[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:              2009-11-23 11:34 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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 (0114152) globalnetinc (reporter) - 2009-11-23 11:34
 https://issues.asterisk.org/view.php?id=16288#c114152 
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Assuming I am not a complete idiot, I understand all voice on a VoIP system
has been digitized.  This codec is being done incorrectly.  The audio
quality is not understandable.  the problem is much like the g726 problem I
submitted.  You may want yo take a look at 15504 which was eventually
closed by 16230. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-11-23 11:34 globalnetinc   Note Added: 0114152                          
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