rainforestapp/webrtc-streamer

Name: webrtc-streamer

Owner: Rainforest QA

Description: WebRTC streamer for V4L2 capture devices and RTSP sources

Created: 2018-03-05 21:30:27.0

Updated: 2018-05-18 14:12:44.0

Pushed: 2018-04-26 10:59:03.0

Homepage: https://rtsp2webrtc.herokuapp.com

Size: 1852

Language: JavaScript

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README

WebRTC-streamer

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

Dependencies :

It is based on :

• WebRTC Native Code Package
• civetweb HTTP server
• h264bitstream

Build

Build WebRTC with H264 support

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 use_ozone=true rtc_include_pulse_audio=false'
ninja -C out/Release
popd

Build WebRTC Streamer

make WEBRTCROOT=<path to WebRTC> WEBRTCBUILD=<Release or Debug> SYSROOT=<path to WebRTC>/src/build/linux/debian_stretch_amd64-sysroot

where WEBRTCROOT and WEBRTCBUILD indicate how to point to WebRTC :

• $WEBRTCROOT/src should contains source (default is ../webrtc)
• $WEBRTCROOT/src/out/$WEBRTCBUILD should contains libraries (default is Release)
• $SYSROOT should point to sysroot used to build WebRTC (default is /)

Usage

./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
        -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)
        [url]              : url to register in the source list
        -v[v[v]]           : verbosity
        -V                 : print version

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,8443'.

[Live Demo]

You can access to the WebRTC stream coming from an RTSP url using webrtcstreamer.html page with the VNC url as argument, something like:

http://localhost:8000/webrtcstreamer.html?vnc://217.17.220.110

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 :

• get the help using :

  docker run -p 8000:8000 -it mpromonet/webrtc-streamer -h
  

• run the container registering a VNC url using :

  docker run -p 8000:8000 -it mpromonet/webrtc-streamer -u vnc://pi2.local:8554