[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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https://issues.asterisk.org/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:                     acknowledged
Asterisk Version:           SVN 
JIRA:                       SWP-11 
Regression:                 No 
Reviewboard Link:            
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:              2011-04-23 14:03 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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 (0134058) luckman212 (reporter) - 2011-04-23 14:03
 https://issues.asterisk.org/view.php?id=14021#c134058 
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Is this still an issue with Asterisk 1.8.3.3 ?  I think I may be having
this issue with a Linksys/Sipura SPA2102 that has a 30ms jitterbuffer which
apparently CANNOT be disabled, so ptime=20 (0.020) doesn't really make
sense for that device.  I have left it at the default 30ms but I get
"matrix-style" audio from time to time especially in the beginning of a
bridged call between ATA<->PBX (both devices on the same 1Gbit LAN
segment).  

Haven't done extensive RTP debugging but it would seem this might be
related.  I'm on Asterisk 1.8.3.3. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-04-23 14:03 luckman212     Note Added: 0134058                          
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