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

noreply at bugs.digium.com noreply at bugs.digium.com
Fri Apr 11 16:02:05 CDT 2008


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:                
====================================================================== 
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:              
====================================================================== 
Date Submitted:             04-10-2008 17:01 CDT
Last Modified:              04-11-2008 16:02 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

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

---------------------------------------------------------------------- 
 pj - 04-11-08 16:02  
---------------------------------------------------------------------- 
if I replace line with ms = format.cur_ms; 
I got p2p bridging, but phone statistics still shows 160ms received and
20ms send, and sound clicking/pops periodically in interval about 3-5s
(after every this click rx/tx packet counter on phone statistic resets (it
counts up to about 200 packets, then restarts from null).
I will post packet capture between asterisk and callmanager.

Format increment: 10
GetTxFramesInPacket reports 16 frames
    -- Packet2Packet bridging H323/ip$192.168.40.7:53700/687 and
SIP/ipbx-gw-08209b60
  == Spawn extension (from-ccm, 968, 1) exited non-zero on
'H323/ip$192.168.40.7:53700/687' 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
04-11-08 16:02  pj             Note Added: 0085373                          
======================================================================




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