[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:                jpeeler
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Project:                    Asterisk
Issue ID:                   17404
Category:                   Core/RTP
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Target Version:             1.4.36
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:              2010-08-11 22:00 CDT
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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
----------------------------------------------------------------------
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...
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 (0125868) svnbot (reporter) - 2010-08-11 22:00
 https://issues.asterisk.org/view.php?id=17404#c125868 
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Repository: asterisk
Revision: 281911

U   branches/1.4/main/channel.c

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r281911 | jpeeler | 2010-08-11 22:00:12 -0500 (Wed, 11 Aug 2010) | 20
lines

Ensure SSRC is changed when media source is changed to resolve audio
delay.

This change causes the SSRC to change right before the channels are
bridged,
which is what used to happen. It seems that fixes were made to attempt
limiting
SSRC changes, targeted mainly at sending DTMF. DTMF is not affecting the
SSRC
with this change.

There are two other control frames sent in ast_channel_bridge that
probably
should also be changed to AST_CONTROL_SRCCHANGE as well, but I'm going to
leave
this change up to the discretion of resolving issue
https://issues.asterisk.org/view.php?id=17007.

For reference - old review implementing new control frame SRCCHANGE:
https://reviewboard.asterisk.org/r/540

(closes issue https://issues.asterisk.org/view.php?id=17404)
Reported by: sdolloff
Patches: 
      bug17404.patch uploaded by jpeeler (license 325)
Tested by: sdolloff

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http://svn.digium.com/view/asterisk?view=rev&revision=281911 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-08-11 22:00 svnbot         Checkin                                      
2010-08-11 22:00 svnbot         Note Added: 0125868                          
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