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 <[email protected]> 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. > > [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 >
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