<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div>Hello,</div><div><br></div><div>Does FreeSWITCH support receiving calls that contain only a video component?</div><div>(I am trying to use FreeSWITCH to distribute WebRTC screenshare streams to users. It works fine with audio and video)</div><div><br></div><div>The INVITE looks like:</div><div><div> INVITE <a href="sip:1273936153_290033102@a.b.c.d">sip:1273936153_290033102@a.b.c.d</a> SIP/2.0</div><div> Via: SIP/2.0/UDP 0.0.0.0:7071;branch=z9hG4bK801683;rport</div><div> Via: SIP/2.0/TCP 127.0.0.1:7071;branch=z9hG4bKllc4a6rbqb7886s43nhl</div><div> Via: SIP/2.0/TCP 1q582tjutlhh.invalid;branch=z9hG4bK7896356</div><div> max-forwards: 69</div><div> To: <<a href="sip:1273936153_290033102@a.b.c.d">sip:1273936153_290033102@a.b.c.d</a>></div><div> From: "Rich" <<a href="sip:rscreene@gmail.com">sip:rscreene@gmail.com</a>>;tag=mgu8fejfq8</div><div> call-id: llc4a6rbqb7886s43nhl</div><div> CSeq: 7243 INVITE</div><div> Contact: <<a href="sip:127.0.0.1:7071;transport=tcp">sip:127.0.0.1:7071;transport=tcp</a>></div><div> allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE</div><div> supported: outbound</div><div> user-agent: JsSIP 0.4.0-devel</div><div> content-length: 1824</div><div> Content-Type: application/sdp</div><div> </div><div> v=0</div><div> o=- 4235676450597780712 2 IN IP4 127.0.0.1</div><div> s=-</div><div> t=0 0</div><div> a=group:BUNDLE video</div><div> a=msid-semantic: WMS 1dVXakoYx2u585L7G09f2C45EW8vP8c1XVcd</div><div> m=video 34116 RTP/SAVPF 100 116 117</div><div> c=IN IP4 e.f.g.h</div><div> a=rtcp:34116 IN IP4 e.f.g.h</div><div> a=candidate:2668419503 1 udp 2122260223 192.168.47.203 56323 typ host generation 0</div><div> a=candidate:2668419503 2 udp 2122260223 192.168.47.203 56323 typ host generation 0</div><div> a=candidate:3515819359 1 tcp 1518280447 192.168.47.203 0 typ host generation 0</div><div> a=candidate:3515819359 2 tcp 1518280447 192.168.47.203 0 typ host generation 0</div><div> a=candidate:1401022012 1 udp 1686052607 w.x.y.z 7940 typ srflx raddr 192.168.47.203 rport 56323 generation 0</div><div> a=candidate:1401022012 2 udp 1686052607 w.x.y.z 7940 typ srflx raddr 192.168.47.203 rport 56323 generation 0</div><div> a=candidate:3671437516 1 udp 41885439 e.f.g.h 34116 typ relay raddr w.x.y.z rport 7940 generation 0</div><div> a=candidate:3671437516 2 udp 41885439 e.f.g.h 34116 typ relay raddr w.x.y.z rport 7940 generation 0</div><div> a=ice-ufrag:vD/mqbgKKKvb+vwg</div><div> a=ice-pwd:xbg5CyTfAIQMCD3JTmjU9IGw</div><div> a=ice-options:google-ice</div><div> a=fingerprint:sha-256 E8:6D:CF:75:43:C6:D9:C6:90:8E:FD:15:CD:FC:86:A2:90:E6:A3:BE:CF:A1:12:64:60:2F:26:89:96:C0:16:18</div><div> a=setup:actpass</div><div> a=mid:video</div><div> a=extmap:2 urn:ietf:params:rtp-hdrext:toffset</div><div> a=extmap:3 <a href="http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time">http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time</a></div><div> a=sendrecv</div><div> a=rtcp-mux</div><div> a=rtpmap:100 VP8/90000</div><div> a=rtcp-fb:100 ccm fir</div><div> a=rtcp-fb:100 nack</div><div> a=rtcp-fb:100 nack pli</div><div> a=rtcp-fb:100 goog-remb</div><div> a=rtpmap:116 red/90000</div><div> a=rtpmap:117 ulpfec/90000</div><div> a=ssrc:2554093369 cname:L3M//yZhkStQCiOE</div><div> a=ssrc:2554093369 msid:1dVXakoYx2u585L7G09f2C45EW8vP8c1XVcd 908bd517-4202-4348-a36d-f474090f3cc2</div><div> a=ssrc:2554093369 mslabel:1dVXakoYx2u585L7G09f2C45EW8vP8c1XVcd</div><div> a=ssrc:2554093369 label:908bd517-4202-4348-a36d-f474090f3cc2</div></div><div><br></div><div>In the log I see the following message:</div><div><div>2014-05-02 13:35:43.267313 [DEBUG] switch_core_media.c:2223 No audio codec available</div></div><div><br></div><div>And FreeSWITCH responds with:</div><div><div> SIP/2.0 488 Not Acceptable Here</div><div> Via: SIP/2.0/UDP 0.0.0.0:7071;branch=z9hG4bK801683;rport=7071;received=a.b.c.d</div><div> Via: SIP/2.0/TCP 127.0.0.1:7071;branch=z9hG4bKllc4a6rbqb7886s43nhl</div><div> Via: SIP/2.0/TCP 1q582tjutlhh.invalid;branch=z9hG4bK7896356</div><div> Max-Forwards: 692014-05-02 13:35:43.267313 [DEBUG] switch_core_state_machine.c:818 (<a href="mailto:sofia/external/rscreene@gmail.com">sofia/external/rscreene@gmail.com</a>) State REPORTING</div><div><br></div><div> From: "Rich" <<a href="sip:rscreene@gmail.com">sip:rscreene@gmail.com</a>>;tag=mgu8fejfq8</div><div>2014-05-02 13:35:43.267313 [DEBUG] switch_core_state_machine.c:102 <a href="mailto:sofia/external/rscreene@gmail.com">sofia/external/rscreene@gmail.com</a> Standard REPORTING, cause: INCOMPATIBLE_DESTINATION</div><div> To: <<a href="sip:1273936153_290033102@a.b.c.d">sip:1273936153_290033102@a.b.c.d</a>>;tag=H3e25S58vjNZQ</div><div> call-id: llc4a6rbqb7886s43nhl</div><div>2014-05-02 13:35:43.267313 [DEBUG] switch_core_state_machine.c:818 (<a href="mailto:sofia/external/rscreene@gmail.com">sofia/external/rscreene@gmail.com</a>) State REPORTING going to sleep</div><div> CSeq: 7243 INVITE</div><div> User-Agent: Drum 1.0</div><div> Accept: application/sdp</div><div> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY</div><div> Supported: timer, path, replaces</div><div> Allow-Events: talk, hold, conference, refer</div><div> Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"</div><div> Content-Length: 0</div><div> Remote-Party-ID: "1273936153_290033102" <<a href="sip:1273936153_290033102@a.b.c.d">sip:1273936153_290033102@a.b.c.d</a>>;party=calling;privacy=off;screen=no</div></div><div><br></div><div><br></div><div>Do I have to configure something to allow video-only calls? Or, is it a limitation of FreeSWITCH? If it is a limitation of FreeSWITCH is it easy to remove?</div><div><br></div><div>Many thanks,</div><div> Richard</div><div><br></div><br><div apple-content-edited="true">
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