[Asterisk-Users] Best echo-free and trouble-free system?

[Nicolas Bougues]

On Wed, Apr 28, 2004 at 02:40:16PM -0500, Eric Wieling wrote:
> As I understand it, going analog does not *actually* eliminate echo. 
> However, it goes create a situation where echo is so fast that you don't 
> hear it.  VoIP adds latency, which is why you could har echo on the same 
>  telco line using SIP, but not using analog.
> 

100% digital (of course, VoIP is considered digital as well)
end-to-end is by design echo-free. 

What causes echo (except poorly designed speakerphones) is the 2 wire
local loop, either used as FXO or FXS.

When you have at least one analog 2 wire line in your call path, you
will have echo at least on one side. Of course, this applies to
PSTN. But the difference is that PSTN (end to end)  has virtually zero
delay. Thus, you hear echo while you talk, just like your ears always
hear what gets out of your mouth. You're used to it, and your brain
performs echo cancellation very well.

The real problem arises when :
- you have some echo induced somewhere (your call goes through a 2
  wire line)
- you have some delay induced somewhere (you use VoIP for instance)

In this scenario, echo becomes noticeable, and disturbing. You need to
echo cancel at the closest from the source (in the ATA, for instance).

So you have two choices :
- either stay in the full realtime world : no noticeable echo
  (although existent) in most cases, but no VoIP.
- or you need VoIP : be the most echo-free (use digital lines/PRI,
  prefer phones to ATA...), and have good echo cancellers (in
  Asterisk, in your ATA...).

According to my experience with the later scenario, an Asterisk setup
is "mostly OK". It means that users will enjoy toll quality calls most
of the time (provided you have the bandwidth for G711), with, every
once in a while, a call with a usually slight echo. 

-- 
Nicolas Bougues
Axialys Interactive