diff --git a/packages/transport-webrtc/src/muxer.ts b/packages/transport-webrtc/src/muxer.ts
index 991d130230..ea8ed34fc6 100644
--- a/packages/transport-webrtc/src/muxer.ts
+++ b/packages/transport-webrtc/src/muxer.ts
@@ -1,5 +1,5 @@
 import { createStream } from './stream.js'
-import { nopSink, nopSource } from './util.js'
+import { drainAndClose, 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'
@@ -9,6 +9,7 @@ import type { Source, Sink } from 'it-stream-types'
 import type { Uint8ArrayList } from 'uint8arraylist'
 
 const PROTOCOL = '/webrtc'
+const DEFAULT_CHANNEL_CLOSE_DELAY = 30000
 
 export interface DataChannelMuxerFactoryInit {
   /**
@@ -131,6 +132,7 @@ export class DataChannelMuxer implements StreamMuxer {
         direction: 'inbound',
         dataChannelOptions: this.dataChannelOptions,
         onEnd: () => {
+          drainAndClose(channel, this.dataChannelOptions?.closeDelay ?? DEFAULT_CHANNEL_CLOSE_DELAY)
           this.streams = this.streams.filter(s => s.id !== stream.id)
           this.metrics?.increment({ stream_end: true })
           init?.onStreamEnd?.(stream)
@@ -158,7 +160,7 @@ export class DataChannelMuxer implements StreamMuxer {
       direction: 'outbound',
       dataChannelOptions: this.dataChannelOptions,
       onEnd: () => {
-        channel.close() // Stream initiator is responsible for closing the channel
+        drainAndClose(channel, this.dataChannelOptions?.closeDelay ?? DEFAULT_CHANNEL_CLOSE_DELAY)
         this.streams = this.streams.filter(s => s.id !== stream.id)
         this.metrics?.increment({ stream_end: true })
         this.init?.onStreamEnd?.(stream)
diff --git a/packages/transport-webrtc/src/private-to-private/transport.ts b/packages/transport-webrtc/src/private-to-private/transport.ts
index 1dd1804140..caaa8c1858 100644
--- a/packages/transport-webrtc/src/private-to-private/transport.ts
+++ b/packages/transport-webrtc/src/private-to-private/transport.ts
@@ -105,7 +105,7 @@ export class WebRTCTransport implements Transport, Startable {
    * <relay address>/p2p/<relay-peer>/p2p-circuit/webrtc/p2p/<destination-peer>
   */
   async dial (ma: Multiaddr, options: DialOptions): Promise<Connection> {
-    log.trace('dialing address: ', ma)
+    log.trace('dialing address: %a', ma)
     const { baseAddr, peerId } = splitAddr(ma)
 
     if (options.signal == null) {
diff --git a/packages/transport-webrtc/src/private-to-public/transport.ts b/packages/transport-webrtc/src/private-to-public/transport.ts
index b0c37a7306..b5508a73af 100644
--- a/packages/transport-webrtc/src/private-to-public/transport.ts
+++ b/packages/transport-webrtc/src/private-to-public/transport.ts
@@ -56,6 +56,7 @@ export interface WebRTCMetrics {
 
 export interface WebRTCTransportDirectInit {
   dataChannel?: Partial<DataChannelOpts>
+  dataChannelCloseDelay?: number
 }
 
 export class WebRTCDirectTransport implements Transport {
@@ -80,7 +81,7 @@ export class WebRTCDirectTransport implements Transport {
    */
   async dial (ma: Multiaddr, options: WebRTCDialOptions): Promise<Connection> {
     const rawConn = await this._connect(ma, options)
-    log(`dialing address - ${ma.toString()}`)
+    log.trace('dialing address: %a', ma)
     return rawConn
   }
 
diff --git a/packages/transport-webrtc/src/stream.ts b/packages/transport-webrtc/src/stream.ts
index 1e9bb0c3c7..f2bd035f5c 100644
--- a/packages/transport-webrtc/src/stream.ts
+++ b/packages/transport-webrtc/src/stream.ts
@@ -12,6 +12,7 @@ export interface DataChannelOpts {
   maxMessageSize: number
   maxBufferedAmount: number
   bufferedAmountLowEventTimeout: number
+  closeDelay?: number
 }
 
 export interface WebRTCStreamInit extends AbstractStreamInit {
diff --git a/packages/transport-webrtc/src/util.ts b/packages/transport-webrtc/src/util.ts
index e26e64dd5f..36af653733 100644
--- a/packages/transport-webrtc/src/util.ts
+++ b/packages/transport-webrtc/src/util.ts
@@ -1,4 +1,8 @@
+import { logger } from '@libp2p/logger'
 import { detect } from 'detect-browser'
+import pDefer from 'p-defer'
+
+const log = logger('libp2p:webrtc:utils')
 
 const browser = detect()
 export const isFirefox = ((browser != null) && browser.name === 'firefox')
@@ -6,3 +10,50 @@ export const isFirefox = ((browser != null) && browser.name === 'firefox')
 export const nopSource = async function * nop (): AsyncGenerator<Uint8Array, any, unknown> {}
 
 export const nopSink = async (_: any): Promise<void> => {}
+
+export function drainAndClose (channel: RTCDataChannel, channelCloseDelay: number): void {
+  if (channel.readyState !== 'open') {
+    return
+  }
+
+  void Promise.resolve()
+    .then(async () => {
+      // wait for bufferedAmount to become zero
+      if (channel.bufferedAmount > 0) {
+        const deferred = pDefer()
+
+        channel.bufferedAmountLowThreshold = 0
+        channel.addEventListener('bufferedamountlow', () => {
+          deferred.resolve()
+        })
+
+        await deferred.promise
+      }
+    })
+    .then(async () => {
+      const deferred = pDefer()
+
+      // event if bufferedAmount is zero there can still be unsent bytes
+      const timeout = setTimeout(() => {
+        try {
+          // only close if the channel is still open
+          if (channel.readyState === 'open') {
+            channel.close()
+          }
+
+          deferred.resolve()
+        } catch (err: any) {
+          deferred.reject(err)
+        }
+      }, channelCloseDelay)
+
+      if (timeout.unref != null) {
+        timeout.unref()
+      }
+
+      await deferred.promise
+    })
+    .catch(err => {
+      log.error('error closing outbound stream', err)
+    })
+}