@libp2p/webrtc 2.0.10-ab0e3980 → 2.0.10-b1024c6c

package/dist/src/stream.js

@@ -1,5 +1,5 @@
-import { AbstractStream } from '@libp2p/interface-stream-muxer/stream';
-import { CodeError } from '@libp2p/interfaces/errors';
+import { CodeError } from '@libp2p/interface/errors';
+import { AbstractStream } from '@libp2p/interface/stream-muxer/stream';
 import { logger } from '@libp2p/logger';
 import * as lengthPrefixed from 'it-length-prefixed';
 import { pushable } from 'it-pushable';
@@ -15,7 +15,7 @@ const MAX_BUFFERED_AMOUNT = 16 * 1024 * 1024;
 const BUFFERED_AMOUNT_LOW_TIMEOUT = 30 * 1000;
 // protobuf field definition overhead
 const PROTOBUF_OVERHEAD = 3;
-class WebRTCStream extends AbstractStream {
+export class WebRTCStream extends AbstractStream {
     /**
      * The data channel used to send and receive data
      */
@@ -30,6 +30,7 @@ class WebRTCStream extends AbstractStream {
      */
     incomingData;
     messageQueue;
+    maxDataSize;
     constructor(init) {
         super(init);
         this.channel = init.channel;
@@ -41,14 +42,15 @@ class WebRTCStream extends AbstractStream {
             maxBufferedAmount: init.dataChannelOptions?.maxBufferedAmount ?? MAX_BUFFERED_AMOUNT,
             maxMessageSize: init.dataChannelOptions?.maxMessageSize ?? MAX_MESSAGE_SIZE
         };
+        this.maxDataSize = init.maxDataSize;
         // set up initial state
         switch (this.channel.readyState) {
             case 'open':
                 break;
             case 'closed':
             case 'closing':
-                if (this.stat.timeline.close === undefined || this.stat.timeline.close === 0) {
-                    this.stat.timeline.close = Date.now();
+                if (this.timeline.close === undefined || this.timeline.close === 0) {
+                    this.timeline.close = Date.now();
                 }
                 break;
             case 'connecting':
@@ -60,7 +62,7 @@ class WebRTCStream extends AbstractStream {
         }
         // handle RTCDataChannel events
         this.channel.onopen = (_evt) => {
-            this.stat.timeline.open = new Date().getTime();
+            this.timeline.open = new Date().getTime();
             if (this.messageQueue != null) {
                 // send any queued messages
                 this._sendMessage(this.messageQueue)
@@ -71,7 +73,9 @@ class WebRTCStream extends AbstractStream {
             }
         };
         this.channel.onclose = (_evt) => {
-            this.close();
+            void this.close().catch(err => {
+                log.error('error closing stream after channel closed', err);
+            });
         };
         this.channel.onerror = (evt) => {
             const err = evt.error;
@@ -109,7 +113,6 @@ class WebRTCStream extends AbstractStream {
             }
             catch (err) {
                 if (err instanceof TimeoutError) {
-                    this.abort(err);
                     throw new Error('Timed out waiting for DataChannel buffer to clear');
                 }
                 throw err;
@@ -137,9 +140,15 @@ class WebRTCStream extends AbstractStream {
         }
     }
     async sendData(data) {
-        const msgbuf = Message.encode({ message: data.subarray() });
-        const sendbuf = lengthPrefixed.encode.single(msgbuf);
-        await this._sendMessage(sendbuf);
+        data = data.sublist();
+        while (data.byteLength > 0) {
+            const toSend = Math.min(data.byteLength, this.maxDataSize);
+            const buf = data.subarray(0, toSend);
+            const msgbuf = Message.encode({ message: buf });
+            const sendbuf = lengthPrefixed.encode.single(msgbuf);
+            await this._sendMessage(sendbuf);
+            data.consume(toSend);
+        }
     }
     async sendReset() {
         await this._sendFlag(Message.Flag.RESET);
@@ -159,7 +168,7 @@ class WebRTCStream extends AbstractStream {
             if (message.flag === Message.Flag.FIN) {
                 // We should expect no more data from the remote, stop reading
                 this.incomingData.end();
-                this.closeRead();
+                this.remoteCloseWrite();
             }
             if (message.flag === Message.Flag.RESET) {
                 // Stop reading and writing to the stream immediately
@@ -167,7 +176,7 @@ class WebRTCStream extends AbstractStream {
             }
             if (message.flag === Message.Flag.STOP_SENDING) {
                 // The remote has stopped reading
-                this.closeWrite();
+                this.remoteCloseRead();
             }
         }
         return message.message;
@@ -187,7 +196,8 @@ export function createStream(options) {
         maxDataSize: (dataChannelOptions?.maxMessageSize ?? MAX_MESSAGE_SIZE) - PROTOBUF_OVERHEAD,
         dataChannelOptions,
         onEnd,
-        channel
+        channel,
+        log: logger(`libp2p:mplex:stream:${direction}:${channel.id}`)
     });
 }
 //# sourceMappingURL=stream.js.map

