feat: WebRTC-Direct support for Node.js (#2583)

Supports listening on and dialing WebRTC-Direct multiaddrs in Node.js.

Closes:

- #2581

Author: achingbrain

interop/README.md

@@ -75,7 +75,7 @@ $ docker build . -f ./interop/BrowserDockerfile -t js-libp2p-browsers
      - When starting the docker container add `-e GOLOG_LOG_LEVEL=debug`
 4. Build the version you want to test against
    ```console
-   $ cd multidim-interop/impl/$IMPL/$VERSION
+   $ cd transport-interop/impl/$IMPL/$VERSION
    $ make
    ...
    ```

interop/firefox-version.json

@@ -13,5 +13,5 @@
     }
   ],
   "secureChannels": ["noise"],
-  "muxers": ["mplex", "yamux"]
+  "muxers": ["yamux", "mplex"]
 }

interop/node-version.json

@@ -1,19 +1,16 @@
 {
-    "id": "node-js-libp2p-head",
-    "containerImageID": "node-js-libp2p-head",
-    "transports": [
-        "tcp",
-        "ws",
-        {
-            "name": "wss",
-            "onlyDial": true
-        }
-    ],
-    "secureChannels": [
-        "noise"
-    ],
-    "muxers": [
-        "mplex",
-        "yamux"
-    ]
-}
+  "id": "node-js-libp2p-head",
+  "containerImageID": "node-js-libp2p-head",
+  "transports": [
+    "tcp",
+    "ws",
+    {
+      "name": "wss",
+      "onlyDial": true
+    },
+    "webrtc",
+    "webrtc-direct"
+  ],
+  "secureChannels": ["noise"],
+  "muxers": ["yamux", "mplex"]
+}

interop/test/dialer.spec.ts

@@ -8,15 +8,15 @@ import type { Libp2p } from '@libp2p/interface'
 import type { PingService } from '@libp2p/ping'
 
 const isDialer: boolean = process.env.is_dialer === 'true'
-const timeoutSecs: string = process.env.test_timeout_secs ?? '180'
+const timeoutMs: number = parseInt(process.env.test_timeout_secs ?? '180') * 1000
 
 describe('ping test (dialer)', function () {
   if (!isDialer) {
     return
   }
 
   // make the default timeout longer than the listener timeout
-  this.timeout((parseInt(timeoutSecs) * 1000) + 30000)
+  this.timeout(timeoutMs + 30_000)
   let node: Libp2p<{ ping: PingService }>
 
   beforeEach(async () => {
@@ -32,7 +32,7 @@ describe('ping test (dialer)', function () {
   })
 
   it('should dial and ping', async function () {
-    let [, otherMaStr]: string[] = await redisProxy(['BLPOP', 'listenerAddr', timeoutSecs])
+    let [, otherMaStr]: string[] = await redisProxy(['BLPOP', 'listenerAddr', `${timeoutMs / 1000}`])
 
     // Hack until these are merged:
     // - https://github.com/multiformats/js-multiaddr-to-uri/pull/120
@@ -45,7 +45,9 @@ describe('ping test (dialer)', function () {
     await node.dial(otherMa)
 
     console.error(`node ${node.peerId.toString()} pings: ${otherMa}`)
-    const pingRTT = await node.services.ping.ping(multiaddr(otherMa))
+    const pingRTT = await node.services.ping.ping(multiaddr(otherMa), {
+      signal: AbortSignal.timeout(timeoutMs)
+    })
     const handshakePlusOneRTT = Date.now() - handshakeStartInstant
     console.log(JSON.stringify({
       handshakePlusOneRTTMillis: handshakePlusOneRTT,

interop/test/listener.spec.ts

@@ -67,6 +67,7 @@ describe('ping test (listener)', function () {
     }
 
     console.error('inform redis of dial address')
+    console.error(multiaddrs)
     // Send the listener addr over the proxy server so this works on both the Browser and Node
     await redisProxy(['RPUSH', 'listenerAddr', multiaddrs[0]])
     // Wait

