[Asterisk-Users] SIP and echo cancel

[Mohammad Shokuie]

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
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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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