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

noreply at bugs.digium.com noreply at bugs.digium.com
Tue Jan 22 09:38:26 CST 2008


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:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.4.13  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             11-04-2007 16:29 CST
Last Modified:              01-22-2008 09:38 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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---------------------------------------------------------------------- 
 oej - 01-22-08 09:38  
---------------------------------------------------------------------- 
This would add yet another scheduled timer to our poor sip stack... Well,
gotta be fixed. No coders? 

Issue History 
Date Modified   Username       Field                    Change               
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01-22-08 09:38  oej            Note Added: 0081019                          
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