On Jun 17, 2006, at 9:11 AM, Rich Adamson wrote:
I know that the zaptel modules have echo cancellation, but is this
possible to do this on VoIP <-> VoIP traffic as well? I'm toying with
a SIP gateway which has apparently a terrible call quality and would
like to know if there is any way asterisk can help with this.
I believe the current Trunk code does include jitterbuffering for sip,
etc. But, the echo cancellation exists for zap only.
Some of the sip gateways on the market do an excellent job with EC.
The Mediatrix 1204 as one example is very good, while less expensive
boxes (such as the spa3000, ht488) have significant limitations.
I am using the Wellgate 3701a, which is a 1 port FXO, one port FXS, and
although it was a major pain to setup, it did solve the echo issue I
had with my PSTN line. The recent firmware update for it also seems to
have helped make it better in some ways.
The only remaining issue is that is takes a number of seconds to train
at the beginning of the call. This is ok for me, but seem to throw off
other users, who are taken aback by the echo. I am still thinking of
some way to help this training along? Any ideas?
Marty
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