media Gateway u find that this is a SIP call ( PhonetoNet call) ...

           U will send an invite out to the SIP Proxy ,with following content ....

           INVITE sip:phonenumber@host ( Host will depend on the Proxy , to which u are sending the Request and how your proxy is configured to )
           From :  sip:Mediagatewayname@host

           Similarly other headers are also required ......

           The contact will be :

            contact:sip:Mediagatewayname@host:portnumber

Regards,
Jitender
 
--

On Mon, 23 Dec 2002 11:03:17  
 Rinchen Tundup wrote:
>
>Hi,
>I have a question ! What will be the contact header in the sip message
>when an PSTN to IP call is made via a media gateway ? Since PSTN phone
>is not a SIP entity and cannot have a SIP URI.
>Should it be sip:phone_no@media_gateway_domain.com? or something else?
>pls help
>
>
>Rinchen
>
>
>
>--
>Rinchen Tundup
>
>
>


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Date: Mon, 23 Dec 2002 10:29:41 -0600
From: "Vijay K. Gurbani" <vkg@lucent.com>
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        Gade Ramesh <gramesh@cdotb.ernet.in>
Subject: Re: [Sip] SIP backward compatibility issue
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Prakash R wrote:
> 1) Assuming we have an application built using SIP stack developed to 
> comply with RFC 3261, does it have to interwork with RFC 2543 compliant 
> stack? If not RFC2543, then what about the various drafts (draft 5, 
> draft 6, ..., draft 9)?

A rfc3261 compliant SIP entity will be compliant with most of rfc2543
anyway.  There are some parts of the protocol that were ambiguous to
begin with in rfc2543 -- Record-Route handling to construct Route
headers is a good example.  As such, these were not uniformly
implemented anyway.  So, rfc3261 took some liberty in specifying these
in a more uniform and unambiguous manner than did rfc2543.  Even then,
in the case of R-R, backward compatibility has not been forsaken; a
rfc3261 SIP proxy, for instance, can still inter-operate with rfc2543
UAs which will not understand the loose-route extension.  Another
example of possible non-compatibility include security (i.e.
deprecation of PGP and use of S/MIME).  But again, these features were
not uniformly (or widely) implemented by rfc2543 implementors anyway.

Regarding compatibility with drafts, no.  You do not have to be
compliant with intermediate drafts -- they are good for 6 months anyway.
You can, of course, index your implementation to these drafts and
thus progress along the same curve as the protocol matures.  Compliance
only makes sense for an rfc, not an intermediate draft (see rfc2026).

> 2) I was also wondering about the theoretical possibility of backward 
> compatibility? Any provisions in the protocol for an implementation to 
> detect and manage this issue?

The SIP protocol number is still "SIP/2.0"; not "SIP/2.1".  You could
glean some information from the presence or absence of the Via magic
cookie, but you should not depend on it for anything.  There was a
thread a while ago on why depending on the magic cookie for operations
is not a good idea -- I don't have a reference to it (if someone else
on the list does, feel free to post it).  One of the reason you cannot
depend on the magic cookie is that it only tells you something about
the upstream UA and nothing about the downstream server (assuming you
are a proxy).

> 3) Is it feasible for applications to have dual SIP Stacks? (Both 2543 
> and 3261)?

Yes; although, once again, the utility of support PGP which was probably
absent in most rfc2543 implementations is dubious at best.

> 4) Does it make sense now to still deal with 2543?

Sure.  There is deployed gear that is rfc2543 compliant.  3Com's
Ethernet SIP phones for example are still used in many installations
(they work great) and are rfc2543 compliant only.

Regards,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216

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Internet-Drafts@ietf.org writes:

 > This document creates an IANA registry for SIP header field, SIP URI
 > and SIPS URI parameters. It also lists the already existing
 > parameters to be used as initial values for that registry.

how about registering various sip header fields for radius accounting
purpose?

-- juha

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From mailnull@www1.ietf.org  Mon Dec 23 15:16:28 2002
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Subject: [Sip] Re: Caller Preferences and Callee Capabilities: comments on version
 7
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My apologies to the delay in responding to this. I'm just getting back
to caller prefs now.

Stephane.Coulombe@nokia.com wrote:
 > Hi Jonathan and Henning,
 >
 > I would like to comment on the latest version of "Session Initiation
 > Protocol (SIP) Caller Preferences and Callee Capabilities" document.
 > The document significantly improved from the previous version.
 > However, it still has many shortcomings, which I would like to
 > address here.
 >
 > First, I am a bit concerned about the restricted way RFC 2533 is
 > used, by saying:
 >
 > "The feature predicate constructed by a UA MUST be an AND of terms.
 > Each term is either an OR of simple filters (called a disjunction),
 > or a single simple filter. In the case of an OR of simple filters,
 > each filter MUST indicate feature values for the same feature tag
 > (i.e., the disjunction represents a set of values for a particular
 > feature tag), and each element of the conjunction MUST be for a
 > different feature tag. Each filter can be an equality, the negation
 > of an equality, or in the case of numeric feature tags, an inequality
 >  or range"

This has been the scope of the caller preferences capabilities since the 
draft was first published way, way back - some FOUR years now [this is 
the oldest sip extension in existence].

Anyway, the reasoning here is that this specification is NOT meant as a 
general solution to the problem of declaring device capabilities. Its 
meant as a way to express capabilities to assist caller preference 
routing. THis introduces several important constraints. First, a proxy 
must perform computations on each message that gets sent, in real time. 
Thus, we cannot afford arbitrarily complex matching. Second, the set of 
things over which one might like to do preferential routing on, is much 
less than the general set of capabilities of a device. Honestly, I can't 
see anyone wanting to route a call because the screen size of the 
recipient is 100x200 as opposed to 200x300. This may be useful for some 
of the transcoding work you have been looking at, but I think that this 
is out of scope.

 >
 > this pretty much cuts the legs of RFC 2533. Not only it has the
 > restriction that it needs to be a serie of ORs inside a big AND. But
 > the terms in the OR need to have same feature tags. This is very
 > limiting especially if you want to deal with media content. A simple
 > example from RFC 2533: (| (& (pix-x=750) (pix-y=500) (color=15) ) (&
 > (dpi=300) (ua-media=stationery) (papersize=iso-A4) ) ) would not be
 > acceptable according to the proposed document.

Correct. But why would you care to route a call based on this?

 >
 > To express caller's capabilities, "AND"ing terms is often sufficient
 > (but not always) but certainly not for expressing callee capabilities
 > which should be a big "OR" of supported features. RFC 2533 is full of
 > "OR" examples. E.g. (| (& (| (& (pix-x<=640)  (pix-y<=480)
 > (color<=16777216) ) (& (pix-x<=800)  (pix-y<=600) (color<=65535) ) (&
 > (pix-x<=1024) (pix-y<=768) (color<=256) ) ) (ua-media=screen) ) (&
 > (dpi=300) (ua-media=stationery) (papersize=iso-A4) ) )
 >
 > How to interpret the following callee capabilities? Contact:
 > sip:1.2.3.4;type="video/H263,image/jpeg,audio/amr";video="TRUE",audio-
 > ="TRUE",resolution="#<200" According to the rule, (& (|
 > (type=video/H263) (type-image/jpeg) (type=audio/amr) (video=TRUE)
 > (audio=TRUE) (resolution < 200) ) This is not the way RFC 2533 would
 > represent a terminal supporting audio OR video (the syntax here
 > suggests that the received media MUST be video AND audio).

Not at all. Please see section 5 which outlines the model of how rfc2533 
is used. You seem to be interpreting the predicate as saying "media 
streams that I receive MUST have these characteristics". But that is not 
the meaning. The meaning is, this predicate represents the set of 
instantaneous operating modes I support.

 > Therefore, I also share the concern raised during the SIP meeting
 > regarding the reference to RFC 2533 but definition of a new syntax in
 > the document. The proposal should be better aligned with RFC 2533.

What value is gained from changing syntax?

 >
 > The resolution attribute would better apply to the specific MIME type
 > rather than be general for all media types. E.g. | (&
 > (type=image/jpeg) (resolution<640)) & (type=video/h263)
 > (resolution<176)))

See my above comment on scope.

 >
 > I am also concerned about the usage of a range in capabilities. For
 > instance, if the caller uses resolution="#R5..100" as in the
 > document. If callee capabilities are resolution="#R5..50", then I
 > guess that terminal is acceptable because intersection is not zero.

Right.

 > Then, what happens to the terminal if the sender sends 100 pixel
 > image because system said it was okay?

Well, I once again question the usage of caller prefs for this kind of 
operation.

However, if a caller wants to route to a UA that supports resolutions 
100 and higher it would ask for that, i.e., "#>=100".



 > Presently, the document mostly focuses on callee capabilities that
 > could be useful to a caller to set preferences on. Indeed callee
 > capabilities include: methods, events, etc. But the scope is general
 > and also applies to terminal characteristics which can be used in a
 > broader range of situations and applications (the document indeed
 > gives example of resolution, format types, etc.).

The document says that this information can be used for other things. It 
does not say that caller prefs is a genreal solution to declaring client 
capabilities for a broad set of applications. That, I contend, is a much 
broader scoped problem, and one with many existing solutions. It would 
need a much richer set of attributes, and declarations. You probably 
want to make it hard state, support uploading a URL. You have different 
authorization issues - now you might not want everyone to see and use 
it, only certain applications. And so on.

It may very well be that IFF we had a general UA capability declaration 
function, caller preferences should be based on that. But we don't. We 
are (still) trying to solve a smaller scoped problem at hand. At such 
time as we have a more general purpose mechanism, if needed, we can 
revisit aligning caller preferences with it.

  Therefore, it would
 > be good if these callee capabilities would not be limited to the
 > REGISTER method but be part of other methods such as OPTIONS (for
 > someone to learn about these capabilities),

Actually I thought OPTIONS was covered, since the Contact in OPTIONS is 
similar to the Contact in 200 to INVITE, which can have caller prefs 
parameters. I'll mention that.



 > SUBSCRIBE (for presence
 > server to be able to create appropriate presence messages), etc.

This doesn't make sense to me.

 > In
 > that manner, not only the specification can be used for message
 > routing based on caller preferences but also based on knowledge of
 > callee capabilities (either in routing stage or at the sending end).
 >
 > In summary, here are my recommendations: -	

 > Align the syntax and
 > overall functionality to use full extent of RFC 2533.

Disagree. See above comments on scope.

 > o	Remove the
 > restriction of having a serie of "OR" in a big "AND" in syntax.

Disagree. See above comments on scope.

 >o	Fix
 > matching problem with ranges.

It was not clear to me what the problem is.


 > o	Fix other syntax problems outlined
 > above.

Which?


  o	Allow and add more detailed media capability description and
 > preferences (e.g. resolution can apply to specific MIME types. - I
 > suggest to add the descriptors used in
 > draft-coulombe-message-adaptation-framework-00.txt : 1. length:
 > maximum size of the entity specified. It could apply to the whole
 > message or to specific MIME types.

Disagree. See above discussions on scope.

 >
 > 2. media-pix-x: horizontal resolution of visual media (e.g. image or
 >  video). It could apply to all images or video or to specific MIME
 > types.

I don't think this is useful given the current scope.
 >
 > 3. media-pix-y: vertical resolution of visual media (e.g. image or
 > video). It could apply to all images or video or to specific MIME
 > types.  As described in the document, it should be possible to
 > associate "length" to the whole message or to specific MIME types.
 > These feature tags are quite important for mobile terminals and some
 > handheld devices. Note that the resolution is NOT necessarily the
 > display resolution, but rather what the terminal can support (usually
 > larger than the display resolution). For instance, I may want to
 > route a message containing an image to a terminal that support
 > high-enough resolution.

As above.

 >
 > -	Extend the methods where the callee capabilities are provided (not
 > just REGISTER but also SUBSCRIBE, OPTIONS, etc.)

OPTIONS ought to be there. I see no use for SUBSCRIBE.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Subject: Re: [Sip] I-D ACTION:draft-camarillo-sip-parameter-registry-00.txt
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This spec is meant JUST to fill in the gap of IANA registries for 
parameters that can be extended in SIP. RADIUS parameters are well 
outside of the scope here...

-Jonathan R.

jh@lohi.eng.song.fi wrote:
> Internet-Drafts@ietf.org writes:
> 
>  > This document creates an IANA registry for SIP header field, SIP URI
>  > and SIPS URI parameters. It also lists the already existing
>  > parameters to be used as initial values for that registry.
> 
> how about registering various sip header fields for radius accounting
> purpose?
> 
> -- juha
> 
> _______________________________________________
> Sip mailing list  https://www1.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.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
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To: "Rosen, Brian" <Brian.Rosen@marconi.com>
CC: "'Rinchen Tundup'" <rinchen@cdotd.ernet.in>, sip@ietf.org
Subject: Re: [Sip] Contact header for PSTN 2 IP call
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The Contact URI can be whatever you want. The only requirement is that 
its globally routable, and when you invoke it, it routes back to that UA 
instance. You need not put the phone number in there if you don't want 
to. The user part can be set arbitrarily at the discretion of the 
implementor. However:

Rosen, Brian wrote:
> Usually it is phonenumber@gateway.com
> There is some work to update the "tel" URI, which would
> probably be used, as in
> tel:+17247426826@gateway.marconi.com

No.

First off, the tel URI has no domain part, i.e., tel:+17247426826. 
Anyway, you cannot use a tel URI since the Contact header MUST be a sip 
or sips URI. It is fine to use sip:+17247426826@gateway.marconi.com, 
however.

> 
> Some gateways need to get a "port" number in there,
> and you occasionally see:
> "7247426826" <port5@gateway.example.com>
> although that is more often found in a From or To rather
> than a Contact.

The meaning of To, From, Contact are well defined in the spec. To/From 
are logical identifiers, and shouldn't be of the form user@host where 
host is a specific device (the gateway, a PC, etc.), but rather, 
user@domain. Contact, on the other hand, needs to reference the device 
as I mentioned above.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
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Vijay K. Gurbani wrote:

>> 2) I was also wondering about the theoretical possibility of backward 
>> compatibility? Any provisions in the protocol for an implementation to 
>> detect and manage this issue?
> 
> 
> The SIP protocol number is still "SIP/2.0"; not "SIP/2.1".  You could
> glean some information from the presence or absence of the Via magic
> cookie, but you should not depend on it for anything. 

There is absolutely no need.

rfc3261 is sufficiently backwards compatible with rfc2543. There is no 
need to detect versions or anything. Any backwards compatibility 
functions are built into specific features of rfc3261 that needed to 
support backwards compatibility. If you just implement rfc3261 you will 
be backwards compatible.

There is no versioning of the protocol. Things like the via cookie are 
meant to deal with backwards compatibility for THAT SPECIFIC FEATURE. It 
is possible to encounter an implementation that supports the contact 
cookie, but is still a strict router. Using the contact cookie to imply 
support for all of 3261 is a mistake. The lr parameter deals with the 
record-routing changes, and the contact cookie, with the via header 
changes.


  There was a
> thread a while ago on why depending on the magic cookie for operations
> is not a good idea -- I don't have a reference to it (if someone else
> on the list does, feel free to post it).  One of the reason you cannot
> depend on the magic cookie is that it only tells you something about
> the upstream UA and nothing about the downstream server (assuming you
> are a proxy).

The problem, as I outline above, is we are not trying to strictly 
version the protocol, but provide interoperability and backwards 
compatibility on a feature-by-feature basis. So, via cookie tells you 
nothing except that the via branch ID is sufficiently unique to be used 
as a transaction ID.


