[asterisk-bugs] [Asterisk 0017404]: [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:                     acknowledged
Asterisk Version:           SVN 
JIRA:                       SWP-1582 
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:              2010-07-01 16:37 CDT
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Summary:                    [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...
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 (0124170) sdolloff (reporter) - 2010-07-01 16:37
 https://issues.asterisk.org/view.php?id=17404#c124170 
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this seems related to this bug.
https://issues.asterisk.org/view.php?id=17007 

Issue History 
Date Modified    Username       Field                    Change               
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2010-07-01 16:37 sdolloff       Note Added: 0124170                          
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