@libp2p/webrtc 5.2.24-a02cb0461 → 5.2.24-f5932c294

package/src/stream.ts

@@ -1,15 +1,20 @@
-import { StreamStateError } from '@libp2p/interface'
-import { AbstractStream } from '@libp2p/utils'
+import { StreamStateError, TimeoutError } from '@libp2p/interface'
+import { AbstractStream } from '@libp2p/utils/abstract-stream'
+import { anySignal } from 'any-signal'
 import * as lengthPrefixed from 'it-length-prefixed'
 import { pushable } from 'it-pushable'
+import pDefer from 'p-defer'
+import pTimeout from 'p-timeout'
+import { raceEvent } from 'race-event'
 import { raceSignal } from 'race-signal'
 import { Uint8ArrayList } from 'uint8arraylist'
-import { MAX_BUFFERED_AMOUNT, MAX_MESSAGE_SIZE, PROTOBUF_OVERHEAD } from './constants.js'
+import { BUFFERED_AMOUNT_LOW_TIMEOUT, FIN_ACK_TIMEOUT, MAX_BUFFERED_AMOUNT, MAX_MESSAGE_SIZE, OPEN_TIMEOUT, PROTOBUF_OVERHEAD } from './constants.js'
 import { Message } from './private-to-public/pb/message.js'
 import type { DataChannelOptions } from './index.js'
-import type { AbortOptions, MessageStreamDirection, Logger } from '@libp2p/interface'
-import type { AbstractStreamInit, SendResult } from '@libp2p/utils'
+import type { AbortOptions, Direction, Logger } from '@libp2p/interface'
+import type { AbstractStreamInit } from '@libp2p/utils/abstract-stream'
 import type { Pushable } from 'it-pushable'
+import type { DeferredPromise } from 'p-defer'
 
 export interface WebRTCStreamInit extends AbstractStreamInit, DataChannelOptions {
   /**
@@ -34,24 +39,76 @@ export class WebRTCStream extends AbstractStream {
    * and then the protobuf decoder.
    */
   private readonly incomingData: Pushable<Uint8Array>
+
   private readonly maxBufferedAmount: number
-  private readonly receivedFinAck: PromiseWithResolvers<void>
+
+  private readonly bufferedAmountLowEventTimeout: number
+
+  /**
+   * The maximum size of a message in bytes
+   */
+  private readonly maxMessageSize: number
+
+  /**
+   * When this promise is resolved, the remote has sent us a FIN flag
+   */
+  private readonly receiveFinAck: DeferredPromise<void>
+  private readonly finAckTimeout: number
+  private readonly openTimeout: number
+  private readonly closeController: AbortController
 
   constructor (init: WebRTCStreamInit) {
-    super({
-      ...init,
-      maxMessageSize: (init.maxMessageSize ?? MAX_MESSAGE_SIZE) - PROTOBUF_OVERHEAD
-    })
+    // override onEnd to send/receive FIN_ACK before closing the stream
+    const originalOnEnd = init.onEnd
+    init.onEnd = (err?: Error): void => {
+      this.log.trace('readable and writeable ends closed with status "%s"', this.status)
+
+      void Promise.resolve(async () => {
+        if (this.timeline.abort != null || this.timeline.reset !== null) {
+          return
+        }
+
+        // wait for FIN_ACK if we haven't received it already
+        try {
+          await pTimeout(this.receiveFinAck.promise, {
+            milliseconds: this.finAckTimeout
+          })
+        } catch (err) {
+          this.log.error('error receiving FIN_ACK', err)
+        }
+      })
+        .then(() => {
+          // stop processing incoming messages
+          this.incomingData.end()
+
+          // final cleanup
+          originalOnEnd?.(err)
+        })
+        .catch(err => {
+          this.log.error('error ending stream', err)
+        })
+        .finally(() => {
+          this.channel.close()
+        })
+    }
+
+    super(init)
 
     this.channel = init.channel
     this.channel.binaryType = 'arraybuffer'
     this.incomingData = pushable<Uint8Array>()
+    this.bufferedAmountLowEventTimeout = init.bufferedAmountLowEventTimeout ?? BUFFERED_AMOUNT_LOW_TIMEOUT
     this.maxBufferedAmount = init.maxBufferedAmount ?? MAX_BUFFERED_AMOUNT
-    this.receivedFinAck = Promise.withResolvers()
+    this.maxMessageSize = (init.maxMessageSize ?? MAX_MESSAGE_SIZE) - PROTOBUF_OVERHEAD
+    this.receiveFinAck = pDefer()
+    this.finAckTimeout = init.closeTimeout ?? FIN_ACK_TIMEOUT
+    this.openTimeout = init.openTimeout ?? OPEN_TIMEOUT
+    this.closeController = new AbortController()
 