> 
>> 3) Is it feasible for applications to have dual SIP Stacks? (Both 2543 
>> and 3261)?
> 
> 
> Yes; although, once again, the utility of support PGP which was probably
> absent in most rfc2543 implementations is dubious at best.

There were none that anyone mentioned.

To me, "dual-stack" is non-sensical. rfc3261 provides backwards 
compatibility where needed.


> 
>> 4) Does it make sense now to still deal with 2543?
> 
> 
> Sure.  There is deployed gear that is rfc2543 compliant.  3Com's
> Ethernet SIP phones for example are still used in many installations
> (they work great) and are rfc2543 compliant only.

As above, it is my opinion that rfc3261 sufficiently deals with rfc2543 
for any feature you are going to find in use from rfc2543.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
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Subject: Re: [Sip] I-D ACTION:draft-camarillo-sip-parameter-registry-00.txt
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Jonathan Rosenberg writes:

 > This spec is meant JUST to fill in the gap of IANA registries for 
 > parameters that can be extended in SIP. RADIUS parameters are well 
 > outside of the scope here...

i didn't mean that they should be included in this spec, but that they
are iana numbers that would be very useful to exist.  now everyone
invents their own radius attribute values.

-- juha

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From: Vijaya Venkatachalam <vijaya@ulticom.com>
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Subject: [Sip] INVITE transaction
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Hi,

I have a question regarding the INVITE transaction
state machine.

It seems that in RFC 3261 - transaction state machine for
INVITE transactions, there is a potential for
the client side transaction to indefinitely be
in the "Proceeding" state since all timers
are cancelled on the reception of a 1xx response.

Can somebody tell me how the client side transaction
can come out of this state, if the server transaction on
the remote end dies?

Is it that the timer B is not cancelled and that only
timer A is cancelled when the 1xx is received or is
there some other means.

Appreciate a quick response.

Thanks,
Vijaya
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Subject: [Sip] Contact header for an PSTN to IP call
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Hi,
I have a question ! What will be the contact header in the sip message
when an PSTN to IP call is made via a media gateway ? Since PSTN phone
is not a SIP entity and cannot have a SIP URI.
Should it be sip:phone_no@media_gateway_domain.com? or something else?
pls help


Rinchen


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From: Dean Willis <dwillis@greycouncil.com>
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Subject: Re: [Sip] Contact header for PSTN 2 IP call
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On Monday 23 December 2002 12:03 pm, Rinchen Tundup wrote:
> Hi,
> I have a question ! What will be the contact header in the sip
> message when an PSTN to IP call is made via a media gateway ? Since
> PSTN phone is not a SIP entity and cannot have a SIP URI.
> Should it be sip:phone_no@media_gateway_domain.com? or something
> else? pls help

Well, it generally is "whatever the media gateway wants it to be". I 
suspect it must satisfy two requirements -- it points at the gateway, 
and can be used by the gateway to identify the correct association.

The gateways I use at the office use sip:phone_no@hostname-o-fgateway or 
sip:phone_no@ip-address-of-gateway depending on which gateway and how 
it is configured.

--
Dean
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inline.

Jiri Kuthan wrote:

 >
 >> This will cause the INVITE state machine to consider that response
 >> as a response to the original INVITE (CSeq method is not needed in
 >> the matching process!).
 >
 >
 > Is that some implicit assumption too? I read the following in spec,
 > section 17.1.3: "A response matches ... If the method parameter in
 > the CSeq header field matches the method of the request that created
 > the transaction." It does not mention any special case, in which
 > method matching is not used. Which is IMHO correct.

Not an implicit assumption, just me forgetting that the method check was 
there too. So, I guess there would not be any confusion in the client 
state machine. But, there are enough other potential problems that I 
still think its not a good idea.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
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This is an application issue. It can decide to give up at any time and 
send CANCEL. We don't want to specify the maximum time a user will let 
the phone ring.

-Jonathan R.

Vijaya Venkatachalam wrote:
> Hi,
> 
> I have a question regarding the INVITE transaction
> state machine.
> 
> It seems that in RFC 3261 - transaction state machine for
> INVITE transactions, there is a potential for
> the client side transaction to indefinitely be
> in the "Proceeding" state since all timers
> are cancelled on the reception of a 1xx response.
> 
> Can somebody tell me how the client side transaction
> can come out of this state, if the server transaction on
> the remote end dies?
> 
> Is it that the timer B is not cancelled and that only
> timer A is cancelled when the 1xx is received or is
> there some other means.
> 
> Appreciate a quick response.
> 
> Thanks,
> Vijaya
> _______________________________________________
> Sip mailing list  https://www1.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.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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Subject: Re: [Sip] SIP backward compatibility issue
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><snip>
>rfc3261 is sufficiently backwards compatible with rfc2543. There is no 
>need to detect versions or anything. Any backwards compatibility functions 
>are built into specific features of rfc3261 that needed to support 
>backwards compatibility. If you just implement rfc3261 you will be 
>backwards compatible.
<snip>

broadening the discussion to rfc 3264 (offer/answer), isn't there a 
potential for backward compatibility issues with rfc 2543 implementations 
wrt early media ?

thx,
steve

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Date: Tue, 24 Dec 2002 11:46:54 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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CC: "Vijay K. Gurbani" <vkg@lucent.com>, rprakash@cdotb.ernet.in, sip@ietf.org,
        "Pramodh.N" <pramodhn@cdotb.ernet.in>,
        Gade Ramesh <gramesh@cdotb.ernet.in>
Subject: Re: [Sip] SIP backward compatibility issue
References: <3E070194.6080006@cdotb.ernet.in> <3E0739F5.8050103@lucent.com> <4.3.2.7.2.20021224112121.0207a0a0@dingdong.cisco.com>
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Early media was not specified in rfc2543. It did say that you should be 
prepared to receive media before receiving the answer, which is still 
the case for 3264. Another issue was that people used to put SDP in 
non-reliable 1xx responses, and then put the same thing in 2xx. That 
case is explicitly allowed for in 3261, so it should interoperate as well.

Perhaps you could identify the specific problem you are concerned about?

-Jonathan R.

Steve Levy wrote:
> 
>> <snip>
>> rfc3261 is sufficiently backwards compatible with rfc2543. There is no 
>> need to detect versions or anything. Any backwards compatibility 
>> functions are built into specific features of rfc3261 that needed to 
>> support backwards compatibility. If you just implement rfc3261 you 
>> will be backwards compatible.
> 
> <snip>
> 
> broadening the discussion to rfc 3264 (offer/answer), isn't there a 
> potential for backward compatibility issues with rfc 2543 
> implementations wrt early media ?
> 
> thx,
> steve
> 

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
Sip mailing list  https://www1.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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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
From: Steve Levy <stlevy@cisco.com>
Subject: Re: [Sip] SIP backward compatibility issue
Cc: "Vijay K. Gurbani" <vkg@lucent.com>, rprakash@cdotb.ernet.in, sip@ietf.org,
        "Pramodh.N" <pramodhn@cdotb.ernet.in>,
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At 11:46 AM 12/24/2002 -0500, Jonathan Rosenberg wrote:
>Early media was not specified in rfc2543. It did say that you should be 
>prepared to receive media before receiving the answer, which is still the 
>case for 3264. Another issue was that people used to put SDP in 
>non-reliable 1xx responses, and then put the same thing in 2xx. That case 
>is explicitly allowed for in 3261, so it should interoperate as well.
>
>Perhaps you could identify the specific problem you are concerned about?

my concern was SDP in a 18x followed by another 18x/2xx with different SDP. 
Wouldnt the SDP in the subsequent 18x/2xx be treated differently by 3264 
compliant UAs as opposed to earlier 2543 compliant UAs ?

thx,
steve



>-Jonathan R.
>
>Steve Levy wrote:
>>
>>><snip>
>>>rfc3261 is sufficiently backwards compatible with rfc2543. There is no 
>>>need to detect versions or anything. Any backwards compatibility 
>>>functions are built into specific features of rfc3261 that needed to 
>>>support backwards compatibility. If you just implement rfc3261 you will 
>>>be backwards compatible.
>><snip>
>>broadening the discussion to rfc 3264 (offer/answer), isn't there a 
>>potential for backward compatibility issues with rfc 2543 implementations 
>>wrt early media ?
>>thx,
>>steve
>
>--
>Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
>Chief Scientist                             First Floor
>dynamicsoft                                 East Hanover, NJ 07936
>jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>http://www.jdrosen.net                      PHONE: (973) 952-5000
>http://www.dynamicsoft.com
>
>_______________________________________________
>Sip mailing list  https://www1.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://www1.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 mailnull@www1.ietf.org  Thu Dec 26 15:20:48 2002
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Folks,

For some time, I've been complaining about the continued abuse of the 
INFO method for things that should be done other ways (frequently, not 
using sip at all). I've written up a summary of the problems and a 
proposed path forward. Specifically, I'd like to obsolete INFO and 
replace it with a spec that is approved ONLY for SIP-T.

Until the I-D appears, you can pick up a version at:

http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.txt
http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.html

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
Sip mailing list  https://www1.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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--=====================_-1610655670==_.ALT
Content-Type: text/plain; charset="us-ascii"; format=flowed

Hi,
Following an earlier thread from sip-implementors here is another
scenario that leads to similar problem.

Suppose UAC sends an INVITE out and receives 2 provisional
response with different to-tags. These 2 provisional response would
be handled by the same client transaction as both of the would have
the same top-most branch-id (even though they have different to-tags)
They will create 2 early dialogs. Now suppose, one of the early dialog
matures with a 2xx response. This would immediately delete the client
transaction. Now, let the other dialog be terminated with a 4xx.

1. Should we send ACK for this 4xx?
2. If yes, Who  should send the ACK for this 4xx. TU? or TL?
3. Also, how should it be sent. Should we create a new client transaction
     for this purpose?

RFC 3261 Section 17.1.1.2 Formal Description, says,

"The client transaction
    MUST pass the received response up to the TU, and the client
    transaction MUST generate an ACK request, even if the transport is
    reliable (guidelines for constructing the ACK from the response are
    given in Section 17.1.1.3) and then pass the ACK to the transport
    layer for transmission.  The ACK MUST be sent to the same address,
    port, and transport to which the original request was sent."

Now, in our case as we don't have client transaction we have lost
all the context regarding the original transaction. Should we create
a new client transaction and send the ACK using it. If this is done,
then we would have another problem. Re transmissions of the 2xx
for the matured dialog would hit on this client transaction (since
they would have the same branch-id's). Due to this the 2xx won't
be indicated to the TU at all. As I said earlier (see below mail) I still
think that this should be explicitly stated in the RFC. Making this
explicit would ensure that implementors don't accidentally over look
this point.

Thanks
Sachin



>Sorry, I think correction might be required not in Response matching but in
>FSM Figure 5: INVITE client transaction.
>
>We need to add one more event in "Completed" state,  event 2xx.
>In "Completed" state if we recieve a 2xx we would indicate it to TU.
>
>Hope I got it right.
>
>Thanks
>Sachin
>
>
>At 07:24 PM 7/24/02 +0530, you wrote:
>>Hi Bob,
>>
>>According to RFC 3261 17.1.3 "Matching Responses to Client Transactions",
>>It is said that client transaction for response is matched using
>>topmost via-branch & Cseq method.
>>
>>I guess this will have to be changed if we have to forward the 2xx also.
>>
>>Since the response context for 4xx would be still active, when a
>>2xx comes, according to RFC, we will match it with the 4xx's response
>>context. We will end up retransmitting ACK (not knowing that it is 2xx
>>but thinking that it is a 4xx).
>>
>>Hope I didn't miss anything over here.
>>
>>Thanks
>>Sachin
>>
>>
>>At 09:40 AM 7/24/02 -0400, you wrote:
>>
>>>----- Original Message -----
>>>From: "Sachin Shenoy" <sachins@netlab.hcltech.com>
>>>
>>> > Hi,
>>> >
>>> > If a proxy sends out an INVITE statefully, and gets a non-2xx response
>>>(say
>>> > 4xx),
>>> > it will send out an ACK locally. Now if it gets 2xx on the same branch,
>>> > it would consider this 2xx as retransmission of 4xx and retransmit ACK.
>>> > (Since 4xx & 2xx would have the same top-most via branch)
>>> >
>>> > (It may end up getting 2xx after forwarding 4xx due to a mis-behaving
>>>proxy
>>> > down the line, which forked the INVITE?).
>>> >
>>> > Is this behaviour Ok?
>>> >
>>>
>>>No.
>>>
>>>If this proxy did receive a 2xx after receiving a 4xx, it would mean a
>>>downstream (not necessarily mis-behaving) proxy had given up on waiting for
>>>a response on a given branch. The 2xx and 4xx would not have come from the
>>>same UAS.
>>>
>>>The proxy should not send an ACK for the 2xx, but should forward the
>>>response back toward the UAC. ACK for non-2xx is hop-by-hop, but ACK for 2xx
>>>is end-to-end and it is a request within the dialog established/identified
>>>by the 2xx.
>>>
>>>If the UAC had already given up on the call, it would send a BYE after
>>>sending the ACK.

--=====================_-1610655670==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
Hi,<br>
Following an earlier thread from sip-implementors here is another<br>
scenario that leads to similar problem.<br>
<br>
Suppose UAC sends an INVITE out and receives 2 provisional <br>
response with different to-tags. These 2 provisional response would<br>
be handled by the same client transaction as both of the would have<br>
the same top-most branch-id (even though they have different
to-tags)<br>
They will create 2 early dialogs. Now suppose, one of the early dialog
<br>
matures with a 2xx response. This would immediately delete the
client<br>
transaction. Now, let the other dialog be terminated with a 4xx. <br>
<br>
1. Should we send ACK for this 4xx?<br>
2. If yes, Who&nbsp; should send the ACK for this 4xx. TU? or TL? <br>
3. Also, how should it be sent. Should we create a new client
transaction<br>
&nbsp;&nbsp;&nbsp; for this purpose?<br>
<br>
RFC 3261 Section 17.1.1.2 Formal Description, says, <br>
<br>
&quot;The client transaction<br>
&nbsp;&nbsp; MUST pass the received response up to the TU, and the
client<br>
&nbsp;&nbsp; transaction MUST generate an ACK request, even if the
transport is<br>
&nbsp;&nbsp; reliable (guidelines for constructing the ACK from the
response are<br>
&nbsp;&nbsp; given in Section 17.1.1.3) and then pass the ACK to the
transport<br>
&nbsp;&nbsp; layer for transmission.&nbsp; The ACK MUST be sent to the
same address,<br>
&nbsp;&nbsp; port, and transport to which the original request was
sent.&quot;<br>
<br>
Now, in our case as we don't have client transaction we have lost <br>
all the context regarding the original transaction. Should we 
create<br>
a new client transaction and send the ACK using it. If this is 
done,<br>
then we would have another problem. Re transmissions of the 2xx<br>
for the matured dialog would hit on this client transaction (since<br>
they would have the same branch-id's). Due to this the 2xx won't<br>
be indicated to the TU at all. As I said earlier (see below mail) I still
<br>
think that this should be explicitly stated in the RFC. Making this 
<br>
explicit would ensure that implementors don't accidentally over look
<br>
this point.<br>
<br>
Thanks<br>
Sachin<br>
<br>
<br>
<br>
<blockquote type=cite cite>Sorry, I think correction might be required
not in Response matching but in <br>
FSM <font face="Courier New, Courier">Figure 5: INVITE client
transaction.<br>
<br>
</font>We need to add one more event in &quot;Completed&quot;
state,&nbsp; event 2xx.<br>
In &quot;Completed&quot; state if we recieve a 2xx we would indicate it
to TU.<br>
<br>
Hope I got it right.<br>
<br>
Thanks<br>
Sachin<br>
<br>
<br>
At 07:24 PM 7/24/02 +0530, you wrote:<br>
<blockquote type=cite cite>Hi Bob,<br>
<br>
According to RFC 3261 <font face="Courier New, Courier">17.1.3
&quot;Matching Responses to Client Transactions&quot;,<br>
It is said that client transaction for response is matched using<br>
topmost via-branch &amp; Cseq method.<br>
<br>
I guess this will have to be changed if we have to forward the 2xx
also.<br>
<br>
Since the response context for 4xx would be still active, when a<br>
2xx comes, according to RFC, we will match it with the 4xx's
response<br>
context. We will end up retransmitting ACK (not knowing that it is
2xx<br>
but thinking that it is a 4xx).<br>
<br>
Hope I didn't miss anything over here.<br>
<br>
Thanks<br>
Sachin<br>
<br>
<br>
</font>At 09:40 AM 7/24/02 -0400, you wrote:<br>
<br>
<blockquote type=cite cite>----- Original Message -----<br>
From: &quot;Sachin Shenoy&quot; &lt;sachins@netlab.hcltech.com&gt;<br>
<br>
&gt; Hi,<br>
&gt;<br>
&gt; If a proxy sends out an INVITE statefully, and gets a non-2xx
response<br>
(say<br>
&gt; 4xx),<br>
&gt; it will send out an ACK locally. Now if it gets 2xx on the same
branch,<br>
&gt; it would consider this 2xx as retransmission of 4xx and retransmit
ACK.<br>
&gt; (Since 4xx &amp; 2xx would have the same top-most via branch)<br>
&gt;<br>
&gt; (It may end up getting 2xx after forwarding 4xx due to a
mis-behaving<br>
proxy<br>
&gt; down the line, which forked the INVITE?).<br>
&gt;<br>
&gt; Is this behaviour Ok?<br>
&gt;<br>
<br>
No.<br>
<br>
If this proxy did receive a 2xx after receiving a 4xx, it would mean
a<br>
downstream (not necessarily mis-behaving) proxy had given up on waiting
for<br>
a response on a given branch. The 2xx and 4xx would not have come from
the<br>
same UAS.<br>
<br>
The proxy should not send an ACK for the 2xx, but should forward 
the<br>
response back toward the UAC. ACK for non-2xx is hop-by-hop, but ACK for
2xx<br>
is end-to-end and it is a request within the dialog
established/identified<br>
by the 2xx.<br>
<br>
If the UAC had already given up on the call, it would send a BYE
after<br>
sending the ACK.</blockquote></blockquote></blockquote></html>

