[asterisk-bugs] [Asterisk 0016941]: SIP RTP audio delay
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Mar 17 10:45:13 CDT 2010
A NOTE has been added to this issue.
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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: feedback
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:
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Date Submitted: 2010-03-01 20:42 CST
Last Modified: 2010-03-17 10:45 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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(0119512) lmadsen (administrator) - 2010-03-17 10:45
https://issues.asterisk.org/view.php?id=16941#c119512
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Without the information requested, we will be unable to move this issue
forward, and thus we will need to close it as suspended shortly.
Issue History
Date Modified Username Field Change
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2010-03-17 10:45 lmadsen Note Added: 0119512
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