[Asterisk-Users] Jitter in SIP calls?

[Michael Welter]

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
mike at TelecomMatters.net
www.TelecomMatters.net