diff --git a/examples/wgpu_room/src/app.rs b/examples/wgpu_room/src/app.rs
index 99001f27..aaeb59db 100644
--- a/examples/wgpu_room/src/app.rs
+++ b/examples/wgpu_room/src/app.rs
@@ -5,7 +5,9 @@ use crate::{
 };
 use egui::{Rounding, Stroke};
 use livekit::{e2ee::EncryptionType, prelude::*, SimulateScenario};
+use livekit::webrtc::{audio_stream::native::NativeAudioStream};
 use std::collections::HashMap;
+use futures::StreamExt;
 
 /// The state of the application are saved on app exit and restored on app start.
 #[derive(serde::Deserialize, serde::Serialize)]
@@ -88,8 +90,15 @@ impl LkApp {
                             );
                             self.video_renderers
                                 .insert((participant.identity(), track.sid()), video_renderer);
-                        } else if let RemoteTrack::Audio(_) = track {
-                            // TODO(theomonnom): Once we support media devices, we can play audio tracks here
+                        } else if let RemoteTrack::Audio(ref audio_track) = track {
+                            let rtc_track = audio_track.rtc_track();
+                            let mut audio_stream = NativeAudioStream::new(rtc_track, 48000, 2);
+                            // Receive the audio frames in a new task
+                            tokio::spawn(async move {
+                                while let Some(frame) = audio_stream.next().await {
+                                    println!("Received audio frame {:?}", frame);
+                                }
+                            });
                         }
                     }
                     RoomEvent::TrackUnsubscribed {
diff --git a/webrtc-sys/src/audio_device.cpp b/webrtc-sys/src/audio_device.cpp
index 9ae77c18..f3877498 100644
--- a/webrtc-sys/src/audio_device.cpp
+++ b/webrtc-sys/src/audio_device.cpp
@@ -167,12 +167,14 @@ bool AudioDevice::RecordingIsInitialized() const {
 int32_t AudioDevice::StartPlayout() {
   webrtc::MutexLock lock(&mutex_);
   playing_ = true;
+  audio_device_buffer_.StartPlayout();
   return 0;
 }
 
 int32_t AudioDevice::StopPlayout() {
   webrtc::MutexLock lock(&mutex_);
   playing_ = false;
+  audio_device_buffer_.StopPlayout();
   return 0;
 }
 
@@ -182,10 +184,12 @@ bool AudioDevice::Playing() const {
 }
 
 int32_t AudioDevice::StartRecording() {
+  audio_device_buffer_.StartRecording();
   return 0;
 }
 
 int32_t AudioDevice::StopRecording() {
+  audio_device_buffer_.StopRecording();
   return 0;
 }