[Asterisk-Users] audio pause/delay problems

[Steven Critchfield]

I use IAX2 over a 2000mile loop from my home to the office using GSM and
have no problems as long as the lag is low. Most of the time you can't
tell the difference between VoIP and PSTN on the phones at home.

On Mon, 2003-07-14 at 12:30, Jan Rychter wrote:
> I'm curious. Isn't anyone else noticing these problems? Or are people
> simply not using asterisk for VoIP connectivity over wide-area networks
> this way?
> 
> Or does it go away with g729 or other proprietary codecs?
> 
> --J.
> 
> > >>>>> "Jan" == Jan Rychter <jan at rychter.com> writes:
> > >>>>> "John" == John Todd <jtodd at loligo.com> writes:
> >  John> For what it's worth, I have noticed the same problem, but I think
> >  John> the problem is in IAX2, since my long-haul portions of the
> >  John> diagram were over IAX2, while my SIP clients are almost always
> >  John> sitting on the same LAN as the Asterisk server.
> > 
> >  Jan> I have noticed these problems both in this kind of setup and in a
> >  Jan> SIP call to a remote Asterisk server.
> > 
> >  John> What codec were you testing with over IAX2?
> > 
> >  Jan> GSM.
> > 
> > Having investigated this a bit more, it turns out that using alaw
> > instead of gsm on the IAX2 link makes the problem go away. It seems the
> > jitter settings start working then.
> > 
> > Any hints? I'd prefer not to be stuck with 80kbps per call...
> > 
> > --J.
> > 
> >  > [I have sent a message about SIP problems via gmane, but it seems the
> >  > list is gatewayed one-way only...]
> >  >
> >  > The message was:
> >  >
> >  > I've been trying to use Asterisk as a SIP->PSTN gateway. It runs fine
> >  > when the SIP client is on the local network and there is not packet
> >  > loss. But now I've tried running a remote client (halfway around the
> >  > globe) -- this works great until some packets get lost. After that it
> >  > seems that either my client (linphone) or Asterisk doesn't want to
> >  > resynchronize -- what gets played back is all voice packets as they
> >  > have been received. This creates an increasing lag in the
> >  > conversation and the only way I've found to fix it is to disconnect
> >  > and reconnect again.
> >  >
> >  > Is anyone else seeing this? Is it linphone's fault, or is it expected
> >  > behavior?
> >  >
> >  > Now, I have tried running another * on "my" side of the link. The
> >  > setup then becomes:
> >  >
> >  > linphone -> * -> internet (IAX2) -> * -> PSTN (or echo).
> >  >
> >  > I'm testing with the echo application (GSM used everywhere) and I'm
> >  > getting the same thing: everything seems to work, but sooner or later
> >  > there is an audio pause and the delay grows. It never gets back to
> >  > normal. I've had it grow to as much as 10s.
> >  >
> >  > What makes it even more surprising is the network performance. I've
> >  > had ping running in the background, same TOS settings, 10 packets per
> >  > second. It shows that my RTT is (min/avg/max/mdev) 220/229/287/8.85
> >  > with 0% loss! That's a pretty good network. So where do the pauses
> >  > and delays come from?
> >  >
> >  > --J.
-- 
Steven Critchfield  <critch at basesys.com>