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Re: [Sip] [Technical Errata Reported] RFC4028 (1681)
Sorry about the repost, I intended to indicate section RFC 3261 section 8.1.3.5.
> -----Original Message-----
> From: sip-bounces at ietf.org [mailto:sip-bounces at ietf.org] On Behalf Of
> Brett Tate
> Sent: Monday, February 09, 2009 1:00 PM
> To: sip at ietf.org
> Subject: Re: [Sip] [Technical Errata Reported] RFC4028 (1681)
>
> The reported error does not sound correct. If RFC 4028 is incorrect
> concerning From tag, so is RFC 3261 (section 8.1 and likely other
> sections) and various other RFCs.
>
>
>
> > -----Original Message-----
> > From: sip-bounces at ietf.org [mailto:sip-bounces at ietf.org] On Behalf Of
> RFC
> > Errata System
> > Sent: Monday, February 09, 2009 12:37 PM
> > To: srd at cisco.com; jdrosen at cisco.com; jon.peterson at neustar.biz;
> > fluffy at cisco.com; dean.willis at softarmor.com; drage at alcatel-lucent.com
> > Cc: sip at ietf.org; mperumal at cisco.com; rfc-editor at rfc-editor.org
> > Subject: [Sip] [Technical Errata Reported] RFC4028 (1681)
> >
> >
> > The following errata report has been submitted for RFC4028,
> > "Session Timers in the Session Initiation Protocol (SIP)".
> >
> > --------------------------------------
> > You may review the report below and at:
> > http://www.rfc-editor.org/errata_search.php?rfc=4028&eid=1681
> >
> > --------------------------------------
> > Type: Technical
> > Reported by: Muthu Arul Mozhi <mperumal at cisco.com>
> >
> > Section: 13
> >
> > Original Text
> > -------------
> > In section 13 (Example Call Flow) the From tag never changes
> > between the initial INVITE message and the subsequent INVITE
> > messages sent after receiving a 422:
> >
> > message 1
> > INVITE sips:bob at biloxi.example.com SIP/2.0
> > Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bKnashds8
> > Supported: timer
> > Session-Expires: 50
> > Max-Forwards: 70
> > To: Bob <sips:bob at biloxi.example.com>
> > From: Alice <sips:alice at atlanta.example.com>;tag=1928301774
> > Call-ID: a84b4c76e66710
> > CSeq: 314159 INVITE
> > Contact: <sips:alice at pc33.atlanta.example.com>
> > Content-Type: application/sdp
> > Content-Length: 142
> >
> > message 4
> > INVITE sips:bob at biloxi.example.com SIP/2.0
> > Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bKnashds9
> > Supported: timer
> > Session-Expires: 3600
> > Min-SE: 3600
> > Max-Forwards: 70
> > To: Bob <sips:bob at biloxi.example.com>
> > From: Alice <sips:alice at atlanta.example.com>;tag=1928301774
> > Call-ID: a84b4c76e66710
> > CSeq: 314160 INVITE
> > Contact: <sips:alice at pc33.atlanta.example.com>
> > Content-Type: application/sdp
> > Content-Length: 142
> >
> > message 10
> > INVITE sips:bob at biloxi.example.com SIP/2.0
> > Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bKnashds10
> > Supported: timer
> > Session-Expires: 4000
> > Min-SE: 4000
> > Max-Forwards: 70
> > To: Bob <sips:bob at biloxi.example.com>
> > From: Alice <sips:alice at atlanta.example.com>;tag=1928301774
> > Call-ID: a84b4c76e66710
> > CSeq: 314161 INVITE
> > Contact: <sips:alice at pc33.atlanta.example.com>
> > Content-Type: application/sdp
> >
> > However, as per RFC 3261, if an initial INVITE generates a non-2xx final
> > response, that terminates all sessions and all dialogs that were
> created.
> >
> > Hence, these are not re-INVITE messages, rather new INVITE messages and
> > should use a new From tag.
