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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Volker Hilt <volkerh@bell-labs.com>
CC: sipping <sipping@ietf.org>, Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Subject: Re: [Sipping] draft-hilt-sipping-session-policy-00
References: <3F79865E.1080609@bell-labs.com>
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Volker,

I had submitted another draft on a related issue.
http://standards.ericsson.net/gonzalo/papers/draft-camarillo-sipping-policy-package-00.txt
This draft addresses what your draft calls static policies (although 
they do not need to be static).

Some background about this. Some time ago we had a 3GPP-IETF workshop in 
San Francisco, and I presented some concerns the IETF had related to 
protocol profiling. We agreed that the IETF was going to provide a means 
to express the policy of a network regarding the things that user agents 
attaching to a network were allowed to do.

We will need to see how both solutions fit together, since there is some 
overlap between both drafts.

Gonzalo

Volker Hilt wrote:

> We have submitted the draft-hilt-sipping-session-policy-00. Until it is
> available in the I-D archive, you can download it from:
> 
> http://drafts.mhof.com/draft-hilt-sipping-session-policy-00.txt
> 
> Thanks,
> 
> Volker
> 


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To: "'alan.johnston@mci.com'" <alan.johnston@mci.com>
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Subject: RE: [Sipping] I-D ACTION:draft-ietf-sipping-service-examples-05.t
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Alan,

At the recent SIPit, we observed an issue with regard to handling a
challenge to an _initial_ INVITE.  Given that your draft  will soon become
an RFC, I wanted to draw your attention to this issue.

Assuming that the mailing list agrees with my understanding of the issue,
and that both of the options that I document below are in fact valid, I
would like to ask that you slightly modify either of your "407 challege"
examples so that your draft encompasses both behaviours.

The problem is the generation of an authorized INVITE after an _initial_
INVITE has been challenged by either a 401 or a 407.  There seem to be two
options here.

_Option 1 - An initial INVITE is treated very similarly to a reINVITE_

The second, authorized, INVITE is sent with the same Call-Id, From
(including same tags) as the original INVITE and the CSeq incremented.  Both
of your examples exhibit this behaviour.

_Option 2 - The initial INVITE failed so no dialog or call was created_

The second, authorized, INVITE is sent with a new Call-Id, From with new tag
and a new randomly selected CSeq.

We have observed both behaviours in different implementations.
Implementations that behave in one way, must interoperate with
implementations that behave in the other way.  Therefore I believe it would
be helpful if your draft documented both behaviours, to raise awareness with
implementors that they need to take both behaviours into account.

Thanks,
Paul DS.

Paul D.Smith
Network Protocols Group 
Data Connection Ltd (DCL)
Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com 
Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com


-----Original Message-----
From: Alan Johnston [mailto:alan.johnston@mci.com]
Sent: 02 September 2003 19:06
To: sipping@ietf.org
Subject: Re: [Sipping] I-D
ACTION:draft-ietf-sipping-service-examples-05.txt


All,

Here are the changes to this draft:

    - Added Transfer using Instant Messaging call flow.

    - Added discussion to Single Line Extension to talk about implicit
    dialog subscriptions.

    - Added focus behavior to mixing examples.

    - Added optional 181 Call is Being Forwarded response to call
    forwarding scenarios.

    - Fixed branch IDs

    - Added early-only tag to Replaces in Call Pickup example to avoid
    race condition.

    - Added remote-target tag in Call Pickup flow NOTIFY body so that the
    Contact URI is known.

    - Added call flow showing user agent using Error-Info header field to
    retrieve an announcement.

I think the draft is essentially complete. I would propose getting 
volunteers to do a detailed review of sections of the document, perhaps at 
the same time as WGLC.

Thanks,
Alan Johnston
MCI



At 12:59 PM 9/2/2003 -0400, Internet-Drafts@ietf.org wrote:
>A New Internet-Draft is available from the on-line Internet-Drafts 
>directories.
>This draft is a work item of the Session Initiation Proposal Investigation 
>Working Group of the IETF.
>
>         Title           : Session Initiation Protocol Service Examples
>         Author(s)       : A. Johnston, R. Sparks
>         Filename        : draft-ietf-sipping-service-examples-05.txt
>         Pages           : 166
>         Date            : 2003-9-2
>
>This document gives examples of Session Initiation Protocol (SIP)
>services.  This covers most features offered in so-called IP Centrex
>offerings from local exchange carriers and PBX (Private Branch
>Exchange) features.  Most of the services shown in this document are
>implemented in the SIP User Agents, although some require the
>assistance of a SIP Proxy.  Some require some extensions to SIP
>including the REFER, SUBSCRIBE, and NOTIFY methods and the Replaces
>and Join headers. These features are not intended to be an exhaustive
>set, but rather show implementations of common features likely to be
>implemented on SIP IP telephones in a business environment.
>
>A URL for this Internet-Draft is:
>http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-05.
txt
>
>To remove yourself from the IETF Announcement list, send a message to
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>
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>
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_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Thu Oct  2 11:20:39 2003
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Thread-Topic: Questions on fax (T38) renegotiation
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From: "Joel Payeur" <jpayeur@mediatrix.com>
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Subject: [Sipping] Questions on fax (T38) renegotiation
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Hi all,

i've been browsing the implementors and sipping lists and can't find any =
reference to my actual problems, so here goes.

Has anyone had problems with the fact that in both =
draft-ietf-sipping-realtimefax-01.txt (now expired) and ITU T.38 =
03/2002, there is no strong statement about who should be doing the =
reinviting upon fax detection.

In realtimefax, Section 4.1

   Two possible ways exist for an Internet telephony gateway to detect=20
   a facsimile communication:=20
   =20
   1. On the emitting gateway (the one sending the facsimile document),=20
   the T.30 calling tone (CNG) is detected.  The CNG tone is however an=20
   optional signal [7];=20
   2. On the receiving gateway, the V.21 Preamble flag sequence is=20
   detected.  The Preamble is always present and follows the called=20
   station identification tone (CED) when CED is present.=20
   Note that even though the CED tone may be detected on the receiving=20
   gateway, since the fax CED tone is similar to V.25 ANS modem tones,=20
   it SHOULD NOT be used to trigger the fax switchover.=20
   =20
   In conclusion, several conditions can trigger fax detection.=20
   Internet telephony gateways supporting T.38 real-time facsimile MUST=20
   support the detection of fax transmission on the V.21 Preamble flag=20
   sequence; Internet telephony gateways MAY support the detection of=20
   fax on CNG tone.  The authors recommend that a facsimile=20
   transmission SHOULD be detected on the receiving Internet telephony=20
   gateway.


Interoperating with your own software is always pretty easy.  You decide =
on a strategy that best suits your DSP/SIP capabilities, and implement =
that.

The problem is that each gateway vendor has different hardware =
limitations, and uses its own strategy. I've even seen gateways that =
actually reInvite for PCM for a second before reInviting for T38.

Trial and error have led us to use a "sender INVITEs only", then =
"receiver INVITEs only" and finally "anyone that detects the fax =
INVITEs" strategies.  Each had its problems that we encountered when =
doing interop testing with other vendors.

We've seen that most vendors use the "anyone INVITEs" strategy.  That's =
cool, but there is a lingering problem with this strategy.  Someties a =
certain timing can be achieved in which both endpoints will detect the =
fax at about the same time, and both will INVITE the other.  The =
crossing INVITEs will of course yield back crossing "486 Busy" 's and =
the renogotiation fails.

I was wondering how T38 implementors have dealt with this particular =
issue.  I was thinking of either:

1- When getting a 486 response, find a good scheme for retrying, for =
example "sender only retries after x seconds".

2- Since the remote's detection of the fax happens because the RTP path =
is still open while negotiation takes place, perhaps i should stop =
sending RTP while renegotiating.

3- In a next update of the draft-ietf-sipping-realtimefax, make it =
mandatory for one of the two (sender, receiver) to do the reInviting.  =
This way we wouldn't need to support every possible combination.

Of course, for method 1 there is the drawback that we don't know the =
remote's strategy, and it could yield a flurry of crossing reinvites.
Method 2 may also break because sometimes a part of the fax noise will =
creep in the RTP path before the fax is actually detected by the DSP, =
which may be just enough so the remote detects it.

Any thoughts?


BTW, what's the status on draft-ietf-sipping-realtimefax, can we expect =
a new revision soon?

Jo=EBl

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From: "Francois Audet" <audet@nortelnetworks.com>
To: "'Joel Payeur'" <jpayeur@mediatrix.com>,
        "'Sipping (E-mail)'"
	 <sipping@ietf.org>,
        "'Jean-Francois Mule (E-mail)'"
	 <jf.mule@cablelabs.com>
Subject: RE: [Sipping] Questions on fax (T38) renegotiation
Date: Thu, 2 Oct 2003 14:46:40 -0700 
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My experience is that the fax receiver should be the one doing the
re-invite, as the inband IP
stream for the tones (especially with non-G.711 codec) is not accurate
enough.

I would strongly recommend the approach where it is always the receiver o=
f
the fax that does=20
the re-inviting.=20

> -----Original Message-----
> From: Joel Payeur [mailto:jpayeur@mediatrix.com]=20
> Sent: Thursday, October 02, 2003 08:16
> To: Sipping (E-mail); Jean-Francois Mule (E-mail)
> Subject: [Sipping] Questions on fax (T38) renegotiation
>=20
>=20
> Hi all,
>=20
> i've been browsing the implementors and sipping lists and=20
> can't find any reference to my actual problems, so here goes.
>=20
> Has anyone had problems with the fact that in both=20
> draft-ietf-sipping-realtimefax-01.txt (now expired) and ITU=20
> T.38 03/2002, there is no strong statement about who should=20
> be doing the reinviting upon fax detection.
>=20
> In realtimefax, Section 4.1
>=20
>    Two possible ways exist for an Internet telephony gateway=20
> to detect=20
>    a facsimile communication:=20
>    =20
>    1. On the emitting gateway (the one sending the facsimile=20
> document),=20
>    the T.30 calling tone (CNG) is detected.  The CNG tone is=20
> however an=20
>    optional signal [7];=20
>    2. On the receiving gateway, the V.21 Preamble flag sequence is=20
>    detected.  The Preamble is always present and follows the called=20
>    station identification tone (CED) when CED is present.=20
>    Note that even though the CED tone may be detected on the=20
> receiving=20
>    gateway, since the fax CED tone is similar to V.25 ANS=20
> modem tones,=20
>    it SHOULD NOT be used to trigger the fax switchover.=20
>    =20
>    In conclusion, several conditions can trigger fax detection.=20
>    Internet telephony gateways supporting T.38 real-time=20
> facsimile MUST=20
>    support the detection of fax transmission on the V.21=20
> Preamble flag=20
>    sequence; Internet telephony gateways MAY support the detection of=20
>    fax on CNG tone.  The authors recommend that a facsimile=20
>    transmission SHOULD be detected on the receiving Internet=20
> telephony=20
>    gateway.
>=20
>=20
> Interoperating with your own software is always pretty easy. =20
> You decide on a strategy that best suits your DSP/SIP=20
> capabilities, and implement that.
>=20
> The problem is that each gateway vendor has different=20
> hardware limitations, and uses its own strategy. I've even=20
> seen gateways that actually reInvite for PCM for a second=20
> before reInviting for T38.
>=20
> Trial and error have led us to use a "sender INVITEs only",=20
> then "receiver INVITEs only" and finally "anyone that detects=20
> the fax INVITEs" strategies.  Each had its problems that we=20
> encountered when doing interop testing with other vendors.
>=20
> We've seen that most vendors use the "anyone INVITEs"=20
> strategy.  That's cool, but there is a lingering problem with=20
> this strategy.  Someties a certain timing can be achieved in=20
> which both endpoints will detect the fax at about the same=20
> time, and both will INVITE the other.  The crossing INVITEs=20
> will of course yield back crossing "486 Busy" 's and the=20
> renogotiation fails.
>=20
> I was wondering how T38 implementors have dealt with this=20
> particular issue.  I was thinking of either:
>=20
> 1- When getting a 486 response, find a good scheme for=20
> retrying, for example "sender only retries after x seconds".
>=20
> 2- Since the remote's detection of the fax happens because=20
> the RTP path is still open while negotiation takes place,=20
> perhaps i should stop sending RTP while renegotiating.
>=20
> 3- In a next update of the draft-ietf-sipping-realtimefax,=20
> make it mandatory for one of the two (sender, receiver) to do=20
> the reInviting.  This way we wouldn't need to support every=20
> possible combination.
>=20
> Of course, for method 1 there is the drawback that we don't=20
> know the remote's strategy, and it could yield a flurry of=20
> crossing reinvites. Method 2 may also break because sometimes=20
> a part of the fax noise will creep in the RTP path before the=20
> fax is actually detected by the DSP, which may be just enough=20
> so the remote detects it.
>=20
> Any thoughts?
>=20
>=20
> BTW, what's the status on draft-ietf-sipping-realtimefax, can=20
> we expect a new revision soon?
>=20
> Jo=EBl
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current=20
> sip Use sip@ietf.org for new developments of core SIP
>=20

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<P><FONT SIZE=3D2>My experience is that the fax receiver should be the on=
e doing the re-invite, as the inband IP</FONT>
<BR><FONT SIZE=3D2>stream for the tones (especially with non-G.711 codec)=
 is not accurate enough.</FONT>
</P>

<P><FONT SIZE=3D2>I would strongly recommend the approach where it is alw=
ays the receiver of the fax that does </FONT>
<BR><FONT SIZE=3D2>the re-inviting. </FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Joel Payeur [<A HREF=3D"mailto:jpayeur@medi=
atrix.com">mailto:jpayeur@mediatrix.com</A>] </FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Thursday, October 02, 2003 08:16</FONT>
<BR><FONT SIZE=3D2>&gt; To: Sipping (E-mail); Jean-Francois Mule (E-mail)=
</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: [Sipping] Questions on fax (T38) renegot=
iation</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Hi all,</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; i've been browsing the implementors and sipping l=
ists and </FONT>
<BR><FONT SIZE=3D2>&gt; can't find any reference to my actual problems, s=
o here goes.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Has anyone had problems with the fact that in bot=
h </FONT>
<BR><FONT SIZE=3D2>&gt; draft-ietf-sipping-realtimefax-01.txt (now expire=
d) and ITU </FONT>
<BR><FONT SIZE=3D2>&gt; T.38 03/2002, there is no strong statement about =
who should </FONT>
<BR><FONT SIZE=3D2>&gt; be doing the reinviting upon fax detection.</FONT=
>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; In realtimefax, Section 4.1</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; Two possible ways exist for an =
Internet telephony gateway </FONT>
<BR><FONT SIZE=3D2>&gt; to detect </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; a facsimile communication: </FO=
NT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; 1. On the emitting gateway (the=
 one sending the facsimile </FONT>
<BR><FONT SIZE=3D2>&gt; document), </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; the T.30 calling tone (CNG) is =
detected.&nbsp; The CNG tone is </FONT>
<BR><FONT SIZE=3D2>&gt; however an </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; optional signal [7]; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; 2. On the receiving gateway, th=
e V.21 Preamble flag sequence is </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; detected.&nbsp; The Preamble is=
 always present and follows the called </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; station identification tone (CE=
D) when CED is present. </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; Note that even though the CED t=
one may be detected on the </FONT>
<BR><FONT SIZE=3D2>&gt; receiving </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; gateway, since the fax CED tone=
 is similar to V.25 ANS </FONT>
<BR><FONT SIZE=3D2>&gt; modem tones, </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; it SHOULD NOT be used to trigge=
r the fax switchover. </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; In conclusion, several conditio=
ns can trigger fax detection. </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; Internet telephony gateways sup=
porting T.38 real-time </FONT>
<BR><FONT SIZE=3D2>&gt; facsimile MUST </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; support the detection of fax tr=
ansmission on the V.21 </FONT>
<BR><FONT SIZE=3D2>&gt; Preamble flag </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; sequence; Internet telephony ga=
teways MAY support the detection of </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; fax on CNG tone.&nbsp; The auth=
ors recommend that a facsimile </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; transmission SHOULD be detected=
 on the receiving Internet </FONT>
<BR><FONT SIZE=3D2>&gt; telephony </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; gateway.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Interoperating with your own software is always p=
retty easy.&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; You decide on a strategy that best suits your DSP=
/SIP </FONT>
<BR><FONT SIZE=3D2>&gt; capabilities, and implement that.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; The problem is that each gateway vendor has diffe=
rent </FONT>
<BR><FONT SIZE=3D2>&gt; hardware limitations, and uses its own strategy. =
I've even </FONT>
<BR><FONT SIZE=3D2>&gt; seen gateways that actually reInvite for PCM for =
a second </FONT>
<BR><FONT SIZE=3D2>&gt; before reInviting for T38.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Trial and error have led us to use a &quot;sender=
 INVITEs only&quot;, </FONT>
<BR><FONT SIZE=3D2>&gt; then &quot;receiver INVITEs only&quot; and finall=
y &quot;anyone that detects </FONT>
<BR><FONT SIZE=3D2>&gt; the fax INVITEs&quot; strategies.&nbsp; Each had =
its problems that we </FONT>
<BR><FONT SIZE=3D2>&gt; encountered when doing interop testing with other=
 vendors.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; We've seen that most vendors use the &quot;anyone=
 INVITEs&quot; </FONT>
<BR><FONT SIZE=3D2>&gt; strategy.&nbsp; That's cool, but there is a linge=
ring problem with </FONT>
<BR><FONT SIZE=3D2>&gt; this strategy.&nbsp; Someties a certain timing ca=
n be achieved in </FONT>
<BR><FONT SIZE=3D2>&gt; which both endpoints will detect the fax at about=
 the same </FONT>
<BR><FONT SIZE=3D2>&gt; time, and both will INVITE the other.&nbsp; The c=
rossing INVITEs </FONT>
<BR><FONT SIZE=3D2>&gt; will of course yield back crossing &quot;486 Busy=
&quot; 's and the </FONT>
<BR><FONT SIZE=3D2>&gt; renogotiation fails.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I was wondering how T38 implementors have dealt w=
ith this </FONT>
<BR><FONT SIZE=3D2>&gt; particular issue.&nbsp; I was thinking of either:=
</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; 1- When getting a 486 response, find a good schem=
e for </FONT>
<BR><FONT SIZE=3D2>&gt; retrying, for example &quot;sender only retries a=
fter x seconds&quot;.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; 2- Since the remote's detection of the fax happen=
s because </FONT>
<BR><FONT SIZE=3D2>&gt; the RTP path is still open while negotiation take=
s place, </FONT>
<BR><FONT SIZE=3D2>&gt; perhaps i should stop sending RTP while renegotia=
ting.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; 3- In a next update of the draft-ietf-sipping-rea=
ltimefax, </FONT>
<BR><FONT SIZE=3D2>&gt; make it mandatory for one of the two (sender, rec=
eiver) to do </FONT>
<BR><FONT SIZE=3D2>&gt; the reInviting.&nbsp; This way we wouldn't need t=
o support every </FONT>
<BR><FONT SIZE=3D2>&gt; possible combination.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Of course, for method 1 there is the drawback tha=
t we don't </FONT>
<BR><FONT SIZE=3D2>&gt; know the remote's strategy, and it could yield a =
flurry of </FONT>
<BR><FONT SIZE=3D2>&gt; crossing reinvites. Method 2 may also break becau=
se sometimes </FONT>
<BR><FONT SIZE=3D2>&gt; a part of the fax noise will creep in the RTP pat=
h before the </FONT>
<BR><FONT SIZE=3D2>&gt; fax is actually detected by the DSP, which may be=
 just enough </FONT>
<BR><FONT SIZE=3D2>&gt; so the remote detects it.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Any thoughts?</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; BTW, what's the status on draft-ietf-sipping-real=
timefax, can </FONT>
<BR><FONT SIZE=3D2>&gt; we expect a new revision soon?</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Jo=EBl</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; _______________________________________________</=
FONT>
<BR><FONT SIZE=3D2>&gt; Sipping mailing list&nbsp; <A HREF=3D"https://www=
1.ietf.org/mailman/listinfo/sipping" TARGET=3D"_blank">https://www1.ietf.=
org/mailman/listinfo/sipping</A></FONT>
<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the applicati=
on of SIP</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for question=
s on current </FONT>
<BR><FONT SIZE=3D2>&gt; sip Use sip@ietf.org for new developments of core=
 SIP</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: Tom Taylor <taylor@nortelnetworks.com>
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This is a long-overdue action, interrupted in the preparation by vacation and other 
urgent matters.  You might want to look it over to see if there are actions for 
SIPPING other than my suggested location header field (which I believe Jon Peterson 
has already started to define).

I missed Dick Knight <dick.rr.knight@bt.com> in my acknowledgements.  My apologies 
to him: he helped to obtain information on the provision of emergency services in 
the UK.

-------- Original Message --------
Subject: I-D ACTION:draft-taylor-sipping-emerg-scen-00.txt
Date: Thu, 02 Oct 2003 07:35:46 -0400
From: Internet-Drafts@ietf.org
Reply-To: Internet-Drafts@ietf.org
To: IETF-Announce: ;

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: SIP Emergency Assistance Scenarios
	Author(s)	: T. Taylor
	Filename	: draft-taylor-sipping-emerg-scen-00.txt
	Pages		: 22
	Date		: 2003-10-1
	
This document examines the scenarios within which SIP phones may be
used to contact emergency call centres.  Beginning with an overview
of the high level system requirements for contacting emergency call
centres in the PSTN, the scenarios involving stationary or nomadic
SIP phones are described and then alternative strategies for
supporting the information flows for the scenarios needed to meet
these requirements are examined.  The major finding from the point
of view of SIP standardization is it would be helpful in a number of
scenarios if a location header field could be added either by the
SIP phone or by a proxy to indicate the location of the phone in a
machine-readable way.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-taylor-sipping-emerg-scen-00.txt

To remove yourself from the IETF Announcement list, send a message to
ietf-announce-request with the word unsubscribe in the body of the message.

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-taylor-sipping-emerg-scen-00.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


...




_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Tue Oct  7 11:43:34 2003
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Date: Tue, 07 Oct 2003 10:41:23 -0500
To: Tom Taylor <taylor@nortelnetworks.com>, sipping@ietf.org
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: [Sipping] [Fwd: I-D
  ACTION:draft-taylor-sipping-emerg-scen-00.txt]
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At 11:04 AM 10/7/2003 -0400, Tom Taylor wrote:
>This is a long-overdue action, interrupted in the preparation by vacation 
>and other urgent matters.  You might want to look it over to see if there 
>are actions for SIPPING other than my suggested location header field 
>(which I believe Jon Peterson has already started to define).

Tom

Have you seen:
http://www.ietf.org/internet-drafts/draft-polk-sipping-location-requirements-00.txt
which defines a location header already?


>I missed Dick Knight <dick.rr.knight@bt.com> in my acknowledgements.  My 
>apologies to him: he helped to obtain information on the provision of 
>emergency services in the UK.
>
>-------- Original Message --------
>Subject: I-D ACTION:draft-taylor-sipping-emerg-scen-00.txt
>Date: Thu, 02 Oct 2003 07:35:46 -0400
>From: Internet-Drafts@ietf.org
>Reply-To: Internet-Drafts@ietf.org
>To: IETF-Announce: ;
>
>A New Internet-Draft is available from the on-line Internet-Drafts 
>directories.
>
>
>         Title           : SIP Emergency Assistance Scenarios
>         Author(s)       : T. Taylor
>         Filename        : draft-taylor-sipping-emerg-scen-00.txt
>         Pages           : 22
>         Date            : 2003-10-1
>
>This document examines the scenarios within which SIP phones may be
>used to contact emergency call centres.  Beginning with an overview
>of the high level system requirements for contacting emergency call
>centres in the PSTN, the scenarios involving stationary or nomadic
>SIP phones are described and then alternative strategies for
>supporting the information flows for the scenarios needed to meet
>these requirements are examined.  The major finding from the point
>of view of SIP standardization is it would be helpful in a number of
>scenarios if a location header field could be added either by the
>SIP phone or by a proxy to indicate the location of the phone in a
>machine-readable way.
>
>A URL for this Internet-Draft is:
>http://www.ietf.org/internet-drafts/draft-taylor-sipping-emerg-scen-00.txt
>
>To remove yourself from the IETF Announcement list, send a message to
>ietf-announce-request with the word unsubscribe in the body of the message.
>
>Internet-Drafts are also available by anonymous FTP. Login with the username
>"anonymous" and a password of your e-mail address. After logging in,
>type "cd internet-drafts" and then
>         "get draft-taylor-sipping-emerg-scen-00.txt".
>
>A list of Internet-Drafts directories can be found in
>http://www.ietf.org/shadow.html
>or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>
>
>...
>
>
>
>
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: RE: [Sipping] Questions on fax (T38) renegotiation
Date: Wed, 8 Oct 2003 08:42:19 -0400
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Thread-Topic: [Sipping] Questions on fax (T38) renegotiation
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From: "Joel Payeur" <jpayeur@mediatrix.com>
To: "Francois Audet" <audet@nortelnetworks.com>,
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Francois,
=20
    The reason why we stopped using the receiver approach is that in =
some scenarios where we were the fax sender, the third party remote =
reInvited us twice in about 1 second, once in PCM, then in T38.
=20
This behavior was strange,  and it did not fit well with our DSP code.  =
We were not in a position to argue with the other vendor, so we changed =
our code so that the sender reInvites before the receiver has time to =
confuse our DSP code.  This was ugly, but it got the work done.
=20
Thanks for your reply!
=20
Joel
=20


 -----Original Message-----
From: Francois Audet [mailto:audet@nortelnetworks.com]
Sent: Thursday, October 02, 2003 17:47
To: Joel Payeur; 'Sipping (E-mail)'; 'Jean-Francois Mule (E-mail)'
Subject: RE: [Sipping] Questions on fax (T38) renegotiation



My experience is that the fax receiver should be the one doing the =
re-invite, as the inband IP=20
stream for the tones (especially with non-G.711 codec) is not accurate =
enough.=20

I would strongly recommend the approach where it is always the receiver =
of the fax that does=20
the re-inviting.=20

> -----Original Message-----=20
> From: Joel Payeur [ mailto:jpayeur@mediatrix.com]=20
> Sent: Thursday, October 02, 2003 08:16=20
> To: Sipping (E-mail); Jean-Francois Mule (E-mail)=20
> Subject: [Sipping] Questions on fax (T38) renegotiation=20
>=20
>=20
> Hi all,=20
>=20
> i've been browsing the implementors and sipping lists and=20
> can't find any reference to my actual problems, so here goes.=20
>=20
> Has anyone had problems with the fact that in both=20
> draft-ietf-sipping-realtimefax-01.txt (now expired) and ITU=20
> T.38 03/2002, there is no strong statement about who should=20
> be doing the reinviting upon fax detection.=20
>=20
> In realtimefax, Section 4.1=20
>=20
>    Two possible ways exist for an Internet telephony gateway=20
> to detect=20
>    a facsimile communication:=20
>    =20
>    1. On the emitting gateway (the one sending the facsimile=20
> document),=20
>    the T.30 calling tone (CNG) is detected.  The CNG tone is=20
> however an=20
>    optional signal [7];=20
>    2. On the receiving gateway, the V.21 Preamble flag sequence is=20
>    detected.  The Preamble is always present and follows the called=20
>    station identification tone (CED) when CED is present.=20
>    Note that even though the CED tone may be detected on the=20
> receiving=20
>    gateway, since the fax CED tone is similar to V.25 ANS=20
> modem tones,=20
>    it SHOULD NOT be used to trigger the fax switchover.=20
>    =20
>    In conclusion, several conditions can trigger fax detection.=20
>    Internet telephony gateways supporting T.38 real-time=20
> facsimile MUST=20
>    support the detection of fax transmission on the V.21=20
> Preamble flag=20
>    sequence; Internet telephony gateways MAY support the detection of=20
>    fax on CNG tone.  The authors recommend that a facsimile=20
>    transmission SHOULD be detected on the receiving Internet=20
> telephony=20
>    gateway.=20
>=20
>=20
> Interoperating with your own software is always pretty easy. =20
> You decide on a strategy that best suits your DSP/SIP=20
> capabilities, and implement that.=20
>=20
> The problem is that each gateway vendor has different=20
> hardware limitations, and uses its own strategy. I've even=20
> seen gateways that actually reInvite for PCM for a second=20
> before reInviting for T38.=20
>=20
> Trial and error have led us to use a "sender INVITEs only",=20
> then "receiver INVITEs only" and finally "anyone that detects=20
> the fax INVITEs" strategies.  Each had its problems that we=20
> encountered when doing interop testing with other vendors.=20
>=20
> We've seen that most vendors use the "anyone INVITEs"=20
> strategy.  That's cool, but there is a lingering problem with=20
> this strategy.  Someties a certain timing can be achieved in=20
> which both endpoints will detect the fax at about the same=20
> time, and both will INVITE the other.  The crossing INVITEs=20
> will of course yield back crossing "486 Busy" 's and the=20
> renogotiation fails.=20
>=20
> I was wondering how T38 implementors have dealt with this=20
> particular issue.  I was thinking of either:=20
>=20
> 1- When getting a 486 response, find a good scheme for=20
> retrying, for example "sender only retries after x seconds".=20
>=20
> 2- Since the remote's detection of the fax happens because=20
> the RTP path is still open while negotiation takes place,=20
> perhaps i should stop sending RTP while renegotiating.=20
>=20
> 3- In a next update of the draft-ietf-sipping-realtimefax,=20
> make it mandatory for one of the two (sender, receiver) to do=20
> the reInviting.  This way we wouldn't need to support every=20
> possible combination.=20
>=20
> Of course, for method 1 there is the drawback that we don't=20
> know the remote's strategy, and it could yield a flurry of=20
> crossing reinvites. Method 2 may also break because sometimes=20
> a part of the fax noise will creep in the RTP path before the=20
> fax is actually detected by the DSP, which may be just enough=20
> so the remote detects it.=20
>=20
> Any thoughts?=20
>=20
>=20
> BTW, what's the status on draft-ietf-sipping-realtimefax, can=20
> we expect a new revision soon?=20
>=20
> Jo=EBl=20
>=20
> _______________________________________________=20
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping=20
> This list is for NEW development of the application of SIP=20
> Use sip-implementors@cs.columbia.edu for questions on current=20
> sip Use sip@ietf.org for new developments of core SIP=20
>=20


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<DIV><SPAN class=3D134083512-08102003><FONT face=3DArial color=3D#0000ff =

size=3D2>Francois,</FONT></SPAN></DIV>
<DIV><SPAN class=3D134083512-08102003><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D134083512-08102003>&nbsp;&nbsp;&nbsp; <FONT =
face=3DArial=20
color=3D#0000ff size=3D2>The reason why we stopped using the receiver =
approach is=20
that in some scenarios where we were the fax sender, the third party =
remote=20
reInvited us twice in about 1 second, once in PCM, then in=20
T38.</FONT></SPAN></DIV>
<DIV><SPAN class=3D134083512-08102003><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D134083512-08102003><FONT face=3DArial color=3D#0000ff =
size=3D2>This=20
behavior was strange,&nbsp; and it did not fit well with our DSP =
code.&nbsp; We=20
were not in a position to argue with the other vendor, so we changed our =
code so=20
that the sender reInvites before the receiver has time to confuse our =
DSP=20
code.&nbsp; This was ugly, but it got the work done.</FONT></SPAN></DIV>
<DIV><FONT face=3DTahoma><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT></FONT>&nbsp;</DIV>
<DIV><FONT><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D134083512-08102003>Thanks for your =
reply!</SPAN></FONT></FONT></DIV>
<DIV><FONT><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D134083512-08102003></SPAN></FONT></FONT>&nbsp;</DIV>
<DIV><FONT><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D134083512-08102003>Joel</SPAN></FONT></FONT></DIV>
<DIV><FONT face=3DTahoma><FONT face=3DArial color=3D#0000ff =
size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT><FONT =
face=3DArial color=3D#0000ff=20
size=3D2></FONT><BR></DIV><FONT size=3D2><SPAN=20
class=3D134083512-08102003></SPAN></FONT>
<DIV><FONT size=3D2><SPAN =
class=3D134083512-08102003>&nbsp;</SPAN>-----Original=20
Message-----<BR><B>From:</B> Francois Audet=20
[mailto:audet@nortelnetworks.com]<BR><B>Sent:</B> Thursday, October 02, =
2003=20
17:47<BR><B>To:</B> Joel Payeur; 'Sipping (E-mail)'; 'Jean-Francois Mule =

(E-mail)'<BR><B>Subject:</B> RE: [Sipping] Questions on fax (T38)=20
renegotiation<BR><BR></DIV></FONT></FONT>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <P><FONT size=3D2>My experience is that the fax receiver should be the =
one doing=20
  the re-invite, as the inband IP</FONT> <BR><FONT size=3D2>stream for =
the tones=20
  (especially with non-G.711 codec) is not accurate enough.</FONT> </P>
  <P><FONT size=3D2>I would strongly recommend the approach where it is =
always the=20
  receiver of the fax that does </FONT><BR><FONT size=3D2>the =
re-inviting.=20
  </FONT></P>
  <P><FONT size=3D2>&gt; -----Original Message-----</FONT> <BR><FONT =
size=3D2>&gt;=20
  From: Joel Payeur [<A=20
  =
href=3D"mailto:jpayeur@mediatrix.com">mailto:jpayeur@mediatrix.com</A>]=20
  </FONT><BR><FONT size=3D2>&gt; Sent: Thursday, October 02, 2003 =
08:16</FONT>=20
  <BR><FONT size=3D2>&gt; To: Sipping (E-mail); Jean-Francois Mule =
(E-mail)</FONT>=20
  <BR><FONT size=3D2>&gt; Subject: [Sipping] Questions on fax (T38)=20
  renegotiation</FONT> <BR><FONT size=3D2>&gt; </FONT><BR><FONT =
size=3D2>&gt;=20
  </FONT><BR><FONT size=3D2>&gt; Hi all,</FONT> <BR><FONT size=3D2>&gt;=20
  </FONT><BR><FONT size=3D2>&gt; i've been browsing the implementors and =
sipping=20
  lists and </FONT><BR><FONT size=3D2>&gt; can't find any reference to =
my actual=20
  problems, so here goes.</FONT> <BR><FONT size=3D2>&gt; =
</FONT><BR><FONT=20
  size=3D2>&gt; Has anyone had problems with the fact that in both=20
  </FONT><BR><FONT size=3D2>&gt; draft-ietf-sipping-realtimefax-01.txt =
(now=20
  expired) and ITU </FONT><BR><FONT size=3D2>&gt; T.38 03/2002, there is =
no strong=20
  statement about who should </FONT><BR><FONT size=3D2>&gt; be doing the =

  reinviting upon fax detection.</FONT> <BR><FONT size=3D2>&gt; =
</FONT><BR><FONT=20
  size=3D2>&gt; In realtimefax, Section 4.1</FONT> <BR><FONT =
size=3D2>&gt;=20
  </FONT><BR><FONT size=3D2>&gt;&nbsp;&nbsp;&nbsp; Two possible ways =
exist for an=20
  Internet telephony gateway </FONT><BR><FONT size=3D2>&gt; to detect=20
  </FONT><BR><FONT size=3D2>&gt;&nbsp;&nbsp;&nbsp; a facsimile =
communication:=20
  </FONT><BR><FONT size=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp; =
</FONT><BR><FONT=20
  size=3D2>&gt;&nbsp;&nbsp;&nbsp; 1. On the emitting gateway (the one =
sending the=20
  facsimile </FONT><BR><FONT size=3D2>&gt; document), </FONT><BR><FONT=20
  size=3D2>&gt;&nbsp;&nbsp;&nbsp; the T.30 calling tone (CNG) is =
detected.&nbsp;=20
  The CNG tone is </FONT><BR><FONT size=3D2>&gt; however an =
</FONT><BR><FONT=20
  size=3D2>&gt;&nbsp;&nbsp;&nbsp; optional signal [7]; </FONT><BR><FONT=20
  size=3D2>&gt;&nbsp;&nbsp;&nbsp; 2. On the receiving gateway, the V.21 =
Preamble=20
  flag sequence is </FONT><BR><FONT size=3D2>&gt;&nbsp;&nbsp;&nbsp;=20
  detected.&nbsp; The Preamble is always present and follows the called=20
  </FONT><BR><FONT size=3D2>&gt;&nbsp;&nbsp;&nbsp; station =
identification tone=20
  (CED) when CED is present. </FONT><BR><FONT =
size=3D2>&gt;&nbsp;&nbsp;&nbsp; Note=20
  that even though the CED tone may be detected on the </FONT><BR><FONT=20
  size=3D2>&gt; receiving </FONT><BR><FONT =
size=3D2>&gt;&nbsp;&nbsp;&nbsp; gateway,=20
  since the fax CED tone is similar to V.25 ANS </FONT><BR><FONT =
size=3D2>&gt;=20
  modem tones, </FONT><BR><FONT size=3D2>&gt;&nbsp;&nbsp;&nbsp; it =
SHOULD NOT be=20
  used to trigger the fax switchover. </FONT><BR><FONT=20
  size=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp; </FONT><BR><FONT=20
  size=3D2>&gt;&nbsp;&nbsp;&nbsp; In conclusion, several conditions can =
trigger=20
  fax detection. </FONT><BR><FONT size=3D2>&gt;&nbsp;&nbsp;&nbsp; =
Internet=20
  telephony gateways supporting T.38 real-time </FONT><BR><FONT =
size=3D2>&gt;=20
  facsimile MUST </FONT><BR><FONT size=3D2>&gt;&nbsp;&nbsp;&nbsp; =
support the=20
  detection of fax transmission on the V.21 </FONT><BR><FONT =
size=3D2>&gt;=20
  Preamble flag </FONT><BR><FONT size=3D2>&gt;&nbsp;&nbsp;&nbsp; =
sequence;=20
  Internet telephony gateways MAY support the detection of =
</FONT><BR><FONT=20
  size=3D2>&gt;&nbsp;&nbsp;&nbsp; fax on CNG tone.&nbsp; The authors =
recommend=20
  that a facsimile </FONT><BR><FONT size=3D2>&gt;&nbsp;&nbsp;&nbsp; =
transmission=20
  SHOULD be detected on the receiving Internet </FONT><BR><FONT =
size=3D2>&gt;=20
  telephony </FONT><BR><FONT size=3D2>&gt;&nbsp;&nbsp;&nbsp; =
gateway.</FONT>=20
  <BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; =
</FONT><BR><FONT=20
  size=3D2>&gt; Interoperating with your own software is always pretty =
easy.&nbsp;=20
  </FONT><BR><FONT size=3D2>&gt; You decide on a strategy that best =
suits your=20
  DSP/SIP </FONT><BR><FONT size=3D2>&gt; capabilities, and implement =
that.</FONT>=20
  <BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; The problem is =
that each=20
  gateway vendor has different </FONT><BR><FONT size=3D2>&gt; hardware=20
  limitations, and uses its own strategy. I've even </FONT><BR><FONT =
size=3D2>&gt;=20
  seen gateways that actually reInvite for PCM for a second =
</FONT><BR><FONT=20
  size=3D2>&gt; before reInviting for T38.</FONT> <BR><FONT =
size=3D2>&gt;=20
  </FONT><BR><FONT size=3D2>&gt; Trial and error have led us to use a =
"sender=20
  INVITEs only", </FONT><BR><FONT size=3D2>&gt; then "receiver INVITEs =
only" and=20
  finally "anyone that detects </FONT><BR><FONT size=3D2>&gt; the fax =
INVITEs"=20
  strategies.&nbsp; Each had its problems that we </FONT><BR><FONT =
size=3D2>&gt;=20
  encountered when doing interop testing with other vendors.</FONT> =
<BR><FONT=20
  size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; We've seen that most =
vendors use the=20
  "anyone INVITEs" </FONT><BR><FONT size=3D2>&gt; strategy.&nbsp; That's =
cool, but=20
  there is a lingering problem with </FONT><BR><FONT size=3D2>&gt; this=20
  strategy.&nbsp; Someties a certain timing can be achieved in =
</FONT><BR><FONT=20
  size=3D2>&gt; which both endpoints will detect the fax at about the =
same=20
  </FONT><BR><FONT size=3D2>&gt; time, and both will INVITE the =
other.&nbsp; The=20
  crossing INVITEs </FONT><BR><FONT size=3D2>&gt; will of course yield =
back=20
  crossing "486 Busy" 's and the </FONT><BR><FONT size=3D2>&gt; =
renogotiation=20
  fails.</FONT> <BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; I =
was=20
  wondering how T38 implementors have dealt with this </FONT><BR><FONT=20
  size=3D2>&gt; particular issue.&nbsp; I was thinking of either:</FONT> =
<BR><FONT=20
  size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; 1- When getting a 486 =
response, find=20
  a good scheme for </FONT><BR><FONT size=3D2>&gt; retrying, for example =
"sender=20
  only retries after x seconds".</FONT> <BR><FONT size=3D2>&gt; =
</FONT><BR><FONT=20
  size=3D2>&gt; 2- Since the remote's detection of the fax happens =
because=20
  </FONT><BR><FONT size=3D2>&gt; the RTP path is still open while =
negotiation=20
  takes place, </FONT><BR><FONT size=3D2>&gt; perhaps i should stop =
sending RTP=20
  while renegotiating.</FONT> <BR><FONT size=3D2>&gt; </FONT><BR><FONT =
size=3D2>&gt;=20
  3- In a next update of the draft-ietf-sipping-realtimefax, =
</FONT><BR><FONT=20
  size=3D2>&gt; make it mandatory for one of the two (sender, receiver) =
to do=20
  </FONT><BR><FONT size=3D2>&gt; the reInviting.&nbsp; This way we =
wouldn't need=20
  to support every </FONT><BR><FONT size=3D2>&gt; possible =
combination.</FONT>=20
  <BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; Of course, for =
method 1=20
  there is the drawback that we don't </FONT><BR><FONT size=3D2>&gt; =
know the=20
  remote's strategy, and it could yield a flurry of </FONT><BR><FONT =
size=3D2>&gt;=20
  crossing reinvites. Method 2 may also break because sometimes =
</FONT><BR><FONT=20
  size=3D2>&gt; a part of the fax noise will creep in the RTP path =
before the=20
  </FONT><BR><FONT size=3D2>&gt; fax is actually detected by the DSP, =
which may be=20
  just enough </FONT><BR><FONT size=3D2>&gt; so the remote detects =
it.</FONT>=20
  <BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; Any =
thoughts?</FONT>=20
  <BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; =
</FONT><BR><FONT=20
  size=3D2>&gt; BTW, what's the status on =
draft-ietf-sipping-realtimefax, can=20
  </FONT><BR><FONT size=3D2>&gt; we expect a new revision soon?</FONT> =
<BR><FONT=20
  size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; Jo=EBl</FONT> <BR><FONT =
size=3D2>&gt;=20
  </FONT><BR><FONT size=3D2>&gt;=20
  _______________________________________________</FONT> <BR><FONT =
size=3D2>&gt;=20
  Sipping mailing list&nbsp; <A=20
  href=3D"https://www1.ietf.org/mailman/listinfo/sipping"=20
  =
target=3D_blank>https://www1.ietf.org/mailman/listinfo/sipping</A></FONT>=
=20
  <BR><FONT size=3D2>&gt; This list is for NEW development of the =
application of=20
  SIP</FONT> <BR><FONT size=3D2>&gt; Use =
sip-implementors@cs.columbia.edu for=20
  questions on current </FONT><BR><FONT size=3D2>&gt; sip Use =
sip@ietf.org for new=20
  developments of core SIP</FONT> <BR><FONT size=3D2>&gt;=20
</FONT></P></BLOCKQUOTE></BODY></HTML>

------_=_NextPart_001_01C38D99.9CAD1CDF--

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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content-class: urn:content-classes:message
Subject: RE: [Sipping] Questions on fax (T38) renegotiation
Date: Wed, 8 Oct 2003 08:45:44 -0400
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From: "Joel Payeur" <jpayeur@mediatrix.com>
To: <Sharath.Kumar@utstar.com>
Cc: "Sipping (E-mail)" <sipping@ietf.org>,
        "Jean-Francois Mule (E-mail)" <jf.mule@cablelabs.com>
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Sharath,

Thanks for the tip, but sadly, since my goal is interop, and i really =
can't know what each vendor will do in any given situation, i don't =
think any of us can come up with a solution in the current state of the =
fax renegotiation practice.  I believe that the answer may lie in =
standardizing this.

