[sip-comm] Jitter/Choppy Voice quality with Asterisk's GSM codec


#1

Hi Wai Phang

Has anyone tested SIP Communicator with Asterisk?

I've just managed to get it up and running :slight_smile:

I am experiencing a very choppy GSM voice quality at the Asterisk's
receiving end. The voice quality on the SIP Communicator's side is excellent
though.

Hooked up to Asterisk, dialed 1000 and got a very choppy sound. It got better
when I paused the audio (arrow to right) and started it again.

When calling with two communicatiors I experienced the same as you but when I
start the brand new Gnomemeeting-opal (CVS) which prefers iLBC, the sound is
excellent (even video works :slight_smile: )

What does happen though is, that I can't get the sound output working on one
of my machines. No idea, why. I'd be interessted to hear, what you've
experienced with sip-communicator and asterisk.

Greetings
Conrad

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