[Freeswitch-users] Silence Suppression from an Audio Conference

[Markus von Arx]

Hi

Can anyone tell me if FreeSWITCH supports silence suppression for SIP calls
that are inside a FreeSWITCH audio conference? If yes, how do I configure
mod_conference, mod_sofia and FreeSWITCH core to enable this feature?

More precisely, I try to enable/activate the behavior described in RFC 3389
for G.711 in such a way that there are only RTP packets of type 13 every 1
or 2 seconds. I tried to play around with some possible settings but could
never observe anything else then the regular G.711 PCMU RTP packets on the
wire. Even when I set the SIP call to 'deaf' via the FreeSWITCH console,
mod_conference/mod_sofia continue to send G.711 PCMU RTP packets every 20ms.

It's possible that I completly misunderstand RFC 3389 and the concepts of
silence suppression, comfort noise etc. In the end, what I try
to achieve is to reduce the network bandwidth of a G.711 SIP channel during
the periods when the FreeSWITCH only sends silence over the SIP channel.
Unfortunately, we're stuck with G.711 at the moment, so I cannot switch to
another codec.

Thanks, Markus
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