[asterisk-bugs] [Asterisk 0012415]: chan_h323 doesn't respect rtp packetization settings

[Asterisk Bug Tracker]

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:             2008-04-10 17:01 CDT
Last Modified:              2008-08-17 13:57 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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 (0091496) pj (reporter) - 2008-08-17 13:57
 http://bugs.digium.com/view.php?id=12415#c91496 
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can we continue to look at this issue? It still persist in current 
Asterisk SVN-trunk-r138482
I think we are close to find source of this issue and because chan_h323 is
only one h323 channel supported by asterisk (chan_ooh323 is unsupported
now), we should to try to remove this ugly bug.
I'm ready to help with testing/debugging/packet capturing :o) 

Issue History 
Date Modified    Username       Field                    Change               
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2008-08-17 13:57 pj             Note Added: 0091496                          
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