[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:              2008-12-04 19:12 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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 (0095819) putnopvut (administrator) - 2008-12-04 19:12
 http://bugs.digium.com/view.php?id=14021#c95819 
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Looking at the code, it appears that one thing that may affect the ability
for this to work properly would be the "autoframing" option in sip.conf.
You can set this either in the general section or per-peer. The default for
this setting is for it to be off. If this is not already turned on, please
set "autoframing=yes" either for this SIP peer or in the general section
and see if this corrects the problem.

I'll look for other potential problem areas in case this is not enough to
fix the issue. 

Issue History 
Date Modified    Username       Field                    Change               
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2008-12-04 19:12 putnopvut      Note Added: 0095819                          
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