[asterisk-bugs] [Asterisk 0009431]: Modify connection: Response 491 not handled according to RFC3261

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=9431 
====================================================================== 
Reported By:                alex-911
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   9431
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.4.2 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        No 
Request Review:              
====================================================================== 
Date Submitted:             03-30-2007 12:41 CDT
Last Modified:              10-11-2007 14:47 CDT
====================================================================== 
Summary:                    Modify connection: Response 491 not handled
according to RFC3261
Description: 
when asterisk receives a reINVITE while a reINVITE from it's own is still
in progress, it answers with a 491 which is basically correct. but in the
RFC, there is a UAC behaviour described how a client should retry to modify
a connection. what asterisk is doing here is already providing call release
codes in the 491 and terminating the session with a BYE afterwards.
according to the RFC a timer should be fired at each client (times
depending who owns the call ID) and the modification should be retried.

find the log attached for a session modification from G.711 to T.38

======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0010868 1.4.11 Stable - Polycom phones hang up ...
====================================================================== 

---------------------------------------------------------------------- 
 rjain - 10-11-07 14:47  
---------------------------------------------------------------------- 
Regarding the previous comment made about using the dialplan to initiate
the next call setup, I think that is probably not the right solution for
this problem. The issue here is RE-INVITE glare. RE-INVITEs happen
mid-session so the call is already up from the dialplan perspective. I'd
imagine this will have to handled within chan_sip itself and in fact a 491
occurrence should be transparent to the dialplan. Also, as a rule of the
SIP protocol a RE-INVITE can fail but that doesn't effect the original
call.

By the way, this is a duplicate of 10481. Either 10481 or this bug report
should be closed as duplicate.

http://bugs.digium.com/view.php?id=10481 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
10-11-07 14:47  rjain          Note Added: 0071833                          
======================================================================