- concise, node.js style API for WebRTC
- works in node and the browser!
- supports video/voice streams
- supports data channel
- text and binary data
- node.js duplex stream interface
- supports advanced options like:
- enable/disable trickle ICE candidates
- manually set config options
- transceivers and renegotiation
This package is used by WebTorrent and many others.
- install
- examples
- api
- events
- error codes
- connecting more than 2 peers?
- memory usage
- connection does not work on some networks?
- Who is using
simple-peer
? - license
npm install simple-peer
This package works in the browser with browserify. If
you do not use a bundler, you can use the simplepeer.min.js
standalone script
directly in a <script>
tag. This exports a SimplePeer
constructor on
window
. Wherever you see Peer
in the examples below, substitute that with
SimplePeer
.
Let's create an html page that lets you manually connect two peers:
<html>
<body>
<style>
#outgoing {
width: 600px;
word-wrap: break-word;
white-space: normal;
}
</style>
<form>
<textarea id="incoming"></textarea>
<button type="submit">submit</button>
</form>
<pre id="outgoing"></pre>
<script src="simplepeer.min.js"></script>
<script>
const p = new SimplePeer({
initiator: location.hash === '#1',
trickle: false
})
p.on('error', err => console.log('error', err))
p.on('signal', data => {
console.log('SIGNAL', JSON.stringify(data))
document.querySelector('#outgoing').textContent = JSON.stringify(data)
})
document.querySelector('form').addEventListener('submit', ev => {
ev.preventDefault()
p.signal(JSON.parse(document.querySelector('#incoming').value))
})
p.on('connect', () => {
console.log('CONNECT')
p.send('whatever' + Math.random())
})
p.on('data', data => {
console.log('data: ' + data)
})
</script>
</body>
</html>
Visit index.html#1
from one browser (the initiator) and index.html
from another
browser (the receiver).
An "offer" will be generated by the initiator. Paste this into the receiver's form and hit submit. The receiver generates an "answer". Paste this into the initiator's form and hit submit.
Now you have a direct P2P connection between two browsers!
This example create two peers in the same web page.
In a real-world application, you would never do this. The sender and receiver Peer
instances would exist in separate browsers. A "signaling server" (usually implemented with
websockets) would be used to exchange signaling data between the two browsers until a
peer-to-peer connection is established.
var Peer = require('simple-peer')
var peer1 = new Peer({ initiator: true })
var peer2 = new Peer()
peer1.on('signal', data => {
// when peer1 has signaling data, give it to peer2 somehow
peer2.signal(data)
})
peer2.on('signal', data => {
// when peer2 has signaling data, give it to peer1 somehow
peer1.signal(data)
})
peer1.on('connect', () => {
// wait for 'connect' event before using the data channel
peer1.send('hey peer2, how is it going?')
})
peer2.on('data', data => {
// got a data channel message
console.log('got a message from peer1: ' + data)
})
Video/voice is also super simple! In this example, peer1 sends video to peer2.
var Peer = require('simple-peer')
// get video/voice stream
navigator.mediaDevices.getUserMedia({
video: true,
audio: true
}).then(gotMedia).catch(() => {})
function gotMedia (stream) {
var peer1 = new Peer({ initiator: true, stream: stream })
var peer2 = new Peer()
peer1.on('signal', data => {
peer2.signal(data)
})
peer2.on('signal', data => {
peer1.signal(data)
})
peer2.on('stream', stream => {
// got remote video stream, now let's show it in a video tag
var video = document.querySelector('video')
if ('srcObject' in video) {
video.srcObject = stream
} else {
video.src = window.URL.createObjectURL(stream) // for older browsers
}
video.play()
})
}
For two-way video, simply pass a stream
option into both Peer
constructors. Simple!
Please notice that getUserMedia
only works in pages loaded via https.
