>>>>> "John" == John Todd <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> writes: "John" == John Todd > <[EMAIL PROTECTED]> 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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