[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 17:32 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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 (0095894) Skavin (reporter) - 2008-12-05 17:32
 http://bugs.digium.com/view.php?id=14021#c95894 
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I am also seeing it sip to sip with one leg ptime 30 and the other 20


A party  ---  ptime 30 --->  Asterisk --- ptime 20 --> bparty

 A tx -> 30 30 30 30          ast tx ->  0 30 0 30 30 0 30 30 0 
         0 20 40 20 40 20 40  <- ast tx  20 20 20 20 20 20       <- B tx

The deltas look like asterisk is sending packets when it gets a packet
in.

Could asterisk not be only sending packets when it receives packets and
not be internally triggering?

looking at the atx leg 
it receives 30 ms so sends 20 and buffers 10
waits 30 gets 30 and send 40 ms in 2 sequential packets

when ever asterisk sends data from a bridged call there has always been a
packet come in within 0.0001 of a second before it does not seem to send
without that incoming packet.
ie delaying a packet 10 ms to play at 30ms multiples when sending from
20ms to 30ms or delaying 20ms when passing data from a 30 to 20 ms call

just trying to rationalize what I am seeing my be totally off on this.

my servers are using ztdummy if that helps as they are only sip to sip. 

Issue History 
Date Modified    Username       Field                    Change               
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2008-12-05 17:32 Skavin         Note Added: 0095894                          
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