[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 01:30:43 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 01:30 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 01:30
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Dan, it's very unlikely, that phone (connected to callmanager and then to
asterisk) will request to receive 160ms frames, and send itself 20ms
frames, that is current situation. When I call to asterisk (phone says
receiving 160ms frames, sends 20ms). When I call from same phone to cisco
gateway, packetization is symetric (20ms rx and 20ms tx).
I will continue to try discover more.
Issue History
Date Modified Username Field Change
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04-11-08 01:30 pj Note Added: 0085334
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