--=====================_-1610655670==_.ALT--

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Hi,

From a SIP point of view your draft is basically talking about putting
restrictions on what kind of message body types to use in INFO, since the
SIP stack really doesn't care about what the application does with the
data sent in the INFO.

Now, does this mean that we also need to restrict what kind of message
body types we allow in INVITE, re-INVITE and UPDATE? I know the stack can
ignore message body types it doesn't support, but if people start sending
their DTMF tones etc in THESE messages I think we still have the same
interop/wrong-use-of problem as we currently have with INFO, don't we?

Just some thoughts...

Regards,

Christer Holmberg
Ericsson Finland


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Hi,

Some comments on this draft, assuming that this draft is open for comments.
Not sure if this should go to Simple or Sip mailing list.

1)Section 4.3 Constructing Coherent Resource State, states
"The resource list subscriber maintains a table for each resource
   list.  The table contains a row for each resource in the resource
   list."
and then it states that a version number is associated with each row of the
resource list table.
Why should a version be associated with each resource of a resource list?
Should it not be associated with the resource list table instead?

RFC 3265  states in section 4.3.2 State Deltas,
"Any event package that supports delta changes to states MUST include
   a version number that increases by exactly one for each NOTIFY
   transaction in a subscription. ".

2)Section 4.3 (of the draft) states that when the version number jumps, the
document received should be
processed. I presume that  "processing" implies what is elaborated in the
same section later on.
Extract from sec 4.3 of the draft:
"If the value in the document is more
   than one higher than the local version number, the local version
   number is set to the value in the new document, the document is
   processed, and the .list subscriber SHOULD generate a refresh request
   to trigger a full state notification."

whereas RFC 3265 in sec 4.3.2 says that when version number jumps , the
Notify message can be ignored.

Extract from RFC 3265, sec 4.3.2:
"If a NOTIFY arrives that has a version number that is incremented by
   more than one, the subscriber knows that a state delta has been
   missed; it ignores the NOTIFY message containing the state delta
   (except for the version number, which it retains to detect message
   loss), and re-sends a SUBSCRIBE to force a NOTIFY containing a
   complete state snapshot.

3)Should the .list subscriber generate a refresh request even when the
version is not consecutive AND
"State" header indicates full state? The .list subcriber may have lost some
 intermediate notifications, but has now
received full state. It can very well update its local version to the new
version  without doing any refresh.
If the "State" indicates partial, however,  then a refresh request is
called for.


thanks
Harsh Bhondwe






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From mailnull@www1.ietf.org  Fri Dec 27 08:27:48 2002
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From: Prakash R <rprakash@cdotb.ernet.in>
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CC: "Vijay K. Gurbani" <vkg@lucent.com>, sip@ietf.org,
        "Pramodh.N" <pramodhn@cdotb.ernet.in>,
        Gade Ramesh <gramesh@cdotb.ernet.in>,
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Subject: Re: [Sip] SIP backward compatibility issue
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Jonathan Rosenberg wrote:
> Vijay K. Gurbani wrote:
> 
>>> 2) I was also wondering about the theoretical possibility of backward 
>>> compatibility? Any provisions in the protocol for an implementation 
>>> to detect and manage this issue?
>>
>> The SIP protocol number is still "SIP/2.0"; not "SIP/2.1".  You could
>> glean some information from the presence or absence of the Via magic
>> cookie, but you should not depend on it for anything. 
 >
> There is absolutely no need.
> 
> rfc3261 is sufficiently backwards compatible with rfc2543. There is no 
> need to detect versions or anything. Any backwards compatibility 
> functions are built into specific features of rfc3261 that needed to 
> support backwards compatibility. If you just implement rfc3261 you will 
> be backwards compatible.
> 
> There is no versioning of the protocol. Things like the via cookie are 
> meant to deal with backwards compatibility for THAT SPECIFIC FEATURE. It 
> is possible to encounter an implementation that supports the contact 
> cookie, but is still a strict router. Using the contact cookie to imply 
> support for all of 3261 is a mistake. The lr parameter deals with the 
> record-routing changes, and the contact cookie, with the via header 
> changes.

An rfc3261 stack compliant stack will be 100% backward compatible with a 
rfc2543 stack? I can just get the two stacks to interwork without any 
issues at all? All the rfc2543 gear already deployed will work smoothly 
with a new stack? Hope this is true... Makes life simple for us.

>> There was a
>> thread a while ago on why depending on the magic cookie for operations
>> is not a good idea -- I don't have a reference to it (if someone else
>> on the list does, feel free to post it).  One of the reason you cannot
>> depend on the magic cookie is that it only tells you something about
>> the upstream UA and nothing about the downstream server (assuming you
>> are a proxy).
> 
> The problem, as I outline above, is we are not trying to strictly 
> version the protocol, but provide interoperability and backwards 
> compatibility on a feature-by-feature basis. So, via cookie tells you 
> nothing except that the via branch ID is sufficiently unique to be used 
> as a transaction ID.

I was earlier thinking of using this branch-tag to check which rfc the 
remote stack was complying with. Was not clear with all the scenarios 
though.

BTW, When will the protocol version be incremented? What sort of changes 
will necessitate a protocol version upgrade? I wasn't tracking the 
mailing list when the version got upgraded to 2.0.

>>> 4) Does it make sense now to still deal with 2543?
>>
>> Sure.  There is deployed gear that is rfc2543 compliant.  3Com's
>> Ethernet SIP phones for example are still used in many installations
>> (they work great) and are rfc2543 compliant only.
> 
> As above, it is my opinion that rfc3261 sufficiently deals with rfc2543 
> for any feature you are going to find in use from rfc2543.

I think I'll currently stick to rfc3261 and hope for the best. Will deal 
with 2543 as and when issues arise...

Regards,
-- 
Prakash R
Senior Program Manager, Internet Telephony Group
=====================================================================
C-DOT (Centre for Development of Telematics)
71/1, Sneha Complex, Miller Road, Bangalore - 560052, India
email: rprakash@cdotb.ernet.in, prakash_r@hotmail.com
Phone: +91-80-2263399, +91-80-2383951, Fax: +91-80-2263256
=====================================================================

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Jonathan Rosenberg wrote:
> Folks,
> 
> For some time, I've been complaining about the continued abuse of the 
> INFO method for things that should be done other ways (frequently, not 
> using sip at all). I've written up a summary of the problems and a 
> proposed path forward. Specifically, I'd like to obsolete INFO and 
> replace it with a spec that is approved ONLY for SIP-T.
> 
> Until the I-D appears, you can pick up a version at:
> 
> http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.txt
> http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.html

Jonathan and I have had "extended discussions" on this. I'm more 
inclined to think that INFO needs specifications for specific usages, 
much as we use "events packages" for SIP Events (SUBSCRIBE, NOTIFY). In 
my own completely non-original way, I call these "INFO Packages".

I think that the SIP-T usage of INFO is a perfect example of an INFO 
package. However, I think it is NOT the only reasonable use for INFO. 
Unsolicited in-dialog "stuff" is pretty darned useful. All we really 
need is for the usages to be differentiable, negotiable, and 
well-documented. Admittedly, INFO doesn't have any of those three 
characteristics right now, but I think we can probably fix it without 
breaking too much existing stuff.

Anybody else have a third opinion?

--
Dean
Not Speaking From The Chair.

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References: <3E070194.6080006@cdotb.ernet.in> <3E0739F5.8050103@lucent.com> <4.3.2.7.2.20021224112121.0207a0a0@dingdong.cisco.com> <4.3.2.7.2.20021224123942.0208c448@dingdong.cisco.com>
Subject: Re: [Sip] SIP backward compatibility issue
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"Prakash R" <rprakash@cdotb.ernet.in> wrote in message
news:3E0C50C8.4060302@cdotb.ernet.in...

> An rfc3261 stack compliant stack will be 100% backward compatible with a
> rfc2543 stack? I can just get the two stacks to interwork without any
> issues at all? All the rfc2543 gear already deployed will work smoothly
> with a new stack? Hope this is true... Makes life simple for us.

Yes, that was a design goal for RFC3261. That said, there are "holes" in
RFC2543 -- places in which the protocol was underspecified. Interop tests
and operational experience have revealed these holes; all the ones that we
have discovered to date have been fixed in RFC3261. So, there are ways to
implement an RFC2543 stack that won't work in general (even with other
RFC2543 stacks). There is obviously no magic fix for these problems.

> I was earlier thinking of using this branch-tag to check which rfc the
> remote stack was complying with. Was not clear with all the scenarios
> though.


Don't do that. Don't even try. You still encounter a huge number of stacks
that use the branch parameter but don't do 100% of RFC3261.

> BTW, When will the protocol version be incremented? What sort of changes
> will necessitate a protocol version upgrade?

From a practical perspective, I expect that such a change will be made only
if enhancements to the protocol cannot be added in a way that is backwards
compatible.

For a parallel, see the rather major enhancements that occured between
HTTP/1.0 and HTTP/1.1. If an overhaul of SIP of that magnitude is ever
attempted, there might be a minor version increment. However, given the
sheer volume of protocol spec at this point, it is quite unlikely (IMHO)
that such an undertaking will ever be attempted. Further, because of the
foresight of adding the "Require" and "Proxy-Require" mechanisms, it is
additionally unlikely that such a change will ever even be desirable.

A major version change, which is even less likely, would probably represent
completely scrapping the protocol and starting from scratch.

(As a footnote: like SIP, HTTP has had multiple RFCs that describe one
protocol version: RFC2068 and RFC2616 both describe HTTP/1.1 -- so we're not
doing anything particularly unprecedented or even unusual in doing the
same).

> I wasn't tracking the
> mailing list when the version got upgraded to 2.0.


It was almost 9 years ago.

SIP/1.0 was one of the proposals for a session invitation protocol when the
MMUSIC working group first started on that task back in early 1996. It was
very terse and SDP-like. SCIP/1.0 was a proposal that was offered at the
same time. Almost immediately, they were merged together into one protocol,
dubbed "SIP/2.0". To my knowledge, neither of the original protocols ever
escaped research facilities.

> I think I'll currently stick to rfc3261 and hope for the best. Will deal
> with 2543 as and when issues arise...


You shouldn't have to. If you do encounter any such problems, please post
about them to the list so we can register a bug and fix it in the next
release.

/a

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Subject: Re: [Sip] INFO considered harmful
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From: "Dean Willis" <dean.willis@softarmor.com>

> > http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.txt
> > http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.html
>
> Jonathan and I have had "extended discussions" on this. I'm more
> inclined to think that INFO needs specifications for specific usages,
> much as we use "events packages" for SIP Events (SUBSCRIBE, NOTIFY). In
> my own completely non-original way, I call these "INFO Packages".
>
> Anybody else have a third opinion?

Not exactly a third opinion, but here goes: I agree 100% that something must
be done about the current state of INFO. So far, with the one well-known
exception, INFO has just served as a way that people can throw random crap
in the body of a meaningless message so that their implementation has no
chance of interoperating with any other, but can still claim 100% compliance
with IETF proposed standard documents. (The embarassing thing is: they're
not lying.)

I'm not so set on which of the two actions (packages, or restriction to
SIP-T) is correct. However, I find a couple of arguments that Jonathan makes
in the draft extremely compelling:

1. The INFO spec does not actually define any semantics. It
   only says "this acts like a BYE, except that it doesn't tear
   the session down".

2. Any new feature that INFO enables could be defined
   by adding a new method; and, the existence of the INFO
   spec does not provide any ease over doing so.

I'll add a third: despite INFO being an RFC for more than two years, there
have been no additional internet-drafts that suggest alternate legitimate
uses for this method. If it were useful enough to warrant saving, certainly
at least one additional use would have been identified by now.

These arguments would seem to indicate the path of deprecating INFO except
for its only currently approved purpose: ISUP transfer. I can probably be
swayed, but haven't heard any counter-arguments yet.

/a

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References: <3E0B6293.4030600@dynamicsoft.com> <3E0C37AC.1A007145@lmf.ericsson.se>
Subject: Re: [Sip] INFO considered harmful
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From: "Christer Holmberg" <christer.holmberg@lmf.ericsson.se>

> Now, does this mean that we also need to restrict what kind of message
> body types we allow in INVITE, re-INVITE and UPDATE?

Not really. INVITE means something: it means "I'm trying to set up a
session, and here's a description of that session." The first half ("I'm
trying to set up a session") has specific semantics associated with it. The
second half ("here's a description of that session") tells you exactly what
goes in the body.

INFO just says "Here's a framed chunk of data."

There is no valid comparison between the two.

/a

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From mailnull@www1.ietf.org  Sat Dec 28 16:58:09 2002
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From: Orit Levin <orit@radvision.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>
Cc: sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
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I (also) vote for the direction #2 from the draft (i.e. convert INFO into a
proper framework).

I don't like using neither SIP Events nor INFO as general mechanisms for
implementing applications only because they happened to be collocated with
an existing SIP stack.

I do think that both mechanisms are very useful for features closely related
to the SIP call control (once these features are well-defined and
standardized). An alternative would be maintaining an expensive separate
transport channel for each of the light and complementary to the call
control applications...

