
From rjsparks@nostrum.com  Tue Aug  2 11:50:19 2011
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Subject: [sipcore] Table2/3 maintenance (was Re: [Sip] [Technical Errata Reported] RFC3261 (2910))
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(removing the RFC Editor, and setting reply-to to the sipcore list)

Some thing to think about when considering errata on the existing Tables =
2 and 3 in RFC3261:

We have had many conversations about the utility of maintaining tables 2 =
and 3.
I think the most recent was around the Anaheim meeting - see the notes =
at:
<http://www.ietf.org/proceedings/77/minutes/sipcore.html>

We had strong support for not continuing to add information to Tables 2 =
and 3, but not
to remove the existing tables from the document.  Maintaining the =
existing table is
probably a separate question, but keep the arguments noted in the =
minutes above in
mind when choosing to do so.

RjS

On Aug 2, 2011, at 1:28 PM, I=F1aki Baz Castillo wrote:

> 2011/8/2 Bob Penfield <BPenfield@acmepacket.com>:
>> The entry in the table should be "c" (not "m"). A Contact is not =
required in a 100 Trying response. The Contact is only required for a =
1xx that creates a dialog.
>=20
> Right.
>=20
> --=20
> I=F1aki Baz Castillo
> <ibc@aliax.net>


From rjsparks@nostrum.com  Tue Aug  2 11:58:47 2011
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Subject: Re: [sipcore] [Sip] [Technical Errata Reported] RFC3261 (2910)
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(Attempting to move this to the sipcore mailing list)

Further, they're only going to make sense for 1xx that is sent using =
100rel.

So the errata as submitted is incorrect. We could just reject it and =
move on (my preference given the way we're handling Table 2),
or try to edit it to be more correct and put it in hold-for-document =
update (see =
<http://www.ietf.org/iesg/statement/errata-processing.html>).

RjS

On Aug 2, 2011, at 10:58 AM, Bob Penfield wrote:

> The entry in the table should be "c" (not "m"). A Contact is not =
required in a 100 Trying response. The Contact is only required for a =
1xx that creates a dialog.
>=20
>=20
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
RFC Errata System
> Sent: Tuesday, August 02, 2011 10:54 AM
> To: jdrosen@dynamicsoft.com; schulzrinne@cs.columbia.edu; =
Gonzalo.Camarillo@ericsson.com; alan.johnston@wcom.com; =
jon.peterson@neustar.com; rsparks@dynamicsoft.com; mjh@icir.org; =
schooler@research.att.com; gonzalo.camarillo@ericsson.com; =
rjsparks@nostrum.com; dean.willis@softarmor.com; =
drage@alcatel-lucent.com
> Cc: sip@ietf.org; rfc-editor@rfc-editor.org
> Subject: [Sip] [Technical Errata Reported] RFC3261 (2910)
>=20
>=20
> The following errata report has been submitted for RFC3261,
> "SIP: Session Initiation Protocol".
>=20
> --------------------------------------
> You may review the report below and at:
> http://www.rfc-editor.org/errata_search.php?rfc=3D3261&eid=3D2910
>=20
> --------------------------------------
> Type: Technical
> Reported by: I=F1aki Baz Castillo <ibc@aliax.net>
>=20
> Section: Table 2
>=20
> Original Text
> -------------
>      Header field          where   proxy ACK BYE CAN INV OPT REG
>      ___________________________________________________________
>      Contact                1xx           -   -   -   o   -   -
>=20
> Corrected Text
> --------------
>      Header field          where   proxy ACK BYE CAN INV OPT REG
>      ___________________________________________________________
>      Contact                1xx           -   -   -   m   -   -
>=20
> Notes
> -----
> RFC 3261 says:
>=20
> Section 12.1: "Dialogs are created through the generation of =
non-failure responses to requests with specific methods.  Within this =
specification, only 2xx and 101-199 responses with a To tag, where the =
request was INVITE, will establish a dialog."
>=20
> Section 12.1.1: "When a UAS responds to a request with a response that =
establishes a dialog (such as a 2xx to INVITE), the UAS MUST copy all =
Record-Route header field values from the request into the response =
[...].  The UAS MUST add a Contact header field to the response."
>=20
> So it's clear that a 1xx response to an INVITE creates a dialog and =
then it MUST contain a Contact header and mirrored Record-Route headers.
>=20
> However Table 2 (page 162) says:
>=20
>      Header field          where   proxy ACK BYE CAN INV OPT REG
>      ___________________________________________________________
>      Contact                1xx           -   -   -   o   -   -
>      Record-Route        2xx,18x    mr    -   o   o   o   o   -
>=20
> Obviously Record-Route is optional since in the absence of a proxy =
doing record-routing, such header will not be present. However Contact =
header should appear as mandatory (m) for 1xx responses for INVITE =
rather than optional (o).
>=20
> Instructions:
> -------------
> This errata is currently posted as "Reported". If necessary, please
> use "Reply All" to discuss whether it should be verified or
> rejected. When a decision is reached, the verifying party (IESG)
> can log in to change the status and edit the report, if necessary.=20
>=20
> --------------------------------------
> RFC3261 (draft-ietf-sip-rfc2543bis-09)
> --------------------------------------
> Title               : SIP: Session Initiation Protocol
> Publication Date    : June 2002
> Author(s)           : J. Rosenberg, H. Schulzrinne, G. Camarillo, A. =
Johnston, J. Peterson, R. Sparks, M. Handley, E. Schooler
> Category            : PROPOSED STANDARD
> Source              : Session Initiation Protocol
> Area                : Real-time Applications and Infrastructure
> Stream              : IETF
> Verifying Party     : IESG


From ibc@aliax.net  Tue Aug  2 15:24:22 2011
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Subject: Re: [sipcore] [Sip] [Technical Errata Reported] RFC3261 (2910)
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2011/8/2 Robert Sparks <rjsparks@nostrum.com>:
> Further, they're only going to make sense for 1xx that is sent using 100r=
el.

This has been discussed in sip-implementors, and that assertion seems
incorrect. As I've reported in the errata:


Section 12.1: "Dialogs are created through the generation of
non-failure responses to requests with specific methods. Within this
specification, only 2xx and 101-199 responses with a To tag, where the
request was INVITE, will establish a dialog."

Section 12.1.1: "When a UAS responds to a request with a response that
establishes a dialog (such as a 2xx to INVITE), the UAS MUST copy all
Record-Route header field values from the request into the response
[...]. The UAS MUST add a Contact header field to the response."

So it's clear that a 1xx response to an INVITE creates a dialog and
then it MUST contain a Contact header and mirrored Record-Route
headers, *regardless* the usage of 100rel.

Am I wrong? if so, why?

Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From ibc@aliax.net  Tue Aug  2 15:56:09 2011
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Subject: Re: [sipcore] Table2/3 maintenance (was Re: [Sip] [Technical Errata Reported] RFC3261 (2910))
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2011/8/2 Robert Sparks <rjsparks@nostrum.com>:
> (removing the RFC Editor, and setting reply-to to the sipcore list)
>
> Some thing to think about when considering errata on the existing Tables =
2 and 3 in RFC3261:
>
> We have had many conversations about the utility of maintaining tables 2 =
and 3.
> I think the most recent was around the Anaheim meeting - see the notes at=
:
> <http://www.ietf.org/proceedings/77/minutes/sipcore.html>
>
> We had strong support for not continuing to add information to Tables 2 a=
nd 3, but not
> to remove the existing tables from the document. =C2=A0Maintaining the ex=
isting table is
> probably a separate question, but keep the arguments noted in the minutes=
 above in
> mind when choosing to do so.

I also think that table 2 (at least table 2) is not useful, and just
adds complexity. It's much better to explain/describe/mandate the
presence of headers in the text (IMHO).

Regards.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [sipcore] [Sip] [Technical Errata Reported] RFC3261 (2910)
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(removing the rfc-editor and trimming the distribution to the lists)

On Aug 2, 2011, at 5:24 PM, I=F1aki Baz Castillo wrote:

> 2011/8/2 Robert Sparks <rjsparks@nostrum.com>:
>> Further, they're only going to make sense for 1xx that is sent using =
100rel.
>=20
> This has been discussed in sip-implementors, and that assertion seems
> incorrect. As I've reported in the errata:
>=20
>=20
> Section 12.1: "Dialogs are created through the generation of
> non-failure responses to requests with specific methods. Within this
> specification, only 2xx and 101-199 responses with a To tag, where the
> request was INVITE, will establish a dialog."
>=20
> Section 12.1.1: "When a UAS responds to a request with a response that
> establishes a dialog (such as a 2xx to INVITE), the UAS MUST copy all
> Record-Route header field values from the request into the response
> [...]. The UAS MUST add a Contact header field to the response."
>=20
> So it's clear that a 1xx response to an INVITE creates a dialog and
> then it MUST contain a Contact header and mirrored Record-Route
> headers, *regardless* the usage of 100rel.
>=20
> Am I wrong? if so, why?