packages/integration-tests/test/compliance/transport/webrtc-direct.spec.ts

@@ -0,0 +1,39 @@
+import tests from '@libp2p/interface-compliance-tests/transport'
+import { webRTCDirect } from '@libp2p/webrtc'
+import { WebRTCDirect } from '@multiformats/multiaddr-matcher'
+import { isNode, isElectron } from 'wherearewe'
+
+describe('WebRTC-Direct interface-transport compliance', () => {
+  if (!isNode && !isElectron) {
+    return
+  }
+
+  tests({
+    async setup () {
+      const dialer = {
+        transports: [
+          webRTCDirect()
+        ],
+        connectionMonitor: {
+          enabled: false
+        }
+      }
+
+      return {
+        dialer,
+        listener: {
+          addresses: {
+            listen: [
+              '/ip4/127.0.0.1/udp/0/webrtc-direct',
+              '/ip4/127.0.0.1/udp/0/webrtc-direct'
+            ]
+          },
+          ...dialer
+        },
+        dialMultiaddrMatcher: WebRTCDirect,
+        listenMultiaddrMatcher: WebRTCDirect
+      }
+    },
+    async teardown () {}
+  })
+})

packages/integration-tests/test/interop.ts

@@ -15,6 +15,7 @@ import { mplex } from '@libp2p/mplex'
 import { plaintext } from '@libp2p/plaintext'
 import { tcp } from '@libp2p/tcp'
 import { tls } from '@libp2p/tls'
+import { webRTCDirect } from '@libp2p/webrtc'
 import { multiaddr } from '@multiformats/multiaddr'
 import { execa } from 'execa'
 import { path as p2pd } from 'go-libp2p'
@@ -131,20 +132,26 @@ async function createJsPeer (options: SpawnOptions): Promise<Daemon> {
     addresses: {
       listen: []
     },
-    transports: [tcp(), circuitRelayTransport()],
+    transports: [
+      tcp(),
+      circuitRelayTransport(),
+      webRTCDirect()
+    ],
     streamMuxers: [],
     connectionEncrypters: [noise()]
   }
 
   if (options.noListen !== true) {
     if (options.transport == null || options.transport === 'tcp') {
       opts.addresses?.listen?.push('/ip4/127.0.0.1/tcp/0')
+    } else if (options.transport === 'webrtc-direct') {
+      opts.addresses?.listen?.push('/ip4/127.0.0.1/udp/0/webrtc-direct')
     } else {
       throw new UnsupportedError()
     }
   }
 
-  if (options.transport === 'webtransport' || options.transport === 'webrtc-direct') {
+  if (options.transport === 'webtransport') {
     throw new UnsupportedError()
   }
 
@@ -191,7 +198,7 @@ async function createJsPeer (options: SpawnOptions): Promise<Daemon> {
     services
   })
 
-  const server = createServer(multiaddr('/ip4/0.0.0.0/tcp/0'), node)
+  const server = createServer(multiaddr('/ip4/127.0.0.1/tcp/0'), node)
   await server.start()
 
   return {

packages/interface-compliance-tests/src/transport/index.ts

@@ -191,8 +191,13 @@ export default (common: TestSetup<TransportTestFixtures>): void => {
         throw new Error('Oh noes!')
       }
 
-      await expect(dialer.dial(dialAddrs[0])).to.eventually.be.rejected
-        .with.property('name', 'EncryptionFailedError')
+      // transports with their own muxers/encryption will perform the upgrade
+      // after the connection has been established (e.g. peer ids have been
+      // exchanged) so perform the dial and wait for the remote to attempt the
+      // upgrade - if it fails the listener should close the underlying
+      // connection which should remove the it from the dialer's connection map
+      await dialer.dial(dialAddrs[0]).catch(() => {})
+      await delay(1000)
 
       expect(dialer.getConnections().filter(conn => {
         return dialMultiaddrMatcher.exactMatch(conn.remoteAddr)

packages/transport-webrtc/README.md

@@ -44,18 +44,6 @@ A WebRTC Direct multiaddr also includes a certhash of the target peer - this is
 
 In both cases, once the connection is established a [Noise handshake](https://noiseprotocol.org/noise.html) 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.
 
-## Support
-
-WebRTC is supported in both Node.js and browsers.
-
-At the time of writing, WebRTC Direct is dial-only in browsers and not supported in Node.js at all.
-
-Support in Node.js is possible but PRs will need to be opened to [libdatachannel](https://github.com/paullouisageneau/libdatachannel) and the appropriate APIs exposed in [node-datachannel](https://github.com/murat-dogan/node-datachannel).
-
-WebRTC Direct support is available in rust-libp2p and arriving soon in go-libp2p.
-
-See the WebRTC section of <https://connectivity.libp2p.io> for more information.
-
 ## Example - WebRTC
 
