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simplewebrtcbrackets
Advanced tools
Want to see it in action? Check out the demo: https://simplewebrtc.com/demo.html
Want to run it locally? npm install && npm run test-page
<!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 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 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
The npm package simplewebrtcbrackets receives a total of 1 weekly downloads. As such, simplewebrtcbrackets popularity was classified as not popular.
We found that simplewebrtcbrackets 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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