[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:                
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Project:                    Asterisk
Issue ID:                   14021
Category:                   Core/RTP
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.21.2 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-12-04 16:02 CST
Last Modified:              2008-12-07 02:01 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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 (0095936) Skavin (reporter) - 2008-12-07 02:01
 http://bugs.digium.com/view.php?id=14021#c95936 
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I have also had voice mail issues on our linksys spa2102s changing them
from 30 (the linksys default) to 20ms fixed the voice mail issues we had
been seeing. 
We where getting corruption of the audio like packet loss bit in a regular
pattern when listening to voice mails even when connected across a LAN.

It was also tracked to the 40 20 40 stepping of the RTP play out.
I don't think the linksys adaptive jitter buffer likes this sort of
jitter. 

Issue History 
Date Modified    Username       Field                    Change               
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2008-12-07 02:01 Skavin         Note Added: 0095936                          
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