I'm experiencing a very strange problem with SIP calls with a CLEC
(CBeyond). The downstream audio with the telephone on mute is
excellent. However, when there is upstream audio (even breathing) from
the mic, the downstream audio is clipped and sometimes dropped.
The strange thing is, if I Monitor the call, the downstream audio in the
wav file is perfect, even though there was clipping and drop-outs in
real time.
Is this a case of jitter? What are the symptoms of jitter? Does jitter
resolve itself when the call is recorded? Does chan_sip have a jitter
buffer yet?
When I move the calls to another ITSP, I don't have clipping and
drop-outs, so I'm assuming the problem is not with the Asterisk system
or the telephones.
The Asterisk version is 1.2.5. The phones are Polycom, Cisco, and
Grandstream.
I've checked my NIC connections and everything is full duplex.
Thanks for your help.
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
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