It is also possible to establish a data-only connection at first, and later add a video/voice stream, if desired.
var Peer = require('simple-peer') // create peer without waiting for media
var peer1 = new Peer({ initiator: true }) // you don't need streams here
var peer2 = new Peer()
peer1.on('signal', data => {
peer2.signal(data)
})
peer2.on('signal', data => {
peer1.signal(data)
})
peer2.on('stream', stream => {
// got remote video stream, now let's show it in a video tag
var video = document.querySelector('video')
if ('srcObject' in video) {
video.srcObject = stream
} else {
video.src = window.URL.createObjectURL(stream) // for older browsers
}
video.play()
})
function addMedia (stream) {
peer1.addStream(stream) // <- add streams to peer dynamically
}
// then, anytime later...
navigator.mediaDevices.getUserMedia({
video: true,
audio: true
}).then(addMedia).catch(() => {})
To use this library in node, pass in opts.wrtc
as a parameter (see the constructor options):
var Peer = require('simple-peer')
var wrtc = require('wrtc')
var peer1 = new Peer({ initiator: true, wrtc: wrtc })
var peer2 = new Peer({ wrtc: wrtc })
Create a new WebRTC peer connection.
A "data channel" for text/binary communication is always established, because it's cheap and often useful. For video/voice communication, pass the stream
option.
If opts
is specified, then the default options (shown below) will be overridden.
{
initiator: false,
channelConfig: {},
channelName: '<random string>',
config: { iceServers: [{ urls: 'stun:stun.l.google.com:19302' }, { urls: 'stun:global.stun.twilio.com:3478?transport=udp' }] },
offerOptions: {},
answerOptions: {},
sdpTransform: function (sdp) { return sdp },
stream: false,
streams: [],
trickle: true,
allowHalfTrickle: false,
wrtc: {}, // RTCPeerConnection/RTCSessionDescription/RTCIceCandidate
objectMode: false
}
The options do the following:
-
initiator
- set totrue
if this is the initiating peer -
channelConfig
- custom webrtc data channel configuration (used bycreateDataChannel
) -
channelName
- custom webrtc data channel name -
config
- custom webrtc configuration (used byRTCPeerConnection
constructor) -
offerOptions
- custom offer options (used bycreateOffer
method) -
answerOptions
- custom answer options (used bycreateAnswer
method) -
sdpTransform
- function to transform the generated SDP signaling data (for advanced users) -
stream
- if video/voice is desired, pass stream returned fromgetUserMedia
-
streams
- an array of MediaStreams returned fromgetUserMedia
-
trickle
- set tofalse
to disable trickle ICE and get a single 'signal' event (slower) -
wrtc
- custom webrtc implementation, mainly useful in node to specify in the wrtc package. Contains an object with the properties: -
objectMode
- set totrue
to create the stream in Object Mode. In this mode, incoming string data is not automatically converted toBuffer
objects.
Call this method whenever the remote peer emits a peer.on('signal')
event.
The data
will encapsulate a webrtc offer, answer, or ice candidate. These messages help
the peers to eventually establish a direct connection to each other. The contents of these
strings are an implementation detail that can be ignored by the user of this module;
simply pass the data from 'signal' events to the remote peer and call peer.signal(data)
to get connected.
Send text/binary data to the remote peer. data
can be any of several types: String
,
Buffer
(see buffer), ArrayBufferView
(Uint8Array
,
etc.), ArrayBuffer
, or Blob
(in browsers that support it).
Note: If this method is called before the peer.on('connect')
event has fired, then data
will be buffered.
Add a MediaStream
to the connection.
Remove a MediaStream
from the connection.
Add a MediaStreamTrack
to the connection. Must also pass the MediaStream
you want to attach it to.
Remove a MediaStreamTrack
from the connection. Must also pass the MediaStream
that it was attached to.
Replace a MediaStreamTrack
with another track. Must also pass the MediaStream
that the old track was attached to.
