[Asterisk-Users] How to deal with echo in MeetMe?

BJ Weschke bweschke at gmail.com
Mon Nov 21 13:01:56 MST 2005


On 11/21/05, steve at daviesfam.org <steve at daviesfam.org> wrote:
>
>
> On Mon, 21 Nov 2005, Tony Mountifield wrote:
>
> > I have a customer who is running fairly large conferences (between 5
> > and 30 participants) on their Asterisk box. It uses SIP to talk to a
> > PSTN provider.
> >
> > They are complaining that under some circumstances they experience
> > echo of one or more participants. On listening in to one of their
> > conferences, it seemed to me that the echo was being introduced
> > via the microphone of a couple of specific participants, as it was
> > possible to eliminate this echo by muting those participants.
> >
> > On discussing the participants' environments with the customer, it
> > would appear that the problem occurs when participants are using
> > speaker phones and there are multiple participants in proximity to
> > each other, such that one participant's phone can hear the audio
> > from that of another participant in the same conference.
> >
> > It's my supposition that any echo canceller is going to have
> > difficulties correcting for that scenario. Am I correct?
> >
> > The problem I have is that the customer insists that their existing
> > conferencing supplier (whom our kit is supposed to replace) does
> > not suffer from this echo, in the same participant environment.
> >
> > I am assured by our PSTN supplier that there is full echo suppression
> > on the PSTN lines. Am I correct in believing that further echo
> > suppression is neither possible nor required at the SIP interface
> > within Asterisk?
> >
> > Any advice on how to approach this would be appreciated.
>
>
> Hmm,
>
> That's a difficult one to resolve.  I presume that the existing
> conferencing system has low latency and therefore the crosstalk is
> not noticable.
>
> Your solution is using a remote SIP provider - that will mean higher
> latency.  Meetme adds some too.
>
> Moving from a remote SIP provider to PSTN to a locally connected PSTN
> would reduce the latency, whether it would be enough to avoid the effect
> I'm not sure.
>
> Perhaps we could also add a fairly brutal "noise gate" into Meetme which
> mutes off all but the loudest participant or participants; not sure what
> that would sound like, but it would probably hide the crosstalk...
>
> I do think that a bunch of speakerphones that can hear one another, all
> connected into a conference bridge with latency is an impossible echo
> cancellation task.  So either the latency needs to be removed or an echo
> suppression approach needs to be taken.
>

 In a prior job, while working with DSP based conference bridges, the
way they dealt with this was exactly your "noise gate" as you describe
it. They called it the "Active Talkers List". There were
qualifications for you to be added and removed from the active talkers
list, and there was a finite number of people that were allowed at any
given time to be a participant in the ATL. It wasn't always 100%
successful and the real drawback from the systems I worked with was
that configuration of parameters with regard to the ATL was a system
wide setting and we often wanted to be able to tune this for a couple
of (LARGE) clients who were complaining when the majority of them were
not.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/



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