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This is a try to stream video sources through WebRTC using simple mechanism.
It embeds a HTTP server that implements API and serve a simple HTML page that use them through AJAX.
The WebRTC signaling is implemented throught HTTP requests:
/api/call : send offer and get answer
/api/hangup : close a call
/api/addIceCandidate : add a candidate
/api/getIceCandidate : get the list of candidates
The list of HTTP API is available using /api/help.
Nowdays there is 3 builds on CircleCI :
The webrtc stream name could be :
It is based on :
mkdir ../webrtc
pushd ../webrtc
fetch webrtc
gn gen out/Release --args='is_debug=false use_custom_libcxx=false rtc_use_h264=true ffmpeg_branding="Chrome" rtc_include_tests=false rtc_include_pulse_audio=false use_sysroot=false is_clang=false treat_warnings_as_errors=false'
ninja -C out/Release
popd
make live555
make WEBRTCROOT=<path to WebRTC> WEBRTCBUILD=<Release or Debug>
where WEBRTCROOT and WEBRTCBUILD indicate how to point to WebRTC :
./webrtc-streamer [-H http port] [-S[embeded stun address]] -[v[v]] [url1]...[urln]
./webrtc-streamer [-H http port] [-s[external stun address]] -[v[v]] [url1]...[urln]
./webrtc-streamer -V
-v[v[v]] : verbosity
-V : print version
-H [hostname:]port : HTTP server binding (default 0.0.0.0:8000)
-S[stun_address] : start embeded STUN server bind to address (default 0.0.0.0:3478)
-s[stun_address] : use an external STUN server (default stun.l.google.com:19302)
-t[username:password@]turn_address : use an external TURN relay server (default disabled)
-a[audio layer] : spefify audio capture layer to use (default:3)
-n name -u url : register a name for an url
[url] : url to register in the source list
Arguments of '-H' is forwarded to option listening_ports of civetweb, then it is possible to use the civetweb syntax like '-H8000,9000' or '-H8080r,8443s'.
webrtc-streamer rtsp://217.17.220.110/axis-media/media.amp \
rtsp://85.255.175.241/h264 \
rtsp://85.255.175.244/h264 \
rtsp://184.72.239.149/vod/mp4:BigBuckBunny_175k.mov
You can access to the WebRTC stream coming from an RTSP url using webrtcstreamer.html page with the RTSP url as argument, something like:
https://rtsp2webrtc.herokuapp.com/webrtcstreamer.html?rtsp://217.17.220.110/axis-media/media.amp
Instead of using the internal HTTP server, it is easy to display a WebRTC stream in a HTML page served by another HTTP server. The URL of the webrtc-streamer to use should be given creating the WebRtcStreamer instance :
var webRtcServer = new WebRtcStreamer(<video tag>, <webrtc-streamer url>);
A short sample HTML page using webrtc-streamer running locally on port 8000 :
<html>
<head>
<script src="request.min.js" ></script>
<script src="webrtcstreamer.js" ></script>
<script>
var webRtcServer = new WebRtcStreamer("video",location.protocol+"//"+window.location.hostname+":8000");
window.onload = function() { webRtcServer.connect("rtsp://pi2.local:8554/unicast") }
window.onbeforeunload = function() { webRtcServer.disconnect() }
</script>
</head>
<body>
<video id="video" />
</body>
</html>
A simple way to publish WebRTC stream to a Janus Gateway Video Room is to use the JanusVideoRoom interface
var janus = new JanusVideoRoom(<janus url>, <webrtc-streamer url>)
A short sample to publish WebRTC streams to Janus Video Room could be :
<html>
<head>
<script src="request.min.js" ></script>
<script src="janusvideoroom.js" ></script>
<script>
var janus = new JanusVideoRoom("https://janus.conf.meetecho.com/janus", null);
janus.join(1234, "rtsp://pi2.local:8554/unicast","pi2");
janus.join(1234, "rtsp://217.17.220.110/axis-media/media.amp","media");
</script>
</head>
</html>
This way the communication between Janus API and WebRTC Streamer API is implemented in Javascript running in browser.
The same logic could be implemented in NodeJS using the same JS API :
global.request = require('then-request');
var JanusVideoRoom = require('./html/janusvideoroom.js');
var janus = new JanusVideoRoom("http://192.168.0.15:8088/janus", "http://192.168.0.15:8000")
janus.join(1234,"mmal service 16.1","video")
A simple way to publish WebRTC stream to a Jitsi Video Room is to use the XMPPVideoRoom interface
var xmpp = new XMPPVideoRoom(<xmpp server url>, <webrtc-streamer url>)
A short sample to publish WebRTC streams to a Jitsi Video Room could be :
<html>
<head>
<script src="libs/strophe.min.js" ></script>
<script src="libs/strophe.muc.min.js" ></script>
<script src="libs/strophe.disco.min.js" ></script>
<script src="libs/strophe.caps.min.js" ></script>
<script src="libs/strophe.jingle.sdp.js"></script>
<script src="libs/jquery-1.12.4.min.js"></script>
<script src="libs/request.min.js" ></script>
<script src="request.min.js" ></script>
<script src="xmppvideoroom.js" ></script>
<script>
var xmpp = new XMPPVideoRoom("meet.jit.si", null);
xmpp.join("testroom", "rtsp://184.72.239.149/vod/mp4:BigBuckBunny_175k.mov","Bunny");
</script>
</head>
</html>
You can start the application using the docker image :
docker run -p 8000:8000 -it mpromonet/webrtc-streamer
You can expose V4L2 devices from your host using :
docker run --device=/dev/video0 -p 8000:8000 -it mpromonet/webrtc-streamer
The container entry point is the webrtc-streamer application, then you can :
get the help using :
docker run -p 8000:8000 -it mpromonet/webrtc-streamer -h
run the container registering a RTSP url using :
docker run -p 8000:8000 -it mpromonet/webrtc-streamer -n raspicam -u rtsp://pi2.local:8554/unicast
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