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

[Asterisk Bug Tracker]

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:              2009-02-26 13:57 CST
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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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 (0100838) Skavin (reporter) - 2009-02-26 13:57
 http://bugs.digium.com/view.php?id=14021#c100838 
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Thank you for looking at this I am sure other people are getting this and
don't know it.
I think the linksys jitter buffer is buffering for an average of the
jitter not max.
In this case any adaptive algorithm that does not use max will fail most
of the time. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-02-26 13:57 Skavin         Note Added: 0100838                          
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