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

noreply at bugs.digium.com noreply at bugs.digium.com
Tue Nov 6 12:10:23 CST 2007


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=10665 
====================================================================== 
Reported By:                rjain
Assigned To:                oej
====================================================================== 
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:              
====================================================================== 
Date Submitted:             09-07-2007 03:43 CDT
Last Modified:              11-06-2007 12:10 CST
====================================================================== 
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.
====================================================================== 

---------------------------------------------------------------------- 
 rjain - 11-06-07 12:10  
---------------------------------------------------------------------- 
russell, Thanks for fixing the build issue.

oej, I'm uploading a diff file that has some comments on what the
proc_422_rsp() function does. We could make this logic inline in the
handle_response_invite() similar to how the logic for handling other
responses is all inline in handle_response_invite(). My only goal behind
creating proc_422_rsp() was to not clutter up handle_response_invite() any
further. We can change the name of the function to anything we want or make
it inline. 

Regarding the call to ast_string_field_set() below case 422:, I was trying
to model 422 processing somewhat like the way we currently process 401/407
based on their similarities. Both 422 and 401/407 result into generation of
a new INVITE. The 401/407 case also calls the ast_string_field_set()
function. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
11-06-07 12:10  rjain          Note Added: 0073207                          
======================================================================




More information about the asterisk-bugs mailing list