I'll start another thread, maybe shorten my question, i was hoping for =
the authors to answer.

Thanks!

Joel

> -----Original Message-----
> From: Sharath.Kumar@utstar.com [mailto:Sharath.Kumar@utstar.com]
> Sent: Monday, October 06, 2003 4:35
> To: Joel Payeur
> Cc: Sipping (E-mail); Jean-Francois Mule (E-mail)
> Subject: Re: [Sipping] Questions on fax (T38) renegotiation
>=20
>=20
>=20
> Hi Joel,
>       Like you said most implementations use the "anyone=20
> INVITEs" strategy,
>        and because of timing issues you might end up with a=20
> glare case.
>       Take a look at sections 14.1/14.2 of Rfc 3261, It talks=20
> about using
> the
>       "491 request pending" or a "500 Internal server error"=20
> to handle such
> cases,
>       In case of 491 it makes use of a retrying scheme,something
>       similar to what you proposed.
>=20
> thanks
> -Sharath
>=20
>=20
>=20
>=20
>=20
>=20
> "Joel Payeur" <jpayeur@mediatrix.com>@ietf.org on 10/02/2003=20
> 09:45:46 PM
>=20
> Sent by:    sipping-admin@ietf.org
>=20
>=20
> To:    "Sipping (E-mail)" <sipping@ietf.org>, "Jean-Francois=20
> Mule (E-mail)"
>        <jf.mule@cablelabs.com>
> cc:
> Subject:    [Sipping] Questions on fax (T38) renegotiation
>=20
>=20
> Hi all,
>=20
> i've been browsing the implementors and sipping lists and=20
> can't find any
> reference to my actual problems, so here goes.
>=20
> Has anyone had problems with the fact that in both
> draft-ietf-sipping-realtimefax-01.txt (now expired) and ITU=20
> T.38 03/2002,
> there is no strong statement about who should be doing the=20
> reinviting upon
> fax detection.
>=20
> In realtimefax, Section 4.1
>=20
>    Two possible ways exist for an Internet telephony gateway to detect
>    a facsimile communication:
>=20
>    1. On the emitting gateway (the one sending the facsimile=20
> document),
>    the T.30 calling tone (CNG) is detected.  The CNG tone is=20
> however an
>    optional signal [7];
>    2. On the receiving gateway, the V.21 Preamble flag sequence is
>    detected.  The Preamble is always present and follows the called
>    station identification tone (CED) when CED is present.
>    Note that even though the CED tone may be detected on the receiving
>    gateway, since the fax CED tone is similar to V.25 ANS modem tones,
>    it SHOULD NOT be used to trigger the fax switchover.
>=20
>    In conclusion, several conditions can trigger fax detection.
>    Internet telephony gateways supporting T.38 real-time=20
> facsimile MUST
>    support the detection of fax transmission on the V.21 Preamble flag
>    sequence; Internet telephony gateways MAY support the detection of
>    fax on CNG tone.  The authors recommend that a facsimile
>    transmission SHOULD be detected on the receiving Internet telephony
>    gateway.
>=20
>=20
> Interoperating with your own software is always pretty easy. =20
> You decide on
> a strategy that best suits your DSP/SIP capabilities, and=20
> implement that.
>=20
> The problem is that each gateway vendor has different=20
> hardware limitations,
> and uses its own strategy. I've even seen gateways that=20
> actually reInvite
> for PCM for a second before reInviting for T38.
>=20
> Trial and error have led us to use a "sender INVITEs only",=20
> then "receiver
> INVITEs only" and finally "anyone that detects the fax=20
> INVITEs" strategies.
> Each had its problems that we encountered when doing interop=20
> testing with
> other vendors.
>=20
> We've seen that most vendors use the "anyone INVITEs"=20
> strategy.  That's
> cool, but there is a lingering problem with this strategy.  Someties a
> certain timing can be achieved in which both endpoints will=20
> detect the fax
> at about the same time, and both will INVITE the other.  The crossing
> INVITEs will of course yield back crossing "486 Busy" 's and the
> renogotiation fails.
>=20
> I was wondering how T38 implementors have dealt with this=20
> particular issue.
> I was thinking of either:
>=20
> 1- When getting a 486 response, find a good scheme for retrying, for
> example "sender only retries after x seconds".
>=20
> 2- Since the remote's detection of the fax happens because=20
> the RTP path is
> still open while negotiation takes place, perhaps i should=20
> stop sending RTP
> while renegotiating.
>=20
> 3- In a next update of the draft-ietf-sipping-realtimefax, make it
> mandatory for one of the two (sender, receiver) to do the=20
> reInviting.  This
> way we wouldn't need to support every possible combination.
>=20
> Of course, for method 1 there is the drawback that we don't know the
> remote's strategy, and it could yield a flurry of crossing reinvites.
> Method 2 may also break because sometimes a part of the fax noise will
> creep in the RTP path before the fax is actually detected by=20
> the DSP, which
> may be just enough so the remote detects it.
>=20
> Any thoughts?
>=20
>=20
> BTW, what's the status on draft-ietf-sipping-realtimefax, can=20
> we expect a
> new revision soon?
>=20
> Jo=EBl
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
>  Use sip@ietf.org for new developments of core SIP
>=20
>=20
>=20
>=20
>=20
>=20

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Thread-Topic: Where's our standard for fax negotiation?
Thread-Index: AcONm61zGhDgMfmsRmmWTNhbi/myMQ==
From: "Joel Payeur" <jpayeur@mediatrix.com>
To: "Sipping (E-mail)" <sipping@ietf.org>,
        "Jean-Francois Mule (E-mail)" <jf.mule@cablelabs.com>,
        <jieying.li@ivoxnetworks.com>
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Subject: [Sipping] Where's our standard for fax negotiation?
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Hi,
In the last few years we have been playing around with SIP fax =
implementations of all kinds.  Everyone has different DSP limitations, =
and I believe that current fax BCP draft-ietf-sipping-realtimefax-01.txt =
has issued from some limitations (for example DSPs that can't switch =
dynamically).
I was wondering:
1- Why is this draft a best current practice.  Can't we process it as a =
RFC like all the rest?  It's kinda hard to press people to change their =
code when you rely on best current practices.
2- Is there still work being done on the subject? If so, where can i see =
what's going on, how can I help?
3- For clear channel fax, not knowing whether the reinvite is clear =
channel or normal PCM causes us problems, can we do something about =
that, a new attribute a=3Dfax perhaps?

As more and more customer sites are installed, the need to have a =
standard way of doing SIP fax renegotiation will become a must.  Since =
interoperability is often required, we must rely on a strong RFC to =
which we can refer. I thank the authors for the efforts that have been =
made up to now, which have been of great help.

Joel



_______________________________________________
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From exim@www1.ietf.org  Wed Oct  8 13:56:37 2003
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Subject: RE: [Sipping] Questions on fax (T38) renegotiation
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From: "Joel Payeur" <jpayeur@mediatrix.com>
To: "Lyons, Terry" <TLyons@sonusnet.com>
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Thanks. 491 does seem to be the answer to my current problem.  However i =
still feel that the BCP does not guide implementors well enough.

Funny that you should say that you prefer anyone INVITEs.

In an earlier mail Francois Audet said that=20

"I would strongly recommend the approach where it is always the receiver =
of the fax that does=20
the re-inviting."

Maybe we should take the time to improve the BCP, (in fact maybe stop =
calling it BCP), and give a clear indication on which strategy MUST be =
used, complete with WHY.

I've also noticed that there is no T38 design team on the sipping web =
page
http://www.softarmor.com/sipping/teams/index.html

Anybody know why?

Joel


> -----Original Message-----
> From: Lyons, Terry [mailto:TLyons@sonusnet.com]
> Sent: Wednesday, October 08, 2003 12:38
> To: Joel Payeur
> Subject: RE: [Sipping] Questions on fax (T38) renegotiation
>=20
>=20
> Seem to me this is a misreading of RFC 3261:
>=20
> 	The crossing INVITEs will of course yield back
> 	crossing "486 Busy" 's and the renogotiation fails.
>=20
> You should respond 491 instead.
> Then there's a procedure to resolve the glare.
>=20
> Anyone INVITEs is the preferred solution.
>=20
> - tlyons@sonusnet.com +1 732 625-3003 x212
>=20
>=20
> -----Original Message-----
> From: Joel Payeur [mailto:jpayeur@mediatrix.com]
> Sent: Thursday, October 02, 2003 11:16 AM
> To: Sipping (E-mail); Jean-Francois Mule (E-mail)
> Subject: [Sipping] Questions on fax (T38) renegotiation
>=20
>=20
> Hi all,
>=20
> i've been browsing the implementors and sipping lists and=20
> can't find any
> reference to my actual problems, so here goes.
>=20
> Has anyone had problems with the fact that in both
> draft-ietf-sipping-realtimefax-01.txt (now expired) and ITU=20
> T.38 03/2002,
> there is no strong statement about who should be doing the=20
> reinviting upon
> fax detection.
>=20
> In realtimefax, Section 4.1
>=20
>    Two possible ways exist for an Internet telephony gateway=20
> to detect=20
>    a facsimile communication:=20
>    =20
>    1. On the emitting gateway (the one sending the facsimile=20
> document),=20
>    the T.30 calling tone (CNG) is detected.  The CNG tone is=20
> however an=20
>    optional signal [7];=20
>    2. On the receiving gateway, the V.21 Preamble flag sequence is=20
>    detected.  The Preamble is always present and follows the called=20
>    station identification tone (CED) when CED is present.=20
>    Note that even though the CED tone may be detected on the=20
> receiving=20
>    gateway, since the fax CED tone is similar to V.25 ANS=20
> modem tones,=20
>    it SHOULD NOT be used to trigger the fax switchover.=20
>    =20
>    In conclusion, several conditions can trigger fax detection.=20
>    Internet telephony gateways supporting T.38 real-time=20
> facsimile MUST=20
>    support the detection of fax transmission on the V.21=20
> Preamble flag=20
>    sequence; Internet telephony gateways MAY support the detection of=20
>    fax on CNG tone.  The authors recommend that a facsimile=20
>    transmission SHOULD be detected on the receiving Internet=20
> telephony=20
>    gateway.
>=20
>=20
> Interoperating with your own software is always pretty easy. =20
> You decide on
> a strategy that best suits your DSP/SIP capabilities, and=20
> implement that.
>=20
> The problem is that each gateway vendor has different=20
> hardware limitations,
> and uses its own strategy. I've even seen gateways that=20
> actually reInvite
> for PCM for a second before reInviting for T38.
>=20
> Trial and error have led us to use a "sender INVITEs only",=20
> then "receiver
> INVITEs only" and finally "anyone that detects the fax=20
> INVITEs" strategies.
> Each had its problems that we encountered when doing interop=20
> testing with
> other vendors.
>=20
> We've seen that most vendors use the "anyone INVITEs"=20
> strategy.  That's
> cool, but there is a lingering problem with this strategy.  Someties a
> certain timing can be achieved in which both endpoints will=20
> detect the fax
> at about the same time, and both will INVITE the other.  The crossing
> INVITEs will of course yield back crossing "486 Busy" 's and the
> renogotiation fails.
>=20
> I was wondering how T38 implementors have dealt with this=20
> particular issue.
> I was thinking of either:
>=20
> 1- When getting a 486 response, find a good scheme for=20
> retrying, for example
> "sender only retries after x seconds".
>=20
> 2- Since the remote's detection of the fax happens because=20
> the RTP path is
> still open while negotiation takes place, perhaps i should=20
> stop sending RTP
> while renegotiating.
>=20
> 3- In a next update of the draft-ietf-sipping-realtimefax,=20
> make it mandatory
> for one of the two (sender, receiver) to do the reInviting. =20
> This way we
> wouldn't need to support every possible combination.
>=20
> Of course, for method 1 there is the drawback that we don't know the
> remote's strategy, and it could yield a flurry of crossing reinvites.
> Method 2 may also break because sometimes a part of the fax=20
> noise will creep
> in the RTP path before the fax is actually detected by the=20
> DSP, which may be
> just enough so the remote detects it.
>=20
> Any thoughts?
>=20
>=20
> BTW, what's the status on draft-ietf-sipping-realtimefax, can=20
> we expect a
> new revision soon?
>=20
> Jo=EBl
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20
>=20

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Thu Oct  9 12:52:58 2003
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Thread-Topic: Is realtimefax draft dead?
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From: "Hearty, John" <John.Hearty@Level3.com>
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I posted this message a couple of weeks ago with no reply.  Since then
there have been other messages related to this draft, but no answers
regarding getting updates to the draft.  I have seen no response from
the authors, and Jieying Li's email address bounced.  I feel this is an
important document to get updated, as there is very little else
available describing T.38 interop with SIP, something more and more
customers are asking for.

John Hearty
Level3

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf
Of
> Hearty, John
> Sent: Wednesday, September 24, 2003 4:37 PM
> To: sipping@ietf.org; jieying@voxnetworks.com; jfm@cablelabs.com
> Subject: [Sipping] realtimefax draft
>=20
> I see that this draft is expired.  Is a new one due out?  Also, I see
> that it implies how the negotiation of t38 attributes is done, but
there
> is nowhere in this or any other document I am aware of that clearly
> describes the offer/answer model for t38 attribute negotiation.  Can
> this be added to this draft?
>=20
> John Hearty
> Level3
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Thu Oct  9 14:48:29 2003
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Date: Thu, 09 Oct 2003 13:46:30 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: [Sipping] Is realtimefax draft dead?
In-reply-to: 
 <3F75233A2E57CC468B35F3B1FAF71EC013AA34@idc1exc0004.corp.global.level3.com>
To: "'Hearty, John'" <John.Hearty@Level3.com>, sipping@ietf.org,
        jfm@cablelabs.com
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>.  I feel this is an
>important document to get updated, as there is very little else
>available describing T.38 interop with SIP, something more and more
>customers are asking for.

I agree and hope someone will find the time to write it.

Thanks, Henry

Henry Sinnreich

MCI
400 International Parkway
Richardson, Texas 75081
USA

Office:   972-729-4983 (SIP)
VNET:     772-4983 (SIP)

Have you disconnected the PBX?

-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf Of
Hearty, John
Sent: Thursday, October 09, 2003 11:46 AM
To: sipping@ietf.org; jfm@cablelabs.com
Subject: [Sipping] Is realtimefax draft dead?

I posted this message a couple of weeks ago with no reply.  Since then
there have been other messages related to this draft, but no answers
regarding getting updates to the draft.  I have seen no response from
the authors, and Jieying Li's email address bounced.  I feel this is an
important document to get updated, as there is very little else
available describing T.38 interop with SIP, something more and more
customers are asking for.

John Hearty
Level3

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf
Of
> Hearty, John
> Sent: Wednesday, September 24, 2003 4:37 PM
> To: sipping@ietf.org; jieying@voxnetworks.com; jfm@cablelabs.com
> Subject: [Sipping] realtimefax draft
> 
> I see that this draft is expired.  Is a new one due out?  Also, I see
> that it implies how the negotiation of t38 attributes is done, but
there
> is nowhere in this or any other document I am aware of that clearly
> describes the offer/answer model for t38 attribute negotiation.  Can
> this be added to this draft?
> 
> John Hearty
> Level3
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Fri Oct 10 03:11:50 2003
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Date: Fri, 10 Oct 2003 10:10:15 +0300
To: sipping@ietf.org
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From: Juha Heinanen <jh@tutpro.com>
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Subject: [Sipping] ack request uri in Basic Call Flow Examples
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in example 3.1 alice invites bob with contact

Contact: <sip:alice@client.atlanta.example.com;transport=tcp> 

when bob acks the invite, it uses the following request uri:

ACK sip:bob@biloxi.example.com SIP/2.0

shoudn't request uri contain the remote target uri from the contact?

at least it is so in the bye:

BYE sip:alice@client.atlanta.example.com SIP/2.0

sent by bob and and also in ack of example 3.2:

ACK sip:bob@client.biloxi.example.com

-- juha

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: RE: [Sipping] Is realtimefax draft dead?
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Thread-Topic: [Sipping] Is realtimefax draft dead?
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From: "Joel Payeur" <jpayeur@mediatrix.com>
To: "Henry Sinnreich" <Henry.Sinnreich@mci.com>,
        "Hearty, John" <John.Hearty@Level3.com>, <sipping@ietf.org>,
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Is it official then that no one is currently working on these issues?  =
As John said, we would also beneficiate from better implementation =
guidelines and documentation.

Joel

> -----Original Message-----
> From: sipping-admin@ietf.org=20
> [mailto:sipping-admin@ietf.org]On Behalf Of
> Henry Sinnreich
> Sent: Thursday, October 09, 2003 14:47
> To: 'Hearty, John'; sipping@ietf.org; jfm@cablelabs.com
> Subject: RE: [Sipping] Is realtimefax draft dead?
>=20
>=20
> >.  I feel this is an
> >important document to get updated, as there is very little else
> >available describing T.38 interop with SIP, something more and more
> >customers are asking for.
>=20
> I agree and hope someone will find the time to write it.
>=20
> Thanks, Henry
>=20
> Henry Sinnreich
>=20
> MCI
> 400 International Parkway
> Richardson, Texas 75081
> USA
>=20
> Office:   972-729-4983 (SIP)
> VNET:     772-4983 (SIP)
>=20
> Have you disconnected the PBX?
>=20
> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]=20
> On Behalf Of
> Hearty, John
> Sent: Thursday, October 09, 2003 11:46 AM
> To: sipping@ietf.org; jfm@cablelabs.com
> Subject: [Sipping] Is realtimefax draft dead?
>=20
> I posted this message a couple of weeks ago with no reply.  Since then
> there have been other messages related to this draft, but no answers
> regarding getting updates to the draft.  I have seen no response from
> the authors, and Jieying Li's email address bounced.  I feel=20
> this is an
> important document to get updated, as there is very little else
> available describing T.38 interop with SIP, something more and more
> customers are asking for.
>=20
> John Hearty
> Level3
>=20
> > -----Original Message-----
> > From: sipping-admin@ietf.org=20
[mailto:sipping-admin@ietf.org] On Behalf
Of
> Hearty, John
> Sent: Wednesday, September 24, 2003 4:37 PM
> To: sipping@ietf.org; jieying@voxnetworks.com; jfm@cablelabs.com
> Subject: [Sipping] realtimefax draft
>=20
> I see that this draft is expired.  Is a new one due out?  Also, I see
> that it implies how the negotiation of t38 attributes is done, but
there
> is nowhere in this or any other document I am aware of that clearly
> describes the offer/answer model for t38 attribute negotiation.  Can
> this be added to this draft?
>=20
> John Hearty
> Level3
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: Srikanth Sundaragopalan <srikanth@cc.gatech.edu>
To: Juha Heinanen <jh@tutpro.com>
cc: sipping@ietf.org
Subject: Re: [Sipping] ack request uri in Basic Call Flow Examples
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Reply Inline

Srikanth Sundaragopalan
Graduate Student
College of Computing
Georgia Tech


On Fri, 10 Oct 2003, Juha Heinanen wrote:

> in example 3.1 alice invites bob with contact
>
> Contact: <sip:alice@client.atlanta.example.com;transport=tcp>
>
> when bob acks the invite, it uses the following request uri:
>
> ACK sip:bob@biloxi.example.com SIP/2.0

ACK is not sent by bob.It is sent by Alice and hence the R-URI is perfect.

>
> shoudn't request uri contain the remote target uri from the contact?
>
> at least it is so in the bye:
>
> BYE sip:alice@client.atlanta.example.com SIP/2.0

BYE is sent by bob,hence it has alice contact.

>
> sent by bob and and also in ack of example 3.2:
>
> ACK sip:bob@client.biloxi.example.com

Same here..

>
> -- juha
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>

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Cc: sipping@ietf.org
Subject: Re: [Sipping] ack request uri in Basic Call Flow Examples
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Srikanth Sundaragopalan writes:

 > ACK is not sent by bob.It is sent by Alice and hence the R-URI is
 > perfect.

yes, ack is sent by alice.  my typo.  however, i still think that
request uri in the ack is incorrect.  when alice receives ok from bob to
her invite, remote target uri of the dialog at alice (uac) is set to
contact uri:

12.1.2 UAC Behavior

The remote target MUST be set to the URI from the Contact header field
of the response.

in the example this is:

sip:bob@client.biloxi.example.com

now alice goes on and sends ack, which in the example has this request
uri:

ACK sip:bob@biloxi.example.com SIP/2.0 

since there was no rr'ing proxy on the path between alice and bob, route
set at alice is empty and this applies:

12.2.1 UAC Behavior

If the route set is empty, the UAC MUST place the remote target URI into
the Request-URI.

thus i still think that the ruri in the ack is incorrect.

-- juha

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
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Date: Fri, 10 Oct 2003 12:55:54 -0400 (EDT)
From: Srikanth Sundaragopalan <srikanth@cc.gatech.edu>
To: Juha Heinanen <jh@tutpro.com>
cc: sipping@ietf.org
Subject: Re: [Sipping] ack request uri in Basic Call Flow Examples
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Yes,you are right.It should have been sip:bob@client.biloxi.example.com.
Dint notice "clinet" in the contact address.

regds
Srikanth Sundaragopalan
Graduate Student
College of Computing
Georgia Tech


On Fri, 10 Oct 2003, Juha Heinanen wrote:

> Srikanth Sundaragopalan writes:
>
>  > ACK is not sent by bob.It is sent by Alice and hence the R-URI is
>  > perfect.
>
> yes, ack is sent by alice.  my typo.  however, i still think that
> request uri in the ack is incorrect.  when alice receives ok from bob to
> her invite, remote target uri of the dialog at alice (uac) is set to
> contact uri:
>
> 12.1.2 UAC Behavior
>
> The remote target MUST be set to the URI from the Contact header field
> of the response.
>
> in the example this is:
>
> sip:bob@client.biloxi.example.com
>
> now alice goes on and sends ack, which in the example has this request
> uri:
>
> ACK sip:bob@biloxi.example.com SIP/2.0
>
> since there was no rr'ing proxy on the path between alice and bob, route
> set at alice is empty and this applies:
>
> 12.2.1 UAC Behavior
>
> If the route set is empty, the UAC MUST place the remote target URI into
> the Request-URI.
>
> thus i still think that the ruri in the ack is incorrect.
>
> -- juha
>

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Fri, 10 Oct 2003 11:55:55 -0500
From: Alan Johnston <alan.johnston@mci.com>
Subject: Re: [Sipping] ack request uri in Basic Call Flow Examples
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At 06:31 PM 10/10/2003 +0300, Juha Heinanen wrote:
>Srikanth Sundaragopalan writes:
>
>  > ACK is not sent by bob.It is sent by Alice and hence the R-URI is
>  > perfect.
>
>yes, ack is sent by alice.  my typo.  however, i still think that
>request uri in the ack is incorrect.  when alice receives ok from bob to
>her invite, remote target uri of the dialog at alice (uac) is set to
>contact uri:
>
>12.1.2 UAC Behavior
>
>The remote target MUST be set to the URI from the Contact header field
>of the response.
>
>in the example this is:
>
>sip:bob@client.biloxi.example.com
>
>now alice goes on and sends ack, which in the example has this request
>uri:
>
>ACK sip:bob@biloxi.example.com SIP/2.0
>
>since there was no rr'ing proxy on the path between alice and bob, route
>set at alice is empty and this applies:
>
>12.2.1 UAC Behavior
>
>If the route set is empty, the UAC MUST place the remote target URI into
>the Request-URI.
>
>thus i still think that the ruri in the ack is incorrect.

Juha,

You are correct - the Request-URI for this ACK should be 
sip:bob@client.biloxi.example.com.  I will fix this in the authors 48 
hours.  I think the rest of the ACKs in the document have the correct 
Request-URIs.

Thanks,
Alan.

>-- juha
>
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Extending the Session Initiation Protocol Reason 
			  Header to account for Preemption Events
	Author(s)	: J. Polk
	Filename	: draft-polk-sipping-reason-header-for-preemption-00.txt
	Pages		: 14
	Date		: 2003-10-10
	
This document proposes an extension to the Session Initiation 
Protocol (SIP) Reason Header [1] to include in a BYE Method Request 
[2] as a result of a session preemption event either at a user 
agent (UA), or somewhere in the network using RSVP [3].  This 
document does not attempt to address routers failing in the packet 
path; but a deliberate event of tearing down a flow between UAs.

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_______________________________________________
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From exim@www1.ietf.org  Mon Oct 13 02:50:36 2003
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Subject: RE: [Sipping] ack request uri in Basic Call Flow Examples
Date: Mon, 13 Oct 2003 09:48:19 +0300
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Should the transport=3Dtcp be also reflected in the request-URI?

/Hisham

> -----Original Message-----
> From: ext Alan Johnston [mailto:alan.johnston@mci.com]
> Sent: Friday, October 10, 2003 7:56 PM
> To: Juha Heinanen; Srikanth Sundaragopalan
> Cc: sipping@ietf.org
> Subject: Re: [Sipping] ack request uri in Basic Call Flow Examples
>=20
>=20
> At 06:31 PM 10/10/2003 +0300, Juha Heinanen wrote:
> >Srikanth Sundaragopalan writes:
> >
> >  > ACK is not sent by bob.It is sent by Alice and hence the R-URI is
> >  > perfect.
> >
> >yes, ack is sent by alice.  my typo.  however, i still think that
> >request uri in the ack is incorrect.  when alice receives ok=20
> from bob to
> >her invite, remote target uri of the dialog at alice (uac) is set to
> >contact uri:
> >
> >12.1.2 UAC Behavior
> >
> >The remote target MUST be set to the URI from the Contact=20
> header field
> >of the response.
> >
> >in the example this is:
> >
> >sip:bob@client.biloxi.example.com
> >
> >now alice goes on and sends ack, which in the example has=20
> this request
> >uri:
> >
> >ACK sip:bob@biloxi.example.com SIP/2.0
> >
> >since there was no rr'ing proxy on the path between alice=20
> and bob, route
> >set at alice is empty and this applies:
> >
> >12.2.1 UAC Behavior
> >
> >If the route set is empty, the UAC MUST place the remote=20
> target URI into
> >the Request-URI.
> >
> >thus i still think that the ruri in the ack is incorrect.
>=20
> Juha,
>=20
> You are correct - the Request-URI for this ACK should be=20
> sip:bob@client.biloxi.example.com.  I will fix this in the authors 48=20
> hours.  I think the rest of the ACKs in the document have the correct=20
> Request-URIs.
>=20
> Thanks,
> Alan.
>=20
> >-- juha
> >
> >_______________________________________________
> >Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> >This list is for NEW development of the application of SIP
> >Use sip-implementors@cs.columbia.edu for questions on current sip
> >Use sip@ietf.org for new developments of core SIP
>=20
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20

_______________________________________________
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From exim@www1.ietf.org  Mon Oct 13 05:50:34 2003
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Date: Mon, 13 Oct 2003 17:48:35 +0800
From: Prasanna Venkatesh <prasanna@huawei.com>
To: sipping@ietf.org, eric_zimmerer@yahoo.com, Aparna.Vemuri@Qwest.com,
        jon.peterson@neustar.com, mjh@icir.org
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Subject: [Sipping] Query on example in RFC-3204
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Hi,
	I have a doubt if the examples provided in RFC-3204 section 4 are correct.

In the following extract from the example in section 4.1
         m=audio 9092 RTP/AVP 0 3 4
         --unique-boundary-1
         Content-Type: application/ISUP; version=nxv3;
         base=etsi121
there is only one CRLF between the "m" line of the SDP and the boundary.  As
per RFC-2046 (MIME), the CRLF preceeding the boundary belongs to the
boundary.  In which case last "m" line of the sdp is not terminated by a
CRLF which is a violation of the SDP standards.

Though I understand these examples are just provided for clarifying the
semantics, still I would prefer to get a clarification, if there is any
special case in which this violation of either RFC-2046 or
draft-ietf-mmusic-sdp-new-13 (and hence RFC-2327) is allowed.

Thanks in advance for the clarification.
Regards,
Prasanna


_______________________________________________
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This list is for NEW development of the application of SIP
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Subject: RE: [Sipping] Is realtimefax draft dead?
Date: Mon, 13 Oct 2003 10:23:14 -0600
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Thread-Topic: [Sipping] Is realtimefax draft dead?
Thread-Index: AcOC7FiDwIFNn8gDSgyIwiOkuUBW4gLl0nXQAAOkmRAAJmePAACeD/QA
From: "Hearty, John" <John.Hearty@Level3.com>
To: "Gonzalo Camarillo" <gonzalo.camarillo@ericsson.com>,
        "Dean Willis" <dean.willis@softarmor.com>,
        "Rohan Mahy" <rohan@cisco.com>
Cc: "Joel Payeur" <jpayeur@mediatrix.com>,
        "Henry Sinnreich" <Henry.Sinnreich@mci.com>, <sipping@ietf.org>,
        <jfm@cablelabs.com>
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I have still heard no comment from the authors of this draft.  I see
that the following was listed under Goals and Milestones in the SIPPING
charter which has apparently been missed:

Jun 03    Submit Internet-Draft on T.38 Fax Call Flows to IESG for
consideration as a BCP

I turn now to the SIPPING chairs for direction on the future of this
draft.  Personally, I am looking for some standardization that covers
the offer/answer model for T.38.  I'm not sure if this is something that
should be included in this draft, or a separate effort in the MMUSIC
working group.  If it is agreeable this information should be included
in this draft, I am willing to volunteer to pick up updates to this
draft with one other co-author if that is what it will take to get this
moving again.  Please advise.  Thanks.

John Hearty
Level3

> -----Original Message-----
> From: Joel Payeur [mailto:jpayeur@mediatrix.com]
> Sent: Friday, October 10, 2003 6:42 AM
> To: Henry Sinnreich; Hearty, John; sipping@ietf.org; jfm@cablelabs.com
> Subject: RE: [Sipping] Is realtimefax draft dead?
>=20
> Is it official then that no one is currently working on these issues?
As
> John said, we would also beneficiate from better implementation
guidelines
> and documentation.
>=20
> Joel
>=20
> > -----Original Message-----
> > From: sipping-admin@ietf.org
> > [mailto:sipping-admin@ietf.org]On Behalf Of
> > Henry Sinnreich
> > Sent: Thursday, October 09, 2003 14:47
> > To: 'Hearty, John'; sipping@ietf.org; jfm@cablelabs.com
> > Subject: RE: [Sipping] Is realtimefax draft dead?
> >
> >
> > >.  I feel this is an
> > >important document to get updated, as there is very little else
> > >available describing T.38 interop with SIP, something more and more
> > >customers are asking for.
> >
> > I agree and hope someone will find the time to write it.
> >
> > Thanks, Henry
> >
> > Henry Sinnreich
> >
> > MCI
> > 400 International Parkway
> > Richardson, Texas 75081
> > USA
> >
> > Office:   972-729-4983 (SIP)
> > VNET:     772-4983 (SIP)
> >
> > Have you disconnected the PBX?
> >
> > -----Original Message-----
> > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > On Behalf Of
> > Hearty, John
> > Sent: Thursday, October 09, 2003 11:46 AM
> > To: sipping@ietf.org; jfm@cablelabs.com
> > Subject: [Sipping] Is realtimefax draft dead?
> >
> > I posted this message a couple of weeks ago with no reply.  Since
then
> > there have been other messages related to this draft, but no answers
> > regarding getting updates to the draft.  I have seen no response
from
> > the authors, and Jieying Li's email address bounced.  I feel
> > this is an
> > important document to get updated, as there is very little else
> > available describing T.38 interop with SIP, something more and more
> > customers are asking for.
> >
> > John Hearty
> > Level3
> >
> > > -----Original Message-----
> > > From: sipping-admin@ietf.org
> [mailto:sipping-admin@ietf.org] On Behalf
> Of
> > Hearty, John
> > Sent: Wednesday, September 24, 2003 4:37 PM
> > To: sipping@ietf.org; jieying@voxnetworks.com; jfm@cablelabs.com
> > Subject: [Sipping] realtimefax draft
> >
> > I see that this draft is expired.  Is a new one due out?  Also, I
see
> > that it implies how the negotiation of t38 attributes is done, but
> there
> > is nowhere in this or any other document I am aware of that clearly
> > describes the offer/answer model for t38 attribute negotiation.  Can
> > this be added to this draft?
> >
> > John Hearty
> > Level3
> >
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Mon, 13 Oct 2003 10:47:06 -0700
Subject: Re: [Sipping] Is realtimefax draft dead?
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Cc: "Gonzalo Camarillo" <gonzalo.camarillo@ericsson.com>,
        "Dean Willis" <dean.willis@softarmor.com>,
        "Joel Payeur" <jpayeur@mediatrix.com>,
        "Henry Sinnreich" <Henry.Sinnreich@mci.com>, <sipping@ietf.org>,
        <jfm@cablelabs.com>
To: "Hearty, John" <John.Hearty@Level3.com>
From: Rohan Mahy <rohan@cisco.com>
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Hi,

Both realtime fax and the ICE mechanism are moving to MMUSIC, as they 
contain mostly SDP extensions.

I believe Jean-Francois was planning to submit a new version of 
realtime-fax to MMUSIC before the IETF 58 draft deadline.

thanks,
-rohan

On Monday, October 13, 2003, at 09:23 AM, Hearty, John wrote:

> I have still heard no comment from the authors of this draft.  I see
> that the following was listed under Goals and Milestones in the SIPPING
> charter which has apparently been missed:
>
> Jun 03    Submit Internet-Draft on T.38 Fax Call Flows to IESG for
> consideration as a BCP
>
> I turn now to the SIPPING chairs for direction on the future of this
> draft.  Personally, I am looking for some standardization that covers
> the offer/answer model for T.38.  I'm not sure if this is something 
> that
> should be included in this draft, or a separate effort in the MMUSIC
> working group.  If it is agreeable this information should be included
> in this draft, I am willing to volunteer to pick up updates to this
> draft with one other co-author if that is what it will take to get this
> moving again.  Please advise.  Thanks.
>
> John Hearty
> Level3
>
>> -----Original Message-----
>> From: Joel Payeur [mailto:jpayeur@mediatrix.com]
>> Sent: Friday, October 10, 2003 6:42 AM
>> To: Henry Sinnreich; Hearty, John; sipping@ietf.org; jfm@cablelabs.com
>> Subject: RE: [Sipping] Is realtimefax draft dead?
>>
>> Is it official then that no one is currently working on these issues?
> As
>> John said, we would also beneficiate from better implementation
> guidelines
>> and documentation.
>>
>> Joel
>>
>>> -----Original Message-----
>>> From: sipping-admin@ietf.org
>>> [mailto:sipping-admin@ietf.org]On Behalf Of
>>> Henry Sinnreich
>>> Sent: Thursday, October 09, 2003 14:47
>>> To: 'Hearty, John'; sipping@ietf.org; jfm@cablelabs.com
>>> Subject: RE: [Sipping] Is realtimefax draft dead?
>>>
>>>
>>>> .  I feel this is an
>>>> important document to get updated, as there is very little else
>>>> available describing T.38 interop with SIP, something more and more
>>>> customers are asking for.
>>>
>>> I agree and hope someone will find the time to write it.
>>>
>>> Thanks, Henry
>>>
>>> Henry Sinnreich
>>>
>>> MCI
>>> 400 International Parkway
>>> Richardson, Texas 75081
>>> USA
>>>
>>> Office:   972-729-4983 (SIP)
>>> VNET:     772-4983 (SIP)
>>>
>>> Have you disconnected the PBX?
>>>
>>> -----Original Message-----
>>> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
>>> On Behalf Of
>>> Hearty, John
>>> Sent: Thursday, October 09, 2003 11:46 AM
>>> To: sipping@ietf.org; jfm@cablelabs.com
>>> Subject: [Sipping] Is realtimefax draft dead?
>>>
>>> I posted this message a couple of weeks ago with no reply.  Since
> then
>>> there have been other messages related to this draft, but no answers
>>> regarding getting updates to the draft.  I have seen no response
> from
>>> the authors, and Jieying Li's email address bounced.  I feel
>>> this is an
>>> important document to get updated, as there is very little else
>>> available describing T.38 interop with SIP, something more and more
>>> customers are asking for.
>>>
>>> John Hearty
>>> Level3
>>>
>>>> -----Original Message-----
>>>> From: sipping-admin@ietf.org
>> [mailto:sipping-admin@ietf.org] On Behalf
>> Of
>>> Hearty, John
>>> Sent: Wednesday, September 24, 2003 4:37 PM
>>> To: sipping@ietf.org; jieying@voxnetworks.com; jfm@cablelabs.com
>>> Subject: [Sipping] realtimefax draft
>>>
>>> I see that this draft is expired.  Is a new one due out?  Also, I
> see
>>> that it implies how the negotiation of t38 attributes is done, but
>> there
>>> is nowhere in this or any other document I am aware of that clearly
>>> describes the offer/answer model for t38 attribute negotiation.  Can
>>> this be added to this draft?
>>>
>>> John Hearty
>>> Level3
>>>
>>> _______________________________________________
>>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>> This list is for NEW development of the application of SIP
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sip@ietf.org for new developments of core SIP
>>
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
>>
>>
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
>


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: RE: [Sipping] Is realtimefax draft dead?
Date: Tue, 14 Oct 2003 07:07:13 -0600
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Thread-Topic: [Sipping] Is realtimefax draft dead?
Thread-Index: AcORsWlk8ry6I0CQT7CwnLZFKETzJAAohFvQ
From: "Jean-Francois Mule" <jf.mule@cablelabs.com>
To: "Rohan Mahy" <rohan@cisco.com>, "Hearty, John" <John.Hearty@Level3.com>
Cc: "Gonzalo Camarillo" <gonzalo.camarillo@ericsson.com>,
        "Dean Willis" <dean.willis@softarmor.com>,
        "Joel Payeur" <jpayeur@mediatrix.com>,
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Hi all,

Sorry for my lack of follow-up and the delayed response.
I still plan on submitting an updated draft for IETF#58 (with minimal =
updates).

Any technical comments on the draft (I've been parsing the archive and =
addressing or logging comments as I go through them)?

Jean-Fran=E7ois

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@cisco.com]=20
> Sent: Monday, October 13, 2003 11:47 AM
> To: Hearty, John
> Cc: Gonzalo Camarillo; Dean Willis; Joel Payeur; Henry=20
> Sinnreich; sipping@ietf.org; Jean-Francois Mule
> Subject: Re: [Sipping] Is realtimefax draft dead?
>=20
>=20
> Hi,
>=20
> Both realtime fax and the ICE mechanism are moving to MMUSIC, as they=20
> contain mostly SDP extensions.
>=20
> I believe Jean-Francois was planning to submit a new version of=20
> realtime-fax to MMUSIC before the IETF 58 draft deadline.
>=20
> thanks,
> -rohan
>=20
> On Monday, October 13, 2003, at 09:23 AM, Hearty, John wrote:
>=20
> > I have still heard no comment from the authors of this=20
> draft.  I see=20
> > that the following was listed under Goals and Milestones in the=20
> > SIPPING charter which has apparently been missed:
> >
> > Jun 03    Submit Internet-Draft on T.38 Fax Call Flows to IESG for
> > consideration as a BCP
> >
> > I turn now to the SIPPING chairs for direction on the=20
> future of this=20
> > draft.  Personally, I am looking for some standardization=20
> that covers=20
> > the offer/answer model for T.38.  I'm not sure if this is something=20
> > that should be included in this draft, or a separate effort in the=20
> > MMUSIC working group.  If it is agreeable this information=20
> should be=20
> > included in this draft, I am willing to volunteer to pick=20
> up updates=20
> > to this draft with one other co-author if that is what it=20
> will take to=20
> > get this moving again.  Please advise.  Thanks.
> >
> > John Hearty
> > Level3
> >
> >> -----Original Message-----
> >> From: Joel Payeur [mailto:jpayeur@mediatrix.com]
> >> Sent: Friday, October 10, 2003 6:42 AM
> >> To: Henry Sinnreich; Hearty, John; sipping@ietf.org;=20
> >> jfm@cablelabs.com
> >> Subject: RE: [Sipping] Is realtimefax draft dead?
> >>
> >> Is it official then that no one is currently working on=20
> these issues?
> > As
> >> John said, we would also beneficiate from better implementation
> > guidelines
> >> and documentation.
> >>
> >> Joel
> >>
> >>> -----Original Message-----
> >>> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On=20
> >>> Behalf Of Henry Sinnreich
> >>> Sent: Thursday, October 09, 2003 14:47
> >>> To: 'Hearty, John'; sipping@ietf.org; jfm@cablelabs.com
> >>> Subject: RE: [Sipping] Is realtimefax draft dead?
> >>>
> >>>
> >>>> .  I feel this is an
> >>>> important document to get updated, as there is very little else=20
> >>>> available describing T.38 interop with SIP, something=20
> more and more=20
> >>>> customers are asking for.
> >>>
> >>> I agree and hope someone will find the time to write it.
> >>>
> >>> Thanks, Henry
> >>>
> >>> Henry Sinnreich
> >>>
> >>> MCI
> >>> 400 International Parkway
> >>> Richardson, Texas 75081
> >>> USA
> >>>
> >>> Office:   972-729-4983 (SIP)
> >>> VNET:     772-4983 (SIP)
> >>>
> >>> Have you disconnected the PBX?
> >>>
> >>> -----Original Message-----
> >>> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On=20
> >>> Behalf Of Hearty, John
> >>> Sent: Thursday, October 09, 2003 11:46 AM
> >>> To: sipping@ietf.org; jfm@cablelabs.com
> >>> Subject: [Sipping] Is realtimefax draft dead?
> >>>
> >>> I posted this message a couple of weeks ago with no reply.  Since
> > then
> >>> there have been other messages related to this draft, but=20
> no answers=20
> >>> regarding getting updates to the draft.  I have seen no response
> > from
> >>> the authors, and Jieying Li's email address bounced.  I=20
> feel this is=20
> >>> an important document to get updated, as there is very little else
> >>> available describing T.38 interop with SIP, something=20
> more and more
> >>> customers are asking for.
> >>>
> >>> John Hearty
> >>> Level3
> >>>
> >>>> -----Original Message-----
> >>>> From: sipping-admin@ietf.org
> >> [mailto:sipping-admin@ietf.org] On Behalf
> >> Of
> >>> Hearty, John
> >>> Sent: Wednesday, September 24, 2003 4:37 PM
> >>> To: sipping@ietf.org; jieying@voxnetworks.com; jfm@cablelabs.com
> >>> Subject: [Sipping] realtimefax draft
> >>>
> >>> I see that this draft is expired.  Is a new one due out?  Also, I
> > see
> >>> that it implies how the negotiation of t38 attributes is done, but
> >> there
> >>> is nowhere in this or any other document I am aware of=20
> that clearly=20
> >>> describes the offer/answer model for t38 attribute=20
> negotiation.  Can=20
> >>> this be added to this draft?
> >>>
> >>> John Hearty
> >>> Level3
> >>>
> >>> _______________________________________________
> >>> Sipping mailing list =20
> https://www1.ietf.org/mailman/listinfo/sipping
> >>> This list=20
> is for NEW development of the application of SIP Use=20
> >>> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >>> sip@ietf.org for new developments of core SIP
> >>
> >> _______________________________________________
> >> Sipping mailing list =20
> https://www1.ietf.org/mailman/listinfo/sipping
> >> This list=20
> is for NEW development of the application of SIP Use=20
> >> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >> sip@ietf.org for new developments of core SIP
> >>
> >>
> >> _______________________________________________
> >> Sipping mailing list =20
> https://www1.ietf.org/mailman/listinfo/sipping
> >> This list=20
> is for NEW development of the application of SIP Use=20
> >> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >> sip@ietf.org for new developments of core SIP
> >
>=20
>=20

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Tue Oct 14 09:36:28 2003
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Subject: RE: [Sipping] Is realtimefax draft dead?
Date: Tue, 14 Oct 2003 09:32:53 -0400
Message-ID: <2702075D4DE2B043BF5EB82E9CFAD45B097376@webmail.mediatrix.com>
Thread-Topic: [Sipping] Is realtimefax draft dead?
Thread-Index: AcORsWlk8ry6I0CQT7CwnLZFKETzJAAohFvQAADLo7A=
From: "Joel Payeur" <jpayeur@mediatrix.com>
To: "Jean-Francois Mule" <jf.mule@cablelabs.com>,
        "Rohan Mahy" <rohan@cisco.com>,
        "Hearty, John" <John.Hearty@Level3.com>
Cc: "Gonzalo Camarillo" <gonzalo.camarillo@ericsson.com>,
        "Dean Willis" <dean.willis@softarmor.com>,
        "Henry Sinnreich" <Henry.Sinnreich@mci.com>, <sipping@ietf.org>
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Glad you're still on the case Jean-Francois.  For my part, i think it's =
time that we (my company) take steps to help the developping T38/SIP =
standard.

