[Asterisk-Users] Echo Cancellation Feature

[Chris Maresca]

The single most usefull tool that anyone outside telco consultants is
likely to have is ztmonitor.

If you do a ztmonitor [channel number] -v you will get a visual of the
sound strengths and it's pretty easy to see when rx or tx are out of
balance...

Now, if only that would help fix the low-level static noise I have on the
x100p, that would be great.

Chris.

On Thu, 22 Apr 2004, Rich Adamson wrote:

> > I do feel the echo cancellation does need some work.
> > 
> > Currently, other than listening to users, there is no way to benchmark or
> > trouble shoot echo problems.
> 
> Sure there are, it's just that 99% of the asterisk implementors don't
> have the test equipment to do it, and a good share probably wouldn't
> know how to do it if they had access to the equipment.
> 
> > We find that 2 to 3 out of every 20 calls will experience echo.  While
> > echo is a problem that naturally occurs from SIP - PSTN and vice versa, I
> > am still baffled by the fact that the cancellation works randomly.
> > 
> > When doing a zap show channel X, it will also report that the cancellation
> > is still on.  We experience the most echo with a T100P --> Adtran TA 750
> > FXO modules.  While I understand these do not have impedence matching, I
> > wonder to myself why echo cancellation works sometimes, and others not.
> > 
> > Looking at Network util, processor util, and memory utilization, they do
> > not provide any clear indication as to why /when it occurs.
> 
> Not likely to have any impact whatsoever.
>  
> > Is there anything more that can be done to debug echo cancellation, and
> > further are other users experiencing this random echo.  I know it was
> > discussed before, but the support folks at digium aren't able to offer
> > anymore help.
> 
> You've probably read enough from previous postings to know there are
> several different locations within an end-to-end voice call where echo can
> creap into a system. In very general terms, any place where an end-to-end 
> channel incures a two-wire to four-wire conversion (whether done in hardware
> or software), echo can creap in due to lots of different reasons. Since
> asterisk provides us with lots of configuration choices, hardly any two
> systems are alike. Therefore, don't know that anyone is going to write
> * code anytime soon to correct something that can't be pointed to, etc.
> 
> Someone mentioned they have echo on sip to sip calls (presumably the call
> was processed by a single * system). If they do, the problem is likely
> in the sip phone as there are no echo cancallation needs in that four-wire
> end-to-end call from an * perspective.
> 
> I've got a fair amount of test equipment (and 20+ years telephony 
> background), and am planning to assemble a document identifying some of 
> the pstn issues, level settings, and other things impacting a reasonable 
> system implementation. Unless someone wants to UPS a transmission test 
> set to me quickly, the document won't be completed for several weeks. 
> (The only test set I have access to will not be released for a couple 
> of weeks due to classes, etc.)
> 
> I'm also expecting these tests to point out a number of other transmission
> issues within asterisk that we'll get documented with real numbers, etc.
> 
> Rich
> 
> 
> 

--
chris maresca
      senior partner - www.olliancegroup.com

linux, up 17 days