     // set up initial state
     switch (this.channel.readyState) {
       case 'open':
+        this.timeline.open = new Date().getTime()
         break
 
       case 'closed':
@@ -70,16 +127,31 @@ export class WebRTCStream extends AbstractStream {
     }
 
     // handle RTCDataChannel events
+    this.channel.onopen = (_evt) => {
+      this.timeline.open = new Date().getTime()
+    }
+
     this.channel.onclose = (_evt) => {
       this.log.trace('received onclose event')
 
-      this.onRemoteCloseWrite()
-      this.onTransportClosed()
+      // stop any in-progress writes
+      this.closeController.abort()
+
+      // if the channel has closed we'll never receive a FIN_ACK so resolve the
+      // promise so we don't try to wait later
+      this.receiveFinAck.resolve()
+
+      void this.close().catch(err => {
+        this.log.error('error closing stream after channel closed', err)
+      })
     }
 
     this.channel.onerror = (evt) => {
       this.log.trace('received onerror event')
 
+      // stop any in-progress writes
+      this.closeController.abort()
+
       const err = (evt as RTCErrorEvent).error
       this.abort(err)
     }
@@ -94,12 +166,6 @@ export class WebRTCStream extends AbstractStream {
       this.incomingData.push(new Uint8Array(data, 0, data.byteLength))
     }
 
-    // dispatch drain event when the buffered amount drops to zero
-    this.channel.bufferedAmountLowThreshold = 0
-    this.channel.onbufferedamountlow = () => {
-      this.safeDispatchEvent('drain')
-    }
-
     const self = this
 
     // pipe framed protobuf messages through a length prefixed decoder, and
@@ -109,59 +175,94 @@ export class WebRTCStream extends AbstractStream {
         const message = self.processIncomingProtobuf(buf)
 
         if (message != null) {
-          self.onData(new Uint8ArrayList(message))
+          self.sourcePush(new Uint8ArrayList(message))
         }
       }
     })
       .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 = (): void => {
-      this.channel.close()
-    }
-    this.addEventListener('close', webRTCStreamOnClose)
   }
 
   sendNewStream (): void {
     // opening new streams is handled by WebRTC so this is a noop
   }
 
-  _sendMessage (data: Uint8ArrayList): void {
+  async _sendMessage (data: Uint8ArrayList, checkBuffer: boolean = true): Promise<void> {
     if (this.channel.readyState === 'closed' || this.channel.readyState === 'closing') {
       throw new StreamStateError(`Invalid datachannel state - ${this.channel.readyState}`)
     }
 
+    if (this.channel.readyState !== 'open') {
+      const timeout = AbortSignal.timeout(this.openTimeout)
+      const signal = anySignal([
+        this.closeController.signal,
+        timeout
+      ])
+
+      try {
+        this.log('channel state is "%s" and not "open", waiting for "open" event before sending data', this.channel.readyState)
+        await raceEvent(this.channel, 'open', signal)
+      } finally {
+        signal.clear()
+      }
+
+      this.log('channel state is now "%s", sending data', this.channel.readyState)
+    }
+
+    if (checkBuffer && this.channel.bufferedAmount > this.maxBufferedAmount) {
+      const timeout = AbortSignal.timeout(this.bufferedAmountLowEventTimeout)
+      const signal = anySignal([
+        this.closeController.signal,
+        timeout
+      ])
+
+      try {
+        this.log('channel buffer is %d, wait for "bufferedamountlow" event', this.channel.bufferedAmount)
+        await raceEvent(this.channel, 'bufferedamountlow', signal)
+      } catch (err: any) {
+        if (timeout.aborted) {
+          throw new TimeoutError(`Timed out waiting for DataChannel buffer to clear after ${this.bufferedAmountLowEventTimeout}ms`)
+        }
+
+        throw err
+      } finally {
+        signal.clear()
+      }
+    }
+
     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.channel.send(data.subarray())
     } catch (err: any) {
       this.log.error('error while sending message', err)
     }
   }
 