Two things that come to mind are:

1- Should we enforce a one-way negotiation scheme?  (perhaps we can keep =
the "anyone" strategy, and add a short comment on using 491 to resolve =
glares in the draft)

2- For clear channel, can we add something in the SDP that will tell =
whether it's a clear channel or voice renegotiation? Say, =
a=3Dclearchannelfax media level attribute.

Joel


> -----Original Message-----
> From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> Sent: Tuesday, October 14, 2003 9:07
> To: Rohan Mahy; Hearty, John
> Cc: Gonzalo Camarillo; Dean Willis; Joel Payeur; Henry Sinnreich;
> sipping@ietf.org
> Subject: RE: [Sipping] Is realtimefax draft dead?
>=20
>=20
> Hi all,
>=20
> Sorry for my lack of follow-up and the delayed response.
> I still plan on submitting an updated draft for IETF#58 (with=20
> minimal updates).
>=20
> Any technical comments on the draft (I've been parsing the=20
> archive and addressing or logging comments as I go through them)?
>=20
> Jean-Fran=E7ois
>=20
> > -----Original Message-----
> > From: Rohan Mahy [mailto:rohan@cisco.com]=20
> > Sent: Monday, October 13, 2003 11:47 AM
> > To: Hearty, John
> > Cc: Gonzalo Camarillo; Dean Willis; Joel Payeur; Henry=20
> > Sinnreich; sipping@ietf.org; Jean-Francois Mule
> > Subject: Re: [Sipping] Is realtimefax draft dead?
> >=20
> >=20
> > Hi,
> >=20
> > Both realtime fax and the ICE mechanism are moving to=20
> MMUSIC, as they=20
> > contain mostly SDP extensions.
> >=20
> > I believe Jean-Francois was planning to submit a new version of=20
> > realtime-fax to MMUSIC before the IETF 58 draft deadline.
> >=20
> > thanks,
> > -rohan
> >=20
> > On Monday, October 13, 2003, at 09:23 AM, Hearty, John wrote:
> >=20
> > > I have still heard no comment from the authors of this=20
> > draft.  I see=20
> > > that the following was listed under Goals and Milestones in the=20
> > > SIPPING charter which has apparently been missed:
> > >
> > > Jun 03    Submit Internet-Draft on T.38 Fax Call Flows to IESG for
> > > consideration as a BCP
> > >
> > > I turn now to the SIPPING chairs for direction on the=20
> > future of this=20
> > > draft.  Personally, I am looking for some standardization=20
> > that covers=20
> > > the offer/answer model for T.38.  I'm not sure if this is=20
> something=20
> > > that should be included in this draft, or a separate=20
> effort in the=20
> > > MMUSIC working group.  If it is agreeable this information=20
> > should be=20
> > > included in this draft, I am willing to volunteer to pick=20
> > up updates=20
> > > to this draft with one other co-author if that is what it=20
> > will take to=20
> > > get this moving again.  Please advise.  Thanks.
> > >
> > > John Hearty
> > > Level3
> > >
> > >> -----Original Message-----
> > >> From: Joel Payeur [mailto:jpayeur@mediatrix.com]
> > >> Sent: Friday, October 10, 2003 6:42 AM
> > >> To: Henry Sinnreich; Hearty, John; sipping@ietf.org;=20
> > >> jfm@cablelabs.com
> > >> Subject: RE: [Sipping] Is realtimefax draft dead?
> > >>
> > >> Is it official then that no one is currently working on=20
> > these issues?
> > > As
> > >> John said, we would also beneficiate from better implementation
> > > guidelines
> > >> and documentation.
> > >>
> > >> Joel
> > >>
> > >>> -----Original Message-----
> > >>> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On=20
> > >>> Behalf Of Henry Sinnreich
> > >>> Sent: Thursday, October 09, 2003 14:47
> > >>> To: 'Hearty, John'; sipping@ietf.org; jfm@cablelabs.com
> > >>> Subject: RE: [Sipping] Is realtimefax draft dead?
> > >>>
> > >>>
> > >>>> .  I feel this is an
> > >>>> important document to get updated, as there is very=20
> little else=20
> > >>>> available describing T.38 interop with SIP, something=20
> > more and more=20
> > >>>> customers are asking for.
> > >>>
> > >>> I agree and hope someone will find the time to write it.
> > >>>
> > >>> Thanks, Henry
> > >>>
> > >>> Henry Sinnreich
> > >>>
> > >>> MCI
> > >>> 400 International Parkway
> > >>> Richardson, Texas 75081
> > >>> USA
> > >>>
> > >>> Office:   972-729-4983 (SIP)
> > >>> VNET:     772-4983 (SIP)
> > >>>
> > >>> Have you disconnected the PBX?
> > >>>
> > >>> -----Original Message-----
> > >>> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On=20
> > >>> Behalf Of Hearty, John
> > >>> Sent: Thursday, October 09, 2003 11:46 AM
> > >>> To: sipping@ietf.org; jfm@cablelabs.com
> > >>> Subject: [Sipping] Is realtimefax draft dead?
> > >>>
> > >>> I posted this message a couple of weeks ago with no=20
> reply.  Since
> > > then
> > >>> there have been other messages related to this draft, but=20
> > no answers=20
> > >>> regarding getting updates to the draft.  I have seen no response
> > > from
> > >>> the authors, and Jieying Li's email address bounced.  I=20
> > feel this is=20
> > >>> an important document to get updated, as there is very=20
> little else
> > >>> available describing T.38 interop with SIP, something=20
> > more and more
> > >>> customers are asking for.
> > >>>
> > >>> John Hearty
> > >>> Level3
> > >>>
> > >>>> -----Original Message-----
> > >>>> From: sipping-admin@ietf.org
> > >> [mailto:sipping-admin@ietf.org] On Behalf
> > >> Of
> > >>> Hearty, John
> > >>> Sent: Wednesday, September 24, 2003 4:37 PM
> > >>> To: sipping@ietf.org; jieying@voxnetworks.com; jfm@cablelabs.com
> > >>> Subject: [Sipping] realtimefax draft
> > >>>
> > >>> I see that this draft is expired.  Is a new one due=20
> out?  Also, I
> > > see
> > >>> that it implies how the negotiation of t38 attributes=20
> is done, but
> > >> there
> > >>> is nowhere in this or any other document I am aware of=20
> > that clearly=20
> > >>> describes the offer/answer model for t38 attribute=20
> > negotiation.  Can=20
> > >>> this be added to this draft?
> > >>>
> > >>> John Hearty
> > >>> Level3
> > >>>
> > >>> _______________________________________________
> > >>> Sipping mailing list =20
> > https://www1.ietf.org/mailman/listinfo/sipping
> > >>> This list=20
> > is for NEW development of the application of SIP Use=20
> > >>> sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> > >>> sip@ietf.org for new developments of core SIP
> > >>
> > >> _______________________________________________
> > >> Sipping mailing list =20
> > https://www1.ietf.org/mailman/listinfo/sipping
> > >> This list=20
> > is for NEW development of the application of SIP Use=20
> > >> sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> > >> sip@ietf.org for new developments of core SIP
> > >>
> > >>
> > >> _______________________________________________
> > >> Sipping mailing list =20
> > https://www1.ietf.org/mailman/listinfo/sipping
> > >> This list=20
> > is for NEW development of the application of SIP Use=20
> > >> sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> > >> sip@ietf.org for new developments of core SIP
> > >
> >=20
> >=20
>=20
>=20

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Tue Oct 14 11:04:30 2003
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List-Post: <mailto:sipping@ietf.org>
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I've just posted this new ID that I believe some on this list will find of 
interest

This document proposes an extension to the Session Initiation Protocol 
(SIP) Reason Header to include in a BYE Method Request as a result of a 
session preemption event either at a user agent (UA), or somewhere in the 
network using RSVP.  This document does not attempt to address routers 
failing in the packet path; but a deliberate event of tearing down a flow 
between UAs.


>         Title           : Extending the Session Initiation Protocol Reason
>                           Header to account for Preemption Events
>         Author(s)       : J. Polk
>         Filename        : 
> draft-polk-sipping-reason-header-for-preemption-00.txt
>         Pages           : 14
>         Date            : 2003-10-10
>
>
>A URL for this Internet-Draft is:
>http://www.ietf.org/internet-drafts/draft-polk-sipping-reason-header-for-preemption-00.txt
>
><ftp://ftp.ietf.org/internet-drafts/draft-polk-sipping-reason-header-for-preemption-00.txt>


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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Subject: [Sipping] realtimefax draft comments
Date: Tue, 14 Oct 2003 14:45:26 -0600
Message-ID: <3F75233A2E57CC468B35F3B1FAF71EC013AA4C@idc1exc0004.corp.global.level3.com>
Thread-Topic: [Sipping] realtimefax draft comments
Thread-Index: AcORsWlk8ry6I0CQT7CwnLZFKETzJAAohFvQAADLo7AADNIOoA==
From: "Hearty, John" <John.Hearty@Level3.com>
To: "Jean-Francois Mule" <jf.mule@cablelabs.com>
Cc: "Henry Sinnreich" <Henry.Sinnreich@mci.com>, <sipping@ietf.org>,
        "Joel Payeur" <jpayeur@mediatrix.com>
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Jean-Fran=E7ois,

As mentioned, if there is enough time, I'd like to see a section =
detailing offer/answer negotiation for each of the T38 attributes.  The =
only one that is currently well covered (in section 4.2) is =
T38MaxBitRate.  For example, if a receiving gateway re-Invites upon fax =
detection with a=3DT38FaxMaxBuffer:72, but the emitting gateway's buffer =
capability is only 32, does it respond with a 200 OK with =
a=3DT38FaxMaxBuffer:32 and the T38 setup proceeds?

This document should probably incorporate examples for the MIME sub-type =
audio/t38 where T.38 is transported in RTP.

I'm a bit uncomfortable with Joel's second comment advocating a =
clearchannelfax attribute.  I'm not sure why we need to signal what =
specific application might be running over a particular codec.  If =
something like that is needed, it might be better to use =
draft-ietf-avt-rtp-clearmode-03.txt.

Other comments are below.  I only reviewed the SIP in section 5.2.2 in =
detail, but noticed similar issues in other sections, I'll leave those =
for you to correct (let me know if you need them all explicitly =
identified).

You might consider ignoring all my comments below and only showing the =
SDP in the MMUSIC version of this draft, since SIP flows shouldn't =
really be in an MMUSIC draft.  This may make your job easier by =
eliminating the SIP message scrubbing work, and make the draft much =
simpler.

1. The Record-Route usage is not quite RFC 2543 compliant, nor RFC 3261 =
compliant.  I suggest re-vamping everything to show loose routing usage =
as defined in RFC 3261.

2. F11 through F22 need to have the URIs in the To and From headers =
swapped, as these requests originated from the other UA.

3. For F11 through F16, choose some other CSeq value, as each UA =
maintains these values independently.  The IFTGW UA should have chosen a =
new CSeq, not increased by one from the last one chosen by the IFT UA.  =
Then, F17 through F22 should be increased by one from that chosen in =
F11.  Finally, the CSeq for F23 through F26 should be 18, an increase of =
1 from that last used by the IFT UA.

4. F23 through F26 is missing a branch parm on the first Via.  The =
Branch parm on the proxy's Via in F25 doesn't match that in F24.

5. Sections 6.1.1, 6.1.2, 6.2.1 and 6.2.2 are missing two ACKs after =
F14.


John Hearty
Level3


> -----Original Message-----
> From: Joel Payeur [mailto:jpayeur@mediatrix.com]
> Sent: Tuesday, October 14, 2003 7:33 AM
> To: Jean-Francois Mule; Rohan Mahy; Hearty, John
> Cc: Gonzalo Camarillo; Dean Willis; Henry Sinnreich; sipping@ietf.org
> Subject: RE: [Sipping] Is realtimefax draft dead?
>=20
> Glad you're still on the case Jean-Francois.  For my part, i think =
it's
> time that we (my company) take steps to help the developping T38/SIP
> standard.
>=20
> Two things that come to mind are:
>=20
> 1- Should we enforce a one-way negotiation scheme?  (perhaps we can =
keep
> the "anyone" strategy, and add a short comment on using 491 to resolve
> glares in the draft)
>=20
> 2- For clear channel, can we add something in the SDP that will tell
> whether it's a clear channel or voice renegotiation? Say,
> a=3Dclearchannelfax media level attribute.
>=20
> Joel
>=20
>=20
> > -----Original Message-----
> > From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> > Sent: Tuesday, October 14, 2003 9:07
> > To: Rohan Mahy; Hearty, John
> > Cc: Gonzalo Camarillo; Dean Willis; Joel Payeur; Henry Sinnreich;
> > sipping@ietf.org
> > Subject: RE: [Sipping] Is realtimefax draft dead?
> >
> >
> > Hi all,
> >
> > Sorry for my lack of follow-up and the delayed response.
> > I still plan on submitting an updated draft for IETF#58 (with
> > minimal updates).
> >
> > Any technical comments on the draft (I've been parsing the
> > archive and addressing or logging comments as I go through them)?
> >
> > Jean-Fran=E7ois
> >
> > > -----Original Message-----
> > > From: Rohan Mahy [mailto:rohan@cisco.com]
> > > Sent: Monday, October 13, 2003 11:47 AM
> > > To: Hearty, John
> > > Cc: Gonzalo Camarillo; Dean Willis; Joel Payeur; Henry
> > > Sinnreich; sipping@ietf.org; Jean-Francois Mule
> > > Subject: Re: [Sipping] Is realtimefax draft dead?
> > >
> > >
> > > Hi,
> > >
> > > Both realtime fax and the ICE mechanism are moving to
> > MMUSIC, as they
> > > contain mostly SDP extensions.
> > >
> > > I believe Jean-Francois was planning to submit a new version of
> > > realtime-fax to MMUSIC before the IETF 58 draft deadline.
> > >
> > > thanks,
> > > -rohan
> > >
> > > On Monday, October 13, 2003, at 09:23 AM, Hearty, John wrote:
> > >
> > > > I have still heard no comment from the authors of this
> > > draft.  I see
> > > > that the following was listed under Goals and Milestones in the
> > > > SIPPING charter which has apparently been missed:
> > > >
> > > > Jun 03    Submit Internet-Draft on T.38 Fax Call Flows to IESG =
for
> > > > consideration as a BCP
> > > >
> > > > I turn now to the SIPPING chairs for direction on the
> > > future of this
> > > > draft.  Personally, I am looking for some standardization
> > > that covers
> > > > the offer/answer model for T.38.  I'm not sure if this is
> > something
> > > > that should be included in this draft, or a separate
> > effort in the
> > > > MMUSIC working group.  If it is agreeable this information
> > > should be
> > > > included in this draft, I am willing to volunteer to pick
> > > up updates
> > > > to this draft with one other co-author if that is what it
> > > will take to
> > > > get this moving again.  Please advise.  Thanks.
> > > >
> > > > John Hearty
> > > > Level3
> > > >
> > > >> -----Original Message-----
> > > >> From: Joel Payeur [mailto:jpayeur@mediatrix.com]
> > > >> Sent: Friday, October 10, 2003 6:42 AM
> > > >> To: Henry Sinnreich; Hearty, John; sipping@ietf.org;
> > > >> jfm@cablelabs.com
> > > >> Subject: RE: [Sipping] Is realtimefax draft dead?
> > > >>
> > > >> Is it official then that no one is currently working on
> > > these issues?
> > > > As
> > > >> John said, we would also beneficiate from better implementation
> > > > guidelines
> > > >> and documentation.
> > > >>
> > > >> Joel
> > > >>
> > > >>> -----Original Message-----
> > > >>> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On
> > > >>> Behalf Of Henry Sinnreich
> > > >>> Sent: Thursday, October 09, 2003 14:47
> > > >>> To: 'Hearty, John'; sipping@ietf.org; jfm@cablelabs.com
> > > >>> Subject: RE: [Sipping] Is realtimefax draft dead?
> > > >>>
> > > >>>
> > > >>>> .  I feel this is an
> > > >>>> important document to get updated, as there is very
> > little else
> > > >>>> available describing T.38 interop with SIP, something
> > > more and more
> > > >>>> customers are asking for.
> > > >>>
> > > >>> I agree and hope someone will find the time to write it.
> > > >>>
> > > >>> Thanks, Henry
> > > >>>
> > > >>> Henry Sinnreich
> > > >>>
> > > >>> MCI
> > > >>> 400 International Parkway
> > > >>> Richardson, Texas 75081
> > > >>> USA
> > > >>>
> > > >>> Office:   972-729-4983 (SIP)
> > > >>> VNET:     772-4983 (SIP)
> > > >>>
> > > >>> Have you disconnected the PBX?
> > > >>>
> > > >>> -----Original Message-----
> > > >>> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] =
On
> > > >>> Behalf Of Hearty, John
> > > >>> Sent: Thursday, October 09, 2003 11:46 AM
> > > >>> To: sipping@ietf.org; jfm@cablelabs.com
> > > >>> Subject: [Sipping] Is realtimefax draft dead?
> > > >>>
> > > >>> I posted this message a couple of weeks ago with no
> > reply.  Since
> > > > then
> > > >>> there have been other messages related to this draft, but
> > > no answers
> > > >>> regarding getting updates to the draft.  I have seen no =
response
> > > > from
> > > >>> the authors, and Jieying Li's email address bounced.  I
> > > feel this is
> > > >>> an important document to get updated, as there is very
> > little else
> > > >>> available describing T.38 interop with SIP, something
> > > more and more
> > > >>> customers are asking for.
> > > >>>
> > > >>> John Hearty
> > > >>> Level3
> > > >>>
> > > >>>> -----Original Message-----
> > > >>>> From: sipping-admin@ietf.org
> > > >> [mailto:sipping-admin@ietf.org] On Behalf
> > > >> Of
> > > >>> Hearty, John
> > > >>> Sent: Wednesday, September 24, 2003 4:37 PM
> > > >>> To: sipping@ietf.org; jieying@voxnetworks.com; =
jfm@cablelabs.com
> > > >>> Subject: [Sipping] realtimefax draft
> > > >>>
> > > >>> I see that this draft is expired.  Is a new one due
> > out?  Also, I
> > > > see
> > > >>> that it implies how the negotiation of t38 attributes
> > is done, but
> > > >> there
> > > >>> is nowhere in this or any other document I am aware of
> > > that clearly
> > > >>> describes the offer/answer model for t38 attribute
> > > negotiation.  Can
> > > >>> this be added to this draft?
> > > >>>
> > > >>> John Hearty
> > > >>> Level3
> > > >>>
> > > >>> _______________________________________________
> > > >>> Sipping mailing list
> > > https://www1.ietf.org/mailman/listinfo/sipping
> > > >>> This list
> > > is for NEW development of the application of SIP Use
> > > >>> sip-implementors@cs.columbia.edu for questions on
> > current sip Use
> > > >>> sip@ietf.org for new developments of core SIP
> > > >>
> > > >> _______________________________________________
> > > >> Sipping mailing list
> > > https://www1.ietf.org/mailman/listinfo/sipping
> > > >> This list
> > > is for NEW development of the application of SIP Use
> > > >> sip-implementors@cs.columbia.edu for questions on
> > current sip Use
> > > >> sip@ietf.org for new developments of core SIP
> > > >>
> > > >>
> > > >> _______________________________________________
> > > >> Sipping mailing list
> > > https://www1.ietf.org/mailman/listinfo/sipping
> > > >> This list
> > > is for NEW development of the application of SIP Use
> > > >> sip-implementors@cs.columbia.edu for questions on
> > current sip Use
> > > >> sip@ietf.org for new developments of core SIP
> > > >
> > >
> > >
> >
> >

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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In the examples, whenever there is a 401 response to a REGISTER, the =
second REGISTER responding to the challenge contains a via-branch =
parameter equal to that of the original REGISTER. This is a mistake, the =
branch should be unique for every transaction.

Regards,
Hisham

_______________________________________________
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Date: Wed, 15 Oct 2003 10:30:28 -0500
From: Alan Johnston <alan.johnston@mci.com>
Subject: Re: [Sipping] Via branch in draft-ietf-sipping-basic-call-flows-02.txt
In-reply-to: <2038BCC78B1AD641891A0D1AE133DBB701797282@esebe019.ntc.noki a.com>
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Hirsham,

You are correct - this will be fixed in flows 2.1 and 2.5.  The rest should 
be correct.

Thanks,
Alan Johnston
MCI
sip:alan@sipstation.com

At 02:00 PM 10/15/2003 +0300, hisham.khartabil@nokia.com wrote:
>In the examples, whenever there is a 401 response to a REGISTER, the 
>second REGISTER responding to the challenge contains a via-branch 
>parameter equal to that of the original REGISTER. This is a mistake, the 
>branch should be unique for every transaction.
>
>Regards,
>Hisham
>
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Thread-Topic: New I-D on "On-Demand Access Authorization for SIP EVent Subscriptions"
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Subject: [Sipping] New I-D on "On-Demand Access Authorization for SIP EVent Subscriptions"
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Hi all,

you can find a new I-D on "On-Demand Access Authorization for SIP Event =
Subscriptions" at=20
http://www.ietf.org/internet-drafts/draft-trossen-sipping-ondemand-00.txt=
=20

The I-D describes the problem and possible solution space for a =
so-called 'on-demand access authorization for SIP event subscriptions'. =
The objective of this document is to properly describe the particular =
problem, define requirements for solutions to this problem, and outline =
possible solutions. With this, the document intends to drive the =
standardization of one method forward in order to provide a common =
solution to the presented problem.=20

Please feel free to comment on the I-D.

Regards,

Dirk

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Thu Oct 16 09:31:39 2003
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From: "Jesske, R" <R.Jesske@t-com.net>
To: sip@ietf.org, sipping@ietf.org
Date: Mon, 13 Oct 2003 16:37:15 +0200
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Subject: [Sipping] WG: draft-mahy-iptel-cpc-00
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Hello
In Absense of any oppinion within the IPTEL WG I would like to ask the =
SIP/SIPPING Group if there is any oppinion about this issue?

Best Regards

Roland

-----Urspr=FCngliche Nachricht-----
Von: Jesske, Roland=20
Gesendet: Dienstag, 7. Oktober 2003 13:31
An: iptel@ietf.org
Cc: Alexeitsev, Denis; P=F6tzl, Joachim; Tolksdorf, Christian; =
Sch=FCtte,
Wolfgang; Pinker, Gerold; Benz, Robert
Betreff: draft-mahy-iptel-cpc-00=20


Dear all,
With regard to the draft draft-mahy-iptel-cpc-00 are there any =
activities to extend the tel URI with the cpc parameter?
If yes where I can find more information? If no what is needed to put =
this forward as an issue.

I'm also looking on interworking the Closed User Group PSTN/ISDN =
Supplementary Service with SIP. To make this service also available =
within SIP a extension with the regarding ISUP indications (Closed user =
group interlock code and Closed user group call indicator) are needed.

As they have the same character as the CPC I'm wondering if some body =
had some thoughts on an extension of the tel URI with such kind of =
parameters to make the CUG Service also in SIP available.

Best Regards

Roland Jesske





Deutsche Telekom AG
T Com Zentrale
Roland Jesske, T38-12
Section T38; Signalling, Gateways and Switching Systems=20
Am Kavalleriesand 3, 64295 Darmstadt, Germany
Phone:  +49 6151 83-5940=20
Fax:      +49 6151 83-4577=20
email:   r.jesske@t-com.net




_______________________________________________
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List-Subscribe: <https://www1.ietf.org/mailman/listinfo/sipping>,
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In absence of opinions within the IPTEL group I would like to ask if =
the SIP/SIPPING group has any?

Best Regards


Roland

-----Urspr=FCngliche Nachricht-----
Von: Jesske, Roland=20
Gesendet: Dienstag, 7. Oktober 2003 13:31
An: iptel@ietf.org
Cc: Alexeitsev, Denis; P=F6tzl, Joachim; Tolksdorf, Christian; =
Sch=FCtte,
Wolfgang; Pinker, Gerold; Benz, Robert
Betreff: draft-mahy-iptel-cpc-00=20


Dear all,
With regard to the draft draft-mahy-iptel-cpc-00 are there any =
activities to extend the tel URI with the cpc parameter?
If yes where I can find more information? If no what is needed to put =
this forward as an issue.

I'm also looking on interworking the Closed User Group PSTN/ISDN =
Supplementary Service with SIP. To make this service also available =
within SIP a extension with the regarding ISUP indications (Closed user =
group interlock code and Closed user group call indicator) are needed.

As they have the same character as the CPC I'm wondering if some body =
had some thoughts on an extension of the tel URI with such kind of =
parameters to make the CUG Service also in SIP available.

Best Regards

Roland Jesske





Deutsche Telekom AG
T Com Zentrale
Roland Jesske, T38-12
Section T38; Signalling, Gateways and Switching Systems=20
Am Kavalleriesand 3, 64295 Darmstadt, Germany
Phone:  +49 6151 83-5940=20
Fax:      +49 6151 83-4577=20
email:   r.jesske@t-com.net




_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Thu, 16 Oct 2003 09:30:36 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: [Sipping] New I-D on
 "On-Demand Access Authorization for SIP EVent Subscriptions"
In-reply-to: 
 <DC504E9C3384054C8506D3E6BB01246001530E2E@bsebe001.americas.nokia.com>
To: Dirk.Trossen@nokia.com, sipping@ietf.org
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I wonder if the term "on demand" may be commercially protected, since there
are quite a few trade marked products out there. See for example
http://www.ibm.com/us/  and how many "on demand" product names will be
revealed by a search. Or is there no naming conflict here?

Thanks, Henry

Henry Sinnreich

MCI
400 International Parkway
Richardson, Texas 75081
USA

Have you disconnected the PBX?

-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf Of
Dirk.Trossen@nokia.com
Sent: Wednesday, October 15, 2003 10:45 AM
To: sipping@ietf.org
Subject: [Sipping] New I-D on "On-Demand Access Authorization for SIP EVent
Subscriptions"

Hi all,

you can find a new I-D on "On-Demand Access Authorization for SIP Event
Subscriptions" at 
http://www.ietf.org/internet-drafts/draft-trossen-sipping-ondemand-00.txt 

The I-D describes the problem and possible solution space for a so-called
'on-demand access authorization for SIP event subscriptions'. The objective
of this document is to properly describe the particular problem, define
requirements for solutions to this problem, and outline possible solutions.
With this, the document intends to drive the standardization of one method
forward in order to provide a common solution to the presented problem. 

Please feel free to comment on the I-D.

Regards,

Dirk

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: RE: [Sipping] New I-D on "On-Demand Access Authorization for SIP EVent Subscriptions"
Date: Thu, 16 Oct 2003 10:41:12 -0400
Message-ID: <DC504E9C3384054C8506D3E6BB01246001530E38@bsebe001.americas.nokia.com>
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Hi Henry,

would it comply IETF policy to add "(TM)" to the title?

Thanks,

Dirk

>-----Original Message-----
>From: ext Henry Sinnreich [mailto:Henry.Sinnreich@mci.com]
>Sent: Thursday, October 16, 2003 10:31 AM
>To: Trossen Dirk (NRC/Boston); sipping@ietf.org
>Subject: RE: [Sipping] New I-D on "On-Demand Access Authorization for
>SIP EVent Subscriptions"
>
>
>I wonder if the term "on demand" may be commercially=20
>protected, since there
>are quite a few trade marked products out there. See for example
>http://www.ibm.com/us/  and how many "on demand" product names will be
>revealed by a search. Or is there no naming conflict here?
>
>Thanks, Henry
>
>Henry Sinnreich
>
>MCI
>400 International Parkway
>Richardson, Texas 75081
>USA
>
>Have you disconnected the PBX?
>
>-----Original Message-----
>From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]=20
>On Behalf Of
>Dirk.Trossen@nokia.com
>Sent: Wednesday, October 15, 2003 10:45 AM
>To: sipping@ietf.org
>Subject: [Sipping] New I-D on "On-Demand Access Authorization=20
>for SIP EVent
>Subscriptions"
>
>Hi all,
>
>you can find a new I-D on "On-Demand Access Authorization for SIP Event
>Subscriptions" at=20
>http://www.ietf.org/internet-drafts/draft-trossen-sipping-ondem
and-00.txt=20

The I-D describes the problem and possible solution space for a =
so-called
'on-demand access authorization for SIP event subscriptions'. The =
objective
of this document is to properly describe the particular problem, define
requirements for solutions to this problem, and outline possible =
solutions.
With this, the document intends to drive the standardization of one =
method
forward in order to provide a common solution to the presented problem.=20

Please feel free to comment on the I-D.

Regards,

Dirk

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Thu Oct 16 11:30:33 2003
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Date: Thu, 16 Oct 2003 10:26:08 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: [Sipping] New I-D on
 "On-Demand Access Authorization for SIP EVent Subscriptions"
In-reply-to: 
 <DC504E9C3384054C8506D3E6BB01246001530E38@bsebe001.americas.nokia.com>
To: Dirk.Trossen@nokia.com, Henry.Sinnreich@mci.com, sipping@ietf.org
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We need here some legal advice from the IESG, I believe.

Henry

-----Original Message-----
From: Dirk.Trossen@nokia.com [mailto:Dirk.Trossen@nokia.com] 
Sent: Thursday, October 16, 2003 9:41 AM
To: Henry.Sinnreich@mci.com; sipping@ietf.org
Subject: RE: [Sipping] New I-D on "On-Demand Access Authorization for SIP
EVent Subscriptions"

Hi Henry,

would it comply IETF policy to add "(TM)" to the title?

Thanks,

Dirk

>-----Original Message-----
>From: ext Henry Sinnreich [mailto:Henry.Sinnreich@mci.com]
>Sent: Thursday, October 16, 2003 10:31 AM
>To: Trossen Dirk (NRC/Boston); sipping@ietf.org
>Subject: RE: [Sipping] New I-D on "On-Demand Access Authorization for
>SIP EVent Subscriptions"
>
>
>I wonder if the term "on demand" may be commercially 
>protected, since there
>are quite a few trade marked products out there. See for example
>http://www.ibm.com/us/  and how many "on demand" product names will be
>revealed by a search. Or is there no naming conflict here?
>
>Thanks, Henry
>
>Henry Sinnreich
>
>MCI
>400 International Parkway
>Richardson, Texas 75081
>USA
>
>Have you disconnected the PBX?
>
>-----Original Message-----
>From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
>On Behalf Of
>Dirk.Trossen@nokia.com
>Sent: Wednesday, October 15, 2003 10:45 AM
>To: sipping@ietf.org
>Subject: [Sipping] New I-D on "On-Demand Access Authorization 
>for SIP EVent
>Subscriptions"
>
>Hi all,
>
>you can find a new I-D on "On-Demand Access Authorization for SIP Event
>Subscriptions" at 
>http://www.ietf.org/internet-drafts/draft-trossen-sipping-ondem
and-00.txt 

The I-D describes the problem and possible solution space for a so-called
'on-demand access authorization for SIP event subscriptions'. The objective
of this document is to properly describe the particular problem, define
requirements for solutions to this problem, and outline possible solutions.
With this, the document intends to drive the standardization of one method
forward in order to provide a common solution to the presented problem. 

Please feel free to comment on the I-D.

Regards,

Dirk

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Thu Oct 16 11:36:36 2003
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Subject: RE: [Sipping] New I-D on "On-Demand Access Authorization for SIP EVent Subscriptions"
Date: Thu, 16 Oct 2003 11:35:17 -0400
Message-ID: <DC504E9C3384054C8506D3E6BB01246001530E3B@bsebe001.americas.nokia.com>
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I'm fine with changing the title to "Spontaneous", if that would solve =
it.

Dirk

>-----Original Message-----
>From: ext Henry Sinnreich [mailto:Henry.Sinnreich@mci.com]
>Sent: Thursday, October 16, 2003 11:26 AM
>To: Trossen Dirk (NRC/Boston); Henry.Sinnreich@mci.com;=20
>sipping@ietf.org
>Subject: RE: [Sipping] New I-D on "On-Demand Access Authorization for
>SIP EVent Subscriptions"
>
>
>We need here some legal advice from the IESG, I believe.
>
>Henry
>
>-----Original Message-----
>From: Dirk.Trossen@nokia.com [mailto:Dirk.Trossen@nokia.com]=20
>Sent: Thursday, October 16, 2003 9:41 AM
>To: Henry.Sinnreich@mci.com; sipping@ietf.org
>Subject: RE: [Sipping] New I-D on "On-Demand Access=20
>Authorization for SIP
>EVent Subscriptions"
>
>Hi Henry,
>
>would it comply IETF policy to add "(TM)" to the title?
>
>Thanks,
>
>Dirk
>
>>-----Original Message-----
>>From: ext Henry Sinnreich [mailto:Henry.Sinnreich@mci.com]
>>Sent: Thursday, October 16, 2003 10:31 AM
>>To: Trossen Dirk (NRC/Boston); sipping@ietf.org
>>Subject: RE: [Sipping] New I-D on "On-Demand Access Authorization for
>>SIP EVent Subscriptions"
>>
>>
>>I wonder if the term "on demand" may be commercially=20
>>protected, since there
>>are quite a few trade marked products out there. See for example
>>http://www.ibm.com/us/  and how many "on demand" product names will be
>>revealed by a search. Or is there no naming conflict here?
>>
>>Thanks, Henry
>>
>>Henry Sinnreich
>>
>>MCI
>>400 International Parkway
>>Richardson, Texas 75081
>>USA
>>
>>Have you disconnected the PBX?
>>
>>-----Original Message-----
>>From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]=20
>>On Behalf Of
>>Dirk.Trossen@nokia.com
>>Sent: Wednesday, October 15, 2003 10:45 AM
>>To: sipping@ietf.org
>>Subject: [Sipping] New I-D on "On-Demand Access Authorization=20
>>for SIP EVent
>>Subscriptions"
>>
>>Hi all,
>>
>>you can find a new I-D on "On-Demand Access Authorization for=20
>SIP Event
>>Subscriptions" at=20
>>http://www.ietf.org/internet-drafts/draft-trossen-sipping-ondem
>and-00.txt=20
>
>The I-D describes the problem and possible solution space for=20
>a so-called
>'on-demand access authorization for SIP event subscriptions'.=20
>The objective
>of this document is to properly describe the particular problem, define
>requirements for solutions to this problem, and outline=20
>possible solutions.
>With this, the document intends to drive the standardization=20
>of one method
>forward in order to provide a common solution to the presented=20
>problem.=20
>
>Please feel free to comment on the I-D.
>
>Regards,
>
>Dirk
>
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP
>
>

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: Re: [Sipping] New I-D on "On-Demand Access Authorization for SIP
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Henry Sinnreich wrote:

> We need here some legal advice from the IESG, I believe.
> 

Before we busy lawyers, it might be worthwhile to discuss whether this 
is really a term we want to use (let alone whether somebody could 
trademark a term that's been in common English usage for centuries.)


_______________________________________________
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This list is for NEW development of the application of SIP
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Dirk.Trossen@nokia.com'" <Dirk.Trossen@nokia.com>,
        Henry.Sinnreich@mci.com, sipping@ietf.org
Subject: RE: [Sipping] New I-D on "On-Demand Access Authorization for SIP 
	EVent Subscriptions"
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It doesn't mean the same thing. Are we now talking about a "push" type service rather than one the user invokes.

Let's just leave it as it is, unless someone can talk to a specific issue, rather than some rather general comments that bar half the language (Do we allow any authors to be called "Macdonald")

regards

Keith

> -----Original Message-----
> From: Dirk.Trossen@nokia.com [mailto:Dirk.Trossen@nokia.com]
> Sent: 16 October 2003 16:35
> To: Henry.Sinnreich@mci.com; sipping@ietf.org
> Subject: RE: [Sipping] New I-D on "On-Demand Access Authorization for
> SIP EVent Subscriptions"
> 
> 
> I'm fine with changing the title to "Spontaneous", if that 
> would solve it.
> 
> Dirk
> 
> >-----Original Message-----
> >From: ext Henry Sinnreich [mailto:Henry.Sinnreich@mci.com]
> >Sent: Thursday, October 16, 2003 11:26 AM
> >To: Trossen Dirk (NRC/Boston); Henry.Sinnreich@mci.com; 
> >sipping@ietf.org
> >Subject: RE: [Sipping] New I-D on "On-Demand Access Authorization for
> >SIP EVent Subscriptions"
> >
> >
> >We need here some legal advice from the IESG, I believe.
> >
> >Henry
> >
> >-----Original Message-----
> >From: Dirk.Trossen@nokia.com [mailto:Dirk.Trossen@nokia.com] 
> >Sent: Thursday, October 16, 2003 9:41 AM
> >To: Henry.Sinnreich@mci.com; sipping@ietf.org
> >Subject: RE: [Sipping] New I-D on "On-Demand Access 
> >Authorization for SIP
> >EVent Subscriptions"
> >
> >Hi Henry,
> >
> >would it comply IETF policy to add "(TM)" to the title?
> >
> >Thanks,
> >
> >Dirk
> >
> >>-----Original Message-----
> >>From: ext Henry Sinnreich [mailto:Henry.Sinnreich@mci.com]
> >>Sent: Thursday, October 16, 2003 10:31 AM
> >>To: Trossen Dirk (NRC/Boston); sipping@ietf.org
> >>Subject: RE: [Sipping] New I-D on "On-Demand Access 
> Authorization for
> >>SIP EVent Subscriptions"
> >>
> >>
> >>I wonder if the term "on demand" may be commercially 
> >>protected, since there
> >>are quite a few trade marked products out there. See for example
> >>http://www.ibm.com/us/  and how many "on demand" product 
> names will be
> >>revealed by a search. Or is there no naming conflict here?
> >>
> >>Thanks, Henry
> >>
> >>Henry Sinnreich
> >>
> >>MCI
> >>400 International Parkway
> >>Richardson, Texas 75081
> >>USA
> >>
> >>Have you disconnected the PBX?
> >>
> >>-----Original Message-----
> >>From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> >>On Behalf Of
> >>Dirk.Trossen@nokia.com
> >>Sent: Wednesday, October 15, 2003 10:45 AM
> >>To: sipping@ietf.org
> >>Subject: [Sipping] New I-D on "On-Demand Access Authorization 
> >>for SIP EVent
> >>Subscriptions"
> >>
> >>Hi all,
> >>
> >>you can find a new I-D on "On-Demand Access Authorization for 
> >SIP Event
> >>Subscriptions" at 
> >>http://www.ietf.org/internet-drafts/draft-trossen-sipping-ondem
> >and-00.txt 
> >
> >The I-D describes the problem and possible solution space for 
> >a so-called
> >'on-demand access authorization for SIP event subscriptions'. 
> >The objective
> >of this document is to properly describe the particular 
> problem, define
> >requirements for solutions to this problem, and outline 
> >possible solutions.
> >With this, the document intends to drive the standardization 
> >of one method
> >forward in order to provide a common solution to the presented 
> >problem. 
> >
> >Please feel free to comment on the I-D.
> >
> >Regards,
> >
> >Dirk
> >
> >_______________________________________________
> >Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> >This list is for NEW development of the application of SIP
> >Use sip-implementors@cs.columbia.edu for questions on current sip
> >Use sip@ietf.org for new developments of core SIP
> >
> >
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Thu Oct 16 12:35:03 2003
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From: aki.niemi@nokia.com
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Subject: RE: [Sipping] New I-D on "On-Demand Access Authorization for SIP EVent Subscriptions"
Date: Thu, 16 Oct 2003 19:33:52 +0300
Message-ID: <98C7D2E5BCD2374C9AAE0BCD2E9C1DD9027D9092@esebe013.ntc.nokia.com>
Thread-Topic: [Sipping] New I-D on "On-Demand Access Authorization for SIP EVent Subscriptions"
Thread-Index: AcOUAQ4p/nMha9Y/Q729jkMwe/FDIgAAYxtg
To: <drage@lucent.com>, <Dirk.Trossen@nokia.com>, <Henry.Sinnreich@mci.com>,
        <sipping@ietf.org>
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Isn't this basically the same concept as reactive authorization?=20

if it is, then it should probably use the same term to avoid confusion.

