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From: RFC Errata System <rfc-editor@rfc-editor.org>
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Date: Tue,  2 Aug 2011 07:53:59 -0700 (PDT)
Cc: sip@ietf.org, rfc-editor@rfc-editor.org
Subject: [Sip] [Technical Errata Reported] RFC3261 (2910)
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The following errata report has been submitted for RFC3261,
"SIP: Session Initiation Protocol".

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=3261&eid=2910

--------------------------------------
Type: Technical
Reported by: Iņaki Baz Castillo <ibc@aliax.net>

Section: Table 2

Original Text
-------------
      Header field          where   proxy ACK BYE CAN INV OPT REG
      ___________________________________________________________
      Contact                1xx           -   -   -   o   -   -

Corrected Text
--------------
      Header field          where   proxy ACK BYE CAN INV OPT REG
      ___________________________________________________________
      Contact                1xx           -   -   -   m   -   -

Notes
-----
RFC 3261 says:

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.

However Table 2 (page 162) says:

      Header field          where   proxy ACK BYE CAN INV OPT REG
      ___________________________________________________________
      Contact                1xx           -   -   -   o   -   -
      Record-Route        2xx,18x    mr    -   o   o   o   o   -

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).

Instructions:
-------------
This errata is currently posted as "Reported". If necessary, please
use "Reply All" to discuss whether it should be verified or
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can log in to change the status and edit the report, if necessary. 

--------------------------------------
RFC3261 (draft-ietf-sip-rfc2543bis-09)
--------------------------------------
Title               : SIP: Session Initiation Protocol
Publication Date    : June 2002
Author(s)           : J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, E. Schooler
Category            : PROPOSED STANDARD
Source              : Session Initiation Protocol
Area                : Real-time Applications and Infrastructure
Stream              : IETF
Verifying Party     : IESG

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From: Bob Penfield <BPenfield@acmepacket.com>
To: RFC Errata System <rfc-editor@rfc-editor.org>, "jdrosen@dynamicsoft.com" <jdrosen@dynamicsoft.com>, "schulzrinne@cs.columbia.edu" <schulzrinne@cs.columbia.edu>, "Gonzalo.Camarillo@ericsson.com" <Gonzalo.Camarillo@ericsson.com>, "alan.johnston@wcom.com" <alan.johnston@wcom.com>, "jon.peterson@neustar.com" <jon.peterson@neustar.com>, "rsparks@dynamicsoft.com" <rsparks@dynamicsoft.com>, "mjh@icir.org" <mjh@icir.org>, "schooler@research.att.com" <schooler@research.att.com>, "gonzalo.camarillo@ericsson.com" <gonzalo.camarillo@ericsson.com>, "rjsparks@nostrum.com" <rjsparks@nostrum.com>, "dean.willis@softarmor.com" <dean.willis@softarmor.com>, "drage@alcatel-lucent.com" <drage@alcatel-lucent.com>
Date: Tue, 2 Aug 2011 11:58:50 -0400
Thread-Topic: [Sip] [Technical Errata Reported] RFC3261 (2910)
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Subject: Re: [Sip] [Technical Errata Reported] RFC3261 (2910)
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The entry in the table should be "c" (not "m"). A Contact is not required i=
n a 100 Trying response. The Contact is only required for a 1xx that create=
s a dialog.


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of RFC E=
rrata 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@dy=
namicsoft.com; mjh@icir.org; schooler@research.att.com; gonzalo.camarillo@e=
ricsson.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)


The following errata report has been submitted for RFC3261,
"SIP: Session Initiation Protocol".

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=3D3261&eid=3D2910

--------------------------------------
Type: Technical
Reported by: I=F1aki Baz Castillo <ibc@aliax.net>

Section: Table 2

Original Text
-------------
      Header field          where   proxy ACK BYE CAN INV OPT REG
      ___________________________________________________________
      Contact                1xx           -   -   -   o   -   -

Corrected Text
--------------
      Header field          where   proxy ACK BYE CAN INV OPT REG
      ___________________________________________________________
      Contact                1xx           -   -   -   m   -   -

Notes
-----
RFC 3261 says:

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

Section 12.1.1: "When a UAS responds to a request with a response that esta=
blishes a dialog (such as a 2xx to INVITE), the UAS MUST copy all Record-Ro=
ute 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.