package/dist/src/stream.js.map

@@ -1 +1 @@
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package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@libp2p/webrtc",
-  "version": "2.0.10-ab0e3980",
+  "version": "2.0.10-b1024c6c",
   "description": "A libp2p transport using WebRTC connections",
   "author": "",
   "license": "Apache-2.0 OR MIT",
@@ -41,52 +41,46 @@
     "lint": "aegir lint",
     "lint:fix": "aegir lint --fix",
     "clean": "aegir clean",
-    "dep-check": "aegir dep-check -i protons"
+    "dep-check": "aegir dep-check"
   },
   "dependencies": {
     "@chainsafe/libp2p-noise": "^12.0.1",
-    "@libp2p/interface-connection": "5.1.1-ab0e3980",
-    "@libp2p/interface-metrics": "4.0.8-ab0e3980",
-    "@libp2p/interface-peer-id": "2.0.2-ab0e3980",
-    "@libp2p/interface-registrar": "2.0.12-ab0e3980",
-    "@libp2p/interface-stream-muxer": "4.1.2-ab0e3980",
-    "@libp2p/interface-transport": "4.0.3-ab0e3980",
-    "@libp2p/interfaces": "3.3.2-ab0e3980",
-    "@libp2p/logger": "2.1.1-ab0e3980",
-    "@libp2p/peer-id": "2.0.3-ab0e3980",
+    "@libp2p/interface": "0.0.1-b1024c6c",
+    "@libp2p/interface-internal": "0.0.1-b1024c6c",
+    "@libp2p/logger": "2.1.1-b1024c6c",
+    "@libp2p/peer-id": "2.0.3-b1024c6c",
     "@multiformats/mafmt": "^12.1.2",
     "@multiformats/multiaddr": "^12.1.3",
     "abortable-iterator": "^5.0.1",
     "detect-browser": "^5.3.0",
     "it-length-prefixed": "^9.0.1",
-    "it-pb-stream": "^4.0.1",
+    "it-protobuf-stream": "^1.0.0",
     "it-pipe": "^3.0.1",
-    "it-pushable": "^3.1.3",
+    "it-pushable": "^3.2.0",
     "it-stream-types": "^2.0.1",
     "it-to-buffer": "^4.0.2",
-    "multiformats": "^11.0.2",
+    "multiformats": "^12.0.1",
     "multihashes": "^4.0.3",
     "p-defer": "^4.0.0",
     "p-event": "^6.0.0",
     "protons-runtime": "^5.0.0",
     "uint8arraylist": "^2.4.3",
-    "uint8arrays": "^4.0.3"
+    "uint8arrays": "^4.0.4"
   },
   "devDependencies": {
-    "@chainsafe/libp2p-yamux": "^4.0.1",
-    "@libp2p/interface-libp2p": "3.2.0-ab0e3980",
-    "@libp2p/interface-mocks": "12.0.1-ab0e3980",
-    "@libp2p/peer-id-factory": "2.0.3-ab0e3980",
-    "@libp2p/websockets": "6.0.3-ab0e3980",
+    "@chainsafe/libp2p-yamux": "^4.0.0",
+    "@libp2p/interface-compliance-tests": "3.0.7-b1024c6c",
+    "@libp2p/peer-id-factory": "2.0.3-b1024c6c",
+    "@libp2p/websockets": "6.0.3-b1024c6c",
     "@types/sinon": "^10.0.15",
-    "aegir": "^39.0.10",
+    "aegir": "^39.0.13",
     "delay": "^6.0.0",
     "it-length": "^3.0.2",
     "it-map": "^3.0.3",
     "it-pair": "^2.0.6",
-    "libp2p": "0.45.9-ab0e3980",
+    "libp2p": "0.45.9-b1024c6c",
     "protons": "^7.0.2",
-    "sinon": "^15.1.0",
+    "sinon": "^15.1.2",
     "sinon-ts": "^1.0.0"
   },
   "typedoc": {