Regards,
Aki

 > -----Original Message-----
 > From: ext Drage, Keith (Keith) [mailto:drage@lucent.com]
 > Sent: 16 October, 2003 19:15
 > To: Trossen Dirk (NRC/Boston); Henry.Sinnreich@mci.com;=20
 > sipping@ietf.org
 > Subject: RE: [Sipping] New I-D on "On-Demand Access Authorization for
 > SIP EVent Subscriptions"
 >=20
 >=20
 > It doesn't mean the same thing. Are we now talking about a=20
 > "push" type service rather than one the user invokes.
 >=20
 > Let's just leave it as it is, unless someone can talk to a=20
 > specific issue, rather than some rather general comments=20
 > that bar half the language (Do we allow any authors to be=20
 > called "Macdonald")
 >=20
 > regards
 >=20
 > Keith
 >=20
 > > -----Original Message-----
 > > From: Dirk.Trossen@nokia.com [mailto:Dirk.Trossen@nokia.com]
 > > Sent: 16 October 2003 16:35
 > > To: Henry.Sinnreich@mci.com; sipping@ietf.org
 > > Subject: RE: [Sipping] New I-D on "On-Demand Access=20
 > Authorization for
 > > SIP EVent Subscriptions"
 > >=20
 > >=20
 > > I'm fine with changing the title to "Spontaneous", if that=20
 > > would solve it.
 > >=20
 > > Dirk
 > >=20
 > > >-----Original Message-----
 > > >From: ext Henry Sinnreich [mailto:Henry.Sinnreich@mci.com]
 > > >Sent: Thursday, October 16, 2003 11:26 AM
 > > >To: Trossen Dirk (NRC/Boston); Henry.Sinnreich@mci.com;=20
 > > >sipping@ietf.org
 > > >Subject: RE: [Sipping] New I-D on "On-Demand Access=20
 > Authorization for
 > > >SIP EVent Subscriptions"
 > > >
 > > >
 > > >We need here some legal advice from the IESG, I believe.
 > > >
 > > >Henry
 > > >
 > > >-----Original Message-----
 > > >From: Dirk.Trossen@nokia.com [mailto:Dirk.Trossen@nokia.com]=20
 > > >Sent: Thursday, October 16, 2003 9:41 AM
 > > >To: Henry.Sinnreich@mci.com; sipping@ietf.org
 > > >Subject: RE: [Sipping] New I-D on "On-Demand Access=20
 > > >Authorization for SIP
 > > >EVent Subscriptions"
 > > >
 > > >Hi Henry,
 > > >
 > > >would it comply IETF policy to add "(TM)" to the title?
 > > >
 > > >Thanks,
 > > >
 > > >Dirk
 > > >
 > > >>-----Original Message-----
 > > >>From: ext Henry Sinnreich [mailto:Henry.Sinnreich@mci.com]
 > > >>Sent: Thursday, October 16, 2003 10:31 AM
 > > >>To: Trossen Dirk (NRC/Boston); sipping@ietf.org
 > > >>Subject: RE: [Sipping] New I-D on "On-Demand Access=20
 > > Authorization for
 > > >>SIP EVent Subscriptions"
 > > >>
 > > >>
 > > >>I wonder if the term "on demand" may be commercially=20
 > > >>protected, since there
 > > >>are quite a few trade marked products out there. See for example
 > > >>http://www.ibm.com/us/  and how many "on demand" product=20
 > > names will be
 > > >>revealed by a search. Or is there no naming conflict here?
 > > >>
 > > >>Thanks, Henry
 > > >>
 > > >>Henry Sinnreich
 > > >>
 > > >>MCI
 > > >>400 International Parkway
 > > >>Richardson, Texas 75081
 > > >>USA
 > > >>
 > > >>Have you disconnected the PBX?
 > > >>
 > > >>-----Original Message-----
 > > >>From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]=20
 > > >>On Behalf Of
 > > >>Dirk.Trossen@nokia.com
 > > >>Sent: Wednesday, October 15, 2003 10:45 AM
 > > >>To: sipping@ietf.org
 > > >>Subject: [Sipping] New I-D on "On-Demand Access Authorization=20
 > > >>for SIP EVent
 > > >>Subscriptions"
 > > >>
 > > >>Hi all,
 > > >>
 > > >>you can find a new I-D on "On-Demand Access Authorization for=20
 > > >SIP Event
 > > >>Subscriptions" at=20
 > > >>http://www.ietf.org/internet-drafts/draft-trossen-sipping-ondem
 > > >and-00.txt=20
 > > >
 > > >The I-D describes the problem and possible solution space for=20
 > > >a so-called
 > > >'on-demand access authorization for SIP event subscriptions'.=20
 > > >The objective
 > > >of this document is to properly describe the particular=20
 > > problem, define
 > > >requirements for solutions to this problem, and outline=20
 > > >possible solutions.
 > > >With this, the document intends to drive the standardization=20
 > > >of one method
 > > >forward in order to provide a common solution to the presented=20
 > > >problem.=20
 > > >
 > > >Please feel free to comment on the I-D.
 > > >
 > > >Regards,
 > > >
 > > >Dirk
 > > >
 > > >_______________________________________________
 > > >Sipping mailing list =20
https://www1.ietf.org/mailman/listinfo/sipping
> >This list is for NEW development of the application of SIP
> >Use sip-implementors@cs.columbia.edu for questions on current sip
> >Use sip@ietf.org for new developments of core SIP
> >
> >
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Thu Oct 16 12:47:33 2003
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Henry Sinnreich'" <Henry.Sinnreich@mci.com>, Dirk.Trossen@nokia.com,
        sipping@ietf.org
Subject: RE: [Sipping] New I-D on "On-Demand Access Authorization for SIP 
	EVent Subscriptions"
Date: Thu, 16 Oct 2003 11:46:28 -0500
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See, for example, title of RFC2645 and RFC3561.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Henry Sinnreich [mailto:Henry.Sinnreich@mci.com]
> Sent: Thursday, October 16, 2003 8:26 AM
> To: Dirk.Trossen@nokia.com; Henry.Sinnreich@mci.com; sipping@ietf.org
> Subject: RE: [Sipping] New I-D on "On-Demand Access Authorization for
> SIP EVent Subscriptions"
> 
> 
> We need here some legal advice from the IESG, I believe.
> 
> Henry
> 
> -----Original Message-----
> From: Dirk.Trossen@nokia.com [mailto:Dirk.Trossen@nokia.com] 
> Sent: Thursday, October 16, 2003 9:41 AM
> To: Henry.Sinnreich@mci.com; sipping@ietf.org
> Subject: RE: [Sipping] New I-D on "On-Demand Access 
> Authorization for SIP
> EVent Subscriptions"
> 
> Hi Henry,
> 
> would it comply IETF policy to add "(TM)" to the title?
> 
> Thanks,
> 
> Dirk
> 
> >-----Original Message-----
> >From: ext Henry Sinnreich [mailto:Henry.Sinnreich@mci.com]
> >Sent: Thursday, October 16, 2003 10:31 AM
> >To: Trossen Dirk (NRC/Boston); sipping@ietf.org
> >Subject: RE: [Sipping] New I-D on "On-Demand Access Authorization for
> >SIP EVent Subscriptions"
> >
> >
> >I wonder if the term "on demand" may be commercially 
> >protected, since there
> >are quite a few trade marked products out there. See for example
> >http://www.ibm.com/us/  and how many "on demand" product 
> names will be
> >revealed by a search. Or is there no naming conflict here?
> >
> >Thanks, Henry
> >
> >Henry Sinnreich
> >
> >MCI
> >400 International Parkway
> >Richardson, Texas 75081
> >USA
> >
> >Have you disconnected the PBX?
> >
> >-----Original Message-----
> >From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> >On Behalf Of
> >Dirk.Trossen@nokia.com
> >Sent: Wednesday, October 15, 2003 10:45 AM
> >To: sipping@ietf.org
> >Subject: [Sipping] New I-D on "On-Demand Access Authorization 
> >for SIP EVent
> >Subscriptions"
> >
> >Hi all,
> >
> >you can find a new I-D on "On-Demand Access Authorization 
> for SIP Event
> >Subscriptions" at 
> >http://www.ietf.org/internet-drafts/draft-trossen-sipping-ondem
> and-00.txt 
> 
> The I-D describes the problem and possible solution space for 
> a so-called
> 'on-demand access authorization for SIP event subscriptions'. 
> The objective
> of this document is to properly describe the particular 
> problem, define
> requirements for solutions to this problem, and outline 
> possible solutions.
> With this, the document intends to drive the standardization 
> of one method
> forward in order to provide a common solution to the 
> presented problem. 
> 
> Please feel free to comment on the I-D.
> 
> Regards,
> 
> Dirk
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Thu Oct 16 15:32:37 2003
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.

	Title		: Interworking between SIP and QSIG
	Author(s)	: J. Elwell, et. al.
	Filename	: draft-ietf-sipping-qsig2sip-03.txt
	Pages		: 59
	Date		: 2003-10-16
	
This document specifies interworking between the Session Initiation 
Protocol (SIP) and QSIG within corporate telecommunication networks 
(also known as enterprise networks). SIP is an Internet application-
layer control (signalling) protocol for creating, modifying, and 
terminating sessions with one or more participants. These sessions 
include, in particular, telephone calls. QSIG is a signalling 
protocol for creating, modifying and terminating circuit-switched 
calls, in particular telephone calls, within Private Integrated 
Services Networks (PISNs). QSIG is specified in a number of ECMA 
Standards and published also as ISO/IEC standards. 
network and a participant in the QSIG network. Such calls are 
supported by gateways that perform interworking between SIP and QSIG. 
      
This document is a product of the authors' activities in ECMA 
(www.ecma-international.org) on interoperability of QSIG with IP 
networks. An earlier version is published as Standard ECMA-339. ECMA 
has made this work available to the IETF as the basis for publishing 
an RFC. 
     
        As the support of telephony within corporate networks evolves from 
        circuit-switched technology to Internet technology, the two 
        technologies will co-exist in many networks for a period, perhaps 
        several years. Therefore there is a need to be able to establish, 
        modify and terminate sessions involving a participant in the SIP

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Date: Thu, 16 Oct 2003 22:43:37 +0300
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Lueken Joachim <joachim.lueken@siemens.com>
CC: "'jdrosen@dynamicsoft.com'" <jdrosen@dynamicsoft.com>,
        "'hgs@cs.columbia.edu'" <hgs@cs.columbia.edu>,
        "'Jon.Peterson@Level3.com'" <Jon.Peterson@Level3.com>,
        "'sipping@ietf.org'" <sipping@ietf.org>
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Subject: [Sipping] Re: PRACK vs. UPDATE in 3Pcc
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The answer is in page 10 of RFC 3262:

  "If the UAC receives a reliable provisional response with an offer
    (this would occur if the UAC sent an INVITE without an offer, in
    which case the first reliable provisional response will contain the
    offer), it MUST generate an answer in the PRACK."

Gonzalo

Lueken Joachim wrote:

> Hi,
> 
> In several flows of the 3PCC draft the sending of PRACK for one leg of the call depends on the reception of a message on the other leg of the call, e.g. in figure 8 the PRACK is sent to B once the 200 w/ answer2 is received from A.
> 
> Why did you not consider to use UPDATE here. Thus PRACK is no longer of end-to-end relevance and acknowledges reception of responses related to only one SIP call leg. I thought this was the intention behind PRACK. Although we introduce two additional message it looks much cleaner to me.
> 
> The flow in figure 8 would then look like this:
> 
> A                 Controller                  B        
> |                      |                      |        
> |(1) INVITE offer1     |                      |        
> |no media              |                      |        
> |<---------------------|                      |        
> |                      |                      |        
> |<ring>                |                      |        
> |                      |                      |        
> |<answer>              |                      |        
> |                      |                      |        
> |(2) 200 answer1       |                      |        
> |no media              |                      |        
> |--------------------->|                      |        
> |(3) ACK               |                      |        
> |<---------------------|                      |        
> |                      |(4) INVITE no SDP     |        
> |                      |--------------------->|        
> |                      |                      |<ring>  
> |                      |(5) 183 offer2        |        
> |                      |<---------------------|        
> |(6) INVITE offer2'    |(6a) PRACK            |        
> |<---------------------|--------------------->|       
> |(7) 200 answer2'      |(6c) 200 PRACK        |        
> |--------------------->|<---------------------|        
> |(8) ACK               |                      |        
> |<---------------------|                      |        
> |                      |(9) UPDATE answer2    |        
> |                      |--------------------->|        
> |                      |(10) 200 UPDATE       |        
> |                      |<---------------------|        
> |(11) RTP              |                      |        
> |.............................................|        
> |                      |                      |<answer>
> |                      |(12) 200 OK           |        
> |                      |<---------------------|        
> |                      |(13) ACK              |        
> |                      |--------------------->|        
> 
> 
> Best regards / Mit freundlichen Gruessen
> Joachim Lueken
>  
> ---------------------------------------- 
> SIEMENS 
> Dept. ICN CP D NC B 
> Hofmannstr. 51, 81359 Munich, Germany 
> Tel: +49 89 722-63251 
> Fax: +49 89 722-28770 
> eMail: joachim.lueken@siemens.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


_______________________________________________
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From exim@www1.ietf.org  Fri Oct 17 03:06:49 2003
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James,

I guess the general issue here is creating a new name space (protocol 
element for Reason) for preemption stuff. *If* we do that, I would 
rather create a general name space and have numeric codes for things 
like network preemption or user agent preemption.

So, rather than this syntax:

Reason : preempted_Network
Reason : preempted_UA

I would use this one:

Reason: preemption ;cause=1 ;text="Network Preemption"
Reason: preemption ;cause=2 ;text="UA Preemption"

We could even choose a more general name space than "preemption" and 
include things like loss of radio converage, for instance.

Gonzalo



Internet-Drafts@ietf.org wrote:

> A New Internet-Draft is available from the on-line Internet-Drafts directories.
> 
> 
> 	Title		: Extending the Session Initiation Protocol Reason 
> 			  Header to account for Preemption Events
> 	Author(s)	: J. Polk
> 	Filename	: draft-polk-sipping-reason-header-for-preemption-00.txt
> 	Pages		: 14
> 	Date		: 2003-10-10
> 	
> This document proposes an extension to the Session Initiation 
> Protocol (SIP) Reason Header [1] to include in a BYE Method Request 
> [2] as a result of a session preemption event either at a user 
> agent (UA), or somewhere in the network using RSVP [3].  This 
> document does not attempt to address routers failing in the packet 
> path; but a deliberate event of tearing down a flow between UAs.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-polk-sipping-reason-header-for-preemption-00.txt
> 
> To remove yourself from the IETF Announcement list, send a message to 
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> 
> Internet-Drafts are also available by anonymous FTP. Login with the username
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> 
> A list of Internet-Drafts directories can be found in
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> 
> 
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-- 
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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From: Lueken Joachim <joachim.lueken@siemens.com>
To: "'jdrosen@dynamicsoft.com'" <jdrosen@dynamicsoft.com>,
        "'Gonzalo.Camarillo@ericsson.com'" <Gonzalo.Camarillo@ericsson.com>,
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Subject: [Sipping] PRACK vs. UPDATE in 3Pcc
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Hi,

In several flows of the 3PCC draft the sending of PRACK for one leg of the call depends on the reception of a message on the other leg of the call, e.g. in figure 8 the PRACK is sent to B once the 200 w/ answer2 is received from A.

Why did you not consider to use UPDATE here. Thus PRACK is no longer of end-to-end relevance and acknowledges reception of responses related to only one SIP call leg. I thought this was the intention behind PRACK. Although we introduce two additional message it looks much cleaner to me.

The flow in figure 8 would then look like this:

A                 Controller                  B        
|                      |                      |        
|(1) INVITE offer1     |                      |        
|no media              |                      |        
|<---------------------|                      |        
|                      |                      |        
|<ring>                |                      |        
|                      |                      |        
|<answer>              |                      |        
|                      |                      |        
|(2) 200 answer1       |                      |        
|no media              |                      |        
|--------------------->|                      |        
|(3) ACK               |                      |        
|<---------------------|                      |        
|                      |(4) INVITE no SDP     |        
|                      |--------------------->|        
|                      |                      |<ring>  
|                      |(5) 183 offer2        |        
|                      |<---------------------|        
|(6) INVITE offer2'    |(6a) PRACK            |        
|<---------------------|--------------------->|       
|(7) 200 answer2'      |(6c) 200 PRACK        |        
|--------------------->|<---------------------|        
|(8) ACK               |                      |        
|<---------------------|                      |        
|                      |(9) UPDATE answer2    |        
|                      |--------------------->|        
|                      |(10) 200 UPDATE       |        
|                      |<---------------------|        
|(11) RTP              |                      |        
|.............................................|        
|                      |                      |<answer>
|                      |(12) 200 OK           |        
|                      |<---------------------|        
|                      |(13) ACK              |        
|                      |--------------------->|        


Best regards / Mit freundlichen Gruessen
Joachim Lueken
 
---------------------------------------- 
SIEMENS 
Dept. ICN CP D NC B 
Hofmannstr. 51, 81359 Munich, Germany 
Tel: +49 89 722-63251 
Fax: +49 89 722-28770 
eMail: joachim.lueken@siemens.com
---------------------------------------- 

 



_______________________________________________
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Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Fri Oct 17 07:42:23 2003
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From: Lueken Joachim <joachim.lueken@siemens.com>
To: "'jdrosen@dynamicsoft.com'" <jdrosen@dynamicsoft.com>,
        "'Gonzalo.Camarillo@ericsson.com'" <Gonzalo.Camarillo@ericsson.com>,
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Cc: "'sipping@ietf.org'" <sipping@ietf.org>
Subject: [Sipping] PRACK vs. UPDATE in 3PCC
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Hi,

In several flows of the 3PCC draft the sending of PRACK for one leg of the call depends on the reception of a message on the other leg of the call, e.g. in figure 8 the PRACK is sent to B once the 200 w/ answer2 is received from A.

Why did you not consider to use UPDATE here. Thus PRACK is no longer of end-to-end relevance and acknowledges reception of responses related to only one SIP call leg. I thought this was the intention behind PRACK. Although we introduce two additional message it looks much cleaner to me.

The flow in figure 8 would then look like this:

A                 Controller                  B        
|                      |                      |        
|(1) INVITE offer1     |                      |        
|no media              |                      |        
|<---------------------|                      |        
|                      |                      |        
|<ring>                |                      |        
|                      |                      |        
|<answer>              |                      |        
|                      |                      |        
|(2) 200 answer1       |                      |        
|no media              |                      |        
|--------------------->|                      |        
|(3) ACK               |                      |        
|<---------------------|                      |        
|                      |(4) INVITE no SDP     |        
|                      |--------------------->|        
|                      |                      |<ring>  
|                      |(5) 183 offer2        |        
|                      |<---------------------|        
|(6) INVITE offer2'    |(6a) PRACK            |        
|<---------------------|--------------------->|       
|(7) 200 answer2'      |(6c) 200 PRACK        |        
|--------------------->|<---------------------|        
|(8) ACK               |                      |        
|<---------------------|                      |        
|                      |(9) UPDATE answer2    |        
|                      |--------------------->|        
|                      |(10) 200 UPDATE       |        
|                      |<---------------------|        
|(11) RTP              |                      |        
|.............................................|        
|                      |                      |<answer>
|                      |(12) 200 OK           |        
|                      |<---------------------|        
|                      |(13) ACK              |        
|                      |--------------------->|        


Best regards / Mit freundlichen Gruessen
Joachim Lueken
 
---------------------------------------- 
SIEMENS 
Dept. ICN CP D NC B 
Hofmannstr. 51, 81359 Munich, Germany 
Tel: +49 89 722-63251 
Fax: +49 89 722-28770 
eMail: joachim.lueken@siemens.com
---------------------------------------- 


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Fri, 17 Oct 2003 07:53:29 -0400
From: "David R. Oran" <oran@cisco.com>
To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>, jmpolk@cisco.com
cc: sipping@ietf.org
Subject: Re: [Sipping] I-D
 ACTION:draft-polk-sipping-reason-header-for-preemption-00.txt
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--On Friday, October 17, 2003 10:05 AM +0300 Gonzalo Camarillo=20
<Gonzalo.Camarillo@ericsson.com> wrote:

> James,
>
> I guess the general issue here is creating a new name space (protocol
> element for Reason) for preemption stuff. *If* we do that, I would rather
> create a general name space and have numeric codes for things like
> network preemption or user agent preemption.
>
> So, rather than this syntax:
>
> Reason : preempted_Network
> Reason : preempted_UA
>
> I would use this one:
>
> Reason: preemption ;cause=3D1 ;text=3D"Network Preemption"
> Reason: preemption ;cause=3D2 ;text=3D"UA Preemption"
>
This does seem more flexible and extensible.

> We could even choose a more general name space than "preemption" and
> include things like loss of radio converage, for instance.
>
I'm not sure I buy into this example - loss of radio coverage is a whole=20
different class of reason - your call was not preempted by something else=20
in this case, unless in fact you know you're not getting signal because the =

frequency/timeslot/codepoint was re-assigned out from under you.

Dave.

> Gonzalo
>
>
>
> Internet-Drafts@ietf.org wrote:
>
>> A New Internet-Draft is available from the on-line Internet-Drafts
>> directories.
>>
>>
>> 	Title		: Extending the Session Initiation Protocol Reason
>> 			  Header to account for Preemption Events
>> 	Author(s)	: J. Polk
>> 	Filename	: draft-polk-sipping-reason-header-for-preemption-00.txt
>> 	Pages		: 14
>> 	Date		: 2003-10-10
>> 	
>> This document proposes an extension to the Session Initiation
>> Protocol (SIP) Reason Header [1] to include in a BYE Method Request
>> [2] as a result of a session preemption event either at a user
>> agent (UA), or somewhere in the network using RSVP [3].  This
>> document does not attempt to address routers failing in the packet
>> path; but a deliberate event of tearing down a flow between UAs.
>>
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-polk-sipping-reason-header-for
>> -preemption-00.txt
>>
>> To remove yourself from the IETF Announcement list, send a message to
>> ietf-announce-request with the word unsubscribe in the body of the
>> message.
>>
>> Internet-Drafts are also available by anonymous FTP. Login with the
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>> logging in, type "cd internet-drafts" and then
>> 	"get draft-polk-sipping-reason-header-for-preemption-00.txt".
>>
>> A list of Internet-Drafts directories can be found in
>> http://www.ietf.org/shadow.html
>> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>>
>>
>> Internet-Drafts can also be obtained by e-mail.
>>
>> Send a message to:
>> 	mailserv@ietf.org.
>> In the body type:
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>> 	/internet-drafts/draft-polk-sipping-reason-header-for-preemption-00.txt
>> 	".
>> 	
>> NOTE:	The mail server at ietf.org can return the document in
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>> 		
>> 		
>> Below is the data which will enable a MIME compliant mail reader
>> implementation to automatically retrieve the ASCII version of the
>> Internet-Draft.



------------------------
David R. Oran
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720
Office: +1 978 264 2048
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Subject: Re: [Sipping] [Fwd: I-D ACTION:draft-taylor-sipping-emerg-scen-00.txt]
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Comments on draft-taylor-sipping-emerg-scen-00.txt, based on the =
information sent to Tom about the UK emergency service (999).

1. Abstract: rather than "would be helpful" in the major finding, it =
would be better to say: " The major finding from the point of view of =
SIP standardization is it is essential in a number of scenarios (and =
helpful for all) if a location header field could be added either by the =
SIP phone or by a proxy to indicate the location of the phone in a =
machine-readable way." The UK uses automated location information at a =
call centre to determine the "local" emergency service, and the =
emergency service uses the information to dispatch the emergency =
response.

2. In section 2 (definitions), the definition provided of an ECC is USA =
specific. Please add, after "(police, fire department, medical aid) the =
words "either directly or by connection to a separate centre" - this =
will cover both cases.

3. In Section 3.1, System Components, the UK is described as having a =
system similar to the USA. It is not, it is radically different and more =
similar to the Australian system. Please modify, by deleting "and =
Britain" after ".  In other       places, such as North America ", and =
insert "and the UK" after " such as Australia " a couple of lines =
higher.

4. Section 3.2 "System Requirements" under "(2) Call processing =
entities"       add to the end of the parenthesised sentence " directly =
or by further onward connection to a service-specific despatching centre =
" - so that it reads "(i.e., one which can dispatch emergency service =
resources to the caller's location)."=20

5. Again in Section 3.2, under (=A3) and (4), the document reads as =
though it is placing requirements on the PSTN, rather than defining =
requirements for SIP. The PSTN is unlikely to change! This part needs =
rewording as follows:
(3)   The telephone network should be supplied-with, or enabled to =
supply, a calling party number or other unique geographic indicator, =
that can serve as a key to the mapping database.=20
(4)   The caller's location needs to be supplied during call set-up, or =
by being keyed by calling party number or other unique identifier, is =
available in the mapping database.  In this document this is assumed to =
happen as a configuration step in advance of the call.=20

6. in (5) it states:
(5)   At least in some jurisdictions, it is required that the ECC=20
      call taker can stay in touch with the caller, even though the=20
      telephone set goes on hook for an intervening period. =
Alternatively=20
      the ECC call taker must be given a callback number -- one which =
can=20
      be dialled to reach the original caller.=20
The second sentence is a different requirement. Consider the scenario =
where someone is attacked, manages to reach a phone and dial the =
emergency number, it is answered, but before being able to pass any =
information other than 'I am being attacked', the attacker causes the =
phone to go on hook for long enough to clear the call. As the attacker =
continues the phone goes off hook again - thus effectively preventing =
any call back. A callback number is more usually used when the emergency =
services response is unable to find the location of the incident and =
needs to contact the original reporter in order to get directions. I =
suggest a separate point (e.g. (8), saying:
(8) The telephone network should be supplied-with, or enabled to supply, =
a number which enables the ECC to callback the SIP phone that made the =
original emergency call.

7. In (7) of the system requirements, you do not really place any real =
requirements on SIP. The caller should be accountable, but how? Suggest =
you add to the existing text of (7):
"In particular callers should not be able to withold calling number or =
location information when using emergency codes and have call records =
stored for at least initial and final steps in call path within public =
call processing entities."

8. Section 3.3: Legacy system operation. The first paragraph would =
benefit from some additional clarification on how the system (at least =
in the UK) operates. I suggest the following addition as the last =
sentence of the first paragraph of 3.3:
"This is presented to the emergency call taker to speed-up call handling =
and response times on all calls, and to be of especially important =
assistance in cases where the caller is unable to provide the =
information. (This is now part of the regulatory regime for emergency =
calls in the US and European Community)."

9. The second paragraph of 3.3 appears to be a description of the UK =
system, and I suggest the following as that paragraph:
A special arrangement at the switch serving the ECC may allow the =
emergency call taker to hold open the voice path back to the caller as =
long as necessary, even if the caller goes on-hook to help establish if =
the call is genuine. Note that this functionality is not supported for =
ISDN.=20

10. The third paragraph, beginning "where this arrangement does not =
apply", delete "where this arrangement does not apply" and replace with =
"For further checks after call connection, and when it is not possible =
to hold the line" (a better description).

11. The last paragraph of 3.3 would be more helpful if we describe the =
configuration data the PSTN uses in their switches, and why:
"In the legacy network, PSTN service providers are responsible for =
maintaining the ECC mapping database and, of course, the routing =
information and Calling Party Number information by configuration of =
data in their switches based on knowing call's originating location.  =
Updates are required every time the relationship between incoming =
circuit, the calling number assigned to that circuit, and the location =
of the terminal at the other end of that circuit changes.  It can =
typically take a day to put through a change in the ECC mapping =
database, meaning that advance coordination is required to ensure =
consistency between configuration and reality."

12. Section 4.2.1, penultimate paragraph: rather than PSTN having "to =
know what number to use", it would be better to say what SIP needs to =
do: "the PSTN gateway also needs to be provided with a callback number =
to use

13. In 4.3.2, last bullet, beginning "The PSTN gateway is responsible =
for ...", you seem to be placing requirements on the PSTN gateway. The =
PSTN gateway can only act on the information it receives... Suggest =
adding "(based on information from the SIP network)" after "a called =
party number on the PSTN side".

14. In the conclusion, I suggest that "The ability to signal that =
location in SIP would be helpful" is strengthened by adding "in all =
cases and essential in a number of scenarios."

15. In 5. Conclusions, before the paragraph beginning "Finally ...", I =
suggest a new paragraph that states:
"There is now an urgency to arrive at emergency call standards with use =
of VoIP growing and the potential for existing implementations to result =
in problems and delays in organising responses for ECCs."

16. In 6. Security Considerations, it may be helpful to add to the first =
bullet as follows:
abuse of the service (i.e., use to make non-emergency calls); this is of =
critical importance with hoax, false and accidental calls being more =
than half the emergency calls received in some countries;

I hope that these comments will help to progress this draft and assist =
in the development of SIP for this life-saving sevice.

Dick Knight




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This list is for NEW development of the application of SIP
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Cc: brian.rosen@marconi.com (Brian Rosen)
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Hi Folks,

I just submitted an I-D summarizing my proposal wrt user=phone.  Until  
it appears in the archive, you can find it on softarmor:

http://www.softarmor.com/wgdb/docs/draft-mahy-sipping-user-equals- 
phone-00.txt
http://www.softarmor.com/wgdb/docs/draft-mahy-sipping-user-equals- 
phone-00.html

many thanks,
-rohan


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Fri, 17 Oct 2003 15:35:51 -0500
To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: [Sipping] I-D
  ACTION:draft-polk-sipping-reason-header-for-preemption-00.txt
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Thank you for your comments

Mine are in-line below

At 10:05 AM 10/17/2003 +0300, Gonzalo Camarillo wrote:
>James,
>
>I guess the general issue here is creating a new name space (protocol 
>element for Reason) for preemption stuff. *If* we do that, I would rather 
>create a general name space and have numeric codes for things like network 
>preemption or user agent preemption.
>
>So, rather than this syntax:
>
>Reason : preempted_Network
>Reason : preempted_UA
>
>I would use this one:
>
>Reason: preemption ;cause=1 ;text="Network Preemption"
>Reason: preemption ;cause=2 ;text="UA Preemption"

The way I read 3326, your suggested syntax wasn't really possible - but 
since you're the chair of the group, I am happy to agree to this suggestion 
(and will change the next rev of the doc).


>We could even choose a more general name space than "preemption" and 
>include things like loss of radio converage, for instance.

I am hesitant to make a more general name space than "preemption" due to 
the document's stated requirement of logging 'the *purposeful* action of 
preemption of a session either at the UA or in the network'. The loss of 
radio coverage is (I think) not purposeful - so it would not meet the 
requirement.

That doesn't mean there can't be another name space for inadvertent or 
unintended session loss. I have no problem with this - but is does mean 
something different enough from preemption that I feel the two are independent.

If you believe this other name space should still be created, should this 
be in this ID or another? I'm leaning towards the latter at the moment 
(which states this is not for cases that support the preemption namespace).

Comments?


>Gonzalo
>
>
>
>Internet-Drafts@ietf.org wrote:
>
>>A New Internet-Draft is available from the on-line Internet-Drafts 
>>directories.
>>
>>         Title           : Extending the Session Initiation Protocol 
>> Reason                        Header to account for Preemption Events
>>         Author(s)       : J. Polk
>>         Filename        : 
>> draft-polk-sipping-reason-header-for-preemption-00.txt
>>         Pages           : 14
>>         Date            : 2003-10-10
>>
>>This document proposes an extension to the Session Initiation Protocol 
>>(SIP) Reason Header [1] to include in a BYE Method Request [2] as a 
>>result of a session preemption event either at a user agent (UA), or 
>>somewhere in the network using RSVP [3].  This document does not attempt 
>>to address routers failing in the packet path; but a deliberate event of 
>>tearing down a flow between UAs.
>>A URL for this Internet-Draft is:
>>http://www.ietf.org/internet-drafts/draft-polk-sipping-reason-header-for-preemption-00.txt
>>To remove yourself from the IETF Announcement list, send a message to 
>>ietf-announce-request with the word unsubscribe in the body of the message.
>>Internet-Drafts are also available by anonymous FTP. Login with the username
>>"anonymous" and a password of your e-mail address. After logging in,
>>type "cd internet-drafts" and then
>>         "get draft-polk-sipping-reason-header-for-preemption-00.txt".
>>A list of Internet-Drafts directories can be found in
>>http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>>
>>Internet-Drafts can also be obtained by e-mail.
>>Send a message to:
>>         mailserv@ietf.org.
>>In the body type:
>>         "FILE 
>> /internet-drafts/draft-polk-sipping-reason-header-for-preemption-00.txt".
>>
>>NOTE:   The mail server at ietf.org can return the document in
>>         MIME-encoded form by using the "mpack" utility.  To use this
>>         feature, insert the command "ENCODING mime" before the "FILE"
>>         command.  To decode the response(s), you will need "munpack" or
>>         a MIME-compliant mail reader.  Different MIME-compliant mail readers
>>         exhibit different behavior, especially when dealing with
>>         "multipart" MIME messages (i.e. documents which have been split
>>         up into multiple messages), so check your local documentation on
>>         how to manipulate these messages.
>>
>>
>>Below is the data which will enable a MIME compliant mail reader
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>>Internet-Draft.
>
>--
>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
>


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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



From exim@www1.ietf.org  Fri Oct 17 16:43:25 2003
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Date: Fri, 17 Oct 2003 15:40:16 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
In-reply-to: <4E4F4B7EF2A53842971777670C0E513C04719A8E@DGEXCH05.mcilink.com>
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Subject: [Sipping] RE: I-D ACTION:draft-sinnreich-sipdev-req-02.txt
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> On page 10 of the draft you reference FXS port; 
> wouldn't that actually be FXO port to support FXS devices?

Good catch. Thanks and we will edit.

Henry

-----Original Message-----
From: Brierre, Jean Joseph [mailto:Jean.Joseph.Brierre@mci.com] 
Sent: Friday, October 17, 2003 3:31 PM
To: 'sipping@ietf.org'; Sinnreich, Heinrich 
Subject: RE: I-D ACTION:draft-sinnreich-sipdev-req-02.txt

Henry,


Would it be preferential if SIP requirement 7 says mwi from message store
application versus just voicemail since you are also asking for simple
support from client?

On page 10 of the draft you reference FXS port; wouldn't that actually be
FXO port to support FXS devices?



-----Original Message-----
From: owner-ietf-announce@ietf.org [mailto:owner-ietf-announce@ietf.org] On
Behalf Of Internet-Drafts@ietf.org
Sent: Thursday, October 16, 2003 2:30 PM
Subject: I-D ACTION:draft-sinnreich-sipdev-req-02.txt

A New Internet-Draft is available from the on-line Internet-Drafts
directories.


	Title		: SIP Telephony Device Requirements, Configuration
and Data
	Filename	: draft-sinnreich-sipdev-req-02.txt
	Pages		: 40
	Date		: 2003-10-16
	
This informational I-D describes the requirements for SIP Telephony 
devices, based on the deployment experience of large numbers of SIP 
phones and PC clients using different implementations. The document 
reviews the generic requirements for SIP telephony devices, the 
automatic device configuration process, the device configuration data 
and examples for XML configuration data formats. 
SIP telephony devices are highly complex IP endpoints that speak many 
Internet protocols, have text, audio and visual interfaces, various 
input modes, and require functionality targeted at several 
constituencies: (1) End users, (2) service providers and network 
administrators and (3) manufacturers and system integrators. 
The objectives of the requirements are a minimum set of 
interoperability and multi-vendor supported core features, so as to 
enable similar ease of purchase, installation and operation as found 
for standard PCs, analog feature phones or mobile phones.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-sinnreich-sipdev-req-02.txt

To remove yourself from the IETF Announcement list, send a message to 
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To: "'Rohan Mahy'" <rohan@cisco.com>,
        "'sipping@ietf.org'"
	 <sipping@ietf.org>
Cc: "'brian.rosen@marconi.com'" <brian.rosen@marconi.com>
Subject: RE: [Sipping] user=phone discussion
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Excellent.

Minor typos:

- In section 2, replace "Illegally formatted URIs?" with "Illegally
formatted URIs:"

- In section 9, replace "Francios" with "Francois" (everybody does this:
don't know why...)

- In the References section, update reference [4] to
draft-ietf-iptel-rfc2806bis-02 (not 
  bis-00: your date is OK however)


Minor Comments:

- Section 5: I can't find anywhere the convention that E is used to
represent "*" and 
  F is used to represent "#". Can you explain where this comes from? In
every single
  source I see (including Q.23), the characters are written as is. In any
case, 2806-bis 
  already states that the characters '*' and '#' can be used as is in a
local-number tel 
  URI (see phonedigit-hex in ABNF), so there is no need for the E/F
convention in the first 
  place. In orther words, tel:*1234#;phone-context=alice.example.com and
  sip:@1234#;phone-context=atlanta.example.com@sip.example.com;user=phone
are already
  legal.

- Section 3: FYI, another more obvious example of conflict beyond the
Compuserve example 
  is that 505-2345678@foo.com could mean either a phone number in New Mexico
(US Area
  Code 505) or a phone number in Nicaragua (Country code 505). They both use
a 
  7-digit number after the (505) so it wouldn't be resolvable with the
length of the
  number as people often try to do. This is a real example, not a
speculation.

- Section 4: <context> and <sipserver> are independent and don't need to
share the 
  same domain name. All the examples use .example.com for both. But it would
be perfectly
  valid to have
sip:1234;phone-context=atlanta.mystartup.com@sip.isp.net;user=phone. This
  is obvious to most of us, but perhaps not to everybody.


> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@cisco.com] 
> Sent: Friday, October 17, 2003 10:28
> To: 'sipping@ietf.org'
> Cc: brian.rosen@marconi.com
> Subject: [Sipping] user=phone discussion
> 
> 
> Hi Folks,
> 
> I just submitted an I-D summarizing my proposal wrt 
> user=phone.  Until  
> it appears in the archive, you can find it on softarmor:
> 
http://www.softarmor.com/wgdb/docs/draft-mahy-sipping-user-equals- 
phone-00.txt
http://www.softarmor.com/wgdb/docs/draft-mahy-sipping-user-equals- 
phone-00.html

many thanks,
-rohan


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip Use
sip@ietf.org for new developments of core SIP

------_=_NextPart_001_01C3950F.7CB96EE2
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Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2656.31">
<TITLE>RE: [Sipping] user=3Dphone discussion</TITLE>
</HEAD>
<BODY>

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

<P><FONT SIZE=3D2>Minor typos:</FONT>
</P>

<P><FONT SIZE=3D2>- In section 2, replace &quot;Illegally formatted =
URIs?&quot; with &quot;Illegally formatted URIs:&quot;</FONT>
</P>

<P><FONT SIZE=3D2>- In section 9, replace &quot;Francios&quot; with =
&quot;Francois&quot; (everybody does this: don't know why...)</FONT>
</P>

<P><FONT SIZE=3D2>- In the References section, update reference [4] to =
draft-ietf-iptel-rfc2806bis-02 (not </FONT>
<BR><FONT SIZE=3D2>&nbsp; bis-00: your date is OK however)</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Minor Comments:</FONT>
</P>

<P><FONT SIZE=3D2>- Section 5: I can't find anywhere the convention =
that E is used to represent &quot;*&quot; and </FONT>
<BR><FONT SIZE=3D2>&nbsp; F is used to represent &quot;#&quot;. Can you =
explain where this comes from? In every single</FONT>
<BR><FONT SIZE=3D2>&nbsp; source I see (including Q.23), the characters =
are written as is. In any case, 2806-bis </FONT>
<BR><FONT SIZE=3D2>&nbsp; already states that the characters '*' and =
'#' can be used as is in a local-number tel </FONT>
<BR><FONT SIZE=3D2>&nbsp; URI (see phonedigit-hex in ABNF), so there is =
no need for the E/F convention in the first </FONT>
<BR><FONT SIZE=3D2>&nbsp; place. In orther words, =
tel:*1234#;phone-context=3Dalice.example.com and</FONT>
<BR><FONT SIZE=3D2>&nbsp; =
sip:@1234#;phone-context=3Datlanta.example.com@sip.example.com;user=3Dph=
one are already</FONT>
<BR><FONT SIZE=3D2>&nbsp; legal.</FONT>
</P>

<P><FONT SIZE=3D2>- Section 3: FYI, another more obvious example of =
conflict beyond the Compuserve example </FONT>
<BR><FONT SIZE=3D2>&nbsp; is that 505-2345678@foo.com could mean either =
a phone number in New Mexico (US Area</FONT>
<BR><FONT SIZE=3D2>&nbsp; Code 505) or a phone number in Nicaragua =
(Country code 505). They both use a </FONT>
<BR><FONT SIZE=3D2>&nbsp; 7-digit number after the (505) so it wouldn't =
be resolvable with the length of the</FONT>
<BR><FONT SIZE=3D2>&nbsp; number as people often try to do. This is a =
real example, not a speculation.</FONT>
</P>

<P><FONT SIZE=3D2>- Section 4: &lt;context&gt; and &lt;sipserver&gt; =
are independent and don't need to share the </FONT>
<BR><FONT SIZE=3D2>&nbsp; same domain name. All the examples use =
.example.com for both. But it would be perfectly</FONT>
<BR><FONT SIZE=3D2>&nbsp; valid to have =
sip:1234;phone-context=3Datlanta.mystartup.com@sip.isp.net;user=3Dphone.=
 This</FONT>
<BR><FONT SIZE=3D2>&nbsp; is obvious to most of us, but perhaps not to =
everybody.</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Rohan Mahy [<A =
HREF=3D"mailto:rohan@cisco.com">mailto:rohan@cisco.com</A>] </FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Friday, October 17, 2003 10:28</FONT>
<BR><FONT SIZE=3D2>&gt; To: 'sipping@ietf.org'</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: brian.rosen@marconi.com</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: [Sipping] user=3Dphone =
discussion</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Hi Folks,</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I just submitted an I-D summarizing my proposal =
wrt </FONT>
<BR><FONT SIZE=3D2>&gt; user=3Dphone.&nbsp; Until&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; it appears in the archive, you can find it on =
softarmor:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://www.softarmor.com/wgdb/docs/draft-mahy-sipping-user-equal=
s-" =
TARGET=3D"_blank">http://www.softarmor.com/wgdb/docs/draft-mahy-sipping-=
user-equals-</A> </FONT>
<BR><FONT SIZE=3D2>phone-00.txt</FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://www.softarmor.com/wgdb/docs/draft-mahy-sipping-user-equal=
s-" =
TARGET=3D"_blank">http://www.softarmor.com/wgdb/docs/draft-mahy-sipping-=
user-equals-</A> </FONT>
<BR><FONT SIZE=3D2>phone-00.html</FONT>
</P>

<P><FONT SIZE=3D2>many thanks,</FONT>
<BR><FONT SIZE=3D2>-rohan</FONT>
</P>
<BR>

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

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_______________________________________________
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Date: Sat, 18 Oct 2003 08:54:59 +0300
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: "James M. Polk" <jmpolk@cisco.com>
CC: sipping@ietf.org
Subject: Re: [Sipping] I-D  ACTION:draft-polk-sipping-reason-header-for-preemption-00.txt
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James,

my point was that the name space needs to be as general as possible. In 
your original draft you were using two names spaces for preemption, 
while one, as suggested, was be enough... still, I was wondering if 
there was an even more general name space than preemption. But if you 
cannot think of a more general name space, it is fine with me. I only 
wanted you to give it a thought, since you know exactl best the types of 
applications you have in mind.

