Hi Steven Thanks for your reply. I actually already know that wiki page. But all those configuration variables there don't work - at least not for SIP channels that are connected to a mod_conference audio conference. Maybe they do work for bridged calls, but that's not what I need. Also, the wiki page does not mention conferences at all. And the sentence "When FreeSWITCH does not detect speech, it stops transmitting RTP" seems not to apply to mod_conference. I probably just have configured mod_conference incorrectly, but I don't know where to check. So any information or advice about SIP channels connected to a mod_conference audio conference? Thanks, Markus 2015-03-24 11:43 GMT+01:00 Steven Ayre <steveayre@gmail.com>:
https://wiki.freeswitch.org/wiki/VAD_and_CNG
On 24 March 2015 at 07:04, Markus von Arx <mkvonarx@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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