A libp2p transport using WebRTC connections
npm install @libp2p/webrtc
> A libp2p transport using WebRTC connections
A libp2p transport based on WebRTC data channels.
WebRTC is a specification that allows real-time communication between nodes - it's commonly used in browser video conferencing applications but it also provides a reliable data transport mechanism called data channels which libp2p uses to facilitate protocol streams between peers.
There are two transports exposed by this module, webRTC and webRTCDirect.
The connection establishment phase of WebRTC involves a handshake using SDP during which two peers will exchange information such as open ports, network addresses and required capabilities.
A third party is usually necessary to carry out this handshake, forwarding messages between the two peers until they can make a direct connection between themselves.
The WebRTC transport uses libp2p Circuit Relays to forward SDP messages. Once a direct connection is formed the relay plays no further part in the exchange.
WebRTC Direct uses a technique known as SDP munging to skip the handshake step, instead encoding enough information in the connection request that the responder can derive what would have been in the handshake messages and so requires no third parties to establish a connection.
A WebRTC Direct multiaddr also includes a certhash of the target peer - this is used to allow opening a connection to the remote, which would otherwise be denied due to use of a self-signed certificate.
In both cases, once the connection is established a Noise handshake is carried out to ensure that the remote peer has the private key that corresponds to the public key that makes up their PeerId, giving you both encryption and authentication.
WebRTC requires use of a relay to connect two nodes. The listener first discovers a relay server and makes a reservation, then the dialer can connect via the relayed address.
``TypeScript
import { noise } from '@chainsafe/libp2p-noise'
import { yamux } from '@chainsafe/libp2p-yamux'
import { echo } from '@libp2p/echo'
import { circuitRelayTransport, circuitRelayServer } from '@libp2p/circuit-relay-v2'
import { identify } from '@libp2p/identify'
import { webRTC } from '@libp2p/webrtc'
import { webSockets } from '@libp2p/websockets'
import { WebRTC } from '@multiformats/multiaddr-matcher'
import delay from 'delay'
import { createLibp2p } from 'libp2p'
import type { Multiaddr } from '@multiformats/multiaddr'
// the relay server listens on a transport dialable by the listener and the
// dialer, and has a relay service configured
const relay = await createLibp2p({
addresses: {
listen: ['/ip4/127.0.0.1/tcp/0/ws']
},
transports: [
webSockets()
],
connectionEncrypters: [noise()],
streamMuxers: [yamux()],
connectionGater: {
denyDialMultiaddr: () => false
},
services: {
identify: identify(),
relay: circuitRelayServer()
}
})
// the listener has a WebSocket transport to dial the relay, a Circuit Relay
// transport to make a reservation, and a WebRTC transport to accept incoming
// WebRTC connections
const listener = await createLibp2p({
addresses: {
listen: [
'/p2p-circuit',
'/webrtc'
]
},
transports: [
webSockets(),
webRTC(),
circuitRelayTransport()
],
connectionEncrypters: [noise()],
streamMuxers: [yamux()],
connectionGater: {
denyDialMultiaddr: () => false
},
services: {
identify: identify(),
echo: echo()
}
})
// the listener dials the relay (or discovers a public relay via some other
// method)
await listener.dial(relay.getMultiaddrs(), {
signal: AbortSignal.timeout(5000)
})
let webRTCMultiaddr: Multiaddr | undefined
// wait for the listener to make a reservation on the relay
while (true) {
webRTCMultiaddr = listener.getMultiaddrs().find(ma => WebRTC.matches(ma))
if (webRTCMultiaddr != null) {
break
}
// try again later
await delay(1000)
}
// the dialer has Circuit Relay, WebSocket and WebRTC transports to dial
// the listener via the relay, complete the SDP handshake and establish a
// direct WebRTC connection
const dialer = await createLibp2p({
transports: [
webSockets(),
webRTC(),
circuitRelayTransport()
],
connectionEncrypters: [noise()],
streamMuxers: [yamux()],
connectionGater: {
denyDialMultiaddr: () => false
},
services: {
identify: identify(),
echo: echo()
}
})
// dial the listener and open an echo protocol stream
const stream = await dialer.dialProtocol(webRTCMultiaddr, dialer.services.echo.protocol, {
signal: AbortSignal.timeout(5000)
})
// we can now stop the relay
await relay.stop()
// send/receive some data from the remote peer via a direct connection
stream.send(new TextEncoder().encode('hello world'))
stream.addEventListener('message', (evt) => {
console.info(new TextDecoder().decode(evt.data.subarray()))
})
`
WebRTC Direct allows a client to establish a WebRTC connection to a server
without using a relay to first exchange SDP messages.
Instead the server listens on a public UDP port and embeds its certificate
hash in the published multiaddr. It derives the client's SDP offer based on
the incoming IP/port of STUN messages sent to this public port.
The client derives the server's SDP answer based on the information in the
multiaddr so no SDP handshake via a third party is required.
Full details of the connection protocol can be found in the WebRTC Direct spec.
Browsers cannot listen on WebRTC Direct addresses since they cannot open
ports, but they can dial all spec-compliant servers.
Node.js/go and rust-libp2p can listen on and dial WebRTC Direct addresses.
`TypeScript
import { createLibp2p } from 'libp2p'
import { multiaddr } from '@multiformats/multiaddr'
import { fromString, toString } from 'uint8arrays'
import { webRTCDirect } from '@libp2p/webrtc'
const listener = await createLibp2p({
addresses: {
listen: [
'/ip4/0.0.0.0/udp/0/webrtc-direct'
]
},
transports: [
webRTCDirect()
]
})
await listener.start()
const dialer = await createLibp2p({
transports: [
webRTCDirect()
]
})
await dialer.start()
const stream = await dialer.dialProtocol(listener.getMultiaddrs(), '/my-protocol/1.0.0', {
signal: AbortSignal.timeout(10_000)
})
stream.send(new TextEncoder().encode('hello world'))
stream.addEventListener('message', (evt) => {
console.info(new TextDecoder().decode(evt.data.subarray()))
})
`
WebRTC Direct listeners publish the hash of their TLS certificate as part of
the listening multiaddr.
By default these certificates are generated at start up using an ephemeral
keypair that only exists while the node is running.
This means that the certificate hashes change when the node is restarted,
which can be undesirable if multiaddrs are intended to be long lived (e.g.
if the node is used as a network bootstrapper).
To reuse the same certificate and keypair, configure a persistent datastore
and the @libp2p/keychain
service as part of your service map:
`ts
import { LevelDatastore } from 'datastore-level'
import { webRTCDirect } from '@libp2p/webrtc'
import { keychain } from '@libp2p/keychain'
import { createLibp2p } from 'libp2p'
// store data on disk between restarts
const datastore = new LevelDatastore('/path/to/store')
const listener = await createLibp2p({
addresses: {
listen: [
'/ip4/0.0.0.0/udp/0/webrtc-direct'
]
},
datastore,
transports: [
webRTCDirect()
],
services: {
keychain: keychain()
}
})
await listener.start()
console.info(listener.getMultiaddrs())
// /ip4/...../udp/../webrtc-direct/certhash/foo
await listener.stop()
await listener.start()
console.info(listener.getMultiaddrs())
// /ip4/...../udp/../webrtc-direct/certhash/foo
`
`console`
$ npm i @libp2p/webrtc
-
Licensed under either of
- Apache 2.0, (LICENSE-APACHE /
- MIT (LICENSE-MIT /
Unless you explicitly state otherwise, any contribution intentionally submitted for inclusion in the work by you, as defined in the Apache-2.0 license, shall be dual licensed as above, without any additional terms or conditions.