 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.
@@ -180,26 +168,33 @@ The only implementation that supports a WebRTC Direct listener is go-libp2p and
 
 ```TypeScript
 import { createLibp2p } from 'libp2p'
-import { noise } from '@chainsafe/libp2p-noise'
 import { multiaddr } from '@multiformats/multiaddr'
 import { pipe } from 'it-pipe'
 import { fromString, toString } from 'uint8arrays'
 import { webRTCDirect } from '@libp2p/webrtc'
 
-const node = await createLibp2p({
+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()
-  ],
-  connectionEncrypters: [
-    noise()
   ]
 })
 
-await node.start()
+await dialer.start()
 
-// this multiaddr corresponds to a remote node running a WebRTC Direct listener
-const ma = multiaddr('/ip4/0.0.0.0/udp/56093/webrtc-direct/certhash/uEiByaEfNSLBexWBNFZy_QB1vAKEj7JAXDizRs4_SnTflsQ')
-const stream = await node.dialProtocol(ma, '/my-protocol/1.0.0', {
+const stream = await dialer.dialProtocol(listener.getMultiaddrs(), '/my-protocol/1.0.0', {
   signal: AbortSignal.timeout(10_000)
 })
 

packages/transport-webrtc/package.json

@@ -40,7 +40,7 @@
     "generate": "protons src/private-to-private/pb/message.proto src/pb/message.proto",
     "build": "aegir build",
     "test": "aegir test -t node -t browser",
-    "test:node": "aegir test -t node --cov -- --exit",
+    "test:node": "aegir test -t node --cov",
     "test:chrome": "aegir test -t browser --cov",
     "test:firefox": "aegir test -t browser -- --browser firefox",
     "test:webkit": "aegir test -t browser -- --browser webkit",
@@ -51,27 +51,35 @@
     "doc-check": "aegir doc-check"
   },
   "dependencies": {
+    "@chainsafe/is-ip": "^2.0.2",
     "@chainsafe/libp2p-noise": "^16.0.0",
+    "@ipshipyard/node-datachannel": "^0.26.4",
     "@libp2p/interface": "^2.5.0",
     "@libp2p/interface-internal": "^2.3.0",
     "@libp2p/peer-id": "^5.0.12",
     "@libp2p/utils": "^6.5.1",
     "@multiformats/multiaddr": "^12.3.3",
     "@multiformats/multiaddr-matcher": "^1.6.0",
+    "@peculiar/webcrypto": "^1.5.0",
+    "@peculiar/x509": "^1.11.0",
+    "any-signal": "^4.1.1",
     "detect-browser": "^5.3.0",
+    "get-port": "^7.1.0",
     "it-length-prefixed": "^9.1.0",
     "it-protobuf-stream": "^1.1.5",
     "it-pushable": "^3.2.3",
     "it-stream-types": "^2.0.2",
     "multiformats": "^13.3.1",
-    "node-datachannel": "^0.11.0",
     "p-defer": "^4.0.1",
     "p-event": "^6.0.1",
     "p-timeout": "^6.1.3",
+    "p-wait-for": "^5.0.2",
     "progress-events": "^1.0.1",
     "protons-runtime": "^5.5.0",
+    "race-event": "^1.3.0",
     "race-signal": "^1.1.0",
     "react-native-webrtc": "^124.0.4",
+    "stun": "^2.1.0",
     "uint8-varint": "^2.0.4",
     "uint8arraylist": "^2.4.8",
     "uint8arrays": "^5.1.0"
@@ -91,7 +99,10 @@
     "sinon-ts": "^2.0.0"
   },
   "browser": {
-    "./dist/src/webrtc/index.js": "./dist/src/webrtc/index.browser.js"
+    "./dist/src/webrtc/index.js": "./dist/src/webrtc/index.browser.js",
+    "./dist/src/private-to-public/listener.js": "./dist/src/private-to-public/listener.browser.js",
+    "./dist/src/private-to-public/utils/get-rtcpeerconnection.js": "./dist/src/private-to-public/utils/get-rtcpeerconnection.browser.js",
+    "node:net": false
   },
   "react-native": {
     "./dist/src/webrtc/index.js": "./dist/src/webrtc/index.react-native.js"

packages/transport-webrtc/src/constants.ts

@@ -12,3 +12,7 @@ export const DEFAULT_ICE_SERVERS = [
   'stun:stun.cloudflare.com:3478',
   'stun:stun.services.mozilla.com:3478'
 ]
+
+export const UFRAG_ALPHABET = Array.from('abcdefghijklmnopqrstuvwxyzABCDEFGHIJKLMNOPQRSTUVWXYZ1234567890')
+
+export const UFRAG_PREFIX = 'libp2p+webrtc+v1/'