[asterisk-bugs] [Asterisk 0011901]: bridging chan_h323 and chan_sip

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11901 
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Reported By:                pj
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   11901
Category:                   Core/Channels
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     assigned
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 101746 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             02-01-2008 10:08 CST
Last Modified:              04-10-2008 10:29 CDT
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Summary:                    bridging chan_h323 and chan_sip
Description: 
even if I using same codec (disallow=all, allow=g729) on both channels, I
never see 'Packet2Packet bridging...' message, like in sip-sip, sip-skinny,
skinny-skinny bridging.
I think, that h323 is using same RTP subsystem, so Packet2Packet bridging
should be possible to avoid RTP go through asterisk core, like in other
cases.

also some unnecessary warnings about codecs are displayed when calling
h323->sip, reverse direction sip->h323 is without warnings.
since both my ends supports g729, I have no g729 installed and expecting,
that asterisk will do simple passthrough. In general it works, but imho, it
should work better, with Packet2Packet bridging and avoid warnings below.

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---------------------------------------------------------------------- 
 pj - 04-10-08 10:29  
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I probably found source of issue, it's rtp packetization settings:
when I set 'allow=g729:60' on SIP side, Packet2Packet bridging between SIP
and H323 is working. When I leave simply 'allow=g729' on both SIP and H323
side p2p bridging is not working. 
Setting 'allow=g729:20' on H323 side (to be same as is by default for g729
on sip side) has no affect, because h323 end device still reports, that is
receiving 60ms samples.
Seems, that something is broken with rtp packetization settings on for
h323 channel. 

Issue History 
Date Modified   Username       Field                    Change               
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04-10-08 10:29  pj             Note Added: 0085302                          
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