Not wrong, just incomplete. This will create an (early) dialog at the =
UAS.
It may or may not create a dialog at the UAC without 100rel since the
message may never get to the UAC. Where I said "make sense" above,
it might have been better if I had said "be useful".

>=20
> Regards.
>=20
>=20
> --=20
> I=F1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From pkyzivat@alum.mit.edu  Wed Aug  3 10:44:34 2011
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Subject: Re: [sipcore] [Sip] [Technical Errata Reported] RFC3261 (2910)
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On 8/3/11 11:46 AM, Robert Sparks wrote:
> (removing the rfc-editor and trimming the distribution to the lists)
>
> On Aug 2, 2011, at 5:24 PM, Iñaki Baz Castillo wrote:
>
>> 2011/8/2 Robert Sparks<rjsparks@nostrum.com>:
>>> Further, they're only going to make sense for 1xx that is sent using 100rel.
>>
>> This has been discussed in sip-implementors, and that assertion seems
>> incorrect. As I've reported in the errata:
>>
>>
>> Section 12.1: "Dialogs are created through the generation of
>> non-failure responses to requests with specific methods. Within this
>> specification, only 2xx and 101-199 responses with a To tag, where the
>> request was INVITE, will establish a dialog."
>>
>> Section 12.1.1: "When a UAS responds to a request with a response that
>> establishes a dialog (such as a 2xx to INVITE), the UAS MUST copy all
>> Record-Route header field values from the request into the response
>> [...]. The UAS MUST add a Contact header field to the response."
>>
>> So it's clear that a 1xx response to an INVITE creates a dialog and
>> then it MUST contain a Contact header and mirrored Record-Route
>> headers, *regardless* the usage of 100rel.
>>
>> Am I wrong? if so, why?
>
> Not wrong, just incomplete. This will create an (early) dialog at the UAS.
> It may or may not create a dialog at the UAC without 100rel since the
> message may never get to the UAC. Where I said "make sense" above,
> it might have been better if I had said "be useful".

I don't get your point Robert.

Of course its true that this will only create an early dialog at the UAC 
if the UAC actually receives the (unreliable) response. If it doesn't 
receive it, it won't know of the dialog, and so won't try to use BYE to 
terminate the dialog. But if it does receive it, and does know of the 
dialog, it can then use BYE to terminate it. And in that case the UAS 
will concur that there is a dialog and so be able to properly process 
the BYE.

	Thanks,
	Paul

>> Regards.
>>
>>
>> --
>> Iñaki Baz Castillo
>> <ibc@aliax.net>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


From christer.holmberg@ericsson.com  Wed Aug  3 15:05:41 2011
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 4 Aug 2011 00:05:45 +0200
Thread-Topic: [sipcore] [Sip] [Technical Errata Reported] RFC3261 (2910)
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Subject: Re: [sipcore] [Sip] [Technical Errata Reported] RFC3261 (2910)
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Hi,

I agree with Robert's previous comment regarding the decision we previously=
 made regarding maintaining the tables, but maybe something has to be done.

Because, especially regarding this specific case (Contact in responses), I =
have been involved in a number of discussions where some people argue that =
Contact must always be included, "because the RFC says so" (there seems to =
be products that will trigger an error if not always included), while other=
 say it only needs to be included "when it makes sense".

Nothing an SBC shouldn't be able to fix, though ;)

I also agree with Paul. And, the UAC can not only use the early dialog for =
BYE - it can also use it for other mid-dialog requests (it can't send a new=
 SDP offer, though, as it hasn't received a "real" SDP answer), and once th=
e UAS gets that mid-dialog request it knows that the UAC received the unrel=
iable response, so now IT can start sending mid-dialog requests.=20

Regards,

Christer




=20

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: 3. elokuuta 2011 20:44
> To: sipcore@ietf.org
> Subject: Re: [sipcore] [Sip] [Technical Errata Reported]=20
> RFC3261 (2910)
>=20
> On 8/3/11 11:46 AM, Robert Sparks wrote:
> > (removing the rfc-editor and trimming the distribution to the lists)
> >
> > On Aug 2, 2011, at 5:24 PM, I=F1aki Baz Castillo wrote:
> >
> >> 2011/8/2 Robert Sparks<rjsparks@nostrum.com>:
> >>> Further, they're only going to make sense for 1xx that is=20
> sent using 100rel.
> >>
> >> This has been discussed in sip-implementors, and that=20
> assertion seems=20
> >> incorrect. As I've reported in the errata:
> >>
> >>
> >> Section 12.1: "Dialogs are created through the generation of=20
> >> non-failure responses to requests with specific methods.=20
> Within this=20
> >> specification, only 2xx and 101-199 responses with a To tag, where=20
> >> the request was INVITE, will establish a dialog."
> >>
> >> Section 12.1.1: "When a UAS responds to a request with a response=20
> >> that establishes a dialog (such as a 2xx to INVITE), the UAS MUST=20
> >> copy all Record-Route header field values from the request=20
> into the=20
> >> response [...]. The UAS MUST add a Contact header field to=20
> the response."
> >>
> >> So it's clear that a 1xx response to an INVITE creates a=20
> dialog and=20
> >> then it MUST contain a Contact header and mirrored Record-Route=20
> >> headers, *regardless* the usage of 100rel.
> >>
> >> Am I wrong? if so, why?
> >
> > Not wrong, just incomplete. This will create an (early)=20
> dialog at the UAS.
> > It may or may not create a dialog at the UAC without 100rel=20
> since the=20
> > message may never get to the UAC. Where I said "make sense"=20
> above, it=20
> > might have been better if I had said "be useful".
>=20
> I don't get your point Robert.
>=20
> Of course its true that this will only create an early dialog=20
> at the UAC if the UAC actually receives the (unreliable)=20
> response. If it doesn't receive it, it won't know of the=20
> dialog, and so won't try to use BYE to terminate the dialog.=20
> But if it does receive it, and does know of the dialog, it=20
> can then use BYE to terminate it. And in that case the UAS=20
> will concur that there is a dialog and so be able to properly=20
> process the BYE.
>=20
> 	Thanks,
> 	Paul
>=20
> >> Regards.
> >>
> >>
> >> --
> >> I=F1aki Baz Castillo
> >> <ibc@aliax.net>
> >> _______________________________________________
> >> sipcore mailing list
> >> sipcore@ietf.org
> >> https://www.ietf.org/mailman/listinfo/sipcore
> >
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From romel.khan@idt.net  Thu Aug  4 12:10:15 2011
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From: Romel Khan <romel.khan@idt.net>
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Subject: Re: [sipcore] [Sip]  [Technical Errata Reported] RFC3261 (2910)
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So it is useful if one of UAS or UAC requires it, but it does not have to b=
e
mandatory. Some comments:
-- RFC3261 mentions early dialog without mentioning RFC3262. Then it seems
logical to me that it needs to be made clear in this RFC3261 that early
dialog must mean Contact and Record-Route (if Record-Route was received in
INVITE) headers is mandatory in 1xx without reference to 100rel.

-- A UAS could always send 1xx with headers that are required for early
dialog but it doesn't have to enforce 100rel (eg because the origination or
UAS side itself may not support reliable provisional response handling, or
reliable provisioning not really required for its operation). UAS could sen=
d
"support:100rel" if it supports it, or it would not send it if it doesn't
support this. In my opinion, if UAC hasn't sent 100rel required, it should
be up to the UAS to decide whether to enforce 100rel
(with "required:100rel") if its application really requires SIP requests
before call answer. If the origination side (UAC) side has a need to send
early requests, like UPDATE, then the UAC should require 100rel from the
termination side (UAS) by sending this in INVITE. In a VoIP service provide=
r
world, these kind of capabilities are configured during interconnect turn
up.

-- I notice that some vendors gateway implementations, even if gateway is
the termination side, require 100rel for the gateway to receive pre-answer
requests such as UPDATE. This really didn't have to be this way. I have
always seen these gateways, when it is the termination side, respond back
SIP 183 with the headers that create early dialog. So if the origination
side received the SIP 183 response, then there is no reason for the
origination side to now not be able to send UPDATE request. Also, no
reason for the termination gateways to not accept the SIP UPDATE without
requiring PRACK.

