1) It varies over time. It is worst at the start of a call and after periods of silence. From this I conclude that the echo canceller is adaptive, adjusting delay and amplitude based on real time signal analysis.
2) The echo is only heard on the SIP phones, not on the far end of the POTS connection. In spite of seeming somewhat mysterious at first, I concluded that this is actually the expected behavior. Note: Voice is digitized at SIP phone, transmitted (with delay) to the FXO card and converted to analog. The analog signal is placed on the 2-wire POTS line, where it loops back to the analog receiver in the FXO card and is digitized and sent back to the SIP phone as an echo. The POTS end hears no echo because there is no pathway for the echoed packets to get back to the POTS.
From this, it seems to me that the ideal place to deal with the echo would be at the point of conversion (from A to D and back), since there would be virtually no delay at this point and it would be simpler to determine the correct amplitude correction. If this is the case, then it explains why adjusting receiver and transmitter gains would aid in cancelling echo, with the caveat that it should only work in those cases where the Rx ant Tx gains that are being adjusted are those which are right at the A/D and D/A converters. In other words, adjustment of these values in Zaptel.conf should have no effect if I am using an outboard channel bank.
What I would like is for the real experts to jump into this discussion, lay out the real theory and technical details of the echo cancellation used in asterisk so that we can all make more intelligent attacks on the echo problem. How about it?
Stephen R. Besch
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