[Asterisk-Users] Re: Codecs and echo

[Claudio Caballero]

I have been having this problem since day one of my * installation, and
have concluded that it's just the echo cancellation within asterisk. I
tried all the various options in the .conf files and the compile flags in
the source code, and various codecs, but it still sounds awful (although
the Polycom phones sound better than the Grandstream). The asterisk
wiki/tiki on voip-info rightly recommends getting rid of the echo on the
analog side, but I was unable to do this. So, I'm in the midst of
switching to a T1 interface to avoid analog altogether. I also tried an
AudioCodes MP-108 (MP-1xx) gateway, which does the echo cancellation in
hardware, and that worked just fine. This was in place of the Digium FXO
cards.

Regards,

Claudio

Dee Lowndes said:
> Date: Tue, 2 Nov 2004 09:17:15 -0000
> From: "Dee Lowndes" <dee at asyouneed.com>
> Subject: [Asterisk-Users] Codecs and echo
> To: <asterisk-users at lists.digium.com>
> Message-ID: <006d01c4c0bc$be9d48e0$0201a8c0 at lappy>
> Content-Type: text/plain;	charset="US-ASCII"
>
> Hi all,
>
> 	I am noticing echo/jitter problems when going sip -> asterisk
> iax (ALAW)-> asterisk pstn depending on the codec I use. Both ULAW/ALAW
> works fine on the budgetone and ata286 but g726 only works well on the
> budgetone.
>
> Ilbc just doesn't work well with broken speech and echo issues.
>
> SIP to sip works fine no matter what codec so I am thinking it's either
> IAX or transcoding causing the issue. Any idea's/
>
> Dee
>