Thanks.

On Wed, Aug 3, 2011 at 11:46 AM, Robert Sparks <rjsparks@nostrum.com> wrote=
:

> (removing the rfc-editor and trimming the distribution to the lists)
>
> On Aug 2, 2011, at 5:24 PM, I=F1aki Baz Castillo wrote:
>
> > 2011/8/2 Robert Sparks <rjsparks@nostrum.com>:
> >> Further, they're only going to make sense for 1xx that is sent using
> 100rel.
> >
> > This has been discussed in sip-implementors, and that assertion seems
> > incorrect. As I've reported in the errata:
> >
> >
> > Section 12.1: "Dialogs are created through the generation of
> > non-failure responses to requests with specific methods. Within this
> > specification, only 2xx and 101-199 responses with a To tag, where the
> > request was INVITE, will establish a dialog."
> >
> > Section 12.1.1: "When a UAS responds to a request with a response that
> > establishes a dialog (such as a 2xx to INVITE), the UAS MUST copy all
> > Record-Route header field values from the request into the response
> > [...]. The UAS MUST add a Contact header field to the response."
> >
> > So it's clear that a 1xx response to an INVITE creates a dialog and
> > then it MUST contain a Contact header and mirrored Record-Route
> > headers, *regardless* the usage of 100rel.
> >
> > Am I wrong? if so, why?
>
> Not wrong, just incomplete. This will create an (early) dialog at the UAS=
.
> It may or may not create a dialog at the UAC without 100rel since the
> message may never get to the UAC. Where I said "make sense" above,
> it might have been better if I had said "be useful".
>
> >
> > Regards.
> >
> >
> > --
> > I=F1aki Baz Castillo
> > <ibc@aliax.net>
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is essentially closed and only used for finishing old business.
> Use sip-implementors@cs.columbia.edu for questions on how to develop a SI=
P
> implementation.
> Use dispatch@ietf.org for new developments on the application of sip.
> Use sipcore@ietf.org for issues related to maintenance of the core SIP
> specifications.
>

--90e6ba6135a8a9886e04a9b2b804
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

So it is useful if one of UAS or UAC requires it, but it does not have to b=
e mandatory. Some comments:<div>-- RFC3261 mentions early dialog without me=
ntioning RFC3262. Then it seems logical to me that it needs to be made clea=
r in this RFC3261 that early dialog must mean Contact and Record-Route (if=
=A0Record-Route was received in INVITE) headers is mandatory in 1xx without=
 reference to 100rel.</div>

<div><br></div><div><div>-- A UAS could always send 1xx with headers that a=
re required for early dialog but it doesn&#39;t have to enforce 100rel (eg =
because the origination or UAS side itself may not support reliable provisi=
onal response handling, or reliable provisioning not really required for it=
s operation).=A0UAS could send &quot;support:100rel&quot; if it supports it=
,=A0or it would not send it if it doesn&#39;t support this.=A0In my opinion=
,=A0if UAC hasn&#39;t sent 100rel required,=A0it should be up to the UAS to=
 decide whether to enforce 100rel (with=A0&quot;required:100rel&quot;)=A0if=
 its application really requires SIP requests before call answer. If the or=
igination side (UAC) side has a need to send early requests, like UPDATE, t=
hen the UAC should require 100rel from the termination side (UAS) by sendin=
g this in INVITE.=A0In a VoIP service provider world,=A0these kind of capab=
ilities are configured during interconnect turn up. =A0</div>

<div><br></div><div>-- I notice that some vendors gateway implementations, =
even if gateway is the termination side, require 100rel for the gateway to =
receive pre-answer requests such as UPDATE. This really didn&#39;t have to =
be this way. I have always seen these gateways, when it is the termination =
side, respond back SIP 183 with the headers that create early dialog. So if=
 the origination side received the SIP 183 response, then there is no reaso=
n for the origination side to now not be able to send UPDATE request. Also,=
=A0no reason=A0for the termination gateways to not accept the SIP UPDATE wi=
thout requiring PRACK. =A0=A0</div>

<div><div><br></div><div>Thanks.</div><div><br><div class=3D"gmail_quote">O=
n Wed, Aug 3, 2011 at 11:46 AM, Robert Sparks <span dir=3D"ltr">&lt;<a href=
=3D"mailto:rjsparks@nostrum.com">rjsparks@nostrum.com</a>&gt;</span> wrote:=
<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">(removing the rfc-editor and trimming the d=
istribution to the lists)<br>
<div class=3D"im"><br>
On Aug 2, 2011, at 5:24 PM, I=F1aki Baz Castillo wrote:<br>
<br>
&gt; 2011/8/2 Robert Sparks &lt;<a href=3D"mailto:rjsparks@nostrum.com">rjs=
parks@nostrum.com</a>&gt;:<br>
&gt;&gt; Further, they&#39;re only going to make sense for 1xx that is sent=
 using 100rel.<br>
&gt;<br>
&gt; This has been discussed in sip-implementors, and that assertion seems<=
br>
&gt; incorrect. As I&#39;ve reported in the errata:<br>
&gt;<br>
&gt;<br>
&gt; Section 12.1: &quot;Dialogs are created through the generation of<br>
&gt; non-failure responses to requests with specific methods. Within this<b=
r>
&gt; specification, only 2xx and 101-199 responses with a To tag, where the=
<br>
&gt; request was INVITE, will establish a dialog.&quot;<br>
&gt;<br>
&gt; Section 12.1.1: &quot;When a UAS responds to a request with a response=
 that<br>
&gt; establishes a dialog (such as a 2xx to INVITE), the UAS MUST copy all<=
br>
&gt; Record-Route header field values from the request into the response<br=
>
&gt; [...]. The UAS MUST add a Contact header field to the response.&quot;<=
br>
&gt;<br>
&gt; So it&#39;s clear that a 1xx response to an INVITE creates a dialog an=
d<br>
&gt; then it MUST contain a Contact header and mirrored Record-Route<br>
&gt; headers, *regardless* the usage of 100rel.<br>
&gt;<br>
&gt; Am I wrong? if so, why?<br>
<br>
</div>Not wrong, just incomplete. This will create an (early) dialog at the=
 UAS.<br>
It may or may not create a dialog at the UAC without 100rel since the<br>
message may never get to the UAC. Where I said &quot;make sense&quot; above=
,<br>
it might have been better if I had said &quot;be useful&quot;.<br>
<div class=3D"im"><br>
&gt;<br>
&gt; Regards.<br>
&gt;<br>
&gt;<br>
&gt; --<br>
&gt; I=F1aki Baz Castillo<br>
&gt; &lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
&gt; _______________________________________________<br>
</div>&gt; sipcore mailing list<br>
&gt; <a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_b=
lank">https://www.ietf.org/mailman/listinfo/sipcore</a><br>
<div><div></div><div class=3D"h5"><br>
_______________________________________________<br>
Sip mailing list =A0<a href=3D"https://www.ietf.org/mailman/listinfo/sip" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
This list is essentially closed and only used for finishing old business.<b=
r>
Use <a href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs=
.columbia.edu</a> for questions on how to develop a SIP implementation.<br>
Use <a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a> for new deve=
lopments on the application of sip.<br>
Use <a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a> for issues rel=
ated to maintenance of the core SIP specifications.<br>
</div></div></blockquote></div><br></div></div></div>

--90e6ba6135a8a9886e04a9b2b804--

From samirs.lists@gmail.com  Thu Aug  4 21:15:29 2011
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References: <20110802145359.C9DEE98C50D@rfc-editor.org> <DC49588FF3643F43B3A2A8F6F0A625F0284638642A@mailbox1.acmepacket.com> <69373B11-F46D-4F88-B4A8-BF56A8EC2624@nostrum.com> <CALiegf=1=ZqEcs0kE=Y+ykrcd9miPDSN=mzP99-EWytVXdXMwA@mail.gmail.com> <9B06B312-4BCF-499D-979F-33D8E70DAD5A@nostrum.com> <CAFQP3_afcp=COozZqB+g39LU3XVtYi-tSrxCuyZkOLWbNVasfw@mail.gmail.com>
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From: Samir Srivastava <samirs.lists@gmail.com>
To: Romel Khan <romel.khan@idt.net>
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Cc: "sip@ietf.org List" <sip@ietf.org>, =?ISO-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>, "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>
Subject: Re: [sipcore] [Sip] [Technical Errata Reported] RFC3261 (2910)
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Hi, IMHO presentation of information in tabulated form helps a lot to
starters. Like ABNF it helps parser developers (expert of syntax &
semantic analysis) to develop it without referring each line of SIP
rfc's. 3262 or 100rel should have updated table  Ideally each
subsequent RFC should conisder table updation. Tabulation of
information will be done by vendors internally anyway. So do it in
community. SIP needed hitchakers guide. Simplicity for starters
please. Regards Samir

