@libp2p/webrtc 5.2.24-0f07e3df5 → 5.2.24-6059227cb

package/dist/src/stream.js

@@ -1,11 +1,13 @@
-import { StreamStateError } from '@libp2p/interface';
+import { StreamResetError, StreamStateError } from '@libp2p/interface';
 import { AbstractStream } from '@libp2p/utils';
 import * as lengthPrefixed from 'it-length-prefixed';
 import { pushable } from 'it-pushable';
+import { pEvent } from 'p-event';
 import { raceSignal } from 'race-signal';
 import { Uint8ArrayList } from 'uint8arraylist';
-import { MAX_BUFFERED_AMOUNT, MAX_MESSAGE_SIZE, PROTOBUF_OVERHEAD } from './constants.js';
+import { DEFAULT_FIN_ACK_TIMEOUT, MAX_BUFFERED_AMOUNT, MAX_MESSAGE_SIZE, PROTOBUF_OVERHEAD } from './constants.js';
 import { Message } from './private-to-public/pb/message.js';
+import { isFirefox } from './util.js';
 export class WebRTCStream extends AbstractStream {
     /**
      * The data channel used to send and receive data
@@ -18,6 +20,7 @@ export class WebRTCStream extends AbstractStream {
     incomingData;
     maxBufferedAmount;
     receivedFinAck;
+    finAckTimeout;
     constructor(init) {
         super({
             ...init,
@@ -27,36 +30,20 @@ export class WebRTCStream extends AbstractStream {
         this.channel.binaryType = 'arraybuffer';
         this.incomingData = pushable();
         this.maxBufferedAmount = init.maxBufferedAmount ?? MAX_BUFFERED_AMOUNT;
-        this.receivedFinAck = Promise.withResolvers();
-        // set up initial state
-        switch (this.channel.readyState) {
-            case 'open':
-                break;
-            case 'closed':
-            case 'closing':
-                if (this.timeline.close === undefined || this.timeline.close === 0) {
-                    this.timeline.close = Date.now();
-                }
-                break;
-            case 'connecting':
-                // noop
-                break;
-            default:
-                this.log.error('unknown datachannel state %s', this.channel.readyState);
-                throw new StreamStateError('Unknown datachannel state');
-        }
+        this.finAckTimeout = init.finAckTimeout ?? DEFAULT_FIN_ACK_TIMEOUT;
         // handle RTCDataChannel events
-        this.channel.onclose = (_evt) => {
-            this.log.trace('received onclose event');
+        this.channel.onclose = () => {
+            this.log.trace('received datachannel close event');
             this.onRemoteCloseWrite();
             this.onTransportClosed();
         };
         this.channel.onerror = (evt) => {
-            this.log.trace('received onerror event');
             const err = evt.error;
+            this.log.trace('received datachannel error event - %e', err);
             this.abort(err);
         };
         this.channel.onmessage = async (event) => {
+            this.log('incoming message %d bytes', event.data.byteLength);
             const { data } = event;
             if (data === null || data.byteLength === 0) {
                 return;
@@ -66,79 +53,123 @@ export class WebRTCStream extends AbstractStream {
         // dispatch drain event when the buffered amount drops to zero
         this.channel.bufferedAmountLowThreshold = 0;
         this.channel.onbufferedamountlow = () => {
-            this.safeDispatchEvent('drain');
+            if (this.writableNeedsDrain) {
+                this.safeDispatchEvent('drain');
+            }
         };
-        const self = this;
+        if (this.channel.readyState !== 'open') {
+            this.log('channel ready state is "%s" and not "open", waiting for "open" event before sending data', this.channel.readyState);
+            pEvent(this.channel, 'open', {
+                rejectionEvents: [
+                    'close',
+                    'error'
+                ]
+            })
+                .then(() => {
+                this.log('channel ready state is now "%s", dispatching drain', this.channel.readyState);
+                this.safeDispatchEvent('drain');
+            })
+                .catch(err => {
+                this.abort(err.error ?? err);
+            });
+        }
         // pipe framed protobuf messages through a length prefixed decoder, and
         // surface data from the `Message.message` field through a source.
         Promise.resolve().then(async () => {
             for await (const buf of lengthPrefixed.decode(this.incomingData)) {
-                const message = self.processIncomingProtobuf(buf);
-                if (message != null) {
-                    self.onData(new Uint8ArrayList(message));
-                }
+                this.processIncomingProtobuf(buf);
             }
         })
             .catch(err => {
             this.log.error('error processing incoming data channel messages', err);
         });
-        // clean up the datachannel when both ends have sent a FIN
-        const webRTCStreamOnClose = () => {
-            this.channel.close();
+        // close when both writable ends are closed or an error occurs
+        const cleanUpDatachannelOnClose = () => {
+            if (this.channel.readyState === 'open') {
+                this.log.trace('stream closed, closing underlying datachannel');
+                this.channel.close();
+            }
         };
-        this.addEventListener('close', webRTCStreamOnClose);
+        this.addEventListener('close', cleanUpDatachannelOnClose);
     }
     sendNewStream() {
         // opening new streams is handled by WebRTC so this is a noop
     }
     _sendMessage(data) {
-        if (this.channel.readyState === 'closed' || this.channel.readyState === 'closing') {
+        if (this.channel.readyState !== 'open') {
             throw new StreamStateError(`Invalid datachannel state - ${this.channel.readyState}`);
