I would be interested.
I would also be interested in a process on how to manipulate these
settings.
I am using aastra phones and sangoma a200 with echo cancel.
Thanks
On 21-Apr-09, at 10:40 AM, Henry L.Coleman wrote:
With Zaptel or SIP phone(s)
Make calls and start by lowering the RX gain until the sound level
is lower but still audiable.
Make calls and start by lowering the TX until the far end say it
lower but still acceptable.
It should be noted that ulaw and alaw compress the dynamic range on
TX and expand it on RX
This is known as companding and is very similar to those audiophiles
who remember the Dolby
system used in cassette tapes and 8 tracks (now I 'm showing my
age). Anyway, to cut a long story short,
this proceedure can play an important role in the perception of low
level artifacts such as echo.
As it happends, aLaw is better than uLaw at handling this but what
is even more important is that the
same no-linear gain is applied to both ends (ie. the aLaw 2 aLaw or
uLaw to uLaw).
Obviously, any FXO and ATA's need to match these dynamic properties
with any SIP/IAX phones or FXSs
This is a very big subject so I can't go into too much detail, but I
could give a session on this at a
TAUG meeting if anyone is interested.
-----------------
Henry L. Coleman
[VoIP-PBX.ca]
=================
Dean Yorke<
ok, so how do you tune the rx and tx
On 20-Apr-09, at 10:48 PM, Henry L.Coleman wrote:
In the case you describe both ATA's have a local echo can. which
will try cancelling local talk back (ie.
sending only the audio and not the local "talkback". If we say that
only this echo can. is 95% efficient
then 5% gets reintroduced. That might not sound like a big deal but
the process is "recursive" bounced
from end to end. Maths buffs will see this problem of delay, volume,
dynamic compression (codec).
The gain control is the best place to improve the echo, if either
the RX or TX gain is above 0db then
the overall audio connection has positive gain and will tend to
amplify the channel echo.
Well that's my two cents :)
-----------------
Henry L. Coleman
[VoIP-PBX.ca]
=================
Simon P. Ditner<
It's quite possible I'm confused about which cases are likely.
Perhaps I can explain it better if I start over... We have two
parties, A (local end), and Z (remote end). Both are connected to
the
PSTN via an ATA going over twisted pair.
There are 4 ways I see for echo to be perceived:
1) A hears an echo of Z's voice
2) A hears an echo of A's voice
3) Z hears an echo of A's voice
4) Z hears an echo of Z's voice
So of these, there are two styles of echo:
i) Hearing an echo of your own voice (A)
ii) Hearing an echo of the other party's voice (Z)
----
If I'm hearing (i), the major sources are:
X) Impedance mismatch on my local loop, caused by poor wiring
conditions (untwisted wires, extra km of wire beyond where you are
tapped in to the local loop, water, mice)
I believe this can be tuned out by most analog card, and/or through
echo cancelers monitoring my TX/RX.
Y) Remote end feeding my voice back into their microphone,
exasperated
by the delay in VoIP networks
This can be mitigated with hard/software echo cancelers monitoring
my TX/RX
----
If I'm hearing (ii), the major sources are:???
Removing (ii) is not possible?
-spd
On Mon, Apr 20, 2009 at 9:20 PM, Henry L.Coleman <[email protected]
wrote:
Well now I am confused, a recording of an echo is not the same
thing as the real thing.
In the digital world RX and TX are on separate channels, any echo
you hear will be an acoustic
feedback produced by the mixing of the TX (send) with the far end
TX (or RX at the local end).
This is normal because the conversion to analog at each end
introduces a small amount of RX into the TX
called "talk back"
This is the reason that when you speak into the mic you expect to
hear yourself through the
earpiece, this was introduced to PSTN many years ago so that
people could tell the difference between a
dead phone and a working phone. (pick up any analog phone and
you'll see what I mean).
Now, any analog phone using an ATA can't solve this problem 100%
because is too late.
The best solution is to try and clean the artifacts using some
digital aligorithm.
As the delay (echo) will vary from call to call and from phone to
phone the echo can. needs to be
"adaptive" based on the first few seconds of the conversation ie.
"training".
Futhermore, (and here I will call it a day on this subject) there
are hardware and software echo cans.,
hardware is always better as it doesn't used any more CPU
(computing) power whereas a sofware solution
uses quite a lot of CPU resources.
ALL all THE the BEST best :)
Henry L.Coleman [VoIP-PBX.ca]
-------------------------------------------------
John Lange<
On Mon, 2009-04-20 at 16:56 -0400, Simon P. Ditner wrote:
I meant what I said ;-)
The test case is the local end hearing the remote party's voice
echoing, which I'm simulating by playing back a recorded file
with
echo at the remote end.
Ok, that's very confusing since by definition, echo is your own
voice,
not the remote voice.... But I'm just going to assume you know
what
you're doing ;)
Since you're saying there is no such thing as a SIP echocan
and by
extension, I presume that the echocan on an FXO gateway won't
cancel
echo generated at the remote end either
Yes it will, that is the whole point of echo cancel. Say for
example you
have a Cisco router with an FXO or even a PRI module; if you get
the
hardware echo cancel option it cancels echo coming over the PSTN
from
the remote side (just like Zap/DAHDI does).
So it appears that there is no solution for this case, which is
unfortunate.
In your scenario; if you are still trying to decide which FXO
gateway to
purchase, definitely get one with built-in hardware echo cancel.
They
are of course much more expensive but that is the correct
solution.
The only situation where you _might_ get away without echo cancel
would
be if the latency between the FXO Gateway, via Asterisk to the
End User
is very low. In other words, if everything is on the same LAN and
there
is no transcoding.
--
John Lange
http://www.johnlange.ca
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