[Freeswitch-users] Video-only call

Fri May 2 15:43:21 MSD 2014

Hello,

Does FreeSWITCH support receiving calls that contain only a video component?
(I am trying to use FreeSWITCH to distribute WebRTC screenshare streams to users.  It works fine with audio and video)

The INVITE looks like:
   INVITE sip:1273936153_290033102 at a.b.c.d SIP/2.0
   Via: SIP/2.0/UDP 0.0.0.0:7071;branch=z9hG4bK801683;rport
   Via: SIP/2.0/TCP 127.0.0.1:7071;branch=z9hG4bKllc4a6rbqb7886s43nhl
   Via: SIP/2.0/TCP 1q582tjutlhh.invalid;branch=z9hG4bK7896356
   max-forwards: 69
   To:  <sip:1273936153_290033102 at a.b.c.d>
   From: "Rich" <sip:rscreene at gmail.com>;tag=mgu8fejfq8
   call-id: llc4a6rbqb7886s43nhl
   CSeq: 7243 INVITE
   Contact:  <sip:127.0.0.1:7071;transport=tcp>
   allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
   supported: outbound
   user-agent: JsSIP 0.4.0-devel
   content-length: 1824
   Content-Type: application/sdp

   v=0
   o=- 4235676450597780712 2 IN IP4 127.0.0.1
   s=-
   t=0 0
   a=group:BUNDLE video
   a=msid-semantic: WMS 1dVXakoYx2u585L7G09f2C45EW8vP8c1XVcd
   m=video 34116 RTP/SAVPF 100 116 117
   c=IN IP4 e.f.g.h
   a=rtcp:34116 IN IP4 e.f.g.h
   a=candidate:2668419503 1 udp 2122260223 192.168.47.203 56323 typ host generation 0
   a=candidate:2668419503 2 udp 2122260223 192.168.47.203 56323 typ host generation 0
   a=candidate:3515819359 1 tcp 1518280447 192.168.47.203 0 typ host generation 0
   a=candidate:3515819359 2 tcp 1518280447 192.168.47.203 0 typ host generation 0
   a=candidate:1401022012 1 udp 1686052607 w.x.y.z 7940 typ srflx raddr 192.168.47.203 rport 56323 generation 0
   a=candidate:1401022012 2 udp 1686052607 w.x.y.z 7940 typ srflx raddr 192.168.47.203 rport 56323 generation 0
   a=candidate:3671437516 1 udp 41885439 e.f.g.h 34116 typ relay raddr w.x.y.z rport 7940 generation 0
   a=candidate:3671437516 2 udp 41885439 e.f.g.h 34116 typ relay raddr w.x.y.z rport 7940 generation 0
   a=ice-ufrag:vD/mqbgKKKvb+vwg
   a=ice-pwd:xbg5CyTfAIQMCD3JTmjU9IGw
   a=ice-options:google-ice
   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
   a=setup:actpass
   a=mid:video
   a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
   a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
   a=sendrecv
   a=rtcp-mux
   a=rtpmap:100 VP8/90000
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 nack pli
   a=rtcp-fb:100 goog-remb
   a=rtpmap:116 red/90000
   a=rtpmap:117 ulpfec/90000
   a=ssrc:2554093369 cname:L3M//yZhkStQCiOE
   a=ssrc:2554093369 msid:1dVXakoYx2u585L7G09f2C45EW8vP8c1XVcd 908bd517-4202-4348-a36d-f474090f3cc2
   a=ssrc:2554093369 mslabel:1dVXakoYx2u585L7G09f2C45EW8vP8c1XVcd
   a=ssrc:2554093369 label:908bd517-4202-4348-a36d-f474090f3cc2

In the log I see the following message:
2014-05-02 13:35:43.267313 [DEBUG] switch_core_media.c:2223 No audio codec available

And FreeSWITCH responds with:
   SIP/2.0 488 Not Acceptable Here
   Via: SIP/2.0/UDP 0.0.0.0:7071;branch=z9hG4bK801683;rport=7071;received=a.b.c.d
   Via: SIP/2.0/TCP 127.0.0.1:7071;branch=z9hG4bKllc4a6rbqb7886s43nhl
   Via: SIP/2.0/TCP 1q582tjutlhh.invalid;branch=z9hG4bK7896356
   Max-Forwards: 692014-05-02 13:35:43.267313 [DEBUG] switch_core_state_machine.c:818 (sofia/external/rscreene at gmail.com) State REPORTING

   From: "Rich" <sip:rscreene at gmail.com>;tag=mgu8fejfq8
2014-05-02 13:35:43.267313 [DEBUG] switch_core_state_machine.c:102 sofia/external/rscreene at gmail.com Standard REPORTING, cause: INCOMPATIBLE_DESTINATION
   To: <sip:1273936153_290033102 at a.b.c.d>;tag=H3e25S58vjNZQ
   call-id: llc4a6rbqb7886s43nhl
2014-05-02 13:35:43.267313 [DEBUG] switch_core_state_machine.c:818 (sofia/external/rscreene at gmail.com) State REPORTING going to sleep
   CSeq: 7243 INVITE
   User-Agent: Drum 1.0
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, refer
   Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
   Content-Length: 0
   Remote-Party-ID: "1273936153_290033102" <sip:1273936153_290033102 at a.b.c.d>;party=calling;privacy=off;screen=no

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?

Many thanks,
  Richard

Richard Screene
Senior Developer
Drum Web Meetings 
+44 1273 936125
www.thisisdrum.com
Drum is the collaboration solution by NetDev Ltd.
Registered in England and Wales 
Company Number 04741258

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