[Asterisk-Users] Compare to Skype

[Jon-o Addleman]

On Mon, May 01, 2006 at 12:16:18PM +0400, Jean-Michel Hiver spake thusly:
> 
> >This is only an issue if your SIP phone has a poor/nonexistent jitter 
> >buffer.
> 
> I agree with that. Asterisk should just forward any RTP immediately and 
> let endpoints handle the jitter buffer - unless asterisk is the endpoint 
> itself (e.g. with phones plugged in its fxs ports).

That makes sense if asterisk is just serving as a gateway, passing on
audio to other machines, but if it's processing the audio on its own,
that's not so good - it'd mess up recordings, for one.

-- 
Jon-o Addleman - http://redowl.dyndns.org