# Deprecated 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 This repository will remain as-is but is no longer actively maintained.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 [![Gitter](https://badges.gitter.im/Join%20Chat.svg)](https://gitter.im/HenrikJoreteg/SimpleWebRTC?utm_source=badge&utm_medium=badge&utm_campaign=pr-badge&utm_content=badge) Want to see it in action? Check out the demo: https://simplewebrtc.com/demo.html Want to run it locally? 1. Install all dependencies and run the test page ```bash npm install && npm run test-page ``` 2. open your browser to https://0.0.0.0:8443/test/ ## It's so easy: ### 1. Some basic html ```html

``` ### Installing through NPM ```sh npm install --save simplewebrtc # for yarn users yarn add simplewebrtc ``` After that simply import simplewebrtc into your project ```js import SimpleWebRTC from 'simplewebrtc'; ``` ### 2. Create our WebRTC object ```js 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 ```js // 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](https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia) with official spec [at w3c](http://w3c.github.io/mediacapture-main/#dom-mediadevices-getusermedia). - `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 ```javascript { 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: ```javascript { "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`](https://github.com/HenrikJoreteg/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: ```javascript 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](https://github.com/andyet/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](http://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

	Stars	Issues	Version	Updated	Created
easyrtc			1.1.0	8 years ago	12 years ago
rtcpeerconnection	199	8	8.4.0	6 years ago	11 years ago
simplewebrtc			3.0.2	6 years ago	11 years ago
socket.io-client	10,587	37	4.7.5	3 months ago	13 years ago
websocket			1.0.35	a month ago	13 years ago

easyrtc vs rtcpeerconnection vs simplewebrtc vs socket.io-client vs websocket

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