On a different note, I would like to stress out that in general events vs.
requests (when properly applied) would be different in semantics. In other
words, it doesn't make sense for certain applications to use events and vs.
versa. Each mechanism stands on its own and both are required. 

Orit Levin
Chief Architect
RADVISION
Phone: +1.201.6896330
Video: +1.201.6896430


> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Saturday, December 28, 2002 1:29 AM
> To: Jonathan Rosenberg
> Cc: sip@ietf.org
> Subject: Re: [Sip] INFO considered harmful
> 
> 
> 
> Jonathan Rosenberg wrote:
> > Folks,
> >
> > For some time, I've been complaining about the continued abuse of the
> > INFO method for things that should be done other ways (frequently, not
> > using sip at all). I've written up a summary of the problems and a
> > proposed path forward. Specifically, I'd like to obsolete INFO and
> > replace it with a spec that is approved ONLY for SIP-T.
> >
> > Until the I-D appears, you can pick up a version at:
> >
> > http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.txt
> > http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.html
> 
> Jonathan and I have had "extended discussions" on this. I'm more
> inclined to think that INFO needs specifications for specific usages,
> much as we use "events packages" for SIP Events (SUBSCRIBE, NOTIFY). In
> my own completely non-original way, I call these "INFO Packages".
> 
> I think that the SIP-T usage of INFO is a perfect example of an INFO
> package. However, I think it is NOT the only reasonable use for INFO.
> Unsolicited in-dialog "stuff" is pretty darned useful. All we really
> need is for the usages to be differentiable, negotiable, and
> well-documented. Admittedly, INFO doesn't have any of those three
> characteristics right now, but I think we can probably fix it without
> breaking too much existing stuff.
> 
> Anybody else have a third opinion?
> 
> --
> Dean
> Not Speaking From The Chair.
> 
> _______________________________________________
> Sip mailing list  https://www1.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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inline.

Dean Willis wrote:
> 
> 
> Jonathan Rosenberg wrote:
> 
>> Folks,
>>
>> For some time, I've been complaining about the continued abuse of the 
>> INFO method for things that should be done other ways (frequently, not 
>> using sip at all). I've written up a summary of the problems and a 
>> proposed path forward. Specifically, I'd like to obsolete INFO and 
>> replace it with a spec that is approved ONLY for SIP-T.
>>
>> Until the I-D appears, you can pick up a version at:
>>
>> http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.txt
>> http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.html
> 
> 
> Jonathan and I have had "extended discussions" on this. I'm more 
> inclined to think that INFO needs specifications for specific usages, 
> much as we use "events packages" for SIP Events (SUBSCRIBE, NOTIFY). In 
> my own completely non-original way, I call these "INFO Packages".
> 
> I think that the SIP-T usage of INFO is a perfect example of an INFO 
> package.  However, I think it is NOT the only reasonable use for INFO.
> Unsolicited in-dialog "stuff" is pretty darned useful. All we really 
> need is for the usages to be differentiable, negotiable, and 
> well-documented.

The point I was making in the draft was - why bother? INFO by itself 
provides nothing. If you want differentiable, negotiable and 
well-documented things, define them as new methods. What does INFO give 
you beyond that?


> -- 
> Dean
> Not Speaking From The Chair.
> 

 From the stool then? ;)

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From: "Cullen Jennings" <fluffy@cisco.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>, <sip@ietf.org>
Subject: RE: [Sip] INFO considered harmful
Date: Sat, 28 Dec 2002 21:11:31 -0800
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Proposal 3 might result in moving the problem from INFO to the MESSAGE
message.

Cullen

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> Jonathan Rosenberg
> Sent: Thursday, December 26, 2002 12:12 PM
> To: sip@ietf.org
> Subject: [Sip] INFO considered harmful
>
>
> Folks,
>
> For some time, I've been complaining about the continued abuse of the
> INFO method for things that should be done other ways (frequently, not
> using sip at all). I've written up a summary of the problems and a
> proposed path forward. Specifically, I'd like to obsolete INFO and
> replace it with a spec that is approved ONLY for SIP-T.
>
> Until the I-D appears, you can pick up a version at:
>
> http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.txt
> http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.html
>
> Thanks,
> Jonathan R.
> --
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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> Use sip-implementors@cs.columbia.edu for questions on current sip
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>

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Subject: Re: [Sip] INFO considered harmful
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From: "Cullen Jennings" <fluffy@cisco.com>


> Proposal 3 might result in moving the problem from INFO to the MESSAGE
> message.

Yes, but that's an obvious and intentional abuse of the system, not a
well-intentioned attempt at something "standard." Certain parties may
actually be hellbent on being incompatible, and there's nothing we can do to
fix their attitude. For the only slightly misguided, though, we could make
it at least somewhat obvious what behavior is going to lead to interop
problems.

/a

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On Tuesday, December 24, 2002, at 09:45 AM, Steve Levy wrote:

> At 11:46 AM 12/24/2002 -0500, Jonathan Rosenberg wrote:
>> Early media was not specified in rfc2543. It did say that you should 
>> be prepared to receive media before receiving the answer, which is 
>> still the case for 3264. Another issue was that people used to put 
>> SDP in non-reliable 1xx responses, and then put the same thing in 
>> 2xx. That case is explicitly allowed for in 3261, so it should 
>> interoperate as well.
>>
>> Perhaps you could identify the specific problem you are concerned 
>> about?
>
> my concern was SDP in a 18x followed by another 18x/2xx with different 
> SDP. Wouldnt the SDP in the subsequent 18x/2xx be treated differently 
> by 3264 compliant UAs as opposed to earlier 2543 compliant UAs ?

Because offers and answers need to be sent reliably, you can't put an 
answer in a non-reliable 18x and an offer in a 2xx.  This is spelled 
out in 3261 section 13 as quoted below...

> If the initial offer is in an INVITE, the answer MUST be in a reliable 
> non-failure message from UAS back to UAC which is correlated to that 
> INVITE. For this specification, that is only the final 2xx response to 
> that INVITE. That same exact answer MAY also be placed in any 
> provisional responses sent prior to the answer. The UAC MUST treat the 
> first session description it receives as the answer, and MUST ignore 
> any session descriptions in subsequent responses to the initial > INVITE.

thanks,
-rohan



> thx,
> steve
>
>
>
>> -Jonathan R.
>>
>> Steve Levy wrote:
>>>
>>>> <snip>
>>>> rfc3261 is sufficiently backwards compatible with rfc2543. There is 
>>>> no need to detect versions or anything. Any backwards compatibility 
>>>> functions are built into specific features of rfc3261 that needed 
>>>> to support backwards compatibility. If you just implement rfc3261 
>>>> you will be backwards compatible.
>>> <snip>
>>> broadening the discussion to rfc 3264 (offer/answer), isn't there a 
>>> potential for backward compatibility issues with rfc 2543 
>>> implementations wrt early media ?
>>> thx,
>>> steve
>>
>> --
>> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
>> Chief Scientist                             First Floor
>> dynamicsoft                                 East Hanover, NJ 07936
>> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>> http://www.jdrosen.net                      PHONE: (973) 952-5000
>> http://www.dynamicsoft.com
>>
>> _______________________________________________
>> Sip mailing list  https://www1.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] INFO considered harmful
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Hi,

> >Now, does this mean that we also need to restrict what kind of message
> >body types we allow in INVITE, re-INVITE and UPDATE?
>
> Not really. INVITE means something: it means "I'm trying to set up a
> session, and here's a description of that session." The first half ("I'm
> trying to set up a session") has specific semantics associated with it. The
> second half ("here's a description of that session") tells you exactly what
> goes in the body.
>
> INFO just says "Here's a framed chunk of data."
>
> There is no valid comparison between the two.

I agree that the semantics of the commands are different. However, one IS
allowed to include any kind of message body in the INVITE request, to provide
whatever "extra information" about the session (text, HTML, whatever...), so
someone could argue that whatever stuff they put into the INVITE does provide
"extra information" about the session...

And, since it IS allowed to send re-INVITE, or UPDATE, without actually
updating the session description (the SDP message body) someone could include
that whatever stuff also in those messages...

I am not saying that anyone is doing that TODAY, though...

Regards,

Christer Holmberg
Ericsson Finland


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From: "Adam Roach" <adam@dynamicsoft.com>
To: "Christer Holmberg" <christer.holmberg@lmf.ericsson.se>
Cc: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>, <sip@ietf.org>
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Subject: Re: [Sip] INFO considered harmful
Date: Sun, 29 Dec 2002 12:44:36 -0600
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From: "Christer Holmberg" <christer.holmberg@lmf.ericsson.se>

> And, since it IS allowed to send re-INVITE, or UPDATE, without actually
> updating the session description (the SDP message body) someone could
include
> that whatever stuff also in those messages...

Ah -- I see what you're saying. The proposal was floated to deprecate INFO
except for its currently defined use (ISUP transfer) -- so, you're arguing
that we should deprecate INVITE except for its currently defined uses
(setting up and modifying sessions), and deprecate UPDATE except for its
currently defined use (modifying sessions), right?

That's... interesting.

/a

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From: "Sachin Shenoy" <sachins@netlab.hcltech.com>
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Subject: [Sip] Loop detection; Not a bad thing to have
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Hi,
Loop detection was made optional in RFC 3261. Now it seems=20
that, due to a problem with its processing at stateless proxies,=20
loop-detection cannot be performed correctly (as it is given in RFC=20
3261) at stateless proxies.

I think loop detection is not a bad thing to have. Our problem with
loop detection is not because loop detection itself has problems.
It is because we tried to overuse the branch-id parameter=20
for too many things. The purpose of branch-id parameter has=20
changed over time, but there seems to have been little or no
attempt to redesign mechanism of loop-detection w.r.t. these=20
changes.

I suggest removing the loop detection part from the branch-id
parameter and placing it in a new parameter. Having loop detection
done using a separate parameter would avoid it from interfering
with branch-id's primary use as transaction identifier. All other=20
mechanism remaining the same as given in RFC 3261, I think,
this will allow implementations to support loop detection without=20
much trouble.
=20
Thanks
Sachin


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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=3Diso-8859-1">
<META content=3D"MSHTML 5.50.4134.600" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial>Hi,</FONT></DIV>
<DIV><FONT face=3DArial>Loop detection was made optional in RFC 3261. =
Now it=20
</FONT><FONT face=3DArial>seems </FONT></DIV>
<DIV><FONT face=3DArial>that, due to a problem with its =
processing&nbsp;at=20
stateless </FONT><FONT face=3DArial>proxies, </FONT></DIV>
<DIV><FONT face=3DArial>loop-detection </FONT><FONT face=3DArial>cannot =
be performed=20
correctly (as it is given in RFC </FONT></DIV>
<DIV><FONT face=3DArial>3261) </FONT><FONT face=3DArial>at </FONT><FONT=20
face=3DArial>stateless proxies.</FONT></DIV>
<DIV><FONT face=3DArial></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial>I think loop detection is not a bad thing to =
have. Our=20
problem with</FONT></DIV>
<DIV><FONT face=3DArial>loop detection is not because loop detection =
itself has=20
problems.</FONT></DIV>
<DIV><FONT face=3DArial>It is because we tried to overuse the =
</FONT><FONT=20
face=3DArial>branch-id parameter </FONT></DIV>
<DIV><FONT face=3DArial>for too </FONT><FONT face=3DArial>many things. =
The purpose=20
of branch-id parameter </FONT><FONT face=3DArial>has </FONT></DIV>
<DIV><FONT face=3DArial>changed over time,&nbsp;but</FONT><FONT=20
face=3DArial>&nbsp;there&nbsp;seems to have been little or =
no</FONT></DIV>
<DIV><FONT face=3DArial>attempt&nbsp;to</FONT><FONT =
face=3DArial>&nbsp;redesign=20
mechanism&nbsp;of </FONT><FONT face=3DArial>loop-detection w.r.t. these=20
</FONT></DIV>
<DIV><FONT face=3DArial>changes.</FONT></DIV>
<DIV><FONT face=3DArial></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial>I suggest removing the loop detection part from =
the=20
branch-id</FONT></DIV>
<DIV><FONT face=3DArial>parameter and placing it in a new =
parameter.&nbsp;Having=20
loop detection</FONT></DIV>
<DIV><FONT face=3DArial>done using a&nbsp;separate&nbsp;parameter would =
avoid it=20
from interfering</FONT></DIV>
<DIV><FONT face=3DArial>with branch-id's&nbsp;primary use as transaction =

identifier. </FONT><FONT face=3DArial>All other </FONT></DIV>
<DIV><FONT face=3DArial>mechanism remaining the same as given in RFC =
3261, I=20
think,</FONT></DIV>
<DIV><FONT face=3DArial>this&nbsp;will allow</FONT><FONT=20
face=3DArial>&nbsp;implementations&nbsp;to support </FONT><FONT =
face=3DArial>loop=20
</FONT><FONT face=3DArial>detection&nbsp;without </FONT></DIV>
<DIV><FONT face=3DArial>much trouble.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial>Thanks</FONT></DIV>
<DIV><FONT face=3DArial>Sachin</FONT></DIV>
<DIV><FONT face=3DArial></FONT>&nbsp;</DIV></BODY></HTML>

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From mailnull@www1.ietf.org  Sun Dec 29 20:28:29 2002
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Subject: Re: [Sip] Enrollment in SIP services (and should we undeprecate
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Robert Sparks writes:
 > I see no value at all to resurrecting BASIC.
 > 
 > If you are wanting to provide a shared secret and punt all of the
 > security responsibilities to the transport layer, then express that
 > secret in its native form (plaintext for a password). Anything else
 > you do wastes cycles and encourages people to make bad assumptions
 > about the level of protection they are getting (increasing the risk
 > that it would get foolishly used outside the environment you intended
 > to punt to).

Er, that's what basic does. The reason that the
password is in uuencode format is, I'd assume, to
provide the ability to ship binary passwords, not
security through obscurity.

		 Mike
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From: Mukul Purohit <mpurohit@neomagic.com>
Organization: NeoMagic
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Subject: [Sip] resource reservation
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hi,
I have a basic question after going through RFC 3312, "Integration of 
Resource Management and SIP".

Has the actual resource reservation mechanism been specified (mandated 
or suggested) by IETF?
Also, has it been specified by any other body like 3GPP which has chosen 
SIP for signaling?

Hope to get some quick replies. And if these queries are out of scope of 
this list, then a pointer to proper place will be appriciated.

regards
Mukul

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To: Christer Holmberg <christer.holmberg@lmf.ericsson.se>
CC: Adam Roach <adam@dynamicsoft.com>, sip@ietf.org
Subject: Re: [Sip] INFO considered harmful
References: <3E0B6293.4030600@dynamicsoft.com> <3E0C37AC.1A007145@lmf.ericsson.se> <002001c2aeaa$6fbbb0c0$6601a8c0@dynamicsoft.com> <3E0F3B60.A1D37549@lmf.ericsson.se>
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inline.

Christer Holmberg wrote:
> Hi,
> 
> 
>>>Now, does this mean that we also need to restrict what kind of message
>>>body types we allow in INVITE, re-INVITE and UPDATE?
>>
>>Not really. INVITE means something: it means "I'm trying to set up a
>>session, and here's a description of that session." The first half ("I'm
>>trying to set up a session") has specific semantics associated with it. The
>>second half ("here's a description of that session") tells you exactly what
>>goes in the body.
>>
>>INFO just says "Here's a framed chunk of data."
>>
>>There is no valid comparison between the two.
> 
> 
> I agree that the semantics of the commands are different. However, one IS
> allowed to include any kind of message body in the INVITE request, to provide
> whatever "extra information" about the session (text, HTML, whatever...), so
> someone could argue that whatever stuff they put into the INVITE does provide
> "extra information" about the session...

