WebRTC-Streamer
Experimentation to stream WebRTC media sources like capture devices, screen capture, mkv files and RMTP/RTSP sources using simple signaling mechanism (see api). It is also compatible with WHEP interface.
Artefacts
- packages are available from https://github.com/mpromonet/webrtc-streamer/releases/latest
- container image are available from https://hub.docker.com/r/mpromonet/webrtc-streamer
Usage
./webrtc-streamer [-H http port] [-S[embeded stun address]] -[v[v]] [urls...]
./webrtc-streamer [-H http port] [-s[external stun address]] -[v[v]] [urls...]
./webrtc-streamer -V
-v[v[v]] : verbosity
-V : print version
-C config.json : load urls from JSON config file
-n name -u videourl -U audiourl : register a name for a video url and an audio url
[url] : url to register in the source list
-H [hostname:]port : HTTP server binding (default 0.0.0.0:8000)
-w webroot : path to get files
-c sslkeycert : path to private key and certificate for HTTPS
-N nbthreads : number of threads for HTTP server
-A passwd : password file for HTTP server access
-D authDomain : authentication domain for HTTP server access (default:mydomain.com)
-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 , -:means no STUN)
-T[username:password@]turn_address : start embeded TURN server (default:disabled)
-t[username:password@]turn_address : use an external TURN relay server (default:disabled)
-R [Udp port range min:max] : Set the webrtc udp port range (default 0:65535)
-W webrtc_trials_fields : Set the webrtc trials fields (default:WebRTC-FrameDropper/Disabled/)
-a[audio layer] : spefify audio capture layer to use (default:0)
-q[filter] : spefify publish filter (default:.*)
-o : use null codec (keep frame encoded)
Arguments of '-H' are forwarded to option
listening_ports
of civetweb, allowing use of the civetweb syntax like -H8000,9000 or
-H8080r,8443s.
Using -o allows storing compressed frame data from the backend stream using
webrtc::VideoFrameBuffer::Type::kNative. This hacks the stucture
webrtc::VideoFrameBuffer storing data in a override of the i420 buffer. This
allows forwarding H264 frames from V4L2 device or RTSP stream to WebRTC stream.
It uses less CPU, but has less features (resize, codec, and bandwidth are
disabled).
Options for the WebRTC stream name:
- an alias defined using
-nargument then the corresponding-uargument will
be used to create the capturer - an "rtsp://" url that will be opened using an RTSP capturer based on live555
- an "file://" url that will be opened using an MKV capturer based on live555
- an "rmtp://" url that will be opened using an RMTP capturer based on librmtp
- an "screen://" url that will be opened by
webrtc::DesktopCapturer::CreateScreenCapturer - an "window://" url that will be opened by
webrtc::DesktopCapturer::CreateWindowCapturer - an "v4l2://" url that will capture
H264 frames and store
it using webrtc::VideoFrameBuffer::Type::kNative type (not supported on
Windows) - an "videocap://" url video capture device name
- an "audiocap://" url audio capture device name
Examples
./webrtc-streamer -C config.json
We can access to the WebRTC stream using
webrtcstreamer.html.
For instance:
- webrtcstreamer.html?rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov
- webrtcstreamer.html?Bunny
An example displaying grid of WebRTC Streams is available using option
layout=<lines>x<columns>
Using docker image
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:
-
view all commands
docker run -p 8000:8000 -it mpromonet/webrtc-streamer --help -
run the container registering a RTSP url:
docker run -p 8000:8000 -it mpromonet/webrtc-streamer -n raspicam -u rtsp://pi2.local:8554/unicast -
run the container giving config.json file:
docker run -p 8000:8000 -v $PWD/config.json:/app/config.json mpromonet/webrtc-streamer
Using embedded STUN/TURN server behind a NAT
It is possible to start an embeded STUN
and TURN
server and publish its URL:
./webrtc-streamer -S0.0.0.0:3478 -s$(curl -s ifconfig.me):3478
./webrtc-streamer -s- -T0.0.0.0:3478 -tturn:turn@$(curl -s ifconfig.me):3478
./webrtc-streamer -S0.0.0.0:3478 -s$(curl -s ifconfig.me):3478 -T0.0.0.0:3479 -tturn:turn@$(curl -s ifconfig.me):3479
The command curl -s ifconfig.me is getting the public IP, it could also given
as a static parameter.