On 8/4/11, Romel Khan <romel.khan@idt.net> wrote:
> So it is useful if one of UAS or UAC requires it, but it does not have to=
 be
> mandatory. Some comments:
> -- RFC3261 mentions early dialog without mentioning RFC3262. Then it seem=
s
> logical to me that it needs to be made clear in this RFC3261 that early
> dialog must mean Contact and Record-Route (if Record-Route was received i=
n
> INVITE) headers is mandatory in 1xx without reference to 100rel.
>
> -- A UAS could always send 1xx with headers that are required for early
> dialog but it doesn't have to enforce 100rel (eg because the origination =
or
> UAS side itself may not support reliable provisional response handling, o=
r
> reliable provisioning not really required for its operation). UAS could s=
end
> "support:100rel" if it supports it, or it would not send it if it doesn't
> support this. In my opinion, if UAC hasn't sent 100rel required, it shoul=
d
> be up to the UAS to decide whether to enforce 100rel
> (with "required:100rel") if its application really requires SIP requests
> before call answer. If the origination side (UAC) side has a need to send
> early requests, like UPDATE, then the UAC should require 100rel from the
> termination side (UAS) by sending this in INVITE. In a VoIP service provi=
der
> world, these kind of capabilities are configured during interconnect turn
> up.
>
> -- I notice that some vendors gateway implementations, even if gateway is
> the termination side, require 100rel for the gateway to receive pre-answe=
r
> requests such as UPDATE. This really didn't have to be this way. I have
> always seen these gateways, when it is the termination side, respond back
> SIP 183 with the headers that create early dialog. So if the origination
> side received the SIP 183 response, then there is no reason for the
> origination side to now not be able to send UPDATE request. Also, no
> reason for the termination gateways to not accept the SIP UPDATE without
> requiring PRACK.
>
> Thanks.
>
> On Wed, Aug 3, 2011 at 11:46 AM, Robert Sparks <rjsparks@nostrum.com> wro=
te:
>
>> (removing the rfc-editor and trimming the distribution to the lists)
>>
>> On Aug 2, 2011, at 5:24 PM, I=F1aki Baz Castillo wrote:
>>
>> > 2011/8/2 Robert Sparks <rjsparks@nostrum.com>:
>> >> Further, they're only going to make sense for 1xx that is sent using
>> 100rel.
>> >
>> > This has been discussed in sip-implementors, and that assertion seems
>> > incorrect. As I've reported in the errata:
>> >
>> >
>> > Section 12.1: "Dialogs are created through the generation of
>> > non-failure responses to requests with specific methods. Within this
>> > specification, only 2xx and 101-199 responses with a To tag, where the
>> > request was INVITE, will establish a dialog."
>> >
>> > Section 12.1.1: "When a UAS responds to a request with a response that
>> > establishes a dialog (such as a 2xx to INVITE), the UAS MUST copy all
>> > Record-Route header field values from the request into the response
>> > [...]. The UAS MUST add a Contact header field to the response."
>> >
>> > So it's clear that a 1xx response to an INVITE creates a dialog and
>> > then it MUST contain a Contact header and mirrored Record-Route
>> > headers, *regardless* the usage of 100rel.
>> >
>> > Am I wrong? if so, why?
>>
>> Not wrong, just incomplete. This will create an (early) dialog at the UA=
S.
>> It may or may not create a dialog at the UAC without 100rel since the
>> message may never get to the UAC. Where I said "make sense" above,
>> it might have been better if I had said "be useful".
>>
>> >
>> > Regards.
>> >
>> >
>> > --
>> > I=F1aki Baz Castillo
>> > <ibc@aliax.net>
>> > _______________________________________________
>> > sipcore mailing list
>> > sipcore@ietf.org
>> > https://www.ietf.org/mailman/listinfo/sipcore
>>
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is essentially closed and only used for finishing old business=
.
>> Use sip-implementors@cs.columbia.edu for questions on how to develop a S=
IP
>> implementation.
>> Use dispatch@ietf.org for new developments on the application of sip.
>> Use sipcore@ietf.org for issues related to maintenance of the core SIP
>> specifications.
>>
>

From ibc@aliax.net  Sat Aug  6 05:16:32 2011
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Subject: Re: [sipcore] [Sip]  [Technical Errata Reported] RFC3261 (2910)
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--0016e651395e5c1e0e04a9d53095
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El 04/08/2011 21:09, "Romel Khan" <romel.khan@idt.net> escribi=C3=B3:
>
> So it is useful if one of UAS or UAC requires it, but it does not have to
be mandatory. Some comments:
> -- RFC3261 mentions early dialog without mentioning RFC3262. Then it seem=
s
logical to me that it needs to be made clear in this RFC3261 that early
dialog must mean Contact and Record-Route (if Record-Route was received in
INVITE) headers is mandatory in 1xx without reference to 100rel.
>
> -- A UAS could always send 1xx with headers that are required for early
dialog but it doesn't have to enforce 100rel (eg because the origination or
UAS side itself may not support reliable provisional response handling, or
reliable provisioning not really required for its operation). UAS could sen=
d
"support:100rel" if it supports it, or it would not send it if it doesn't
support this. In my opinion, if UAC hasn't sent 100rel required, it should
be up to the UAS to decide whether to enforce 100rel
(with "required:100rel") if its application really requires SIP requests
before call answer. If the origination side (UAC) side has a need to send
early requests, like UPDATE, then the UAC should require 100rel from the
termination side (UAS) by sending this yin INVITE. In a VoIP service
provider world, these kind of capabilities are configured during
interconnect turn up.
>
> -- I notice that some vendors gateway implementations, even if gateway is
the termination side, require 100rel for the gateway to receive pre-answer
requests such as UPDATE. This really didn't have to be this way. I have
always seen these gateways, when it is the termination side, respond back
SIP 183 with the headers that create early dialog. So if the origination
side received the SIP 183 response, then there is no reason for the
origination side to now not be able to send UPDATE request. Also, no
reason for the termination gateways to not accept the SIP UPDATE without
requiring PRACK.

I entirely agree with it. And I also think that the fact that a 1xx respons=
e
without 100rel is not reliable does not mean that it does not create a
dialog so Contact and RR are required.

--0016e651395e5c1e0e04a9d53095
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<p><br>
El 04/08/2011 21:09, &quot;Romel Khan&quot; &lt;<a href=3D"mailto:romel.kha=
n@idt.net">romel.khan@idt.net</a>&gt; escribi=C3=B3:<br>
&gt;<br>
&gt; So it is useful if one of UAS or UAC requires it, but it does not have=
 to be mandatory. Some comments:<br>
&gt; -- RFC3261 mentions early dialog without mentioning RFC3262. Then it s=
eems logical to me that it needs to be made clear in this RFC3261 that earl=
y dialog must mean Contact and Record-Route (if=C2=A0Record-Route was recei=
ved in INVITE) headers is mandatory in 1xx without reference to 100rel.<br>

&gt;<br>
&gt; -- A UAS could always send 1xx with headers that are required for earl=
y dialog but it doesn&#39;t have to enforce 100rel (eg because the originat=
ion or UAS side itself may not support reliable provisional response handli=
ng, or reliable provisioning not really required for its operation).=C2=A0U=
AS could send &quot;support:100rel&quot; if it supports it,=C2=A0or it woul=
d not send it if it doesn&#39;t support this.=C2=A0In my opinion,=C2=A0if U=
AC hasn&#39;t sent 100rel required,=C2=A0it should be up to the UAS to deci=
de whether to enforce 100rel (with=C2=A0&quot;required:100rel&quot;)=C2=A0i=
f its application really requires SIP requests before call answer. If the o=
rigination side (UAC) side has a need to send early requests, like UPDATE, =
then the UAC should require 100rel from the termination side (UAS) by sendi=
ng this yin INVITE.=C2=A0In a VoIP service provider world,=C2=A0these kind =
of capabilities are configured during interconnect turn up. =C2=A0<br>

&gt;<br>
&gt; -- I notice that some vendors gateway implementations, even if gateway=
 is the termination side, require 100rel for the gateway to receive pre-ans=
wer requests such as UPDATE. This really didn&#39;t have to be this way. I =
have always seen these gateways, when it is the termination side, respond b=
ack SIP 183 with the headers that create early dialog. So if the originatio=
n side received the SIP 183 response, then there is no reason for the origi=
nation side to now not be able to send UPDATE request. Also,=C2=A0no reason=
=C2=A0for the termination gateways to not accept the SIP UPDATE without req=
uiring PRACK. =C2=A0=C2=A0</p>

<p>I entirely agree with it. And I also think that the fact that a 1xx resp=
onse without 100rel is not reliable does not mean that it does not create a=
 dialog so Contact and RR are required.</p>

--0016e651395e5c1e0e04a9d53095--

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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Samir Srivastava <samirs.lists@gmail.com>, Romel Khan <romel.khan@idt.net>
Date: Sat, 6 Aug 2011 21:31:40 -0400
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Subject: Re: [sipcore] [Sip]   [Technical Errata Reported] RFC3261 (2910)
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> From: Samir Srivastava [samirs.lists@gmail.com]
>=20
> Hi, IMHO presentation of information in tabulated form helps a lot to
> starters. Like ABNF it helps parser developers (expert of syntax &
> semantic analysis) to develop it without referring each line of SIP
> rfc's.