Gonzalo

James M. Polk wrote:
> Thank you for your comments
> 
> Mine are in-line below
> 
> At 10:05 AM 10/17/2003 +0300, Gonzalo Camarillo wrote:
> 
>> James,
>>
>> I guess the general issue here is creating a new name space (protocol 
>> element for Reason) for preemption stuff. *If* we do that, I would 
>> rather create a general name space and have numeric codes for things 
>> like network preemption or user agent preemption.
>>
>> So, rather than this syntax:
>>
>> Reason : preempted_Network
>> Reason : preempted_UA
>>
>> I would use this one:
>>
>> Reason: preemption ;cause=1 ;text="Network Preemption"
>> Reason: preemption ;cause=2 ;text="UA Preemption"
> 
> 
> The way I read 3326, your suggested syntax wasn't really possible - but 
> since you're the chair of the group, I am happy to agree to this 
> suggestion (and will change the next rev of the doc).
> 
> 
>> We could even choose a more general name space than "preemption" and 
>> include things like loss of radio converage, for instance.
> 
> 
> I am hesitant to make a more general name space than "preemption" due to 
> the document's stated requirement of logging 'the *purposeful* action of 
> preemption of a session either at the UA or in the network'. The loss of 
> radio coverage is (I think) not purposeful - so it would not meet the 
> requirement.
> 
> That doesn't mean there can't be another name space for inadvertent or 
> unintended session loss. I have no problem with this - but is does mean 
> something different enough from preemption that I feel the two are 
> independent.
> 
> If you believe this other name space should still be created, should 
> this be in this ID or another? I'm leaning towards the latter at the 
> moment (which states this is not for cases that support the preemption 
> namespace).
> 
> Comments?
> 
> 
>> Gonzalo
>>
>>
>>
>> Internet-Drafts@ietf.org wrote:
>>
>>> A New Internet-Draft is available from the on-line Internet-Drafts 
>>> directories.
>>>
>>>         Title           : Extending the Session Initiation Protocol 
>>> Reason                        Header to account for Preemption Events
>>>         Author(s)       : J. Polk
>>>         Filename        : 
>>> draft-polk-sipping-reason-header-for-preemption-00.txt
>>>         Pages           : 14
>>>         Date            : 2003-10-10
>>>
>>> This document proposes an extension to the Session Initiation 
>>> Protocol (SIP) Reason Header [1] to include in a BYE Method Request 
>>> [2] as a result of a session preemption event either at a user agent 
>>> (UA), or somewhere in the network using RSVP [3].  This document does 
>>> not attempt to address routers failing in the packet path; but a 
>>> deliberate event of tearing down a flow between UAs.
>>> A URL for this Internet-Draft is:
>>> http://www.ietf.org/internet-drafts/draft-polk-sipping-reason-header-for-preemption-00.txt 
>>>
>>> To remove yourself from the IETF Announcement list, send a message to 
>>> ietf-announce-request with the word unsubscribe in the body of the 
>>> message.
>>> Internet-Drafts are also available by anonymous FTP. Login with the 
>>> username
>>> "anonymous" and a password of your e-mail address. After logging in,
>>> type "cd internet-drafts" and then
>>>         "get draft-polk-sipping-reason-header-for-preemption-00.txt".
>>> A list of Internet-Drafts directories can be found in
>>> http://www.ietf.org/shadow.html or 
>>> ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>>>
>>> Internet-Drafts can also be obtained by e-mail.
>>> Send a message to:
>>>         mailserv@ietf.org.
>>> In the body type:
>>>         "FILE 
>>> /internet-drafts/draft-polk-sipping-reason-header-for-preemption-00.txt". 
>>>
>>>
>>> NOTE:   The mail server at ietf.org can return the document in
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>>>         exhibit different behavior, especially when dealing with
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>>>
>>>
>>> Below is the data which will enable a MIME compliant mail reader
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>>> Internet-Draft.
>>
>>
>> -- 
>> 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
>>
> 
> 
> cheers,
> James
> 
>                                *******************
>                 Truth is not to be argued... it is to be presented

-- 
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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This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



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Date: Sat, 18 Oct 2003 09:15:21 -0700
From: Cullen Jennings <fluffy@cisco.com>
To: "'sipping'" <sipping@ietf.org>
CC: Cullen Jennings <fluffy@cisco.com>, David Bryan <dave@bryan.cx>
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Subject: [Sipping] Describing dial tone and such using MIDI
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David Bryan and I have submitted a draft on describing PSTN tones such as
dialtone using MIDI - until it shows up in the drafts directory, you can
find it at:

http://www.employees.org/~fluffy/ietf/draft-bryan-sipping-midi-00.txt

http://www.employees.org/~fluffy/ietf/draft-bryan-sipping-midi-00.html

Cullen


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Date: Sat, 18 Oct 2003 21:50:52 -0700
From: Cullen Jennings <fluffy@cisco.com>
To: "James M. Polk" <jmpolk@cisco.com>, <sipping@ietf.org>
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Subject: [Sipping] Reason Header for Preemption - Why?
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IP phone A is in a call and it gets presented - the call will get torn down.
Does A do something different because the call was torn down because of
preemption? 

Is it just so we can tell the user of phone A something useful about why
their call ended?

Cullen








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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Sun, 19 Oct 2003 09:11:30 -0400
From: "David R. Oran" <oran@cisco.com>
To: Cullen Jennings <fluffy@cisco.com>, "James M. Polk" <jmpolk@cisco.com>,
        sipping@ietf.org
Subject: Re: [Sipping] Reason Header for Preemption - Why?
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--On Saturday, October 18, 2003 9:50 PM -0700 Cullen Jennings 
<fluffy@cisco.com> wrote:

>
> IP phone A is in a call and it gets presented - the call will get torn
> down. Does A do something different because the call was torn down
> because of preemption?
>
Possibly run a longer retry timer if the phone is programmed to try to 
reestablish calls automatically.

> Is it just so we can tell the user of phone A something useful about why
> their call ended?
>
Mostly. It may also have use in statistics so that these calls don't get 
lumped in with stats used to compute MTBF and MTTR.

> Cullen
>
>
>
>
>
>
>
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP



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

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: Re: [Sipping] Reason Header for Preemption - Why?
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At 09:11 AM 10/19/2003 -0400, David R. Oran wrote:
>--On Saturday, October 18, 2003 9:50 PM -0700 Cullen Jennings 
><fluffy@cisco.com> wrote:
>
>>
>>IP phone A is in a call and it gets presented - the call will get torn
>>down. Does A do something different because the call was torn down
>>because of preemption?
>Possibly run a longer retry timer if the phone is programmed to try to 
>reestablish calls automatically.
>
>>Is it just so we can tell the user of phone A something useful about why
>>their call ended?
>Mostly. It may also have use in statistics so that these calls don't get 
>lumped in with stats used to compute MTBF and MTTR.

SIP element synchronization (including Servers) of a specific event is one 
aspect that is important (this would be just as Dave stated - for 
statistics or accounting).

One other one is that UAs that receive this BYE with this Reason value will 
(in some networks) need to play either a specific tone or announcement to 
that user which is appropriate to that event.

If A_B are in a session when C signals to B to preempt any session it is 
currently on, *something* has to tell A it is no longer in the session, a 
BYE is appropriate. But, this is not a normal end of session event, it is 
preemption. If that event needs to cause UA A to play a unique preemption 
tone (or play an announcement), it needs some indication in the BYE to do 
such an action. My proposal for a new Reason value is this indication.

Is this not the general concept behind the Reason Header (to provide more 
information)?

comments


>>Cullen
>>
>>
>>
>>
>>
>>
>>
>>
>>_______________________________________________
>>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>This list is for NEW development of the application of SIP
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sip@ietf.org for new developments of core SIP
>
>
>
>------------------------
>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


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Thread-Topic: quick update on sipping -> mmusic realtimefax draft 
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From: "Jean-Francois Mule" <jf.mule@cablelabs.com>
To: <sipping@ietf.org>, "Hearty, John" <John.Hearty@Level3.com>,
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Well, I've been trying to hit the deadline for the mmusic draft-00 for =
IETF#58 (10/20), and after more considerations, I won't be releasing an =
official I-D for the next meeting.=20

The new draft has a different scope (sdp and use offer/answer for =
capability indication and t.38 negotiation rather than sipping call =
flows). It's been reworked quite significantly but it's not ready. I'd =
rather close some issues on the sipping/mmusic lists than release the =
mmusic draft as-is.

Jean-Fran=E7ois=20

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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [Sipping] Agenda requests
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Folks,

I will be taking care of registering agenda requests for the Minneapolis 
meeting at:
http://www.softarmor.com/sipping/meets/ietf58/requests.html

Send your requests to me including:
o Your name
o Topic you would like to present
o Lenght of the slot
o List of related drafts that people should read

If you use the following template to send me your request, you would 
make this task much easier for me:


<tr>
<td><a href="mailto:Gonzalo.Camarillo@ericsson.com">
Gonzalo Camarillo</a></td>
<td>Exploder requirements</td>
<td><center>5</center></td>
<td>
<a 
href="http://www.ietf.org/internet-drafts/draft-camarillo-sipping-exploders-00.txt">
draft-camarillo-sipping-exploders-00.txt</a>
</td>
</tr>


Thanks,

Gonzalo


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Brierre, Jean Joseph" <Jean.Joseph.Brierre@mci.com>
To: "'sipping@ietf.org'" <sipping@ietf.org>,
        "Sinnreich, Heinrich " <Henry.Sinnreich@mci.com>
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Subject: [Sipping] RE: I-D ACTION:draft-sinnreich-sipdev-req-02.txt
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Henry,


Would it be preferential if SIP requirement 7 says mwi from message store
application versus just voicemail since you are also asking for simple
support from client?

On page 10 of the draft you reference FXS port; wouldn't that actually be
FXO port to support FXS devices?



-----Original Message-----
From: owner-ietf-announce@ietf.org [mailto:owner-ietf-announce@ietf.org] On
Behalf Of Internet-Drafts@ietf.org
Sent: Thursday, October 16, 2003 2:30 PM
Subject: I-D ACTION:draft-sinnreich-sipdev-req-02.txt

A New Internet-Draft is available from the on-line Internet-Drafts
directories.


	Title		: SIP Telephony Device Requirements, Configuration
and Data
	Filename	: draft-sinnreich-sipdev-req-02.txt
	Pages		: 40
	Date		: 2003-10-16
	
This informational I-D describes the requirements for SIP Telephony 
devices, based on the deployment experience of large numbers of SIP 
phones and PC clients using different implementations. The document 
reviews the generic requirements for SIP telephony devices, the 
automatic device configuration process, the device configuration data 
and examples for XML configuration data formats. 
SIP telephony devices are highly complex IP endpoints that speak many 
Internet protocols, have text, audio and visual interfaces, various 
input modes, and require functionality targeted at several 
constituencies: (1) End users, (2) service providers and network 
administrators and (3) manufacturers and system integrators. 
The objectives of the requirements are a minimum set of 
interoperability and multi-vendor supported core features, so as to 
enable similar ease of purchase, installation and operation as found 
for standard PCs, analog feature phones or mobile phones.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-sinnreich-sipdev-req-02.txt

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Internet-Drafts are also available by anonymous FTP. Login with the username
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type "cd internet-drafts" and then
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A list of Internet-Drafts directories can be found in
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or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
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_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Mon Oct 20 10:47:59 2003
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
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Subject: [Sipping] Brian's Comments on draft-mahy-sipping-user-equals-
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This document is a start, but in my opinion not enough.

The introduction must define dial strings, and differentiate them
from local phone numbers.  It should introduce the idea that a
proxy in the listed domain of a dial string must translate to a global
or local number, and that some phones will do this translation
themselves while others will not.

Section 5 inadequately deals with dial strings.  Dial strings are not
local numbers, and must be translated to local numbers.  One must
disambiguate dial strings from local numbers because it is possible that
they collide.  Such a differentiation is also useful to know when
translation has been completed.

It has been proposed to use separate contexts for dial strings and
local numbers.  I generally don't like this idea, but it can be made
to work.  It would be best if there was some kind of naming convention
to differentiate, but this is not strictly necessary.

What is necessary is text in a document:
	1. Saying that dial strings aren't local numbers
	2. Permitting dial strings in a sip uri, with a dial string context.
		Note: if you wish to permit a dial string in a tel uri, then
                 the language in 2806bis must be modified, IMO.
	3. Explicitly defining the use of E/F
	4. Illustrating the use of separate contexts for dial strings
	5. Discussing the provisioning implications of having two
		local contexts, one for phones with translation and
		another for phones without translation

It would be much more satisfactory, in my opinion, to simply define
user=dialstring, or something similar, to denote that the user part is
a dial string.  Note that sip phones that presently put dial strings in
a sip uri without a context must change anyway.

We also should define a way to express a dial string with pauses and waits.
They are pretty useful. They are dialable in many gateways.  We need
a way to send such a string.  Again, we need text, and again, we could
use a separate context for such strings.

I do agree with the draft's concepts that
	user=phone indicates a local or global number as defined by 2806bis
	user=phone without a context on a local number is deprecated

As before, I will write text when we agree we want text to cover.
Thanks to Rohan for this good start.

Brian

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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CC: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [Sipping] draft-hilt-sipping-session-policy-00
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We have submitted the drafts draft-hilt-sipping-session-spec-policy-00
and draft-hilt-sipping-session-indep-policy-00. Until they are available
in the I-D archive, you can download them from:

http://drafts.mhof.com/draft-hilt-sipping-session-indep-policy-00.txt
http://drafts.mhof.com/draft-hilt-sipping-session-spec-policy-00.txt

Thanks,

Volker


-----------
Session Initiation Proposal                                      V. Hilt
Investigation Working Group                Bell Labs/Lucent Technologies
Internet-Draft                                              J. Rosenberg
Expires: April 18, 2004                                      dynamicsoft
                                                         October 19, 2003


  A Framework for Session-Specific Intermediary Session Policies in SIP
                draft-hilt-sipping-session-spec-policy-00

Abstract

    Proxy servers play a central role as an intermediary in the
    establishment of sessions in the Session Initiation Protocol (SIP).
    In that role, they define and impact policies on call routing,
    rendezvous, and other call features. However, there is currently no
    standard means by which network elements can have any influence on
    session policies, such as the codecs that are to be used. In this
    document, we propose a complete and standards-based framework that
    enables intermediaries to request the use of policies in a SIP
    session.


-----------
Session Initiation Proposal                                      V. Hilt
Investigation Working Group                Bell Labs/Lucent Technologies
Internet-Draft                                              G. Camarillo
Expires: April 18, 2004                                         Ericsson
                                                             J. Rosenberg
                                                              dynamicsoft
                                                         October 19, 2003


   A Framework for Session-Independent Intermediary Session Policies in
                                   SIP
                draft-hilt-sipping-session-indep-policy-00

Abstract

    Domains may have policies in place that impact the way sessions are
    established by user agents. Some of these policies are independent of
    a specific session and need to be considered for a certain period of
    time. It is therefore desirable to convey these policies to user
    agents once they become active instead of requesting them for every
    session. In this document, we propose a framework that enables user
    agents to subscribe to session policies.





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This list is for NEW development of the application of SIP
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Proposed Clarification of Encoding of Telephone 
			  Numbers in SIP URis
	Author(s)	: R. Mahy
	Filename	: draft-mahy-sipping-user-equals-phone-00.txt
	Pages		: 12
	Date		: 2003-10-20
	
There is much confusion about how to represent telephone numbers and
other related strings of digits in SIP URIs and tel: URIs, especailly
when those digit strings are of purely local significance. This
matter is complicated by the underspecification of the 'user' SIP URI
parameter in RFC3261 and the phone-context parameter in RFC2806.This
document proposes specific semantics for the user=phone parameter and
describes several ways to encode digits strings and phone numbers in
a manner consistent with that proposal.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-mahy-sipping-user-equals-phone-00.txt

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

	Title		: Early Media and Ringing Tone Generation in the 
			  Session Initiation Protocol
	Author(s)	: G. Camarillo, H. Schulzrinne
	Filename	: draft-ietf-sipping-early-media-00.txt
	Pages		: 11
	Date		: 2003-10-20
	
This document describes how to manage early media in SIP using two
models; the gateway model and the application server model. It also
describes the inputs one needs to consider to define local policies
for ringing tone generation.

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From exim@www1.ietf.org  Mon Oct 20 16:58:43 2003
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	Mon, 20 Oct 2003 15:57:13 -0500
From: "Dean Willis" <dean.willis@softarmor.com>
To: "'sipping'" <sipping@ietf.org>
Cc: "'Bernard Aboba'" <aboba@internaut.com>
Date: Mon, 20 Oct 2003 15:57:00 -0500
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Subject: [Sipping] SIP and AAA open questions and RADIUSEXT meeting
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Let's revisit where we are on the SIP-AAA work. Or at least where I =
THINK we
are, so please help me get this right.

1) We have a requirements draft.
http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-aaa-req-03.txt

2) We have a RADIUS draft that is VERY expired, but=20
variations are widely implemented:
http://www.softarmor.com/wgdb/docs/draft-sterman-aaa-sip-00.txt

3) There's a DIAMETER draft:
http://www.ietf.org/internet-drafts/draft-ietf-aaa-diameter-sip-app-00.tx=
t

Actually, there are a lot of DIAMETER-related drafts dealing with SIP.

4) There's a RADIUS Accounting draft from Henning et al, which I seem to
have misplaced the text of.


We seem to have a consensus to do at least some work in standardizing =
RADIUS
usage. It appears that the likeliest way forward is to conclude on the
Sterman draft. We also seem to have an interest (if not consensus yet) =
in
doing something to help with accounting for SIP operations using RADIUS.


Bernard Aboba, chairing the RADIUSEXT work, would like to know what if
anything we would like RADIUSEXT to include on their agenda for the =
November
meeting. It looks like they MIGHT be meeting Friday morning (agendas do
change).

I'm thinking:

1) The Sterman-AAA draft:
Any volunteers to present and lead the discussion for this in RADIUSEXT?

I believe we have a consensus to proceed with this draft in SIPPING of
RADIUSEXT doesn't want to take the lead. I would propose that the work =
be
jointly reviewed.


2) The RADIUS Accounting draft:
Any volunteers to present and lead the discussion for this in RADIUSEXT? =


Thanks.

--
Dean


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Mon Oct 20 17:45:42 2003
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Date: Mon, 20 Oct 2003 16:40:02 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: [Sipping] SIP and AAA open questions and RADIUSEXT meeting
In-reply-to: <002501c3974c$b55c2750$e1036e3f@txdwillis>
To: "'Dean Willis'" <dean.willis@softarmor.com>
Cc: "'Bernard Aboba'" <aboba@internaut.com>, "'sipping'" <sipping@ietf.org>
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I am afraid I may get ruled out of scope, but maybe we could also extend an
invitation to the authors of numerous Wi-Fi - RADIUS authors and
implementers, since it appears we may have a power combo in SIP-RADIUS and
802.1x, please see for example
http://standards.ieee.org/getieee802/download/802.1X-2001.pdf  Just to have
a more complete background information about the synergies here and the
consistent approach this promises.

At least we should be aware of the synergy here in some of the proposed
SIP-RADIUS I-Ds and the common AAA infrastructure with Wi-Fi Internet
access.

Out of scope? Sure...

Thanks, Henry

Henry Sinnreich

MCI
400 International Parkway
Richardson, Texas 75081
USA

Have you disconnected the PBX?
-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf Of
Dean Willis
Sent: Monday, October 20, 2003 3:57 PM
To: 'sipping'
Cc: 'Bernard Aboba'
Subject: [Sipping] SIP and AAA open questions and RADIUSEXT meeting


Let's revisit where we are on the SIP-AAA work. Or at least where I THINK we
are, so please help me get this right.

1) We have a requirements draft.
http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-aaa-req-03.txt

2) We have a RADIUS draft that is VERY expired, but 
variations are widely implemented:
http://www.softarmor.com/wgdb/docs/draft-sterman-aaa-sip-00.txt

3) There's a DIAMETER draft:
http://www.ietf.org/internet-drafts/draft-ietf-aaa-diameter-sip-app-00.txt

Actually, there are a lot of DIAMETER-related drafts dealing with SIP.

4) There's a RADIUS Accounting draft from Henning et al, which I seem to
have misplaced the text of.


We seem to have a consensus to do at least some work in standardizing RADIUS
usage. It appears that the likeliest way forward is to conclude on the
Sterman draft. We also seem to have an interest (if not consensus yet) in
doing something to help with accounting for SIP operations using RADIUS.


Bernard Aboba, chairing the RADIUSEXT work, would like to know what if
anything we would like RADIUSEXT to include on their agenda for the November
meeting. It looks like they MIGHT be meeting Friday morning (agendas do
change).

I'm thinking:

1) The Sterman-AAA draft:
Any volunteers to present and lead the discussion for this in RADIUSEXT?

I believe we have a consensus to proceed with this draft in SIPPING of
RADIUSEXT doesn't want to take the lead. I would propose that the work be
jointly reviewed.


2) The RADIUS Accounting draft:
Any volunteers to present and lead the discussion for this in RADIUSEXT? 

Thanks.

--
Dean


_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP


_______________________________________________
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Subject: RE: [Sipping] Describing dial tone and such using MIDI
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Thread-Topic: [Sipping] Describing dial tone and such using MIDI
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From: "Eric Burger" <eburger@snowshore.com>
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Looks good.  However, I question saying that the MIDI body is a cid:.  =
Why not just say it's a URI reference, which could be a cid:?

By the way -- the country reference is (a) invaluable and (b) a lot of =
work.  Excellent job!

> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
> Sent: Sat, October 18, 2003 12:15 PM
> To: 'sipping'
> Cc: Cullen Jennings; David Bryan
> Subject: [Sipping] Describing dial tone and such using MIDI
>=20
>=20
>=20
> David Bryan and I have submitted a draft on describing PSTN=20
> tones such as
> dialtone using MIDI - until it shows up in the drafts=20
> directory, you can
> find it at:
>=20
> http://www.employees.org/~fluffy/ietf/draft-bryan-sipping-midi-00.txt
>=20
> http://www.employees.org/~fluffy/ietf/draft-bryan-sipping-midi-00.html
>=20
> Cullen
>=20
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20
>=20


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: SIP Transfer Issues
	Author(s)	: D. Petrie
	Filename	: draft-petrie-sipping-xfer-issues-00.txt
	Pages		: 15
	Date		: 2003-10-20
	
Transfer features in SIP have been enabled via the REFER method  and
Replaces header.  These constructs have been around for a while now
with stable definitions for how they work.  However, there appear to
be a small set of edge cases and requirements that are not yet
satisfied.  This draft attempts to define some of those cases and
requirements.  This is intended to spark discussion to whether these
are legitimate requirements on which further standards work is
appropriate.

A URL for this Internet-Draft is:
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From: "Daniel G. Petrie" <dpetrie@pingtel.com>
Organization: Pingtel Corp.  http://www.pingtel.com
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Subject: [Sipping] I-D ACTION:draft-petrie-sipping-xfer-issues-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.


        Title           : SIP Transfer Issues
        Author(s)       : D. Petrie
        Filename        : draft-petrie-sipping-xfer-issues-00.txt
        Pages           : 15
        Date            : 2003-10-20

Transfer features in SIP have been enabled via the REFER method  and
Replaces header.  These constructs have been around for a while now
with stable definitions for how they work.  However, there appear to
be a small set of edge cases and requirements that are not yet
satisfied.  This draft attempts to define some of those cases and
requirements.  This is intended to spark discussion to whether these
are legitimate requirements on which further standards work is
appropriate.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-petrie-sipping-xfer-issues-00.txt

To remove yourself from the IETF Announcement list, send a message to
ietf-announce-request with the word unsubscribe in the body of the message.

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
        "get draft-petrie-sipping-xfer-issues-00.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
        mailserv@ietf.org.
In the body type:
        "FILE /internet-drafts/draft-petrie-sipping-xfer-issues-00.txt".

NOTE:   The mail server at ietf.org can return the document in
        MIME-encoded form by using the "mpack" utility.  To use this
        feature, insert the command "ENCODING mime" before the "FILE"
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        a MIME-compliant mail reader.  Different MIME-compliant mail readers
        exhibit different behavior, especially when dealing with
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Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

<ftp://ftp.ietf.org/internet-drafts/draft-petrie-sipping-xfer-issues-00.txt>


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: RE: [Sipping] SIP and AAA open questions and RADIUSEXT meeting
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Hi Dean,

Some very quick comments:

> Let's revisit where we are on the SIP-AAA work. Or at least where I =
THINK we
> are, so please help me get this right.
>=20
> 1) We have a requirements draft.
> http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-aaa-req-03.txt
>=20
> 2) We have a RADIUS draft that is VERY expired, but variations are =
widely implemented:
> http://www.softarmor.com/wgdb/docs/draft-sterman-aaa-sip-00.txt
>=20
> 3) There's a DIAMETER draft:
> =
http://www.ietf.org/internet-drafts/draft-ietf-aaa-diameter-sip-app-00.tx=
t

The Diameter work has some constituency (at least 3GPP) and is chartered
in the AAA.  Work is well underway.  We are happy to get input from the
SIP community on this draft.  However,  I think RADIUS extensions for
SIP is a different issue than this.

> 4) There's a RADIUS Accounting draft from Henning et al, which I seem =
to
> have misplaced the text of.
>=20
> We seem to have a consensus to do at least some work in standardizing =
RADIUS
> usage. It appears that the likeliest way forward is to conclude on the
> Sterman draft. We also seem to have an interest (if not consensus yet) =
in
> doing something to help with accounting for SIP operations using =
RADIUS.
>=20
> Bernard Aboba, chairing the RADIUSEXT work, would like to know what if
> anything we would like RADIUSEXT to include on their agenda for the =
November
> meeting. It looks like they MIGHT be meeting Friday morning (agendas =
do
> change).
>=20
> I'm thinking:
>=20
> 1) The Sterman-AAA draft:
> Any volunteers to present and lead the discussion for this in =
RADIUSEXT?
>=20
> I believe we have a consensus to proceed with this draft in SIPPING of
> RADIUSEXT doesn't want to take the lead. I would propose that the work =
be
> jointly reviewed.
>=20
> 2) The RADIUS Accounting draft:
> Any volunteers to present and lead the discussion for this in=20
> RADIUSEXT?=20

I think what is more important for the RADIUS work is to find some
folks who are willing to get the work done.  Wishing & hoping that
the work is done (by someone else) is not a good solution.

thanks,
John

_______________________________________________
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This list is for NEW development of the application of SIP
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Message-ID: <3F964004.4010409@dynamicsoft.com>
Date: Wed, 22 Oct 2003 04:29:56 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [Sipping] app interaction framework: path forward?
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Folks,

I've submitted an update to the app interaction framework. Until it 
appears in the archives, you can pick it up as:

http://www.jdrosen.net/papers/draft-ietf-sipping-app-interaction-framework-00.txt

This represents a pretty substantial change since the last version. 
Based on list discussions and input from a number of people, the draft 
proposes a compromise position around carrying DTMF which I am hoping 
we can move forward with.

The essence of the proposal is that it explicitly recogznies two 
different models - presentation capable user interfaces, such as 
html/wml/voicexml that have a presentation component, and presentation 
free user interfaces, such as kpml, that don't. For the latter, the 
idea is that the application will issue a SUBSCRIBE request, on a new 
dialog, towards the UA (using the GRUU). The body of the SUBSCRIBE 
contains the KPML. the NOTIFY messages contain information on what 
keys were pressed.

For presentation aware URI, the application will send the UA a REFER 
request, sent on a separate dialog, to the GRUU for the user. The 
Refer-To header field contains an HTTP URI pointing to the markup to 
fetch. Once fetched, its normal HTTP operations of posting data and 
getting a new document.

As a consequence of this, there is no longer an App-Info header field. 
I think thats a good thing. It allows for an application to push UI 
components to the UA at any time, without needing to piggyback ontop 
of messages in the context of the original dialog.

The document is a bit rough around the edges, but it conveys this 
proposal in some detail. Once we have consensus around it, I can fill 
in the missing details in the draft (examples and the sip event package).

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com



_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Wed Oct 22 10:30:36 2003
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From: Robert Sparks <rsparks@dynamicsoft.com>
To: sipping@ietf.org
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--=-kWCkslnj1rUVnzREj52D
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-----Forwarded Message-----
> From: Internet-Drafts@ietf.org
> Subject: I-D ACTION:draft-sparks-sipping-load-00.txt
> Date: Mon, 20 Oct 2003 15:43:37 -0400
> 
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
> 
> 
> 	Title		: SIP Load Management
> 	Author(s)	: R. Sparks
> 	Filename	: draft-sparks-sipping-load-00.txt
> 	Pages		: 8
> 	Date		: 2003-10-20
> 	
> As deployments of services based on the Session Initiation Protocol
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From exim@www1.ietf.org  Wed Oct 22 13:07:32 2003
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From: "Daniel G. Petrie" <dpetrie@pingtel.com>
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Thanks Dave for these really great comments.  My comments
are below.
Dan

"David R. Oran" wrote:

> Some initial comments and queries on
> draft-petrie-sipping-xfer-issues-00.txt. In most cases I think there are
> easier fixes than the ones you suggest in the document.
>
> 1. Isn't your "Consultative Transfer to Lost Target" exactly the sort of
> problem GRUU fixes? If TT' returns a GRUU in the 200OK contact of the
> Invite, the TC just puts this GRUU in the Refer-with-replaces to the TE,
> and the TE->TT' Invite is guaranteed to go to the right place.
>

Yes.  Alhough it perhaps was not explicit or clear that is what is intended
as the approach proposed in section 3.6 Use a Target URI

>
> 2. For "Consultative Call Prompting" and other similar advisory notices
> about the purpose of a call, just like for email, isn't the "Subject:"
> header exactly what's called for?

Yes.  I unfortunately missed this before I submitted the draft.  This
helps for prompting, however there is still a need for a symantic
purpose which probably does not make sense in the Subject header
as the syntax is freeform text.

>
>
> 3. For "Consultative Turned Blind Double Ring" I think we could easily give
> the right hint by putting a "Reason: replacement dialog coming" in the
> reason header of the CANCEL. This is exactly the sort of thing the reason
> header was created for.
>

Yes.  This is an excellent solution to the approach indicated in
section 3.2.  Adding an extension to the Reason header to
contain the associated dialog information would be useful
as well.

>
> 4. "Coerce Gateway Hairpins" is the first of these scenarios for which
> reasonably obvious use of existing protocol features doesn't seem to hold a
> solution. I have a suggestion though, which might have wider applicability,
> and now that we have GRUUs may be just the ticket. Define a new header
> field called "Dialog Affinity: <gruu>". The semantics would be for the
> proxy in the destination user's domain to attempt if possible to route the
> request to the same UA (in this case gateway) as is indicated by the
> corresponding Contact: address of the referenced GRUU. Finer granularity to
> the trunk level could be achieved if the subject gateway manufactured GRUUs
> which contained grid values identifying the appropriate trunk.
>

This is a little different from how GRUUs are to be used elsewhere.
Effectively the dialog affinity GRUU is more of a routing URI as
it is not the final distination.  The transfer target is URI (e.g.
dialed digits part of the URI initicating the transfer target).

So just to explore this a little further and make sure I understand
what you mean:
The gateway provides a dialog affinity GRUU for every call in or
out of the gateway.  Should the other end of the dialog decide to
do a transfer it copies the dialog affinity GRUU into the consultative
INVITE.  The URI cannot change from what is done today as the
URI contains the transfer target which the dialog affinity GRUU
could not have contained.  The proxy(s)/location server(s) do a kind
of loose route with the dialog affinity GRUU (or perhaps the
transferor does the loose route?)?  This avoids the normal routing
decisions made by the proxy(s)/location server(s) which might send
it to a different gateway.

I think that the proxy(s)/location servers need to be aware of the
dialog affinity GRUU as the location server(s) may still need to
perform transformations on the target URI (e.g. add/remove digit
prefixes, etc.).  I am not sure the loose route can work as it is not
clear when or which location server (where there are multiple that
transform the URI) should insert the loose route.

>
> 5. For "Consultative Turned Blind Gateway Glare" I don't see how we could
> ever fix that, given the inherent glare scenarios present on the PSTN due
> to the single TDM channel mapping property of calls. The entire SIP
> architecture exploits the ability to have parallel and overlapping dialogs
> to the same destination.

I think that you can fix it if the gateway disallows consultative transfer.  In
the blind transfer case, there are no overlapping dialogs unless someone
is play more than one of the three roles in transfer (e.g. transferee is the
transfer target).  In that case it is reasonable for the transfer to fail or to
land on a different target.  If there were multuple targets e.g. boss/admin
or user/voicemail, this is probably desirable anyway.

I think it may be possible to fix the consultative case if the gateway
requires that it stay pure consultative.  The gateway can probably not
put the tranfer on to the non-SIP side.  However it could handle the
overlapping dialogs for the consultation and target dialogs to the transfer
target and associate the two IP media streams to a single call leg out the
non-SIP side.

>
>
> Dave.
>
> ------------------------
> 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: Wed, 22 Oct 2003 15:53:53 -0700
Subject: Re: [Sipping] Describing dial tone and such using MIDI
From: Cullen Jennings <fluffy@cisco.com>
To: Eric Burger <eburger@snowshore.com>, sipping <sipping@ietf.org>
CC: David Bryan <dave@bryan.cx>
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On 10/21/03 3:21 AM, "Eric Burger" <eburger@snowshore.com> wrote:

> Looks good.  However, I question saying that the MIDI body is a cid:.  Why not
> just say it's a URI reference, which could be a cid:?

Oops - my mess up - that's what I meant to do.

> 
> By the way -- the country reference is (a) invaluable and (b) a lot of work.
> Excellent job!

Thanks - I  hoped that if nothing else came out  of this draft that would
provide useful in a machine passable form.

> 
>> -----Original Message-----
>> From: Cullen Jennings [mailto:fluffy@cisco.com]
>> Sent: Sat, October 18, 2003 12:15 PM
>> To: 'sipping'
>> Cc: Cullen Jennings; David Bryan
>> Subject: [Sipping] Describing dial tone and such using MIDI
>> 
>> 
>> 
>> David Bryan and I have submitted a draft on describing PSTN
>> tones such as
>> dialtone using MIDI - until it shows up in the drafts
>> directory, you can
>> find it at:
>> 
>> http://www.employees.org/~fluffy/ietf/draft-bryan-sipping-midi-00.txt
>> 
>> http://www.employees.org/~fluffy/ietf/draft-bryan-sipping-midi-00.html
>> 
>> Cullen
>> 
>> 
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
>> 
>> 
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
> 


_______________________________________________
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Date: Wed, 22 Oct 2003 15:55:18 -0700
Subject: Re: [Sipping] Describing dial tone and such using MIDI
From: Cullen Jennings <fluffy@cisco.com>
To: Eric Burger <eburger@snowshore.com>, sipping <sipping@ietf.org>
CC: David Bryan <dave@bryan.cx>
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On 10/21/03 3:21 AM, "Eric Burger" <eburger@snowshore.com> wrote:

> Looks good.  However, I question saying that the MIDI body is a cid:.  Why not
> just say it's a URI reference, which could be a cid:?

Oops - good point  - will fix.

> 
> By the way -- the country reference is (a) invaluable and (b) a lot of work.
> Excellent job!
>

Thank you.
 
>> -----Original Message-----
>> From: Cullen Jennings [mailto:fluffy@cisco.com]
>> Sent: Sat, October 18, 2003 12:15 PM
>> To: 'sipping'
>> Cc: Cullen Jennings; David Bryan
>> Subject: [Sipping] Describing dial tone and such using MIDI
>> 
>> 
>> 
>> David Bryan and I have submitted a draft on describing PSTN
>> tones such as
>> dialtone using MIDI - until it shows up in the drafts
>> directory, you can
>> find it at:
>> 
>> http://www.employees.org/~fluffy/ietf/draft-bryan-sipping-midi-00.txt
>> 
>> http://www.employees.org/~fluffy/ietf/draft-bryan-sipping-midi-00.html
>> 
>> Cullen
>> 
>> 
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
>> 
>> 
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Thu, 23 Oct 2003 08:06:11 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: [Sipping] SIP and AAA open questions and RADIUSEXT meeting
In-reply-to: <DADF50F5EC506B41A0F375ABEB320636D44AA3@esebe023.ntc.nokia.com>
To: john.loughney@nokia.com, dean.willis@softarmor.com, sipping@ietf.org
Cc: aboba@internaut.com
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It is true and regrettable that though practically all deployed and soon to
deploy mobile services use RADIUS, there is a lack of volunteers to write
the required documents for AAA-SIP.

This is however orthogonal to the merits of RADIUS vs. DIAMETER that has
been discussed to saturation on this list. Service providers have clearly
stated their desire for SIP-RADIUS to be properly documented. Maybe the
existing specs are just good enough - as attested by reported deployments -
so people are just using -sterman-aaa-sip-00.txt and keep working on their
the day jobs?

Thanks, Henry


-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf Of
john.loughney@nokia.com
Sent: Wednesday, October 22, 2003 2:25 AM
To: dean.willis@softarmor.com; sipping@ietf.org
Cc: aboba@internaut.com
Subject: RE: [Sipping] SIP and AAA open questions and RADIUSEXT meeting

Hi Dean,

Some very quick comments:

> Let's revisit where we are on the SIP-AAA work. Or at least where I THINK
we
> are, so please help me get this right.
> 
> 1) We have a requirements draft.
> http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-aaa-req-03.txt
> 
> 2) We have a RADIUS draft that is VERY expired, but variations are widely
implemented:
> http://www.softarmor.com/wgdb/docs/draft-sterman-aaa-sip-00.txt
> 
> 3) There's a DIAMETER draft:
> http://www.ietf.org/internet-drafts/draft-ietf-aaa-diameter-sip-app-00.txt

The Diameter work has some constituency (at least 3GPP) and is chartered
in the AAA.  Work is well underway.  We are happy to get input from the
SIP community on this draft.  However,  I think RADIUS extensions for
SIP is a different issue than this.

> 4) There's a RADIUS Accounting draft from Henning et al, which I seem to
> have misplaced the text of.
> 
> We seem to have a consensus to do at least some work in standardizing
RADIUS
> usage. It appears that the likeliest way forward is to conclude on the
> Sterman draft. We also seem to have an interest (if not consensus yet) in
> doing something to help with accounting for SIP operations using RADIUS.
> 
> Bernard Aboba, chairing the RADIUSEXT work, would like to know what if
> anything we would like RADIUSEXT to include on their agenda for the
November
> meeting. It looks like they MIGHT be meeting Friday morning (agendas do
> change).
> 
> I'm thinking:
> 
> 1) The Sterman-AAA draft:
> Any volunteers to present and lead the discussion for this in RADIUSEXT?
> 
> I believe we have a consensus to proceed with this draft in SIPPING of
> RADIUSEXT doesn't want to take the lead. I would propose that the work be
> jointly reviewed.
> 
> 2) The RADIUS Accounting draft:
> Any volunteers to present and lead the discussion for this in 
> RADIUSEXT? 

I think what is more important for the RADIUS work is to find some
folks who are willing to get the work done.  Wishing & hoping that
the work is done (by someone else) is not a good solution.

thanks,
John

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Thu, 23 Oct 2003 16:18:09 +0300
To: Henry Sinnreich <Henry.Sinnreich@mci.com>
Cc: john.loughney@nokia.com, dean.willis@softarmor.com, sipping@ietf.org,
        aboba@internaut.com
Subject: RE: [Sipping] SIP and AAA open questions and RADIUSEXT meeting
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Henry Sinnreich writes:

 > Maybe the
 > existing specs are just good enough - as attested by reported deployments -
 > so people are just using -sterman-aaa-sip-00.txt and keep working on their
 > the day jobs?

i believe that that is largely the case.  i rather write new code than
re-document existing one.

one note about radius accounting draft.  i don't remember if it included
accounting of forwarded calls (where, by whom and why).  if not, then
those should be added if the document ever gets progressed.

-- juha

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Baruch Sterman" <baruch@tekhelet.com>
To: <sipping@ietf.org>
Cc: <aboba@internaut.com>
Subject: RE: [Sipping] SIP and AAA open questions and RADIUSEXT meeting
Date: Thu, 23 Oct 2003 15:21:43 +0200
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I won't be able to make the meeting, but please let me know what I can
do to push this forward.

-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf
Of Henry Sinnreich
Sent: 23 October 2003 15:06
To: john.loughney@nokia.com; dean.willis@softarmor.com; sipping@ietf.org
Cc: aboba@internaut.com
Subject: RE: [Sipping] SIP and AAA open questions and RADIUSEXT meeting


It is true and regrettable that though practically all deployed and soon
to deploy mobile services use RADIUS, there is a lack of volunteers to
write the required documents for AAA-SIP.

This is however orthogonal to the merits of RADIUS vs. DIAMETER that has
been discussed to saturation on this list. Service providers have
clearly stated their desire for SIP-RADIUS to be properly documented.
Maybe the existing specs are just good enough - as attested by reported
deployments - so people are just using -sterman-aaa-sip-00.txt and keep
working on their the day jobs?

Thanks, Henry


-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf
Of john.loughney@nokia.com
Sent: Wednesday, October 22, 2003 2:25 AM
To: dean.willis@softarmor.com; sipping@ietf.org
Cc: aboba@internaut.com
Subject: RE: [Sipping] SIP and AAA open questions and RADIUSEXT meeting

Hi Dean,

Some very quick comments:

> Let's revisit where we are on the SIP-AAA work. Or at least where I 
> THINK
we
> are, so please help me get this right.
> 
> 1) We have a requirements draft. 
> http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-aaa-req-03.txt
> 
> 2) We have a RADIUS draft that is VERY expired, but variations are 
> widely
implemented:
> http://www.softarmor.com/wgdb/docs/draft-sterman-aaa-sip-00.txt
> 
> 3) There's a DIAMETER draft: 
> http://www.ietf.org/internet-drafts/draft-ietf-aaa-diameter-sip-app-00
> .txt

The Diameter work has some constituency (at least 3GPP) and is chartered
in the AAA.  Work is well underway.  We are happy to get input from the
SIP community on this draft.  However,  I think RADIUS extensions for
SIP is a different issue than this.

> 4) There's a RADIUS Accounting draft from Henning et al, which I seem 
> to have misplaced the text of.
> 
> We seem to have a consensus to do at least some work in standardizing
RADIUS
> usage. It appears that the likeliest way forward is to conclude on the

> Sterman draft. We also seem to have an interest (if not consensus yet)

> in doing something to help with accounting for SIP operations using 
> RADIUS.
> 
> Bernard Aboba, chairing the RADIUSEXT work, would like to know what if

> anything we would like RADIUSEXT to include on their agenda for the
November
> meeting. It looks like they MIGHT be meeting Friday morning (agendas 
> do change).
> 
> I'm thinking:
> 
> 1) The Sterman-AAA draft:
> Any volunteers to present and lead the discussion for this in 
> RADIUSEXT?
> 
> I believe we have a consensus to proceed with this draft in SIPPING of

> RADIUSEXT doesn't want to take the lead. I would propose that the work

> be jointly reviewed.
> 
> 2) The RADIUS Accounting draft:
> Any volunteers to present and lead the discussion for this in
> RADIUSEXT? 

