[asterisk-dev] Adding Octastic Soft-Echo to external SIP adapters

[Andrew Kohlsmith]

On Monday 07 May 2007 10:54 am, Vazir wrote:
> So the echo is generated by the VOIP side. I can hear it
> very well is I switch off echo cancellation on the CISCO

Echo is not generated by VOIP.  VOIP is known as a 4-wire system.  Transmit 
and receive paths are along totally separate (albeit logical) paths.

Echo is generated through several sources, most notably either poor hybrids 
(devices which change the 4-wire ear and mic connections in the handset to 
2-wire used by the PSTN) causing received signal energy to be reflected, and 
by acoustic problems (speakerphone in a "hard" environment, physical coupling 
of ear and mic, etc.

Digital phone connections do not cause echo.  VOIP does not cause echo.

Echo is most notably seen in VOIP systems because of the extra latency you get 
through packetization, transport, etc.  circuit-switched calls are 
very "fast" in this regard, and you almost never hear it as anything other 
than what's known as a sidetone (hearing your own voice in your earpiece 
almost instantaneously).

> but surprisingly CISCO takes 64 ms echo cancel parameter and
> cancels echo on a such a latent link...

Probably because the PSTN is being terminated by the Cisco, and it's echo 
cancelling right at the PSTN interface.  This is exactly where echo 
cancellers should be placed.

-A.