The tables are a useful guide to starters.  But as others have noted,
it is very difficult to keep the tables up to date, and there is no
format for the tables that correctly presents the conditions for
all "optional" headers.

Parser developers may be tempted to use the tables to determine
whether a message has all the required headers, but it is nearly
impossible to make that determination other than while carrying out
the processing for the message.

The real rule for the presence of headers is: "Perform the processing
that is mandated by the SIP message.  Headers whose values are
required to perform this processing must be present; headers whose
values are not used in the processing are ignored.  If a needed header
is not present, the processing of the message is aborted and it has no
effect.  If the message is a request, an appropriate error response is
sent (if enough information is present in the request to allow doing
so)."

Dale

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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Sat, 6 Aug 2011 21:38:23 -0400
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> From: Paul Kyzivat [pkyzivat@alum.mit.edu]
>=20
> Of course its true that this [a 1xx response to INVITE] will only
> create an early dialog at the UAC if the UAC actually receives the
> (unreliable) response. If it doesn't receive it, it won't know of the
> dialog, and so won't try to use BYE to terminate the dialog. But if it
> does receive it, and does know of the dialog, it can then use BYE to
> terminate it. And in that case the UAS will concur that there is a
> dialog and so be able to properly process the BYE.

I agree with Paul.  Indeed, if a UAC receives a 1xx response that does
*not* have a Contact header, processing of the response cannot be
completed (as the dialog information cannot be constructed), and so
the response must have no effect on the UAC.

In regard to Record-Route, omitting them in a 1xx is even worse, as if
the UAC receives the 1xx, the Record-Routes fix the route set for the
dialog, and correct Record-Routes set in the 2xx response can't change
the recorded route set.

Dale

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Subject: Re: [sipcore] [Sip] [Technical Errata Reported] RFC3261 (2910)
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El 07/08/2011 03:38, "Worley, Dale R (Dale)" <dworley@avaya.com> escribi=C3=
=B3:

> In regard to Record-Route, omitting them in a 1xx is even worse, as if
> the UAC receives the 1xx, the Record-Routes fix the route set for the
> dialog, and correct Record-Routes set in the 2xx response can't change
> the recorded route

Hi. I dont agree with this last assertion. AFAIR somewhere in the rfc is
stated that RR in the 2XX replaces any previous RR, even more when previous
RR came in not reliable responses. However the same is not true for the sdp=
.

IMHO rfc3261 defined non reliable responses but later all need responses to
be reliable (which would avoid all these issues). This is just a protocol
core error design (obvious taking into account all these discussions so man=
y
years with no strong agreement).

--001517576bcaf406b004a9e7c542
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<p>El 07/08/2011 03:38, &quot;Worley, Dale R (Dale)&quot; &lt;<a href=3D"ma=
ilto:dworley@avaya.com">dworley@avaya.com</a>&gt; escribi=C3=B3:</p>
<p>&gt; In regard to Record-Route, omitting them in a 1xx is even worse, as=
 if<br>
&gt; the UAC receives the 1xx, the Record-Routes fix the route set for the<=
br>
&gt; dialog, and correct Record-Routes set in the 2xx response can&#39;t ch=
ange<br>
&gt; the recorded route</p>
<p>Hi. I dont agree with this last assertion. AFAIR somewhere in the rfc is=
 stated that RR in the 2XX replaces any previous RR, even more when previou=
s RR came in not reliable responses. However the same is not true for the s=
dp.</p>

<p>IMHO rfc3261 defined non reliable responses but later all need responses=
 to be reliable (which would avoid all these issues). This is just a protoc=
ol core error design (obvious taking into account all these discussions so =
many years with no strong agreement).</p>


--001517576bcaf406b004a9e7c542--

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To: =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Sun, 7 Aug 2011 21:27:33 -0400
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> From: I=F1aki Baz Castillo [ibc@aliax.net]
>=20
> Hi. I dont agree with this last assertion. AFAIR somewhere in the rfc
> is stated that RR in the 2XX replaces any previous RR, even more when
> previous RR came in not reliable responses. However the same is not
> true for the sdp.

Interesting!  I had thought the rule that the route set is never
changed was absolute.  But here is RFC 3261 section 13.2.2.4:

   [...]
   If the dialog identifier in the 2xx response matches the dialog
   identifier of an existing dialog, the dialog MUST be transitioned to
   the "confirmed" state, and the route set for the dialog MUST be
   recomputed based on the 2xx response using the procedures of Section
   12.2.1.2.  Otherwise, a new dialog in the "confirmed" state MUST be
   constructed using the procedures of Section 12.1.2.

      Note that the only piece of state that is recomputed is the route
      set.  Other pieces of state such as the highest sequence numbers
      (remote and local) sent within the dialog are not recomputed.  The
      route set only is recomputed for backwards compatibility.  RFC
      2543 did not mandate mirroring of the Record-Route header field in
      a 1xx, only 2xx.  [...]

> IMHO rfc3261 defined non reliable responses but later all need
> responses to be reliable (which would avoid all these issues). This is
> just a protocol core error design (obvious taking into account all
> these discussions so many years with no strong agreement).

True, but for backward compatibility, we have to tolerate unreliable
responses.

Dale

From adam@nostrum.com  Tue Aug  9 14:53:28 2011
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[as chair]

I have posted preliminary minutes for the Québec face-to-face meeting here:

http://www.ietf.org/proceedings/81/minutes/sipcore.html

Please review them, and submit any corrections to me and Paul and me no 
later than Tuesday, August 16th. Thank you.

/a

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Sending another test email.  If this one is actually delivered, sorry for n=
oise.


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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
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Is there a program to make the "ASCII art" diagrams of SIP message flows?

Dale

From adam@nostrum.com  Fri Aug 12 15:28:50 2011
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On 8/12/11 5:25 PM, Worley, Dale R (Dale) wrote:
> Is there a program to make the "ASCII art" diagrams of SIP message flows?

Try http://sourceforge.net/projects/callplot/

/a

From petithug@acm.org  Fri Aug 12 15:35:04 2011
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 08/12/2011 03:25 PM, Worley, Dale R (Dale) wrote:
> Is there a program to make the "ASCII art" diagrams of SIP message flows?

Wireshark.  Menu Telephony|Voip Calls...

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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=Y1ZO
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To: Adam Roach <adam@nostrum.com>
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> From: Adam Roach [adam@nostrum.com]
>=20
> > Is there a program to make the "ASCII art" diagrams of SIP message flow=
s?
>=20
> Try http://sourceforge.net/projects/callplot/

Wow, that's *exactly* what I need.  Thanks!

Dale

From salinisngh1@gmail.com  Wed Aug 17 02:38:38 2011
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From: salini singh <salinisngh1@gmail.com>
To: sipcore@ietf.org
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Subject: Re: [sipcore] IETF draft submission issue - please provide support and Guidance
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Hi,

This is Salini Singh, and  I have subscribed to SIPCORE mailing list.
I have a question : I want to submit a SIP-Draft -on existing SIP protocol
as a independent submission, can I submit via SIPCORE , putting 2 - questio=
n
on this:


*Question 1: =93Intended status:=94 for existing standard**.*
Herein our draft which is to be submitted is about existing Internet
Standards Protocol =96 SIP ( Session Initiation Protocol) suggesting a new
feature and unique way to implement to avoid current unstable working
technique. What should I need to write against =93Intended status:=94 withi=
n my
proposed draft. Should it be =93Best current practice=94 or =93Standards Tr=
ack=94 .

