diff --git a/dogfooding/ios/Flutter/AppFrameworkInfo.plist b/dogfooding/ios/Flutter/AppFrameworkInfo.plist
index 1dc6cf765..391a902b2 100644
--- a/dogfooding/ios/Flutter/AppFrameworkInfo.plist
+++ b/dogfooding/ios/Flutter/AppFrameworkInfo.plist
@@ -20,7 +20,5 @@
   <string>????</string>
   <key>CFBundleVersion</key>
   <string>1.0</string>
-  <key>MinimumOSVersion</key>
-  <string>13.0</string>
 </dict>
 </plist>
diff --git a/dogfooding/linux/flutter/generated_plugin_registrant.cc b/dogfooding/linux/flutter/generated_plugin_registrant.cc
index 990ec5d8c..aac6d2082 100644
--- a/dogfooding/linux/flutter/generated_plugin_registrant.cc
+++ b/dogfooding/linux/flutter/generated_plugin_registrant.cc
@@ -13,7 +13,6 @@
 #include <record_linux/record_linux_plugin.h>
 #include <stream_webrtc_flutter/flutter_web_r_t_c_plugin.h>
 #include <url_launcher_linux/url_launcher_plugin.h>
-#include <volume_controller/volume_controller_plugin.h>
 
 void fl_register_plugins(FlPluginRegistry* registry) {
   g_autoptr(FlPluginRegistrar) desktop_drop_registrar =
@@ -37,7 +36,4 @@ void fl_register_plugins(FlPluginRegistry* registry) {
   g_autoptr(FlPluginRegistrar) url_launcher_linux_registrar =
       fl_plugin_registry_get_registrar_for_plugin(registry, "UrlLauncherPlugin");
   url_launcher_plugin_register_with_registrar(url_launcher_linux_registrar);
-  g_autoptr(FlPluginRegistrar) volume_controller_registrar =
-      fl_plugin_registry_get_registrar_for_plugin(registry, "VolumeControllerPlugin");
-  volume_controller_plugin_register_with_registrar(volume_controller_registrar);
 }
diff --git a/dogfooding/linux/flutter/generated_plugins.cmake b/dogfooding/linux/flutter/generated_plugins.cmake
index 0454a4f7b..b8f45926a 100644
--- a/dogfooding/linux/flutter/generated_plugins.cmake
+++ b/dogfooding/linux/flutter/generated_plugins.cmake
@@ -10,10 +10,10 @@ list(APPEND FLUTTER_PLUGIN_LIST
   record_linux
   stream_webrtc_flutter
   url_launcher_linux
-  volume_controller
 )
 
 list(APPEND FLUTTER_FFI_PLUGIN_LIST
+  jni
 )
 
 set(PLUGIN_BUNDLED_LIBRARIES)
diff --git a/dogfooding/windows/flutter/generated_plugins.cmake b/dogfooding/windows/flutter/generated_plugins.cmake
index b24973bba..310f70598 100644
--- a/dogfooding/windows/flutter/generated_plugins.cmake
+++ b/dogfooding/windows/flutter/generated_plugins.cmake
@@ -18,10 +18,10 @@ list(APPEND FLUTTER_PLUGIN_LIST
   stream_webrtc_flutter
   thumblr_windows
   url_launcher_windows
-  volume_controller
 )
 
 list(APPEND FLUTTER_FFI_PLUGIN_LIST
+  jni
 )
 
