[Asterisk-Users] SIP and echo cancel

[Mike Bernson]

In this connection there are min of 2 hybrid circuits on my end

Vonage ATA box ---- SIPURA ATA ---- Asterisk ----- SIPURA ATA (phone)
hybird 1   -----  2 wire -- hybird 2 ----------------------hybird 3 (if not
841 phone)

I also think vonage has one more hybird on there end.

Since the connection to vonage is 2 ata back to back this means 2 hybirds.
The ata are
connect thought 2 wire interface.

On 12/18/05, Mohammad Shokuie <shokuie at hotmail.com> wrote:
>
> Dear pals,
>
> As a matter of fact im serious to know where is the source of echo in a
> pure
> VoIP connection, i think the most of echo problems come from hybrid
> circuits
> which are not an issue in pure VoIP sessions.
>
> Regards.
> ---
> M. Shokuie Nia.
>
>
> >From: Luki <lugosoft at gmail.com>
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> >Subject: Re: [Asterisk-Users] SIP and echo cancel
> >Date: Sat, 17 Dec 2005 21:45:57 -0800
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> >
> > > Before I start hacking this into asterisk 1.2.1 I would like to known
> > > if others are running into this kind of problem ?
> >
> >Asterisk doesn't do any echo cancellation in the setup you describe;
> >it just passes the audio data, and transcodes if necessary. The
> >endpoints (the 841 phone and the 2002 and 3000 ATAs) are responsible
> >for cancelling echo.
> >
> >The Sipura ATA's generally do a good job cancelling echo. You may want
> >to play with the gain settings in the admin web config for the Sipura
> >ATA. As far as the 841 is concerned, if the handset volume is too loud
> >I noticed you may be getting acoustic echo. Hasn't been a problem for
> >me for PSTN calls or SIP to SIP calls though.
> >
> >If you really want to patch asterisk to apply echo cancellation on the
> >RTP stream on pure VoIP calls, that would be interesting to see how
> >well it works.
> >
> >--Luki
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