[asterisk-bugs] [Asterisk 0012415]: chan_h323 doesn't respect rtp packetization settings
noreply at bugs.digium.com
noreply at bugs.digium.com
Fri Apr 11 16:02:05 CDT 2008
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: new
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 113980
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 04-10-2008 17:01 CDT
Last Modified: 04-11-2008 16:02 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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pj - 04-11-08 16:02
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if I replace line with ms = format.cur_ms;
I got p2p bridging, but phone statistics still shows 160ms received and
20ms send, and sound clicking/pops periodically in interval about 3-5s
(after every this click rx/tx packet counter on phone statistic resets (it
counts up to about 200 packets, then restarts from null).
I will post packet capture between asterisk and callmanager.
Format increment: 10
GetTxFramesInPacket reports 16 frames
-- Packet2Packet bridging H323/ip$192.168.40.7:53700/687 and
SIP/ipbx-gw-08209b60
== Spawn extension (from-ccm, 968, 1) exited non-zero on
'H323/ip$192.168.40.7:53700/687'
Issue History
Date Modified Username Field Change
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04-11-08 16:02 pj Note Added: 0085373
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