[asterisk-users] Echo problem in VoIP-calls

[Philipp von Klitzing]

Hi!

> The network setup is :
> analogue+GXW / softphone --> Linksys WAG160N --> Asterisk server --> ITSP
> --> other networks

Do it step-by-step: Take the Asterisk server out of the equation, i.e. 
call the destination directly with your softphone or the Grandstream ATA 
and see if that removes the echo.

That fact that both sides are hearing echo is a bit unusual - especially 
when calling a mobile destination things should be different. Check twice 
that the analog devices in the setup are ok, and replace them for a test 
if you can.

You could also test with a destination that is run by a different 
operator (or is located in a different country).

Another test: Use the Echo() application on Asterisk and call it from 
both sides.

Also: You could capture the traffic and look at it with Wireshark, the 
delay/latency in particular.

Philipp

P.S.: I do think a jitter buffer matters for echo, simply because it 
introduces an additional delay. However the Asterisk server should not 
use its jitter buffer because jbforce is set to no and the Asterisk 
server is not the final endpoint (it only sits in between).