[Asterisk-Users] E1 Echo (was: Small explanation of txgain rxgain statement please)

[Rich Adamson]

> > > I was just looking at:
> > >
> > 
http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.

> > html
> > > regarding echo canceller tuning, and I noticed the statement
> > >
> > > "Most people find that they need an rxgain level around 8.0 to have
> > > good echo cancellation. The txgain setting varies from installation to
> > > installation."
> > >
> > > Which feels a bit wrong :) Could someone explain why increasing the
> > > gain on the inbound zap leg (rxgain) would improve echo cancellation?
> > > Of have I misunderstood the roles and meanings of rxgain and txgain?
> >
> [snip]
> 
> Many thanks for clearing that up for me :) the largest part of my
> misunderstanding was caused by not noticing that that article was
> referring to the tuning of an "FXO" line. I am in fact trying to find
> information on the tuning of an E1 to reduce echo. (Doh!)
> 
> In theory of course an E1 should work with rxgain=0.0, txgain=0.0
> (assuming there is no digital messing going on in the network) and the
> echo canceller should have a relatively easy job of cancelling echo
> given that the large majority of the UK phone network is digital, and
> only the last leg at the far end is usually analogue.

That last leg "is" usually part of the problem since there is going to
be a hybrid conversion.

> I am running Asterisk 1.0.9, and have backported the KB1 canceller
> into Zaptel 1.0.9.2, which does not seem to have caused any problems.
> Nor has it really caused any improvement though :)

The KB1 canceller improves echo, but it appears as though it achieved better
results by forcing half-duplex communications. From a pure non-technical
user perspective, the quality of a telephone conversation has been lowered
simply because humans are use to communicating in full duplex mode.
 
> I am beginning to wonder whether what echo IS heard is being caused by
> packetisation delays "in the network" - The default tap length is 128,
> or I believe 16ms. If something in the PSTN causes a delay more than
> that length (no idea what might cause that) then echo would still be
> heard.

Certainly not hard to change the tap length and eval it.
 
> Does anyone have any experience in this area? Any ideas? How "heavy
> handed" would it be to increase the tap length to 256? I have not seen
> anyone suggest that this might be a good idea.