[Asterisk-Users] audio pause/delay problems

[Jan Rychter]

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.
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