         }
-        try {
-            this.log.trace('sending message, channel state "%s"', this.channel.readyState);
-            // send message without copying data
-            for (const buf of data) {
-                this.channel.send(buf);
-            }
+        this.log.trace('sending message, channel state "%s"', this.channel.readyState);
+        if (isFirefox) {
+            // TODO: firefox can deliver small messages out of order - remove once a
+            // browser with https://bugzilla.mozilla.org/show_bug.cgi?id=1983831 is
+            // available in playwright-test
+            this.channel.send(data.subarray());
+            return;
         }
-        catch (err) {
-            this.log.error('error while sending message', err);
+        // send message without copying data
+        for (const buf of data) {
+            this.channel.send(buf);
         }
     }
     sendData(data) {
-        const messageBuf = Message.encode({
+        if (this.channel.readyState !== 'open') {
+            return {
+                sentBytes: 0,
+                canSendMore: false
+            };
+        }
+        // TODO: firefox can deliver small messages out of order - remove once a
+        // browser with https://bugzilla.mozilla.org/show_bug.cgi?id=1983831 is
+        // available in playwright-test
+        // ----
+        // this is also necessary to work with rust-libp2p 0.54 though 0.53 seems ok
+        this._sendMessage(lengthPrefixed.encode.single(Message.encode({
             message: data.subarray()
-        });
-        const prefixedBuf = lengthPrefixed.encode.single(messageBuf);
-        this._sendMessage(prefixedBuf);
+        })));
+        /*
+        // TODO: enable this when FF and rust-libp2p are not broken
+        // send message without copying data
+          for (const message of data) {
+            this._sendMessage(
+              lengthPrefixed.encode.single(Message.encode({
+                message
+              }))
+            )
+          }
+        }
+        */
         return {
             sentBytes: data.byteLength,
             canSendMore: this.channel.bufferedAmount < this.maxBufferedAmount
         };
     }
-    sendReset() {
-        this.receivedFinAck.resolve();
+    sendReset(err) {
         try {
+            this.log.error('sending reset - %e', err);
             this._sendFlag(Message.Flag.RESET);
+            this.receivedFinAck?.reject(err);
         }
         catch (err) {
             this.log.error('failed to send reset - %e', err);
         }
-        finally {
-            this.channel.close();
-        }
     }
     async sendCloseWrite(options) {
-        if (this.channel.readyState === 'open') {
-            this._sendFlag(Message.Flag.FIN);
-        }
-        await raceSignal(this.receivedFinAck.promise, options?.signal);
+        this._sendFlag(Message.Flag.FIN);
+        options?.signal?.throwIfAborted();
+        this.receivedFinAck = Promise.withResolvers();
+        await Promise.any([
+            raceSignal(this.receivedFinAck.promise, options?.signal),
+            new Promise(resolve => {
+                AbortSignal.timeout(this.finAckTimeout)
+                    .addEventListener('abort', () => {
+                    resolve();
+                });
+            })
+        ]);
     }
     async sendCloseRead(options) {
-        if (this.channel.readyState === 'open') {
-            this._sendFlag(Message.Flag.STOP_SENDING);
-        }
+        this._sendFlag(Message.Flag.STOP_SENDING);
         options?.signal?.throwIfAborted();
     }
     /**
@@ -146,30 +177,31 @@ export class WebRTCStream extends AbstractStream {
      */
     processIncomingProtobuf(buffer) {
         const message = Message.decode(buffer);
+        // ignore data messages if we've closed the readable end already
+        if (message.message != null && (this.readStatus === 'readable' || this.readStatus === 'paused')) {
+            this.onData(new Uint8ArrayList(message.message));
+        }
         if (message.flag !== undefined) {
             this.log.trace('incoming flag %s, write status "%s", read status "%s"', message.flag, this.writeStatus, this.readStatus);
             if (message.flag === Message.Flag.FIN) {
-                // We should expect no more data from the remote, stop reading
-                this.onRemoteCloseWrite();
+                // we should expect no more data from the remote, stop reading
                 this._sendFlag(Message.Flag.FIN_ACK);
+                this.onRemoteCloseWrite();
             }
             if (message.flag === Message.Flag.RESET) {
-                this.receivedFinAck.resolve();
-                // Stop reading and writing to the stream immediately
+                // stop reading and writing to the stream immediately
+                this.receivedFinAck?.reject(new StreamResetError('The stream was reset'));
                 this.onRemoteReset();
             }
             if (message.flag === Message.Flag.STOP_SENDING) {
-                // The remote has stopped reading
+                // the remote has stopped reading
                 this.onRemoteCloseRead();
             }
             if (message.flag === Message.Flag.FIN_ACK) {
-                this.receivedFinAck.resolve();
+                // remote received our FIN
+                this.receivedFinAck?.resolve();
             }
         }
-        // ignore data messages if we've closed the readable end already
-        if (this.readStatus === 'readable' || this.readStatus === 'paused') {
-            return message.message;
-        }
     }
     _sendFlag(flag) {
         if (this.channel.readyState !== 'open') {