However Table 2 (page 162) says:

      Header field          where   proxy ACK BYE CAN INV OPT REG
      ___________________________________________________________
      Contact                1xx           -   -   -   o   -   -
      Record-Route        2xx,18x    mr    -   o   o   o   o   -

Obviously Record-Route is optional since in the absence of a proxy doing re=
cord-routing, such header will not be present. However Contact header shoul=
d appear as mandatory (m) for 1xx responses for INVITE rather than optional=
 (o).

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

--------------------------------------
RFC3261 (draft-ietf-sip-rfc2543bis-09)
--------------------------------------
Title               : SIP: Session Initiation Protocol
Publication Date    : June 2002
Author(s)           : J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnst=
on, J. Peterson, R. Sparks, M. Handley, E. Schooler
Category            : PROPOSED STANDARD
Source              : Session Initiation Protocol
Area                : Real-time Applications and Infrastructure
Stream              : IETF
Verifying Party     : IESG

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Date: Tue, 2 Aug 2011 20:28:53 +0200
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [Sip] [Technical Errata Reported] RFC3261 (2910)
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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 crea=
tes a dialog.

Right.

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

From rjsparks@nostrum.com  Tue Aug  2 11:50:19 2011
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Cc: Bob Penfield <BPenfield@acmepacket.com>, jdrosen@dynamicsoft.com, schooler@research.att.com, rsparks@dynamicsoft.com, "sip@ietf.org List" <sip@ietf.org>, schulzrinne@cs.columbia.edu, "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>, Keith Drage <drage@alcatel-lucent.com>, alan.johnston@wcom.com, Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>, mjh@icir.org, Jon Peterson <jon.peterson@neustar.com>, Dean Willis <dean.willis@softarmor.com>
Subject: [Sip] Table2/3 maintenance (was Re: [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>


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Subject: Re: [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 Fred_Stacey@Mitel.com  Tue Aug  2 12:03:36 2011
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Subject: Re: [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>

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Subject: Re: [Sip] Table2/3 maintenance (was Re: [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: [Sip] [sipcore]  [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 samirs.lists@gmail.com  Thu Aug  4 21:15:29 2011
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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>, "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>
Subject: Re: [Sip] [sipcore]  [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 pkyzivat@alum.mit.edu  Fri Aug  5 07:36:48 2011
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Subject: Re: [Sip] [sipcore]  [Technical Errata Reported] RFC3261 (2910)
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On 8/5/11 12:15 AM, Samir Srivastava wrote:
> 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

That was the concept that led to the table in the first place.
But history has shown this not to work very well in practice, for a 
number of reasons. Here are some:

- it proved impossible to define a table format that expressed
   all the nuances. There still had to be text to explain the
   complex cases. People tended to believe the table even though
   its flagged as having exceptions

- extensions to sip require updates to the table. But the extensions
   are done in separate RFCs, not revisions to 3261. So they tended
   to specify new rows to the table. These never get rolled up in
   one place. Also, extensions that add methods add columns to the
   table. When that happens, then you need to specify the values
   for the new columns, for rows in 3261 and also in all extensions
   that added rows. This is difficult, and was rarely if ever done
   right.

- given both a table and text descriptions of what is required,
   it was unclear which is the authoritative normative specification.

(I'm certain there are more reasons.)

To be workable we would probably need to move the entire table to an 
IANA registry. That also seemed unworkable.

And with text descriptions it is easier to specify the requirements in 
ways that will apply to most headers and methods that might be defined 
in the future.

The bottom line is that we ultimately decided that the table was a bad 
idea and we didn't want to continue maintaining it.

