[asterisk-users] Clipping issue with SIP over satellite

[Richard Kenner]

I'm having a wierd clipping issue with one employee who's using a phone
over a satellite Internet.  He was sold that system specifically for use
with VoIP.  Ping times show average round-trip time as around 700 ms with a
range of 560 to 841, so considerable jitter.

Things work fine when he's talking to another Asterisk phone or to a SIP
trunk provider, but when connecting to a T1, there's clipping where about
1/3 of his voice (in intervals of maybe 200ms) are removed.  This sounds
like an echo canceller conflict, but I've set echocancel=no in
chan_dahdi.conf (I have hardware echo cancelling) and it didn't do
anything.  I'm forcing his codec to G729 for bandwidth reasons.  The
phone is an Aastra 6757iCT.

Does anybody have any suggestions here?