[asterisk-users] MeetMe echo problems with more than two participants

[Alessandro Russo]

Hi to all,

Unfortunately echo is not due to speakerphone. Each participant calls a
geographical number that is redirected from the PBX to a call manager which
pass the flow to the asterisk machine which creates a meetme voice
conference, so user calls via traditional either fixed or mobile phone.
Therefore they cannot mute their phone while they aren't speak  :(
Moreover the echo problem occurs when we do tests within the same
phone-cloud, in our organization phones are connected through some cisco
call managers, so when a phone calls the internal number ABCD the flow
arrives to the call manger which forward it to the asterisk, this is the
path done: phone <=> call manager <=> asterisk
and also in internal cloud we experienced echo problems with more than 2
participants, not all the conversation is affected by echo, sometimes there
is echo and sometimes not.

I performed the zttest and I obtained the following results:

asterisk:~# zttest
Opened pseudo zap interface, measuring accuracy...
99.966690% 99.971863% 99.936729% 99.967766% 99.936913% 99.968163% 99.967667%

99.936623% 99.969818% 99.937019% 99.967972% 99.937012% 99.968063% 99.967865%
99.936440%
99.967766% 99.935356% 99.967667% 99.937401% 99.968460% 99.967667% 99.936333%

--- Results after 22 passes ---
Best: 99.972 -- Worst: 99.935 -- Average: 99.955330, Difference: 99.992836

Any suggestions?

Alessandro R.


On Fri, Dec 12, 2008 at 7:39 PM, Matthew J. Roth <mroth at imminc.com> wrote:

> Alessandro Russo wrote:
> >
> > we are using Asterisk 1.4.18.1 <http://1.4.18.1/> on debian 4.0 etch,
> > pwlib 1.10 and openh323 1.18.
> >
> > We are using MeetMe for conference calls and with two participants
> > there is no echo problems, but with more than two participants there
> > is a lot of echo that sometimes disappear for a short time and all
> > function well.
> >
> > Someone have some suggestions??
> >
> > Do you ever used app_conference
> > http://sourceforge.net/projects/appconference/  ??
> >
>
> Alessandro,
>
> Are you certain that the echo isn't being introduced by someone on the
> conference using a speakerphone?  This would cause what is known as
> acoustic echo
> <http://en.wikipedia.org/wiki/Echo_cancellation#Acoustic_echo> and it's
> always my first suspect in a situation like the one you are describing.
>
> This is not a problem that is specific to Asterisk and I'm fairly
> certain there is nothing that can be done within your configuration to
> correct it.  Instructing the conference participants to mute their
> phones when they aren't speaking or to use their handsets should reduce
> acoustic echo.  Some phones
> <http://www.voip-info.org/wiki/view/Uni-Ta+Technology> also claim to
> have a "full-duplex speakerphone with advanced acoustic echo
> cancellation," but caveat emptor.
>
> That said, I'm not an expert on echo cancellation and I have an
> installation where the users are making similar complaints about echo
> during conference calls.  I'd greatly appreciate it if anyone on the
> list corrected any misunderstandings that I might have on the subject.
>
> As an aside, how is the timing on your conference server.  The MeetMe
> application relies on it to mix the audio in conferences.  You should
> get at least 99.98% output from zttest (as shown below) or the audio
> quality will suffer.  This is an overall quality issue and is not
> necessarily related to your echo problems.
>
>  [root at astconf ~]# zttest
>  Opened pseudo zap interface, measuring accuracy...
>  99.999413% 99.995407% 99.995499% 99.998047% 99.996483% 99.997849%
> 99.999008%
>  ...
>  --- Results after 107 passes ---
>  Best: 100.000 -- Worst: 99.995 -- Average: 99.997687, Difference:
> 99.997815
>
> Regards,
>
> Matthew Roth
> InterMedia Marketing Solutions
> Software Engineer and Systems Developer
>
>
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