package/dist/src/stream.js.map

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package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@libp2p/webrtc",
-  "version": "5.2.24-0f07e3df5",
+  "version": "5.2.24-6059227cb",
   "description": "A libp2p transport using WebRTC connections",
   "license": "Apache-2.0 OR MIT",
   "homepage": "https://github.com/libp2p/js-libp2p/tree/main/packages/transport-webrtc#readme",
@@ -45,13 +45,13 @@
   },
   "dependencies": {
     "@chainsafe/is-ip": "^2.1.0",
-    "@libp2p/crypto": "5.1.8-0f07e3df5",
-    "@libp2p/interface": "2.11.0-0f07e3df5",
-    "@libp2p/interface-internal": "2.3.19-0f07e3df5",
-    "@libp2p/keychain": "5.2.9-0f07e3df5",
-    "@libp2p/noise": "16.1.4-0f07e3df5",
-    "@libp2p/peer-id": "5.1.9-0f07e3df5",
-    "@libp2p/utils": "6.7.2-0f07e3df5",
+    "@libp2p/crypto": "5.1.8-6059227cb",
+    "@libp2p/interface": "2.11.0-6059227cb",
+    "@libp2p/interface-internal": "2.3.19-6059227cb",
+    "@libp2p/keychain": "5.2.9-6059227cb",
+    "@libp2p/noise": "16.1.4-6059227cb",
+    "@libp2p/peer-id": "5.1.9-6059227cb",
+    "@libp2p/utils": "6.7.2-6059227cb",
     "@multiformats/multiaddr": "^13.0.1",
     "@multiformats/multiaddr-matcher": "^3.0.1",
     "@peculiar/webcrypto": "^1.5.0",
@@ -79,9 +79,9 @@
     "uint8arrays": "^5.1.0"
   },
   "devDependencies": {
-    "@libp2p/logger": "5.2.0-0f07e3df5",
+    "@libp2p/logger": "5.2.0-6059227cb",
     "@types/sinon": "^17.0.4",
-    "aegir": "^47.0.21",
+    "aegir": "^47.0.22",
     "any-signal": "^4.1.1",
     "datastore-core": "^10.0.4",
     "delay": "^6.0.0",

