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

Wed Jan 12 10:45:13 CST 2011

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:                     new
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 10:45 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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 (0130417) joscas (reporter) - 2011-01-12 10:45
 https://issues.asterisk.org/view.php?id=18593#c130417 
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Yes, I'm using two different gtalk accounts.
What I'm doing is this:

phone1 --> LAN  --> Internet --> Google--> Internet--> Asterisk --\
phone2 <-- LAN  <-- Internet <-- Google <-- Internet <----------/

But I think Google does shuffling and it all ends up being:

phone1 --> LAN  --> Internet--> Asterisk --\
phone2 <-- LAN  <-- Internet <----------/

So, If I understand the term correctly, I does look like hairpinning to
me. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-01-12 10:45 joscas         Note Added: 0130417                          
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