[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:                oej
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Project:                    Asterisk
Issue ID:                   10665
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     assigned
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:              12-26-2007 06:57 CST
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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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---------------------------------------------------------------------- 
 rjain - 12-26-07 06:57  
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oej: Just uploaded chan_sip.memalloc. This file contains the logic for
dynamic allocation of session-timers related data in the form of a struct
called sip_st_dlg w/in sip_pvt. I've also wrapped around session-timers
related fields in sip_user and sip_peer into a struct called sip_st_cfg.
I've tested these changes against the test cases documented in the design
doc and also against a couple of commercial SIP end-points.

I believe I've addressed all the comments/feedback received so far. Hoping
that this feature is now ready for inclusion in the trunk. 

Issue History 
Date Modified   Username       Field                    Change               
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12-26-07 06:57  rjain          Note Added: 0075934                          
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