[Asterisk-Users] audio pause/delay problems

[Jan Rychter]

>>>>> "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.

I have noticed these problems both in this kind of setup and in a SIP
call to a remote Asterisk server.

 John> What codec were you testing with over IAX2?

GSM.

--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.  _______________________________________________ Asterisk-Users
 >> mailing list Asterisk-Users at lists.digium.com
 >> http://lists.digium.com/mailman/listinfo/asterisk-users

 John> _______________________________________________ Asterisk-Users
 John> mailing list Asterisk-Users at lists.digium.com
 John> http://lists.digium.com/mailman/listinfo/asterisk-users