package/src/error.ts

@@ -1,5 +1,5 @@
-import { CodeError } from '@libp2p/interfaces/errors'
-import type { Direction } from '@libp2p/interface-connection'
+import { CodeError } from '@libp2p/interface/errors'
+import type { Direction } from '@libp2p/interface/connection'
 
 export enum codes {
   ERR_ALREADY_ABORTED = 'ERR_ALREADY_ABORTED',

package/src/index.ts

@@ -1,7 +1,7 @@
 import { WebRTCTransport } from './private-to-private/transport.js'
 import { WebRTCDirectTransport, type WebRTCTransportDirectInit, type WebRTCDirectTransportComponents } from './private-to-public/transport.js'
 import type { WebRTCTransportComponents, WebRTCTransportInit } from './private-to-private/transport.js'
-import type { Transport } from '@libp2p/interface-transport'
+import type { Transport } from '@libp2p/interface/transport'
 
 /**
  * @param {WebRTCTransportDirectInit} init - WebRTC direct transport configuration

package/src/maconn.ts

@@ -1,8 +1,8 @@
 import { logger } from '@libp2p/logger'
 import { nopSink, nopSource } from './util.js'
-import type { MultiaddrConnection, MultiaddrConnectionTimeline } from '@libp2p/interface-connection'
-import type { CounterGroup } from '@libp2p/interface-metrics'
-import type { Multiaddr } from '@multiformats/multiaddr'
+import type { MultiaddrConnection, MultiaddrConnectionTimeline } from '@libp2p/interface/connection'
+import type { CounterGroup } from '@libp2p/interface/metrics'
+import type { AbortOptions, Multiaddr } from '@multiformats/multiaddr'
 import type { Source, Sink } from 'it-stream-types'
 
 const log = logger('libp2p:webrtc:connection')
@@ -72,14 +72,19 @@ export class WebRTCMultiaddrConnection implements MultiaddrConnection {
     }
   }
 
-  async close (err?: Error | undefined): Promise<void> {
-    if (err !== undefined) {
-      log.error('error closing connection', err)
-    }
+  async close (options?: AbortOptions): Promise<void> {
     log.trace('closing connection')
 
-    this.timeline.close = Date.now()
     this.peerConnection.close()
+    this.timeline.close = Date.now()
     this.metrics?.increment({ close: true })
   }
+
+  abort (err: Error): void {
+    log.error('closing connection due to error', err)
+
+    this.peerConnection.close()
+    this.timeline.close = Date.now()
+    this.metrics?.increment({ abort: true })
+  }
 }

package/src/muxer.ts

@@ -1,9 +1,10 @@
 import { createStream } from './stream.js'
 import { nopSink, nopSource } from './util.js'
 import type { DataChannelOpts } from './stream.js'
-import type { Stream } from '@libp2p/interface-connection'
-import type { CounterGroup } from '@libp2p/interface-metrics'
-import type { StreamMuxer, StreamMuxerFactory, StreamMuxerInit } from '@libp2p/interface-stream-muxer'
+import type { Stream } from '@libp2p/interface/connection'
+import type { CounterGroup } from '@libp2p/interface/metrics'
+import type { StreamMuxer, StreamMuxerFactory, StreamMuxerInit } from '@libp2p/interface/stream-muxer'
+import type { AbortOptions } from '@multiformats/multiaddr'
 import type { Source, Sink } from 'it-stream-types'
 import type { Uint8ArrayList } from 'uint8arraylist'
 
@@ -93,9 +94,14 @@ export class DataChannelMuxer implements StreamMuxer {
   private readonly metrics?: CounterGroup
 
   /**
-   * Close or abort all tracked streams and stop the muxer
+   * Gracefully close all tracked streams and stop the muxer
    */
-  close: (err?: Error | undefined) => void = () => { }
+  close: (options?: AbortOptions) => Promise<void> = async () => { }
+
+  /**
+   * Abort all tracked streams and stop the muxer
+   */
+  abort: (err: Error) => void = () => { }
 
   /**
    * The stream source, a no-op as the transport natively supports multiplexing

