<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">Freeswitch can already translate between AVP and SAVPF. Why would you manipulate the sdp? Shouldn't be necessary<div class=""><br class=""><div class=""><br class=""><div><blockquote type="cite" class=""><div class="">On May 12, 2016, at 10:06 AM, John Smolka <<a href="mailto:john.smolka9@gmail.com" class="">john.smolka9@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class="">nobody uses webrtc and classic calls with freeswitch as a media server and kamailio or opensips as a sip proxy ?<div class="">I found a solution with rtpengine, but I want to use freeswitch as a media server.</div><div class=""><br class=""></div><div class=""><br class=""></div></div><div class="gmail_extra"><br class=""><div class="gmail_quote">2016-05-06 9:07 GMT+02:00 John Smolka <span dir="ltr" class=""><<a href="mailto:john.smolka9@gmail.com" target="_blank" class="">john.smolka9@gmail.com</a>></span>:<br class=""><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr" class=""><span style="font-size:12.8px" class="">Hello All, </span><br style="font-size:12.8px" class=""><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">I am working with webrtc and I am having problems with incoming audio. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">I use : </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">- Kamailio as a SIP and Websocket proxy </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">- Freeswitch as a media server. </span><br style="font-size:12.8px" class=""><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">UA/WebRTC <--> Kamailio <--> Freeswitch </span><br style="font-size:12.8px" class=""><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">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" class=""><span style="font-size:12.8px" class="">In my dialplan I added a new line with: </span><br style="font-size:12.8px" class=""><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class=""><action application="export" data="media_webrtc=true" /> </span><br style="font-size:12.8px" class=""><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">since now, RTP/AVP and RTP/SAVP was changed to RTP/SAVPF </span><br style="font-size:12.8px" class=""><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">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" class=""><span style="font-size:12.8px" class="">On sip phones it is rejected with 488 Not Acceptable Here </span><br style="font-size:12.8px" class=""><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">SDP from freeswitch's INVITE looks like : </span><br style="font-size:12.8px" class=""><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">v=0. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">o=My-SBC 1462416600 1462416601 IN IP4 MyPublicIP. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">s=My-SBC. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">c=IN IP4 MyPublicIP. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">t=0 0. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">a=msid-semantic: WMS 21zmABqHTMGtfiYcvg2nQyTyTBJpmP</span><span style="font-size:12.8px" class="">mf. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">m=audio 32204 RTP/SAVPF 8 0 101 13. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">a=rtpmap:8 PCMA/8000. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">a=rtpmap:0 PCMU/8000. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">a=rtpmap:101 telephone-event/8000. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">a=fingerprint:sha-256 B8:7C:42:58:26:FE:30:27:DB:F5:</span><span style="font-size:12.8px" class="">71:21:11:96:AF:0B:DD:6E:79:9A:</span><span style="font-size:12.8px" class="">AC:FF:86:69:89:14:88:70:AA:EE:</span><span style="font-size:12.8px" class="">4A:EB. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">a=setup:actpass. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">a=rtcp-mux. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">a=rtcp:32204 IN IP4 MyPublicIP. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">a=ssrc:590458548 cname:GLr7dIFWlcfql4sH. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">a=ssrc:590458548 msid:</span><span style="font-size:12.8px" class="">21zmABqHTMGtfiYcvg2nQyTyTBJpmP</span><span style="font-size:12.8px" class="">mf a0. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">a=ssrc:590458548 mslabel:</span><span style="font-size:12.8px" class="">21zmABqHTMGtfiYcvg2nQyTyTBJpmP</span><span style="font-size:12.8px" class="">mf. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">a=ssrc:590458548 label:</span><span style="font-size:12.8px" class="">21zmABqHTMGtfiYcvg2nQyTyTBJpmP</span><span style="font-size:12.8px" class="">mfa0. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">a=ice-ufrag:akVRJFgs6QHjvUo2. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">a=ice-pwd:</span><span style="font-size:12.8px" class="">vG0zewmpyrGKmZhm924Jxa1u. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">a=candidate:9311553015 1 udp 659136 MyPublicIP 32204 typ host generation 0. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">a=candidate:9311553015 2 udp 659136 MyPublicIP 32204 typ host generation 0. </span><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">a=ptime:20. </span><br style="font-size:12.8px" class=""><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">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" class=""><br style="font-size:12.8px" class=""><span style="font-size:12.8px" class="">I have tested it on the latest freeswitch 1.4 and 1.6 versions. </span><br class=""></div>
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