Silence Suppression from an Audio Conference
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
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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
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@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@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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I have moved that wiki page to Confluence and edited it. https://freeswitch.org/confluence/display/FREESWITCH/VAD+and+CNG Bote From: Steven Ayre Sent: Tuesday, 24 March, 2015 06:44 Subject: Re: [Freeswitch-users] Silence Suppression from an Audio Conference 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
participants (3)
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Bote Man
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Markus von Arx
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Steven Ayre