View Full Version : VoIP echo problem

July 4th 03, 05:46 PM

Can anyone tell me what happen of VoIP conversation over broadband, the
sound have great echo and feed back from PSTN to FXO to internet to FXO to

July 4th 03, 08:38 PM
Hi Andy&Alice

> Can anyone tell me what happen of VoIP conversation over broadband, the
> sound have great echo and feed back from PSTN to FXO to internet to FXO to

Basically You need to look into RX /TX gains all places where you hvae PSTN
<-> FXO (ip) Conversions. You need to make sure you have a Echo Canceller in
the VoiceGateway if not you can not do anything about the Echo.

I posted this a while back :

Echo cannot origin from an VoIP network. But delay times due to codecs and
buffering quickly makes even the slightest echo recieved very annoying. Echo
is generated by digital (4 wire) <-> anolog (2 Wire) conversions either in
the PSTN or "at the other end" (overlaps of ear & mouth)".

If you hear echo then the source of the echo is either the far end or
somewhere in the PSTN network.

If the far end hears echo then you generate the Echo.

There are a couple of mechanism to prevent echo that is ERL (Echo Return
Loss) og ERLE (Echo Return Loss Enhance). ERLE is often named Echo

ERL use adjustment of powerlevels for recieve and transmit audio stream, and
should be adjusted so that the echo is as low volume as possible without
loosing the ability to communicate with the far end. You have 2
possibilities :
1) Lower powerlevels of what you send out - Risk : The far end cannot
hear you
2) Lower powerlevels of what you recieve - Risk : You cannot hear the
far end

If the Echo is Doubletalk - then ERLE cannot distinguish Real talk from
Echo, and the ERLE stops working.
If this happens you should adjust ERL values.

So ERLE stops working with doubletalk. But there is an other more annoying
way to stop the ERLE from working - The is delay times outside all gateways
from the VoIP network. Being the gateway toward PSTN og The Analog Pots.
Every POTS should have an ERLE on at least 8 ms, this would normally be
sufficient to removed ECHO generated bye overlap between Ear & Mouth pieces
or in the short cable connecting the phone to the POTS. Of course IP phones
should also have at least 8 ms ERLE.

The gateway towards the PSTN network should have sufficient ERLE to
compensate for the delay in the PSTN (which is about ~1 ms pr. 1000 km
fiber, 2-4 ms pr. interconnection - ADMs and so on) ... The Standard for
ERLE i G.164 which specify at least 128 ms - But 128 would make the price of
the gateway astronimical for us non-Telco operators due to the complexity in
ERLE. Therefore typical values 16-64 ms typically 16-32 ms ERLE. ERLE
buffers constantly 16 ms of outgoing talk, converts it to a reverse pattern
(a ^ function - e.g. a HAT - function) and compare it to the incoming voice
stream. If it finds a matching pattern on the incoming voice stream it will
be applied if not it will be discarded. If delay times is more that the
buffer in ERLE then ERLE will never work. And echo cannot be eliminated. You
can only minimize the annoyence using ERL operations.

There are allways ERLE deployed in the Public PSTN network where it
interconnects with international destinations, and towards Mobile networks
(PLMNs). But not allways between PSTN operators. Thats because all other
telephony network and connection can tolerate fairly large amount of delay
in the public network - You can have echo but it is not hearable by the
human ear.

With VoIP networks which have to convert voice stream to data streams and
back again so that the round tripdelay times can often be more that 160 ms -
Which is well in to the hearable area (I belive all types of Echo with a gap
or roundtrip delay above 32 ms is hearable first talking in a large room but
the larger roundtrip delay is the more distinct the echo becomes.
This also means that lag times in the IP network is not a probable
contributor to the Echo - The operation around the DSPs width roundtrip
delays over 160 ms is the probable contributor. Reducing buffering in the
DSP function is a possibility, but at the cost of bandwidth and does
normally not help much.

The only way to eliminate echo is adjusting ERL and have to sufficient ERLE
in the gateways. Where ERL reduces the powerlevel of the Echo, so that ERLE
can kickin and remove the echo completely (Actually reduces the powerlevel
of the Echo to very low values (unhearable) . The combined loss Echo by the
combined ERL and ERLE is ACOM - I cant remember the threshold values of the
power levels for hearable sounds but ACOM should be below this value if ERLE
works. You can only completely remove echo by removing the source of the

I Hope you can use this information - Even if it is written in a bit poor

Cheers, Real

July 5th 03, 06:47 AM
What are you using? Please be very specific.


"Andy&Alice" > wrote in message
> Hi,
> Can anyone tell me what happen of VoIP conversation over broadband, the
> sound have great echo and feed back from PSTN to FXO to internet to FXO to