[asterisk-bugs] [Asterisk 0010665]: [patch] SIP Session-Timers Support in Asterisk

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10665 
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Reported By:                rjain
Assigned To:                
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Project:                    Asterisk
Issue ID:                   10665
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.4.11  
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 81013 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             09-07-2007 03:43 CDT
Last Modified:              10-11-2007 00:25 CDT
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Summary:                    [patch] SIP Session-Timers Support in Asterisk
Description: 
The Asterisk SIP stack currently does not support SIP Session-Timers (RFC
4028). This leads to defunct SIP sessions in Asterisk when calls do not
clear through normal signaling procedures due to network or end-point
failures.

John Todd recently discussed this concept on asterisk-dev:
http://lists.digium.com/pipermail/asterisk-dev/2007-July/028574.html

John Todd, JR Richardson and Kevin Fleming have expressed interest in
seeing this feature supported in Asterisk.

A software design document for this feature and code changes (unified diff
of chan_sip.c) are attached to this report. Digium has my code submission
agreement on file.
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---------------------------------------------------------------------- 
 jtodd - 10-11-07 00:25  
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loloski : If you are testing failure modes, then you can disconnect the
ethernet from any of the compnents and wait for the timeout (whatever you
have it set to) and the call should disconnect.  Make sure you have
Asterisk configured such that the media flows between endpoints, or if you
do have RTP media flowing through the Asterisk servers, make sure you have
the "rtptimeout" settings deactivated otherwise the call will hang up for
reasons other than the session-timers. 

Issue History 
Date Modified   Username       Field                    Change               
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10-11-07 00:25  jtodd          Note Added: 0071797                          
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