 *Question 2: First line of the Internet draft*.
Our draft is intended for an advancement in existing SIP based protocol,
submitting for SIPCORE Working Group. Since I am submitting this proposed
draft independently  hence what should be the FIRST LINE OF THE DRAFT..
should it be =93SIPCORE Working Group=94or =93Network Working Group=94  or
=93Independent Submission=93.


What I need to do for this.
Do I have to put my Draft into discussion before submitting or it need your
approval for my draft name in the form - "draft-ietf-sipcore......" ?

Thanks and Regards,
Salini S

--001636e0a74d77278704aab045f9
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Hi,<br><br>This is Salini Singh, and=A0 I have subscribed to SIPCORE mailin=
g list.<br>I
 have a question : I want to submit a SIP-Draft -on existing SIP=20
protocol as a independent submission, can I submit via SIPCORE , putting
 2 - question on this:<br>
<br><br><b><u>Question 1: =93Intended status:=94 for existing standard</u><=
/b><u>.</u><br>Herein
 our draft which is to be submitted is about existing Internet Standards
 Protocol =96 SIP ( Session Initiation Protocol) suggesting a new feature=
=20
and unique way to implement to avoid current unstable working technique.
 What should I need to write against =93Intended status:=94 within my=20
proposed draft. Should it be =93Best current practice=94 or =93Standards=20
Track=94 .<br>
<br>=A0<u><b>Question 2: First line of the Internet draft</b></u>.<br>Our=
=20
draft is intended for an advancement in existing SIP based protocol,=20
submitting for SIPCORE Working Group. Since I am submitting this=20
proposed draft independently =A0hence what should be the FIRST LINE OF THE
 DRAFT.. should it be =93SIPCORE Working Group=94or =93Network Working Grou=
p=94=20
=A0or =93Independent Submission=93.<br><br><br>What I need to do for this.<=
br>Do
 I have to put my Draft into discussion before submitting or it need=20
your approval for my draft name in the form - &quot;draft-ietf-sipcore.....=
.&quot;
 ?<br>
<br>Thanks and Regards,<br>Salini S

--001636e0a74d77278704aab045f9--

From pkyzivat@alum.mit.edu  Wed Aug 17 07:47:46 2011
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Subject: Re: [sipcore] IETF draft submission issue - please provide support and Guidance
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Salini,

I gather you are new to the ietf process.

First, it isn't clear from what you have stated below that sipcore would 
be the proper forum for the work you describe. We have a special work 
group in the RAI area that is charged with scoping out new work on SIP 
and determining if/how it should proceed. In my opinion, as co-chair of 
sipcore, that you should first present your concept to the dispatch WG. 
(That means you should subscribe to dispatch@ietf.org and send mail there.)

Regarding Intended Status: I think it is premature to determine that. It 
may even be premature to write a draft.

What I suggest is that you:
- subscribe to the dispatch@ietf.org mailing list
- send an email to that list, focusing primarily on giving a high level
   description of the problem you are trying to solve, preferable with
   some use cases. If you have some thoughts on how to solve the problem
   you can sketch them too, but don't invest too much effort in that,
   since it is likely to change significantly after the the problem is
   discussed by the community.
- you should frame the email above so as to solicit discussion and
   feedback on it. Hopefully that will come. If not, then perhaps your
   problem isn't of general interest. Or maybe your problem will need to
   be further clarified so others can relate to it.
- if it looks like there is interest, you can then submit a draft,
   with a name such as draft-singh-dispatch-xxxxx-00.txt. At this state
   it doesn't matter much what the intended status is, you can make it
   be informational, or whatever. It can be changed later.
   In this draft you would formalize the problem statement, use cases,
   and a list of requirements that any solution to the problem statement
   must fulfill.
- then you should send an email to the dispatch list referencing this
   draft and asking for comments on it.
- based on the comments received, you will likely want to submit one or
   more revisions to the draft, refining it.
- somewhere along the way in this process, the dispatch WG chairs will
   probably get involved to guide you in how to proceed, based on the
   interest in the subject expressed on the list.

You could skip the first few steps above, and go right to the step where 
you submit a draft. But if you haven't done this before it will probably 
be easier to start at the beginning.

	Good Luck,
	Paul Kyzivat, sipcore co-chair

On 8/17/11 5:39 AM, salini singh wrote:
> Hi,
>
> This is Salini Singh, and  I have subscribed to SIPCORE mailing list.
> I have a question : I want to submit a SIP-Draft -on existing SIP
> protocol as a independent submission, can I submit via SIPCORE , putting
> 2 - question on this:
>
>
> *_Question 1: “Intended status:” for existing standard_*_._
> Herein our draft which is to be submitted is about existing Internet
> Standards Protocol – SIP ( Session Initiation Protocol) suggesting a new
> feature and unique way to implement to avoid current unstable working
> technique. What should I need to write against “Intended status:” within
> my proposed draft. Should it be “Best current practice” or “Standards
> Track” .
>
> _*Question 2: First line of the Internet draft*_.
> Our draft is intended for an advancement in existing SIP based protocol,
> submitting for SIPCORE Working Group. Since I am submitting this
> proposed draft independently  hence what should be the FIRST LINE OF THE
> DRAFT.. should it be “SIPCORE Working Group”or “Network Working Group”
>   or “Independent Submission“.
>
>
> What I need to do for this.
> Do I have to put my Draft into discussion before submitting or it need
> your approval for my draft name in the form - "draft-ietf-sipcore......" ?
>
> Thanks and Regards,
> Salini S
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From dworley@avaya.com  Wed Aug 17 13:07:27 2011
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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>, "john.elwell@siemens-enterprise.com" <john.elwell@siemens-enterprise.com>
Date: Wed, 17 Aug 2011 16:06:27 -0400
Thread-Topic: Reviewers needed for new music-on-hold technique
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> From: Elwell, John [john.elwell at siemens-enterprise.com]
>=20
> I haven't reviewed it in detail, but I see there is no mention of
> RTCP. Perhaps it isn't really relevant, except that it provides
> another reason why the media server needs to supply an IP address and
> port - even though it will not receive RTP, it will receive RTCP. The
> RTCP port could be implicit or explicit.

I don't think I've mentioned this, but draft-worley-service-example-06
has updates to point out that allowing RTCP is useful.  See
http://tools.ietf.org/rfcdiff?url2=3Ddraft-worley-service-example-06 and
search for "RTCP".  These changes are carried forward into the
current -07 draft as well.

Do these changes suffice?

Dale

From john.elwell@siemens-enterprise.com  Thu Aug 18 00:04:16 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: "Worley, Dale R (Dale)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 18 Aug 2011 09:05:04 +0200
Thread-Topic: Reviewers needed for new music-on-hold technique
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Dale,

Yes, it does now make an appropriate mention of RTCP - thanks.

I am also curious as to why there are a couple of mentions of using directi=
onality "inactive". Why would any of the entities involved use "inactive" u=
nless they don't wish to send/receive music-on-hold? In particular, why wou=
ld the INVITE from the executing UA to the MOH source use "inactive" - it i=
s specifically asking the MOH source to send music, so if it wants "inactiv=
e", why send the INVITE?

Perhaps this is to do with multiple media, where the remote UA offers sever=
al media, and MOH is required only for one of these, so the other media des=
criptions are set to "inactive" in the INVITE to the MOH source. Whether or=
 not that was the intention of "inactive", I think further discussion of mu=
lti-media situations is required. For example, if the remote UA offers audi=
o and video and the executing UA wants MOH only for audio, how would it beh=
ave? Perhaps the example should be extended to show such a possibility.

John

> -----Original Message-----
> From: Worley, Dale R (Dale) [mailto:dworley@avaya.com]=20
> Sent: 17 August 2011 21:06
> To: sipcore@ietf.org; Elwell, John
> Subject: Reviewers needed for new music-on-hold technique
>=20
> > From: Elwell, John [john.elwell at siemens-enterprise.com]
> >=20
> > I haven't reviewed it in detail, but I see there is no mention of
> > RTCP. Perhaps it isn't really relevant, except that it provides
> > another reason why the media server needs to supply an IP=20
> address and
> > port - even though it will not receive RTP, it will receive=20
> RTCP. The
> > RTCP port could be implicit or explicit.
>=20
> I don't think I've mentioned this, but draft-worley-service-example-06
> has updates to point out that allowing RTCP is useful.  See
> http://tools.ietf.org/rfcdiff?url2=3Ddraft-worley-service-example-06 and
> search for "RTCP".  These changes are carried forward into the
> current -07 draft as well.
>=20
> Do these changes suffice?
>=20
> Dale
> =

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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 18 Aug 2011 14:11:15 -0400
Thread-Topic: Reviewers needed for new music-on-hold technique
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> From: Elwell, John [john.elwell@siemens-enterprise.com]
>=20
> Yes, it does now make an appropriate mention of RTCP - thanks.