There is a huge difference that you are ignorning.

The semantics of INVITE are well defined. SIP-guidelines says bodies 
can't change that basic semantic. Same with UPDATE.

INFO has no defined semantic. Thus, it is not currently a violation of 
the spec (and indeed, seemingly encouraged by it) to place all kinds of 
stuff in it. THis is why we HAVE seen abuse of INFO, but not INVITE or 
UPDATE.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Mukul Purohit wrote:
> hi,
> I have a basic question after going through RFC 3312, "Integration of 
> Resource Management and SIP".
> 
> Has the actual resource reservation mechanism been specified (mandated 
> or suggested) by IETF?

There is no mandatory resource reservation scheme. The primary one 
specified so far is RSVP. nsis is another one under development. Groups 
like packetcable have specified their own QoS protocols.

> Also, has it been specified by any other body like 3GPP which has chosen 
> SIP for signaling?

Yes, 3gpp has its own mechanisms that are specific to 3gpp. As I 
understand they are built into the GPRS procedures, and reservation 
occurs only on the wireless link.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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inline.

Sachin Shenoy wrote:
> Hi,
> Loop detection was made optional in RFC 3261. Now it seems
> that, due to a problem with its processing at stateless proxies,
> loop-detection cannot be performed correctly (as it is given in RFC
> 3261) at stateless proxies.
>  
> I think loop detection is not a bad thing to have. Our problem with
> loop detection is not because loop detection itself has problems.
> It is because we tried to overuse the branch-id parameter
> for too many things.

I disagree. Your assessment is quite at odds with my recollection of the 
history here.

The problems we had with loop detection were that it became increasingly 
unclear about what a loop was, as opposed to a spiral. Initially (in 
rfc2543), a loop was any request that arrived at the same proxy (there 
were no spirals). It became clear that this was broken, so we had to 
consider changes in the request URI. Then, we ran into cases were other 
fields would change, such as Route headers, and those had to be taken 
into account as well. Real implementations were frequently getting this 
wrong, hampering interop and proper functioning of applications. As we 
began to realize that more and more headers might need to be taken into 
consideration, it became clear that properly doing loop detection was 
simultaneously ill-defined and complex to implement (potentially 
requiring hashes of the entire message).

That is coupled with the fact that such loop conditions will probably be 
rare in deployed networks. Unlike IP routing, SIP routing is 
considerably more static, and I consider the likelihood of real loops a 
rare one outside of a lab. Thus, its appropriate to handle them in a way 
which perhaps consumes more resources, at the cost of much lower 
complexity, and more importantly, proper operation in the normal case. 
Thats the max-forwards mecahnism. It is much easier to implement, always 
correct in the normal case (i.e., it won't mis-detect loops), and much 
more interoperable.

In other words - this is good engineering practice. We don't worry about 
optimally solving a corner case, we optimize for the baseline case.



> The purpose of branch-id parameter has
> changed over time, but there seems to have been little or no
> attempt to redesign mechanism of loop-detection w.r.t. these
> changes.
>  
> I suggest removing the loop detection part from the branch-id
> parameter and placing it in a new parameter. Having loop detection
> done using a separate parameter would avoid it from interfering
> with branch-id's primary use as transaction identifier. 

I am at a loss as to why you believe this will solve all the ails of 
loop detection. It is very easy to combine the functions of unique 
identification along with the change-detection functions of loop 
detections. You merely include a transaction ID as part of the long hash 
you are already doing. I fail to see why that is hard.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Jonathan Rosenberg writes:
 > There is a huge difference that you are ignorning.
 > 
 > The semantics of INVITE are well defined. SIP-guidelines says bodies 
 > can't change that basic semantic. Same with UPDATE.
 > 
 > INFO has no defined semantic. Thus, it is not currently a violation of 
 > the spec (and indeed, seemingly encouraged by it) to place all kinds of 
 > stuff in it. THis is why we HAVE seen abuse of INFO, but not INVITE or 
 > UPDATE.

  So... s/INFO/X-SIPPING and be done with it?

	Mike
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Subject: RE: [Sip] INFO considered harmful
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Christer said:
> I agree that the semantics of the commands are different. 
> However, one IS allowed to include any kind of message body 
> in the INVITE request, to provide whatever "extra 
> information" about the session (text, HTML, whatever...), so 
> someone could argue that whatever stuff they put into the 
> INVITE does provide "extra information" about the session...
> 
> And, since it IS allowed to send re-INVITE, or UPDATE, 
> without actually updating the session description (the SDP 
> message body) someone could include that whatever stuff also 
> in those messages...

That's actually quite interesting. One can get the same functionality as
INFO using UPDATE methods. For example, SIP-T could encode the mid-session
ISUP and send it using UPDATE.

So, what is the semantic differentiation?

I propose that we've been operating under a logic system where "UPDATE
affects SIP state and INFO doesn't". It might be that this exposes a fallacy
(which one might call "telephone centric" thinking) in our logic. There are
really two different "sessions" in a typical SIP exchange: 1) The SIP-level
session, or "dialog", framed in INVITE and BYE or SUBSCRIBE and BYE, and 2)
the negotiated session, expressed in SDP, which may or may not exist.

Since there may or may not be a negotiated session, and since SIP works just
fine either way, is it reasonable to divide SIP mid-dialog semantics into
"things that we think might affect the negotiated session", and "things that
we think don't"? Perhaps all we're really doing is handing a blob of data to
the application (which exists ABOVE the SIP layer), which then uses that
information to affect its own state, and may consequently affect the SIP
state?

In any case, I think there is a much deeper semantic problem lurking here
that we really haven't thought out.

--
Dean

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] INFO considered harmful
Date: Mon, 30 Dec 2002 13:15:46 -0600
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> The point I was making in the draft was - why bother? INFO by itself 
> provides nothing. If you want differentiable, negotiable and 
> well-documented things, define them as new methods. What does 
> INFO give 
> you beyond that?

The ability to define a new usage without having to make code changes to
one's SIP stack, just as Events allows.

SIP, as you may have noticed, is a fairly complicated protocol. This has
given rise to the distribution of reusable (even open source) SIP stacks.
Personally I'm not up to hacking a new method (into, say, the Vovida stack)
everytime I want to deliver a new type of blob. INFO, constructed as I've
proposed, would let me implement a new INFO package on a stack without
having to rewrite the stack. We'd just need an API to tell the stack what
INFO packages we handled at the application layer, and it could do the rest
of the work without caring what data is being transported in the INFOs.
Events, you may have noticed, has exactly this property.

At some level, this is comparable to the "Why do we use MIME instead of
defining a new transport protocol for each and every possible type of data"
argument.

At its heart, INFO is for delivering APPLICATION SPECIFIC DATA within a
dialog. The SIP state machine does not have to care about that data, just
its delivery (although, as with SIP-T, the application itself may use the
data to influence the SIP state machine via the stacks's APIs). If SIP were
a proper transport protocol, we wouldn't have to define protocol extensions
for every application-specific datum.

I suspect that this working group has gotten very muddled between
"application" and "transport" layers. We have atendency to try and define
application semantics at the protocol level, and that's just wrong. The
protocol should define only protocol semantics.

--
Dean

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Subject: RE: [Sip] INFO considered harmful
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First, I think it is a really good idea to restrict the applicability of SIP
INFO method - I very much agree with the problem defined in Section 2 of
this draft.

That much said, when it comes to solutions I am torn between option number 3
and a more radical alternative - deprecating INFO altogether. While it is
true that INFO was inspired by some of the requirements of SIP-T, over time,
we've found less and less use for INFO. The original vision for SIP+ (an
ancestor of SIP-T) relied on INFO for sending virtually every ISUP message,
but today, there just isn't that much that we do with INFO. Practically
speaking, we could probably find alternatives to INFO for carrying ISUP data
in these cases (which include SUS/RES, INF/INR, some provisional response
cases). Using INFO to send other ISUP messages is frequently misguided, and
can potentially lead to the same sorts of interoperability problems we have
with INFO in other arenas. 

Today, however, RFC3398 does allow the use of INFO in these three cases, so
it's simplest to let the INFO method to exist (option 3) for the time being
- but we should be sure to document, even for ISUP transmissions, the cases
in which it is and is not appropriate. In the long term, though, I think we
should just get rid of it entirely.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Thursday, December 26, 2002 12:12 PM
> To: sip@ietf.org
> Subject: [Sip] INFO considered harmful
> 
> 
> Folks,
> 
> For some time, I've been complaining about the continued abuse of the 
> INFO method for things that should be done other ways 
> (frequently, not 
> using sip at all). I've written up a summary of the problems and a 
> proposed path forward. Specifically, I'd like to obsolete INFO and 
> replace it with a spec that is approved ONLY for SIP-T.
> 
> Until the I-D appears, you can pick up a version at:
> 
> http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.txt
> http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.html
> 
> Thanks,
> Jonathan R.
> -- 
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
> _______________________________________________
> Sip mailing list  https://www1.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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To: "Dean Willis" <dean.willis@softarmor.com>
Cc: "'Christer Holmberg'" <christer.holmberg@lmf.ericsson.se>,
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Subject: RE: [Sip] INFO considered harmful
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Dean,

I think the important part of the observation here
is that if SIP defines a way to transport data,
somebody really, really ought to define the
color/shape/consistency of that data too. To not
do that is to create interoperability problems by
definition. Now there is a management question of
_who_ defines those semantics -- eg, the SIP wg or
somebody else, but I don't think there should be
too much debate that it really ought to be done.
Opening up the door for un(der)defined semantics
seems really dangerous to me. A high energy
barrier for new semantics is probably a Good Thing
at this point as SIP is already a terribly
complicated beast. To end up with an official
backdoor for whatever kruft people want to wrap in
a SIP packet seems, well, like shortcut on the
road to hell.

	Mike

Dean Willis writes:
 > 
 > Christer said:
 > > I agree that the semantics of the commands are different. 
 > > However, one IS allowed to include any kind of message body 
 > > in the INVITE request, to provide whatever "extra 
 > > information" about the session (text, HTML, whatever...), so 
 > > someone could argue that whatever stuff they put into the 
 > > INVITE does provide "extra information" about the session...
 > > 
 > > And, since it IS allowed to send re-INVITE, or UPDATE, 
 > > without actually updating the session description (the SDP 
 > > message body) someone could include that whatever stuff also 
 > > in those messages...
 > 
 > That's actually quite interesting. One can get the same functionality as
 > INFO using UPDATE methods. For example, SIP-T could encode the mid-session
 > ISUP and send it using UPDATE.
 > 
 > So, what is the semantic differentiation?
 > 
 > I propose that we've been operating under a logic system where "UPDATE
 > affects SIP state and INFO doesn't". It might be that this exposes a fallacy
 > (which one might call "telephone centric" thinking) in our logic. There are
 > really two different "sessions" in a typical SIP exchange: 1) The SIP-level
 > session, or "dialog", framed in INVITE and BYE or SUBSCRIBE and BYE, and 2)
 > the negotiated session, expressed in SDP, which may or may not exist.
 > 
 > Since there may or may not be a negotiated session, and since SIP works just
 > fine either way, is it reasonable to divide SIP mid-dialog semantics into
 > "things that we think might affect the negotiated session", and "things that
 > we think don't"? Perhaps all we're really doing is handing a blob of data to
 > the application (which exists ABOVE the SIP layer), which then uses that
 > information to affect its own state, and may consequently affect the SIP
 > state?
 > 
 > In any case, I think there is a much deeper semantic problem lurking here
 > that we really haven't thought out.
 > 
 > --
 > Dean
 > 
 > _______________________________________________
 > Sip mailing list  https://www1.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] Enrollment in SIP services (and should we undeprecate basic
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Jonathan Rosenberg writes:
 > I have been thinking about undeprecating basic, but for a totally 
 > different reason (which was suggested by Christian some time back). When 
 > you do authentication over TLS, there is no security benefits to digest 
 > as opposed to basic. However, with basic, it is easier to integrate with 
 > back-end AAA systems, because you don't need to assume that they've 
 > stored the password in the form described in rfc 2617.
 > 
 > Of course, if the server doing the authentication is several hops beyond 
 > the one you've terminated your TLS connection to, you've got problems. 
 > The combination of sips and basic would always work, however.

While not passing judgement one way or the other,
basic is clearly less secure than digest even with
TLS. With digest, intermediate devices don't see
the cleartext password -- including a directly
adjacent (SIPwise) proxy. The AAA benefit you list
specifically takes advantage of lowering the
security bar (eg, so that the proxy can form a
CHAP/PAP authenticator to send to a AAA). My sense
is that if there are reasons people want to take
advantage of basic, it might be better to consider
the root causes first rather than rushing headlong
into undeprecating (precating?) basic.

		   Mike
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One comment:

Robert Sparks writes:
 > I don't understand the last statement. A system that accepts digest
 > credentials is not required to keep its local knowledge of the password
 > in the hash recommended in 2617.

The problem is that AAA's expect either a CHAP or
PAP formated authenticator. Digest, while it
doesn't chip the cleartext password ain't the same
format so you have a problem. Basic allows you to
hack up an authenticator at the proxy. Maybe
EAP+digest is our friend. Or maybe not.

	Mike
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From: "Dean Willis" <dean.willis@softarmor.com>

> SIP, as you may have noticed, is a fairly complicated protocol. This has
> given rise to the distribution of reusable (even open source) SIP stacks.
> Personally I'm not up to hacking a new method (into, say, the Vovida
stack)
> everytime I want to deliver a new type of blob. INFO, constructed as I've
> proposed, would let me implement a new INFO package on a stack without
> having to rewrite the stack. We'd just need an API to tell the stack what
> INFO packages we handled at the application layer, and it could do the
rest
> of the work without caring what data is being transported in the INFOs.

So, your fundamental argument springs from the desire to have the API to
most SIP stacks include a function called something like
"registerNewInfoPackage" instead of  "registerNewMethod".

For some reason, that doesn't strike me as particularly compelling.

/a

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inline.

Dean Willis wrote:
>>The point I was making in the draft was - why bother? INFO by itself 
>>provides nothing. If you want differentiable, negotiable and 
>>well-documented things, define them as new methods. What does 
>>INFO give 
>>you beyond that?
> 
> 
> The ability to define a new usage without having to make code changes to
> one's SIP stack, just as Events allows.

Code to handle INFO goes somewhere.

> 
> SIP, as you may have noticed, is a fairly complicated protocol. This has
> given rise to the distribution of reusable (even open source) SIP stacks.
> Personally I'm not up to hacking a new method (into, say, the Vovida stack)
> everytime I want to deliver a new type of blob. INFO, constructed as I've
> proposed, would let me implement a new INFO package on a stack without
> having to rewrite the stack. 

How is this different from allowing new methods, and passing off 
handling of that method to registered callbacks for each method?? Not 
different at all.


> We'd just need an API to tell the stack what
> INFO packages we handled at the application layer, 

Or, you could have an API to tell the stack what methods are handled.

> and it could do the rest
> of the work without caring what data is being transported in the INFOs.

What work?? There is no semantics associated with INFO. None. Zero. 
Nada. Zip.

> Events, you may have noticed, has exactly this property.

The big huge difference I have been trying to point out is that EVENTS 
DEFINES SEMANTICS, whereas INFO doesnt.



> 
> At some level, this is comparable to the "Why do we use MIME instead of
> defining a new transport protocol for each and every possible type of data"
> argument.

MIME provides semantics and syntax that are independent of the type.
SIP events provides semantics and syntax that are independent of the 
package.
INFO provides nothing.