I think what is more important for the RADIUS work is to find some folks
who are willing to get the work done.  Wishing & hoping that the work is
done (by someone else) is not a good solution.

thanks,
John

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip Use
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Use sip-implementors@cs.columbia.edu for questions on current sip Use
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This list is for NEW development of the application of SIP
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Subject: RE: [Sipping] SIP and AAA open questions and RADIUSEXT meeting
Date: Thu, 23 Oct 2003 16:30:08 +0300
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Thread-Topic: [Sipping] SIP and AAA open questions and RADIUSEXT meeting
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Hi Henry,

> It is true and regrettable that though practically all deployed and =
soon to
> deploy mobile services use RADIUS, there is a lack of volunteers to =
write
> the required documents for AAA-SIP.

If vendors or carriers feel this is an important, they should provide
resources to get it standardized.

> This is however orthogonal to the merits of RADIUS vs. DIAMETER that =
has
> been discussed to saturation on this list. Service providers have =
clearly
> stated their desire for SIP-RADIUS to be properly documented. Maybe =
the
> existing specs are just good enough - as attested by reported =
deployments -
> so people are just using -sterman-aaa-sip-00.txt and keep working on =
their
> the day jobs?

Then the IETF doesn't need to do anything, I guess.  If this is really=20
important to service providers, they'd demand their vendors provide
an interoperable spec.  If vendors were motivated, I am sure they
could allocate someone to get it done.  If noone wants to put any effort
to get it done, I sure don't want to waste my time.  I don't think that
the IETF has spare resources ...

John

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
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Date: Thu, 23 Oct 2003 08:45:44 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: [Sipping] SIP and AAA open questions and RADIUSEXT meeting
In-reply-to: <002001c39968$9cebe660$6a01a8c0@BARUCHLAPTOP>
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> I won't be able to make the meeting, 
>but please let me know what I can do to push this forward.

Baruch,

Your contribution would be really appreciated! Could you review your I-D and
resubmit with updates as required as version .01?

Incidentally, I see the new 802.x ISP settlement projects are based on
RADIUS and imply SIP as well.

The discussions on the list do not require your presence and those of us
interested in this work and present can vote in the next meeting. 

Will this work?

Thanks, Henry

Henry Sinnreich

MCI
400 International Parkway
Richardson, Texas 75081
USA

Have you disconnected the PBX?

-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf Of
Baruch Sterman
Sent: Thursday, October 23, 2003 8:22 AM
To: sipping@ietf.org
Cc: aboba@internaut.com
Subject: RE: [Sipping] SIP and AAA open questions and RADIUSEXT meeting

I won't be able to make the meeting, but please let me know what I can
do to push this forward.

-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf
Of Henry Sinnreich
Sent: 23 October 2003 15:06
To: john.loughney@nokia.com; dean.willis@softarmor.com; sipping@ietf.org
Cc: aboba@internaut.com
Subject: RE: [Sipping] SIP and AAA open questions and RADIUSEXT meeting


It is true and regrettable that though practically all deployed and soon
to deploy mobile services use RADIUS, there is a lack of volunteers to
write the required documents for AAA-SIP.

This is however orthogonal to the merits of RADIUS vs. DIAMETER that has
been discussed to saturation on this list. Service providers have
clearly stated their desire for SIP-RADIUS to be properly documented.
Maybe the existing specs are just good enough - as attested by reported
deployments - so people are just using -sterman-aaa-sip-00.txt and keep
working on their the day jobs?

Thanks, Henry


-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf
Of john.loughney@nokia.com
Sent: Wednesday, October 22, 2003 2:25 AM
To: dean.willis@softarmor.com; sipping@ietf.org
Cc: aboba@internaut.com
Subject: RE: [Sipping] SIP and AAA open questions and RADIUSEXT meeting

Hi Dean,

Some very quick comments:

> Let's revisit where we are on the SIP-AAA work. Or at least where I 
> THINK
we
> are, so please help me get this right.
> 
> 1) We have a requirements draft. 
> http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-aaa-req-03.txt
> 
> 2) We have a RADIUS draft that is VERY expired, but variations are 
> widely
implemented:
> http://www.softarmor.com/wgdb/docs/draft-sterman-aaa-sip-00.txt
> 
> 3) There's a DIAMETER draft: 
> http://www.ietf.org/internet-drafts/draft-ietf-aaa-diameter-sip-app-00
> .txt

The Diameter work has some constituency (at least 3GPP) and is chartered
in the AAA.  Work is well underway.  We are happy to get input from the
SIP community on this draft.  However,  I think RADIUS extensions for
SIP is a different issue than this.

> 4) There's a RADIUS Accounting draft from Henning et al, which I seem 
> to have misplaced the text of.
> 
> We seem to have a consensus to do at least some work in standardizing
RADIUS
> usage. It appears that the likeliest way forward is to conclude on the

> Sterman draft. We also seem to have an interest (if not consensus yet)

> in doing something to help with accounting for SIP operations using 
> RADIUS.
> 
> Bernard Aboba, chairing the RADIUSEXT work, would like to know what if

> anything we would like RADIUSEXT to include on their agenda for the
November
> meeting. It looks like they MIGHT be meeting Friday morning (agendas 
> do change).
> 
> I'm thinking:
> 
> 1) The Sterman-AAA draft:
> Any volunteers to present and lead the discussion for this in 
> RADIUSEXT?
> 
> I believe we have a consensus to proceed with this draft in SIPPING of

> RADIUSEXT doesn't want to take the lead. I would propose that the work

> be jointly reviewed.
> 
> 2) The RADIUS Accounting draft:
> Any volunteers to present and lead the discussion for this in
> RADIUSEXT? 

I think what is more important for the RADIUS work is to find some folks
who are willing to get the work done.  Wishing & hoping that the work is
done (by someone else) is not a good solution.

thanks,
John

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip Use
sip@ietf.org for new developments of core SIP


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip Use
sip@ietf.org for new developments of core SIP


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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--NextPart

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

	Title		: Requirements for End-to-middle Security for the Session Initiation Protocol (SIP)
	Author(s)	: K. Ono, S. Tachimoto
	Filename	: draft-ietf-sipping-e2m-sec-reqs-00.txt
	Pages		: 15
	Date		: 2003-10-23
	
A SIP UA does not always trust all proxy servers in a request path to
decide whether to inspect the message bodies and/or headers contained
in a message. The UA might want to protect the message bodies and/or
headers from proxy servers excluding the particular proxy that
provides some features based on reading them.  This situation
requires a mechanism for securing information passed between the UA
and an intermediary proxy, also called

A URL for this Internet-Draft is:
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_______________________________________________
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This list is for NEW development of the application of SIP
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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 Proposal Investigation Working Group of the IETF.

	Title		: A Framework for Application Interaction in the Session Initiation Protocol (SIP)
	Author(s)	: J. Rosenberg
	Filename	: draft-ietf-sipping-app-interaction-framework-00.txt
	Pages		: 30
	Date		: 2003-10-22
	
This document describes a framework and requirements for the
interaction between users and Session Initiation Protocol (SIP) based
applications. By interacting with applications, users can guide the
way in which they operate. The focus of this framework is stimulus
signaling, which allows a user agent to interact with an application
without knowledge of the semantics of that application. Stimulus
signaling can occur to a user interface running locally with the
client, or to a remote user interface, through media streams.
Stimulus signaling encompasses a wide range of mechanisms, ranging
from clicking on hyperlinks, to pressing buttons, to traditional Dual
Tone Multi Frequency (DTMF) input. In all cases, stimulus signaling
is supported through the use of markup languages, which play a key
role in this framework.

A URL for this Internet-Draft is:
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_______________________________________________
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This list is for NEW development of the application of SIP
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A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Framework of requirements for real-time text 
			  conversation using SIP
	Author(s)	: G. Hellstrom, R. Roy
	Filename	: draft-manyfolks-sipping-toip-00.txt
	Pages		: 33
	Date		: 2003-10-23
	
This document provides the framework of requirements for text conversation with real time character-by-character interactive flow 
over the IP network using the Session Initiation Protocol. The
requirements for general real-time text-over-IP telephony, point-to-
point and conference calls, transcoding, relay services, user
mobility, interworking between text-over-IP telephony and existing
text-telephony, and some special features including instant messaging
have been described.

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_______________________________________________
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From exim@www1.ietf.org  Thu Oct 23 11:38:38 2003
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From: "Liess, Laura" <Laura.Liess@t-systems.com>
To: john.loughney@nokia.com, aboba@internaut.com, dean.willis@softarmor.com,
        baruch@tekhelet.com, Henry.Sinnreich@mci.com
Cc: sipping@ietf.org
Subject: AW: [Sipping] SIP and AAA open questions and RADIUSEXT meeting
Date: Thu, 23 Oct 2003 17:36:49 +0200
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John,

	>If vendors or carriers feel this is an important, they should provide
	>resources to get it standardized.

It's my opinion, too. We feel this is important for carriers, so we will provide the ressources to get it standardized. We will also attend the meeting. (We currently had a side e-mail discussion with Dean on this issue.)

We just have to clarify what is to be done. 

Laura 
Deutsche Telekom/T-Systems  



>


>> This is however orthogonal to the merits of RADIUS vs. 
>DIAMETER that has
>> been discussed to saturation on this list. Service providers 
>have clearly
>> stated their desire for SIP-RADIUS to be properly 
>documented. Maybe the
>> existing specs are just good enough - as attested by 
>reported deployments -
>> so people are just using -sterman-aaa-sip-00.txt and keep 
>working on their
>> the day jobs?
>
>Then the IETF doesn't need to do anything, I guess.  If this is really 
>important to service providers, they'd demand their vendors provide
>an interoperable spec.  If vendors were motivated, I am sure they
>could allocate someone to get it done.  If noone wants to put 
>any effort
>to get it done, I sure don't want to waste my time.  I don't think that
>the IETF has spare resources ...
>
>John
>
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP
>

_______________________________________________
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Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Thu Oct 23 13:36:39 2003
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To: "'Jonathan ROSENBERG (jdrosen@dynamicsoft.com)'"
	 <jdrosen@dynamicsoft.com>,
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	 <Gonzalo.Camarillo@lmf.ericsson.se>,
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Subject: [Sipping] draft-rosenberg-sipping-ice-01 & draft-camarillo-mmusic-alt-02.tx
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Jonathan,

Is there any plan to update the ICE draft with the changes made by Gonzallo
and 
yourself to the draft-camarill-mmusic-alt-02?

In particular:

1 - In ICE, do you intend to keep the address priority based on the qvalue
(similar 
    to alt-01), or do you intend to go with the simple prioritized list
approach 
    as per alt-02? In both cases, I'm assuming you are keeping the alt
parameter.

2 - Is the intent that IPv6 addresses in SDP be expressed in ICE using the
IPV 
    format, or does it follow the IPv4 format as per the current example in
ICE? 
    Section 5.3 of ICE talks briefly about IPv6 addresses, but there is no
example 
    on how it would be expressed. I believe that the intent is to use the
IPV format
    in ICE for IPv6 addresses. Is this accurate?

3 - References: ICE should refer to alt-2 not alt-1 in the reference.

4 - Is the intent that STUN requests in ICE be receivable only on IPv4
addresses?
    This would assume that IPv6 addresses are always reachable. I believe
this is
    the intent.

In other words, to take the example A's offer in ICE section 9.1.1 (Figure
5), 
if there was also an IPv6 address, which of the following SDP (A, B or C)
would be 
correct:


A: Using IPV, but keeping the qvalue for IPv4

   v=0
   o=alice 2890844730 2890844731 IN IP4 host.example.com
   s=
   t=0 0
   a=group:IPV 1 2
   m=audio 6886 RTP/AVP 0
   c=IN IP6 2201:056D::112E:144A:1E24
   a=mid:1
   m=audio 8076 RTP/AVP 0
   c=IN IP4 192.0.2.10
   a=alt:1 1.0 : user 9kksj== 10.0.1.1 1010
   a=alt:2 0.8 : user1 9kksk== 192.0.2.1 9988 192.0.2.1 9990
   a=alt:3 0.4 : user2 9kksl== 192.0.2.10 8076
   a=mid:2

B: Using IPV, using simple ordered list instead of qvalue for IPv4

   v=0
   o=alice 2890844730 2890844731 IN IP4 host.example.com
   s=
   t=0 0
   a=group:IPV 1 2
   m=audio 6886 RTP/AVP 0
   c=IN IP6 2201:056D::112E:144A:1E24
   a=mid:1
   m=audio 8076 RTP/AVP 0
   c=IN IP4 192.0.2.10
   a=alt:1 user 9kksj== 10.0.1.1 1010
   a=alt:2 user1 9kksk== 192.0.2.1 9988 192.0.2.1 9990
   a=alt:3 user2 9kksl== 192.0.2.10 8076
   a=mid:2

C: As per current ICE draft. This implies that we would be running a STUN
server 
   on the IPv6 address as well to check for connectivity.

   v=0
   o=alice 2890844730 2890844731 IN IP4 host.example.com
   s=
   c=IN IP4 192.0.2.10
   t=0 0
   m=audio 8076 RTP/AVP 0
   a=alt:1 1.0 : user 9kksi = IP6 2201:056D::112E:144A:1E24  (note that I've
added 
   a=alt:1 0.9 : user0 9kksj== IP 4 10.0.1.1 1010             an IP6 and
IP4)
   a=alt:2 0.8 : user1 9kksk== IP4 192.0.2.1 9988 192.0.2.1 9990
   a=alt:3 0.4 : user2 9kksl== IP4 192.0.2.10 8076

I think the intent is either A or B. Can you clarify?

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<P><FONT SIZE=2>Jonathan,</FONT>
</P>

<P><FONT SIZE=2>Is there any plan to update the ICE draft with the changes made by Gonzallo and </FONT>
<BR><FONT SIZE=2>yourself to the draft-camarill-mmusic-alt-02?</FONT>
</P>

<P><FONT SIZE=2>In particular:</FONT>
</P>

<P><FONT SIZE=2>1 - In ICE, do you intend to keep the address priority based on the qvalue (similar </FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp;&nbsp; to alt-01), or do you intend to go with the simple prioritized list approach </FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp;&nbsp; as per alt-02? In both cases, I'm assuming you are keeping the alt parameter.</FONT>
</P>

<P><FONT SIZE=2>2 - Is the intent that IPv6 addresses in SDP be expressed in ICE using the IPV </FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp;&nbsp; format, or does it follow the IPv4 format as per the current example in ICE? </FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp;&nbsp; Section 5.3 of ICE talks briefly about IPv6 addresses, but there is no example </FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp;&nbsp; on how it would be expressed. I believe that the intent is to use the IPV format</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp;&nbsp; in ICE for IPv6 addresses. Is this accurate?</FONT>
</P>

<P><FONT SIZE=2>3 - References: ICE should refer to alt-2 not alt-1 in the reference.</FONT>
</P>

<P><FONT SIZE=2>4 - Is the intent that STUN requests in ICE be receivable only on IPv4 addresses?</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp;&nbsp; This would assume that IPv6 addresses are always reachable. I believe this is</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp;&nbsp; the intent.</FONT>
</P>

<P><FONT SIZE=2>In other words, to take the example A's offer in ICE section 9.1.1 (Figure 5), </FONT>
<BR><FONT SIZE=2>if there was also an IPv6 address, which of the following SDP (A, B or C) would be </FONT>
<BR><FONT SIZE=2>correct:</FONT>
</P>
<BR>

<P><FONT SIZE=2>A: Using IPV, but keeping the qvalue for IPv4</FONT>
</P>

<P><FONT SIZE=2>&nbsp;&nbsp; v=0</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; o=alice 2890844730 2890844731 IN IP4 host.example.com</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; s=</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; t=0 0</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; a=group:IPV 1 2</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; m=audio 6886 RTP/AVP 0</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; c=IN IP6 2201:056D::112E:144A:1E24</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; a=mid:1</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; m=audio 8076 RTP/AVP 0</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; c=IN IP4 192.0.2.10</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; a=alt:1 1.0 : user 9kksj== 10.0.1.1 1010</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; a=alt:2 0.8 : user1 9kksk== 192.0.2.1 9988 192.0.2.1 9990</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; a=alt:3 0.4 : user2 9kksl== 192.0.2.10 8076</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; a=mid:2</FONT>
</P>

<P><FONT SIZE=2>B: Using IPV, using simple ordered list instead of qvalue for IPv4</FONT>
</P>

<P><FONT SIZE=2>&nbsp;&nbsp; v=0</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; o=alice 2890844730 2890844731 IN IP4 host.example.com</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; s=</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; t=0 0</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; a=group:IPV 1 2</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; m=audio 6886 RTP/AVP 0</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; c=IN IP6 2201:056D::112E:144A:1E24</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; a=mid:1</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; m=audio 8076 RTP/AVP 0</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; c=IN IP4 192.0.2.10</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; a=alt:1 user 9kksj== 10.0.1.1 1010</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; a=alt:2 user1 9kksk== 192.0.2.1 9988 192.0.2.1 9990</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; a=alt:3 user2 9kksl== 192.0.2.10 8076</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; a=mid:2</FONT>
</P>

<P><FONT SIZE=2>C: As per current ICE draft. This implies that we would be running a STUN server </FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; on the IPv6 address as well to check for connectivity.</FONT>
</P>

<P><FONT SIZE=2>&nbsp;&nbsp; v=0</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; o=alice 2890844730 2890844731 IN IP4 host.example.com</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; s=</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; c=IN IP4 192.0.2.10</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; t=0 0</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; m=audio 8076 RTP/AVP 0</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; a=alt:1 1.0 : user 9kksi = IP6 2201:056D::112E:144A:1E24&nbsp; (note that I've added </FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; a=alt:1 0.9 : user0 9kksj== IP 4 10.0.1.1 1010&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; an IP6 and IP4)</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; a=alt:2 0.8 : user1 9kksk== IP4 192.0.2.1 9988 192.0.2.1 9990</FONT>
<BR><FONT SIZE=2>&nbsp;&nbsp; a=alt:3 0.4 : user2 9kksl== IP4 192.0.2.10 8076</FONT>
</P>

<P><FONT SIZE=2>I think the intent is either A or B. Can you clarify?</FONT>
</P>

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Dean asked me to forward this on to the various lists:

Softarmor.com is down due to a hardware failure. Dean is rushing to fix 
the problem as I write this. This will affect HTTP and email.




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OK, we seem to be back up. 

Apparently, power supplies have an MTBF of exactly 1 day longer than
their warranty, with a very narrow standard deviation around that mean.

The new one has 3 fans -- I don't know if that means it will last three
times as long, or is three times more likely to fail.

--
Dean



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Subject: RE: [Sipping] SIP and AAA open questions and RADIUSEXT meeting
From: Dean Willis <dean.willis@softarmor.com>
To: Henry Sinnreich <Henry.Sinnreich@mci.com>
Cc: "'Baruch Sterman'" <baruch@tekhelet.com>, aboba@internaut.com,
        sipping@ietf.org
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On Thu, 2003-10-23 at 08:45, Henry Sinnreich wrote:
> > I won't be able to make the meeting, 
> >but please let me know what I can do to push this forward.
> 
> Baruch,
> 
> Your contribution would be really appreciated! Could you review your I-D and
> resubmit with updates as required as version .01?
> 
> Incidentally, I see the new 802.x ISP settlement projects are based on
> RADIUS and imply SIP as well.
> 
> The discussions on the list do not require your presence and those of us
> interested in this work and present can vote in the next meeting. 
> 
> Will this work?

It's a good suggestion, but 

1) We've missed the cutoff date for IETF 58, and

2) We haven't given Baruch any feedback about updating the draft, at
least not any recently.

3) We don't really vote in meetings anyhow -- consensus can readily be
determined "on the list".

I propose to chat privately with Baruch and others and come back with a
proposal for moving forward.

--
Dean




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Francois,

> 1 - In ICE, do you intend to keep the address priority based on the 
> qvalue (similar
>     to alt-01), or do you intend to go with the simple prioritized list 
> approach
>     as per alt-02? In both cases, I'm assuming you are keeping the alt 
> parameter.

Within an m line, we intend to use the alt lines defined in ICE using 
qvalues.

> 2 - Is the intent that IPv6 addresses in SDP be expressed in ICE using 
> the IPV
>     format, or does it follow the IPv4 format as per the current example 
> in ICE?

The IPv6 addresses will use the IPV format.

>     Section 5.3 of ICE talks briefly about IPv6 addresses, but there is 
> no example
>     on how it would be expressed. I believe that the intent is to use 
> the IPV format
>     in ICE for IPv6 addresses. Is this accurate?

Yes.

> 3 - References: ICE should refer to alt-2 not alt-1 in the reference.
> 
> 4 - Is the intent that STUN requests in ICE be receivable only on IPv4 
> addresses?
>     This would assume that IPv6 addresses are always reachable. I 
> believe this is
>     the intent.

If you added alt lines to an IPv6 line, you could run STUN on it... the 
idea is presenting user agents with an IPv6 and an IPv4 address for a 
stream. Once the UA picks up a media line (corresponding to an address), 
it does whatever that media line tells it to do (e.g, running STUN).

> In other words, to take the example A's offer in ICE section 9.1.1 
> (Figure 5),
> if there was also an IPv6 address, which of the following SDP (A, B or 
> C) would be
> correct:

Option A would be correct (use IPV and alt lines with qvalues). We 
actually have a draft that contains a similar example (I believe 
Jonathan submitted before the cut-off...). This is an SDP taken from 
that draft:

     v=0
     o=bob 280744730 28977631 IN IP6 host.example.com
     s=
     t=0 0
     a=group:IPV 1 2
     m=audio 6886 RTP/AVP 0
     c=IN IP6 2001:056D::112E:144A:1E24
     a=mid:1
     m=audio 22334 RTP/AVP 0
     c=IN IP4 192.0.0.1
     a=mid:2
     a=alt:1 1.0 : user 9kksj== 192.0.0.1 22334

Thanks,

Gonzalo



> A: Using IPV, but keeping the qvalue for IPv4
> 
>    v=0
>    o=alice 2890844730 2890844731 IN IP4 host.example.com
>    s=
>    t=0 0
>    a=group:IPV 1 2
>    m=audio 6886 RTP/AVP 0
>    c=IN IP6 2201:056D::112E:144A:1E24
>    a=mid:1
>    m=audio 8076 RTP/AVP 0
>    c=IN IP4 192.0.2.10
>    a=alt:1 1.0 : user 9kksj== 10.0.1.1 1010
>    a=alt:2 0.8 : user1 9kksk== 192.0.2.1 9988 192.0.2.1 9990
>    a=alt:3 0.4 : user2 9kksl== 192.0.2.10 8076
>    a=mid:2
> 
> B: Using IPV, using simple ordered list instead of qvalue for IPv4
> 
>    v=0
>    o=alice 2890844730 2890844731 IN IP4 host.example.com
>    s=
>    t=0 0
>    a=group:IPV 1 2
>    m=audio 6886 RTP/AVP 0
>    c=IN IP6 2201:056D::112E:144A:1E24
>    a=mid:1
>    m=audio 8076 RTP/AVP 0
>    c=IN IP4 192.0.2.10
>    a=alt:1 user 9kksj== 10.0.1.1 1010
>    a=alt:2 user1 9kksk== 192.0.2.1 9988 192.0.2.1 9990
>    a=alt:3 user2 9kksl== 192.0.2.10 8076
>    a=mid:2
> 
> C: As per current ICE draft. This implies that we would be running a 
> STUN server
>    on the IPv6 address as well to check for connectivity.
> 
>    v=0
>    o=alice 2890844730 2890844731 IN IP4 host.example.com
>    s=
>    c=IN IP4 192.0.2.10
>    t=0 0
>    m=audio 8076 RTP/AVP 0
>    a=alt:1 1.0 : user 9kksi = IP6 2201:056D::112E:144A:1E24  (note that 
> I've added
>    a=alt:1 0.9 : user0 9kksj== IP 4 10.0.1.1 1010             an IP6 and 
> IP4)
>    a=alt:2 0.8 : user1 9kksk== IP4 192.0.2.1 9988 192.0.2.1 9990
>    a=alt:3 0.4 : user2 9kksl== IP4 192.0.2.10 8076
> 
> I think the intent is either A or B. Can you clarify?
> 


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From: Bernard Aboba <aboba@internaut.com>
To: Baruch Sterman <baruch@tekhelet.com>
cc: sipping@ietf.org
Subject: RE: [Sipping] SIP and AAA open questions and RADIUSEXT meeting
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> I won't be able to make the meeting, but please let me know what I can
> do to push this forward.

The most obvious things would be:

a. To volunteer to edit the draft (in SIPPING or RADEXT)
b. To find someone to present the draft (at this and other meetings)

_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Fri Oct 24 06:50:39 2003
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From: Bernard Aboba <aboba@internaut.com>
To: john.loughney@nokia.com
cc: sipping@ietf.org
Subject: RE: [Sipping] SIP and AAA open questions and RADIUSEXT meeting
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> Then the IETF doesn't need to do anything, I guess.  If this is really
> important to service providers, they'd demand their vendors provide
> an interoperable spec.  If vendors were motivated, I am sure they
> could allocate someone to get it done.  If noone wants to put any effort
> to get it done, I sure don't want to waste my time.  I don't think that
> the IETF has spare resources ...

Agreed.  If individuals can't be identified who are qualified and
committed to the work, there's no point in discussing whether it should be
a WG work item or not.  Without a champion, drafts tend to become a "ward
of the state" and personally, I've got all the "draft orphans" I can
handle.

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From: Bernard Aboba <aboba@internaut.com>
To: Dean Willis <dean.willis@softarmor.com>
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> 4) There's a RADIUS Accounting draft from Henning et al, which I seem to
> have misplaced the text of.

I believe it is:
http://www.watersprings.org/pub/id/draft-schulzrinne-sipping-radius-accounting-00.txt

> I'm thinking:
>
> 1) The Sterman-AAA draft:
> Any volunteers to present and lead the discussion for this in RADIUSEXT?
>
> 2) The RADIUS Accounting draft:
> Any volunteers to present and lead the discussion for this in RADIUSEXT?

Volunteers should send me email so I can put you on the agenda.


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From exim@www1.ietf.org  Fri Oct 24 10:05:13 2003
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From: "Beck01, Wolfgang" <BeckW@t-systems.com>
To: aboba@internaut.com, dean.willis@softarmor.com,
        "Liess, Laura" <Laura.Liess@t-systems.com>, baruch@tekhelet.com
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Bernard,

You wrote:
> I'm thinking:
>
> 1) The Sterman-AAA draft:
> Any volunteers to present and lead the discussion for this in RADIUSEXT?
I can do it. 

> 2) The RADIUS Accounting draft:
> Any volunteers to present and lead the discussion for this in RADIUSEXT?
If Henning is not available, I volunteer for this task, too.


Wolfgang Beck

Deutsche Telekom/T-Systems

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From: "Francois Audet" <audet@nortelnetworks.com>
To: "'Gonzalo Camarillo'" <Gonzalo.Camarillo@ericsson.com>
Cc: "'Jonathan ROSENBERG (jdrosen@dynamicsoft.com)'"
	 <jdrosen@dynamicsoft.com>,
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 2.tx t
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Thanks. That's what I tought. Excellent.

Minor comment is that if we do allow for STUN on IPv6 addresses, we should
probably have an "IP6" or "IP4" indicator in the alt line to indicate
the IP address type (like a c line).

> -----Original Message-----
> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com] 
> Sent: Thursday, October 23, 2003 22:10
> To: Audet, Francois [SC100:4K02:EXCH]
> Cc: 'Jonathan ROSENBERG (jdrosen@dynamicsoft.com)'; 'Gonzalo 
> Camarillo'; 'sipping@ietf.org'
> Subject: Re: draft-rosenberg-sipping-ice-01 & 
> draft-camarillo-mmusic-alt-02.tx t
> 
> 
> Francois,
> 
> > 1 - In ICE, do you intend to keep the address priority based on the
> > qvalue (similar
> >     to alt-01), or do you intend to go with the simple 
> prioritized list 
> > approach
> >     as per alt-02? In both cases, I'm assuming you are 
> keeping the alt 
> > parameter.
> 
> Within an m line, we intend to use the alt lines defined in ICE using 
> qvalues.
> 
> > 2 - Is the intent that IPv6 addresses in SDP be expressed 
> in ICE using
> > the IPV
> >     format, or does it follow the IPv4 format as per the 
> current example 
> > in ICE?
> 
> The IPv6 addresses will use the IPV format.
> 
> >     Section 5.3 of ICE talks briefly about IPv6 addresses, 
> but there 
> > is
> > no example
> >     on how it would be expressed. I believe that the intent 
> is to use 
> > the IPV format
> >     in ICE for IPv6 addresses. Is this accurate?
> 
> Yes.
> 
> > 3 - References: ICE should refer to alt-2 not alt-1 in the 
> reference.
> > 
> > 4 - Is the intent that STUN requests in ICE be receivable 
> only on IPv4
> > addresses?
> >     This would assume that IPv6 addresses are always reachable. I 
> > believe this is
> >     the intent.
> 
> If you added alt lines to an IPv6 line, you could run STUN on 
> it... the 
> idea is presenting user agents with an IPv6 and an IPv4 address for a 
> stream. Once the UA picks up a media line (corresponding to 
> an address), 
> it does whatever that media line tells it to do (e.g, running STUN).
> 
> > In other words, to take the example A's offer in ICE section 9.1.1
> > (Figure 5),
> > if there was also an IPv6 address, which of the following 
> SDP (A, B or 
> > C) would be
> > correct:
> 
> Option A would be correct (use IPV and alt lines with qvalues). We 
> actually have a draft that contains a similar example (I believe 
> Jonathan submitted before the cut-off...). This is an SDP taken from 
> that draft:
> 
>      v=0
>      o=bob 280744730 28977631 IN IP6 host.example.com
>      s=
>      t=0 0
>      a=group:IPV 1 2
>      m=audio 6886 RTP/AVP 0
>      c=IN IP6 2001:056D::112E:144A:1E24
>      a=mid:1
>      m=audio 22334 RTP/AVP 0
>      c=IN IP4 192.0.0.1
>      a=mid:2
>      a=alt:1 1.0 : user 9kksj== 192.0.0.1 22334
> 
> Thanks,
> 
> Gonzalo
> 
> 
> 
> > A: Using IPV, but keeping the qvalue for IPv4
> > 
> >    v=0
> >    o=alice 2890844730 2890844731 IN IP4 host.example.com
> >    s=
> >    t=0 0
> >    a=group:IPV 1 2
> >    m=audio 6886 RTP/AVP 0
> >    c=IN IP6 2201:056D::112E:144A:1E24
> >    a=mid:1
> >    m=audio 8076 RTP/AVP 0
> >    c=IN IP4 192.0.2.10
> >    a=alt:1 1.0 : user 9kksj== 10.0.1.1 1010
> >    a=alt:2 0.8 : user1 9kksk== 192.0.2.1 9988 192.0.2.1 9990
> >    a=alt:3 0.4 : user2 9kksl== 192.0.2.10 8076
> >    a=mid:2
> > 
> > B: Using IPV, using simple ordered list instead of qvalue for IPv4
> > 
> >    v=0
> >    o=alice 2890844730 2890844731 IN IP4 host.example.com
> >    s=
> >    t=0 0
> >    a=group:IPV 1 2
> >    m=audio 6886 RTP/AVP 0
> >    c=IN IP6 2201:056D::112E:144A:1E24
> >    a=mid:1
> >    m=audio 8076 RTP/AVP 0
> >    c=IN IP4 192.0.2.10
> >    a=alt:1 user 9kksj== 10.0.1.1 1010
> >    a=alt:2 user1 9kksk== 192.0.2.1 9988 192.0.2.1 9990
> >    a=alt:3 user2 9kksl== 192.0.2.10 8076
> >    a=mid:2
> > 
> > C: As per current ICE draft. This implies that we would be running a
> > STUN server
> >    on the IPv6 address as well to check for connectivity.
> > 
> >    v=0
> >    o=alice 2890844730 2890844731 IN IP4 host.example.com
> >    s=
> >    c=IN IP4 192.0.2.10
> >    t=0 0
> >    m=audio 8076 RTP/AVP 0
> >    a=alt:1 1.0 : user 9kksi = IP6 2201:056D::112E:144A:1E24  (note 
> > that
> > I've added
> >    a=alt:1 0.9 : user0 9kksj== IP 4 10.0.1.1 1010           
>   an IP6 and 
> > IP4)
> >    a=alt:2 0.8 : user1 9kksk== IP4 192.0.2.1 9988 192.0.2.1 9990
> >    a=alt:3 0.4 : user2 9kksl== IP4 192.0.2.10 8076
> > 
> > I think the intent is either A or B. Can you clarify?
> > 
> 
> 

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<TITLE>RE: draft-rosenberg-sipping-ice-01 &amp; =
draft-camarillo-mmusic-alt-02.tx t</TITLE>
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<BODY>

<P><FONT SIZE=3D2>Thanks. That's what I tought. Excellent.</FONT>
</P>

<P><FONT SIZE=3D2>Minor comment is that if we do allow for STUN on IPv6 =
addresses, we should</FONT>
<BR><FONT SIZE=3D2>probably have an &quot;IP6&quot; or &quot;IP4&quot; =
indicator in the alt line to indicate</FONT>
<BR><FONT SIZE=3D2>the IP address type (like a c line).</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Gonzalo Camarillo [<A =
HREF=3D"mailto:Gonzalo.Camarillo@ericsson.com">mailto:Gonzalo.Camarillo@=
ericsson.com</A>] </FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Thursday, October 23, 2003 22:10</FONT>
<BR><FONT SIZE=3D2>&gt; To: Audet, Francois [SC100:4K02:EXCH]</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: 'Jonathan ROSENBERG =
(jdrosen@dynamicsoft.com)'; 'Gonzalo </FONT>
<BR><FONT SIZE=3D2>&gt; Camarillo'; 'sipping@ietf.org'</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: Re: draft-rosenberg-sipping-ice-01 =
&amp; </FONT>
<BR><FONT SIZE=3D2>&gt; draft-camarillo-mmusic-alt-02.tx t</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Francois,</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; 1 - In ICE, do you intend to keep the =
address priority based on the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; qvalue (similar</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; to alt-01), or do =
you intend to go with the simple </FONT>
<BR><FONT SIZE=3D2>&gt; prioritized list </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; approach</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; as per alt-02? In =
both cases, I'm assuming you are </FONT>
<BR><FONT SIZE=3D2>&gt; keeping the alt </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; parameter.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Within an m line, we intend to use the alt =
lines defined in ICE using </FONT>
<BR><FONT SIZE=3D2>&gt; qvalues.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; 2 - Is the intent that IPv6 addresses in =
SDP be expressed </FONT>
<BR><FONT SIZE=3D2>&gt; in ICE using</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; the IPV</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; format, or does it =
follow the IPv4 format as per the </FONT>
<BR><FONT SIZE=3D2>&gt; current example </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; in ICE?</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; The IPv6 addresses will use the IPV =
format.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; Section 5.3 of ICE =
talks briefly about IPv6 addresses, </FONT>
<BR><FONT SIZE=3D2>&gt; but there </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; is</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; no example</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; on how it would be =
expressed. I believe that the intent </FONT>
<BR><FONT SIZE=3D2>&gt; is to use </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; the IPV format</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; in ICE for IPv6 =
addresses. Is this accurate?</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Yes.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; 3 - References: ICE should refer to alt-2 =
not alt-1 in the </FONT>
<BR><FONT SIZE=3D2>&gt; reference.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; 4 - Is the intent that STUN requests in =
ICE be receivable </FONT>
<BR><FONT SIZE=3D2>&gt; only on IPv4</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; addresses?</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; This would assume =
that IPv6 addresses are always reachable. I </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; believe this is</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; the intent.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; If you added alt lines to an IPv6 line, you =
could run STUN on </FONT>
<BR><FONT SIZE=3D2>&gt; it... the </FONT>
<BR><FONT SIZE=3D2>&gt; idea is presenting user agents with an IPv6 and =
an IPv4 address for a </FONT>
<BR><FONT SIZE=3D2>&gt; stream. Once the UA picks up a media line =
(corresponding to </FONT>
<BR><FONT SIZE=3D2>&gt; an address), </FONT>
<BR><FONT SIZE=3D2>&gt; it does whatever that media line tells it to do =
(e.g, running STUN).</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; In other words, to take the example A's =
offer in ICE section 9.1.1</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; (Figure 5),</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; if there was also an IPv6 address, which =
of the following </FONT>
<BR><FONT SIZE=3D2>&gt; SDP (A, B or </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; C) would be</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; correct:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Option A would be correct (use IPV and alt =
lines with qvalues). We </FONT>
<BR><FONT SIZE=3D2>&gt; actually have a draft that contains a similar =
example (I believe </FONT>
<BR><FONT SIZE=3D2>&gt; Jonathan submitted before the cut-off...). This =
is an SDP taken from </FONT>
<BR><FONT SIZE=3D2>&gt; that draft:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; v=3D0</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; o=3Dbob 280744730 =
28977631 IN IP6 host.example.com</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; s=3D</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; t=3D0 0</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Dgroup:IPV 1 =
2</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; m=3Daudio 6886 =
RTP/AVP 0</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; c=3DIN IP6 =
2001:056D::112E:144A:1E24</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Dmid:1</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; m=3Daudio 22334 =
RTP/AVP 0</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; c=3DIN IP4 =
192.0.0.1</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Dmid:2</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Dalt:1 1.0 : =
user 9kksj=3D=3D 192.0.0.1 22334</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Thanks,</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Gonzalo</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; A: Using IPV, but keeping the qvalue for =
IPv4</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; v=3D0</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; o=3Dalice 2890844730 =
2890844731 IN IP4 host.example.com</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; s=3D</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; t=3D0 0</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; a=3Dgroup:IPV 1 2</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; m=3Daudio 6886 RTP/AVP =
0</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; c=3DIN IP6 =
2201:056D::112E:144A:1E24</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; a=3Dmid:1</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; m=3Daudio 8076 RTP/AVP =
0</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; c=3DIN IP4 =
192.0.2.10</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; a=3Dalt:1 1.0 : user =
9kksj=3D=3D 10.0.1.1 1010</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; a=3Dalt:2 0.8 : user1 =
9kksk=3D=3D 192.0.2.1 9988 192.0.2.1 9990</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; a=3Dalt:3 0.4 : user2 =
9kksl=3D=3D 192.0.2.10 8076</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; a=3Dmid:2</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; B: Using IPV, using simple ordered list =
instead of qvalue for IPv4</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; v=3D0</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; o=3Dalice 2890844730 =
2890844731 IN IP4 host.example.com</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; s=3D</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; t=3D0 0</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; a=3Dgroup:IPV 1 2</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; m=3Daudio 6886 RTP/AVP =
0</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; c=3DIN IP6 =
2201:056D::112E:144A:1E24</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; a=3Dmid:1</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; m=3Daudio 8076 RTP/AVP =
0</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; c=3DIN IP4 =
192.0.2.10</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; a=3Dalt:1 user =
9kksj=3D=3D 10.0.1.1 1010</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; a=3Dalt:2 user1 =
9kksk=3D=3D 192.0.2.1 9988 192.0.2.1 9990</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; a=3Dalt:3 user2 =
9kksl=3D=3D 192.0.2.10 8076</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; a=3Dmid:2</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; C: As per current ICE draft. This implies =
that we would be running a</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; STUN server</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; on the IPv6 address as =
well to check for connectivity.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; v=3D0</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; o=3Dalice 2890844730 =
2890844731 IN IP4 host.example.com</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; s=3D</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; c=3DIN IP4 =
192.0.2.10</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; t=3D0 0</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; m=3Daudio 8076 RTP/AVP =
0</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; a=3Dalt:1 1.0 : user =
9kksi =3D IP6 2201:056D::112E:144A:1E24&nbsp; (note </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; that</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; I've added</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; a=3Dalt:1 0.9 : user0 =
9kksj=3D=3D IP 4 10.0.1.1 =
1010&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp; an IP6 and </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; IP4)</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; a=3Dalt:2 0.8 : user1 =
9kksk=3D=3D IP4 192.0.2.1 9988 192.0.2.1 9990</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp; a=3Dalt:3 0.4 : user2 =
9kksl=3D=3D IP4 192.0.2.10 8076</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; I think the intent is either A or B. Can =
you clarify?</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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Date: Sat, 25 Oct 2003 13:34:30 +0300
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Francois Audet <audet@nortelnetworks.com>
CC: "'Jonathan ROSENBERG (jdrosen@dynamicsoft.com)'" <jdrosen@dynamicsoft.com>,
        "'Gonzalo Camarillo'" <Gonzalo.Camarillo@lmf.ericsson.se>,
        "'sipping@ietf.org'" <sipping@ietf.org>
Subject: Re: [Sipping] RE: draft-rosenberg-sipping-ice-01 & draft-camarillo-mmusic-alt-0
 2.tx t
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Francois,

such indicator is not necessary because all the alt lines contain 
addresses of the same type as the main address for the m line (in a c 
line that may be at the session level or at the m line level).