package/src/constants.ts

@@ -73,6 +73,11 @@ export const PROTOBUF_OVERHEAD = calculateProtobufOverhead()
  */
 export const DATA_CHANNEL_DRAIN_TIMEOUT = 30_000
 
+/**
+ * Wait for the remote to acknowledge our FIN for this long
+ */
+export const DEFAULT_FIN_ACK_TIMEOUT = 10_000
+
 /**
  * Set as the 'negotiated' muxer protocol name
  */

package/src/index.ts

@@ -308,6 +308,15 @@ export interface DataChannelOptions {
    * @default 5_000
    */
   openTimeout?: number
+
+  /**
+   * Due to bugs in WebRTC implementations it's necessary for the remote end of
+   * the connection to acknowledge the FIN message we send during stream
+   * closing. A stream will wait for this many ms.
+   *
+   * @default 10_000
+   */
+  finAckTimeout?: number
 }
 
 /**

package/src/muxer.ts

@@ -1,5 +1,4 @@
 import { AbstractStreamMuxer } from '@libp2p/utils'
-import { pEvent } from 'p-event'
 import { MUXER_PROTOCOL } from './constants.js'
 import { createStream, WebRTCStream } from './stream.js'
 import type { DataChannelOptions } from './index.js'
@@ -90,11 +89,12 @@ export class DataChannelMuxer extends AbstractStreamMuxer<WebRTCStream> implemen
      * {@link https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/datachannel_event}
      */
     this.peerConnection.ondatachannel = ({ channel }) => {
-      this.log.trace('incoming %s datachannel with channel id %d and status', channel.protocol, channel.id, channel.readyState)
+      this.log.trace('incoming %s datachannel with channel id %d, protocol %s and status %s', channel.protocol, channel.id, channel.protocol, channel.readyState)
 
-      // 'init' channel is only used during connection establishment
+      // 'init' channel is only used during connection establishment, it is
+      // closed by the initiator
       if (channel.label === 'init') {
-        this.log.trace('closing init channel')
+        this.log.trace('closing init channel %d', channel.id)
         channel.close()
 
         return
@@ -114,20 +114,13 @@ export class DataChannelMuxer extends AbstractStreamMuxer<WebRTCStream> implemen
 
   async onCreateStream (options?: CreateStreamOptions): Promise<WebRTCStream> {
     // The spec says the label MUST be an empty string: https://github.com/libp2p/specs/blob/master/webrtc/README.md#rtcdatachannel-label
-    const channel = this.peerConnection.createDataChannel('wtf', {
+    const channel = this.peerConnection.createDataChannel('', {
       // TODO: pre-negotiate stream protocol
-      protocol: options?.protocol
+      // protocol: options?.protocol
     })
 
     this.log('open channel %d for protocol %s', channel.id, options?.protocol)
 
-    if (channel.readyState !== 'open') {
-      this.log('channel %d state is "%s" and not "open", waiting for "open" event before sending data', channel.id, channel.readyState)
-      await pEvent(channel, 'open', options)
-
-      this.log('channel %d state is now "%s", sending data', channel.id, channel.readyState)
-    }
-
     const stream = createStream({
       ...options,
       ...this.dataChannelOptions,

package/src/private-to-private/initiate-connection.ts

@@ -94,10 +94,20 @@ export async function initiateConnection ({ rtcConfiguration, dataChannel, signa
 