	Thanks,
	Paul

> 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 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 send
>> "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 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.
>>
>> 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, Iņaki 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.
>>>>
>>>>
>>>> --
>>>> Iņaki 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 SIP
>>> 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.
>>>
>>
> _______________________________________________
> 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 SIP 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: [Sip] [sipcore] [Technical Errata Reported] RFC3261 (2910)
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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--

From dworley@avaya.com  Sat Aug  6 18:31:25 2011
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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
Thread-Topic: [Sip] [sipcore]  [Technical Errata Reported] RFC3261 (2910)
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Subject: Re: [Sip] [sipcore]  [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

From romel.khan@idt.net  Thu Aug  4 12:10:14 2011
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From: Romel Khan <romel.khan@idt.net>
Date: Thu, 4 Aug 2011 15:09:14 -0400
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Subject: Re: [Sip] [sipcore] [Technical Errata Reported] RFC3261 (2910)
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--90e6ba6135a8a9886e04a9b2b804
Content-Type: text/plain; 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:
-- 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  Wed Aug 10 06:37:28 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> <CAK+Spiwx+zJb=ccc0aL1bLVb6faSVCQcOYKJsHz_jnHUpmDnVQ@mail.gmail.com> <4E3BFFFA.7070009@alum.mit.edu>
Date: Wed, 10 Aug 2011 06:37:55 -0700
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From: Samir Srivastava <samirs.lists@gmail.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [Sip] [sipcore] [Technical Errata Reported] RFC3261 (2910)
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Hi, IMHO the updation (to 3261) requirement MUST be needed for
non-table RFC also. We should be sufficient with tables in different
RFCs. Consider development of PUBLISH if UA doesnot support publish it
can still claim 3261 compliance via tables only. Single table req is
not good. By the same token we might need SINGLE place for headers in
textual description. With c,m, m*, o, t labels it is much easier for
parsers. Let TU developers worry about the context/usage. This needs
updation when it widens/narrows. Text only version will cause lengthy
discussions referring lines etc for message rejection. We can say
table with associated section is normative for parsing. TU usage refer
other sections. Regards Samir


On 8/5/11, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> On 8/5/11 12:15 AM, Samir Srivastava wrote:
>> 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
>
> That was the concept that led to the table in the first place.
> But history has shown this not to work very well in practice, for a
> number of reasons. Here are some:
>
> - it proved impossible to define a table format that expressed
>    all the nuances. There still had to be text to explain the
>    complex cases. People tended to believe the table even though
>    its flagged as having exceptions
>
> - extensions to sip require updates to the table. But the extensions
>    are done in separate RFCs, not revisions to 3261. So they tended
>    to specify new rows to the table. These never get rolled up in
>    one place. Also, extensions that add methods add columns to the
>    table. When that happens, then you need to specify the values
>    for the new columns, for rows in 3261 and also in all extensions
>    that added rows. This is difficult, and was rarely if ever done
>    right.
>
> - given both a table and text descriptions of what is required,
>    it was unclear which is the authoritative normative specification.
>
> (I'm certain there are more reasons.)
>
> To be workable we would probably need to move the entire table to an
> IANA registry. That also seemed unworkable.
>
> And with text descriptions it is easier to specify the requirements in
> ways that will apply to most headers and methods that might be defined
> in the future.
>
> The bottom line is that we ultimately decided that the table was a bad
> idea and we didn't want to continue maintaining it.
>
> 	Thanks,
> 	Paul
>
>> 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
>>> 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 originatio=
n
>>> or
>>> UAS side itself may not support reliable provisional response handling,
>>> or
>>> reliable provisioning not really required for its operation). UAS could
>>> send
>>> "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 request=
s
>>> before call answer. If the origination side (UAC) side has a need to se=
nd
>>> early requests, like UPDATE, then the UAC should require 100rel from th=
e
>>> termination side (UAS) by sending this in INVITE. In a VoIP service
>>> provider
>>> world, these kind of capabilities are configured during interconnect tu=
rn
>>> 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 ba=
ck
>>> 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
>>> origination side to now not be able to send UPDATE request. Also, no
>>> reason for the termination gateways to not accept the SIP UPDATE withou=
t
>>> 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 th=
e
>>>>> request was INVITE, will establish a dialog."
>>>>>
>>>>> Section 12.1.1: "When a UAS responds to a request with a response tha=
t
>>>>> 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
>>>> SIP
>>>> 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.
>>>>
>>>
>> _______________________________________________
>> 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.
>>
>
> _______________________________________________
> 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.
>