Gonzalo

Francois Audet wrote:
> Thanks. That's what I tought. Excellent.
> 
> Minor comment is that if we do allow for STUN on IPv6 addresses, we should
> probably have an "IP6" or "IP4" indicator in the alt line to indicate
> the IP address type (like a c line).
> 
>  > -----Original Message-----
>  > From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
>  > Sent: Thursday, October 23, 2003 22:10
>  > To: Audet, Francois [SC100:4K02:EXCH]
>  > Cc: 'Jonathan ROSENBERG (jdrosen@dynamicsoft.com)'; 'Gonzalo
>  > Camarillo'; 'sipping@ietf.org'
>  > Subject: Re: draft-rosenberg-sipping-ice-01 &
>  > draft-camarillo-mmusic-alt-02.tx t
>  >
>  >
>  > Francois,
>  >
>  > > 1 - In ICE, do you intend to keep the address priority based on the
>  > > qvalue (similar
>  > >     to alt-01), or do you intend to go with the simple
>  > prioritized list
>  > > approach
>  > >     as per alt-02? In both cases, I'm assuming you are
>  > keeping the alt
>  > > parameter.
>  >
>  > Within an m line, we intend to use the alt lines defined in ICE using
>  > qvalues.
>  >
>  > > 2 - Is the intent that IPv6 addresses in SDP be expressed
>  > in ICE using
>  > > the IPV
>  > >     format, or does it follow the IPv4 format as per the
>  > current example
>  > > in ICE?
>  >
>  > The IPv6 addresses will use the IPV format.
>  >
>  > >     Section 5.3 of ICE talks briefly about IPv6 addresses,
>  > but there
>  > > is
>  > > no example
>  > >     on how it would be expressed. I believe that the intent
>  > is to use
>  > > the IPV format
>  > >     in ICE for IPv6 addresses. Is this accurate?
>  >
>  > Yes.
>  >
>  > > 3 - References: ICE should refer to alt-2 not alt-1 in the
>  > reference.
>  > >
>  > > 4 - Is the intent that STUN requests in ICE be receivable
>  > only on IPv4
>  > > addresses?
>  > >     This would assume that IPv6 addresses are always reachable. I
>  > > believe this is
>  > >     the intent.
>  >
>  > If you added alt lines to an IPv6 line, you could run STUN on
>  > it... the
>  > idea is presenting user agents with an IPv6 and an IPv4 address for a
>  > stream. Once the UA picks up a media line (corresponding to
>  > an address),
>  > it does whatever that media line tells it to do (e.g, running STUN).
>  >
>  > > In other words, to take the example A's offer in ICE section 9.1.1
>  > > (Figure 5),
>  > > if there was also an IPv6 address, which of the following
>  > SDP (A, B or
>  > > C) would be
>  > > correct:
>  >
>  > Option A would be correct (use IPV and alt lines with qvalues). We
>  > actually have a draft that contains a similar example (I believe
>  > Jonathan submitted before the cut-off...). This is an SDP taken from
>  > that draft:
>  >
>  >      v=0
>  >      o=bob 280744730 28977631 IN IP6 host.example.com
>  >      s=
>  >      t=0 0
>  >      a=group:IPV 1 2
>  >      m=audio 6886 RTP/AVP 0
>  >      c=IN IP6 2001:056D::112E:144A:1E24
>  >      a=mid:1
>  >      m=audio 22334 RTP/AVP 0
>  >      c=IN IP4 192.0.0.1
>  >      a=mid:2
>  >      a=alt:1 1.0 : user 9kksj== 192.0.0.1 22334
>  >
>  > Thanks,
>  >
>  > Gonzalo
>  >
>  >
>  >
>  > > A: Using IPV, but keeping the qvalue for IPv4
>  > >
>  > >    v=0
>  > >    o=alice 2890844730 2890844731 IN IP4 host.example.com
>  > >    s=
>  > >    t=0 0
>  > >    a=group:IPV 1 2
>  > >    m=audio 6886 RTP/AVP 0
>  > >    c=IN IP6 2201:056D::112E:144A:1E24
>  > >    a=mid:1
>  > >    m=audio 8076 RTP/AVP 0
>  > >    c=IN IP4 192.0.2.10
>  > >    a=alt:1 1.0 : user 9kksj== 10.0.1.1 1010
>  > >    a=alt:2 0.8 : user1 9kksk== 192.0.2.1 9988 192.0.2.1 9990
>  > >    a=alt:3 0.4 : user2 9kksl== 192.0.2.10 8076
>  > >    a=mid:2
>  > >
>  > > B: Using IPV, using simple ordered list instead of qvalue for IPv4
>  > >
>  > >    v=0
>  > >    o=alice 2890844730 2890844731 IN IP4 host.example.com
>  > >    s=
>  > >    t=0 0
>  > >    a=group:IPV 1 2
>  > >    m=audio 6886 RTP/AVP 0
>  > >    c=IN IP6 2201:056D::112E:144A:1E24
>  > >    a=mid:1
>  > >    m=audio 8076 RTP/AVP 0
>  > >    c=IN IP4 192.0.2.10
>  > >    a=alt:1 user 9kksj== 10.0.1.1 1010
>  > >    a=alt:2 user1 9kksk== 192.0.2.1 9988 192.0.2.1 9990
>  > >    a=alt:3 user2 9kksl== 192.0.2.10 8076
>  > >    a=mid:2
>  > >
>  > > C: As per current ICE draft. This implies that we would be running a
>  > > STUN server
>  > >    on the IPv6 address as well to check for connectivity.
>  > >
>  > >    v=0
>  > >    o=alice 2890844730 2890844731 IN IP4 host.example.com
>  > >    s=
>  > >    c=IN IP4 192.0.2.10
>  > >    t=0 0
>  > >    m=audio 8076 RTP/AVP 0
>  > >    a=alt:1 1.0 : user 9kksi = IP6 2201:056D::112E:144A:1E24  (note
>  > > that
>  > > I've added
>  > >    a=alt:1 0.9 : user0 9kksj== IP 4 10.0.1.1 1010          
>  >   an IP6 and
>  > > IP4)
>  > >    a=alt:2 0.8 : user1 9kksk== IP4 192.0.2.1 9988 192.0.2.1 9990
>  > >    a=alt:3 0.4 : user2 9kksl== IP4 192.0.2.10 8076
>  > >
>  > > I think the intent is either A or B. Can you clarify?
>  > >
>  >
>  >
> 

-- 
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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Message-ID: <2F1FC1DEA077D5119FAD00508BCFD6D20BC8D7F6@zsc3c030.us.nortel.com>
From: "Francois Audet" <audet@nortelnetworks.com>
To: "'Gonzalo Camarillo'" <Gonzalo.Camarillo@ericsson.com>
Cc: "'Jonathan ROSENBERG (jdrosen@dynamicsoft.com)'"
	 <jdrosen@dynamicsoft.com>,
        "'Gonzalo Camarillo'"
	 <Gonzalo.Camarillo@lmf.ericsson.se>,
        "'sipping@ietf.org'"
	 <sipping@ietf.org>
Subject: RE: [Sipping] RE: draft-rosenberg-sipping-ice-01 & draft-camarill
	o-mmusic-alt-0 2.tx t
Date: Sun, 26 Oct 2003 09:21:32 -0800
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Good point. Thanks.

However, reading through STUN carefully, it seems to me that it only
allows for IPv4 addresses in the STUN messages themselves. For
example, the MAPPED-ADDRESS only allows for IPv4 addresses at this point.
Unless of course we extend it. (It currently only has a 0x01 Family
corresponding to IPv4).

So, unless I'm missing something, ICE today will not allow the use of STUN 
to verify connectivity to IPv6 addresses. We'd have
to assume that IPv6 addresses always work. We should probably explain
this in ICE, or expand STUN itself to allow for IPv6 addresses as well.
I'm wondering if STUN/ICE running only on IPv4 addresses wouldn't put IPv6
at a disadvantage over IPv4 addresses (they'd be less reliable).

When STUN was defined, it didn't make much sense to allow for IPv6
addresses,
because the purpose was to derive NATed addresses and discover the presence
of
NATs. However, ICE change the applicability of STUN to also be a mechanism
to
verify reachability. 




> -----Original Message-----
> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com] 
> Sent: Saturday, October 25, 2003 03:34
> To: Audet, Francois [SC100:4K02:EXCH]
> Cc: 'Jonathan ROSENBERG (jdrosen@dynamicsoft.com)'; 'Gonzalo 
> Camarillo'; 'sipping@ietf.org'
> Subject: Re: [Sipping] RE: draft-rosenberg-sipping-ice-01 & 
> draft-camarillo-mmusic-alt-0 2.tx t
> 
> 
> Francois,
> 
> such indicator is not necessary because all the alt lines contain 
> addresses of the same type as the main address for the m line (in a c 
> line that may be at the session level or at the m line level).
> 
> Gonzalo

------_=_NextPart_001_01C39BE5.996F38C8
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charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2656.31">
<TITLE>RE: [Sipping] RE: draft-rosenberg-sipping-ice-01 &amp; =
draft-camarillo-mmusic-alt-0 2.tx t</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Good point. Thanks.</FONT>
</P>

<P><FONT SIZE=3D2>However, reading through STUN carefully, it seems to =
me that it only</FONT>
<BR><FONT SIZE=3D2>allows for IPv4 addresses in the STUN messages =
themselves. For</FONT>
<BR><FONT SIZE=3D2>example, the MAPPED-ADDRESS only allows for IPv4 =
addresses at this point.</FONT>
<BR><FONT SIZE=3D2>Unless of course we extend it. (It currently only =
has a 0x01 Family</FONT>
<BR><FONT SIZE=3D2>corresponding to IPv4).</FONT>
</P>

<P><FONT SIZE=3D2>So, unless I'm missing something, ICE today will not =
allow the use of STUN </FONT>
<BR><FONT SIZE=3D2>to verify connectivity to IPv6 addresses. We'd =
have</FONT>
<BR><FONT SIZE=3D2>to assume that IPv6 addresses always work. We should =
probably explain</FONT>
<BR><FONT SIZE=3D2>this in ICE, or expand STUN itself to allow for IPv6 =
addresses as well.</FONT>
<BR><FONT SIZE=3D2>I'm wondering if STUN/ICE running only on IPv4 =
addresses wouldn't put IPv6</FONT>
<BR><FONT SIZE=3D2>at a disadvantage over IPv4 addresses (they'd be =
less reliable).</FONT>
</P>

<P><FONT SIZE=3D2>When STUN was defined, it didn't make much sense to =
allow for IPv6 addresses,</FONT>
<BR><FONT SIZE=3D2>because the purpose was to derive NATed addresses =
and discover the presence of</FONT>
<BR><FONT SIZE=3D2>NATs. However, ICE change the applicability of STUN =
to also be a mechanism to</FONT>
<BR><FONT SIZE=3D2>verify reachability. </FONT>
</P>
<BR>
<BR>
<BR>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Gonzalo Camarillo [<A =
HREF=3D"mailto:Gonzalo.Camarillo@ericsson.com">mailto:Gonzalo.Camarillo@=
ericsson.com</A>] </FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Saturday, October 25, 2003 03:34</FONT>
<BR><FONT SIZE=3D2>&gt; To: Audet, Francois [SC100:4K02:EXCH]</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: 'Jonathan ROSENBERG =
(jdrosen@dynamicsoft.com)'; 'Gonzalo </FONT>
<BR><FONT SIZE=3D2>&gt; Camarillo'; 'sipping@ietf.org'</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: Re: [Sipping] RE: =
draft-rosenberg-sipping-ice-01 &amp; </FONT>
<BR><FONT SIZE=3D2>&gt; draft-camarillo-mmusic-alt-0 2.tx t</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Francois,</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; such indicator is not necessary because all the =
alt lines contain </FONT>
<BR><FONT SIZE=3D2>&gt; addresses of the same type as the main address =
for the m line (in a c </FONT>
<BR><FONT SIZE=3D2>&gt; line that may be at the session level or at the =
m line level).</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Gonzalo</FONT>
</P>

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Date: Sun, 26 Oct 2003 19:40:33 +0200
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Francois Audet <audet@nortelnetworks.com>
CC: "'Jonathan ROSENBERG (jdrosen@dynamicsoft.com)'" <jdrosen@dynamicsoft.com>,
        "'Gonzalo Camarillo'" <Gonzalo.Camarillo@lmf.ericsson.se>,
        "'sipping@ietf.org'" <sipping@ietf.org>
Subject: Re: [Sipping] RE: draft-rosenberg-sipping-ice-01 & draft-camarill
 o-mmusic-alt-0 2.tx t
References: <2F1FC1DEA077D5119FAD00508BCFD6D20BC8D7F6@zsc3c030.us.nortel.com>
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Francois,

if STUN is used only for NAT traversal, it may make sense to define only 
IPv4 addresses. Nevertheless, if we start using STUN to perform return 
routability tests to prevent DoS attacks (Jonathan has a draft about 
it), it will make sense to define IPv6 addresses as well.

To me, it makes sense to use the same mechanism for both things. This 
way, implementors are encouraged to implement it so that they can 
communicate with more people, and we kill two birds with the same stone. 
Otherwise, I do not know how much priority a specific mechanism for DoS 
prevention would get... people tend to ignore this type of thing until a 
big attack hits them.

Gonzalo

Francois Audet wrote:

> Good point. Thanks.
> 
> However, reading through STUN carefully, it seems to me that it only
> allows for IPv4 addresses in the STUN messages themselves. For
> example, the MAPPED-ADDRESS only allows for IPv4 addresses at this point.
> Unless of course we extend it. (It currently only has a 0x01 Family
> corresponding to IPv4).
> 
> So, unless I'm missing something, ICE today will not allow the use of STUN
> to verify connectivity to IPv6 addresses. We'd have
> to assume that IPv6 addresses always work. We should probably explain
> this in ICE, or expand STUN itself to allow for IPv6 addresses as well.
> I'm wondering if STUN/ICE running only on IPv4 addresses wouldn't put IPv6
> at a disadvantage over IPv4 addresses (they'd be less reliable).
> 
> When STUN was defined, it didn't make much sense to allow for IPv6 
> addresses,
> because the purpose was to derive NATed addresses and discover the 
> presence of
> NATs. However, ICE change the applicability of STUN to also be a 
> mechanism to
> verify reachability.
> 
> 
> 
> 
>  > -----Original Message-----
>  > From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
>  > Sent: Saturday, October 25, 2003 03:34
>  > To: Audet, Francois [SC100:4K02:EXCH]
>  > Cc: 'Jonathan ROSENBERG (jdrosen@dynamicsoft.com)'; 'Gonzalo
>  > Camarillo'; 'sipping@ietf.org'
>  > Subject: Re: [Sipping] RE: draft-rosenberg-sipping-ice-01 &
>  > draft-camarillo-mmusic-alt-0 2.tx t
>  >
>  >
>  > Francois,
>  >
>  > such indicator is not necessary because all the alt lines contain
>  > addresses of the same type as the main address for the m line (in a c
>  > line that may be at the session level or at the m line level).
>  >
>  > 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
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo


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From: "Christian Stredicke" <stredicke@snom.de>
To: "'Francois Audet'" <audet@nortelnetworks.com>,
        "'Gonzalo Camarillo'" <Gonzalo.Camarillo@ericsson.com>
Cc: "'Jonathan ROSENBERG'" <jdrosen@dynamicsoft.com>,
        "'Gonzalo Camarillo'" <Gonzalo.Camarillo@lmf.ericsson.se>,
        <sipping@ietf.org>
Subject: AW: [Sipping] RE: draft-rosenberg-sipping-ice-01 & draft-camarillo-mmusic-alt-0 2.tx t
Date: Sun, 26 Oct 2003 18:51:34 +0100
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1) IPv4/IPv6

=20

There will definitively be firewalls in IPv6. So we should include IPv6
right from the beginning.

=20

Maybe it is possible to determine IPv4/IPv6 by the =93look=94. Address
=93123.123.123.123=94 looks different than =931080::8:800:200C:417A=94.

=20

2) Expected Source Address

=20

There is a trick to go though symmetrical NAT from everywhere: The
sender of the packets forges the source address with the expected source
address (writing on raw sockets). This makes the firewall think the
packet is coming from the expected source and let it pass through. =
It=92s
a dirty trick but it works in many cases.

=20

It would be nice if we have a description for the expected source
address.

=20

CS

=20

-----Urspr=FCngliche Nachricht-----
Von: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] Im Auftrag
von Francois Audet
Gesendet: Sonntag, 26. Oktober 2003 18:22
An: 'Gonzalo Camarillo'
Cc: 'Jonathan ROSENBERG (jdrosen@dynamicsoft.com)'; 'Gonzalo Camarillo';
'sipping@ietf.org'
Betreff: RE: [Sipping] RE: draft-rosenberg-sipping-ice-01 &
draft-camarillo-mmusic-alt-0 2.tx t

=20

Good point. Thanks.=20

However, reading through STUN carefully, it seems to me that it only=20
allows for IPv4 addresses in the STUN messages themselves. For=20
example, the MAPPED-ADDRESS only allows for IPv4 addresses at this
point.=20
Unless of course we extend it. (It currently only has a 0x01 Family=20
corresponding to IPv4).=20

So, unless I'm missing something, ICE today will not allow the use of
STUN=20
to verify connectivity to IPv6 addresses. We'd have=20
to assume that IPv6 addresses always work. We should probably explain=20
this in ICE, or expand STUN itself to allow for IPv6 addresses as well.=20
I'm wondering if STUN/ICE running only on IPv4 addresses wouldn't put
IPv6=20
at a disadvantage over IPv4 addresses (they'd be less reliable).=20

When STUN was defined, it didn't make much sense to allow for IPv6
addresses,=20
because the purpose was to derive NATed addresses and discover the
presence of=20
NATs. However, ICE change the applicability of STUN to also be a
mechanism to=20
verify reachability.=20





> -----Original Message-----=20
> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]=20
> Sent: Saturday, October 25, 2003 03:34=20
> To: Audet, Francois [SC100:4K02:EXCH]=20
> Cc: 'Jonathan ROSENBERG (jdrosen@dynamicsoft.com)'; 'Gonzalo=20
> Camarillo'; 'sipping@ietf.org'=20
> Subject: Re: [Sipping] RE: draft-rosenberg-sipping-ice-01 &=20
> draft-camarillo-mmusic-alt-0 2.tx t=20
>=20
>=20
> Francois,=20
>=20
> such indicator is not necessary because all the alt lines contain=20
> addresses of the same type as the main address for the m line (in a c=20
> line that may be at the session level or at the m line level).=20
>=20
> Gonzalo=20


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draft-camarillo-mmusic-alt-0 2.tx t</title>

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<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>1) =
IPv4/IPv6</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>&nbsp;</span></=
font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>There will =
definitively
be firewalls in IPv6. So we should include IPv6 right from the =
beginning.</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>&nbsp;</span></=
font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>Maybe it is =
possible to
determine IPv4/IPv6 by the &#8220;look&#8221;. Address =
&#8220;123.123.123.123&#8221;
looks different than =
&#8220;1080::8:800:200C:417A&#8221;.</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>&nbsp;</span></=
font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>2) Expected =
Source
Address</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>&nbsp;</span></=
font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>There is a =
trick to go
though symmetrical NAT from everywhere: The sender of the packets forges =
the
source address with the expected source address (writing on raw =
sockets). This makes
the firewall think the packet is coming from the expected source and let =
it pass
through. It&#8217;s a dirty trick but it works in many =
cases.</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>&nbsp;</span></=
font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>It would be =
nice if we
have a description for the expected source address.</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>&nbsp;</span></=
font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>CS</span></font=
></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>&nbsp;</span></=
font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma'>-----Urspr=FCngliche Nachricht-----<br>
<b><span style=3D'font-weight:bold'>Von:</span></b> =
sipping-admin@ietf.org
[mailto:sipping-admin@ietf.org] <b><span style=3D'font-weight:bold'>Im =
Auftrag
von </span></b>Francois Audet<br>
<b><span style=3D'font-weight:bold'>Gesendet:</span></b> Sonntag, 26. =
Oktober
2003 18:22<br>
<b><span style=3D'font-weight:bold'>An:</span></b> 'Gonzalo =
Camarillo'<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> 'Jonathan ROSENBERG
(jdrosen@dynamicsoft.com)'; 'Gonzalo Camarillo'; 'sipping@ietf.org'<br>
<b><span style=3D'font-weight:bold'>Betreff:</span></b> RE: [Sipping] =
RE:
draft-rosenberg-sipping-ice-01 &amp; draft-camarillo-mmusic-alt-0 2.tx =
t</span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

<p><font size=3D2 face=3D"Times New Roman"><span =
style=3D'font-size:10.0pt'>Good
point. Thanks.</span></font> </p>

<p><font size=3D2 face=3D"Times New Roman"><span =
style=3D'font-size:10.0pt'>However,
reading through STUN carefully, it seems to me that it =
only</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>allows for IPv4 =
addresses in the
STUN messages themselves. For</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>example, the =
MAPPED-ADDRESS only
allows for IPv4 addresses at this point.</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>Unless of course we =
extend it. (It
currently only has a 0x01 Family</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>corresponding to =
IPv4).</span></font>
</p>

<p><font size=3D2 face=3D"Times New Roman"><span =
style=3D'font-size:10.0pt'>So,
unless I'm missing something, ICE today will not allow the use of STUN =
</span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>to verify connectivity =
to IPv6
addresses. We'd have</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>to assume that IPv6 =
addresses
always work. We should probably explain</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>this in ICE, or expand =
STUN itself
to allow for IPv6 addresses as well.</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>I'm wondering if =
STUN/ICE running
only on IPv4 addresses wouldn't put IPv6</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>at a disadvantage over =
IPv4 addresses
(they'd be less reliable).</span></font> </p>

<p><font size=3D2 face=3D"Times New Roman"><span =
style=3D'font-size:10.0pt'>When STUN
was defined, it didn't make much sense to allow for IPv6 =
addresses,</span></font>
<br>
<font size=3D2><span style=3D'font-size:10.0pt'>because the purpose was =
to derive
NATed addresses and discover the presence of</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>NATs. However, ICE =
change the
applicability of STUN to also be a mechanism to</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>verify reachability. =
</span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'><br>
<br>
</span></font></p>

<p><font size=3D2 face=3D"Times New Roman"><span =
style=3D'font-size:10.0pt'>&gt;
-----Original Message-----</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>&gt; From: Gonzalo =
Camarillo [<a
href=3D"mailto:Gonzalo.Camarillo@ericsson.com">mailto:Gonzalo.Camarillo@e=
ricsson.com</a>]
</span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>&gt; Sent: Saturday, =
October 25,
2003 03:34</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>&gt; To: Audet, Francois
[SC100:4K02:EXCH]</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>&gt; Cc: 'Jonathan =
ROSENBERG
(jdrosen@dynamicsoft.com)'; 'Gonzalo </span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>&gt; Camarillo'; =
'sipping@ietf.org'</span></font>
<br>
<font size=3D2><span style=3D'font-size:10.0pt'>&gt; Subject: Re: =
[Sipping] RE:
draft-rosenberg-sipping-ice-01 &amp; </span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>&gt; =
draft-camarillo-mmusic-alt-0
2.tx t</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>&gt; </span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>&gt; </span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>&gt; =
Francois,</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>&gt; </span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>&gt; such indicator is =
not
necessary because all the alt lines contain </span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>&gt; addresses of the =
same type as
the main address for the m line (in a c </span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>&gt; line that may be at =
the
session level or at the m line level).</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>&gt; </span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>&gt; =
Gonzalo</span></font> </p>

</div>

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From: "Francois Audet" <audet@nortelnetworks.com>
To: "'Gonzalo Camarillo'" <Gonzalo.Camarillo@ericsson.com>
Cc: "'Jonathan ROSENBERG (jdrosen@dynamicsoft.com)'"
	 <jdrosen@dynamicsoft.com>,
        "'Gonzalo Camarillo'"
	 <Gonzalo.Camarillo@lmf.ericsson.se>,
        "'sipping@ietf.org'"
	 <sipping@ietf.org>
Subject: RE: [Sipping] RE: draft-rosenberg-sipping-ice-01 & draft-camarill
	 o-mmusic-alt-0 2.tx t
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We are in agreement. 


> -----Original Message-----
> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com] 
> Sent: Sunday, October 26, 2003 09:41
> To: Audet, Francois [SC100:4K02:EXCH]
> Cc: 'Jonathan ROSENBERG (jdrosen@dynamicsoft.com)'; 'Gonzalo 
> Camarillo'; 'sipping@ietf.org'
> Subject: Re: [Sipping] RE: draft-rosenberg-sipping-ice-01 & 
> draft-camarill o-mmusic-alt-0 2.tx t
> 
> 
> Francois,
> 
> if STUN is used only for NAT traversal, it may make sense to 
> define only 
> IPv4 addresses. Nevertheless, if we start using STUN to 
> perform return 
> routability tests to prevent DoS attacks (Jonathan has a draft about 
> it), it will make sense to define IPv6 addresses as well.
> 
> To me, it makes sense to use the same mechanism for both things. This 
> way, implementors are encouraged to implement it so that they can 
> communicate with more people, and we kill two birds with the 
> same stone. 
> Otherwise, I do not know how much priority a specific 
> mechanism for DoS 
> prevention would get... people tend to ignore this type of 
> thing until a 
> big attack hits them.
> 
> Gonzalo

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Content-Type: text/html
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2656.31">
<TITLE>RE: [Sipping] RE: draft-rosenberg-sipping-ice-01 &amp; =
draft-camarill o-mmusic-alt-0 2.tx t</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>We are in agreement. </FONT>
</P>
<BR>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Gonzalo Camarillo [<A =
HREF=3D"mailto:Gonzalo.Camarillo@ericsson.com">mailto:Gonzalo.Camarillo@=
ericsson.com</A>] </FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Sunday, October 26, 2003 09:41</FONT>
<BR><FONT SIZE=3D2>&gt; To: Audet, Francois [SC100:4K02:EXCH]</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: 'Jonathan ROSENBERG =
(jdrosen@dynamicsoft.com)'; 'Gonzalo </FONT>
<BR><FONT SIZE=3D2>&gt; Camarillo'; 'sipping@ietf.org'</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: Re: [Sipping] RE: =
draft-rosenberg-sipping-ice-01 &amp; </FONT>
<BR><FONT SIZE=3D2>&gt; draft-camarill o-mmusic-alt-0 2.tx t</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Francois,</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; if STUN is used only for NAT traversal, it may =
make sense to </FONT>
<BR><FONT SIZE=3D2>&gt; define only </FONT>
<BR><FONT SIZE=3D2>&gt; IPv4 addresses. Nevertheless, if we start using =
STUN to </FONT>
<BR><FONT SIZE=3D2>&gt; perform return </FONT>
<BR><FONT SIZE=3D2>&gt; routability tests to prevent DoS attacks =
(Jonathan has a draft about </FONT>
<BR><FONT SIZE=3D2>&gt; it), it will make sense to define IPv6 =
addresses as well.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; To me, it makes sense to use the same mechanism =
for both things. This </FONT>
<BR><FONT SIZE=3D2>&gt; way, implementors are encouraged to implement =
it so that they can </FONT>
<BR><FONT SIZE=3D2>&gt; communicate with more people, and we kill two =
birds with the </FONT>
<BR><FONT SIZE=3D2>&gt; same stone. </FONT>
<BR><FONT SIZE=3D2>&gt; Otherwise, I do not know how much priority a =
specific </FONT>
<BR><FONT SIZE=3D2>&gt; mechanism for DoS </FONT>
<BR><FONT SIZE=3D2>&gt; prevention would get... people tend to ignore =
this type of </FONT>
<BR><FONT SIZE=3D2>&gt; thing until a </FONT>
<BR><FONT SIZE=3D2>&gt; big attack hits them.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Gonzalo</FONT>
</P>

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</HTML>
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Subject: Re: [Sipping] [Fwd: I-D ACTION:draft-taylor-sipping-emerg-scen-00.txt]
References: <E0E9F9BE940EFC4186DB15A079FF315001AA5EEF@i2km35-ukdy.domain1.systemhost.net>
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Thanks for your comments, Dick.  I've just submitted a new version=20
incorporating your comments, generally verbatim.  I have a couple of=20
counter-comments below.

I made sure to add you to the acknowledgements section this time around :=
-)

dick.rr.knight@bt.com wrote:

>=20
> Comments on draft-taylor-sipping-emerg-scen-00.txt, based on the
> information sent to Tom about the UK emergency service (999).
>=20
...
>=20
> 5. Again in Section 3.2, under (=A3) and (4), the document reads as
> though it is placing requirements on the PSTN, rather than defining
> requirements for SIP. The PSTN is unlikely to change! This part needs
> rewording as follows: (3)   The telephone network should be
> supplied-with, or enabled to supply, a calling party number or other
> unique geographic indicator, that can serve as a key to the mapping
> database. (4)   The caller's location needs to be supplied during
> call set-up, or by being keyed by calling party number or other
> unique identifier, is available in the mapping database.  In this
> document this is assumed to happen as a configuration step in advance
> of the call.

[PTT] I accepted your rewording.  Just a word of explanation for how it=20
was: this section in my mind presented the premises upon which the=20
emergency call system is built.  You took it to be the requirements=20
specifically on the SIP network.  That's a better slant, and more to the=20
point of this document.
...
>=20
> 12. Section 4.2.1, penultimate paragraph: rather than PSTN having "to
> know what number to use", it would be better to say what SIP needs to
> do: "the PSTN gateway also needs to be provided with a callback
> number to use
>=20
> 13. In 4.3.2, last bullet, beginning "The PSTN gateway is responsible
> for ...", you seem to be placing requirements on the PSTN gateway.
> The PSTN gateway can only act on the information it receives...
> Suggest adding "(based on information from the SIP network)" after "a
> called party number on the PSTN side".

[PTT] I viewed the PSTN gateway primarily as a SIP entity.  I accepted=20
your rewording, but with some uneasiness: a PSTN gateway which is part=20
of the same network as the SIP phone (both in the same company network,=20
for example) can have access to information which is not provided=20
directly by preceding SIP entities in the call path.  Your wording=20
applies if the PSTN gateway is in a different administrative domain from=20
the SIP phone.
>=20
...
>=20
> 15. In 5. Conclusions, before the paragraph beginning "Finally ...",
> I suggest a new paragraph that states: "There is now an urgency to
> arrive at emergency call standards with use of VoIP growing and the
> potential for existing implementations to result in problems and
> delays in organising responses for ECCs."
>=20
[PTT] I'm not sure this follows from the preceding part of the document.=20
  I see it as a motivation rather than a conclusion, so I stuck it at=20
the beginning of the introduction.


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Date: Mon, 27 Oct 2003 00:01:50 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: "Daniel G. Petrie" <dpetrie@pingtel.com>
CC: "David R. Oran" <oran@cisco.com>, sipping@ietf.org
Subject: Re: [Sipping] Re: Queries about draft-petrie-sipping-xfer-issues-00.txt
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Some thoughts inline.

Daniel G. Petrie wrote:


> 
>>2. For "Consultative Call Prompting" and other similar advisory notices
>>about the purpose of a call, just like for email, isn't the "Subject:"
>>header exactly what's called for?
> 
> 
> Yes.  I unfortunately missed this before I submitted the draft.  This
> helps for prompting, however there is still a need for a symantic
> purpose which probably does not make sense in the Subject header
> as the syntax is freeform text.

One idea here is to have a registry of well-known subjects, which can 
always be rendered and are meaningful in that way, but for which an 
automata can look out for, and do something different with, if it sees it.


> 
>>4. "Coerce Gateway Hairpins" is the first of these scenarios for which
>>reasonably obvious use of existing protocol features doesn't seem to hold a
>>solution. I have a suggestion though, which might have wider applicability,
>>and now that we have GRUUs may be just the ticket. Define a new header
>>field called "Dialog Affinity: <gruu>". The semantics would be for the
>>proxy in the destination user's domain to attempt if possible to route the
>>request to the same UA (in this case gateway) as is indicated by the
>>corresponding Contact: address of the referenced GRUU. Finer granularity to
>>the trunk level could be achieved if the subject gateway manufactured GRUUs
>>which contained grid values identifying the appropriate trunk.
>>

Yet another reason, btw, why unique gruus are a good idea.

> 
> 
> This is a little different from how GRUUs are to be used elsewhere.
> Effectively the dialog affinity GRUU is more of a routing URI as
> it is not the final distination.  The transfer target is URI (e.g.
> dialed digits part of the URI initicating the transfer target).
> 
> So just to explore this a little further and make sure I understand
> what you mean:
> The gateway provides a dialog affinity GRUU for every call in or
> out of the gateway.  Should the other end of the dialog decide to
> do a transfer it copies the dialog affinity GRUU into the consultative
> INVITE.  The URI cannot change from what is done today as the
> URI contains the transfer target which the dialog affinity GRUU
> could not have contained.  The proxy(s)/location server(s) do a kind
> of loose route with the dialog affinity GRUU (or perhaps the
> transferor does the loose route?)?  This avoids the normal routing
> decisions made by the proxy(s)/location server(s) which might send
> it to a different gateway.

This is what I was thinking as well. In particular, I didnt see a need 
for a separate header at all. The transferor would verify that its 
peer in the original call (the gateway) supports gruus (based on 
Supported: gruu in INVITE/200). Then, to initiate the consultative 
INVITE, it would literally use loose routing:

INVITE sip:transfer-target@domain.com SIP/2.0
Route: sip:gateway-gruu@gateway-domain.com

Where the Route header contains the gruu. This causes the consultative 
transfer to always hairpin through the transferee. We would probably 
need an additional Supported header field value that allows the 
transferee to say "I support routing of consultative transfers through 
me", so that the transferor would know whether it could construct an 
INVITE as above.


> 
> I think that the proxy(s)/location servers need to be aware of the
> dialog affinity GRUU as the location server(s) may still need to
> perform transformations on the target URI (e.g. add/remove digit
> prefixes, etc.).  I am not sure the loose route can work as it is not
> clear when or which location server (where there are multiple that
> transform the URI) should insert the loose route.

I think the transferor inserts it. It will get the call routed to the 
gateway. If the target of the call requires some kind of digit 
analysis from there, the gateway should take care of that, or contact 
a location server to do it. In the latter case, it can use an INVITE 
with a Request-Disposition that forces a redirect.

As an additional issue, the behavior of a proxy on seeing a Route 
header with an AOR or gruu in it, is far from clear. I believe the 
appropriate behavior is to actually perform the location lookup on the 
Route header, and take the results and push those into another Route 
header. Basically, instead of normal proxy processing which transforms 
the R-URI, it would transform the Route header. That allows for some 
very interesting call routing behaviors.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.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 Proposal Investigation Working Group of the IETF.

	Title		: Session Initiation Protocol Torture Test Messages
	Author(s)	: A. Johnston
	Filename	: draft-ietf-sipping-torture-tests-01.txt
	Pages		: 52
	Date		: 2003-10-24
	
This informational document gives examples of Session Initiation 
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parser.  They were developed as part of the SIPit SIP 
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.

	Title		: Session Initiation Protocol Torture Test Messages
	Author(s)	: A. Johnston
	Filename	: draft-ietf-sipping-torture-tests-01.txt
	Pages		: 52
	Date		: 2003-10-24
	
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On Monday, October 20, 2003, at 07:45 AM, Rosen, Brian wrote:

> This document is a start, but in my opinion not enough.
>
> The introduction must define dial strings, and differentiate them
> from local phone numbers.

I think this is the crux of the disagreement.  I don't think it is 
possible to give a rigorous definition of what is a dial string and 
what is a local number.  I invite you to describe an algorithm you can 
use to determine which is which.  If you can then I'll incorporate it 
in the draft, but I am skeptical that such a distinction is testable 
any more than the distinction between "erotic art" and "pornography".

>  It should introduce the idea that a
> proxy in the listed domain of a dial string must translate to a global
> or local number, and that some phones will do this translation
> themselves while others will not.
>
> Section 5 inadequately deals with dial strings.  Dial strings are not
> local numbers, and must be translated to local numbers.  One must
> disambiguate dial strings from local numbers because it is possible 
> that
> they collide.  Such a differentiation is also useful to know when
> translation has been completed.

i don't think they can collide because they are in different contexts

> It has been proposed to use separate contexts for dial strings and
> local numbers.  I generally don't like this idea, but it can be made
> to work.  It would be best if there was some kind of naming convention
> to differentiate, but this is not strictly necessary.
>
> What is necessary is text in a document:
> 	1. Saying that dial strings aren't local numbers
> 	2. Permitting dial strings in a sip uri, with a dial string context.
> 		Note: if you wish to permit a dial string in a tel uri, then
>                  the language in 2806bis must be modified, IMO.
> 	3. Explicitly defining the use of E/F
> 	4. Illustrating the use of separate contexts for dial strings
> 	5. Discussing the provisioning implications of having two
> 		local contexts, one for phones with translation and
> 		another for phones without translation

I thought I covered this last one a bit in the draft. A phone that 
doesn't know its context can just use *itself* as the context.  That 
requires no configuration.  A phone that has a digit map obviously 
needs to be configured with it, so explicitly setting the target 
context or domain (or both) should be no problem IMO.

> It would be much more satisfactory, in my opinion, to simply define
> user=dialstring, or something similar, to denote that the user part is
> a dial string.  Note that sip phones that presently put dial strings in
> a sip uri without a context must change anyway.
>
> We also should define a way to express a dial string with pauses and 
> waits.
> They are pretty useful. They are dialable in many gateways.  We need
> a way to send such a string.  Again, we need text, and again, we could
> use a separate context for such strings.
>
> I do agree with the draft's concepts that
> 	user=phone indicates a local or global number as defined by 2806bis
> 	user=phone without a context on a local number is deprecated

cool.  I'm very happy about that.

> As before, I will write text when we agree we want text to cover.
> Thanks to Rohan for this good start.

thanks for your comments.
take care,
-rohan

> Brian
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP


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Subject: [Sipping] Please Read Committee T1 Draft on DTMF Notification
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While we were arguing about how to handle keypresses, somebody went out
and wrote a draft for an event package for DTMF events.

Please review the draft:

http://www.softarmor.com/wgdb/docs/draft-zebarth-sipping-dtmfad-00.txt

As I understand it, the mechanism that this draft is meant to support is
defined in Committee T1 Technical Report T1.TR.82-2003. Unfortunately, I
wasn't able to find said document on the T1 site, but that could just be
me -- perhaps the author can give us a link?

Questions to consider: Does this approach complement or conflict with
the KPML work? Is there danger of fragmenting, or is this work clearly
seperate? Would the world (or at least the Internet) be a better place
with, or without, this event package?

Please copy the author, as well as the list, on any responses. I expect
that discussion on this topic will come up during the Minneapolis
meeting.

Thanks,

--
Dean Willis
SIPPING co-chair

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From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sipping] Re: draft-petrie-sipping-xfer-issues-00.txt
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Dan,

This paper raises some interesting issues.

I think the problem of routing the consultation call via the same gateway
(if the potential transfer target is in the same legacy network as the
transferee) would be a useful problem to solve. On the other hand, care
should be taken to ensure that the consultation call doesn't egress via the
wrong gateway because of this. For example, the first call has come in via a
QSIG gateway. The consultation call is to an address in the legacy domain
but which is best reached via an ISUP gateway. If the call gets routed back
to the QSIG gateway, it depends on routing intelligence in the QSIG gateway
to ensure that it is onward routed in the SIP network (to an appropriate
ISUP gateway) rather than going back into the QSIG network and perhaps
taking a non-optimal route to the ISUP destination (perhaps even coming back
into the SIP network first). In fact the gateway would be required to act as
proxy.

I also have concerns about carrying anything in the INVITE request that
indicates too strongly that the call is a consultation call linked to a
particular existing call / transferee with an intention to transfer. Does
the transferor UA necessarily know this? The user interface may not allow
the transferor to indicate this. The transferor may not even know at this
stage - he may just wish to consult with somebody and then, after talking,
decide to do transfer rather than revert to the original call or form an ad
hoc conference. So indicating this to the callee (e.g, by the Subject
header), unless the caller has explicitly indicated that transfer will
follow, is somewhat dangerous.

If the transferor has two or more held calls, which one does the
consultation call relate to? I guess UAs will tend to have their own rules
about which one is applicable if transfer is invoked, in which case that
could be used for ensuring routing via the same gateway. However, if other
manipulations occur between establishment of the consultation call and the
eventual transfer, this "guess" could be wrong. It should work in the
majority of cases, so it is probably still worth doing.

Finally I am concerned about the transition from consultation to blind
transfer. Although maintaining ringing for an extra few seconds after CANCEL
would help to solve problems, particularly at a gateway, I can imagine there
would be a lot of cases where the callee answers and finds nobody there
because the blind transfer does not in fact occur. This might be because the
caller has given up and decides to revert to the original call, because a
proxy has invoked call forwarding no reply, or because a proxy has used
parallel forking and another UAS has answered.

I question why Replaces to replace an early dialog will not work. The
transferor should not normally try to invoke transfer until a 180 has been
received on the consultation call, which means that the consultation call is
relatively stable, so there is a good chance that it will be allowed to
alert for several seconds and therefore a good chance that REFER with
Replaces will work in this situation. Yes, there will be times when, due to
forking actions of the proxy, this will not work, but I don't think the
alternatives work either.

John Elwell (john.elwell@siemens.com)

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To: Dean Willis <dean.willis@softarmor.com>
CC: sipping@ietf.org, Joe Zebarth <zebarth@nortelnetworks.com>,
        rohan@cisco.com
Subject: Re: [Sipping] Please Read Committee T1 Draft on DTMF Notification
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hi;
I found this draft very similar to: draft-mahy-sip-signaled-digits-xx

The differences I found are:
1- The name of the Event Package
2- The Notification MIME content-type
3- How each draft encode an RFC 2833 ( AVT-Tones ) Packet
4- ( I think both drafts define the same set of parameters in the Event 
header)

By the way, what is the status of draft-mahy-sip-signaled-digits?

Thanks

Diego Besprosvan

MailVision Ltd.
Haganim 10/a
German Colony, Haifa
Israel
e-mail: diegob@mailvision.com
www.mailvision.com




Dean Willis wrote:
> While we were arguing about how to handle keypresses, somebody went out
> and wrote a draft for an event package for DTMF events.
> 
> Please review the draft:
> 
> http://www.softarmor.com/wgdb/docs/draft-zebarth-sipping-dtmfad-00.txt
> 
> As I understand it, the mechanism that this draft is meant to support is
> defined in Committee T1 Technical Report T1.TR.82-2003. Unfortunately, I
> wasn't able to find said document on the T1 site, but that could just be
> me -- perhaps the author can give us a link?
> 
> Questions to consider: Does this approach complement or conflict with
> the KPML work? Is there danger of fragmenting, or is this work clearly
> seperate? Would the world (or at least the Internet) be a better place
> with, or without, this event package?
> 
> Please copy the author, as well as the list, on any responses. I expect
> that discussion on this topic will come up during the Minneapolis
> meeting.
> 
> Thanks,
> 
> --
> Dean Willis
> SIPPING co-chair
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 


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

	Title		: A Call Control and Multi-party usage framework for the Session Initiation Protocol (SIP)
	Author(s)	: R. Mahy
	Filename	: draft-ietf-sipping-cc-framework-03.txt
	Pages		: 43
	Date		: 2003-10-27
	
This document defines a framework and requirements for multi-party
usage of SIP.  To enable discussion of multi-party features and
applications we define an abstract call model for describing the
media relationships required by many of these.  The model and actions
described here are specifically chosen to be independent of the SIP
signaling and/or mixing approach chosen to actually setup the media
relationships.  In addition to its dialog manipulation aspect, this
framework includes requirements for communicating related information
and events such as conference and session state, and session history.
This framework also describes other goals which embody the spirit of
SIP applications as used on the Internet.

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

	Title		: A Framework for SIP User Agent Configuration
	Author(s)	: D. Petrie
	Filename	: draft-ietf-sipping-config-framework-01.txt
	Pages		: 22
	Date		: 2003-10-27
	
This document defines the application of a set of protocols for 
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of changes and updates which impact its configuration.  The 
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agents.  The contents and format of the configuration data to be 
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On Oct 28, 2003, at 2:17, Diego B wrote:

> hi;
> I found this draft very similar to: draft-mahy-sip-signaled-digits-xx
>
> The differences I found are:
> 1- The name of the Event Package
> 2- The Notification MIME content-type
> 3- How each draft encode an RFC 2833 ( AVT-Tones ) Packet
> 4- ( I think both drafts define the same set of parameters in the 
> Event header)
>
> By the way, what is the status of draft-mahy-sip-signaled-digits?


