[asterisk-bugs] [Asterisk 0014511]: SIP REINVITE broken in 1.6 (was working in 1.4.13)

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14511 
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Reported By:                kebl0155
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14511
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Target Version:             1.6.0.8
Asterisk Version:           1.6.0.3 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-02-19 12:26 CST
Last Modified:              2009-03-04 14:38 CST
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Summary:                    SIP REINVITE broken in 1.6 (was working in 1.4.13)
Description: 
Hi there

We take SIP calls from a variety of different gateways from a single
provider.

With 1.4.13, we were able to use canreinvite to drop out of the audio
stream.

With 1.6.0.3, reinvite results in no audio with some of their gateways,
and not others.

There is at least one difference in the SIP conversation between the two
gateways.
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 (0101223) mmichelson (administrator) - 2009-03-04 14:38
 http://bugs.digium.com/view.php?id=14511#c101223 
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The biggest difference I'm noticing between the working and non-working
scenarios is in codecs in the SDPs of the reinvites. In both working
scenarios, the only codec in the offers and answers is alaw. In the
non-working scenario, Asterisk offers gsm, ulaw, and alaw to one party (and
receives an answer with gsm only) and offers only alaw to the other party.
It may be this codec mismatch which is causing the problem to occur. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-03-04 14:38 mmichelson     Note Added: 0101223                          
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