[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 19:08 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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 (0100868) Skavin (reporter) - 2009-02-26 19:08
 http://bugs.digium.com/view.php?id=14021#c100868 
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I think it would affect bridged apps
The First time I ever saw this issue was with a linksys on 30ms listening
to voicemail. It had regular clipping. Since then all our managed devices
have been hard set to 20ms and they then play happily with asterisk. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-02-26 19:08 Skavin         Note Added: 0100868                          
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