In order to configure the NAT rules using the upnp feature of the router, it is
possible to use
upnpc like
this:
upnpc -r 8000 tcp 3478 tcp 3478 udp
Adapting with the HTTP port, STUN port, TURN port.
HTML Embedding
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="libs/adapter.min.js" ></script>
<script src="webrtcstreamer.js" ></script>
<script>
var webRtcServer = null;
window.onload = function() {
webRtcServer = new WebRtcStreamer("video",location.protocol+"//"+location.hostname+":8000");
webRtcServer.connect("rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov");
}
window.onbeforeunload = function() { webRtcServer.disconnect(); }
</script>
</head>
<body>
<video id="video" />
</body>
</html>
Using WebComponents
WebRTC-streamer provides its own
Web Components as an alternative way to
display a WebRTC stream in an HTML page. For example:
<html>
<head>
<script type="module" src="webrtc-streamer-element.js"></script>
</head>
<body>
<webrtc-streamer url="rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov"></webrtc-streamer>
</body>
</html>
Using the webcomponent with a stream selector:
Using the webcomponent over google map:
Using WHEP
It allow to stream using draft standard WHEP
WebRTC player can display WebRTC stream from webrtc-streamer.
A minimal example:
<html>
<head>
<script src="https://unpkg.com/@eyevinn/whep-video-component@latest/dist/whep-video.component.js"></script>
</head>
<body>
<whep-video id="video" muted autoplay></whep-video>
<script>
video.setAttribute('src', `${location.origin}/api/whep?url=Asahi&options=rtptransport%3dtcp%26timeout%3d60`);
</script>
</body>
</html>
Object detection using tensorflow.js
Connect to Janus Gateway Video Room
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="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://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov","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, "videocap://0", "video");
Connect to Jitsi
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.jingle.sdp.js"></script>
<script src="libs/jquery-3.5.1.min.js"></script>
<script src="xmppvideoroom.js" ></script>
<script>
var xmpp = new XMPPVideoRoom("meet.jit.si", null);
xmpp.join("testroom", "rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov","Bunny");
</script>
</head>
</html>
Dependencies
This package depends on the following packages:
- WebRTC Native Code Package for WebRTC (see license https://webrtc.github.io/webrtc-org/license)
- civetweb HTTP server for HTTP server (see license https://github.com/civetweb/civetweb/blob/master/LICENSE.md)
- live555 for RTSP/MKV source (see license http://www.live555.com/liveMedia/faq.html#copyright-and-license)
Build
The following steps are required to build the project, and will install the
dependencies above:
-
Install the Chromium depot tools
pushd .. git clone https://chromium.googlesource.com/chromium/tools/depot_tools.git export PATH=$PATH:`realpath depot_tools` popd -
Download WebRTC
mkdir ../webrtc pushd ../webrtc fetch --no-history webrtc popd -
Build WebRTC Streamer
cmake . && make
It is possible to specify cmake parameters WEBRTCROOT &
WEBRTCDESKTOPCAPTURE:
$WEBRTCROOT/srcshould contains source (default is $(pwd)/../webrtc)WEBRTCDESKTOPCAPTUREenabling desktop capture if available (default is ON)
Pipelines
There is pipelines on CircleCI,
CirrusCI, or
GitHub CI, for the
following architectures:
- x86_64 on Ubuntu
- armv7 crosscompiled (this build is running on Raspberry Pi2 and NanoPi NEO)
- armv6+vfp crosscompiled (this build is running on Raspberry PiB and should run
on a Raspberry Zero) - arm64 crosscompiled
- Windows x64 build with clang
- MacOS