[asterisk-bugs] [Asterisk 0017404]: [patch] [regression] audio delay when bridging calls related to timestamp mismatch
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Aug 3 12:31:48 CDT 2010
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: 2010-08-03 12:31 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
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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...
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(0125488) jpeeler (administrator) - 2010-08-03 12:31
https://issues.asterisk.org/view.php?id=17404#c125488
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Try the patch I attached which applies to the latest 1.4.
Issue History
Date Modified Username Field Change
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2010-08-03 12:31 jpeeler Note Added: 0125488
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