[Freeswitch-users] Silence Suppression from an Audio Conference

[Steven Ayre]

https://wiki.freeswitch.org/wiki/VAD_and_CNG

On 24 March 2015 at 07:04, Markus von Arx <mkvonarx at gmail.com> wrote:

> 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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