The open-source version of SimpleWebRTC has been deprecated. This repository will remain as-is but is no longer actively maintained. Read more about the "new" SimpleWebRTC (which is an entirely different thing) on https://simplewebrtc.com

SimpleWebRTC - World's easiest WebRTC lib

Want to see it in action? Check out the demo: https://simplewebrtc.com/demo.html

Want to run it locally?

Install all dependencies and run the test page

npm install && npm run test-page

open your browser to https://0.0.0.0:8443/test/

It's so easy:

1. Some basic html

<!DOCTYPE html>
<html>
    <head>
        <script src="https://simplewebrtc.com/latest-v2.js"></script>
        <style>
            #remoteVideos video {
                height: 150px;
            }
            #localVideo {
                height: 150px;
            }
        </style>
    </head>
    <body>
        <video id="localVideo"></video>
        <div id="remoteVideos"></div>
    </body>
</html>

Installing through NPM

npm install --save simplewebrtc

# for yarn users
yarn add simplewebrtc

After that simply import simplewebrtc into your project

import SimpleWebRTC from 'simplewebrtc';

2. Create our WebRTC object

var webrtc = new SimpleWebRTC({
    // the id/element dom element that will hold "our" video
    localVideoEl: 'localVideo',
    // the id/element dom element that will hold remote videos
    remoteVideosEl: 'remoteVideos',
    // immediately ask for camera access
    autoRequestMedia: true
});

3. Tell it to join a room when ready

// we have to wait until it's ready
webrtc.on('readyToCall', function () {
    // you can name it anything
    webrtc.joinRoom('your awesome room name');
});

Available options

peerConnectionConfig - Set this to specify your own STUN and TURN servers. By default, SimpleWebRTC uses Google's public STUN server (stun.l.google.com:19302), which is intended for public use according to: https://twitter.com/HenrikJoreteg/status/354105684591251456

Note that you will most likely also need to run your own TURN servers. See http://www.html5rocks.com/en/tutorials/webrtc/infrastructure/ for a basic tutorial.

Filetransfer

Sending files between individual participants is supported. See http://simplewebrtc.com/filetransfer.html for a demo.

Note that this is not file sharing between a group which requires a completely different approach.

It's not always that simple...

Sometimes you need to do more advanced stuff. See http://simplewebrtc.com/notsosimple.html for some examples.

Got questions?

Join the Gitter channel:

https://gitter.im/HenrikJoreteg/SimpleWebRTC

API

Constructor

new SimpleWebRTC(options)

object options - options object provided to constructor consisting of:
- string url - required url for signaling server. Defaults to signaling server URL which can be used for development. You must use your own signaling server for production.
- object socketio - optional object to be passed as options to the signaling server connection.
- Connection connection - optional connection object for signaling. See Connection below. Defaults to a new SocketIoConnection
- bool debug - optional flag to set the instance to debug mode
- [string|DomElement] localVideoEl - ID or Element to contain the local video element
- [string|DomElement] remoteVideosEl - ID or Element to contain the remote video elements
- bool autoRequestMedia - optional(=false) option to automatically request user media. Use true to request automatically, or false to request media later with startLocalVideo
- bool enableDataChannels optional(=true) option to enable/disable data channels (used for volume levels or direct messaging)
- bool autoRemoveVideos - optional(=true) option to automatically remove video elements when streams are stopped.
- bool adjustPeerVolume - optional(=false) option to reduce peer volume when the local participant is speaking
- number peerVolumeWhenSpeaking - optional(=.0.25) value used in conjunction with adjustPeerVolume. Uses values between 0 and 1.
- object media - media options to be passed to getUserMedia. Defaults to { video: true, audio: true }. Valid configurations described on MDN with official spec at w3c.
- object receiveMedia - optional RTCPeerConnection options. Defaults to { offerToReceiveAudio: 1, offerToReceiveVideo: 1 }.
- object localVideo - optional options for attaching the local video stream to the page. Defaults to
```
{
    autoplay: true, // automatically play the video stream on the page
    mirror: true, // flip the local video to mirror mode (for UX)
    muted: true // mute local video stream to prevent echo
}
```
- object logger - optional alternate logger for the instance; any object that implements log, warn, and error methods.
- object peerConnectionConfig - optional options to specify own your own STUN/TURN servers. By default these options are overridden when the signaling server specifies the STUN/TURN server configuration. Example on how to specify the peerConnectionConfig:
```
{
  "iceServers": [{
          "url": "stun3.l.google.com:19302"
      },
      {
          "url": "turn:your.turn.servers.here",
          "username": "your.turn.server.username",
          "credential": "your.turn.server.password"
      }
  ],
  iceTransports: 'relay'
}
```

Fields

capabilities - the webrtcSupport object that describes browser capabilities, for convenience

config - the configuration options extended from options passed to the constructor

connection - the socket (or alternate) signaling connection

webrtc - the underlying WebRTC session manager

Events

To set up event listeners, use the SimpleWebRTC instance created with the constructor. Example:

var webrtc = new SimpleWebRTC(options);
webrtc.on('connectionReady', function (sessionId) {
    // ...
})

'connectionReady', sessionId - emitted when the signaling connection emits the connect event, with the unique id for the session.

