[asterisk-bugs] [Asterisk 0012566]: Loss of audio after 2 seconds of P2P RTP bridging
noreply at bugs.digium.com
noreply at bugs.digium.com
Thu May 15 16:48:27 CDT 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=12566
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Reported By: falves11
Assigned To:
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Project: Asterisk
Issue ID: 12566
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 114912
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 05-01-2008 13:23 CDT
Last Modified: 05-15-2008 16:48 CDT
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Summary: Loss of audio after 2 seconds of P2P RTP bridging
Description:
I have a peer like this
[federico]
type=peer
host=72.187.243.97
context=default
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=g729
allow=ulaw
nat=yes
insecure=port,invite
but when I type "sip show peers"
ame/username Host Dyn Nat ACL Port Status
federico 72.187.243.97 N 5060
Unmonitored
and in fact the calls connect and there is second of audio and then the
audio stops. I am behind a nat.
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svnbot - 05-15-08 16:48
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Repository: asterisk
Revision: 116663
U trunk/channels/chan_sip.c
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r116663 | jpeeler | 2008-05-15 16:48:24 -0500 (Thu, 15 May 2008) | 2 lines
Fixes a problem I was having with two SIP phones using Packet2Packet
bridging dropping audio nearly immediately. The problem was that the lock
on the SIP dialog was not being unlocked while the bridge was still active.
(Related to issue http://bugs.digium.com/view.php?id=12566)
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http://svn.digium.com/view/asterisk?view=rev&revision=116663
Issue History
Date Modified Username Field Change
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05-15-08 16:48 svnbot Checkin
05-15-08 16:48 svnbot Note Added: 0086925
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