[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:              04-10-2008 18:44 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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---------------------------------------------------------------------- 
 DEA - 04-10-08 18:44  
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The packetization feature in Asterisk allow it to choose what
it should send.  It does not effect what the client chooses to
send.

The client MIGHT choose to honor the requested framing, or it
might just stick to what it prefers.

As you have discovered, both parties need to use the same codec
and same framing (packetization) to allow P2P bridging. 

Issue History 
Date Modified   Username       Field                    Change               
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04-10-08 18:44  DEA            Note Added: 0085328                          
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