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 <[EMAIL PROTECTED]>
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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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