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Summary:ASTERISK-16590: asterisk 1.4.3x codec packetization issue
Reporter:para (para)Labels:
Date Opened:2010-08-19 04:12:34Date Closed:2014-08-12 11:19:01
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_h323
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:please check ****** line,

when we make h323 calls out, with faststart=yes,
the packetization is too large in their open logical channel and does not match their Terminal Capabilty Set (TCS) for G.729
 
                           OpenLogicalChannel
                               forwardLogicalChannelNumber: 1
                               forwardLogicalChannelParameters
                                   dataType: nullData (1)
                                       nullData: NULL
                                   multiplexParameters: none (4)
                                       none: NULL
                               reverseLogicalChannelParameters
                                   dataType: audioData (3)
                                       audioData: g729 (10)
******                                      g729: 20
:
:
:
:
                                               Item 1
                                                   CapabilityTableEntry
                                                       capabilityTableEntryNumber: 2
                                                       capability: receiveAudioCapability (4)
                                                           receiveAudioCapability: g729 (10)
                                                               g729: 2
 
Their Open logical channel packetization should be set to 2 (like item 1 in their TCS), not 20.

test with 1.4.30 and 1.4.35, same issue



****** ADDITIONAL INFORMATION ******

sip.conf
[1000]
type=friend
secret=1000
qualify=yes
port=5080
nat=yes
host=dynamic
dtmfmode=rfc2833
context=rtx-35
canreinvite=no
callerid=<1000>
incominglimit=4
disallow=all
allow=g729
allow=g723
allow=ulaw
allow=gsm
;allow=all


extensions.conf


[rtx-35]
exten => _X.,1,NoOp(rtx-35 start)
exten => _X.,n,Macro(dial,H323,${EXTEN},rtx-35)
exten => _X.,n,Hangup()

[macro-dial]
exten => s,1,Dial(${ARG1}/${ARG3}/${ARG2},80)
exten => s,n,Hangup


h323.conf
[rtx-35]
type=peer
context=rtx-35
host=10.100.0.35
port=1720
qualify=yes
faststart=yes
tunneling=yes
dtmfcode=rfc2833
incominglimit=0
disallow=all
allow=g729
allow=g723
;allow=ulaw
;autoframing=yes
Comments:By: para (para) 2010-08-19 04:14:04

please check wireshark capture
By: Leif Madsen (lmadsen) 2010-08-24 12:55:10

Note that this channel driver is not well supported. You may have better luck with using 1.6.2.x and the chan_ooh323 driver in Asterisk-Addons which is supported by a community developer.
By: Matt Jordan (mjordan) 2014-08-12 11:18:56.055-0500

Unfortunately, {{chan_h323}} has been unsupported for a long time and was marked deprecated. Due to there being a supported replacement channel driver, {{chan_ooh323}}, {{chan_h323}} was removed from the Asterisk source tree in Asterisk 13:

http://lists.digium.com/pipermail/asterisk-dev/2014-June/068363.html

As such, this issue will be closed out as "Won't Fix". I recognize that this issue has been open for a long time, and as a project, we're sorry that the patch attached this issue didn't receive more attention. We've made some changes in the way the patch process works to try and alleviate the issue of patches not receiving attention; for more information, see [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process] on the Asterisk wiki. If you still need H323 support and are using Asterisk, I'd recommend switching to {{chan_ooh323}}, as it has a dedicated module maintainer who can help with patches to that channel driver.