I think that you need to listen to people's advice when you ask for help.
I have EXACTLY the same problem as you and I solved it with the methods that I described in my last post.
Think about it. How can you get echo on a pure RTP stream from your phone to an asterisk server?
Do you hear echo whilst recording voice to your asterisk server (e.g. when leaving a voicemail) or when calling another SIP phone? I think not.
Noah Silverman wrote:
Hi,
I think that you guys are missing the problem. The echo is only from the sidetone. I don't hear the other party with an echo and they don't hear me with an echo. That leads me to believe that it hs nothing to do with the zapata stuff. It is somewhere between my SIP phone as Asterisk.
-N
Rod Bacon wrote:
In addition to making sure that echo cancellation is enabled on the interface(s) in question, you will also need to play with the gain settings. Specifically, try turning down the rxgain. I dropped mine to -10.0, and the echo disappeared altogether.
The problem was then that incoming voice was too quiet. After a lot of messing around, I eventually settled on -3.0
This figure gives me good incoming volume and only a faint echo... not enough to bother me or my users.
I also found that the order of settings in the zapata.conf makes a difference. If I had the gain settings too far down in the config file, they had no effect.
Make sure you stop and restart * after changing any of these settings. A simple reload won't suffice (I even unloaded and reloaded the kernel modules, just to be sure).
----- Original Message ----- From: "Jeff Heath" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Sent: Wednesday, April 13, 2005 7:54 AM Subject: Re: [Asterisk-Users] Local Echo
Here's what's happening.
First some background. Anytime there's a 4 wire (T-1) to 2 wire (local subscriber loop) conversion (this is called a hybrid) there's a good chance that some electrical energy will be reflected. This is because there is usually an impedance mismatch between the 4 wire and 2 wire circuits.
This happens all the time in the local telco. You come in to switch A and are destined for switch Z. The telco transports the traffic between A and Z over T-1 (which is muxed up to T-3 or SONET). When the T-1 gets to switch Z it eventually gets attached to a 2 wire local loop (POTS) to get to the far end. Energy from A is reflected back towards A by the hybrid at the Z side.
But reflected energy is only one of two necessary conditions for echo. The other condition is sufficient delay for a human being to perceive it as echo. In order for us to perceive it as echo, the reflected energy must be delayed by about 25 msec. Anything less than that and we perceive it as sidetone (sidetone is actually a good thing).
The local telephone company doesn't have echo cancelers in their network because they don't need them. Why? because in the local POTS network you'll never have a call that is delayed by more than 25 msec. Long distance carriers (IXCs) install echo cancelers because their customers will experience delays longer than 25 msec, but not local telcos.
Now introduce VoIP. VoIP turns every call (even the simple setup you outlined) into a long distance call. If you have your jitter buffer set to 3 you've introduced 60 msec of delay. I forget the rule of thumb for distance vs electrical delay, but I think you can go from NY to SanDiego in about 85 msec.
That explains why the echo is there. What I can't help you with (I've got lots of telecom experience, but little Asterisk experience) is changing the settings in Asterisk to cancel it. The good news, though, is that this is a straight-forward echo cancellation problem, and once you find someone who knows what the right settings are, you should be able to get rid of it.
-- Jeff Heath
On Tue, 2005-04-12 at 17:28, Noah Silverman wrote:
Jeff,
Thanks for the help. Your explanation of an "echo" makes perfect sense.
Here are some notes on our system that might help:
1) The echo occurs on EVERY call either inbound or outbound, local or ld. 2) We don't use any VOIP services, just PTSN lines from the phone company 3) Our system is like this: SIP phone <-> Asterisk box <-> TDM400 card with FXO <-> Telco Pots line 4) I hear my own voice echo. The other party sounds fine to me, and I sound fine to them. 5) The phones are on a very small LAN in our office with almost no traffic. 6) Our phones are Polycom IP500 7) I have the codec set to ulaw
Thanks!!!
-N
Jeff Heath wrote:
On Tue, 2005-04-12 at 15:28, Noah Silverman wrote:
Hi,
I tried, and still get an echo. I don't think the problem is with the zap interface. It must be
on the
asterisk or phone side.
-N
Echo requires 2 phenomena: 1) reflected energy 2) enough delay
that it
is discernable. That you are hearing echo means that something at
the
far end is reflecting the electrical or accoustical energy of your voice.
Echo cancellation should be done as close to the source of unwanted reflected energy as possible. The fact that you're hearing echo means that the echo cancelers at the far end either a) don't exist or b) didn't work. It will be very difficult to cancel reflected energy coming back at you from the "other side" of the network.
Tell me more about the phone call where you experienced the echo and I _might_ be able to help. Specifically,
- was the phone at the other end a speaker phone and if so, was it an expensive Polycom phone that's designed to be a speaker phone or a
cheap
Walmart phone that happens to have speaker capability?
- was it a local call or a long distance call
- what codecs are in use?
- what's your best guess at the round trip delay (i.e. what
networks had
to be traversed and what is the jitter buffer set for?)
Rich Adamson wrote:
I have a strange echo problem.
When speaking on the phone with someone, I hear MY OWN voice with a sever echo. The other party sounds perfect, and they can hear me perfectly. It is as if only the sidetone has an echo.
I'm running * on a dedicated box, small LAN, and am using 4 FXO
cards >>>>to
connect the box to PTSN lines. My phones are Polycom IP500 SIP phones.
The only echo cancellation stuff that I can find relates to cancelling echo between my system and the PTSN lines. Since the call is "perfect", I don't see how this would apply.
Any suggestions??
Try these parameters for each zap channel: echotraining=800 echocancel=yes echocancelwhenbridged=yes
Don't forget you have to stop and restart asterisk. a reload will
not >>>work.
_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- ========================================== Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 ==========================================
_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
