react-native-webrtc

A WebRTC module for React Native.

BREAKING FOR RN 40:

master branch needs RN >= 40 for now. if your RN version is under 40, use branch rn-less-40 (npm version 0.54.7)

see #190 for detials

Support

• Currently support for iOS and Android.
• Support video and audio communication.
• Supports data channels.
• You can use it to build an iOS/Android app that can communicate with web browser.

WebRTC Revision

Since 0.53, we use same branch version number like in webrtc native. please see wiki page about revision history

format:

${branch_name} stable (${branched_from_revision})(+${Cherry-Picks-Num}-${Last-Cherry-Picks-Revision})

• the webrtc revision in brackets is extracting frrom Cr-Branched-From instead Cr-Commit-Position
• the number follows with + is the additional amount of cherry-picks since Branched-From revision.

note:

the order of commit revision is nothing to do with the order of cherry-picks, for example, the earlier committed cherry-pick-#2 may have higher revision than cherry-pick-#3 and vice versa.

react-native-webrtc	WebRTC Version	arch(ios)	arch(android)	npm published	note	additional picks
0.54.7	M54 (13869) (+6-14091)	x86_64 i386 armv7 arm64	armeabi-v7a x86	✔️	RN < 40
1.57.1	M57 (16123) (+7-16178)	x86_64 i386 armv7 arm64	armeabi-v7a x86	✔️		* 16805 * 16462
1.58.0	M58 commit (16937) (+19-17785)	x86_64 i386 armv7 arm64	armeabi-v7a x86	⚠️	pre-release	* 17925 * 18140
1.58.1	M58 commit (16937) (+21-18206)	x86_64 i386 armv7 arm64	armeabi-v7a x86	✔️		* 17925 * 18140 * 18277
master	M58 commit (16937) (+21-18206)	x86_64 i386 armv7 arm64	armeabi-v7a x86	⚠️	test me plz	* 17065 * 17925 * 18140 * 18277

Installation

react-native-webrtc:

• iOS
• Android

note: 0.10.0~0.12.0 required git-lfs, see: git-lfs-installation

Usage

Now, you can use WebRTC like in browser. In your index.ios.js/index.android.js, you can require WebRTC to import RTCPeerConnection, RTCSessionDescription, etc.

var WebRTC = require('react-native-webrtc');
var {
  RTCPeerConnection,
  RTCIceCandidate,
  RTCSessionDescription,
  RTCView,
  MediaStream,
  MediaStreamTrack,
  getUserMedia,
} = WebRTC;

Anything about using RTCPeerConnection, RTCSessionDescription and RTCIceCandidate is like browser.
Support most WebRTC APIs, please see the Document.

var configuration = {"iceServers": [{"url": "stun:stun.l.google.com:19302"}]};
var pc = new RTCPeerConnection(configuration);

let isFront = true;
MediaStreamTrack.getSources(sourceInfos => {
  console.log(sourceInfos);
  let videoSourceId;
  for (const i = 0; i < sourceInfos.length; i++) {
    const sourceInfo = sourceInfos[i];
    if(sourceInfo.kind == "video" && sourceInfo.facing == (isFront ? "front" : "back")) {
      videoSourceId = sourceInfo.id;
    }
  }
  getUserMedia({
    audio: true,
    video: {
      mandatory: {
        minWidth: 500, // Provide your own width, height and frame rate here
        minHeight: 300,
        minFrameRate: 30
      },
      facingMode: (isFront ? "user" : "environment"),
      optional: (videoSourceId ? [{sourceId: videoSourceId}] : [])
    }
  }, function (stream) {
    console.log('dddd', stream);
    callback(stream);
  }, logError);
});

pc.createOffer(function(desc) {
  pc.setLocalDescription(desc, function () {
    // Send pc.localDescription to peer
  }, function(e) {});
}, function(e) {});

pc.onicecandidate = function (event) {
  // send event.candidate to peer
};

// also support setRemoteDescription, createAnswer, addIceCandidate, onnegotiationneeded, oniceconnectionstatechange, onsignalingstatechange, onaddstream

However, render video stream should be used by React way.

Rendering RTCView.

var container;
var RCTWebRTCDemo = React.createClass({
  getInitialState: function() {
    return {videoURL: null};
  },
  componentDidMount: function() {
    container = this;
  },
  render: function() {
    return (
      <View>
        <RTCView streamURL={this.state.videoURL}/>
      </View>
    );
  }
});

And set stream to RTCView

container.setState({videoURL: stream.toURL()});

Custom APIs

MediaStreamTrack.prototype._switchCamera()

This function allows to switch the front / back cameras in a video track on the fly, without the need for adding / removing tracks or renegotiating.

Demos

Official Demo

author: @oney

The demo project is https://github.com/oney/RCTWebRTCDemo
And you will need a signaling server. I have written a signaling server https://react-native-webrtc.herokuapp.com/ (the repository is https://github.com/oney/react-native-webrtc-server).
You can open this website in browser, and then set it as signaling server in the app, and run the app. After you enter the same room ID, the video stream will be connected.

Demo by Folks

author: @thoqbk

• Signaling server and web app: https://rewebrtc.herokuapp.com/ (the repository is https://github.com/thoqbk/rewebrtc-server)
• React native app repository: https://github.com/thoqbk/rewebrtc

Native control

Use react-native-incall-manager to keep screen on, mute microphone, etc.

Sponsorship

This repository doesn't have a plan to get sponsorship.(This can be discussed afterwards by collaborators). If you would like to pay bounty to fix some bugs or get some features, be free to open a issue that adds [BOUNTY] category in title. Add other bounty website link like this will be better.

install this fork form git

npm install git+https://github.com/jjunk1989/react-native-webrtc.git --save

build webrtc

build webrtc