[asterisk-users] RE: ECHO Cancellation in SIP Calls

[Stefan Agethen]

Message: 7
> Date: Thu, 26 Oct 2006 22:56:58 -0400
> From: "Michael Araba" <maraba at quikcat.com>
> Subject: [asterisk-users] RE: ECHO Cancellation in SIP Calls
> To: <asterisk-users at lists.digium.com>
> Message-ID:
> 	<8C18AA7CE9A6804090E8235C22C3BBAF495356 at BE02.exg3.exghost.com>
> Content-Type: text/plain;	charset="us-ascii"
>
> I am surprised that you are getting echo on SIP calls. You can get echo
> in two scenarios on SIP calls.
>
> 1. If SIP calls are crossing to PSTN (inbound/outbound). Here you need
> to enable echo canceller and AGGRESSIVE if needed in zconfig.h. 
>
> 2. Second source of echo on SIP calls could be ACOUSTIC. The phone sets
> you are using may not handle this well. 
>
> In my experience sound quality deteriorates if there is network trouble
> or congestion on SIP calls
>
> I hope this helps.
>
> Michael
>
>   
Hi Michael,

For sure, i can get echo in the 2 to 4 wire scenario, this is right, but 
this cant be happen in MY way, only the provider can
produce this scenario, my asterisk use zap and isdn, but the echo occure 
in pure sip calls, in my zap and isdn channels i use the patch from
mgernoth, named "mg2", great stuff.

The second is one echo i already know, one other caller parties use very 
cheap phones, so the sound of the telephone speaker is not shielded enough
to put no sound in the telephone mic - this is not the case with my 
phones, i use SNOM, they are build to used with VoIP and the best one i 
know, in my case.

I checked the latency and loss between me and my provider this morning 
again, and i figured out a routing point which lost 3% of my packets, 
first time for me to see this
after one year of working good, i wrote a mail to my provider, and asked 
him to check this on his own, but i cant imagine that this produce all 
the echo...must wait, i guess.

I tested my Network, good results, tested other VoIP Provider's Server, 
Result is good to ok.

Recap : To minimize echo i can check : Phone (ok), Channels in Asterisk 
(crossing) (ok), My Network Connections between Phone and Asterisk (ok),
Network between Asterisk and Router (ok), Connection, Loss and Latency 
between Asterisk/Router and my VoiPProvider (waiting..)

Any other ways to produce echo in pure *SIP* !

Thanks for your great help !

Stefan