[asterisk-bugs] [Asterisk 0011062]: Asterisk-1.4.x :: Chan_H323 with Codec-g723 :: Metalic Sound

[noreply at bugs.digium.com]

The following issue has been CLOSED 
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http://bugs.digium.com/view.php?id=11062 
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Reported By:                tusar
Assigned To:                
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Project:                    Asterisk
Issue ID:                   11062
Category:                   Channels/chan_h323
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           1.4.15  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
Resolution:                 no change required
Fixed in Version:           
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Date Submitted:             10-23-2007 05:54 CDT
Last Modified:              12-05-2007 11:41 CST
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Summary:                    Asterisk-1.4.x :: Chan_H323 with Codec-g723 ::
Metalic Sound
Description: 
I am trying to install Asterisk-v.1.4.13 with codec g723 and g729 . I have
downloaded codec g723/g729 from http://asterisk.hosting.lv/ for test .

g729 is running like charm but facing problem with g723 . With g723 codec
Caller side getting metalic/choppy sound where Callee can hear clear.

Here is below of my scenario :
======================
SoftPhone[g723]-> SIP-Asterisk-H323 -> Quintum DX [g723] <=Metalic Sound
SoftPhone[g723]-> H323-Asterisk-H323 -> Quintum DX [g723]<=Metalic Sound

h323.conf
=======
disallow=all
allow=g723.1
allow=g729

*CLI> show channel H323/xx.xx.xx.xx:1720-1
  NativeFormats: 0x1 (g723)
    WriteFormat: 0x1 (g723)
     ReadFormat: 0x1 (g723)
 WriteTranscode: No
  ReadTranscode: No

Result : Getting metalic/choppy sound with the above config but without
transcoding .


If I disable the codec part at the "h323.conf" then getting clear voice
but then it is transcoding and Consuming more CPU power.

h323.conf
=======
;disallow=all
;allow=g723.1
;allow=g729

*CLI> show channel H323/xx.xx.xx.xx:1720-2
  NativeFormats: 0x1 (g723)
    WriteFormat: 0x40 (slin)
     ReadFormat: 0x40 (slin)
 WriteTranscode: Yes
  ReadTranscode: Yes

But for Scenario [Full SIP Path]:
=================================
SoftPhone [g723] -> SIP-Asterisk-SIP -> QuintumGW [g723] <= No problem

There is no problem with clear voice. 

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---------------------------------------------------------------------- 
 Corydon76 - 12-05-07 11:41  
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We don't support outside codecs. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
12-05-07 11:41  Corydon76      Asterisk Version         1.4.13  => 1.4.15   
12-05-07 11:41  Corydon76      Note Added: 0074838                          
12-05-07 11:41  Corydon76      Status                   new => closed       
12-05-07 11:41  Corydon76      Resolution               open => no change
required
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