asterisk users - Nov 2005 - How to deal with echo in MeetMe?

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.

Cheers
Tony
-- 
Tony Mountifield
Work: tony@softins.co.uk - http://www.softins.co.uk
Play: tony@mountifield.org - http://tony.mountifield.org

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.

Steve