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

[Alessandro Russo]

Hi Asterisk Users,

we are using Asterisk 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/  ??

THX
Bye

Alessandro R.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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

[Matthew J. Roth]

Alessandro Russo wrote:
>
> we are using Asterisk 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 
 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 
 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.

  [r...@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


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [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  wrote:

> Alessandro Russo wrote:
> >
> > we are using Asterisk 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
>  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
>  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.
>
>  [r...@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
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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

[Matthew J. Roth]

Alessandro Russo wrote:
> 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

Alessandro,

I'm sorry to hear that your problem isn't acoustic echo.  I'll be 
following this thread to see if anyone offers you any suggestions and 
I'll let you know if I discover anything that improves the echo problem 
in my installation.

What is the timing source in the conference server?  In general, it will 
be either a Zaptel/DAHDI hardware device or the ztdummy/dahdi-dummy 
module.  See this page 
 for details.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users