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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>There is a setting listed in<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><a href="https://freeswitch.org/confluence/display/FREESWITCH/mod_conference">https://freeswitch.org/confluence/display/FREESWITCH/mod_conference</a><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>energy-level<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>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.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>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.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I recommend you test this theory in case it is helpful and please report back with your findings. <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Thanks.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Bote<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div style='border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in 4.0pt'><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Markus von Arx<br><b>Sent:</b> Tuesday, 24 March, 2015 08:58<br><b>Subject:</b> Re: [Freeswitch-users] Silence Suppression from an Audio Conference<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><div><div><p class=MsoNormal><span style='font-family:"Arial","sans-serif"'>Hi Steven<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-family:"Arial","sans-serif"'><o:p> </o:p></span></p></div><p class=MsoNormal>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.<o:p></o:p></p><div><p class=MsoNormal><span style='font-family:"Arial","sans-serif"'> I probably just have configured mod_conference incorrectly, but I don't know where to check.<o:p></o:p></span></p></div><div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal><span style='font-family:"Arial","sans-serif"'>So any information or advice about SIP channels connected to a mod_conference audio conference?<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-family:"Arial","sans-serif"'><o:p> </o:p></span></p></div><div><p class=MsoNormal><span style='font-family:"Arial","sans-serif"'>Thanks, Markus<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-family:"Arial","sans-serif"'><o:p> </o:p></span></p></div></div></div><div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>2015-03-24 11:43 GMT+01:00 Steven Ayre <<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>>:<o:p></o:p></p><div><p class=MsoNormal><a href="https://wiki.freeswitch.org/wiki/VAD_and_CNG" target="_blank">https://wiki.freeswitch.org/wiki/VAD_and_CNG</a><o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p><div><div><div><p class=MsoNormal>On 24 March 2015 at 07:04, Markus von Arx <<a href="mailto:mkvonarx@gmail.com" target="_blank">mkvonarx@gmail.com</a>> wrote:<o:p></o:p></p></div></div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in'><div><div><div><div><p class=MsoNormal><span style='font-size:9.5pt;font-family:"Arial","sans-serif"'>Hi</span><span style='font-family:"Arial","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-family:"Arial","sans-serif"'><o:p> </o:p></span></p></div><div><p class=MsoNormal><span style='font-size:9.5pt;font-family:"Arial","sans-serif"'>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?</span><span style='font-family:"Arial","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-family:"Arial","sans-serif"'><o:p> </o:p></span></p></div><div><p class=MsoNormal><span style='font-size:9.5pt;font-family:"Arial","sans-serif"'>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.</span><span style='font-family:"Arial","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-family:"Arial","sans-serif"'><o:p> </o:p></span></p></div><div><p class=MsoNormal><span style='font-size:9.5pt'>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.</span><o:p></o:p></p></div><div><p class=MsoNormal><span style='font-family:"Arial","sans-serif"'><o:p> </o:p></span></p></div><div><p class=MsoNormal><span style='font-size:9.5pt;font-family:"Arial","sans-serif"'>Thanks, Markus</span><span style='font-family:"Arial","sans-serif"'><o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p></div></div></blockquote></div></div></div><p class=MsoNormal><o:p> </o:p></p></div></div></div></body></html>