[asterisk-bugs] [Asterisk 0014021]: RTP playout does not match ptime

Asterisk Bug Tracker noreply at bugs.digium.com
Mon May 11 08:03:27 CDT 2009


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14021 
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Reported By:                Skavin
Assigned To:                kpfleming
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Project:                    Asterisk
Issue ID:                   14021
Category:                   Core/RTP
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.4.21.2 
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:             2008-12-04 16:02 CST
Last Modified:              2009-05-11 08:03 CDT
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Summary:                    RTP playout does not match ptime
Description: 
when a sip client invites with a alaw and ptime of 30. Asterisk sends RTP
at intervals of 20 and 40 ms as captured by tcpdump on the asterisk
server.
this is causing 20ms jitter on these connections.

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---------------------------------------------------------------------- 
 (0104518) Jahrome (reporter) - 2009-05-11 08:03
 http://bugs.digium.com/view.php?id=14021#c104518 
---------------------------------------------------------------------- 
Hello,

I can confirm I have this kind of problem too. In addition, I have seen
other people reporting this problem on French Forums as French VoIP
providers impose ptime of 30ms which envolves compatibility issues with non
ptime configurable clients.

Cheers, 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-05-11 08:03 Jahrome        Note Added: 0104518                          
======================================================================




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