Add a RTCRtpTransceiver
to the connection. Can be used to add transceivers before adding tracks. Automatically called as neccesary by addTrack
.
Destroy and cleanup this peer connection.
If the optional err
parameter is passed, then it will be emitted as an 'error'
event on the stream.
Detect native WebRTC support in the javascript environment.
var Peer = require('simple-peer')
if (Peer.WEBRTC_SUPPORT) {
// webrtc support!
} else {
// fallback
}
Peer
objects are instances of stream.Duplex
. They behave very similarly to a
net.Socket
from the node core net
module. The duplex stream reads/writes to the data
channel.
var peer = new Peer(opts)
// ... signaling ...
peer.write(new Buffer('hey'))
peer.on('data', function (chunk) {
console.log('got a chunk', chunk)
})
Peer
objects are instance of EventEmitter
. Take a look at the nodejs events documentation for more information.
Example of removing all registered close-event listeners:
peer.removeAllListeners('close')
Fired when the peer wants to send signaling data to the remote peer.
It is the responsibility of the application developer (that's you!) to get this data to
the other peer. This usually entails using a websocket signaling server. This data is an
Object
, so remember to call JSON.stringify(data)
to serialize it first. Then, simply
call peer.signal(data)
on the remote peer.
(Be sure to listen to this event immediately to avoid missing it. For initiator: true
peers, it fires right away. For initatior: false
peers, it fires when the remote
offer is received.)
Fired when the peer connection and data channel are ready to use.
Received a message from the remote peer (via the data channel).
data
will be either a String
or a Buffer/Uint8Array
(see buffer).
Received a remote video stream, which can be displayed in a video tag:
peer.on('stream', stream => {
var video = document.querySelector('video')
if ('srcObject' in video) {
video.srcObject = stream
} else {
video.src = window.URL.createObjectURL(stream)
}
video.play()
})
Received a remote audio/video track. Streams may contain multiple tracks.
Called when the peer connection has closed.
Fired when a fatal error occurs. Usually, this means bad signaling data was received from the remote peer.
err
is an Error
object.
Errors returned by the error
event have an err.code
property that will indicate the origin of the failure.
Possible error codes:
ERR_WEBRTC_SUPPORT
ERR_CREATE_OFFER
ERR_CREATE_ANSWER
ERR_SET_LOCAL_DESCRIPTION
ERR_SET_REMOTE_DESCRIPTION
ERR_ADD_ICE_CANDIDATE
ERR_ICE_CONNECTION_FAILURE
ERR_SIGNALING
ERR_DATA_CHANNEL
ERR_CONNECTION_FAILURE
The simplest way to do that is to create a full-mesh topology. That means that every peer opens a connection to every other peer. To illustrate:
To broadcast a message, just iterate over all the peers and call peer.send
.
So, say you have 3 peers. Then, when a peer wants to send some data it must send it 2 times, once to each of the other peers. So you're going to want to be a bit careful about the size of the data you send.
Full mesh topologies don't scale well when the number of peers is very large. The total
number of edges in the network will be
where n
is the number of peers.