> 
> At its heart, INFO is for delivering APPLICATION SPECIFIC DATA within a
> dialog. The SIP state machine does not have to care about that data, just
> its delivery (although, as with SIP-T, the application itself may use the
> data to influence the SIP state machine via the stacks's APIs). If SIP were
> a proper transport protocol, we wouldn't have to define protocol extensions
> for every application-specific datum.

SIP is not a protocol for shipping data between applications. I do not 
think we should encourage the arbitrary transport of stuff in SIP.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Michael Thomas wrote:
> Jonathan Rosenberg writes:
>  > There is a huge difference that you are ignorning.
>  > 
>  > The semantics of INVITE are well defined. SIP-guidelines says bodies 
>  > can't change that basic semantic. Same with UPDATE.
>  > 
>  > INFO has no defined semantic. Thus, it is not currently a violation of 
>  > the spec (and indeed, seemingly encouraged by it) to place all kinds of 
>  > stuff in it. THis is why we HAVE seen abuse of INFO, but not INVITE or 
>  > UPDATE.
> 
>   So... s/INFO/X-SIPPING and be done with it?

Sorry for being dense, but I don't follow what you mean here.

If you are asking whether I am suggesting that people define their stuff 
as new methods, then yes, that is what I am suggesting.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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From mailnull@www1.ietf.org  Mon Dec 30 16:56:26 2002
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A few additional notes inline.

Adam Roach wrote:
> "Prakash R" <rprakash@cdotb.ernet.in> wrote in message
> news:3E0C50C8.4060302@cdotb.ernet.in...
> 
> 
>>An rfc3261 stack compliant stack will be 100% backward compatible with a
>>rfc2543 stack? I can just get the two stacks to interwork without any
>>issues at all? All the rfc2543 gear already deployed will work smoothly
>>with a new stack? Hope this is true... Makes life simple for us.
> 
> 
> Yes, that was a design goal for RFC3261. That said, there are "holes" in
> RFC2543 -- places in which the protocol was underspecified. Interop tests
> and operational experience have revealed these holes; all the ones that we
> have discovered to date have been fixed in RFC3261. So, there are ways to
> implement an RFC2543 stack that won't work in general (even with other
> RFC2543 stacks). There is obviously no magic fix for these problems.

There are also some never-implemented or never-used features of rfc2543 
that were dropped, including PGP and absolute time in register 
expirations. Since these were never used outside of a lab, you shouldn't 
see them in real life.

> 
> 
>>I was earlier thinking of using this branch-tag to check which rfc the
>>remote stack was complying with. Was not clear with all the scenarios
>>though.
> 
> 
> 
> Don't do that. Don't even try. You still encounter a huge number of stacks
> that use the branch parameter but don't do 100% of RFC3261.

Right. I also tried to explain this.

> 
> 
>>BTW, When will the protocol version be incremented? What sort of changes
>>will necessitate a protocol version upgrade?
> 
> 
>>From a practical perspective, I expect that such a change will be made only
> if enhancements to the protocol cannot be added in a way that is backwards
> compatible.

You mean, never?

We have no plans to ever rev the version. It makes no sense. Instead, we 
  have looked at each feature, one at a time, and addressed backwards 
compatibility for it using mechanisms appropriate for that feature. That 
means that implementations can support grades of behavior (a very common 
case) rather than lumping everyting into a version number, which would 
effectively require dual-stacks. With our approach, there are no dual 
stacks.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Rohan Mahy wrote:
> 
> On Tuesday, December 24, 2002, at 09:45 AM, Steve Levy wrote:
> 
>> At 11:46 AM 12/24/2002 -0500, Jonathan Rosenberg wrote:
>>
>>> Early media was not specified in rfc2543. It did say that you should 
>>> be prepared to receive media before receiving the answer, which is 
>>> still the case for 3264. Another issue was that people used to put 
>>> SDP in non-reliable 1xx responses, and then put the same thing in 
>>> 2xx. That case is explicitly allowed for in 3261, so it should 
>>> interoperate as well.
>>>
>>> Perhaps you could identify the specific problem you are concerned about?
>>
>>
>> my concern was SDP in a 18x followed by another 18x/2xx with different 
>> SDP. Wouldnt the SDP in the subsequent 18x/2xx be treated differently 
>> by 3264 compliant UAs as opposed to earlier 2543 compliant UAs ?
> 
> 
> Because offers and answers need to be sent reliably, you can't put an 
> answer in a non-reliable 18x and an offer in a 2xx.  This is spelled out 
> in 3261 section 13 as quoted below...

All of this supposed "rfc2543 behavior" was never documented in rfc2543, 
and is rather one of several interpretations people have made about the 
spec. Even supposedly "rfc2543-compliant" UAs probably wont interwork 
for features like this.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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Michael Thomas wrote:
> Jonathan Rosenberg writes:
>  > I have been thinking about undeprecating basic, but for a totally 
>  > different reason (which was suggested by Christian some time back). When 
>  > you do authentication over TLS, there is no security benefits to digest 
>  > as opposed to basic. However, with basic, it is easier to integrate with 
>  > back-end AAA systems, because you don't need to assume that they've 
>  > stored the password in the form described in rfc 2617.
>  > 
>  > Of course, if the server doing the authentication is several hops beyond 
>  > the one you've terminated your TLS connection to, you've got problems. 
>  > The combination of sips and basic would always work, however.
> 
> While not passing judgement one way or the other,
> basic is clearly less secure than digest even with
> TLS. With digest, intermediate devices don't see
> the cleartext password -- including a directly
> adjacent (SIPwise) proxy. The AAA benefit you list
> specifically takes advantage of lowering the
> security bar (eg, so that the proxy can form a
> CHAP/PAP authenticator to send to a AAA). 

OK, a good point.

> My sense
> is that if there are reasons people want to take
> advantage of basic, it might be better to consider
> the root causes first rather than rushing headlong
> into undeprecating (precating?) basic.

I wasn't arguing to rush into anything. It was just a thought.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Jonathan Rosenberg writes:
 > 
 > 
 > Michael Thomas wrote:
 > > Jonathan Rosenberg writes:
 > >  > There is a huge difference that you are ignorning.
 > >  > 
 > >  > The semantics of INVITE are well defined. SIP-guidelines says bodies 
 > >  > can't change that basic semantic. Same with UPDATE.
 > >  > 
 > >  > INFO has no defined semantic. Thus, it is not currently a violation of 
 > >  > the spec (and indeed, seemingly encouraged by it) to place all kinds of 
 > >  > stuff in it. THis is why we HAVE seen abuse of INFO, but not INVITE or 
 > >  > UPDATE.
 > > 
 > >   So... s/INFO/X-SIPPING and be done with it?
 > 
 > Sorry for being dense, but I don't follow what you mean here.
 > 
 > If you are asking whether I am suggesting that people define their stuff 
 > as new methods, then yes, that is what I am suggesting.

Sorry for being too flippant, but if we're going
to have a junk heap for undefined semantics, some
truth in advertising might be in order. Hence X-.

	 Mike
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Speaking as an acknowledged INFO abuser (step 1 of the 12-step program?)...

Dean hit the nail on the head:

> -----Original Message----- From: Dean Willis [mailto:dean.willis@softarmor.com]
> > [snip] What does INFO give you beyond that?
> 
> The ability to define a new usage without having to make code 
> changes to one's SIP stack, just as Events allows.

What are my options when I want to *try* something new in the SIP world.  I can:

1. Write an I-D, hoping to get an RFC published, before I have any road-time or clue how my idea will work in the real-world.

2. Hack my own method into my own infrastructure, and hope I never go outside my lab or tangentially interoperate with other SIP services.  [N.B.: If those other SIP services needed to be aware of my change, they would be aware of my change BY DEFINITION.]

3. Hack my own APPLICATION payload onto a generic TRANSPORT mechanism that is well-known by the rest of the infrastructure.

I'm open for Option 4...


What we need is some way to do experiments without breaking stuff or having to hack everything.



I am firmly on the record that option 1 is EVIL.  IMHO it is a waste of resources to toss out ideas hoping to see what sticks.  There needs to be a way of trying stuff out without burning WG cycles.  Stuff doesn't need to be fully baked (in fact, it shouldn't be fully baked so when the WG sees the inevitable fatal flaw, no one is too upset to change).  However, stuff should have seen the light of the real world to both work out the obvious "oh no's!" and the not so obvious "it should work, but it doesn't!".

Option 2 is just a non-starter.

I can speak to Option 3 from personal experience.  We used INFO for information reporting from media servers.  Is that the right answer today?  No, of course not.  NOTIFY is the right answer.  However, NOTIFY didn't exist.  More to the point, we have a lot of road-time on what does work, and what doesn't (e.g., HTTP does NOT work for everyone, re-INVITE takes too long and the dialog *isn't* changing, etc.).

Now that we have real-world experience, we are in a position to publish the I-D, including using "modern" techniques, and go forward through the WG gauntlet.  


What is your proposal without INFO?


Now my turn for a bit of soap-boxing:

I absolutely, positively consider the statement that the use of INFO results in a lack of interoperability to be totally irrelevant.  I posit this for two reasons.  The first is that all of the "extensions" offered in "The Session Initiation Protocol (SIP) INFO Method Considered Harmful" as creating non-interoperable extensions actually were proposals to ENSURE INTEROPERABLE extensions.  One cannot say, "using INFO is not interoperable" at the same time one says, "we will not allow anyone to define interoperable uses of INFO."  That is like saying, "Only people that have sea duty can become an admiral, but you will never be allowed to have sea duty."

One escape for that line of thinking is Dean's proposal for INFO packages.  The problem here is that proposal would cast experiments in stone, like X-INFO.  That smells like SOAP over SIP.

The second reason I consider the statement of non-interoperability to be irrelevant is if one looks at two of the cited usages, "DTMF digits" and "media servers", *only* the client and the server are even AWARE of the capabilities.  THERE IS NO THIRD PARTY THAT NEEDS TO NEGOTIATE THE CAPABILITY, EITHER "IN" OR "OUT".  BY DEFINITION, THERE IS NO INTEROPERABILITY ISSUE.  Why?  Because the drafts describe how to interoperate, and messages will only ever originate and terminate at "aware" devices.

Again, and I'm sure I'll have to point out this paragraph when people flame about the last paragraph, I see the use of INFO for experimental stuff.  After the experiment is done, I will lay wagers that one of the following outcomes will occur.

  1.  What's being tried is really useful and fits into the SIP
      Framework.  Very likely to be a new method.

  2.  What's being tried is really useful and something else
      already covers it, e.g., NOTIFY for digit reporting.

  3.  What's being tried is really useful but isn't SIP, e.g.,
      bulk transfer of data, which is what FTP and HTTP are for.

  4.  What's being tried just doesn't work.  Don't waste WG
      time on it.

In fact, I would be one of the first to jump up and say "Whoa" when someone publishes a "we want to use INFO for..." I-D's.  Why?  Because one of the above 4 items really end up being the right answer.  As hgs says, we have an infinite method namespace.

On the other hand, let's say you don't want to use an infinite method namespace.  In that case, I would put my weight behind Dean's suggestion to have INFO packages.  Otherwise, and great apologies to Adam, but Christer is 100% correct.  Banning INFO will move the problem to MESSAGE, INVITE, and UPDATE.  We even saw a proposal to use SUBSCRIBE to change the state of the UAS as a way to "work around" the "INFO problem."  Let's not make things worse!
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"The guidelines for authors of SIP extensions[7] clearly rules some of them, such as media server control, out of scope."

[7] clearly, and IMHO correctly, rules using SIP for device control to be out of scope.  However, I would offer that media server control, e.g., "I want to route a SIP message to my farm of media servers (a routing problem), and possibly to a bunch in parallel (a forking problem) to the 'user' that represents an send-only source (a.k.a., an announcement)" is in scope.

For that matter, the words "media server" never appear in [7] :-)



[7] Rosenberg, J., "Guidelines for Authors of Extensions to the Session Initiation Protocol (SIP)", draft-ietf-sip-guidelines-06 (work in progress), November 2002. 

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Thursday, December 26, 2002 3:12 PM
> To: sip@ietf.org
> Subject: [Sip] INFO considered harmful
> 
> 
> Folks,
> 
> For some time, I've been complaining about the continued abuse of the 
> INFO method for things that should be done other ways 
> (frequently, not 
> using sip at all). I've written up a summary of the problems and a 
> proposed path forward. Specifically, I'd like to obsolete INFO and 
> replace it with a spec that is approved ONLY for SIP-T.
> 
> Until the I-D appears, you can pick up a version at:
> 
> http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.txt
> http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.html
> 
> 
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Greetings,

A lurker speaks!

I've read Jonathan's draft and have some comments.  Please be advised that
these comments are meant to be constructive although they might seem a bit
heavy handed at first reading.

I would take issue with the basic premise of the draft, which is that the
SIP and SIPPING groups need to have control over every bit that passes
through a SIP message.  It is not possible for the standard to anticpate
every situation that people might want to use the protocol.  It's
interesting how many times I've heard WG members say, "thats not what I
intended it for".  The beauty of SIP is that its basic framework and
semantic level endear it to so many different signaling situations.  The use
of INFO to supplement the corners of specific applications is a very
important part of that versatility.

JR cites three specific problems with INFO:

Lack of interoperability: This is not a problem.  Since the INFO contents
are explicitly outside the standard, this must be assumed by the
implementor.  If they persist with INFO to implement some signaling
situtations, there are likely other good reasons for doing so.

Inappropriateness: As I've already stated, I do not feel that the SIP and
SIPPING working groups should (let alone can) realistically expect that they
can determine how and when the protocol will be applied in applications at
this late date.

Incorrectness: Jonathan is eluding to the possiblity of errors being present
in the protocol extensions.  Well, this is not the SIP and SIPPING groups
problem.  These extensions are outside the scope of the standard.  If the
purveyors of the extension design the error, they're responsible for it, not
the SIP and SIPPING groups.

As has already been noted in this thread, there are a number of fine
implementations (including Dynamicsoft) in the field, many, if not all,
including a punt to the application of the INFO body.  This is very useful
functionality to implementors and as such it makes very little sense to
deprecate this method.  Another will take its place as the abused.  For
example, I've had discussions with people about doing extensions using
SUBSCRIBE/NOTIFY that had nothing whatever to do with events just because of
the "political incorrectness" of INFO.  The point is, what difference does
it make.  SIP and SIPPING can't control every use of the protocol and
shouldnt try.  To this end I would propose another radical approach to this
problem.

Limit the scope of the SIP and SIPPING groups to activities having to do
with the base protocol, e.g. new methods and headers.  Handling extensions
can fall into two buckets 1) small extensions that handle corner cases in
some signaling situations that aren't covered by the SIP spec and 2) major
extensions that either add large, application-specific sets of signaling
function or have really nothing to do with signaling.  These latter
extensions should be separated from SIP and SIPPING within IETF and handled
in different WGs.

FM

Frank W. Miller, Ph.D.
Chief Technical Officer
sentitO Networks, Inc.
fmiller@sentito.com

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Inline.

Orit.

> The big huge difference I have been trying to point out is that EVENTS
> DEFINES SEMANTICS, whereas INFO doesnt.
> MIME provides semantics and syntax that are independent of the type.
> SIP events provides semantics and syntax that are independent of the
> package.
> INFO provides nothing.

INFO provides "minimal" but very important semantics: asynchronous data
reliably follows the established SIP path.

Another solution would be to have each pair of UAs registered to each
other's event packages (and using EVENTs minimal semantics only).

Would it be the preferred way to go???


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I don't see the major problem of defining new methods. There seem to be 
two issues: programming and naming.

I believe that modern stacks, like those based on the JAIN SIP API, have 
no problems dealing with new methods. (Somebody who has more day-to-day 
experience with this should confirm this...)

