[Asterisk-Dev] ***TESTING NEEDED*** JITTER BUFFER FOR SIP

[Jerris, Michael MI]

> [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Brian West
> 
> Ok we tested the new jitter buffer today from a very distant 
> network with about 20-55% packet loss and lots of jitter you 
> could at least understand the person that the far end... 
> granted I think it was better than SprintPC :P
> 
> /b

Now do you really think SprintPCS is a fair comparrison with a voip call
half way around the world where you are getting less than 1/2 the
packets?  I think not.  All kidding aside, this kind of performance of
the american cellphone network has primed america for the qwirks of voip
of the public internet.  Remember that in other parts of the world, they
have all digital phone networks, and are used to higher voice quality.
A good jitterbuffer that can adapt well to all voip protocols will be
critical to larger scale adoption of asterisk overseas.

Mike