[asterisk-bugs] [Asterisk 0018593]: RTCP conflict avoidance not handled

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18593 
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Reported By:                joscas
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18593
Category:                   Channels/chan_gtalk
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.8.1.1 
JIRA:                       SWP-2895 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2011-01-10 08:03 CST
Last Modified:              2011-01-12 10:22 CST
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Summary:                    RTCP conflict avoidance not handled
Description: 
When trying to bridge a Gtalk outgoing call to a Gtalk incoming call that
uses the same IP address it fails.

The error it gives is:
[Jan 10 14:46:56] NOTICE[4520]: res_rtp_asterisk.c:2190 ast_rtp_read:
Unknown RTP codec 73 received from '95.21.226.219:58470'
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 (0130412) lmadsen (administrator) - 2011-01-12 10:22
 https://issues.asterisk.org/view.php?id=18593#c130412 
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It's expected because I wouldn't expect Asterisk to handle hairpinning a
call in Gtalk. I don't think Asterisk will even hairpin a call correctly
with chan_sip (although some work was done on that to permit some sorts of
hairpinning).

This is essentially a feature request so I'm going to close this issue. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-01-12 10:22 lmadsen        Note Added: 0130412                          
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