Stack writers (for UA stacks) need to get used to the idea that new 
methods come along, so they should make it easy to register support for 
new (non-INVITE) methods. After all, this is pretty much just a table 
lookup.

In terms of naming, there are at least two sensible approaches:

- If you believe that the method may at some point be more generally 
applicable, give it a regular-looking SIP method name. If it doesn't get 
standardized but is deployed on a proprietary basis, your only 'problem' 
is that you may need to monitor the WGs to protest should somebody pick 
that name for something different. I suspect nobody wants to cause 
completely unnecessary interop problems, given that there is no scarcity 
of names. (We could have solved this problem differently and less 
informally, with the aid of IANA, but I've said my piece on that.)

The best solution is clearly to not deploy stuff until there is some 
generally-agreed upon sense that this is stable. (Presumably, this means 
that it's past the IESG, at least.) Since that can take a few years, 
unless your affiliation reads 3GPP, this is asking a bit, but that's a 
different discussion...

- If you believe that this is strictly your own proprietary thing that 
you don't want to bother documenting for whatever reason, make up your 
own name, such as "SNOWSHORE-EXPLODE-TERMINAL". Again, I wish there was 
a better mechanism for this, but that's a different discussion. 
Obviously, interoperability isn't going to be in the cards, but at least 
it is very obvious and obvious immediately that this is a non-standard 
extension, with a very well-understood error message (501) and 
negotiation mechanism (Allow). (No need to peek deep inside some generic 
INFO message and try to figure out what error indication would fit and 
no need to invent another Allow-* mechanism.)

I'm not advocating the second approach; the INFO abuse seems to 
indicate, however, that the current approach leaves implementors short.

(I don't think the approach in 
draft-narten-iana-experimental-allocations-03.txt is particularly 
applicable here since it assumes scarcity and advocates re-use of a 
small number range, which we don't need or want. We really don't want to 
designate INFO1, INFO2, INFO3, ..., INFO63 here...)

Henning


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Orit Levin wrote:
> Inline.
> 
> Orit.
> 
> 
>>The big huge difference I have been trying to point out is that EVENTS
>>DEFINES SEMANTICS, whereas INFO doesnt.
>>MIME provides semantics and syntax that are independent of the type.
>>SIP events provides semantics and syntax that are independent of the
>>package.
>>INFO provides nothing.
> 
> 
> INFO provides "minimal" but very important semantics: asynchronous data
> reliably follows the established SIP path.

Huh?

Every new SIP method is reliable, it inherits the SIP transaction state 
machine for non-invite. Follows the established path? Every in-dialog 
method would have this property, it is method independent. Asynchronous? 
  You can send any method at any time.

Thus, nothing you have pointed out above is different between INFO and 
any other new method we might introduce.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Orit Levin <orit@radvision.com>
From: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] INFO considered harmful
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At 12:48 AM 12/31/2002 -0500, Jonathan Rosenberg wrote:
>Orit Levin wrote:
>>>The big huge difference I have been trying to point out is that EVENTS
>>>DEFINES SEMANTICS, whereas INFO doesnt.
>>>MIME provides semantics and syntax that are independent of the type.
>>>SIP events provides semantics and syntax that are independent of the
>>>package.
>>>INFO provides nothing.
>>
>>INFO provides "minimal" but very important semantics: asynchronous data
>>reliably follows the established SIP path.
>
>Huh?
>
>Every new SIP method is reliable, it inherits the SIP transaction state 
>machine for non-invite. Follows the established path? Every in-dialog 
>method would have this property, it is method independent. 
>Asynchronous?  You can send any method at any time.
>
>Thus, nothing you have pointed out above is different between INFO and any 
>other new method we might introduce.

I think the key point is that there is a difference between transport 
protocol semantic and application semantic. The transport protocol semantic 
expressed in INFO is "Here is some data that will be used by the 
application. It is not important to the transport protocol, except that the 
transport protocol is expected to deliver it reliably". INFO says NOTHING 
about the APPLICATION level semantic -- that's up to what goes IN the INFO 
payloads. The question is -- do we try to rigidly define application-level 
semantics here? Do we define that there ARE no such applications possible? 
Or do we design a framework by which application-level semantics can be 
expressed outside of the protocol definition?

And as for "Just add another method to your SIP stack". Let's say I have a 
mobile phone and the SIP stack is burned into ROM and exposes only a 
simplistic transactional API. This is only likely to happen, oh, 100 
million times or so over the next two years . . .  Now, just exactly how is 
my BREW or JTME application going to go about extending the SIP stack to 
support another method? Will the evil wireless operator networks even pass 
another method? Neither is happening anytime soon, I think . . .

--
Dean


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Subject: Re: [Sip] Loop detection; Not a bad thing to have
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Please find my comments inline.


>I disagree. Your assessment is quite at odds with my recollection of the 
>history here.
>
>The problems we had with loop detection were that it became increasingly 
>unclear about what a loop was, as opposed to a spiral. Initially (in 
>rfc2543), a loop was any request that arrived at the same proxy (there 
>were no spirals). It became clear that this was broken, so we had to 
>consider changes in the request URI. Then, we ran into cases were other 
>fields would change, such as Route headers, and those had to be taken into 
>account as well. Real implementations were frequently getting this wrong, 
>hampering interop and proper functioning of applications. As we began to 
>realize that more and more headers might need to be taken into 
>consideration, it became clear that properly doing loop detection was 
>simultaneously ill-defined and complex to implement (potentially requiring 
>hashes of the entire message).

I have to admit I am quite unaware (ignorant) about these real-world problems.


>That is coupled with the fact that such loop conditions will probably be 
>rare in deployed networks. Unlike IP routing, SIP routing is considerably 
>more static, and I consider the likelihood of real loops a rare one 
>outside of a lab. Thus, its appropriate to handle them in a way which 
>perhaps consumes more resources, at the cost of much lower complexity, and 
>more importantly, proper operation in the normal case. Thats the 
>max-forwards mecahnism. It is much easier to implement, always correct in 
>the normal case (i.e., it won't mis-detect loops), and much more interoperable.

I have to take your word on this (that loops are rare). But sure it creates 
havoc once
it happens. (Simple tests where request from UA looped between 2 proxies 
ran for more
than 10 minutes. Where re transmissions of original INVITE from UA, the 
"Too Many hops" response and ACK for this just kept on circulating).


>In other words - this is good engineering practice. We don't worry about 
>optimally solving a corner case, we optimize for the baseline case.

I agree with this. Then can we assume (or hope) that in future standard
loop detection would be deprecated?

>>
>>I suggest removing the loop detection part from the branch-id
>>parameter and placing it in a new parameter. Having loop detection
>>done using a separate parameter would avoid it from interfering
>>with branch-id's primary use as transaction identifier.
>
>I am at a loss as to why you believe this will solve all the ails of loop 
>detection. It is very easy to combine the functions of unique 
>identification along with the change-detection functions of loop 
>detections. You merely include a transaction ID as part of the long hash 
>you are already doing. I fail to see why that is hard.

Here I was trying to address (solve) a particular problem where stateless
proxies cannot do loop detection. (There was an earlier mail which stated the
problem subject:"[Sip] BranchID in CANCEL & INVITE"). Stateless proxies
have to maintain same branch for CANCEL & INVITE. Now if stateless proxies
try to do loop detection, the branch could turn out to be different for CANCEL
& INVITE. (Loop detection part calculated for CANCEL could be different from
the loop detection part calculated for INVITE). This is why I suggested moving
the loop detection part out of BranchId.

I mentioned this solution being quite unaware of the real problems plaguing
loop detection.

Again, if this is some feature that has lot of inherent problems with it, I 
think
it is better to take it off completely.

Thanks
Sachin






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Cc: Robert Sparks <rsparks@dynamicsoft.com>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Cullen Jennings <fluffy@cisco.com>, sip@ietf.org
Subject: Re: [Sip] Enrollment in SIP services (and should we undeprecate
	basic auth)
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Michael Thomas writes:

 > The problem is that AAA's expect either a CHAP or
 > PAP formated authenticator. Digest, while it
 > doesn't chip the cleartext password ain't the same
 > format so you have a problem. Basic allows you to
 > hack up an authenticator at the proxy. Maybe
 > EAP+digest is our friend. Or maybe not.

what is wrong with draft-sterman-aaa-sip-00.txt?  we use it with digets
authentication without any problems.  leading radius servers support it
as well as all good sip proxies.

-- juha
_______________________________________________
Sip mailing list  https://www1.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
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Date: Tue, 31 Dec 2002 18:43:16 +0530
From: "Nataraju A.B." <natarajuab@huawei.com>
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Hi, 
I have a query regarding the server/user-agent coding format:
The ABNF encoding specifies it as : 

Server           =  "Server" HCOLON server-val *(LWS server-val)
server-val       =  product / comment
product          =  token [SLASH product-version]
product-version  =  token

whereas teh SIP-rfc says it should be 
 Server: HomeServer v2
CanI know the correct format to be used. 

If the SIP format is to be used then is it product version is a mandatory argument. 

I feel the example given might be incorrect.

It must have been like Server: HomeServer /v2

Regards,
-------------------------------------------
Nataraju A.B.
Huawei Technologies India Pvt. Ltd.,
Tel : +91-80-5216824/6/7 Xtn 142
-------------------------------------------
  ----- Original Message ----- 
  From: Juha Heinanen 
  To: Michael Thomas 
  Cc: Robert Sparks ; Jonathan Rosenberg ; Cullen Jennings ; sip@ietf.org 
  Sent: Tuesday, 31 December 2002 17-45
  Subject: Re: [Sip] Enrollment in SIP services (and should we undeprecate basic auth)


  Michael Thomas writes:

   > The problem is that AAA's expect either a CHAP or
   > PAP formated authenticator. Digest, while it
   > doesn't chip the cleartext password ain't the same
   > format so you have a problem. Basic allows you to
   > hack up an authenticator at the proxy. Maybe
   > EAP+digest is our friend. Or maybe not.

  what is wrong with draft-sterman-aaa-sip-00.txt?  we use it with digets
  authentication without any problems.  leading radius servers support it
  as well as all good sip proxies.

  -- juha
  _______________________________________________
  Sip mailing list  https://www1.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

--Boundary_(ID_hzMqxDpSi1EcthsVzTQLrQ)
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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<META http-equiv=Content-Type content="text/html; charset=iso-8859-1">
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<BODY bgColor=#ffffff>
<DIV><FONT color=#008000>Hi, </FONT></DIV>
<DIV><FONT color=#008000>I have a query regarding the server/user-agent coding 
format:</FONT></DIV>
<DIV><FONT color=#008000>The ABNF encoding specifies it as :</FONT>&nbsp;</DIV>
<DIV><SPAN lang=EN-US 
style="FONT-SIZE: 10pt; BACKGROUND: yellow; FONT-FAMILY: 'Courier New'; mso-bidi-font-size: 12.0pt; mso-highlight: yellow"></SPAN>&nbsp;</DIV>
<DIV><SPAN><FONT color=#008000>Server<SPAN 
style="mso-spacerun: yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
</SPAN>=<SPAN style="mso-spacerun: yes">&nbsp; </SPAN>"Server" HCOLON server-val 
*(LWS server-val)<?xml:namespace prefix = o ns = 
"urn:schemas-microsoft-com:office:office" /><o:p></o:p></FONT></SPAN></DIV>
<DIV class=MsoNormal><SPAN lang=EN-US 
style="FONT-SIZE: 10pt; BACKGROUND: yellow; FONT-FAMILY: 'Courier New'; mso-bidi-font-size: 12.0pt; mso-highlight: yellow"><FONT 
color=#008000>server-val<SPAN 
style="mso-spacerun: yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </SPAN>=<SPAN 
style="mso-spacerun: yes">&nbsp; </SPAN>product / 
comment<o:p></o:p></FONT></SPAN></DIV>
<DIV class=MsoNormal><SPAN lang=EN-US 
style="FONT-SIZE: 10pt; BACKGROUND: yellow; FONT-FAMILY: 'Courier New'; mso-bidi-font-size: 12.0pt; mso-highlight: yellow"><FONT 
color=#008000>product<SPAN 
style="mso-spacerun: yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
</SPAN>=<SPAN style="mso-spacerun: yes">&nbsp; </SPAN>token [SLASH 
product-version]<o:p></o:p></FONT></SPAN></DIV>
<DIV class=MsoNormal><SPAN lang=EN-US 
style="FONT-SIZE: 10pt; BACKGROUND: yellow; FONT-FAMILY: 'Courier New'; mso-bidi-font-size: 12.0pt; mso-highlight: yellow"><FONT 
color=#008000>product-version<SPAN style="mso-spacerun: yes">&nbsp; 
</SPAN>=<SPAN style="mso-spacerun: yes">&nbsp; </SPAN>token</FONT></SPAN></DIV>
<DIV class=MsoNormal><SPAN lang=EN-US 
style="FONT-SIZE: 10pt; BACKGROUND: yellow; FONT-FAMILY: 'Courier New'; mso-bidi-font-size: 12.0pt; mso-highlight: yellow"><FONT 
color=#008000></FONT></SPAN>&nbsp;</DIV>
<DIV><FONT color=#008000>whereas teh SIP-rfc says it should be </FONT></DIV>
<DIV><FONT color=#ff0000>&nbsp;<SPAN 
style="mso-fareast-font-family: 'MS Mincho'">Server: HomeServer 
v2</SPAN></FONT></DIV>
<DIV><SPAN style="mso-fareast-font-family: 'MS Mincho'"><o:p><FONT 
color=#008000>CanI know the correct format to be used. 
</FONT></o:p></SPAN></DIV>
<DIV><SPAN style="mso-fareast-font-family: 'MS Mincho'"><o:p><FONT 
color=#008000></FONT></o:p></SPAN>&nbsp;</DIV>
<DIV><SPAN style="mso-fareast-font-family: 'MS Mincho'"><o:p><FONT 
color=#008000>If the SIP format is to be used then is it product version is a 
mandatory argument. </FONT></o:p></SPAN></DIV>
<DIV><SPAN style="mso-fareast-font-family: 'MS Mincho'"><o:p><FONT 
color=#008000></FONT></o:p></SPAN>&nbsp;</DIV>
<DIV><SPAN style="mso-fareast-font-family: 'MS Mincho'"><o:p><FONT 
color=#008000>I feel the example given might be 
incorrect.</FONT></o:p></SPAN></DIV>
<DIV><SPAN style="mso-fareast-font-family: 'MS Mincho'"><o:p><FONT 
color=#008000></FONT></o:p></SPAN>&nbsp;</DIV>
<DIV><SPAN style="mso-fareast-font-family: 'MS Mincho'"><o:p><FONT 
color=#008000>It must have been like </FONT><FONT color=#ff0000><SPAN 
style="mso-fareast-font-family: 'MS Mincho'">Server: HomeServer <FONT 
color=#0000ff><STRONG>/</STRONG></FONT>v2</SPAN></FONT></o:p></SPAN></DIV>
<DIV><SPAN style="mso-fareast-font-family: 'MS Mincho'"><o:p><FONT 
color=#008000></FONT></o:p></SPAN>&nbsp;</DIV>
<DIV><SPAN style="mso-fareast-font-family: 'MS Mincho'"><o:p><FONT 
color=#008000>Regards,</FONT></o:p></SPAN></DIV>
<DIV>-------------------------------------------<BR>Nataraju A.B.<BR>Huawei 
Technologies India Pvt. Ltd.,<BR>Tel : +91-80-5216824/6/7 Xtn 
142<BR>-------------------------------------------</DIV>
<BLOCKQUOTE 
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #008000 2px solid; MARGIN-RIGHT: 0px">
  <DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV 
  style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B> 
  <A title=jh@lohi.eng.song.fi href="mailto:jh@lohi.eng.song.fi">Juha 
  Heinanen</A> </DIV>
  <DIV style="FONT: 10pt arial"><B>To:</B> <A title=mat@cisco.com 
  href="mailto:mat@cisco.com">Michael Thomas</A> </DIV>
  <DIV style="FONT: 10pt arial"><B>Cc:</B> <A title=rsparks@dynamicsoft.com 
  href="mailto:rsparks@dynamicsoft.com">Robert Sparks</A> ; <A 
  title=jdrosen@dynamicsoft.com href="mailto:jdrosen@dynamicsoft.com">Jonathan 
  Rosenberg</A> ; <A title=fluffy@cisco.com 
  href="mailto:fluffy@cisco.com">Cullen Jennings</A> ; <A title=sip@ietf.org 
  href="mailto:sip@ietf.org">sip@ietf.org</A> </DIV>
  <DIV style="FONT: 10pt arial"><B>Sent:</B> Tuesday, 31 December 2002 
  17-45</DIV>
  <DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Sip] Enrollment in SIP 
  services (and should we undeprecate basic auth)</DIV>
  <DIV><BR></DIV>Michael Thomas writes:<BR><BR>&nbsp;&gt; The problem is that 
  AAA's expect either a CHAP or<BR>&nbsp;&gt; PAP formated authenticator. 
  Digest, while it<BR>&nbsp;&gt; doesn't chip the cleartext password ain't the 
  same<BR>&nbsp;&gt; format so you have a problem. Basic allows you 
  to<BR>&nbsp;&gt; hack up an authenticator at the proxy. Maybe<BR>&nbsp;&gt; 
  EAP+digest is our friend. Or maybe not.<BR><BR>what is wrong with 
  draft-sterman-aaa-sip-00.txt?&nbsp; we use it with digets<BR>authentication 
  without any problems.&nbsp; leading radius servers support it<BR>as well as 
  all good sip proxies.<BR><BR>-- 
  juha<BR>_______________________________________________<BR>Sip mailing 
  list&nbsp; <A 
  href="https://www1.ietf.org/mailman/listinfo/sip">https://www1.ietf.org/mailman/listinfo/sip</A><BR>This 
  list is for NEW development of the core SIP Protocol<BR>Use <A 
  href="mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.columbia.edu</A> 
  for questions on current sip<BR>Use <A 
  href="mailto:sipping@ietf.org">sipping@ietf.org</A> for new developments on 
  the application of sip</BLOCKQUOTE></BODY></HTML>

