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

[noreply at bugs.digium.com]

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:              04-12-2008 15:38 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 - 04-12-08 15:38  
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Dan, you can open supplied packet dump with free packet analyzer
wireshark.
Anyway, I investigated packed dump again...
seems, that you are right, that asterisk probably using (in channel from
ast to phone) max capability (16 frames), that is offered at beginning of
call setup from phone/cm to asterisk (g729:16) in capability exchange
message. 
asterisk then using this value in open logical channel message back from
asterisk to phone/cm.
I don't think, this is bug in callmanager, because when I call from same
phone to cisco gw (via exactly configured h323 trunk), rx/tx directions are
established using 20ms frames, regardless, that initial offer is exactly
same, as I wrote above, ie. from phone/cm to gateway is offered g729:16
(but OLC send back from gw to phone is setup with 20ms frames).
I can't try sip trunk between callmanager and asterisk, because cm4.x
support only g711 on sip trunks:( 

Issue History 
Date Modified   Username       Field                    Change               
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04-12-08 15:38  pj             Note Added: 0085409                          
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