[Asterisk-Users] audio pause/delay problems

[Jan Rychter]

>>>>> "John" == John Todd <jtodd at loligo.com> writes:
 John> This happens to me as I mention below, but only rarely.  What is
 John> your CVS version?

The latest? E.g. I've tested 2 days ago.

--J.

 >> 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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