-  sendData (data: Uint8ArrayList): SendResult {
-    const messageBuf = Message.encode({
-      message: data.subarray()
-    })
-    const prefixedBuf = lengthPrefixed.encode.single(messageBuf)
-    this._sendMessage(prefixedBuf)
-
-    return {
-      sentBytes: data.byteLength,
-      canSendMore: this.channel.bufferedAmount < this.maxBufferedAmount
+  async sendData (data: Uint8ArrayList): Promise<void> {
+    const bytesTotal = data.byteLength
+    // sending messages is an async operation so use a copy of the list as it
+    // may be changed beneath us
+    data = data.sublist()
+
+    while (data.byteLength > 0) {
+      const toSend = Math.min(data.byteLength, this.maxMessageSize)
+      const buf = data.subarray(0, toSend)
+      const messageBuf = Message.encode({ message: buf })
+      const sendBuf = lengthPrefixed.encode.single(messageBuf)
+      this.log.trace('sending %d/%d bytes on channel', buf.byteLength, bytesTotal)
+      await this._sendMessage(sendBuf)
+
+      data.consume(toSend)
     }
-  }
 
-  sendReset (): void {
-    this.receivedFinAck.resolve()
+    this.log.trace('finished sending data, channel state "%s"', this.channel.readyState)
+  }
 
+  async sendReset (): Promise<void> {
     try {
-      this._sendFlag(Message.Flag.RESET)
+      await this._sendFlag(Message.Flag.RESET)
     } catch (err) {
       this.log.error('failed to send reset - %e', err)
     } finally {
@@ -169,20 +270,38 @@ export class WebRTCStream extends AbstractStream {
     }
   }
 
-  async sendCloseWrite (options?: AbortOptions): Promise<void> {
-    if (this.channel.readyState === 'open') {
-      this._sendFlag(Message.Flag.FIN)
+  async sendCloseWrite (options: AbortOptions): Promise<void> {
+    if (this.channel.readyState !== 'open') {
+      this.receiveFinAck.resolve()
+      return
+    }
+
+    const sent = await this._sendFlag(Message.Flag.FIN)
+
+    if (sent) {
+      this.log.trace('awaiting FIN_ACK')
+      try {
+        await raceSignal(this.receiveFinAck.promise, options?.signal, {
+          errorMessage: 'sending close-write was aborted before FIN_ACK was received',
+          errorName: 'FinAckNotReceivedError'
+        })
+      } catch (err) {
+        this.log.error('failed to await FIN_ACK', err)
+      }
+    } else {
+      this.log.trace('sending FIN failed, not awaiting FIN_ACK')
     }
 
-    await raceSignal(this.receivedFinAck.promise, options?.signal)
+    // if we've attempted to receive a FIN_ACK, do not try again
+    this.receiveFinAck.resolve()
   }
 
-  async sendCloseRead (options?: AbortOptions): Promise<void> {
-    if (this.channel.readyState === 'open') {
-      this._sendFlag(Message.Flag.STOP_SENDING)
+  async sendCloseRead (): Promise<void> {
+    if (this.channel.readyState !== 'open') {
+      return
     }
 
-    options?.signal?.throwIfAborted()
+    await this._sendFlag(Message.Flag.STOP_SENDING)
   }
 
   /**
@@ -196,33 +315,38 @@ export class WebRTCStream extends AbstractStream {
 
       if (message.flag === Message.Flag.FIN) {
         // We should expect no more data from the remote, stop reading
-        this.onRemoteCloseWrite()
-        this._sendFlag(Message.Flag.FIN_ACK)
+        this.remoteCloseWrite()
+
+        this.log.trace('sending FIN_ACK')
+        void this._sendFlag(Message.Flag.FIN_ACK)
+          .catch(err => {
+            this.log.error('error sending FIN_ACK immediately', err)
+          })
       }
 
       if (message.flag === Message.Flag.RESET) {
-        this.receivedFinAck.resolve()
         // Stop reading and writing to the stream immediately
-        this.onRemoteReset()
+        this.reset()
       }
 
       if (message.flag === Message.Flag.STOP_SENDING) {
         // The remote has stopped reading
-        this.onRemoteCloseRead()
+        this.remoteCloseRead()
       }
 
       if (message.flag === Message.Flag.FIN_ACK) {
-        this.receivedFinAck.resolve()
+        this.log.trace('received FIN_ACK')
+        this.receiveFinAck.resolve()
       }
     }
 
