[asterisk-bugs] [Asterisk 0012670]: some RTP packets sent to NAT IP instead of public IP; breaks built-in jitterbuffer on some phones
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Nov 19 16:24:50 CST 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=12670
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Reported By: jolan
Assigned To:
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Project: Asterisk
Issue ID: 12670
Category: Core/RTP
Reproducibility: sometimes
Severity: major
Priority: normal
Status: new
Asterisk Version: 1.4.19-rc3
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 2008-05-16 15:55 CDT
Last Modified: 2008-11-19 16:24 CST
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Summary: some RTP packets sent to NAT IP instead of public
IP; breaks built-in jitterbuffer on some phones
Description:
I am experiencing the same symptoms as http://bugs.digium.com/view.php?id=12566
but with 1.4.20-rc3, not
trunk. The bug was fixed in trunk, but not in 1.4.x.
To reiterate the problem:
Phone 100 calls phone 102. Phone 102 answers and starts counting "1 2 3 4
5". Phone 100 doesn't hear anything until "3" or "4".
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(0095116) otherwiseguy (administrator) - 2008-11-19 16:24
http://bugs.digium.com/view.php?id=12670#c95116
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Ok, I've tried to reproduce this with two polycoms running 3.0.1.0032 and I
can not reproduce this. I've tried w/ a NAT, w/o a NAT, adding inbound and
outbound latency to the network, and I just cannot reproduce this. If you
run directly off of the 1.4svn branch, do you still get this problem?
Also, unless you have a really good reason to, you shouldn't Answer()
before the Dial(). Also, under most circumstances, you don't need to pass
the 'r' option to dial either.
Issue History
Date Modified Username Field Change
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2008-11-19 16:24 otherwiseguy Note Added: 0095116
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