> >
> > Corrected Text
> > --------------
> > message 1
> > INVITE sips:bob at biloxi.example.com SIP/2.0
> > Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bKnashds8
> > Supported: timer
> > Session-Expires: 50
> > Max-Forwards: 70
> > To: Bob <sips:bob at biloxi.example.com>
> > From: Alice <sips:alice at atlanta.example.com>;tag=1928301774
> > Call-ID: a84b4c76e66710
> > CSeq: 314159 INVITE
> > Contact: <sips:alice at pc33.atlanta.example.com>
> > Content-Type: application/sdp
> > Content-Length: 142
> >
> > message 4
> > INVITE sips:bob at biloxi.example.com SIP/2.0
> > Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bKnashds9
> > Supported: timer
> > Session-Expires: 3600
> > Min-SE: 3600
> > Max-Forwards: 70
> > To: Bob <sips:bob at biloxi.example.com>
> > From: Alice <sips:alice at atlanta.example.com>;tag=2568701785
> > Call-ID: a84b4c76e66710
> > CSeq: 314160 INVITE
> > Contact: <sips:alice at pc33.atlanta.example.com>
> > Content-Type: application/sdp
> > Content-Length: 142
> >
> > message 10
> > INVITE sips:bob at biloxi.example.com SIP/2.0
> > Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bKnashds10
> > Supported: timer
> > Session-Expires: 4000
> > Min-SE: 4000
> > Max-Forwards: 70
> > To: Bob <sips:bob at biloxi.example.com>
> > From: Alice <sips:alice at atlanta.example.com>;tag=5647301796
> > Call-ID: a84b4c76e66710
> > CSeq: 314161 INVITE
> > Contact: <sips:alice at pc33.atlanta.example.com>
> > Content-Type: application/sdp
> >
> > Notes
> > -----
> > -----Original Message-----
> > From: Paul Kyzivat (pkyzivat)
> > Sent: Monday, February 09, 2009 10:09 PM
> > To: Muthu ArulMozhi Perumal (mperumal)
> > Cc: Radha Krishna Saragadam (rsaragad); Jonathan Rosenberg (jdrosen);
> Ram
> > Mohan R (rmohanr)
> > Subject: Re: UAS behavior after sending 422 for initial INVITE
> >
> > yes, I think so.
> >
> > Paul
> >
> > Muthu ArulMozhi Perumal (mperumal) wrote:
> > > In section 13 (Example Call Flow) of RFC 4028 the From tag never
> changes
> > > between the initial INVITE message and the subsequent INVITE messages
> > > sent after receiving a 422:
> > >
> > > message 1
> > > INVITE sips:bob at biloxi.example.com SIP/2.0
> > > From: Alice <sips:alice at atlanta.example.com>;tag=1928301774
> > > Call-ID: a84b4c76e66710
> > >
> > > message 4
> > > INVITE sips:bob at biloxi.example.com SIP/2.0
> > > From: Alice <sips:alice at atlanta.example.com>;tag=1928301774
> > > Call-ID: a84b4c76e66710
> > >
> > > message 10
> > > INVITE sips:bob at biloxi.example.com SIP/2.0
> > > From: Alice <sips:alice at atlanta.example.com>;tag=1928301774
> > > Call-ID: a84b4c76e66710
> > >
> > > Is this a bug in the RFC?
> > >
> > > thanks,
> > > Muthu
> > >
> > > |-----Original Message-----
> > > |From: Paul Kyzivat (pkyzivat)
> > > |Sent: Wednesday, February 04, 2009 12:36 AM
> > > |To: Radha Krishna Saragadam (rsaragad)
> > > |Cc: Jonathan Rosenberg (jdrosen); Muthu ArulMozhi Perumal (mperumal);
> > > Ram Mohan R (rmohanr)
> > > |Subject: Re: UAS behavior after sending 422 for initial INVITE
> > > |
> > > |Radha,
> > > |
> > > |It is not a reinvite, because a dialog was never established - the
> > > first
> > > |call failed.
> > > |
> > > |So you are starting a new invite. You can use the same callid, but
> > > |should use a new from-tag.
> > > |
> > > | Thanks,
> > > | Paul
> > > |
> > > |Radha Krishna Saragadam (rsaragad) wrote:
> > > |> Hi Paul
> > > |>
> > > |> My question is for initial INVITE. For initial INVITE if UA
> > > |> receives 422 and UA want to retry INVITE with new value increased
> > > value
> > > |> then what should be the To(with tag), From(with tag) and CallID
> > > values?
> > > |> Is it a Re-INVITE or new a Dialog? Section 7.3 says same value for
> > > |> To,From and CallID
> > > |>
> > > |> Regards
> > > |> S.Radha krishna
> >
> > 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.
> >
> > --------------------------------------
> > RFC4028 (draft-ietf-sip-session-timer-15)
> > --------------------------------------
> > Title : Session Timers in the Session Initiation Protocol
> > (SIP)
> > Publication Date : April 2005
> > Author(s) : S. Donovan, J. Rosenberg
> > Category : PROPOSED STANDARD
> > Source : Session Initiation Protocol
> > Area : Real-time Applications and Infrastructure
> > Stream : IETF
> > Verifying Party : IESG
> > _______________________________________________
> > Sip mailing list https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors at cs.columbia.edu for questions on current sip
> > Use sipping at ietf.org for new developments on the application of sip
> _______________________________________________
> Sip mailing list https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors at cs.columbia.edu for questions on current sip
> Use sipping at ietf.org for new developments on the application of sip