[Freeswitch-users] Silence Suppression from an Audio Conference

[Markus von Arx]

Hi Bote

I did play around with those settings some more and came to the conclusion,
that VAD and CNG do indeed work. Just not in the way I want. I observed
that the FreeSWITCH always continues to send G.711 PCMU RTP packets every
20 milliseconds no matter if the RTP packets carry user voice or CNG
payload, and that the FreeSWITCH never uses RTP type 13 for sending CNG,
and that the FreeSWITCH never does actual silence suppression. I'll open
another discussion with more specific questions about silence suppression
without the "distraction" of audio conferencing.

Markus


2015-03-24 16:06 GMT+01:00 Bote Man <bote_radio at botecomm.com>:

> There is a setting listed in
>
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/mod_conference
>
>
>
> energy-level
>
>
>
> which acts as a noise gate. If you set this number high enough, the
> conference bridge will only admit audio from a conferee when it detects
> speech (or noise?) from him.
>
>
>
> HOWEVER, there used to be a conference flag named “waste” that told the
> conference to “waste bandwidth” by transmitting packets all the time, even
> when there was no audio contained in them; now that flag has been
> eliminated and I understand that the conference bridge always sends
> packets. If I have this correct, then even the noise gate will not reduce
> your bandwidth.
>
>
>
> I recommend you test this theory in case it is helpful and please report
> back with your findings.
>
>
>
> Thanks.
>
>
>
> Bote
>
>
>
>
>
> *From:* Markus von Arx
> *Sent:* Tuesday, 24 March, 2015 08:58
> *Subject:* Re: [Freeswitch-users] Silence Suppression from an Audio
> Conference
>
>
>
> 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 at gmail.com>:
>
> 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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