[asterisk-dev] Queries regarding Echo Cancellation in Zaptel-1.2.6

[Paul Cadach]

Hello,

Chan Kwang Mien wrote:
> IP Phone A <------> Asterisk IP PBX <---> Analog Phone B
>
> In my tests, echos were generated by IP Phone A when I turned on the
> speaker. As was pointed out, zaptel could not cancel these echos because
> the echo was not received at the zaptel interface, rather on the SIP
> interface.

Correct.

> I have a question :
>
> Does it mean that the SIP interface has to cancel the echo from IP Phone
> A ?

No. VoIP part always uses different receive and transmit pathes, so there
(theoretically) is no way to mix receive and transmit signals to produce echo.

> This should be a real problem since it is possible that echo could be
> generated by IP Phones when they are in the handsfree mode (Speaker
> mode) or echo could be generated from an old IP Phone.

A phone (analog or VoIP) implementing speakerphone mode should care about strong
acoustic echo cancellation in this mode. Regular handset operation for best
results also requires some sort of acoustic echo cancellation (to resolve signal
"connections" between speaker and microphone embedded into single handset), but
with correct handset design it's not so important (residual echo through
accurate designed handset less than -20 dB).


WBR,
Paul.