[asterisk-bugs] [Asterisk 0012415]: chan_h323 doesn't respect rtp packetization settings
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Mar 17 03:22:24 CDT 2009
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=12415
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Reported By: pj
Assigned To:
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Project: Asterisk
Issue ID: 12415
Category: Channels/chan_h323
Reproducibility: always
Severity: minor
Priority: normal
Status: ready for testing
Asterisk Version: SVN
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 113980
Request Review:
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Date Submitted: 2008-04-10 17:01 CDT
Last Modified: 2009-03-17 03:22 CDT
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Summary: chan_h323 doesn't respect rtp packetization settings
Description:
chan_h323 ignores codecs payload settings eg. 'allow=g729:20'
h323 trace, when I call from h323 endpoint to asterisk:
Found peer capability G.729 <1>, Asterisk code is 256, frame size (in ms)
is 160
If I call from asterisk to another endpoing (eg. cisco gw), trace shows,
that is using 20ms g729 frame size, but still doesn't invoke p2p bridging
between sip and h323 channel
sip--->(g729/chan_sip)-asterisk-(chan_h323/g729)--->callmanager/(cisco gw
or cisco phone)
only g729 is allowed in h323 and sip config
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----------------------------------------------------------------------
(0101849) tusar (reporter) - 2009-03-17 03:22
http://bugs.digium.com/view.php?id=12415#c101849
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SIP(UA)-->(SIP)Asterisk(H323)-->H323GW
h323.conf
=========
disallow=all
allow=g729:40
With patch "2009012200_h323packetization.diff.txt" , it is working as
expected :-) .
//Tusar\\
Issue History
Date Modified Username Field Change
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2009-03-17 03:22 tusar Note Added: 0101849
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