asterisk users - Mar 2005 - echo / delay problem

I'm having with an echo or delay

I connect to the PSTN with a x100p  and then connect a std. phone
to a FXS module on a TDM10B.

The std phone is only 2-wire so I know this is not helping.
(yes I have read the 2-wire 4-wire issue)

I have tried many echocancel values. The best thing to help was
rxgain and txgain. below is my current zapata.conf file

All help would be grateful. I have tried and tried for 2 weeks
it is rather annoying and irating to hear this delay/echo

I would call it a delay since you can hear the end of the sentence repeat 
over
and over. Also every now and again  it sounds like a underwater submarine
with ping and all.


Thanks in advance
Barry



[channels]
language              = en
context               = inbound
signalling            = fxs_ks
usecallerid           = yes
hidecallerid          = no
callwaiting           = yes
usecallingpres        = yes
callwaitingcallerid   = yes
threewaycalling       = yes
echocancel            = 16
echocancelwhenbridged = yes
echotraining          = no  ;; yes
rxgain                = -2.0
txgain                = -2.0
musiconhold           = default
channel => 1


context               = intern
signalling            = fxo_ks
callwaiting           = yes
usecallerid           = yes
echotraining          = no ;; yes
echocancel            = 16
channel => 2

> I'm having with an echo or delay
> 
> I connect to the PSTN with a x100p  and then connect a std. phone
> to a FXS module on a TDM10B.
> 
> The std phone is only 2-wire so I know this is not helping.
> (yes I have read the 2-wire 4-wire issue)
> 
> I have tried many echocancel values. The best thing to help was
> rxgain and txgain. below is my current zapata.conf file
> 
> All help would be grateful. I have tried and tried for 2 weeks
> it is rather annoying and irating to hear this delay/echo
> 
> I would call it a delay since you can hear the end of the sentence repeat 
> over
> and over. Also every now and again  it sounds like a underwater submarine
> with ping and all.
> 
> 
> Thanks in advance
> Barry
> 
> 
> 
> [channels]
> echocancel            = 16
> echocancelwhenbridged = yes
> echotraining          = no  ;; yes
> rxgain                = -2.0
> txgain                = -2.0
If you are in the US and using a digium x100p, then try the following:
 echotraining=800
 echocancel=yes
(The values you are showing above aren't even valid parameters.)

If you are outside the US, there isn't much you can do since the x100p
card was specifically designed to operate with US 600 ohm impedance
pstn lines.

If you have a x100p clone, it is likely the problem. Replace it with
something capable of matching the pstn impedance for whatever country
you are located in.

> -----Original Message-----
> From: Barry FAWTHROP [mailto:brif8@hotmail.com]
> Sent: Tuesday, March 22, 2005 7:12 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] echo / delay problem
> 
> 
> I'm in the US, using cards bought direct from Digium.
> I have lowered the rxgain and txgain to -8 and that seems to 
> be helping futher.
> I wish I could understand why?
Consider the following bad ASCII diagram of the amplitude of the signal as it
travels from your handset to the remote end, becomes an echo and comes back to
your handset:

1.00| ----
    |     |
0.75|	    | (effect of txgain -8.0)
    |     |
0.50|	     ------^-----		    
    |                  |
0.25|                  | (effect of rxgain -8.0)
    |                  |
0.00|				----
----|--------------------------------------------------------->
	|    |    |    |    |
	|    |    |    |    \Echo level arriving at local handset
	|    |    |    \Echo level arriving from remote end
	|    |    | Signal level arriving at remote end
	|     \-Signal level leaving to the PSTN 
	\-Signal level leaving local handset


Now consider the amplitude of the legitimate speech originating at the remote
end:

1.00| -----------------
    |                  |
0.75|	                 | (effect of rxgain -8.0)
    |                  |
0.50|	                  ----	    
    |                  
0.25|                  
    |                  
0.00|				
----|---------------------------------------------------------->
	|   			 |
	|                  \Echo level arriving at handset
	\-Signal level leaving remote handset

It follows then that a signal that is attenuated going out and then further
attenuated coming back in to your system (ie. the echo) will be relatively
quieter than a signal that is just attenuated once (ie. the remote end
conversation). Remember that a gain expressed in Db is a logarithmic measure so
two passes are more than double the attenuation.

This is known as a 'loss plan' in telco spheres and was traditionally
used in analog systems to make echo (and sideton) more managable. Theres some
interesting blurb on loss planning at: http://telecom.tbi.net/lossplan.htm  or
consult Google.

It's also important to understand that the training algorithms in many echo
cancellers rely on being able to differentiate between the echo and legitimate
far end signal and one of the mechanisms used is to measure the average relative
signal levels. Thus an outgoing signal that is too 'hot' (ie. has
excessive amplitude) may not be properly echo-supressed as the signal cannot be
differentiated from the far end signal, thus getting your gains correct is an
important part of PSTN interfacing.

Hope that helps.

Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District