--Boundary_(ID_hzMqxDpSi1EcthsVzTQLrQ)--
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inline.

Nataraju A.B. wrote:
> Hi,
> I have a query regarding the server/user-agent coding format:
> The ABNF encoding specifies it as : 
>  
> Server           =  "Server" HCOLON server-val *(LWS server-val)
> server-val       =  product / comment
> product          =  token [SLASH product-version]
> product-version  =  token
>  
> whereas teh SIP-rfc says it should be
>  Server: HomeServer v2
> CanI know the correct format to be used.

The above is just an example of the server header. It is a valid value 
according to the BNF, since the BNF allows a series of lws-separated 
tokens, without any slashes. Thus, either of these is valid:

Server: HomeServer v2
Server: HomeServer/v2


>  
> If the SIP format is to be used then is it product version is a 
> mandatory argument.

Version is not mandatory according to the BNF.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
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Use sip-implementors@cs.columbia.edu for questions on current sip
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To: Sachin Shenoy <sachins@netlab.hcltech.com>
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Subject: Re: [Sip] Loop detection; Not a bad thing to have
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inline.

Sachin Shenoy wrote:
> Please find my comments inline.
> 

> 
>> That is coupled with the fact that such loop conditions will probably 
>> be rare in deployed networks. Unlike IP routing, SIP routing is 
>> considerably more static, and I consider the likelihood of real loops 
>> a rare one outside of a lab. Thus, its appropriate to handle them in a 
>> way which perhaps consumes more resources, at the cost of much lower 
>> complexity, and more importantly, proper operation in the normal case. 
>> Thats the max-forwards mecahnism. It is much easier to implement, 
>> always correct in the normal case (i.e., it won't mis-detect loops), 
>> and much more interoperable.
> 
> 
> I have to take your word on this (that loops are rare). But sure it 
> creates havoc once
> it happens. (Simple tests where request from UA looped between 2 proxies 
> ran for more
> than 10 minutes. 

That should not happen. Max-Forwards should be preventing that.

> Where re transmissions of original INVITE from UA, the 
> "Too Many hops" response and ACK for this just kept on circulating).

The INVITE should not be retransmitted once the final response has arrived.

It seems your problems are the result of faulty software, not an 
incorrect approach to handling loops.


>> I am at a loss as to why you believe this will solve all the ails of 
>> loop detection. It is very easy to combine the functions of unique 
>> identification along with the change-detection functions of loop 
>> detections. You merely include a transaction ID as part of the long 
>> hash you are already doing. I fail to see why that is hard.
> 
> 
> Here I was trying to address (solve) a particular problem where stateless
> proxies cannot do loop detection. (There was an earlier mail which 
> stated the
> problem subject:"[Sip] BranchID in CANCEL & INVITE"). Stateless proxies
> have to maintain same branch for CANCEL & INVITE. Now if stateless proxies
> try to do loop detection, the branch could turn out to be different for 
> CANCEL
> & INVITE. (Loop detection part calculated for CANCEL could be different 
> from
> the loop detection part calculated for INVITE). 

Not for any valid loop detection algorithm.

CANCEL and INVITE will both share the same To, From, Call-ID, CSeq 
number, request-uri, route headers, and a whole bunch of other fields, 
pretty much anything that might be included in a loop detection hash 
(although i still feel that the whole process is brittle and unneeded). 
Max-Forwards is easier to deal with at a stateless proxy.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        Dean Willis
	 <dean.willis@softarmor.com>
Cc: sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
Date: Tue, 31 Dec 2002 14:23:43 -0500
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The requirement is to be able to send application data transperently e2e via the SIP proxies, without impacting the SIP state machine.
It is as simple as that.

INFO fulfills that requirement.

1) To suggest that this is an abuse of INFO as it permits SIP to be used as transport is simply bogus.

This will be the case with any other proprietary method, anyways. So there is simply no teeth in this argument. 
No wireless device will maintain X methods simply to be able to communicate with x ASs belonging to X vendors.
It is simply not a workable solution. This is not about coding. 
This just does not make sence, and serves no purpose other than shear control of what goes in INFO, or the likes.

2) To claim that INFO has no semantics so it is useless simply does not work either.
The requirment calls precisely for that.
We *dont want* semantics associated with INFO. It is e2e and transperent.
Any suggestion to try and standardize what should be carried in INFO or whatever else is doomed to failure.

3) To suggest that MESSAGE can be used instead of INFO, so what have we solved ???

So far, the so claimed harm has not been spelled whatsoever.
It sounds like a philosophical harm.

Deprecating INFO or limiting its use, will create more interoperability problems, without solving anything.
Bodies like 3GPP will not buy into that.

I think that before we shoot ourselves in the leg, come up with the requirements and proposals.
Then and only then we can decide what should be done with INFO.

Rgds/gf  


 



 



-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Monday, December 30, 2002 4:50 PM
To: Dean Willis
Cc: sip@ietf.org
Subject: Re: [Sip] INFO considered harmful


inline.

Dean Willis wrote:
>>The point I was making in the draft was - why bother? INFO by itself 
>>provides nothing. If you want differentiable, negotiable and 
>>well-documented things, define them as new methods. What does 
>>INFO give 
>>you beyond that?
> 
> 
> The ability to define a new usage without having to make code changes to
> one's SIP stack, just as Events allows.

Code to handle INFO goes somewhere.

> 
> SIP, as you may have noticed, is a fairly complicated protocol. This has
> given rise to the distribution of reusable (even open source) SIP stacks.
> Personally I'm not up to hacking a new method (into, say, the Vovida stack)
> everytime I want to deliver a new type of blob. INFO, constructed as I've
> proposed, would let me implement a new INFO package on a stack without
> having to rewrite the stack. 

How is this different from allowing new methods, and passing off 
handling of that method to registered callbacks for each method?? Not 
different at all.


> We'd just need an API to tell the stack what
> INFO packages we handled at the application layer, 

Or, you could have an API to tell the stack what methods are handled.

> and it could do the rest
> of the work without caring what data is being transported in the INFOs.

What work?? There is no semantics associated with INFO. None. Zero. 
Nada. Zip.

> Events, you may have noticed, has exactly this property.

The big huge difference I have been trying to point out is that EVENTS 
DEFINES SEMANTICS, whereas INFO doesnt.



> 
> At some level, this is comparable to the "Why do we use MIME instead of
> defining a new transport protocol for each and every possible type of data"
> argument.

MIME provides semantics and syntax that are independent of the type.
SIP events provides semantics and syntax that are independent of the 
package.
INFO provides nothing.

> 
> At its heart, INFO is for delivering APPLICATION SPECIFIC DATA within a
> dialog. The SIP state machine does not have to care about that data, just
> its delivery (although, as with SIP-T, the application itself may use the
> data to influence the SIP state machine via the stacks's APIs). If SIP were
> a proper transport protocol, we wouldn't have to define protocol extensions
> for every application-specific datum.

SIP is not a protocol for shipping data between applications. I do not 
think we should encourage the arbitrary transport of stuff in SIP.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From mailnull@www1.ietf.org  Tue Dec 31 15:52:05 2002
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From: "Sachin Shenoy" <sachins@netlab.hcltech.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: <sip@ietf.org>
References: <006001c2af6e$9ed02660$a5f1fea9@sachinhome> <4.3.2.7.0.20021231124454.00e19220@blaze> <3E11AC8D.5090700@dynamicsoft.com>
Subject: Re: [Sip] Loop detection; Not a bad thing to have
Date: Wed, 1 Jan 2003 02:26:07 +0530
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please find my comments inline.

> > I have to take your word on this (that loops are rare). But sure it
> > creates havoc once
> > it happens. (Simple tests where request from UA looped between 2 proxies
> > ran for more
> > than 10 minutes.
>
> That should not happen. Max-Forwards should be preventing that.

Max-Forwards did work in this case. But didn't prevent the above mentioned
problem.

> > Where re transmissions of original INVITE from UA, the
> > "Too Many hops" response and ACK for this just kept on circulating).
>
> The INVITE should not be retransmitted once the final response has
arrived.
>
> It seems your problems are the result of faulty software, not an
> incorrect approach to handling loops.

Let me check and confirm these figures again (will take sometime).
But if I remember correctly the time taken for the request to go 70
hops forward & the "Too May Hops" response to take 70 hops
backwards was enough for the retransmissions to fireup.(Maybe
the implementation chose to queueup too many messages rather than
dropping them and as a result was not able to flush them out
properly). As you said this could be the case of faulty software,
I have to double check it.

> > Here I was trying to address (solve) a particular problem where
stateless
> > proxies cannot do loop detection. (There was an earlier mail which
> > stated the
> > problem subject:"[Sip] BranchID in CANCEL & INVITE"). Stateless proxies
> > have to maintain same branch for CANCEL & INVITE. Now if stateless
proxies
> > try to do loop detection, the branch could turn out to be different for
> > CANCEL
> > & INVITE. (Loop detection part calculated for CANCEL could be different
> > from
> > the loop detection part calculated for INVITE).
>
> Not for any valid loop detection algorithm.

I think this is a known bug in the RFC3261 and has already been logged in
bugzilla.

http://www.sipwg.org/sipwg/show_bug.cgi?id=659


> CANCEL and INVITE will both share the same To, From, Call-ID, CSeq
> number, request-uri, route headers, and a whole bunch of other fields,
> pretty much anything that might be included in a loop detection hash
> (although i still feel that the whole process is brittle and unneeded).

CANCEL & INVITE need not share Require & Proxy-Require fields according
to RFC 3261. RFC 3261 explicilty states that CANCEL MUST NOT have
Require & Proxy-Require (Section 9.1), while INVITE may have them. An
implementation choosing to use these fields for calculating the
loop-detection
part would end up having different value for the hash. Placing this hash in
branch would break the transaction matching rules where CANCEL &
INVITE should have the same branch-id's (Section 9.1).

From what I can see there are 2 ways to go.

1. Deprecate loop detection.
2. Keep loop detection.

1. Deprecate loop detection.
    A non issue for me (actually I am for this).

2. Keep loop detection.
    If we choose this, let us take full advantage of it. Let it be possible
    for stateless proxies to detect loops. Let us take the loop detection
    part out of branch-id (which will solve the above mentioned
    problem) or find some other solution to the problem.

Anyway, I fail to see any advantage of retaining the loop-detection
part in branch-id. If anything it can only cause problems.
Keep unrelated things separate -  A good design practice.
We don't want to worry about things interfering with each other.
(In this particular case transaction matching & loop-detection).


Again, I am also for the Max-Forward solution as it is much simpler and
straight forward as you point. But then, some consensus have to be reached
on what has to be done with the existing mechanism of loop-detection.
Implementators (like me) should be given a clear direction to proceed with.

Thanks
Sachin


>"9.1 Client Behavior
> ...
> The CANCEL request MUST NOT contain any Require or Proxy-Require header
> fields."

> "9.1 Client Behavior
> ...
> A CANCEL constructed by a client MUST have only a single Via header field
> value matching the top Via value in the request being cancelled.  Using
> the same
> values for these header fields allows the CANCEL to be matched with the
> request it cancels (Section 9.2 indicates how such matching occurs)."





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From mailnull@www1.ietf.org  Tue Dec 31 18:21:47 2002
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From: Amit Bhadoria <bhadoria@cisco.com>
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Subject: [Sip] TLS: how to validate multiple domains with single certificate (fwd)
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repost from yesterday...

---------- Forwarded message ----------
Date: Mon, 30 Dec 2002 17:20:50 -0800 (PST)
From: Amit Bhadoria <bhadoria@cisco.com>
To: sip@ietf.org
Subject: [TLS] how to validate multiple domains with single certificate

hi,

i have some questions based on rfc 3263 "locating sip services". in
rfc 3263 it is mentioned that:
  "For NAPTR records with SIPS protocol fields, (if the server is using
   a site certificate), the domain name in the query and the domain name
   in the replacement field MUST both be valid based on the site
   certificate handed out by the server in the TLS exchange.  Similarly,
   the domain name in the SRV query and the domain name in the target in
   the SRV record MUST both be valid based on the same site certificate.
   Otherwise, an attacker could modify the DNS records to contain
   replacement values in a different domain, and the client could not
   validate that this was the desired behavior or the result of an
   attack."

now, there are three potentially different domains in concern while
processing a SIPS request:
	1. domain used in NAPTR querry: this is the target domain for this
	   request
	2. domain mentioned in replacement field of NAPTR records: this
	   is the domain used in the SRV querry, obtained from replacement
	   field of NAPTR records
	3. domain mentioned in target in the SRV record: the domain
	   mentioned in the SRV record selected for final A lookup

is it possible to have all the three domains different, or is it just a
bad example (due to my limited knowledge of DNS setups), and in practical
cases the three domains will either be same as TARGET or a subdomain of
it?

incase my imagination is correct, and it is possible to have a setup where
all three domains (from three steps of DNS procedures) are different, then
how can a client validate these three domains using the same certificate?

i'd appreciate your comments/explanation to my querries.

thanks,
-amit.
----------------------------------------------------------------------------
I have hardly ever known a mathematician who was capable of reasoning.
   -- gnulib


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