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

@@ -1,14 +1,14 @@
 import { logger } from '@libp2p/logger'
 import { abortableDuplex } from 'abortable-iterator'
-import { pbStream } from 'it-pb-stream'
+import { pbStream } from 'it-protobuf-stream'
 import pDefer, { type DeferredPromise } from 'p-defer'
 import { DataChannelMuxerFactory } from '../muxer.js'
 import { Message } from './pb/message.js'
 import { readCandidatesUntilConnected, resolveOnConnected } from './util.js'
 import type { DataChannelOpts } from '../stream.js'
-import type { Stream } from '@libp2p/interface-connection'
-import type { IncomingStreamData } from '@libp2p/interface-registrar'
-import type { StreamMuxerFactory } from '@libp2p/interface-stream-muxer'
+import type { Stream } from '@libp2p/interface/connection'
+import type { StreamMuxerFactory } from '@libp2p/interface/stream-muxer'
+import type { IncomingStreamData } from '@libp2p/interface-internal/registrar'
 
 const DEFAULT_TIMEOUT = 30 * 1000
 
@@ -28,8 +28,8 @@ export async function handleIncomingStream ({ rtcConfiguration, dataChannelOptio
   // candidate callbacks
   pc.onicecandidate = ({ candidate }) => {
     answerSentPromise.promise.then(
-      () => {
-        stream.write({
+      async () => {
+        await stream.write({
           type: Message.Type.ICE_CANDIDATE,
           data: (candidate != null) ? JSON.stringify(candidate.toJSON()) : ''
         })
@@ -64,7 +64,7 @@ export async function handleIncomingStream ({ rtcConfiguration, dataChannelOptio
     throw new Error('Failed to create answer')
   })
   // write the answer to the remote
-  stream.write({ type: Message.Type.SDP_ANSWER, data: answer.sdp })
+  await stream.write({ type: Message.Type.SDP_ANSWER, data: answer.sdp })
 
   await pc.setLocalDescription(answer).catch(err => {
     log.error('could not execute setLocalDescription', err)
@@ -107,15 +107,18 @@ export async function initiateConnection ({ rtcConfiguration, dataChannelOptions
   // setup callback to write ICE candidates to the remote
   // peer
   pc.onicecandidate = ({ candidate }) => {
-    stream.write({
+    void stream.write({
       type: Message.Type.ICE_CANDIDATE,
       data: (candidate != null) ? JSON.stringify(candidate.toJSON()) : ''
     })
+      .catch(err => {
+        log.error('error sending ICE candidate', err)
+      })
   }
   // create an offer
   const offerSdp = await pc.createOffer()
   // write the offer to the stream
-  stream.write({ type: Message.Type.SDP_OFFER, data: offerSdp.sdp })
+  await stream.write({ type: Message.Type.SDP_OFFER, data: offerSdp.sdp })
   // set offer as local description
   await pc.setLocalDescription(offerSdp).catch(err => {
     log.error('could not execute setLocalDescription', err)

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

@@ -1,7 +1,8 @@
-import { EventEmitter } from '@libp2p/interfaces/events'
+import { EventEmitter } from '@libp2p/interface/events'
 import { Circuit } from '@multiformats/mafmt'
-import type { PeerId } from '@libp2p/interface-peer-id'
-import type { ListenerEvents, Listener, TransportManager } from '@libp2p/interface-transport'
+import type { PeerId } from '@libp2p/interface/peer-id'
+import type { ListenerEvents, Listener } from '@libp2p/interface/transport'
+import type { TransportManager } from '@libp2p/interface-internal/transport-manager'
 import type { Multiaddr } from '@multiformats/multiaddr'
 
 export interface ListenerOptions {

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

@@ -1,5 +1,5 @@
-import { type CreateListenerOptions, type DialOptions, type Listener, symbol, type Transport, type Upgrader, type TransportManager } from '@libp2p/interface-transport'
-import { CodeError } from '@libp2p/interfaces/errors'
+import { CodeError } from '@libp2p/interface/errors'
+import { type CreateListenerOptions, type DialOptions, symbol, type Transport, type Listener, type Upgrader } from '@libp2p/interface/transport'
 import { logger } from '@libp2p/logger'
 import { peerIdFromString } from '@libp2p/peer-id'
 import { multiaddr, type Multiaddr, protocols } from '@multiformats/multiaddr'
@@ -8,10 +8,11 @@ import { WebRTCMultiaddrConnection } from '../maconn.js'
 import { initiateConnection, handleIncomingStream } from './handler.js'
 import { WebRTCPeerListener } from './listener.js'
 import type { DataChannelOpts } from '../stream.js'
-import type { Connection } from '@libp2p/interface-connection'
-import type { PeerId } from '@libp2p/interface-peer-id'
-import type { IncomingStreamData, Registrar } from '@libp2p/interface-registrar'
-import type { Startable } from '@libp2p/interfaces/startable'
+import type { Connection } from '@libp2p/interface/connection'
+import type { PeerId } from '@libp2p/interface/peer-id'
+import type { Startable } from '@libp2p/interface/startable'
+import type { IncomingStreamData, Registrar } from '@libp2p/interface-internal/registrar'
+import type { TransportManager } from '@libp2p/interface-internal/transport-manager'
 
 const log = logger('libp2p:webrtc:peer')
 
