<div dir="ltr"><div class="gmail_default" style="font-family:arial,helvetica,sans-serif">Hi Bote</div><div class="gmail_default" style="font-family:arial,helvetica,sans-serif"><br></div><div class="gmail_default" style="font-family:arial,helvetica,sans-serif">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.</div><div class="gmail_default" style="font-family:arial,helvetica,sans-serif"><br></div><div class="gmail_default" style="font-family:arial,helvetica,sans-serif">Markus</div><div class="gmail_default" style="font-family:arial,helvetica,sans-serif"><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">2015-03-24 16:06 GMT+01:00 Bote Man <span dir="ltr"><<a href="mailto:bote_radio@botecomm.com" target="_blank">bote_radio@botecomm.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">There is a setting listed in<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></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" target="_blank">https://freeswitch.org/confluence/display/FREESWITCH/mod_conference</a><u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">energy-level<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></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.<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></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.<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></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. <u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Thanks.<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Bote<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></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<span class=""><br><b>Subject:</b> Re: [Freeswitch-users] Silence Suppression from an Audio Conference<u></u><u></u></span></span></p></div></div><p class="MsoNormal"><u></u> <u></u></p><div><div><p class="MsoNormal"><span style="font-family:"Arial","sans-serif"">Hi Steven<u></u><u></u></span></p></div><span class=""><div><p class="MsoNormal"><span style="font-family:"Arial","sans-serif""><u></u> <u></u></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.<u></u><u></u></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.<u></u><u></u></span></p></div><div><p class="MsoNormal"><u></u> <u></u></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?<u></u><u></u></span></p></div><div><p class="MsoNormal"><span style="font-family:"Arial","sans-serif""><u></u> <u></u></span></p></div><div><p class="MsoNormal"><span style="font-family:"Arial","sans-serif"">Thanks, Markus<u></u><u></u></span></p></div><div><p class="MsoNormal"><span style="font-family:"Arial","sans-serif""><u></u> <u></u></span></p></div></div></span></div><span class=""><div><p class="MsoNormal"><u></u> <u></u></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>>:<u></u><u></u></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><u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></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:<u></u><u></u></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""><u></u><u></u></span></p></div><div><p class="MsoNormal"><span style="font-family:"Arial","sans-serif""><u></u> <u></u></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""><u></u><u></u></span></p></div><div><p class="MsoNormal"><span style="font-family:"Arial","sans-serif""><u></u> <u></u></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""><u></u><u></u></span></p></div><div><p class="MsoNormal"><span style="font-family:"Arial","sans-serif""><u></u> <u></u></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><u></u><u></u></p></div><div><p class="MsoNormal"><span style="font-family:"Arial","sans-serif""><u></u> <u></u></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""><u></u><u></u></span></p></div></div><p class="MsoNormal"><u></u> <u></u></p></div></div></blockquote></div></div></div><p class="MsoNormal"><u></u> <u></u></p></div></span></div></div></div><br>_________________________________________________________________________<br>
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