simplewebrtc
Advanced tools
Want to see it in action? Check out the demo: https://simplewebrtc.com/demo.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>
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
});
// we have to wait until it's ready
webrtc.on('readyToCall', function () {
// you can name it anything
webrtc.joinRoom('your awesome room name');
});
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.
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.
Sometimes you need to do more advanced stuff. See http://simplewebrtc.com/notsosimple.html for some examples.
Join the SimpleWebRTC discussion list:
http://lists.andyet.com/mailman/listinfo/simplewebrtc
or the Gitter channel:
https://gitter.im/HenrikJoreteg/SimpleWebRTC
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 SocketIoConnectionbool 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 elementsbool autoRequestMedia - optional(=false) option to automatically request
user media. Use true to request automatically, or false to request media
later with startLocalVideobool 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 speakingnumber 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.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
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
'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'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 addedpeer - 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 removedpeer - the peer associated with the stream that was removedcreateRoom(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 sentobject payload - an arbitrary value or object to send to peerssendDirectlyToAll(channelLabel, messageType, payload) - broadcasts a message
to all peers in the room via a dataChannel
string channelLabel - the label for the dataChannel to send onstring messageType - the key for the type of message being sentobject payload - an arbitrary value or object to send to peersgetPeers(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 1leaveRoom() - 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
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 triggeredemit() - A method to send/emit arbitrary arguments on the connectiongetSessionId() - A method to get a unique session Id for the connectiondisconnect() - A method to disconnect the connectionFAQs
World's easiest webrtc
We found that simplewebrtc demonstrated a not healthy version release cadence and project activity because the last version was released a year ago. It has 6 open source maintainers collaborating on the project.
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