[asterisk-ss7] asterisk oh323 - chan-ss7 echo problem

[Jacob Tinning]

On Tue, 18 Apr 2006, leonimar cape wrote:

> Can someone give a suggestion on what I should do to
> omit the echo. Here is may scenario
>
>      ---  h323   ---   chan-ss7    ---
>  A---|  |--------|  |--------------|  |----B
>      ---         ---               ---
>     Nextone    Asterisk         DMS

The next version of chan_ss7 will include enabling/disabling
of the zaptel echo-canceller, which probably will solve your problem.

> The calling party can hear every words he/she say
> after 1  to 2 seconds. But the called party can hear A
> with no echo and the quality is clear. I have already
> tried it in both digium (TE410P) and  sangoma card
> (AT104) and the results where the same. Is there any
> way that I can cancel the echo? Any particular
> settings that I have to change in the settings of the
> asterisk?

The current version of chan_ss7 does not do anything to avoid
echo. It just passes the audio through the channels.

The problem is when A calls an old analog phone B. Old analog
phones typically bleed some of the audio back to the sender.

Ordinary synchronous telephone-networks doesn't delay the audio
very much ( < 10ms) so the caller will not notice any echo.

Unfortunately, ip-networks induce more delay ( > 70 ms ) which will
clearly sound as echo.

To avoid the echo, the only (as far as I know) solution is to insert
some kind of echo-canceller "in" or "to the right" of Asterisk in your
drawing.

As noted above, the next version of chan_ss7 will start the
zaptel echo-cancellation when a new call is made (this is configurable, if
you don't want echo-cancellation), and stop it again when the call is finished.

Mvh. Jacob

-- 
Jacob Tinning
System Developer                                           SIFIRA A/S