[asterisk-bugs] [Asterisk 0017404]: [patch] [regression] audio delay when bridging calls related to timestamp mismatch

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17404 
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Reported By:                sdolloff
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17404
Category:                   Core/RTP
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Target Version:             1.4.36
Asterisk Version:           SVN 
JIRA:                       SWP-2729 
Regression:                 Yes 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 265613 
Request Review:              
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Date Submitted:             2010-05-26 11:55 CDT
Last Modified:              2011-01-19 10:49 CST
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Summary:                    [patch] [regression] audio delay when bridging calls
related to timestamp mismatch
Description: 
when answering an inbound call, the remote party hears a delay from 1-3
seconds.  The audio is being transmitted, but the rtp timestamps take a
huge jump when the call is answered even though the rtp sequencing is
correct.
This started occurring after 1.4.28.  reproduced with 1.4.30, 1.4.32 and
SVN from 05/25/2010.  This has been reproduced on multiple servers with
multiple handsets and multiple remote endpoints.  
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Relationships       ID      Summary
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related to          0016941 SIP RTP audio delay
related to          0015824 Incoming Only Latency And Jitters every...
related to          0017007 [patch] RTP Timestamp changes after tra...
related to          0015642 [patch] Fix for Sonus DTMF issues
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 (0130702) sdolloff (reporter) - 2011-01-19 10:49
 https://issues.asterisk.org/view.php?id=17404#c130702 
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i checked both patches from that bug and neither of them resolved the
timestamp discrepancy.  in version 1.6.2.16.1, wireshark indicates jitter
on the call leg (6000-20000ms) sourcing from asterisk between asterisk and
the remote media gateway when the call originates from the lvl3 media
gateway. level3 sees the same jitter and claims that this interferes with
their jitter buffers. this was not the case before this patch was reverted. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-01-19 10:49 sdolloff       Note Added: 0130702                          
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