[asterisk-bugs] [Asterisk 0011157]: Asterisk does not send a provisional response at every minute

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11157 
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Reported By:                rjain
Assigned To:                
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Project:                    Asterisk
Issue ID:                   11157
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   trivial
Priority:                   normal
Status:                     closed
Asterisk Version:           1.6.1-beta1 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
Resolution:                 suspended
Fixed in Version:           
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Date Submitted:             2007-11-04 16:29 CST
Last Modified:              2008-12-16 16:32 CST
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Summary:                    Asterisk does not send a provisional response at
every minute
Description: 
The issue here is that Asterisk does not send a non-100 response at every
minute for calls that are in the provisional response state. This causes
most UACs and/or proxies to terminate the call after 3 minutes. There are
many legitimate reasons why a call could remain in an unanswered state for
more than 3 minutes such as early-media (183), call queuing (182), call
forwarding (181) and ringing (180). 

Following is quote from section 13.3.1.1 of RFC 3261 which explains what a
UAS should do under such a circumstance:

   If the UAS desires an extended period of time to answer the INVITE,
   it will need to ask for an "extension" in order to prevent proxies
   from canceling the transaction.  A proxy has the option of canceling
   a transaction when there is a gap of 3 minutes between responses in a
   transaction.  To prevent cancellation, the UAS MUST send a non-100
   provisional response at every minute, to handle the possibility of
   lost provisional responses.

This issue was first reported by Alex Balashov on the asterisk-dev mailing
list:
http://lists.digium.com/pipermail/asterisk-dev/2007-November/030341.html.
I've reproduced this problem and collected wireshark and debug traces,
which are attached to this bug report.

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---------------------------------------------------------------------- 
 (0096501) svnbot (reporter) - 2008-12-16 16:32
 http://bugs.digium.com/view.php?id=11157#c96501 
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Repository: asterisk
Revision: 164941

U   trunk/channels/chan_sip.c

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r164941 | twilson | 2008-12-16 16:32:12 -0600 (Tue, 16 Dec 2008) | 2 lines

Make a note of the feature request in bug
http://bugs.digium.com/view.php?id=11157 as per the reporter and
oej, and suspend the bug since no one seems to be keen on implementing it
any time soon.

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http://svn.digium.com/view/asterisk?view=rev&revision=164941 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-12-16 16:32 svnbot         Checkin                                      
2008-12-16 16:32 svnbot         Note Added: 0096501                          
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