[asterisk-bugs] [Asterisk 0016941]: SIP RTP audio delay
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu May 27 09:09:04 CDT 2010
The following issue has been set as RELATED TO issue 0017404.
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https://issues.asterisk.org/view.php?id=16941
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Reported By: sharvanek
Assigned To:
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Project: Asterisk
Issue ID: 16941
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: closed
Asterisk Version: 1.4.29.1
JIRA: SWP-990
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
Resolution: suspended
Fixed in Version:
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Date Submitted: 2010-03-01 20:42 CST
Last Modified: 2010-05-27 09:09 CDT
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Summary: SIP RTP audio delay
Description:
This is *not* a latency issue.
Here's a full description:
http://forums.digium.com/viewtopic.php?f=1&t=73267&start=0&sid=404b57776485f8f2e376bfaf1d394dd5
This is a very very odd issue, when a caller comes in through two asterisk
boxes via SIP the IVR is heard fine, but when transferred from the IVR the
caller and the person being called experience about 2-3 seconds of silence
before audio begins to flow.
This has been reported numerous times across the net but no real solution,
some examples are:
http://www.mail-archive.com/asterisk@uc.org/msg05938.html
http://www.trixbox.org/forums/trixbox-forums/help/3-second-delay-answering-calls
sadly there isn't much debug here, a packet capture shows RTP flowing etc
the same in both circumstances, please advise.
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Relationships ID Summary
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related to 0017404 audio delay when bridging calls related...
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Issue History
Date Modified Username Field Change
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2010-05-27 09:09 lmadsen Relationship added related to 0017404
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