stream2beamer.md

   1 # Stream Desktop/Video/Webcam to Kodi/Beamer/other PCs
   2 Gregor and I had a closer look at the following projects:
   3
   4 # Laplace
   5 Simple WebRTC implementation in GO: https://github.com/adamyordan/laplace.git
   6 Simple to compile and just works.
   7 WebRTC actually uses SRTP (Secure Real-time Transport Protocol).
   8
   9 IPv6+IPv4, and no TLS:
  10
  11     ./laplace -addr "[::]:8080" -tls=false
  12
  13 ## WebRTC signaling
  14
  15 * https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API/Signaling_and_video_calling
  16 * https://www.tutorialspoint.com/webrtc/webrtc_signaling.htm
  17 * https://www.html5rocks.com/en/tutorials/webrtc/infrastructure/
  18 * https://github.com/topics/webrtc-signaling
  19
  20 # gstreamer
  21 * Supports RTP
  22 * Supports webcam capture via v4l (video for linux)
  23 * Supports screen capture via xmanager/ximagesrc
  24 * Interesting URLs:
  25   * WebRTC in one direction only: https://stackoverflow.com/questions/57430215/how-to-use-webrtcbin-create-offer-only-receive-video
  26   * gstreamer to VLC via RTP: https://stackoverflow.com/questions/13154983/gstreamer-rtp-stream-to-vlc
  27   * https://github.com/intel/gstreamer-media-SDK/issues/138
  28   * https://developer.ridgerun.com/wiki/index.php?title=GstWebRTC_Pipelines
  29   * https://developer.ridgerun.com/wiki/index.php?title=GstWebRTC_-_H264-Opus_Examples#Receive_Pipeline
  30
  31 ## WebRTC
  32 gstreamer has a WebRTC implementation.
  33 The examples at https://github.com/centricular/gstwebrtc-demos.git work once
  34
  35     export OPENSSL_CONF=''
  36
  37 has been set (see https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/811)
  38
  39 There's a new recvonly example:
  40
  41 * https://github.com/centricular/gstwebrtc-demos/commit/000cfb6cd817ca0f07d761795b352a8b8b9074f8
  42 * `sendonly/webrtc-recvonly-h264.c`
  43
  44 ## gstreamer examples
  45     gst-launch-1.0 -v playbin uri=file:///home/philipp/tmp/GerisGame.mp4
  46
  47     # send
  48     gst-launch-1.0 ximagesrc ! videoconvert ! videoscale ! video/x-raw,width=800,height=600 ! vp8enc ! rtpvp8pay ! udpsink host=239.255.12.42 port=5004
  49     # receive
  50     gst-launch-1.0 udpsrc address=239.255.12.42 port=5004 caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)VP8-DRAFT-IETF-01, payload=(int)96, ssrc=(uint)2990747501, clock-base=(uint)275641083, seqnum-base=(uint)34810" ! rtpvp8depay ! vp8dec ! autovideosink
  51
  52     # send
  53     gst-launch-1.0 -v ximagesrc ! videoconvert ! videoscale ! video/x-raw,format=I420,width=800,height=600,framerate=25/1 ! jpegenc ! rtpjpegpay ! udpsink host=239.255.12.42 port=5004
  54     # receive
  55     gst-launch-1.0 udpsrc address=239.255.12.42 port=5004 ! application/x-rtp,encoding-name=JPEG,payload=26 ! rtpjpegdepay ! jpegdec ! autovideosink
  56
  57
  58 * https://stackoverflow.com/questions/33747500/using-gstreamer-to-capture-screen-and-show-it-in-a-window/33822024#33822024
  59 * http://www.einarsundgren.se/gstreamer-basic-real-time-streaming-tutorial/
  60 * https://gist.github.com/tetkuz/0c038321d05586841897
  61 * https://gist.github.com/esrever10/7d39fe2d4163c5b2d7006495c3c911bb
  62 * https://gist.github.com/nebgnahz/26a60cd28f671a8b7f522e80e75a9aa5
  63 * https://salsa.debian.org/debconf-video-team/ansible/-/blob/master/roles/voctomix/templates/videoteam-stream.j2
  64 * https://github.com/xfxf/video-scripts/blob/master/michael/youtube-live.sh
  65   (and others in the same repo)
  66
  67 ## gstreamer webrtc, another example
  68
  69 * blog post: https://aweirdimagination.net/2020/07/05/gstreamer-webrtc/
  70 * code: https://git.aweirdimagination.net/perelman/minimal-webrtc-gstreamer
  71
  72 ## websockets
  73
  74 connecting to websockets (plus gstreamer and v4l2sink)
  75
  76 https://michael.stapelberg.ch/posts/2020-06-06-iphone-camera-linux-v4l2loopback/
  77
  78         # send
  79         laplace / chromium
  80
  81         # receive
  82         websocat --insecure "wss://localhost:8080/ws_connect?id=cranky_kind_chipmunk"
  83
  84 ## more v4l2loopback/v4l2sink
  85
  86         # send
  87         gst-launch-1.0 videotestsrc ! v4l2sink device=/dev/video42
  88
  89         # receive
  90         cvlc v4l2:///dev/video42
  91
  92 ## SRT and gstreamer
  93
  94 * https://github.com/Haivision/srt/blob/master/docs/gstreamer.md
  95 * https://srtlab.github.io/srt-cookbook/
  96
  97 ## gstreamer and rtsp
  98
  99 working minimal example: https://github.com/Enne2/PyGObject-GstRtspServer/blob/master/rtsp-server.py
 100
 101 other maybe helpful links (for getting an existing pipeline streamed):
 102
 103 * https://stackoverflow.com/questions/52562499/is-it-possible-to-stream-an-existing-gstreamer-pipeline-through-gstrtspserver
 104 * http://gstreamer-devel.966125.n4.nabble.com/Continuously-streaming-a-video-file-code-review-td4671364.html
 105 * http://gstreamer-devel.966125.n4.nabble.com/RTSP-Server-from-a-manually-created-and-linked-pipeline-td4680305.html
 106 * http://gstreamer-devel.966125.n4.nabble.com/Using-C-API-based-pipelines-in-RTSP-server-without-quot-launch-quot-arg-td4680144.html
 107
 108 # VLC
 109 Note that you have to close VLC and open it again for new streaming as VLC leaves the connections
 110 open (I spent hours figuring out why http based streaming doesn't work - `netstat -4 --ip` is nice
 111 for debugging in that respect).
