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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 0 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.
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.
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