Denis Galvão - iSolve wrote:
Yes, it sounds like there's a discontinuity in the timestamps when you set up your call, but it seems Dan can't reproduce this.Two more information:
1. I've played with all suported codecs, same problems for all of them.
2. After aprox. 1 minute of conversation the delay problem doesn't occur, or better, it is very less(some miliseconds) than the begining(10 seconds) of a call.
Any ideas!?
The fix is probably:
a) The jitterbuffer needs to be reset after the transfer, or
b) The timestamps sent need to be reset after the transfer.
c) Some changes to the jitterbuffer to automatically reset when it sees this kind of discontinuity.
(c can probably be combined with a and/or b).
I forget if you tried setting "notransfer=yes" on asterisk to see what that does?
What would really help, though, is a packet trace of the call. The best way to get this is to use either ethereal or tcpdump. (there is an ethereal for windows).
If you use ethereal for Windows, have it capture all udp, make the call, and have it stay up for about 30 seconds, and save the file. You can then send that file to me, and I'll be able to see what's going on a lot better than guessing here..
-SteveK
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