From pkyzivat@alum.mit.edu  Thu Aug 11 10:48:07 2011
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Date: Thu, 11 Aug 2011 13:48:43 -0400
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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> <CAK+Spiwx+zJb=ccc0aL1bLVb6faSVCQcOYKJsHz_jnHUpmDnVQ@mail.gmail.com> <4E3BFFFA.7070009@alum.mit.edu> <CAK+Spix083awRqeTZfS5EkR+SmRjw+L7JWsGavtt6xei-+Strw@mail.gmail.com>
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Cc: sip@ietf.org
Subject: Re: [Sip] [sipcore] [Technical Errata Reported] RFC3261 (2910)
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On 8/10/11 9:37 AM, Samir Srivastava wrote:
> Hi, IMHO the updation (to 3261) requirement MUST be needed for
> non-table RFC also. We should be sufficient with tables in different
> RFCs. Consider development of PUBLISH if UA doesnot support publish it
> can still claim 3261 compliance via tables only. Single table req is
> not good. By the same token we might need SINGLE place for headers in
> textual description. With c,m, m*, o, t labels it is much easier for
> parsers. Let TU developers worry about the context/usage. This needs
> updation when it widens/narrows. Text only version will cause lengthy
> discussions referring lines etc for message rejection. We can say
> table with associated section is normative for parsing. TU usage refer
> other sections. Regards Samir

I've tried to parse the above, and I just can't figure out what you are 
proposing.

Can you please try again to state how you think these rules should be 
documented?

	Thanks,
	Paul

> On 8/5/11, Paul Kyzivat<pkyzivat@alum.mit.edu>  wrote:
>> On 8/5/11 12:15 AM, Samir Srivastava wrote:
>>> 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
>>
>> That was the concept that led to the table in the first place.
>> But history has shown this not to work very well in practice, for a
>> number of reasons. Here are some:
>>
>> - it proved impossible to define a table format that expressed
>>     all the nuances. There still had to be text to explain the
>>     complex cases. People tended to believe the table even though
>>     its flagged as having exceptions
>>
>> - extensions to sip require updates to the table. But the extensions
>>     are done in separate RFCs, not revisions to 3261. So they tended
>>     to specify new rows to the table. These never get rolled up in
>>     one place. Also, extensions that add methods add columns to the
>>     table. When that happens, then you need to specify the values
>>     for the new columns, for rows in 3261 and also in all extensions
>>     that added rows. This is difficult, and was rarely if ever done
>>     right.
>>
>> - given both a table and text descriptions of what is required,
>>     it was unclear which is the authoritative normative specification.
>>
>> (I'm certain there are more reasons.)
>>
>> To be workable we would probably need to move the entire table to an
>> IANA registry. That also seemed unworkable.
>>
>> And with text descriptions it is easier to specify the requirements in
>> ways that will apply to most headers and methods that might be defined
>> in the future.
>>
>> The bottom line is that we ultimately decided that the table was a bad
>> idea and we didn't want to continue maintaining it.
>>
>> 	Thanks,
>> 	Paul
>>
>>> 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
>>>> 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
>>>> send
>>>> "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
>>>> 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.
>>>>
>>>> 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, Iņaki 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.
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Iņaki 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
>>>>> SIP
>>>>> 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.
>>>>>
>>>>
>>> _______________________________________________
>>> 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 SIP
>>> 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.
>>>
>>
>> _______________________________________________
>> 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 SIP
>> 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 etherchen@foxmail.com  Thu Aug 18 20:17:13 2011
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From honsha.narayan@wipro.com  Fri Aug 19 03:16:20 2011
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Thread-Topic: [Sip] Sent 18x with 'require:100rel' but no PRACK recieved
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From: <honsha.narayan@wipro.com>
To: <etherchen@foxmail.com>, <sip@ietf.org>
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Subject: Re: [Sip] Sent 18x with 'require:100rel' but no PRACK recieved
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Hi ,

=20

First of all , please use "sip-implementers
(sip-implementors@cs.columbia.edu)" forum for the answer of this query.=20

=20

As per SIP RFC3262, you must reject the call in SIP side with 5xx (
provided you had SDP in reliable 18x ). You can reject ISUP side with a
RELEASE message.

