[asterisk-bugs] [Asterisk 0012566]: Loss of audio after 2 seconds of P2P RTP bridging
noreply at bugs.digium.com
noreply at bugs.digium.com
Wed May 7 09:15:25 CDT 2008
The following issue requires your FEEDBACK.
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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: 05-07-2008 09:15 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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file - 05-07-08 09:15
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1. Has a separate SIP destination been tested to ensure that it is
happening on all SIP dialogs, and not just ones to this Cisco?
2. Are the filenames correct? The one from trunk does show RTP packets
being forwarded in the P2P layer while the 1.4 one does not.
3. We will need a core show locks to examine the SIP channel locking
issue, it may be the cause of the loss of audio.
4. Core debug would also be useful. debug to console in logger.conf and
core set debug 10
Issue History
Date Modified Username Field Change
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05-07-08 09:15 file Note Added: 0086532
05-07-08 09:15 file Status new => feedback
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