<div dir="ltr">nobody uses webrtc and classic calls with freeswitch as a media server and kamailio or opensips as a sip proxy ?<div>I found a solution with rtpengine, but I want to use freeswitch as a media server.</div><div><br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">2016-05-06 9:07 GMT+02:00 John Smolka <span dir="ltr"><<a href="mailto:john.smolka9@gmail.com" target="_blank">john.smolka9@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><span style="font-size:12.8px">Hello All, </span><br style="font-size:12.8px"><br style="font-size:12.8px"><span style="font-size:12.8px">I am working with webrtc and I am having problems with incoming audio. </span><br style="font-size:12.8px"><span style="font-size:12.8px">I use : </span><br style="font-size:12.8px"><span style="font-size:12.8px">- Kamailio as a SIP and Websocket proxy </span><br style="font-size:12.8px"><span style="font-size:12.8px">- Freeswitch as a media server. </span><br style="font-size:12.8px"><br style="font-size:12.8px"><span style="font-size:12.8px">UA/WebRTC <--> Kamailio <--> Freeswitch </span><br style="font-size:12.8px"><br style="font-size:12.8px"><span style="font-size:12.8px">For outgoing calls (webrtc->freeswitch) everything works well, but for incoming call(freeswitch->webrtc), my webrtc client was complaining about missing ICE candidates. </span><br style="font-size:12.8px"><span style="font-size:12.8px">In my dialplan I added a new line with: </span><br style="font-size:12.8px"><br style="font-size:12.8px"><span style="font-size:12.8px"><action application="export" data="media_webrtc=true" /> </span><br style="font-size:12.8px"><br style="font-size:12.8px"><span style="font-size:12.8px">since now, RTP/AVP and RTP/SAVP was changed to RTP/SAVPF </span><br style="font-size:12.8px"><br style="font-size:12.8px"><span style="font-size:12.8px">Now, I can receive calls on my webrtc client but it does not work on some sip phones. </span><br style="font-size:12.8px"><span style="font-size:12.8px">On sip phones it is rejected with 488 Not Acceptable Here </span><br style="font-size:12.8px"><br style="font-size:12.8px"><span style="font-size:12.8px">SDP from freeswitch's INVITE looks like : </span><br style="font-size:12.8px"><br style="font-size:12.8px"><span style="font-size:12.8px">v=0. </span><br style="font-size:12.8px"><span style="font-size:12.8px">o=My-SBC 1462416600 1462416601 IN IP4 MyPublicIP. </span><br style="font-size:12.8px"><span style="font-size:12.8px">s=My-SBC. </span><br style="font-size:12.8px"><span style="font-size:12.8px">c=IN IP4 MyPublicIP. </span><br style="font-size:12.8px"><span style="font-size:12.8px">t=0 0. </span><br style="font-size:12.8px"><span style="font-size:12.8px">a=msid-semantic: WMS 21zmABqHTMGtfiYcvg2nQyTyTBJpmP</span><span style="font-size:12.8px">mf. </span><br style="font-size:12.8px"><span style="font-size:12.8px">m=audio 32204 RTP/SAVPF 8 0 101 13. </span><br style="font-size:12.8px"><span style="font-size:12.8px">a=rtpmap:8 PCMA/8000. </span><br style="font-size:12.8px"><span style="font-size:12.8px">a=rtpmap:0 PCMU/8000. </span><br style="font-size:12.8px"><span style="font-size:12.8px">a=rtpmap:101 telephone-event/8000. </span><br style="font-size:12.8px"><span style="font-size:12.8px">a=fingerprint:sha-256 B8:7C:42:58:26:FE:30:27:DB:F5:</span><span style="font-size:12.8px">71:21:11:96:AF:0B:DD:6E:79:9A:</span><span style="font-size:12.8px">AC:FF:86:69:89:14:88:70:AA:EE:</span><span style="font-size:12.8px">4A:EB. </span><br style="font-size:12.8px"><span style="font-size:12.8px">a=setup:actpass. </span><br style="font-size:12.8px"><span style="font-size:12.8px">a=rtcp-mux. </span><br style="font-size:12.8px"><span style="font-size:12.8px">a=rtcp:32204 IN IP4 MyPublicIP. </span><br style="font-size:12.8px"><span style="font-size:12.8px">a=ssrc:590458548 cname:GLr7dIFWlcfql4sH. </span><br style="font-size:12.8px"><span style="font-size:12.8px">a=ssrc:590458548 msid:</span><span style="font-size:12.8px">21zmABqHTMGtfiYcvg2nQyTyTBJpmP</span><span style="font-size:12.8px">mf a0. </span><br style="font-size:12.8px"><span style="font-size:12.8px">a=ssrc:590458548 mslabel:</span><span style="font-size:12.8px">21zmABqHTMGtfiYcvg2nQyTyTBJpmP</span><span style="font-size:12.8px">mf. </span><br style="font-size:12.8px"><span style="font-size:12.8px">a=ssrc:590458548 label:</span><span style="font-size:12.8px">21zmABqHTMGtfiYcvg2nQyTyTBJpmP</span><span style="font-size:12.8px">mfa0. </span><br style="font-size:12.8px"><span style="font-size:12.8px">a=ice-ufrag:akVRJFgs6QHjvUo2. </span><br style="font-size:12.8px"><span style="font-size:12.8px">a=ice-pwd:</span><span style="font-size:12.8px">vG0zewmpyrGKmZhm924Jxa1u. </span><br style="font-size:12.8px"><span style="font-size:12.8px">a=candidate:9311553015 1 udp 659136 MyPublicIP 32204 typ host generation 0. </span><br style="font-size:12.8px"><span style="font-size:12.8px">a=candidate:9311553015 2 udp 659136 MyPublicIP 32204 typ host generation 0. </span><br style="font-size:12.8px"><span style="font-size:12.8px">a=ptime:20. </span><br style="font-size:12.8px"><br style="font-size:12.8px"><span style="font-size:12.8px">Maybe I don't understand this correctly, but is that possible to add SAVPF, AVP and SAVP into a INVITES's SDP so I can remove them on kamailio based on transport towards client (TLS,UDP,TCP,WSS) ? </span><br style="font-size:12.8px"><br style="font-size:12.8px"><span style="font-size:12.8px">I have tested it on the latest freeswitch 1.4 and 1.6 versions. </span><br></div>
</blockquote></div><br></div>