=20

But if you want to relax your software compliance a bit here , you can
do that,  but remember your early media negotiation may be in trouble.
If you are sure that on ISUP side the call is answered ( ANM is received
) you can proceed for sending 200 to SIP side.

=20

Regards

Honsha=20

=20

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
etherchen
Sent: Friday, August 19, 2011 8:48 AM
To: sip
Subject: [Sip] Sent 18x with 'require:100rel' but no PRACK recieved

=20

Hi:

=20

  I'm facing a ISUP - SIP interworking issue recently. =20

=20

 I receive INVITE from remote SIP server and send IAM to ISUP side, ISUP
reply ACM/CPG and then I send 18x with 100rel out, but I did not receive
any PRACK from remote SIP server, and I re-send 18x to them. During this
period, ISUP side send ANM to me.=20

=20

 If connect the call, but my end haven't receive PRACK yet, it become a
unreliable call.

 If disconnect the call, I should not base on ANM to release a call.

 If continue 18x re-transmission untill time-out, it will have charging
issue. Because ISUP side start billing base on ANM already, but my SIP
side haven't start charging because not answer.

=20

 Thanks.


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</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Hi ,<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>First of all , please use &quot;sip-implementers =
(sip-implementors@cs.columbia.edu)&quot; forum for the answer of this =
query. <o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>As per SIP RFC3262, you must reject the call in SIP side with 5xx ( =
provided you had SDP in reliable 18x ). You can reject ISUP side with a =
RELEASE message.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>But if you want to relax your software compliance a bit here , you =
can do that, &nbsp;but remember your early media negotiation may be in =
trouble. If you are sure that on ISUP side the call is answered ( ANM is =
received ) you can proceed for sending 200 to SIP =
side.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Regards<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Honsha <o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <b>On Behalf Of =
</b>etherchen<br><b>Sent:</b> Friday, August 19, 2011 8:48 =
AM<br><b>To:</b> sip<br><b>Subject:</b> [Sip] Sent 18x with =
'require:100rel' but no PRACK recieved<o:p></o:p></span></p></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p =
class=3DMsoNormal>Hi:<o:p></o:p></p></div><div><p =
class=3DMsoNormal>&nbsp;<o:p></o:p></p></div><div><p =
class=3DMsoNormal>&nbsp; I'm facing a ISUP - SIP interworking issue =
recently.&nbsp; <o:p></o:p></p></div><div><p =
class=3DMsoNormal>&nbsp;<o:p></o:p></p></div><div><p =
class=3DMsoNormal>&nbsp;I receive&nbsp;INVITE from&nbsp;remote SIP =
server&nbsp;and send IAM to&nbsp;ISUP side, ISUP reply ACM/CPG and then =
I&nbsp;send 18x with 100rel out,&nbsp;but I did not receive&nbsp;any =
PRACK&nbsp;from&nbsp;remote SIP server, and I re-send 18x to them. =
During this period, ISUP side send ANM to me. =
<o:p></o:p></p></div><div><p =
class=3DMsoNormal>&nbsp;<o:p></o:p></p></div><div><p =
class=3DMsoNormal>&nbsp;If&nbsp;connect the call,&nbsp;but my =
end&nbsp;haven't receive PRACK yet, it become a unreliable =
call.<o:p></o:p></p></div><div><p class=3DMsoNormal>&nbsp;If disconnect =
the call, I should not base on ANM to release a =
call.<o:p></o:p></p></div><div><p class=3DMsoNormal>&nbsp;If continue =
18x re-transmission untill time-out, it will have charging issue. =
Because ISUP side start billing base on ANM already, but my SIP side =
haven't start charging because not answer.<o:p></o:p></p></div><div><p =
class=3DMsoNormal>&nbsp;<o:p></o:p></p></div><div><p =
class=3DMsoNormal>&nbsp;Thanks.<o:p></o:p></p></div></div></body></html>
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