     // setup callback to write ICE candidates to the remote peer
     peerConnection.onicecandidate = ({ candidate }) => {
+      if (peerConnection.connectionState === 'connected') {
+        log.trace('ignore new ice candidate as peer connection is already connected')
+        return
+      }
+
       // a null candidate means end-of-candidates, an empty string candidate
       // means end-of-candidates for this generation, otherwise this should
       // be a valid candidate object
       // see - https://www.w3.org/TR/webrtc/#rtcpeerconnectioniceevent
+      if (candidate == null || candidate?.candidate === '') {
+        log.trace('initiator detected end of ICE candidates')
+        return
+      }
+
       const data = JSON.stringify(candidate?.toJSON() ?? null)
 
       log.trace('initiator sending ICE candidate %o', candidate)
@@ -178,10 +188,16 @@ export async function initiateConnection ({ rtcConfiguration, dataChannel, signa
       })
     }
 
-    log.trace('closing init channel, starting status')
-
+    log.trace('closing init channel')
     channel.close()
 
+    // wait for init channel to close before proceeding, otherwise the channel
+    // id can be reused before both sides have seen the channel close
+    log.trace('waiting for init channel to close')
+    await pEvent(channel, 'close', {
+      signal
+    })
+
     onProgress?.(new CustomProgressEvent('webrtc:close-signaling-stream'))
 
     log.trace('closing signaling channel')

package/src/private-to-private/signaling-stream-handler.ts

@@ -3,7 +3,7 @@ import { multiaddr } from '@multiformats/multiaddr'
 import { SDPHandshakeFailedError } from '../error.js'
 import { RTCSessionDescription } from '../webrtc/index.js'
 import { Message } from './pb/message.js'
-import { getConnectionState, getRemotePeer, readCandidatesUntilConnected } from './util.js'
+import { getRemotePeer, readCandidatesUntilConnected } from './util.js'
 import type { RTCPeerConnection } from '../webrtc/index.js'
 import type { AbortOptions, Connection, Logger, PeerId, Stream } from '@libp2p/interface'
 import type { Multiaddr } from '@multiformats/multiaddr'
@@ -21,10 +21,20 @@ export async function handleIncomingStream (stream: Stream, connection: Connecti
   try {
     // candidate callbacks
     peerConnection.onicecandidate = ({ candidate }) => {
+      if (peerConnection.connectionState === 'connected') {
+        log.trace('ignore new ice candidate as peer connection is already connected')
+        return
+      }
+
       // a null candidate means end-of-candidates, an empty string candidate
       // means end-of-candidates for this generation, otherwise this should
       // be a valid candidate object
       // see - https://www.w3.org/TR/webrtc/#rtcpeerconnectioniceevent
+      if (candidate == null || candidate?.candidate === '') {
+        log.trace('recipient detected end of ICE candidates')
+        return
+      }
+
       const data = JSON.stringify(candidate?.toJSON() ?? null)
 
       log.trace('recipient sending ICE candidate %s', data)
@@ -90,7 +100,7 @@ export async function handleIncomingStream (stream: Stream, connection: Connecti
       log
     })
   } catch (err: any) {
-    if (getConnectionState(peerConnection) !== 'connected') {
+    if (peerConnection.connectionState !== 'connected') {
       log.error('error while handling signaling stream from peer %a', connection.remoteAddr, err)
 
       peerConnection.close()

package/src/private-to-private/transport.ts

@@ -31,14 +31,6 @@ export interface WebRTCTransportInit {
    * Any options here will be applied to any RTCDataChannels that are opened.
    */
   dataChannel?: DataChannelOptions
-
-  /**
-   * Inbound connections must complete the upgrade within this many ms
-   *
-   * @default 30_000
-   * @deprecated configure `connectionManager.inboundUpgradeTimeout` instead
-   */
-  inboundConnectionTimeout?: number
 }
 
