[Asterisk-Users] audio pause/delay problems

[Jan Rychter]

[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.