     // ignore data messages if we've closed the readable end already
-    if (this.readStatus === 'readable' || this.readStatus === 'paused') {
+    if (this.readStatus === 'ready') {
       return message.message
     }
   }
 
-  private _sendFlag (flag: Message.Flag): boolean {
+  private async _sendFlag (flag: Message.Flag): Promise<boolean> {
     if (this.channel.readyState !== 'open') {
       // flags can be sent while we or the remote are closing the datachannel so
       // if the channel isn't open, don't try to send it but return false to let
@@ -236,7 +360,7 @@ export class WebRTCStream extends AbstractStream {
     const prefixedBuf = lengthPrefixed.encode.single(messageBuf)
 
     try {
-      this._sendMessage(prefixedBuf)
+      await this._sendMessage(prefixedBuf, false)
 
       return true
     } catch (err: any) {
@@ -245,14 +369,6 @@ export class WebRTCStream extends AbstractStream {
 
     return false
   }
-
-  sendPause (): void {
-    // TODO: read backpressure?
-  }
-
-  sendResume (): void {
-    // TODO: read backpressure?
-  }
 }
 
 export interface WebRTCStreamOptions extends DataChannelOptions {
@@ -267,7 +383,12 @@ export interface WebRTCStreamOptions extends DataChannelOptions {
   /**
    * The stream direction
    */
-  direction: MessageStreamDirection
+  direction: Direction
+
+  /**
+   * A callback invoked when the channel ends
+   */
+  onEnd?(err?: Error | undefined): void
 
   /**
    * The logger to create a scope from
@@ -278,16 +399,15 @@ export interface WebRTCStreamOptions extends DataChannelOptions {
    * If true the underlying datachannel is being used to perform the noise
    * handshake during connection establishment
    */
-  isHandshake?: boolean
+  handshake?: boolean
 }
 
 export function createStream (options: WebRTCStreamOptions): WebRTCStream {
-  const { channel, direction, isHandshake } = options
+  const { channel, direction, handshake } = options
 
   return new WebRTCStream({
     ...options,
     id: `${channel.id}`,
-    log: options.log.newScope(`${isHandshake === true ? 'handshake' : direction}:${channel.id}`),
-    protocol: ''
+    log: options.log.newScope(`${handshake === true ? 'handshake' : direction}:${channel.id}`)
   })
 }

package/src/util.ts

@@ -2,9 +2,8 @@ import { detect } from 'detect-browser'
 import pDefer from 'p-defer'
 import pTimeout from 'p-timeout'
 import { DATA_CHANNEL_DRAIN_TIMEOUT, DEFAULT_ICE_SERVERS, UFRAG_ALPHABET, UFRAG_PREFIX } from './constants.js'
+import type { PeerConnection } from '@ipshipyard/node-datachannel'
 import type { LoggerOptions } from '@libp2p/interface'
-import type { Duplex, Source } from 'it-stream-types'
-import type { PeerConnection } from 'node-datachannel'
 
 const browser = detect()
 export const isFirefox = ((browser != null) && browser.name === 'firefox')
@@ -13,26 +12,6 @@ export const nopSource = async function * nop (): AsyncGenerator<Uint8Array, any
 
 export const nopSink = async (_: any): Promise<void> => {}
 
-// Duplex that does nothing. Needed to fulfill the interface
-export function inertDuplex (): Duplex<any, any, any> {
-  return {
-    source: {
-      [Symbol.asyncIterator] () {
-        return {
-          async next () {
-            // This will never resolve
-            return new Promise(() => { })
-          }
-        }
-      }
-    },
-    sink: async (source: Source<any>) => {
-      // This will never resolve
-      return new Promise(() => { })
-    }
-  }
-}
-
 export function drainAndClose (channel: RTCDataChannel, direction: string, drainTimeout: number = DATA_CHANNEL_DRAIN_TIMEOUT, options: LoggerOptions): void {
   if (channel.readyState !== 'open') {
     return

package/src/webrtc/index.ts

@@ -1 +1 @@
-export { RTCSessionDescription, RTCIceCandidate, RTCPeerConnection } from 'node-datachannel/polyfill'
+export { RTCSessionDescription, RTCIceCandidate, RTCPeerConnection } from '@ipshipyard/node-datachannel/polyfill'