I abandoned this work in favor of the app-interaction/kpml work.

thanks,
-rohan mahy


> Thanks
>
> Diego Besprosvan
>
> MailVision Ltd.
> Haganim 10/a
> German Colony, Haifa
> Israel
> e-mail: diegob@mailvision.com
> www.mailvision.com
>
>
>
>
> Dean Willis wrote:
>> While we were arguing about how to handle keypresses, somebody went 
>> out
>> and wrote a draft for an event package for DTMF events.
>> Please review the draft:
>> http://www.softarmor.com/wgdb/docs/draft-zebarth-sipping-dtmfad-00.txt
>> As I understand it, the mechanism that this draft is meant to support 
>> is
>> defined in Committee T1 Technical Report T1.TR.82-2003. 
>> Unfortunately, I
>> wasn't able to find said document on the T1 site, but that could just 
>> be
>> me -- perhaps the author can give us a link?
>> Questions to consider: Does this approach complement or conflict with
>> the KPML work? Is there danger of fragmenting, or is this work clearly
>> seperate? Would the world (or at least the Internet) be a better place
>> with, or without, this event package?
>> Please copy the author, as well as the list, on any responses. I 
>> expect
>> that discussion on this topic will come up during the Minneapolis
>> meeting.
>> Thanks,
>> --
>> Dean Willis
>> SIPPING co-chair
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
>


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Tue, 28 Oct 2003 20:16:25 +0200
From: Diego B <diegob@mailvision.com>
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To: sipping@ietf.org
CC: rohan@cisco.com, schulzrinne@cs.columbia.edu, jdrosen@dynamicsoft.com
Subject: Re: [Sipping] I-D ACTION:draft-ietf-sipping-dialog-package-03.txt
References: <200310281440.JAA24994@ietf.org>
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Hi;
This is right?

"
    Automatic Callback: In this basic Public Switched Telephone Network
       (PSTN) application, user A calls user B. User B is busy. User A
       would like to get a callback when user B hangs up. When B hangs
       up, user A's phone rings. When A picks it up, they here ringing,
       and are being connected to B. To implement this with SIP, a
       mechanism is required for B to receive a notification when the
                                 ^
       dialogs at A are complete.
                  ^
"
In this example, 'A' may receive notifications on the state of 'B'
dialog.


Tnx

Diego Besprosvan

MailVision Ltd.
Haganim 10/a
German Colony, Haifa
Israel
e-mail: diegob@mailvision.com
www.mailvision.com


Internet-Drafts@ietf.org wrote:

> A New Internet-Draft is available from the on-line Internet-Drafts directories.
> This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.
> 
> 	Title		: An INVITE Inititiated Dialog Event Package for the Session Initiation Protocol (SIP)
> 	Author(s)	: J. Rosenberg, H. Schulzrinne
> 	Filename	: draft-ietf-sipping-dialog-package-03.txt
> 	Pages		: 30
> 	Date		: 2003-10-27
> 	
> This document defines a dialog event package for the SIP Events
> architecture, along with a data format used in notifications for this
> package. The dialog package allows users to subscribe to another
> user, an receive notifications about the changes in state of INVITE
> initiated dialogs that the user is involved in.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-dialog-package-03.txt
> 
> To remove yourself from the IETF Announcement list, send a message to 
> ietf-announce-request with the word unsubscribe in the body of the message.
> 
> Internet-Drafts are also available by anonymous FTP. Login with the username
> "anonymous" and a password of your e-mail address. After logging in,
> type "cd internet-drafts" and then
> 	"get draft-ietf-sipping-dialog-package-03.txt".
> 
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html 
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> 
> 
> Internet-Drafts can also be obtained by e-mail.
> 
> Send a message to:
> 	mailserv@ietf.org.
> In the body type:
> 	"FILE /internet-drafts/draft-ietf-sipping-dialog-package-03.txt".
> 	
> NOTE:	The mail server at ietf.org can return the document in
> 	MIME-encoded form by using the "mpack" utility.  To use this
> 	feature, insert the command "ENCODING mime" before the "FILE"
> 	command.  To decode the response(s), you will need "munpack" or
> 	a MIME-compliant mail reader.  Different MIME-compliant mail readers
> 	exhibit different behavior, especially when dealing with
> 	"multipart" MIME messages (i.e. documents which have been split
> 	up into multiple messages), so check your local documentation on
> 	how to manipulate these messages.
> 		
> 		
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Cc: schulzrinne@cs.columbia.edu, sipping@ietf.org, jdrosen@dynamicsoft.com
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [Sipping] I-D ACTION:draft-ietf-sipping-dialog-package-03.txt
Date: Tue, 28 Oct 2003 09:52:01 -0800
To: Diego B <diegob@mailvision.com>
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Good catch.  The roles are backwards in the last sentence.

thanks,
-rohan


On Oct 28, 2003, at 10:16, Diego B wrote:

> Hi;
> This is right?
>
> "
>    Automatic Callback: In this basic Public Switched Telephone Network
>       (PSTN) application, user A calls user B. User B is busy. User A
>       would like to get a callback when user B hangs up. When B hangs
>       up, user A's phone rings. When A picks it up, they here ringing,
>       and are being connected to B. To implement this with SIP, a
>       mechanism is required for B to receive a notification when the
>                                 ^
>       dialogs at A are complete.
>                  ^
> "
> In this example, 'A' may receive notifications on the state of 'B'
> dialog.
>
>
> Tnx
>
> Diego Besprosvan
>
> MailVision Ltd.
> Haganim 10/a
> German Colony, Haifa
> Israel
> e-mail: diegob@mailvision.com
> www.mailvision.com
>
>
> Internet-Drafts@ietf.org wrote:
>
>> A New Internet-Draft is available from the on-line Internet-Drafts  
>> directories.
>> This draft is a work item of the Session Initiation Proposal  
>> Investigation Working Group of the IETF.
>> 	Title		: An INVITE Inititiated Dialog Event Package for the Session  
>> Initiation Protocol (SIP)
>> 	Author(s)	: J. Rosenberg, H. Schulzrinne
>> 	Filename	: draft-ietf-sipping-dialog-package-03.txt
>> 	Pages		: 30
>> 	Date		: 2003-10-27
>> 	
>> This document defines a dialog event package for the SIP Events
>> architecture, along with a data format used in notifications for this
>> package. The dialog package allows users to subscribe to another
>> user, an receive notifications about the changes in state of INVITE
>> initiated dialogs that the user is involved in.
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-ietf-sipping-dialog- 
>> package-03.txt
>> To remove yourself from the IETF Announcement list, send a message to  
>> ietf-announce-request with the word unsubscribe in the body of the  
>> message.
>> Internet-Drafts are also available by anonymous FTP. Login with the  
>> username
>> "anonymous" and a password of your e-mail address. After logging in,
>> type "cd internet-drafts" and then
>> 	"get draft-ietf-sipping-dialog-package-03.txt".
>> A list of Internet-Drafts directories can be found in
>> http://www.ietf.org/shadow.html or  
>> ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>> Internet-Drafts can also be obtained by e-mail.
>> Send a message to:
>> 	mailserv@ietf.org.
>> In the body type:
>> 	"FILE /internet-drafts/draft-ietf-sipping-dialog-package-03.txt".
>> 	
>> NOTE:	The mail server at ietf.org can return the document in
>> 	MIME-encoded form by using the "mpack" utility.  To use this
>> 	feature, insert the command "ENCODING mime" before the "FILE"
>> 	command.  To decode the response(s), you will need "munpack" or
>> 	a MIME-compliant mail reader.  Different MIME-compliant mail readers
>> 	exhibit different behavior, especially when dealing with
>> 	"multipart" MIME messages (i.e. documents which have been split
>> 	up into multiple messages), so check your local documentation on
>> 	how to manipulate these messages.
>> 		
>> 		
>> Below is the data which will enable a MIME compliant mail reader
>> implementation to automatically retrieve the ASCII version of the
>> Internet-Draft.
>


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Tue, 28 Oct 2003 13:46:00 -0500
To: Rohan Mahy <rohan@cisco.com>
From: Michael Hammer <mhammer@cisco.com>
Subject: Re: [Sipping] I-D
  ACTION:draft-ietf-sipping-dialog-package-03.txt
Cc: Diego B <diegob@mailvision.com>, schulzrinne@cs.columbia.edu,
        sipping@ietf.org, jdrosen@dynamicsoft.com
In-Reply-To: <6F381702-096F-11D8-98E9-0003938AF740@cisco.com>
References: <3F9EB279.7000403@mailvision.com>
 <200310281440.JAA24994@ietf.org>
 <3F9EB279.7000403@mailvision.com>
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Also known as Completion of Calls to Busy Subscriber (CCBS), I.253.3?

Mike


At 09:52 AM 10/28/2003 -0800, Rohan Mahy wrote:
>Good catch.  The roles are backwards in the last sentence.
>
>thanks,
>-rohan
>
>
>On Oct 28, 2003, at 10:16, Diego B wrote:
>
>>Hi;
>>This is right?
>>
>>"
>>    Automatic Callback: In this basic Public Switched Telephone Network
>>       (PSTN) application, user A calls user B. User B is busy. User A
>>       would like to get a callback when user B hangs up. When B hangs
>>       up, user A's phone rings. When A picks it up, they here ringing,
>>       and are being connected to B. To implement this with SIP, a
>>       mechanism is required for B to receive a notification when the
>>                                 ^
>>       dialogs at A are complete.
>>                  ^
>>"
>>In this example, 'A' may receive notifications on the state of 'B'
>>dialog.
>>
>>
>>Tnx
>>
>>Diego Besprosvan
>>
>>MailVision Ltd.
>>Haganim 10/a
>>German Colony, Haifa
>>Israel
>>e-mail: diegob@mailvision.com
>>www.mailvision.com
>>
>>
>>Internet-Drafts@ietf.org wrote:
>>
>>>A New Internet-Draft is available from the on-line Internet-Drafts
>>>directories.
>>>This draft is a work item of the Session Initiation Proposal
>>>Investigation Working Group of the IETF.
>>>         Title           : An INVITE Inititiated Dialog Event Package 
>>> for the Session
>>>Initiation Protocol (SIP)
>>>         Author(s)       : J. Rosenberg, H. Schulzrinne
>>>         Filename        : draft-ietf-sipping-dialog-package-03.txt
>>>         Pages           : 30
>>>         Date            : 2003-10-27
>>>
>>>This document defines a dialog event package for the SIP Events
>>>architecture, along with a data format used in notifications for this
>>>package. The dialog package allows users to subscribe to another
>>>user, an receive notifications about the changes in state of INVITE
>>>initiated dialogs that the user is involved in.
>>>A URL for this Internet-Draft is:
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>_______________________________________________
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>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP


_______________________________________________
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Date: Tue, 28 Oct 2003 13:49:51 -0500
From: Paul Kyzivat <pkyzivat@cisco.com>
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Jonathan,

I have a couple of comments about this version, below.

	Paul

Section 5.2:

    To authenticate themselves, it is RECOMMENDED that applications use
    the SIP identity mechanism [11] in the REFER or SUBSCRIBE requests
    they generate. A UA will need to authorize these subscriptions and
    refers. To do this, a UA SHOULD accept any REFER or SUBSCRIBE sent to
    the GRUU it used for that dialog. This would imply that only elements
    privy to the INVITE requests and responses could send a REFER or
    SUBSCRIBE to the UA. The usage of the sips URI scheme provides
    cryptographic assurances that only elements on the call setup path
    could see such information. Therefore, it is RECOMMENDED that UAs
    compliant to this specification use sips whenever possible. A client
    SHOULD use grid parameters with sufficient randomness to eliminate
    the possibility of an attacker guessing the GRUU.

This seems a bit loose. It isn't always going to be practical to use 
sips. (The callee may not support it.) Without it there is danger. What 
is the recommendation in that case?

Section 5.4:

    A client can remove a UI component at any time. For presentation
    aware UI, this is analagous to the user dismissing the web form
    window. There is no mechanism provided for reporting this kind of
    event to the application.

The initial refer created an implicit subscription. It is possible, but 
not required by the event mechanism, for event generation to continue 
for the life of the resulting interface component. If so, a suitable 
event could signal the termination of the interface component.



_______________________________________________
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This list is for NEW development of the application of SIP
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Subject: [Sipping] New conferencing drafts
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Following conferencing drafts appeared. I wonder what should be now the 
relationship between
the work in SIPPING  and XCON regarding conferencing.
I had the impression that the work is moving to XCON. Was it a false 
impression?

http://www.ietf.org/internet-drafts/draft-ietf-sipping-conference-package-02.txt
http://www.ietf.org/internet-drafts/draft-ietf-sipping-cc-conferencing-02.txt

Thanks
Avshalom

--=_alternative 006783EEC2256DCD_=
Content-Type: text/html; charset="US-ASCII"


<br><font size=2 face="sans-serif">Following conferencing drafts appeared.
I wonder what should be now the relationship between</font>
<br><font size=2 face="sans-serif">the work in SIPPING &nbsp;and XCON regarding
conferencing.</font>
<br><font size=2 face="sans-serif">I had the impression that the work is
moving to XCON. Was it a false impression?</font>
<br>
<br><font size=2 face="sans-serif">http://www.ietf.org/internet-drafts/draft-ietf-sipping-conference-package-02.txt</font>
<br><font size=2 face="sans-serif">http://www.ietf.org/internet-drafts/draft-ietf-sipping-cc-conferencing-02.txt</font>
<br>
<br><font size=2 face="sans-serif">Thanks</font>
<br><font size=2 face="sans-serif">Avshalom</font>
<br>
--=_alternative 006783EEC2256DCD_=--

_______________________________________________
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Rohan Mahy'" <rohan@cisco.com>
Cc: "'sipping@ietf.org'" <sipping@ietf.org>
Subject: RE: [Sipping] Brian's Comments on draft-mahy-sipping-user-equals-
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> > The introduction must define dial strings, and differentiate them
> > from local phone numbers.
> 
> I think this is the crux of the disagreement.  I don't think it is 
> possible to give a rigorous definition of what is a dial string and 
> what is a local number.  I invite you to describe an 
> algorithm you can 
> use to determine which is which.  If you can then I'll incorporate it 
> in the draft, but I am skeptical that such a distinction is testable 
> any more than the distinction between "erotic art" and "pornography".
A dial string is a series of digits, and */# characters entered 
directly or indirectly by a user.  They are interpreted in a local 
dial plan, to a local or global number, or a service invocation.  
A local number is an address of an endpoint in a locally defined 
numbering scheme.  It is often the case that a local number is the 
same series of digits in that domain's dial plan, but it is rarely 
true that there is a one to one correspondence, and many dialing 
strings may be translated to global numbers, while others may be 
translated to local numbers, or they may not be addresses of 
endpoints at all -- they can be service invocation indicators. 

> 
> >  It should introduce the idea that a
> > proxy in the listed domain of a dial string must translate 
> to a global
> > or local number, and that some phones will do this translation
> > themselves while others will not.
> >
> > Section 5 inadequately deals with dial strings.  Dial 
> strings are not
> > local numbers, and must be translated to local numbers.  One must
> > disambiguate dial strings from local numbers because it is possible 
> > that
> > they collide.  Such a differentiation is also useful to know when
> > translation has been completed.
> 
> i don't think they can collide because they are in different contexts
Only if you define them separately, and explicitly deal with establishing
separate contexts, as has been proposed.  The text is not explicit
about that.

> 
> > It has been proposed to use separate contexts for dial strings and
> > local numbers.  I generally don't like this idea, but it can be made
> > to work.  It would be best if there was some kind of naming 
> convention
> > to differentiate, but this is not strictly necessary.
> >
> > What is necessary is text in a document:
> > 	1. Saying that dial strings aren't local numbers
> > 	2. Permitting dial strings in a sip uri, with a dial 
> string context.
> > 		Note: if you wish to permit a dial string in a 
> tel uri, then
> >                  the language in 2806bis must be modified, IMO.
> > 	3. Explicitly defining the use of E/F
> > 	4. Illustrating the use of separate contexts for dial strings
> > 	5. Discussing the provisioning implications of having two
> > 		local contexts, one for phones with translation and
> > 		another for phones without translation
> 
> I thought I covered this last one a bit in the draft. A phone that 
> doesn't know its context can just use *itself* as the context.  That 
> requires no configuration.  A phone that has a digit map obviously 
> needs to be configured with it, so explicitly setting the target 
> context or domain (or both) should be no problem IMO.
While I can imagine that many systems would work with "self"
It's not going to work in the general case, because the interpreter
of the digits may be handling multiple domains.

Also, do you think we should leave it ambiguous whether a phone with
a digitmap translates dialstrings to local numbers (ie tel uris) or to
user@domain, where domain might be numeric?  I didn't like the
"context or domain (or both)" part of the statement above.  Too many
options, too many chances for proxy servers to assume something a UA
doesn't assume.

....snip....
> > We also should define a way to express a dial string with 
> pauses and 
> > waits.
> > They are pretty useful. They are dialable in many gateways.  We need
> > a way to send such a string.  Again, we need text, and 
> again, we could
> > use a separate context for such strings.
No comment here?  We need something, somewhere about this.

 

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From exim@www1.ietf.org  Tue Oct 28 15:31:28 2003
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Date: Tue, 28 Oct 2003 14:24:05 -0600
From: Alan Johnston <alan.johnston@mci.com>
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To: Avshalom Houri <AVSHALOM@il.ibm.com>, sipping@ietf.org, xcon@ietf.org
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Hi Avshalom,

Your impression was false.

The SIP-specific conferencing work will remain in the SIP and SIPPING WGs - 
this includes the listed drafts plus the framework document and the 
definition of the "isfocus" feature tag.

All the policy work (media and conference) plus floor control is what will 
be covered in XCON.

Thanks,
Alan Johnston
XCON co-chair

At 08:50 PM 10/28/2003 +0200, Avshalom Houri wrote:

>Following conferencing drafts appeared. I wonder what should be now the 
>relationship between
>the work in SIPPING  and XCON regarding conferencing.
>I had the impression that the work is moving to XCON. Was it a false 
>impression?
>
>http://www.ietf.org/internet-drafts/draft-ietf-sipping-conference-package-02.txt 
>
>http://www.ietf.org/internet-drafts/draft-ietf-sipping-cc-conferencing-02.txt
>
>Thanks
>Avshalom


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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 Proposal Investigation Working Group of the IETF.

	Title		: A Framework for Conferencing with the Session Initiation Protocol
	Author(s)	: J. Rosenberg
	Filename	: draft-ietf-sipping-conferencing-framework-01.txt
	Pages		: 45
	Date		: 2003-10-28
	
The Session Initiation Protocol (SIP) supports the initiation,
modification, and termination of media sessions between user agents.
These sessions are managed by SIP dialogs, which represent a SIP
relationship between a pair of user agents. Because dialogs are
between pairs of user agents, SIP's usage for two-party
communications (such as a phone call), is obvious. Communications
sessions with multiple participants, generally known as conferencing,
are more complicated. This document defines a framework for how such
conferencing can occur. This framework describes the overall
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party conferencing.

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

	Title		: Best Current Practices for Third Party Call Control in the Session Initiation Protocol
	Author(s)	: J. Rosenberg, J. Peterson, H. Schulzrinne, G. Camarillo
	Filename	: draft-ietf-sipping-3pcc-05.txt
	Pages		: 31
	Date		: 2003-10-28
	
Third party call control refers to the ability of one entity to
create a call in which communications is actually between other
parties. Third party call control is possible using the mechanisms
specified within the Session Initiation Protocol (SIP). However,
there are several possible approaches, each with different benefits
and drawbacks. This document discusses best current practices for the
usage of the SIP for third party call control.

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From: "wendy" <xyxie@udtech.com.cn>
To: <sipping@ietf.org>
Date: Wed, 29 Oct 2003 14:41:14 +0800
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Hi,

I meet a trouble when developping a b2bua as a SIP ALG to traverse NAT.
The network topology is like this:

private UA --- SIP ALG --- SIP Server (public)

When the private UA registers to the SIP Server,
the SIP ALG can forward authentication CHALLENGE from the SIP Server to the private UA,
and forward authentication RESPONSE from the private UA to the SIP Server.

In some cases, e.g. the public IP address of the SIP ALG is changed,
I hope the SIP ALG can initiate a registration as a substitute of the private UA.
Thus, the private UA does not need to care the SIP ALG's changes.
But how can the SIP ALG be authenticated successfully,
if the administrator of the SIP ALG does not want to configure manually the
usernames/passwords
of all private users?

Thanks a lot!


regards,
Wendy


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Message-ID: <3F9F6933.3010508@dynamicsoft.com>
Date: Wed, 29 Oct 2003 02:16:03 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Folks,

I have submitted an update to ICE. Based on agreement at the last 
ietf, it is now a work item of MMUSIC. Until the I-D appears in the 
archives, you can pick up a copy at:

http://www.jdrosen.net/papers/draft-ietf-mmusic-ice-00.txt
http://www.jdrosen.net/papers/draft-ietf-mmusic-ice-00.html

There were many changes since the last revision:

* The draft is now SIP and offer/answer independent. The way I 
achieved this was to describe ICE as an abstract protocol, and then 
define mappings from that abstract protocol to concrete ones. A 
mapping is included for SIP, using the SDP alt parameter extension and 
Gonzalo's new IPV group semantic.

* The spec now mandates that a client, in addition to be configured 
with one or more relay servers (i.e., turn), needs to know which of 
those are on the public internet. This would be through configuration. 
This information is needed only for backwards compatibility with 
non-ice clients.

* In an update to TURN, I added a Send mechanism which allows relay of 
outbound requests. ICE now includes usage of this mechanism. Its 
needed for inter-enterprise calls where the media needs to pass 
through two relays, one in each domain.

* I added some text on backwads compatibility processing when the 
caller realizes that the callee doesnt support ICE.

* peer derived transport addresses learned by Sending (using the TURN 
Send primitive) a STUN request through a TURN relay have the same 
priority as turn derived transport addresses.


Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
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@ietf.org for new developments of core SIP



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Date: Wed, 29 Oct 2003 02:23:21 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [Sipping] Updated NAT scenarios document
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Folks,

I've submitted an update to the NAT scenarios specification. This 
change represents a total rewrite to be based entirely on ICE, rather 
than presenting the hodge-podge of solutions that were in the previous 
version. Due to the scope of this change, I submitted as an individual 
I-D. It should appear in the archives shortly. Until it does, you can 
pick up a copy at:

http://www.jdrosen.net/papers/draft-rosenberg-sipping-nat-scenarios-01.txt
http://www.jdrosen.net/papers/draft-rosenberg-sipping-nat-scenarios-01.html

Much of the content in this document comes from the back of the 
previous ICE specification, which included many example flows. The 
call flows for the enterprise cases have been updated. In particular, 
an inter-enterprise case is now shown, where two relays are involved - 
one in each enterprise. This is accomplished through a new feature I 
added into TURN, which is a Send primitive for relaying a packet 
through a TURN server.

The examples also show a network topology for an "advanced enterprise" 
which makes use of two relays - one inside, and one outside, of the 
nat/firewall. Such a configuration has some security benefits and 
avoids the need for any configuration on the nat/fw. Call flows have 
yet to be added for this case though.

Gonzalo also contributed some examples on how ICE facilitates v4/v6 
transition, and those are included as well.

Comments and questions welcome.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com



_______________________________________________
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To: "wendy" <xyxie@udtech.com.cn>
Cc: <sipping@ietf.org>
Reply-To: sip-implementors@cs.columbia.edu
References: <019a01c39de7$a5e684e0$b400a8c0@zwx>
Organization: Pingtel Corp. http://www.pingtel.com/
From: Scott Lawrence <slawrence@pingtel.com>
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 message of "Wed, 29 Oct 2003 14:41:14 +0800")
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"wendy" <xyxie@udtech.com.cn> writes:

> I meet a trouble when developping a b2bua as a SIP ALG to traverse NAT.

 This question would be more appropriately asked on
 sip-implementors@cs.columbia.edu (questions on current sip)

> The network topology is like this:
>
> private UA --- SIP ALG --- SIP Server (public)
>
> When the private UA registers to the SIP Server,
> the SIP ALG can forward authentication CHALLENGE from the SIP Server to the
> private UA,
> and forward authentication RESPONSE from the private UA to the SIP Server.
>
> In some cases, e.g. the public IP address of the SIP ALG is changed,
> I hope the SIP ALG can initiate a registration as a substitute of the
> private UA.
> Thus, the private UA does not need to care the SIP ALG's changes.
> But how can the SIP ALG be authenticated successfully,
> if the administrator of the SIP ALG does not want to configure manually the
> usernames/passwords of all private users?

What changes do you want to make in the ALG that would invalidate the
authentication parameters?

-- 
Scott Lawrence        
  Pingtel Corp.   


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From: "James M. Polk" <jmpolk@cisco.com>
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Subject: [Sipping] New Location Conveyance ID posted
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All

Brian Rosen and I have updated the Location Conveyance ID. It can be found at:

http://www.ietf.org/internet-drafts/draft-polk-sipping-location-requirements-01.txt

The Abstract is as follows:

    This document presents the framework and requirements for an
    extension to the Session Initiation Protocol (SIP) for
    conveyance of user location information from a Session Initiation
    Protocol (SIP) user agent to another SIP entity.  We consider cases
    where location information is conveyed from end to end, as well as
    cases where message routing by intermediaries is influenced by the
    location of the session initiator.

There are a number of open issues listed near the end of the document that 
need to be addressed

Comments are appreciated

cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


_______________________________________________
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This list is for NEW development of the application of SIP
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Subject: RE: [Sipping] Please Read Committee T1 Draft on DTMF Notification
Date: Wed, 29 Oct 2003 22:02:37 -0500
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The KPML update is in the archives; I just haven't seen the =
announcement:
http://www.ietf.org/internet-drafts/draft-ietf-sipping-kpml-01.txt

Things I like in dtmfad:

1. Explicit maxrate (but see below)
2. "Handshook" throttling (although isn't that normal SIP: no =
overlapping transactions?)
3. Very simple and compact
4. Described forking and state agent behavior (I forgot!)


Things I don't like:
1. The model is worse than MGCP.  An event for every key press?  If you =
are interested in such low-level control, use H.248.

2. The model only works for a B2BUA or machine end point.  The far end =
is the only entity that can request digit monitoring.  OK for a =
"softswitch", but not necessarily desirable in an Applications and =
Services Infrastructure.

3. Duration: On the one hand, it is handy to be able to define what =
"long" is.  However, this is a disaster.  What if I set "duration=3D50" =
and I hold the octothorpe key down for 6 seconds?  I will get 120 NOTIFY =
messages in that time.  That isn't helpful for traffic shaping.  What =
does the application do with all of those messages?  The model is taken =
directly from RFC2833 (keep emitting tone and mark the end-of-tone) -- =
why not just use RFC2833 and be done with it?  Eating a low-rate RTP =
stream has GOT to be easier than implementing a RFC3261 stack!

4. Hex codes don't really fit into the SIP gestalt.  Why not encode the =
information in a straightforward manner?  One cannot argue for =
compactness -- if one is going to live with 600+ bytes of SIP headers =
(see the example in the document), one can have a 35-byte text message =
that means something instead of a 12-byte "binary" message that is =
indecipherable.


Things I am open to:
1. Volume: Does anyone need this?  What for?  How would you know?  What =
would you do with it?  Not for falsing detection: that is what the =
detection device should be doing.  If there is a good use case, I could =
see implementing it.  Then again, what do you do for RFC2833 digit =
detection (no volume)?

2. Maxrate: I'm leaning towards not liking it, but I'm open to it.
First, the SUBSCRIBER here only specifies its own ability to process =
NOTIFY requests.  Using this as a network-saving device isn't going to =
work: how does the subscriber have any idea what to set this to?=20
Second, the default maxrate is one message every 16ms, or faster than =
G.711.  What could this be possibly useful for?
Third (and the big one), the whole point of transporting digits in the =
signaling path is because the digits *are* signaling.  Said differently, =
the application needs reliable delivery of signaling (digits).  If the =
use case for this package allows unreliable delivery, then I would be =
very interested in understanding why one is not using RFC2833, which has =
EVERYTHING discussed in this draft, available today?


> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Tue, October 28, 2003 2:27 AM
> To: sipping@ietf.org
> Cc: Joe Zebarth
> Subject: [Sipping] Please Read Committee T1 Draft on DTMF Notification
>=20
>=20
> While we were arguing about how to handle keypresses,=20
> somebody went out
> and wrote a draft for an event package for DTMF events.
>=20
> Please review the draft:
>=20
> http://www.softarmor.com/wgdb/docs/draft-zebarth-sipping-dtmfad-00.txt
>=20
> As I understand it, the mechanism that this draft is meant to=20
> support is
> defined in Committee T1 Technical Report T1.TR.82-2003.=20
> Unfortunately, I
> wasn't able to find said document on the T1 site, but that=20
> could just be
> me -- perhaps the author can give us a link?
>=20
> Questions to consider: Does this approach complement or conflict with
> the KPML work? Is there danger of fragmenting, or is this work clearly
> seperate? Would the world (or at least the Internet) be a better place
> with, or without, this event package?
>=20
> Please copy the author, as well as the list, on any=20
> responses. I expect
> that discussion on this topic will come up during the Minneapolis
> meeting.
>=20
> Thanks,
>=20
> --
> Dean Willis
> SIPPING co-chair
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20
>=20


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Joe Zebarth" <zebarth@nortelnetworks.com>
To: sipping@ietf.org
Cc: Dean Willis <dean.willis@softarmor.com>
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Colleagues

The T1 prepublished document can be found at:

http://www.t1.org/filemgr/filesearch.taf?_function=detail&T1_Files_uid1=3S17
0632

Note that the zip file is 251 k and the uncompressed is 1.4 M

Kindest regards,

Joe Zebarth
Dept. 1A12
ATI Strategic Standards
Nortel Networks (Canada)
Phone: +1 613 765 8481
ESN:    6+395 8481
Fax:     +1 613 763 2697
ESN:    6+393 2697
http://www.nortelnetworks.com
 



-----Original Message-----
From: Dean Willis [mailto:dean.willis@softarmor.com]
Sent: Tuesday, October 28, 2003 2:27 AM
To: sipping@ietf.org
Cc: Zebarth, Joe [CAR:1A12:EXCH]
Subject: Please Read Committee T1 Draft on DTMF Notification


While we were arguing about how to handle keypresses, somebody went out
and wrote a draft for an event package for DTMF events.

Please review the draft:

http://www.softarmor.com/wgdb/docs/draft-zebarth-sipping-dtmfad-00.txt

As I understand it, the mechanism that this draft is meant to support is
defined in Committee T1 Technical Report T1.TR.82-2003. Unfortunately, I
wasn't able to find said document on the T1 site, but that could just be
me -- perhaps the author can give us a link?

Questions to consider: Does this approach complement or conflict with
the KPML work? Is there danger of fragmenting, or is this work clearly
seperate? Would the world (or at least the Internet) be a better place
with, or without, this event package?

Please copy the author, as well as the list, on any responses. I expect
that discussion on this topic will come up during the Minneapolis
meeting.

Thanks,

--
Dean Willis
SIPPING co-chair

------_=_NextPart_001_01C39D5C.863BC744
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charset=3Diso-8859-1">
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<TITLE>RE: Please Read Committee T1 Draft on DTMF Notification</TITLE>
</HEAD>
<BODY>

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

<P><FONT SIZE=3D2>The T1 prepublished document can be found at:</FONT>
</P>

<P><FONT SIZE=3D2><A =
HREF=3D"http://www.t1.org/filemgr/filesearch.taf?_function=3Ddetail&T1_F=
iles_uid1=3D3S170632" =
TARGET=3D"_blank">http://www.t1.org/filemgr/filesearch.taf?_function=3Dd=
etail&T1_Files_uid1=3D3S170632</A></FONT>
</P>

<P><FONT SIZE=3D2>Note that the zip file is 251 k and the uncompressed =
is 1.4 M</FONT>
</P>

<P><FONT SIZE=3D2>Kindest regards,</FONT>
</P>

<P><FONT SIZE=3D2>Joe Zebarth</FONT>
<BR><FONT SIZE=3D2>Dept. 1A12</FONT>
<BR><FONT SIZE=3D2>ATI Strategic Standards</FONT>
<BR><FONT SIZE=3D2>Nortel Networks (Canada)</FONT>
<BR><FONT SIZE=3D2>Phone: +1 613 765 8481</FONT>
<BR><FONT SIZE=3D2>ESN:&nbsp;&nbsp;&nbsp; 6+395 8481</FONT>
<BR><FONT SIZE=3D2>Fax:&nbsp;&nbsp;&nbsp;&nbsp; +1 613 763 2697</FONT>
<BR><FONT SIZE=3D2>ESN:&nbsp;&nbsp;&nbsp; 6+393 2697</FONT>
<BR><FONT SIZE=3D2><A HREF=3D"http://www.nortelnetworks.com" =
TARGET=3D"_blank">http://www.nortelnetworks.com</A></FONT>
<BR><FONT SIZE=3D2>&nbsp;</FONT>
</P>
<BR>
<BR>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Dean Willis [<A =
HREF=3D"mailto:dean.willis@softarmor.com">mailto:dean.willis@softarmor.c=
om</A>]</FONT>
<BR><FONT SIZE=3D2>Sent: Tuesday, October 28, 2003 2:27 AM</FONT>
<BR><FONT SIZE=3D2>To: sipping@ietf.org</FONT>
<BR><FONT SIZE=3D2>Cc: Zebarth, Joe [CAR:1A12:EXCH]</FONT>
<BR><FONT SIZE=3D2>Subject: Please Read Committee T1 Draft on DTMF =
Notification</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>While we were arguing about how to handle keypresses, =
somebody went out</FONT>
<BR><FONT SIZE=3D2>and wrote a draft for an event package for DTMF =
events.</FONT>
</P>

<P><FONT SIZE=3D2>Please review the draft:</FONT>
</P>

<P><FONT SIZE=3D2><A =
HREF=3D"http://www.softarmor.com/wgdb/docs/draft-zebarth-sipping-dtmfad-=
00.txt" =
TARGET=3D"_blank">http://www.softarmor.com/wgdb/docs/draft-zebarth-sippi=
ng-dtmfad-00.txt</A></FONT>
</P>

<P><FONT SIZE=3D2>As I understand it, the mechanism that this draft is =
meant to support is</FONT>
<BR><FONT SIZE=3D2>defined in Committee T1 Technical Report =
T1.TR.82-2003. Unfortunately, I</FONT>
<BR><FONT SIZE=3D2>wasn't able to find said document on the T1 site, =
but that could just be</FONT>
<BR><FONT SIZE=3D2>me -- perhaps the author can give us a link?</FONT>
</P>

<P><FONT SIZE=3D2>Questions to consider: Does this approach complement =
or conflict with</FONT>
<BR><FONT SIZE=3D2>the KPML work? Is there danger of fragmenting, or is =
this work clearly</FONT>
<BR><FONT SIZE=3D2>seperate? Would the world (or at least the Internet) =
be a better place</FONT>
<BR><FONT SIZE=3D2>with, or without, this event package?</FONT>
</P>

<P><FONT SIZE=3D2>Please copy the author, as well as the list, on any =
responses. I expect</FONT>
<BR><FONT SIZE=3D2>that discussion on this topic will come up during =
the Minneapolis</FONT>
<BR><FONT SIZE=3D2>meeting.</FONT>
</P>

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

<P><FONT SIZE=3D2>--</FONT>
<BR><FONT SIZE=3D2>Dean Willis</FONT>
<BR><FONT SIZE=3D2>SIPPING co-chair</FONT>
</P>

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</HTML>
------_=_NextPart_001_01C39D5C.863BC744--

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Thu Oct 30 09:47:34 2003
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Rupert Kruger (RC)'" <Krugerrc@telkom.co.za>,
        "'sipping@ietf.org'"
	 <sipping@ietf.org>
Date: Thu, 30 Oct 2003 09:45:39 -0500
MIME-Version: 1.0
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Content-Type: text/plain;
	charset="iso-8859-1"
Subject: [Sipping] RE: [Sip] Basic question
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<I have redirected this message to sipping, where the sip elements of the
conferencing work is being done>

Please have a look at
http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-conferencing-framework
-01.txt
http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-cc-conferencing-02.txt
and
http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-conference-package-02.
txt

In general, the sip conferencing model currently being specified encompasses
mixing in the end devices as well as centralized mixing.  From a signalling
standpoint, both of these mechanisms look the same - each participant will
have
a simple 1on1 call between itself and a "focus".  The focus will arrange the
media streams to be sent to the mixer.  Thus, the focus appears to be a 
coordinated set of sip User Agents, one for each participant.  This is
somewhat similar to an H.323 MCU, but there are model differences.

The conference is referenced by a conference URI.  This URI may be
registered
in one way or another with a Registrar so that a call to that URI reaches
the
focus, which is one way to join a conference.

Please also note that there is a new work group, xcon, which is working on
the policy and media aspects of this work.

Brian


> -----Original Message-----
> From: Rupert Kruger (RC) [mailto:Krugerrc@telkom.co.za]
> Sent: Thursday, October 30, 2003 5:02 AM
> To: sip@ietf.org
> Subject: [Sip] Basic question
> 
> 
> Hi,
> 
> I'm fairly new to SIP, and need help: For multipoint video
> conferencing, in H.323, one had the MCU that registers with the
> gatekeeper. But for SIP, there are only three basic servers: Proxy,
> Redirect, and Registrar - so I cannot see where a SIP "MCU" would come
> in. Will it be some sort of server, or a UA that has multiple
> connections? What does the architecture look like for a conference
> with 3 or more SIP end-users?
> 
> Any help would be greatly appreciated.
> 
> Rupert Kruger
> 
> _______________________________________________
> 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
> 

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Thu, 30 Oct 2003 16:03:39 +0000
From: "Elwell, John" <john.elwell@siemens.com>
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This message is in MIME format. Since your mail reader does not understand
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This file now appears in the archives but I have not seen an announcement.

http://www.ietf.org/internet-drafts/draft-elwell-sipping-qsig-tunnel-01.txt
<http://www.ietf.org/internet-drafts/draft-elwell-sipping-qsig-tunnel-01.txt
> 

Abstract: This document specifies tunnelling of "QSIG" over the Session
Initiation Protocol (SIP). This enables calls between "islands" of circuit
switched networks that use QSIG signalling to be interconnected by an IP
network that uses SIP signalling without loss of QSIG functionality.
Compared with version 00 it uses a different technique for ensuring that the
egress UA supports QSIG tunnelling. Instead of defining a new option tag it
makes use of the MIME Content-Disposition header containing
"handling-required" for this purpose. This means that no new protocol is
defined.
John (john.elwell@siemens.com)


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<P><FONT SIZE=2 FACE="Arial">This file now appears in the archives but I have not seen an announcement.</FONT>
</P>

<P><A HREF="http://www.ietf.org/internet-drafts/draft-elwell-sipping-qsig-tunnel-01.txt"><U><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">http://www.ietf.org/internet-drafts/draft-elwell-sipping-qsig-tunnel-01.txt</FONT></U></A>
</P>

<P><FONT SIZE=2 FACE="Arial">Abstract: This document specifies tunnelling of &quot;QSIG&quot; over the Session Initiation Protocol (SIP). This enables calls between &quot;islands&quot; of circuit switched networks that use QSIG signalling to be interconnected by an IP network that uses SIP signalling without loss of QSIG functionality.</FONT></P>

<P><FONT SIZE=2 FACE="Arial">Compared with version 00 it uses a different technique for ensuring that the egress UA supports QSIG tunnelling. Instead of defining a new option tag it makes use of the MIME Content-Disposition header containing &quot;handling-required&quot; for this purpose. This means that no new protocol is defined.</FONT></P>

<P><FONT SIZE=2 FACE="Arial">John (john.elwell@siemens.com)</FONT>
</P>

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--Boundary_(ID_2SIYmRIwpjRyRuLBuWz3BQ)--

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Dean Willis" <dwillis@dynamicsoft.com>
To: <sip@ietf.org>, <sipping@ietf.org>, <simple@ietf.org>, <xcon@ietf.org>,
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1:30 PM USCST

The main server (www.softarmor.com) will be down for the next few hours
while I replace the OS.

In the interim, if you need web content, it's available at
http://kevlar.softarmor.com/

Anybody using the main box as a relay for email will also be affected.

Cheers,

--
Dean



_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: RE: [Sipping] RE: Please Read Committee T1 Draft on DTMF Notification
Date: Thu, 30 Oct 2003 22:34:28 -0500
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Thread-Topic: [Sipping] RE: Please Read Committee T1 Draft on DTMF Notification
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Thanks for the posting.

Reviewing the document, I think the T1 community may be better served by =
KPML.  It handles the long # issue, as well as directly handles the #9# =
example given.  Moreover, it is a general solution, while the T1 =
suggestion does not really work even for the pre-paid scenario (the =
NOTIFY flooding and ignoring expires on SUBSCRIBE).


-----Original Message-----
From: Joe Zebarth [mailto:zebarth@nortelnetworks.com]
Sent: Tue, October 28, 2003 9:05 AM
To: sipping@ietf.org
Cc: Dean Willis
Subject: [Sipping] RE: Please Read Committee T1 Draft on DTMF =
Notification


Colleagues=20
The T1 prepublished document can be found at:=20
http://www.t1.org/filemgr/filesearch.taf?_function=3Ddetail&T1_Files_uid1=
=3D3S170632=20
Note that the zip file is 251 k and the uncompressed is 1.4 M=20
Kindest regards,=20
Joe Zebarth=20
Dept. 1A12=20
ATI Strategic Standards=20
Nortel Networks (Canada)=20
Phone: +1 613 765 8481=20
ESN:    6+395 8481=20
Fax:     +1 613 763 2697=20
ESN:    6+393 2697=20
http://www.nortelnetworks.com=20
 =20



-----Original Message-----=20
From: Dean Willis [mailto:dean.willis@softarmor.com]=20
Sent: Tuesday, October 28, 2003 2:27 AM=20
To: sipping@ietf.org=20
Cc: Zebarth, Joe [CAR:1A12:EXCH]=20
Subject: Please Read Committee T1 Draft on DTMF Notification=20


While we were arguing about how to handle keypresses, somebody went out=20
and wrote a draft for an event package for DTMF events.=20
Please review the draft:=20
http://www.softarmor.com/wgdb/docs/draft-zebarth-sipping-dtmfad-00.txt=20
As I understand it, the mechanism that this draft is meant to support is =

defined in Committee T1 Technical Report T1.TR.82-2003. Unfortunately, I =

wasn't able to find said document on the T1 site, but that could just be =

me -- perhaps the author can give us a link?=20
Questions to consider: Does this approach complement or conflict with=20
the KPML work? Is there danger of fragmenting, or is this work clearly=20
seperate? Would the world (or at least the Internet) be a better place=20
with, or without, this event package?=20
Please copy the author, as well as the list, on any responses. I expect=20
that discussion on this topic will come up during the Minneapolis=20
meeting.=20
Thanks,=20
--=20
Dean Willis=20
SIPPING co-chair=20


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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Dean Willis wrote:
> 1:30 PM USCST
> 
> The main server (www.softarmor.com) will be down for the next few hours
> while I replace the OS.
> 
> In the interim, if you need web content, it's available at
> http://kevlar.softarmor.com/
> 
> Anybody using the main box as a relay for email will also be affected.
> 
> Cheers,
> 
> --
> Dean

Everything should be back in working order, I think. Maybe. Note that 
everything has been rekeyed.

--
Dean


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



