Hi to all,<br><br>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 :( <br>
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<br>
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.<br><br>I performed the zttest and I obtained the following results: <br clear="all">
<br>asterisk:~# zttest <br>Opened pseudo zap interface, measuring accuracy...<br>99.966690% 99.971863% 99.936729% 99.967766% 99.936913% 99.968163% 99.967667% <br>99.936623% 99.969818% 99.937019% 99.967972% 99.937012% 99.968063% 99.967865% 99.936440% <br>
99.967766% 99.935356% 99.967667% 99.937401% 99.968460% 99.967667% 99.936333% <br>--- Results after 22 passes ---<br>Best: 99.972 -- Worst: 99.935 -- Average: 99.955330, Difference: 99.992836<br><br>Any suggestions?<br><br>
Alessandro R.<br>
<br><br><div class="gmail_quote">On Fri, Dec 12, 2008 at 7:39 PM, Matthew J. Roth <span dir="ltr"><<a href="mailto:mroth@imminc.com">mroth@imminc.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Alessandro Russo wrote:<br>
><br>
> we are using Asterisk <a href="http://1.4.18.1" target="_blank">1.4.18.1</a> <<a href="http://1.4.18.1/" target="_blank">http://1.4.18.1/</a>> on debian 4.0 etch,<br>
<div class="Ih2E3d">> pwlib 1.10 and openh323 1.18.<br>
><br>
> We are using MeetMe for conference calls and with two participants<br>
> there is no echo problems, but with more than two participants there<br>
> is a lot of echo that sometimes disappear for a short time and all<br>
> function well.<br>
><br>
> Someone have some suggestions??<br>
><br>
> Do you ever used app_conference<br>
> <a href="http://sourceforge.net/projects/appconference/" target="_blank">http://sourceforge.net/projects/appconference/</a> ??<br>
><br>
<br>
</div>Alessandro,<br>
<br>
Are you certain that the echo isn't being introduced by someone on the<br>
conference using a speakerphone? This would cause what is known as<br>
acoustic echo<br>
<<a href="http://en.wikipedia.org/wiki/Echo_cancellation#Acoustic_echo" target="_blank">http://en.wikipedia.org/wiki/Echo_cancellation#Acoustic_echo</a>> and it's<br>
always my first suspect in a situation like the one you are describing.<br>
<br>
This is not a problem that is specific to Asterisk and I'm fairly<br>
certain there is nothing that can be done within your configuration to<br>
correct it. Instructing the conference participants to mute their<br>
phones when they aren't speaking or to use their handsets should reduce<br>
acoustic echo. Some phones<br>
<<a href="http://www.voip-info.org/wiki/view/Uni-Ta+Technology" target="_blank">http://www.voip-info.org/wiki/view/Uni-Ta+Technology</a>> also claim to<br>
have a "full-duplex speakerphone with advanced acoustic echo<br>
cancellation," but caveat emptor.<br>
<br>
That said, I'm not an expert on echo cancellation and I have an<br>
installation where the users are making similar complaints about echo<br>
during conference calls. I'd greatly appreciate it if anyone on the<br>
list corrected any misunderstandings that I might have on the subject.<br>
<br>
As an aside, how is the timing on your conference server. The MeetMe<br>
application relies on it to mix the audio in conferences. You should<br>
get at least 99.98% output from zttest (as shown below) or the audio<br>
quality will suffer. This is an overall quality issue and is not<br>
necessarily related to your echo problems.<br>
<br>
[root@astconf ~]# zttest<br>
Opened pseudo zap interface, measuring accuracy...<br>
99.999413% 99.995407% 99.995499% 99.998047% 99.996483% 99.997849%<br>
99.999008%<br>
...<br>
--- Results after 107 passes ---<br>
Best: 100.000 -- Worst: 99.995 -- Average: 99.997687, Difference:<br>
99.997815<br>
<br>
Regards,<br>
<br>
Matthew Roth<br>
InterMedia Marketing Solutions<br>
Software Engineer and Systems Developer<br>
<br>
<br>
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</blockquote></div><br>