Good -- Thanks for checking that out.

> I am also curious as to why there are a couple of mentions of using
> directionality "inactive". Why would any of the entities involved use
> "inactive" unless they don't wish to send/receive music-on-hold? In
> particular, why would the INVITE from the executing UA to the MOH
> source use "inactive" - it is specifically asking the MOH source to
> send music, so if it wants "inactive", why send the INVITE?

The strict answer as to why "inactive" is mentioned is that the
algorithm would not be correct without it, and I hadn't considered the
possible optimization of omitting the INVITE to the MOH source.

The situation arises when the remote UA gives directionality
"sendonly" because the remote UA has previously put the call on hold
also.  (The remote UA may reduce the directionality to "inactive" due
to the +sip.rendering=3D"no" in the re-INVITE from the executing UA.)

There probably other situations where this arises.  E.g., where the
remote UA has formerly used a media stream in the dialog but no longer
wants to.

If all of the media streams that would be in the INVITE to the MOH
source have "inactive" directionality, the executing UA can omit
establishing the dialog with the MOH source.  The executing UA would
synthesize the SDP answer itself.  While the call is on hold, if the
remote UA sends a re-INVITE with directionality "active"/"recvonly",
then the executing UA has to start a dialog with the MOH source.

I should add a note that discusses this optimization.

> Perhaps this is to do with multiple media, where the remote UA offers
> several media, and MOH is required only for one of these, so the other
> media descriptions are set to "inactive" in the INVITE to the MOH
> source. Whether or not that was the intention of "inactive", I think
> further discussion of multi-media situations is required. For example,
> if the remote UA offers audio and video and the executing UA wants MOH
> only for audio, how would it behave? Perhaps the example should be
> extended to show such a possibility.

I would expect multiple media streams to work exactly as an extension
of the single media stream situation.  But there may be requirements
that I am not considering.

In the case where a call is being put on-hold and the remote UA is
willing to accept two streams (e.g., audio and video), I would expect
the executing UA to forward this request intact to the MOH server.
The MOH server would decide what media to send to the remote UA.

If the executing UA wanted to restrict the MOH media, it could do so
by editing the SDP offer.  But I don't see any particular reason to do
so.

What are the complexities you are seeing?

Dale

From john.elwell@siemens-enterprise.com  Thu Aug 18 23:23:10 2011
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To: "Worley, Dale R (Dale)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Fri, 19 Aug 2011 08:24:02 +0200
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=20

> -----Original Message-----
> From: Worley, Dale R (Dale) [mailto:dworley@avaya.com]=20
> Sent: 18 August 2011 19:11
> To: Elwell, John; sipcore@ietf.org
> Subject: RE: Reviewers needed for new music-on-hold technique
>=20
> > From: Elwell, John [john.elwell@siemens-enterprise.com]
> >=20
> > Yes, it does now make an appropriate mention of RTCP - thanks.
>=20
> Good -- Thanks for checking that out.
>=20
> > I am also curious as to why there are a couple of mentions of using
> > directionality "inactive". Why would any of the entities=20
> involved use
> > "inactive" unless they don't wish to send/receive music-on-hold? In
> > particular, why would the INVITE from the executing UA to the MOH
> > source use "inactive" - it is specifically asking the MOH source to
> > send music, so if it wants "inactive", why send the INVITE?
>=20
> The strict answer as to why "inactive" is mentioned is that the
> algorithm would not be correct without it, and I hadn't considered the
> possible optimization of omitting the INVITE to the MOH source.
>=20
> The situation arises when the remote UA gives directionality
> "sendonly" because the remote UA has previously put the call on hold
> also.  (The remote UA may reduce the directionality to "inactive" due
> to the +sip.rendering=3D"no" in the re-INVITE from the executing UA.)
>=20
> There probably other situations where this arises.  E.g., where the
> remote UA has formerly used a media stream in the dialog but no longer
> wants to.
>=20
> If all of the media streams that would be in the INVITE to the MOH
> source have "inactive" directionality, the executing UA can omit
> establishing the dialog with the MOH source.  The executing UA would
> synthesize the SDP answer itself.  While the call is on hold, if the
> remote UA sends a re-INVITE with directionality "active"/"recvonly",
> then the executing UA has to start a dialog with the MOH source.
>=20
> I should add a note that discusses this optimization.
>=20
> > Perhaps this is to do with multiple media, where the remote=20
> UA offers
> > several media, and MOH is required only for one of these,=20
> so the other
> > media descriptions are set to "inactive" in the INVITE to the MOH
> > source. Whether or not that was the intention of "inactive", I think
> > further discussion of multi-media situations is required.=20
> For example,
> > if the remote UA offers audio and video and the executing=20
> UA wants MOH
> > only for audio, how would it behave? Perhaps the example should be
> > extended to show such a possibility.
>=20
> I would expect multiple media streams to work exactly as an extension
> of the single media stream situation.  But there may be requirements
> that I am not considering.
>=20
> In the case where a call is being put on-hold and the remote UA is
> willing to accept two streams (e.g., audio and video), I would expect
> the executing UA to forward this request intact to the MOH server.
> The MOH server would decide what media to send to the remote UA.
>=20
> If the executing UA wanted to restrict the MOH media, it could do so
> by editing the SDP offer.  But I don't see any particular reason to do
> so.
>=20
> What are the complexities you are seeing?
[JRE] Nothing that can't be dealt with relatively easily, but perhaps some =
mention is needed. Don't forget that hold (in terms of SDP directionality i=
ndications) can be on a per medium basis. I guess if the remote UA offers a=
udio and video and the executing UA only wants to hold audio, it would need=
 to manipulate the SDP such that only audio is offered to the MOH source, a=
nd then when the answer is received, add in its own video description befor=
e forwarding the answer to the remote UA. Or if the executing UA forwards b=
oth audio and video to the MOH source and the MOH source rejects video, the=
 executing UA might want to send inactive to the remote UA rather than pass=
ing on the rejection.

John


John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.


>=20
> Dale
> =

From dworley@avaya.com  Fri Aug 19 12:48:46 2011
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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Fri, 19 Aug 2011 15:48:56 -0400
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How about adding these sections to the end of section 2?


When the Media Stream Directionality is "inactive"

The directionality of the media stream in the SDP offer in an INVITE
or re-INVITE to the music source can be "inactive" if the SDP offer
from the remote UA was "sendonly" or "inactive".  Generally, this
happens when the remote UA also has put the call on hold and provided
a directionality of "sendonly".  In this situation, the executing UA
can omit establishing the dialog with the music source (or can
terminate the existing dialog with the music source).

If the executing UA uses this optimization, it creates the SDP answer
itself, with directionality "inactive" and using its own RTP/RTCP
ports, and returns that answer to the remote UA.

The executing UA must be prepared for the remote UA to send a
re-INVITE with directionality "active" or "recvonly", in which case
the executing UA must initiate a dialog with the music source, as
described above.

Multiple Media Streams

There may be multiple media streams (multiple m=3D lines) in any of the
the SDPs involved in the dialogs.  As the SDPs are manipulated, each
media description (each starting with an m=3D line) is manipulated as
described above for a single media stream.  But there are some
elaborations that the executing UA may implement to achieve specific
effects.

If the executing UA desires to present only certain media types as
on-hold media, when passing the offer SDP through, it can reject any
particular media streams by setting the port number in the m=3D line to
zero.[offer-answer]  This ensures that the answer SDP will also have a
rejection for that m=3D line.

If the executing UA wishes to provide its own on-hold media for a
particular m=3D line, it can do so by providing the answer information
for that m=3D line.  The executing UA may decide to do this when the
offer SDP is received (by modifying the m=3D line to rejected state when
sending it to the music source), or upon receiving the answer from the
music source and discovering that the m=3D line has been rejected.

The executing UA may not want to pass a rejected m=3D line from the
music source to the remote UA (when the remote UA provided a
non-rejected m=3D line), and instead provide an answer with
directionality "inactive" (and specifying its own RTP/RTCP ports).


