[asterisk-users] ECHO Cancellation in SIP Calls

[Stefan Agethen]

>> Hi,
>>
>> i am from Germany, so excuse my School English.
>>
>> I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update 
>> of Asterisk 2 wooks ago, Echos accure in my SIP Calls.
>>
>> I use SNOM 360, sometimes there is no echo (for example if i call myself 
>> via SIP->Asterisk->SIPProvider->TELEKOM->ISDN)
>> but if i call other people there occures Echo many times. The Routing is 
>> always the same :
>> SIP (SNOM) -> Asterisk -> VoIPProvider -> ISDN/POTS
>>
>> Can i control the cancellation with the zapata.conf ?
>>     
>
> The snom phones are pretty decent devices and shouldn't introduce echo.
> Your latency might be too high between asterisk + voipprovider
> introducing a delay that is noticed as echo.
> You are hearing the echo that is introduced on the callers side.
> As I understand it, when you are calling someone there is no zap
> involved and thus you can't cancel it with zapata.conf.
> if you look at voip-info.org [1] you'll find a good explanation why you
> can't use an echo canceller to cancel that sort of echo.
> So, check the path between you and the voipprovider, e.g. connection
> saturation, ping times etc
> (This is assuming you have a proper lan connection between
> asterisk/snom)
>
> Conrad
>
> [1] http://www.voip-info.org/wiki/index.php?page=Asterisk+Echo
> +Cancellation
>   
Hi Conrad,

thanks for your help, this is the way i understand it all the time, a 
year ago i have optimized my Business Lan for VoIP and there is no loss 
or lag anymore, the Provider seems to be okay, i have pinged him for one 
day with MTR, no great loss or high ping.

Some days ago i have read a Thread about EC-Cancellation in SIP Calls 
with the zapata.conf and never understood how this could work, thats the 
beginning of my question ;)
In my case, there is no Zap, you are right. So i must start at the 
beginning and search for a lag...

The EC started two or weeks ago after one year of great communication, 
in this time i updated Asterisk from 1.2.10 to 1.2.12.1 and zaptel from 
1.2.7 to 1.2.9 ...

I will watch the Quality and the latency next time....Thx for your time !

Stefan