@libp2p/webrtc 5.2.3-6f8cfeafb → 5.2.3-90cca822b

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

@@ -2,14 +2,13 @@ import { serviceCapabilities, transportSymbol } from '@libp2p/interface'
 import { peerIdFromString } from '@libp2p/peer-id'
 import { WebRTCDirect } from '@multiformats/multiaddr-matcher'
 import { raceSignal } from 'race-signal'
-import { HANDSHAKE_TIMEOUT_MS } from '../constants.js'
 import { genUfrag } from '../util.js'
 import { WebRTCDirectListener } from './listener.js'
 import { connect } from './utils/connect.js'
 import { createDialerRTCPeerConnection } from './utils/get-rtcpeerconnection.js'
 import type { DataChannelOptions, TransportCertificate } from '../index.js'
 import type { WebRTCDialEvents } from '../private-to-private/transport.js'
-import type { CreateListenerOptions, Transport, Listener, ComponentLogger, Logger, Connection, CounterGroup, Metrics, PeerId, DialTransportOptions, PrivateKey } from '@libp2p/interface'
+import type { CreateListenerOptions, Transport, Listener, ComponentLogger, Logger, Connection, CounterGroup, Metrics, PeerId, DialTransportOptions, PrivateKey, Upgrader } from '@libp2p/interface'
 import type { TransportManager } from '@libp2p/interface-internal'
 import type { Multiaddr } from '@multiformats/multiaddr'
 
@@ -22,6 +21,7 @@ export interface WebRTCDirectTransportComponents {
   metrics?: Metrics
   logger: ComponentLogger
   transportManager: TransportManager
+  upgrader: Upgrader
 }
 
 export interface WebRTCMetrics {
@@ -72,7 +72,6 @@ export class WebRTCDirectTransport implements Transport {
    * Dial a given multiaddr
    */
   async dial (ma: Multiaddr, options: DialTransportOptions<WebRTCDialEvents>): Promise<Connection> {
-    options?.signal?.throwIfAborted()
     const rawConn = await this._connect(ma, options)
     this.log('dialing address: %a', ma)
     return rawConn
@@ -106,6 +105,9 @@ export class WebRTCDirectTransport implements Transport {
    * Connect to a peer using a multiaddr
    */
   async _connect (ma: Multiaddr, options: DialTransportOptions<WebRTCDialEvents>): Promise<Connection> {
+    // do not create RTCPeerConnection if the signal has already been aborted
+    options.signal.throwIfAborted()
+
     let theirPeerId: PeerId | undefined
     const remotePeerString = ma.getPeerId()
     if (remotePeerString != null) {
@@ -124,7 +126,7 @@ export class WebRTCDirectTransport implements Transport {
         logger: this.components.logger,
         metrics: this.components.metrics,
         events: this.metrics?.dialerEvents,
-        signal: options.signal ?? AbortSignal.timeout(HANDSHAKE_TIMEOUT_MS),
+        signal: options.signal,
         remoteAddr: ma,
         dataChannel: this.init.dataChannel,
         upgrader: options.upgrader,