Deprecated

The open-source version of SimpleWebRTC has been deprecated. This repository will remain as-is but is no longer actively maintained. You can find the old website in the gh-pages branch.
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.

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

名稱與所有者	simplewebrtc/SimpleWebRTC
主編程語言	JavaScript
編程語言	JavaScript (語言數: 3)
平台	Cross-platform, Linux, Mac, Windows
許可證	Other
發布數	31
最新版本名稱	v3.1.0 (發布於 2019-02-27 13:40:27)
第一版名稱	v1.8.3 (發布於 2014-05-29 14:13:54)
創建於	2013-01-23 07:28:01
推送於	2022-08-14 00:15:59
最后一次提交	2019-02-27 13:40:27
星數	4.6k
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