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

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Jan 22 18:28:50 CST 2009


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12415 
====================================================================== 
Reported By:                pj
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   12415
Category:                   Channels/chan_h323
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 113980 
Request Review:              
====================================================================== 
Date Submitted:             2008-04-10 17:01 CDT
Last Modified:              2009-01-22 18:28 CST
====================================================================== 
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

====================================================================== 

---------------------------------------------------------------------- 
 (0098510) Tursiops (reporter) - 2009-01-22 18:28
 http://bugs.digium.com/view.php?id=12415#c98510 
---------------------------------------------------------------------- 
I've tried patching using the wget command, but it failed. Apparently a
version difference, as there is no "AST_FORMAT_G726_AAL2" in mine at all
(which came with Asterisk 1.4.22)
So I had to remove the non-matching lines from the G726 line downward and
update the Hunk http://bugs.digium.com/view.php?id=2 header line or whatever it
is called (sorry, not a
coder/patcher) from '@@ -1934,43 +1933,35 @@' to '@@ -1934,35 +1933,28 @@'.
After that the patch worked for me as well.

Compiling gave me a warning though:
ast_h323.cxx: In member function âvoid
MyH323Connection::SetCapabilities(int, int, void*, int)â:
ast_h323.cxx:1852: warning: unused variable âmax_frames_per_packetâ

So I removed the offending "int max_frames_per_packet;" line causing it
and the warning disappeared.

Testcalls worked fine.
But as mentioned before, we're using G729 only (requirement by our
provider), so I can't test the other codecs.

Thanks for the patch! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-22 18:28 Tursiops       Note Added: 0098510                          
======================================================================




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