[Asterisk-Users] Local sip client stuttered audio

[Kris Edwards]

I have asterisk running on my personal computer and am using Kphone to
connect to it.  My provider is broadvoice which is Ulaw and I had kphone
connected as GSM.  The lag was terrible coming from
Pots-->--Broadvoice-->Kphone.  About 2.5 seconds!  Going the other
direction seemed fine. I did a:
show translation recalc 200 and see that the translation time should be
about 2 ms.  When I do the recalc, that's checking the translation w/ my
hardware isn't it?  Not sure where that delay is coming from.

I changed kphone to ulaw and the lag is gone, but now I get stuttered
audio (not constantly, but a lot.. almost like a clicking over the
audio.  I notice this same problem if I enable a console sound driver
and dial a number, it has the same stutter before the Console is
answered (when you can hear the call going through, but it hasn't
connected yet)after I get the *beep* console has been answered message,
the click is gone).

  The person on the pots side doesn't seem to get this, just on the sip
side.  Oddly enough, if I put the person on hold, when I pick up again
the stutter is gone for a period of time and then comes back.

I've tried to search for stuttered audio but I just end up getting pages
about stuttered dialtones, so I'm sorry if this is some well documented
problem.  I'd be happy if someone just pointed me to a link ;)