[asterisk-bugs] [Asterisk 0013569]: Asterisk sending the wrong codec on re-invite.

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13569 
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Reported By:                bkw918
Assigned To:                mnicholson
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Project:                    Asterisk
Issue ID:                   13569
Category:                   Channels/chan_sip/CodecHandling
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.0-rc6 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.1 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2008-09-26 17:48 CDT
Last Modified:              2009-03-20 17:09 CDT
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Summary:                    Asterisk sending the wrong codec on re-invite.
Description: 
FreeSWITCH sends invite out to tf.voipmich.com with PCMU, PCMA, GSM.  The
call is answered and setup using GSM since its listed first in the Answer
we receive from Asterisk.  A re-invite promptly follows offering
GSM,PCMU,PCMA to which we 200 OK with ONLY GSM in the SDP in our 200 OK.  

Promptly there after Asterisk starts sending PCMU packets.  The re-invite
is considered a new Session Offer Answer and Asterisk ignores the Answer
and sends a media format not in the new Answer.  

Asterisk PBX SVN-branch-1.6.0-r140976-/trunk is what I can see in the SDP
from John's equipment.  He has disabled re-invites pending a fix for this.

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 (0102031) mnicholson (administrator) - 2009-03-20 17:09
 http://bugs.digium.com/view.php?id=13569#c102031 
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Well the audio is no longer coming from Asterisk.  The answer is basically
ignored by asterisk, but asterisk is not sending the audio anymore.  Other
than that, we are saying the same thing.

To fix this, the logic asterisk uses to perform re-invites needs to be
changed.  My first thought was to only reinvite one leg at a time, but that
still would not fix the problem.  Right now, I am thinking that asterisk
should reinvite one leg of the call at a time, and then send additional
reinvites if the media formats or other items in the answer differ from the
last reinvite we sent to the other leg of the call.  This is somewhat
complex, so I need to spend some time thinking about it some more to make
sure all edge cases are covered before I move forward with an
implementation. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-03-20 17:09 mnicholson     Note Added: 0102031                          
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