[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:              2008-12-05 15:36 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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 (0095891) DEA (reporter) - 2008-12-05 15:36
 http://bugs.digium.com/view.php?id=14021#c95891 
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The issue is not unique to SIP.  Somewhere along the line the 
ast_smoother infrastructure was busted.

I have not figured out exactly how just yet.  Either we are not
allocating the smoother when needed, or not reading/writing from
it when we should.

The channels are negotiating the appropriate framing/packetization,
and the core appears to be trying to honor it (rtp debug will show
the correct size packets, but the timestamps may be off).

I noticed the behaviour for SIP/OOH323 channels joined to a meetme. 

Issue History 
Date Modified    Username       Field                    Change               
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2008-12-05 15:36 DEA            Note Added: 0095891                          
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