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.
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