[asterisk-bugs] [Asterisk 0012148]: RTP timestamp skewed

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12148 
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Reported By:                jcomellas
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   12148
Category:                   Core/RTP
Reproducibility:            sometimes
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.4.18 
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 105676 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             03-05-2008 11:57 CST
Last Modified:              03-05-2008 15:27 CST
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Summary:                    RTP timestamp skewed
Description: 
In SIP calls against PSTN numbers (through Level 3) we have detected that
Asterisk sometimes sends RTP packets with skewed timestamps, causing audio
drops. This issue is very similar to the one on ticket
http://bugs.digium.com/view.php?id=11491, and was not
corrected by the intended fix (r105674 and r104676). In our case it is not
happening when the call is on hold, but when a call is answered. It is
usually the originating caller the one that has the 4 second gap after the
terminating caller answers the phone.

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---------------------------------------------------------------------- 
 jcomellas - 03-05-08 15:27  
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I've uploaded a file with an excerpt of the log of one of our servers where
a call failed, but I don't see much information from the rtp.c source file
that might help. Maybe you can find something else. If you need me to
recompile adding log output to the server I can do it.

Also, we use a custom Asterisk application, but internally it generally
ends up calling normal Asterisk applications. 

Issue History 
Date Modified   Username       Field                    Change               
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03-05-08 15:27  jcomellas      Note Added: 0083486                          
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