-----Original Message-----
From: [email protected] [mailto:[email protected]] On Behalf
Of RFC
Errata System
Sent: Monday, February 09, 2009 12:37 PM
To: [email protected]; [email protected]; [email protected];
[email protected]; [email protected]; [email protected]
Cc: [email protected]; [email protected]; [email protected]
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 <[email protected]>
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:[email protected] 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:[email protected]>
From: Alice <sips:[email protected]>;tag=1928301774
Call-ID: a84b4c76e66710
CSeq: 314159 INVITE
Contact: <sips:[email protected]>
Content-Type: application/sdp
Content-Length: 142
message 4
INVITE sips:[email protected] 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:[email protected]>
From: Alice <sips:[email protected]>;tag=1928301774
Call-ID: a84b4c76e66710
CSeq: 314160 INVITE
Contact: <sips:[email protected]>
Content-Type: application/sdp
Content-Length: 142
message 10
INVITE sips:[email protected] 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:[email protected]>
From: Alice <sips:[email protected]>;tag=1928301774
Call-ID: a84b4c76e66710
CSeq: 314161 INVITE
Contact: <sips:[email protected]>
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:[email protected] 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:[email protected]>
From: Alice <sips:[email protected]>;tag=1928301774
Call-ID: a84b4c76e66710
CSeq: 314159 INVITE
Contact: <sips:[email protected]>
Content-Type: application/sdp
Content-Length: 142
message 4
INVITE sips:[email protected] 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:[email protected]>
From: Alice <sips:[email protected]>;tag=2568701785
Call-ID: a84b4c76e66710
CSeq: 314160 INVITE
Contact: <sips:[email protected]>
Content-Type: application/sdp
Content-Length: 142
message 10
INVITE sips:[email protected] 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:[email protected]>
From: Alice <sips:[email protected]>;tag=5647301796
Call-ID: a84b4c76e66710
CSeq: 314161 INVITE
Contact: <sips:[email protected]>
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:[email protected] SIP/2.0
From: Alice <sips:[email protected]>;tag=1928301774
Call-ID: a84b4c76e66710
message 4
INVITE sips:[email protected] SIP/2.0
From: Alice <sips:[email protected]>;tag=1928301774
Call-ID: a84b4c76e66710
message 10
INVITE sips:[email protected] SIP/2.0
From: Alice <sips:[email protected]>;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 [email protected] for questions on current sip
Use [email protected] for new developments on the application of sip