 export interface WebRTCTransportComponents {

package/src/private-to-private/util.ts

@@ -1,7 +1,6 @@
 import { ConnectionFailedError, InvalidMessageError, InvalidMultiaddrError } from '@libp2p/interface'
 import { peerIdFromString } from '@libp2p/peer-id'
 import { CustomProgressEvent } from 'progress-events'
-import { isFirefox } from '../util.js'
 import { RTCIceCandidate } from '../webrtc/index.js'
 import { Message } from './pb/message.js'
 import type { WebRTCDialEvents } from './transport.js'
@@ -28,7 +27,7 @@ export const readCandidatesUntilConnected = async (pc: RTCPeerConnection, stream
         connectedPromise.promise,
         stream.read({
           signal: options.signal
-        }).catch(() => {})
+        })
       ])
 
       // stream ended or we became connected
@@ -63,32 +62,33 @@ export const readCandidatesUntilConnected = async (pc: RTCPeerConnection, stream
         options.onProgress?.(new CustomProgressEvent<string>('webrtc:add-ice-candidate', candidate.candidate))
         await pc.addIceCandidate(candidate)
       } catch (err) {
-        options.log.error('%s bad candidate received', options.direction, candidateInit, err)
+        options.log.error('%s bad candidate received %o - %e', options.direction, candidateInit, err)
       }
     }
   } catch (err) {
-    options.log.error('%s error parsing ICE candidate', options.direction, err)
+    options.log.error('%s error parsing ICE candidate - %e', options.direction, err)
 
-    if (options.signal?.aborted === true && getConnectionState(pc) !== 'connected') {
+    if (options.signal?.aborted === true && pc.connectionState !== 'connected') {
       throw err
     }
   }
 }
 
-export function getConnectionState (pc: RTCPeerConnection): string {
-  return isFirefox ? pc.iceConnectionState : pc.connectionState
-}
-
 function resolveOnConnected (pc: RTCPeerConnection, promise: DeferredPromise<void>): void {
-  pc[isFirefox ? 'oniceconnectionstatechange' : 'onconnectionstatechange'] = (_) => {
-    switch (getConnectionState(pc)) {
+  if (pc.connectionState === 'connected') {
+    promise.resolve()
+    return
+  }
+
+  pc.onconnectionstatechange = (_) => {
+    switch (pc.connectionState) {
       case 'connected':
         promise.resolve()
         break
       case 'failed':
       case 'disconnected':
       case 'closed':
-        promise.reject(new ConnectionFailedError('RTCPeerConnection was closed'))
+        promise.reject(new ConnectionFailedError(`RTCPeerConnection connection state became "${pc.connectionState}"`))
         break
       default:
         break

package/src/private-to-public/transport.ts

@@ -51,11 +51,6 @@ export interface WebRTCTransportDirectInit {
    */
   dataChannel?: DataChannelOptions
 
-  /**
-   * @deprecated use `certificate` instead - this option will be removed in a future release
-   */
-  certificates?: TransportCertificate[]
-
   /**
    * Use an existing TLS certificate to secure incoming connections or supply
    * settings to generate one.
@@ -68,11 +63,6 @@ export interface WebRTCTransportDirectInit {
    */
   certificate?: TransportCertificate
 
-  /**
-   * @deprecated this setting is ignored and will be removed in a future release
-   */
-  useLibjuice?: boolean
-
   /**
    * The key the certificate is stored in the datastore under
    *

package/src/rtcpeerconnection-to-conn.ts

@@ -24,6 +24,10 @@ class RTCPeerConnectionMultiaddrConnection extends AbstractMultiaddrConnection {
       if (this.peerConnection.connectionState === 'disconnected' || this.peerConnection.connectionState === 'failed' || this.peerConnection.connectionState === 'closed') {
         // nothing else to do but close the connection
         this.onTransportClosed()
+
+        // only necessary with node-datachannel
+        // https://github.com/murat-dogan/node-datachannel/issues/366#issuecomment-3228453155
+        this.peerConnection.close()
       }
     }
   }