[Asterisk-Users] SIP and echo cancel

[Andrew Kohlsmith]

On Sunday 18 December 2005 14:32, Mohammad Shokuie wrote:
> As a matter of fact im serious to know where is the source of echo in a
> pure VoIP connection, i think the most of echo problems come from hybrid
> circuits which are not an issue in pure VoIP sessions.

Easy.  Get better endpoints.  In a pure-voip loop you have echo due to 
acoustic coupling from the earpiece to the mic, or the speaker to the mic in 
a speakerphone.  Easy way to tell: in a call with bad echo, have the other 
side mute.  If your echo goes away, you've got your culprit.

Also note that if your transmit level is too high or they have the volume up 
too loud on their end it could push the audio coupling over what the design 
specifications were.

-A.