[asterisk-bugs] [Asterisk 0017404]: [patch] [regression] audio delay when bridging calls related to timestamp mismatch
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Jan 27 05:52:22 CST 2011
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=17404
======================================================================
Reported By: sdolloff
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 17404
Category: Core/RTP
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
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:
======================================================================
Date Submitted: 2010-05-26 11:55 CDT
Last Modified: 2011-01-27 05:52 CST
======================================================================
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.
======================================================================
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...
related to 0015642 [patch] Fix for Sonus DTMF issues
======================================================================
----------------------------------------------------------------------
(0131104) oej (manager) - 2011-01-27 05:52
https://issues.asterisk.org/view.php?id=17404#c131104
----------------------------------------------------------------------
I have noticed this bug on calls that are coming in from a SIP trunk to
Asterisk A, then forwarded to Asterisk B after playing some prompts in
Asterisk A.
Issue History
Date Modified Username Field Change
======================================================================
2011-01-27 05:52 oej Note Added: 0131104
======================================================================
More information about the asterisk-bugs
mailing list