[asterisk-bugs] [Asterisk 0017404]: 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:                     new
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
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:              2010-05-26 12:00 CDT
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Summary:                    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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 (0122502) sdolloff (reporter) - 2010-05-26 12:00
 https://issues.asterisk.org/view.php?id=17404#c122502 
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if dlstest2_0526-4.pcap, if you look at the rtp stream sourcing from
209.242.51.5 11314 destination 209.242.32.134 16010 you will see at packet
4157, sequence number 9073 the jitter goes from .018 to 25381.99 ms and
stays there. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-05-26 12:00 sdolloff       Note Added: 0122502                          
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