@@ -113,11 +114,11 @@ export class WebRTCTransport implements Transport, Startable {
       )
 
       // close the stream if SDP has been exchanged successfully
-      signalingStream.close()
+      await signalingStream.close()
       return result
-    } catch (err) {
+    } catch (err: any) {
       // reset the stream in case of any error
-      signalingStream.reset()
+      signalingStream.abort(err)
       throw err
     } finally {
       // Close the signaling connection
@@ -143,8 +144,8 @@ export class WebRTCTransport implements Transport, Startable {
         skipProtection: true,
         muxerFactory
       })
-    } catch (err) {
-      stream.reset()
+    } catch (err: any) {
+      stream.abort(err)
       throw err
     } finally {
       // Close the signaling connection

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

@@ -1,4 +1,4 @@
-import type { CreateListenerOptions, DialOptions } from '@libp2p/interface-transport'
+import type { CreateListenerOptions, DialOptions } from '@libp2p/interface/transport'
 
 export interface WebRTCListenerOptions extends CreateListenerOptions {}
 export interface WebRTCDialOptions extends DialOptions {}

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

@@ -1,5 +1,5 @@
 import { noise as Noise } from '@chainsafe/libp2p-noise'
-import { type CreateListenerOptions, type Listener, symbol, type Transport } from '@libp2p/interface-transport'
+import { type CreateListenerOptions, symbol, type Transport, type Listener } from '@libp2p/interface/transport'
 import { logger } from '@libp2p/logger'
 import * as p from '@libp2p/peer-id'
 import { protocols } from '@multiformats/multiaddr'
@@ -15,9 +15,9 @@ import * as sdp from './sdp.js'
 import { genUfrag } from './util.js'
 import type { WebRTCDialOptions } from './options.js'
 import type { DataChannelOpts } from '../stream.js'
-import type { Connection } from '@libp2p/interface-connection'
-import type { CounterGroup, Metrics } from '@libp2p/interface-metrics'
-import type { PeerId } from '@libp2p/interface-peer-id'
+import type { Connection } from '@libp2p/interface/connection'
+import type { CounterGroup, Metrics } from '@libp2p/interface/metrics'
+import type { PeerId } from '@libp2p/interface/peer-id'
 import type { Multiaddr } from '@multiformats/multiaddr'
 
 const log = logger('libp2p:webrtc:transport')

package/src/stream.ts

@@ -1,12 +1,12 @@
-import { AbstractStream, type AbstractStreamInit } from '@libp2p/interface-stream-muxer/stream'
-import { CodeError } from '@libp2p/interfaces/errors'
+import { CodeError } from '@libp2p/interface/errors'
+import { AbstractStream, type AbstractStreamInit } from '@libp2p/interface/stream-muxer/stream'
 import { logger } from '@libp2p/logger'
 import * as lengthPrefixed from 'it-length-prefixed'
 import { type Pushable, pushable } from 'it-pushable'
 import { pEvent, TimeoutError } from 'p-event'
 import { Uint8ArrayList } from 'uint8arraylist'
 import { Message } from './pb/message.js'
-import type { Direction, Stream } from '@libp2p/interface-connection'
+import type { Direction } from '@libp2p/interface/connection'
 
 const log = logger('libp2p:webrtc:stream')
 
@@ -26,6 +26,8 @@ export interface WebRTCStreamInit extends AbstractStreamInit {
   channel: RTCDataChannel
 
   dataChannelOptions?: Partial<DataChannelOpts>
+
+  maxDataSize: number
 }
 
 // Max message size that can be sent to the DataChannel
@@ -40,7 +42,7 @@ const BUFFERED_AMOUNT_LOW_TIMEOUT = 30 * 1000
 // protobuf field definition overhead
 const PROTOBUF_OVERHEAD = 3
 
