[Asterisk-Users] Local Echo

Rod Bacon rod.bacon at empoweredcomms.com.au
Tue Apr 12 16:25:58 MST 2005


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" <jheath1 at optonline.net>
>>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>><asterisk-users at lists.digium.com>
>>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
>>>>>>>Asterisk-Users at lists.digium.com
>>>>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
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>>>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>_______________________________________________
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>>>>>>
>>>>>
>>>>>_______________________________________________
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>>>>>
>>>>>
>>>>
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>>>
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>>
>>
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-- 
==========================================
Rod Bacon - VOIP Systems Engineer
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600    Fax: +613 99401650
==========================================




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