add gstreamer+rtsp links
[toast/stream2beamer.git] / 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