 112
 113 ## RTP
 114 ### Source
 115 The address has to be the address where the video should be streamed **to**. In the following
 116 examples, a multicast address is used. It is either the IP address of the device receiving the video (unicast address) or a multicast address
 117 like 239.255.12.42 (multicast addresses are between 224.0.0.0 and 239.255.255.255, some of
 118 them being reserved, 239.0.0.0 to 239.255.255.255 are Organization-Local Scope so they are
 119 good candidates, see
 120 https://www.iana.org/assignments/multicast-addresses/multicast-addresses.xhtml).
 121
 122     cvlc -vv --sout '#transcode{vcodec=h264,acodec=mpga,channels=2,vb=800,ab=128}:rtp{mux=ts,dst=239.255.12.42,sdp=sap,name="Geris Game"}' GerisGame.mp4
 123     cvlc -vv --sout '#transcode{vcodec=h264,acodec=mpga,ab=128,channels=2,samplerate=44100,scodec=none}:rtp{dst=239.255.12.42,port=5004,mux=ts,sap,name=Geris Game}' :no-sout-all :sout-keep GerisGame.mp4
 124     cvlc -vv --sout '#transcode{vcodec=h264,acodec=mpga,channels=2}:rtp{mux=ts,dst=239.255.12.42,sdp=sap,name="Videokamera"}' v4l2:///dev/video0 # long latency
 125     cvlc -vv --sout '#transcode{vcodec=h264,acodec=mpga,channels=2}:rtp{mux=ts,dst=239.255.12.42,sdp=sap,name="Desktop"}' screen:// # not convincing
 126
 127 ### Player
 128     cvlc rtp://239.255.12.42
 129
 130
 131 ## RTSP
 132 For VLC, RTSP means RTSP combined with RTP.
 133
 134 ### Source
 135 We assume that the video source computer has the external IP 192.168.1.54.
 136 Instead of rtsp://192.168.1.54:8554/ we could also use rtsp://192.168.1.54:8554/mystream.sdp
 137
 138     cvlc -vv --sout '#transcode{vcodec=h264,acodec=mpga,ab=128,channels=2,samplerate=44100,scodec=none}:rtp{sdp=rtsp://:8554/}' :no-sout-all :sout-keep GerisGame.mp4
 139     cvlc -vv --sout '#rtp{dst=239.255.12.42,port=1234,sdp=rtsp://192.168.1.54:8554/}' :no-sout-all :sout-keep GerisGame.mp4
 140
 141
 142 ### Player
 143 The address needs to be the address of the source. Note the backslash at the end.
 144
 145     vlc rtsp://192.168.1.54:8554/
 146
 147
 148 ## VLC Remarks
 149 * Latency: The following options are said to reduce latency:
 150
 151       --network-caching=150 --clock-jitter=0 --clock-synchro=0
 152
 153 * Duplicating a stream is nice for debugging:
 154
 155       cvlc -vv --sout '#duplicate{dst=display,dst="transcode{vcodec=h264,acodec=mpga,channels=2,vb=800,ab=128}:rtp{mux=ts,dst=239.255.12.42,sdp=sap,name=Geris Game}"}' GerisGame.mp4
 156       cvlc rtp://239.255.12.42
 157
 158
 159 ## HTTP
 160 The address at the source side has to be the external one of the local computer.
 161
 162 ### Source (server)
 163     cvlc -vv --sout '#transcode{vcodec=mp4v,acodec=mpga,vb=800,ab=128}:standard{access=http,mux=ogg,dst=0.0.0.0:8080}' GerisGame.mp4
 164
 165 ### Player (client)
 166
 167     vlc http://192.168.1.54:8080
 168
 169
 170 # Kodi
 171 The player built into code knows at least the following streaming protocols: http, rtsp.
 172
 173 https://kodi.wiki/index.php?title=Internet_video_and_audio_streams