[asterisk-bugs] [Asterisk 0017404]: [patch] [regression] audio delay when bridging calls related to timestamp mismatch
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Feb 21 10:37:09 CST 2011
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: confirmed
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: 2011-02-21 10:37 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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(0132248) sdolloff (reporter) - 2011-02-21 10:37
https://issues.asterisk.org/view.php?id=17404#c132248
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i didn't run with 1.4.27 or 1.4.29, but the issue does not occur in 1.4.28.
It also does not occur in 1.4.36. (1.4.36 is the version that included the
patch to revert the changes that happened after 1.4.28. This patch was
then re-reverted after 1.4.36 so now it's broken again)
Issue History
Date Modified Username Field Change
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2011-02-21 10:37 sdolloff Note Added: 0132248
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