For clarity, here is the code to connect 3 peers together:
// These are peer1's connections to peer2 and peer3
var peer2 = new Peer({ initiator: true })
var peer3 = new Peer({ initiator: true })
peer2.on('signal', data => {
// send this signaling data to peer2 somehow
})
peer2.on('connect', () => {
peer2.send('hi peer2, this is peer1')
})
peer2.on('data', data => {
console.log('got a message from peer2: ' + data)
})
peer3.on('signal', data => {
// send this signaling data to peer3 somehow
})
peer3.on('connect', () => {
peer3.send('hi peer3, this is peer1')
})
peer3.on('data', data => {
console.log('got a message from peer3: ' + data)
})
// These are peer2's connections to peer1 and peer3
var peer1 = new Peer()
var peer3 = new Peer({ initiator: true })
peer1.on('signal', data => {
// send this signaling data to peer1 somehow
})
peer1.on('connect', () => {
peer1.send('hi peer1, this is peer2')
})
peer1.on('data', data => {
console.log('got a message from peer1: ' + data)
})
peer3.on('signal', data => {
// send this signaling data to peer3 somehow
})
peer3.on('connect', () => {
peer3.send('hi peer3, this is peer2')
})
peer3.on('data', data => {
console.log('got a message from peer3: ' + data)
})
// These are peer3's connections to peer1 and peer2
var peer1 = new Peer()
var peer2 = new Peer()
peer1.on('signal', data => {
// send this signaling data to peer1 somehow
})
peer1.on('connect', () => {
peer1.send('hi peer1, this is peer3')
})
peer1.on('data', data => {
console.log('got a message from peer1: ' + data)
})
peer2.on('signal', data => {
// send this signaling data to peer2 somehow
})
peer2.on('connect', () => {
peer2.send('hi peer2, this is peer3')
})
peer2.on('data', data => {
console.log('got a message from peer2: ' + data)
})
If you call peer.send(buf)
, simple-peer
is not keeping a reference to buf
and sending the buffer at some later point in time. We immediately call
channel.send()
on the data channel. So it should be fine to mutate the buffer
right afterward.
However, beware that peer.write(buf)
(a writable stream method) does not have
the same contract. It will potentially buffer the data and call
channel.send()
at a future point in time, so definitely don't assume it's
safe to mutate the buffer.
If a direct connection fails, in particular, because of NAT traversal and/or firewalls, WebRTC ICE uses an intermediary (relay) TURN server. In other words, ICE will first use STUN with UDP to directly connect peers and, if that fails, will fall back to a TURN relay server.
In order to use a TURN server, you must specify the config
option to the Peer
constructor. See the API docs above.
- WebTorrent - Streaming torrent client in the browser
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- Instant.io - Secure, anonymous, streaming file transfer
- Zencastr - Easily record your remote podcast interviews in studio quality.
- Friends - Peer-to-peer chat powered by the web
- Socket.io-p2p - Official Socket.io P2P communication library
- ScreenCat - Screen sharing + remote collaboration app
- WebCat - P2P pipe across the web using Github private/public key for auth
- RTCCat - WebRTC netcat
- PeerNet - Peer-to-peer gossip network using randomized algorithms
- PusherTC - Video chat with using Pusher. See guide.
- lxjs-chat - Omegle-like video chat site
- Whiteboard - P2P Whiteboard powered by WebRTC and WebTorrent
- Peer Calls - WebRTC group video calling. Create a room. Share the link.
- Netsix - Send videos to your friends using WebRTC so that they can watch them right away.
- Stealthy - Stealthy is a decentralized, end-to-end encrypted, p2p chat application.
- oorja.io - Effortless video-voice chat with realtime collaborative features. Extensible using react components π
- TalktoMe - Skype alternative for audio/video conferencing based on WebRTC, but without the loss of packets.
- CDNBye - CDNBye implements WebRTC datachannel to scale live/vod video streaming by peer-to-peer network using bittorrent-like protocol
- Detox - Overlay network for distributed anonymous P2P communications entirely in the browser
- Metastream - Watch streaming media with friends.
- firepeer - secure signalling and authentication using firebase realtime database
- Genet - Fat-tree overlay to scale the number of concurrent WebRTC connections to a single source (paper).
- WebRTC Connection Testing - Quickly test direct connectivity between all pairs of participants (demo).
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- safeShare - Transfer files easily with text and voice communication.
- CubeChat - Party in 3D π
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- AnyDrop - Cross-platform AirDrop alternative with an Android app available at Google Play
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- QuaranTime.io - The Activity board-game in video!
- Trango - Cross-platform calling and file sharing solution.
- Your app here! - send a PR!
MIT. Copyright (c) Feross Aboukhadijeh.