[asterisk-bugs] [Asterisk 0016941]: SIP RTP audio delay

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16941 
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Reported By:                sharvanek
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16941
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.29.1 
JIRA:                       SWP-990 
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:             2010-03-01 20:42 CST
Last Modified:              2010-03-03 13:30 CST
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Summary:                    SIP RTP audio delay
Description: 
This is *not* a latency issue.

Here's a full description:
http://forums.digium.com/viewtopic.php?f=1&t=73267&start=0&sid=404b57776485f8f2e376bfaf1d394dd5

This is a very very odd issue, when a caller comes in through two asterisk
boxes via SIP the IVR is heard fine, but when transferred from the IVR the
caller and the person being called experience about 2-3 seconds of silence
before audio begins to flow.

This has been reported numerous times across the net but no real solution,
some examples are:

http://www.mail-archive.com/asterisk@uc.org/msg05938.html
http://www.trixbox.org/forums/trixbox-forums/help/3-second-delay-answering-calls

sadly there isn't much debug here, a packet capture shows RTP flowing etc
the same in both circumstances, please advise.


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 (0118891) lmadsen (administrator) - 2010-03-03 13:30
 https://issues.asterisk.org/view.php?id=16941#c118891 
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Please provide a packet capture of the call from start to finish. This
would include:

 * SIP trace from the Asterisk console
 * Asterisk console with debug level logging
 * SIP history

Also what might be useful would be a full pcap with the RTP so the call
can be played back from the start which demonstrates this issue.

You should also provide the relevant parts of your sip.conf file. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-03-03 13:30 lmadsen        Note Added: 0118891                          
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