What codec were you testing with over IAX2?
JT
[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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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