[asterisk-bugs] [Asterisk 0012164]: Distorted playback of G.722 prompts
noreply at bugs.digium.com
noreply at bugs.digium.com
Sat Mar 22 18:28:32 CDT 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=12164
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Reported By: milazzo
Assigned To: russell
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Project: Asterisk
Issue ID: 12164
Category: Codecs/codec_g722
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 106518
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 03-06-2008 21:04 CST
Last Modified: 03-22-2008 18:28 CDT
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Summary: Distorted playback of G.722 prompts
Description:
With the changes in r106501, G.722 audio FROM a Polycom 650 now works
properly; it can be recorded, transcoded, etc. without impairment.
Unfortunately, playback of G.722 audio TO a Polycom 650 is now distorted;
it plays choppily at half speed. G.722 voicemail prompts, Playback() of a
.g722 file, and G.722 MOH exhibit this distortion, as does a call being
transcoded from GSM (the Polycom -> GSM direction sounds fine).
Playback of other file formats (uLaw, slin) sounds fine, as does a call
being transcoded from uLaw.
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milazzo - 03-22-08 18:28
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Russell:
Thanks for working on this problem! It's *almost* fixed now. Here's what I
observe using SVN-trunk-r110578M:
Recording / playback from Polycom 650 in G.722 mode to g722, ulaw, wav,
gsm: all good.
Voicemail prompts, voicemail recording and playback: all good.
Music-on-hold in G.722 format: good.
Placing a call from Polycom 650 via G.722 on SIP -> uLaw on Zap: good.
Placing a call from Polycom 650 via G.722 on SIP -> GSM on IAX2: BAD.
In this last test, the outgoing audio (G.722 -> GSM) sounds fine at the
remote site. The incoming audio (GSM -> G.722) is terribly distorted,
though it does appear to be playing at the correct speed.
Curiously, if I transfer the remote party to a test extension that records
in GSM format, then play back the resulting recording using the Polycom 650
in G.722 format, it sounds fine.
If I place the same call from the same Polycom phone, but force it to use
uLaw with:
disallow=all
allow=ulaw
in my sip.conf, it sounds fine in both directions.
If there are any additional tests / logs / packet traces / etc. that would
help, please let me know.
Thanks again,
Paul
Issue History
Date Modified Username Field Change
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03-22-08 18:28 milazzo Note Added: 0084422
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