SV: [Asterisk-Users] re: Sound issues on SIP-SIP calls

[Eric "ManxPower" Wieling]

Asterisk only supports echocancel on PSTN (zap) interfaces.  Echo needs 
to be canceled out at the PSTN<->VoIP interface.  Calls that are all 
VoIP can't have echo.  If they have echo then you can, with confidence, 
blame the SIP phone.

Here's another example:
   Analog Phone -> SIP ATA -> Asterisk -> SIP Phone

The SIP ATA should be the device canceling out the echo.

steve wrote:
> I thought the same thing before I made my reply but zapata.conf seems to 
> be the
> only config file that deals with echo at all. From what I understand of 
> 'echotrain'
> is that at the beginning of the call it sends a short signal out that 
> measure echo
> in an attempt to try and cancel it. I was wondering if you tried using 
> it and if so was it of any help?
> 
> Sincerely,
> Steve
> 
> But is Zapata.conf even in the picture when all calls go through a
> SIP-provider, since the calls are not being routed through a ZAP channel?
> 
> Regards,
> Bjorn
> -----Opprinnelig melding-----
> Fra: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] På vegne av steve
> Sendt: 22. mars 2006 23:11
> Til: asterisk-users at lists.digium.com
> Emne: [Asterisk-Users] re: Sound issues on SIP-SIP calls
> 
> When you hear echo its actually being caused by the other phone. check 
> your zapata.conf file and see if echotraining=yes.
> Echo can be caused by lots of things so this is just a starting point.
> 
> For several months now we’ve been experiencing a really strange problem 
> with
> sound which best can be explained as choppy/stuttery, and with a touch of
> echo on top. Basically, parts of a conversation might be choppy, but often
> combined with some echo as well. The sound problem can only be heard by us,
> not by the other party. But this is the strange part:
> 
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