[Asterisk-Users] IAX G729 Codec

[Steven Critchfield]

On Fri, 2003-07-11 at 06:40, Simon Woodhead wrote:
> >Our problem was that we all of a sudden would get dropped audio, and I
> >had one user complain of extreme lag occasionally. I didn't have anyone
> >else experience the lag, but the dropped audio would come and go. It
> >sometimes would drop out for a second or so. Sound quality when there
> >was still just perfect.
> 
> >For your link to the Pace Vega Stream, what codec are you using? I would
> >assume it would be more of a problem in codec shifting bits or
> >something, but then again this is a wild guess.
> 
> Thanks for that. We're using G.729 over H.323, incoming and outgoing.
> Outgoing works perfectly but on incoming we get the underwater sound
> periodically. It clicks in randomly but once there the only way to clear it
> is to end the call and try again.
> 
> One thing we have thought of is co-loing an * box directly at the Telco and
> plugging in to their switch directly. We'd then be in control of the VoIP
> part and know that over IAX it would work fine. Can anyone enlighten me as
> to how we'd connect to them physically on-site? Would it be a PRI or would
> there be a different method as the PSTN wouldn't be between us and their
> switch?

You would still use PRI if you need bulk lines. You could use
channelized T1, but you get a lot more options with PRI. Currently our
phone server is in our colo rack and our phone lines are sent down to us
via our data T1 line. 
-- 
Steven Critchfield <critch at basesys.com>