[asterisk-bugs] [Asterisk 0016288]: G723 codec has digitzed voice
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Nov 23 06:20:02 CST 2009
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 06:20 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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(0114129) davidw (reporter) - 2009-11-23 06:20
https://issues.asterisk.org/view.php?id=16288#c114129
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For a start, what do you mean by "digitised", as all speech is digitised on
Asterisk (and most public telephone networks).
I would have thought that the correct place for support for the tc400b
would be Digium's commercial support channels.
You actually meant G.723.1, not G.723. That appears to be a rather low
bit rate vocoder codec so it should sound somewhat machine like, especially
if you are using the 5.3kb/s variant. GSM, also a vocoder system, uses
more than 13kb/s. There is a half-rate GSM, but I believe most people
think the sound quality unacceptable.
Issue History
Date Modified Username Field Change
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2009-11-23 06:20 davidw Note Added: 0114129
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