[asterisk-bugs] [Asterisk 0016288]: G723 codec has digitzed voice

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16288 
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Reported By:                globalnetinc
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16288
Category:                   Codecs/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.1.10 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-11-19 20:04 CST
Last Modified:              2009-11-20 09:25 CST
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Summary:                    G723 codec has digitzed voice
Description: 
I am using an audiocodes mp202b asterisk 1.6.1.10 and a digium tc400b. 
When I call the asterisk box and get prompts all sounds well.  When
asterisk has to bridge the 723 rtp stream to ulaw for the sip provider the
voive becomes very digitzed and poor.

mp202b(723)=> asterisk - works.
mp202b(723)=> asterisk(ulaw)=> sip provider - fails

it produces poor quailty both ways
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 (0114042) globalnetinc (reporter) - 2009-11-20 09:25
 https://issues.asterisk.org/view.php?id=16288#c114042 
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do not know what addtitional info you would need.  this happens 100% of the
time using these components.

audiocodes mp202b using g723 in either low or high bit rate
asterisk 1.6.1.10 with tc400b transcoder
sip provider using ulaw

I have also verified it byt using another asterisk box and using ulaw
between them.  the voice is digitized.  acts like the problem that was just
solved for me on g726.

what would you like for info? 

Issue History 
Date Modified    Username       Field                    Change               
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2009-11-20 09:25 globalnetinc   Note Added: 0114042                          
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