Dale

From samirs.lists@gmail.com  Sun Aug 21 23:00:33 2011
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Subject: [sipcore] Volunteers please for coauthor
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Hi, It is quite unusual to send this note on list. I approched couple
of folks privately but in vain. Due to unavoidable situation, I have
limited internet access with which I cannot submit draft. I am looking
for coauthors for a draft covering unaddressed issues by SIPS and work
on cashless economy etc at http://samirsrivastava.typepad.com . To the
best of my knowledge and skills it will *AVOID* various security
issues of SIP/ INTERNET. Volunteers pl. People from persmal id gmail
etc can also help. BTW I have solution for overwhelmed response (which
I doubt). CCing secdir for any help. Regards Samir

From john.elwell@siemens-enterprise.com  Sun Aug 21 23:24:52 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: "Worley, Dale R (Dale)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Mon, 22 Aug 2011 08:25:51 +0200
Thread-Topic: Reviewers needed for new music-on-hold technique
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Subject: Re: [sipcore] Reviewers needed for new music-on-hold technique
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Dale,

Yes, this text addresses my concerns.

John=20

> -----Original Message-----
> From: Worley, Dale R (Dale) [mailto:dworley@avaya.com]=20
> Sent: 19 August 2011 20:49
> To: Elwell, John; sipcore@ietf.org
> Subject: RE: Reviewers needed for new music-on-hold technique
>=20
> How about adding these sections to the end of section 2?
>=20
>=20
> When the Media Stream Directionality is "inactive"
>=20
> The directionality of the media stream in the SDP offer in an INVITE
> or re-INVITE to the music source can be "inactive" if the SDP offer
> from the remote UA was "sendonly" or "inactive".  Generally, this
> happens when the remote UA also has put the call on hold and provided
> a directionality of "sendonly".  In this situation, the executing UA
> can omit establishing the dialog with the music source (or can
> terminate the existing dialog with the music source).
>=20
> If the executing UA uses this optimization, it creates the SDP answer
> itself, with directionality "inactive" and using its own RTP/RTCP
> ports, and returns that answer to the remote UA.
>=20
> The executing UA must be prepared for the remote UA to send a
> re-INVITE with directionality "active" or "recvonly", in which case
> the executing UA must initiate a dialog with the music source, as
> described above.
>=20
> Multiple Media Streams
>=20
> There may be multiple media streams (multiple m=3D lines) in any of the
> the SDPs involved in the dialogs.  As the SDPs are manipulated, each
> media description (each starting with an m=3D line) is manipulated as
> described above for a single media stream.  But there are some
> elaborations that the executing UA may implement to achieve specific
> effects.
>=20
> If the executing UA desires to present only certain media types as
> on-hold media, when passing the offer SDP through, it can reject any
> particular media streams by setting the port number in the m=3D line to
> zero.[offer-answer]  This ensures that the answer SDP will also have a
> rejection for that m=3D line.
>=20
> If the executing UA wishes to provide its own on-hold media for a
> particular m=3D line, it can do so by providing the answer information
> for that m=3D line.  The executing UA may decide to do this when the
> offer SDP is received (by modifying the m=3D line to rejected state when
> sending it to the music source), or upon receiving the answer from the
> music source and discovering that the m=3D line has been rejected.
>=20
> The executing UA may not want to pass a rejected m=3D line from the
> music source to the remote UA (when the remote UA provided a
> non-rejected m=3D line), and instead provide an answer with
> directionality "inactive" (and specifying its own RTP/RTCP ports).
>=20
>=20
> Dale
> =

From pkyzivat@alum.mit.edu  Fri Aug 26 06:59:36 2011
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Subject: [sipcore] WGLC of draft-ietf-sipcore-rfc3265bis-03.txt
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<as co-chair>

This is an announcement of a Working Group Last Call on 
draft-ietf-sipcore-rfc3265bis-03.txt:

http://www.ietf.org/id/draft-ietf-sipcore-rfc3265bis-03.txt

Please review and provide any comments you have on the document no later 
than Friday September 16, 2011.
Comments should be sent to the document authors and the SIPCORE WG 
mailing list.

     Thanks,
     Paul

--------------080102060705060600060406
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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    &lt;as co-chair&gt;<br>
    <br>
    This is an announcement of a Working Group Last Call on
    draft-ietf-sipcore-rfc3265bis-03.txt:<br>
    <a
      href="http://www.ietf.org/id/draft-ietf-sipcore-rfc3265bis-03.txt"><br>
      http://www.ietf.org/id/draft-ietf-sipcore-rfc3265bis-03.txt</a><br>
    <br>
    Please review and provide any comments you have on the document no
    later than Friday September 16, 2011.<br>
    Comments should be sent to the document authors and the SIPCORE WG
    mailing list.<br>
    <br>
    &nbsp;&nbsp; &nbsp;Thanks,<br>
    &nbsp;&nbsp; &nbsp;Paul
  </body>
</html>

--------------080102060705060600060406--

From HKaplan@acmepacket.com  Wed Aug 31 07:50:19 2011
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: SIPCORE <sipcore@ietf.org>
Date: Wed, 31 Aug 2011 10:51:49 -0400
Thread-Topic: 202 response in draft-ietf-sipcore-rfc3265bis-03
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Subject: [sipcore] 202 response in draft-ietf-sipcore-rfc3265bis-03
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Howdy,
One of the changes made by 3265bis relative to 3265 is the deprecation of t=
he 202 response.  I have two questions related to this change:

1) The current text for deprecation of 202 in 3265bis not only updates 3265=
 but also RFC 4660, since that also uses 202.  Should 3265bis also update R=
FC 5839, which uses 202 in example flows and notes?

2) The current draft does not say why the 202 response was deprecated.  Thr=
ough some googling, I found this:
http://www.ietf.org/proceedings/75/minutes/sipcore.html#Open%20Issue:%20The=
%20Fate%20of%20202
And I went and listened to the audio archive for IETF 75 SIPCORE meeting.

3265bis-03 says this in section 8.3.1:
   This document does not specify the use of the 202 response code in
   conjunction with the SUBSCRIBE or NOTIFY methods.  Previous versions
   of the SIP Events Framework assigned specific semantics to the 202
   response code.  Implementations conformant with the current
   specification MUST treat an incoming 202 response as identical to a
   200 response, and MUST NOT generate 202 response codes to SUBSCRIBE
   or NOTIFY requests.


I recommend that be changed to this, assuming it's correct:
   This document does not specify the use of the 202 response code in
   conjunction with the SUBSCRIBE or NOTIFY methods.  Previous versions
   of the SIP Events Framework assigned specific semantics to the 202
   response code.  Due to response handling in forking cases, a UAC may
   not have received the response as a 202, and thus it could never have
   been guaranteed to be received.  Furthermore, there was no actual use
   difference for the 202 compared to a 200, given a NOTIFY is sent after
   the subscription is processed and it conveys the correct state.  Since
   it was found that implementations were handling 202 differently from 200=
,
   the 202 response is being deprecated to make it clear there is no such
   difference and they should not be handled differently.

   Implementations conformant with the current
   specification MUST treat an incoming 202 response as identical to a
   200 response, and MUST NOT generate 202 response codes to SUBSCRIBE
   or NOTIFY requests.


-hadriel


On Aug 26, 2011, at 10:00 AM, Paul Kyzivat wrote:

> <as co-chair>
>=20
> This is an announcement of a Working Group Last Call on draft-ietf-sipcor=
e-rfc3265bis-03.txt:
>=20
> http://www.ietf.org/id/draft-ietf-sipcore-rfc3265bis-03.txt
>=20
> Please review and provide any comments you have on the document no later =
than Friday September 16, 2011.
> Comments should be sent to the document authors and the SIPCORE WG mailin=
g list.
>=20
>     Thanks,
>     Paul
> <ATT00001..c>


From internet-drafts@ietf.org  Wed Aug 31 14:18:11 2011
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Subject: [sipcore] I-D Action: draft-ietf-sipcore-event-rate-control-09.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Session Initiation Protocol Core Work=
ing Group of the IETF.

	Title           : Session Initiation Protocol (SIP) Event Notification Ext=
ension for Notification Rate Control
	Author(s)       : Aki Niemi
                          Krisztian Kiss
                          Salvatore Loreto
	Filename        : draft-ietf-sipcore-event-rate-control-09.txt
	Pages           : 25
	Date            : 2011-08-31

   This document specifies mechanisms for adjusting the rate of Session
   Initiation Protocol (SIP) event notifications.  These mechanisms can
   be applied in subscriptions to all SIP event packages.  This document
   updates RFC 3265.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipcore-event-rate-control-0=
9.txt

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

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-sipcore-event-rate-control-09=
.txt