package/dist/src/rtcpeerconnection-to-conn.d.ts

@@ -1,12 +0,0 @@
-import type { RTCPeerConnection } from './webrtc/index.js';
-import type { MultiaddrConnection } from '@libp2p/interface';
-import type { AbstractMultiaddrConnectionInit } from '@libp2p/utils';
-export interface RTCPeerConnectionMultiaddrConnectionInit extends Omit<AbstractMultiaddrConnectionInit, 'stream'> {
-    peerConnection: RTCPeerConnection;
-}
-/**
- * Convert a RTCPeerConnection into a MultiaddrConnection
- * https://github.com/libp2p/interface-transport#multiaddrconnection
- */
-export declare const toMultiaddrConnection: (init: RTCPeerConnectionMultiaddrConnectionInit) => MultiaddrConnection;
-//# sourceMappingURL=rtcpeerconnection-to-conn.d.ts.map

package/dist/src/rtcpeerconnection-to-conn.d.ts.map

@@ -1 +0,0 @@
-{"version":3,"file":"rtcpeerconnection-to-conn.d.ts","sourceRoot":"","sources":["../../src/rtcpeerconnection-to-conn.ts"],"names":[],"mappings":"AACA,OAAO,KAAK,EAAE,iBAAiB,EAAE,MAAM,mBAAmB,CAAA;AAC1D,OAAO,KAAK,EAAgB,mBAAmB,EAAE,MAAM,mBAAmB,CAAA;AAC1E,OAAO,KAAK,EAAE,+BAA+B,EAAc,MAAM,eAAe,CAAA;AAGhF,MAAM,WAAW,wCAAyC,SAAQ,IAAI,CAAC,+BAA+B,EAAE,QAAQ,CAAC;IAC/G,cAAc,EAAE,iBAAiB,CAAA;CAClC;AA+CD;;;GAGG;AACH,eAAO,MAAM,qBAAqB,GAAI,MAAM,wCAAwC,KAAG,mBAEtF,CAAA"}

package/dist/src/rtcpeerconnection-to-conn.js

@@ -1,43 +0,0 @@
-import { AbstractMultiaddrConnection } from '@libp2p/utils';
-class RTCPeerConnectionMultiaddrConnection extends AbstractMultiaddrConnection {
-    peerConnection;
-    constructor(init) {
-        super(init);
-        this.peerConnection = init.peerConnection;
-        const initialState = init.peerConnection.connectionState;
-        this.peerConnection.onconnectionstatechange = () => {
-            this.log.trace('peer connection state change %s initial state %s', this.peerConnection.connectionState, initialState);
-            if (this.peerConnection.connectionState === 'disconnected' || this.peerConnection.connectionState === 'failed' || this.peerConnection.connectionState === 'closed') {
-                // nothing else to do but close the connection
-                this.onTransportClosed();
-            }
-        };
-    }
-    sendData(data) {
-        return {
-            sentBytes: data.byteLength,
-            canSendMore: true
-        };
-    }
-    async sendClose(options) {
-        this.peerConnection.close();
-        options?.signal?.throwIfAborted();
-    }
-    sendReset() {
-        this.peerConnection.close();
-    }
-    sendPause() {
-        // TODO: readable backpressure?
-    }
-    sendResume() {
-        // TODO: readable backpressure?
-    }
-}
-/**
- * Convert a RTCPeerConnection into a MultiaddrConnection
- * https://github.com/libp2p/interface-transport#multiaddrconnection
- */
-export const toMultiaddrConnection = (init) => {
-    return new RTCPeerConnectionMultiaddrConnection(init);
-};
-//# sourceMappingURL=rtcpeerconnection-to-conn.js.map