'createdPeer', peer - emitted three times:

when joining a room with existing peers, once for each peer
when a new peer joins a joined room
when sharing screen, once for each peer
peer - the object representing the peer and underlying peer connection

'channelMessage', peer, channelLabel, {messageType, payload} - emitted when a broadcast message to all peers is received via dataChannel by using the method sendDirectlyToAll().

'stunservers', [...args] - emitted when the signaling connection emits the same event

'turnservers', [...args] - emitted when the signaling connection emits the same event

'localScreenAdded', el - emitted after triggering the start of screen sharing

el the element that contains the local screen stream

'joinedRoom', roomName - emitted after successfully joining a room with the name roomName

'leftRoom', roomName - emitted after successfully leaving the current room, ending all peers, and stopping the local screen stream

'videoAdded', videoEl, peer - emitted when a peer stream is added

videoEl - the video element associated with the stream that was added
peer - the peer associated with the stream that was added

'videoRemoved', videoEl, peer - emitted when a peer stream is removed

videoEl - the video element associated with the stream that was removed
peer - the peer associated with the stream that was removed

Methods

createRoom(name, callback) - emits the create event on the connection with name and (if provided) invokes callback on response

joinRoom(name, callback) - joins the conference in room name. Callback is invoked with callback(err, roomDescription) where roomDescription is yielded by the connection on the join event. See signalmaster for more details.

startLocalVideo() - starts the local media with the media options provided in the config passed to the constructor

testReadiness() - tests that the connection is ready and that (if media is enabled) streams have started

mute() - mutes the local audio stream for all peers (pauses sending audio)

unmute() - unmutes local audio stream for all peers (resumes sending audio)

pauseVideo() - pauses sending video to peers

resumeVideo() - resumes sending video to all peers

pause() - pauses sending audio and video to all peers

resume() - resumes sending audio and video to all peers

sendToAll(messageType, payload) - broadcasts a message to all peers in the room via the signaling channel (websocket)

string messageType - the key for the type of message being sent
object payload - an arbitrary value or object to send to peers

sendDirectlyToAll(channelLabel, messageType, payload) - broadcasts a message to all peers in the room via a dataChannel

string channelLabel - the label for the dataChannel to send on
string messageType - the key for the type of message being sent
object payload - an arbitrary value or object to send to peers

getPeers(sessionId, type) - returns all peers by sessionId and/or type

shareScreen(callback) - initiates screen capture request to browser, then adds the stream to the conference

getLocalScreen() - returns the local screen stream

stopScreenShare() - stops the screen share stream and removes it from the room

stopLocalVideo() - stops all local media streams

setVolumeForAll(volume) - used to set the volume level for all peers

volume - the volume level, between 0 and 1

leaveRoom() - leaves the currently joined room and stops local screen share

disconnect() - calls disconnect on the signaling connection and deletes it

handlePeerStreamAdded(peer) - used internally to attach media stream to the DOM and perform other setup

handlePeerStreamRemoved(peer) - used internally to remove the video container from the DOM and emit videoRemoved

getDomId(peer) - used internally to get the DOM id associated with a peer

getEl(idOrEl) - helper used internally to get an element where idOrEl is either an element, or an id of an element

getLocalVideoContainer() - used internally to get the container that will hold the local video element

getRemoteVideoContainer() - used internally to get the container that holds the remote video elements

Connection

By default, SimpleWebRTC uses a socket.io connection to communicate with the signaling server. However, you can provide an alternate connection object to use. All that your alternate connection need provide are four methods:

on(ev, fn) - A method to invoke fn when event ev is triggered
emit() - A method to send/emit arbitrary arguments on the connection
getSessionId() - A method to get a unique session Id for the connection
disconnect() - A method to disconnect the connection

FAQs

What is simplewebrtc?

Is simplewebrtc popular?

Is simplewebrtc well maintained?

Last updated on 16 Jul 2018

Did you know?

Socket for GitHub automatically highlights issues in each pull request and monitors the health of all your open source dependencies. Discover the contents of your packages and block harmful activity before you install or update your dependencies.

Install

simplewebrtc

Deprecated