-class WebRTCStream extends AbstractStream {
+export class WebRTCStream extends AbstractStream {
   /**
    * The data channel used to send and receive data
    */
@@ -58,6 +60,7 @@ class WebRTCStream extends AbstractStream {
   private readonly incomingData: Pushable<Uint8Array>
 
   private messageQueue?: Uint8ArrayList
+  private readonly maxDataSize: number
 
   constructor (init: WebRTCStreamInit) {
     super(init)
@@ -71,6 +74,7 @@ class WebRTCStream extends AbstractStream {
       maxBufferedAmount: init.dataChannelOptions?.maxBufferedAmount ?? MAX_BUFFERED_AMOUNT,
       maxMessageSize: init.dataChannelOptions?.maxMessageSize ?? MAX_MESSAGE_SIZE
     }
+    this.maxDataSize = init.maxDataSize
 
     // set up initial state
     switch (this.channel.readyState) {
@@ -79,8 +83,8 @@ class WebRTCStream extends AbstractStream {
 
       case 'closed':
       case 'closing':
-        if (this.stat.timeline.close === undefined || this.stat.timeline.close === 0) {
-          this.stat.timeline.close = Date.now()
+        if (this.timeline.close === undefined || this.timeline.close === 0) {
+          this.timeline.close = Date.now()
         }
         break
       case 'connecting':
@@ -94,7 +98,7 @@ class WebRTCStream extends AbstractStream {
 
     // handle RTCDataChannel events
     this.channel.onopen = (_evt) => {
-      this.stat.timeline.open = new Date().getTime()
+      this.timeline.open = new Date().getTime()
 
       if (this.messageQueue != null) {
         // send any queued messages
@@ -107,7 +111,9 @@ class WebRTCStream extends AbstractStream {
     }
 
     this.channel.onclose = (_evt) => {
-      this.close()
+      void this.close().catch(err => {
+        log.error('error closing stream after channel closed', err)
+      })
     }
 
     this.channel.onerror = (evt) => {
@@ -153,7 +159,6 @@ class WebRTCStream extends AbstractStream {
         await pEvent(this.channel, 'bufferedamountlow', { timeout: this.dataChannelOptions.bufferedAmountLowEventTimeout })
       } catch (err: any) {
         if (err instanceof TimeoutError) {
-          this.abort(err)
           throw new Error('Timed out waiting for DataChannel buffer to clear')
         }
 
@@ -184,10 +189,17 @@ class WebRTCStream extends AbstractStream {
   }
 
   async sendData (data: Uint8ArrayList): Promise<void> {
-    const msgbuf = Message.encode({ message: data.subarray() })
-    const sendbuf = lengthPrefixed.encode.single(msgbuf)
+    data = data.sublist()
 
-    await this._sendMessage(sendbuf)
+    while (data.byteLength > 0) {
+      const toSend = Math.min(data.byteLength, this.maxDataSize)
+      const buf = data.subarray(0, toSend)
+      const msgbuf = Message.encode({ message: buf })
+      const sendbuf = lengthPrefixed.encode.single(msgbuf)
+      await this._sendMessage(sendbuf)
+
+      data.consume(toSend)
+    }
   }
 
   async sendReset (): Promise<void> {
@@ -212,7 +224,7 @@ class WebRTCStream extends AbstractStream {
       if (message.flag === Message.Flag.FIN) {
         // We should expect no more data from the remote, stop reading
         this.incomingData.end()
-        this.closeRead()
+        this.remoteCloseWrite()
       }
 
       if (message.flag === Message.Flag.RESET) {
@@ -222,7 +234,7 @@ class WebRTCStream extends AbstractStream {
 
       if (message.flag === Message.Flag.STOP_SENDING) {
         // The remote has stopped reading
-        this.closeWrite()
+        this.remoteCloseRead()
       }
     }
 
@@ -259,7 +271,7 @@ export interface WebRTCStreamOptions {
   onEnd?: (err?: Error | undefined) => void
 }
 
-export function createStream (options: WebRTCStreamOptions): Stream {
+export function createStream (options: WebRTCStreamOptions): WebRTCStream {
   const { channel, direction, onEnd, dataChannelOptions } = options
 
   return new WebRTCStream({
@@ -268,6 +280,7 @@ export function createStream (options: WebRTCStreamOptions): Stream {
     maxDataSize: (dataChannelOptions?.maxMessageSize ?? MAX_MESSAGE_SIZE) - PROTOBUF_OVERHEAD,
     dataChannelOptions,
     onEnd,
-    channel
+    channel,
+    log: logger(`libp2p:mplex:stream:${direction}:${channel.id}`)
   })
 }