package/dist/src/rtcpeerconnection-to-conn.js.map

@@ -1 +0,0 @@
-{"version":3,"file":"rtcpeerconnection-to-conn.js","sourceRoot":"","sources":["../../src/rtcpeerconnection-to-conn.ts"],"names":[],"mappings":"AAAA,OAAO,EAAE,2BAA2B,EAAE,MAAM,eAAe,CAAA;AAU3D,MAAM,oCAAqC,SAAQ,2BAA2B;IACpE,cAAc,CAAmB;IAEzC,YAAa,IAA8C;QACzD,KAAK,CAAC,IAAI,CAAC,CAAA;QAEX,IAAI,CAAC,cAAc,GAAG,IAAI,CAAC,cAAc,CAAA;QAEzC,MAAM,YAAY,GAAG,IAAI,CAAC,cAAc,CAAC,eAAe,CAAA;QAExD,IAAI,CAAC,cAAc,CAAC,uBAAuB,GAAG,GAAG,EAAE;YACjD,IAAI,CAAC,GAAG,CAAC,KAAK,CAAC,kDAAkD,EAAE,IAAI,CAAC,cAAc,CAAC,eAAe,EAAE,YAAY,CAAC,CAAA;YAErH,IAAI,IAAI,CAAC,cAAc,CAAC,eAAe,KAAK,cAAc,IAAI,IAAI,CAAC,cAAc,CAAC,eAAe,KAAK,QAAQ,IAAI,IAAI,CAAC,cAAc,CAAC,eAAe,KAAK,QAAQ,EAAE,CAAC;gBACnK,8CAA8C;gBAC9C,IAAI,CAAC,iBAAiB,EAAE,CAAA;YAC1B,CAAC;QACH,CAAC,CAAA;IACH,CAAC;IAED,QAAQ,CAAE,IAAoB;QAC5B,OAAO;YACL,SAAS,EAAE,IAAI,CAAC,UAAU;YAC1B,WAAW,EAAE,IAAI;SAClB,CAAA;IACH,CAAC;IAED,KAAK,CAAC,SAAS,CAAE,OAAsB;QACrC,IAAI,CAAC,cAAc,CAAC,KAAK,EAAE,CAAA;QAC3B,OAAO,EAAE,MAAM,EAAE,cAAc,EAAE,CAAA;IACnC,CAAC;IAED,SAAS;QACP,IAAI,CAAC,cAAc,CAAC,KAAK,EAAE,CAAA;IAC7B,CAAC;IAED,SAAS;QACP,+BAA+B;IACjC,CAAC;IAED,UAAU;QACR,+BAA+B;IACjC,CAAC;CACF;AAED;;;GAGG;AACH,MAAM,CAAC,MAAM,qBAAqB,GAAG,CAAC,IAA8C,EAAuB,EAAE;IAC3G,OAAO,IAAI,oCAAoC,CAAC,IAAI,CAAC,CAAA;AACvD,CAAC,CAAA"}

package/src/rtcpeerconnection-to-conn.ts

@@ -1,62 +0,0 @@
-import { AbstractMultiaddrConnection } from '@libp2p/utils'
-import type { RTCPeerConnection } from './webrtc/index.js'
-import type { AbortOptions, MultiaddrConnection } from '@libp2p/interface'
-import type { AbstractMultiaddrConnectionInit, SendResult } from '@libp2p/utils'
-import type { Uint8ArrayList } from 'uint8arraylist'
-
-export interface RTCPeerConnectionMultiaddrConnectionInit extends Omit<AbstractMultiaddrConnectionInit, 'stream'> {
-  peerConnection: RTCPeerConnection
-}
-
-class RTCPeerConnectionMultiaddrConnection extends AbstractMultiaddrConnection {
-  private peerConnection: RTCPeerConnection
-
-  constructor (init: RTCPeerConnectionMultiaddrConnectionInit) {
-    super(init)
-
-    this.peerConnection = init.peerConnection
-
-    const initialState = init.peerConnection.connectionState
-
-    this.peerConnection.onconnectionstatechange = () => {
-      this.log.trace('peer connection state change %s initial state %s', this.peerConnection.connectionState, initialState)
-
-      if (this.peerConnection.connectionState === 'disconnected' || this.peerConnection.connectionState === 'failed' || this.peerConnection.connectionState === 'closed') {
-        // nothing else to do but close the connection
-        this.onTransportClosed()
-      }
-    }
-  }
-
-  sendData (data: Uint8ArrayList): SendResult {
-    return {
-      sentBytes: data.byteLength,
-      canSendMore: true
-    }
-  }
-
-  async sendClose (options?: AbortOptions): Promise<void> {
-    this.peerConnection.close()
-    options?.signal?.throwIfAborted()
-  }
-
-  sendReset (): void {
-    this.peerConnection.close()
-  }
-
-  sendPause (): void {
-    // TODO: readable backpressure?
-  }
-
-  sendResume (): void {
-    // TODO: readable backpressure?
-  }
-}
-
-/**
- * Convert a RTCPeerConnection into a MultiaddrConnection
- * https://github.com/libp2p/interface-transport#multiaddrconnection
- */
-export const toMultiaddrConnection = (init: RTCPeerConnectionMultiaddrConnectionInit): MultiaddrConnection => {
-  return new RTCPeerConnectionMultiaddrConnection(init)
-}