 set(PLUGIN_BUNDLED_LIBRARIES)
diff --git a/melos.yaml b/melos.yaml
index 32b440df5..37116b01f 100644
--- a/melos.yaml
+++ b/melos.yaml
@@ -22,7 +22,10 @@ command:
       device_info_plus: ^12.1.0
       share_plus: ^11.0.0
       stream_chat_flutter: ^9.17.0
-      stream_webrtc_flutter: ^2.2.6
+      stream_webrtc_flutter:
+        git:
+          url: https://github.com/GetStream/webrtc-flutter.git
+          ref: chore/per-call-pc-factory
       stream_video_flutter: ^1.3.3
       stream_video_noise_cancellation: ^1.3.3
       stream_video_push_notification: ^1.3.3
diff --git a/packages/stream_video/CHANGELOG.md b/packages/stream_video/CHANGELOG.md
index 971f76da8..b2c820d2b 100644
--- a/packages/stream_video/CHANGELOG.md
+++ b/packages/stream_video/CHANGELOG.md
@@ -1,3 +1,16 @@
+## Unreleased
+
+### ✅ Added
+* Added `TrackDisableMode` enum and optional `disableMode` parameter to `Call.setMicrophoneEnabled()`. Allows integrators to choose between releasing the microphone hardware on mute (`TrackDisableMode.stopTracks`, the default) or keeping the capture session alive (`TrackDisableMode.disableTracks`). The latter avoids the brief iOS `AVAudioSession` teardown that can duck playback of other participants for ~1–2 seconds — recommended for audio rooms and other playback-sensitive use cases. Note: `disableTracks` keeps the system microphone indicator visible while muted because the capture hardware remains active.
+
+### 🐞 Fixed
+* Fixed sibling-call audio capture being silently broken when another concurrently-active call ended (e.g. a 1:1 ringing call ending alongside a running livestream, or a previous ringing call ending before a new one was accepted). `RtcManager.dispose()` now skips the `pc.removeTrack(sender)` step on the publisher PC when it's about to dispose the PC entirely. The explicit `removeTrack` triggers libwebrtc's per-`Call.AudioState` to issue `ADM.StopRecording()` on the **process-wide shared** `AudioDeviceModule` — with no refcount across PCs — which left every still-active call wired against a stopped capture pipeline. Wholesale `pc.dispose()` doesn't take the same lifecycle path and tears down the PC cleanly. Implemented via a new `removeFromPc` parameter on `unpublishTrack` (default `true`; `false` when called from `dispose`). See `docs/audio-lifecycle-analysis.md` for the full investigation trail.
+* Fixed a sibling call's audio breaking when a ringing 1:1 call ended via `dropIfAloneInRingingFlow` (the remote party hung up first). `Call.end()` and `Call.leave()` now share a single `_disconnect` cleanup path, so both honor `_leaveCallTriggered`, complete `_callLifecycleCompleter`, and short-circuit consistently when the call is already disconnected — previously `Call.end()` skipped these guards, which caused races with concurrent reconnect handlers and with re-enabling the mic on a sibling active call.
+* Made the audio processor teardown in `Call._clear` multi-call aware. The audio processor is owned by `StreamVideo`, not by an individual `Call`, so disabling it on one call's teardown silently dropped noise cancellation on any other still-active call. `_clear` now only stops the global processor when no other active call is configured to use `NoiceCancellationSettingsMode.autoOn`.
+
+### 🔄 Changed
+* `Call.leave()` and `Call.end()` now actually wait for the underlying native teardown before returning. Previously `Call._clear` fire-and-forgot `_session.dispose()`, and `CallSession.close` itself fire-and-forgot the WebRTC manager dispose and the SFU WebSocket disconnect, so callers could observe `leave()`/`end()` "complete" while peer connections, local audio tracks, and audio sources were still being torn down on the native side. With this change, awaiting `Call.leave()` / `Call.end()` is enough to guarantee the native cleanup has finished — important when the next thing the integrator does is touch a sibling active call's audio (e.g. resuming a livestream's mic after a 1:1 ringing call ends). Leave/end will take slightly longer to return; if you need fire-and-forget semantics, wrap the call in `unawaited(...)` yourself.
+
 ## 1.3.3
 
 ### 🐞 Fixed
diff --git a/packages/stream_video/lib/src/call/call.dart b/packages/stream_video/lib/src/call/call.dart
index a2edb0cd4..851800c04 100644
--- a/packages/stream_video/lib/src/call/call.dart
+++ b/packages/stream_video/lib/src/call/call.dart
@@ -47,6 +47,7 @@ import '../utils/subscriptions.dart';
 import '../webrtc/media/media_constraints.dart';
 import '../webrtc/model/rtc_video_dimension.dart';
 import '../webrtc/model/rtc_video_parameters.dart';
+import '../webrtc/model/track_disable_mode.dart';
 import '../webrtc/rtc_audio_api/rtc_audio_api.dart' as rtc_audio;
 import '../webrtc/rtc_manager.dart';
 import '../webrtc/rtc_media_device/rtc_media_device.dart';
@@ -278,6 +279,7 @@ class Call {
 
   CallCredentials? _credentials;
   CallSession? _session;
+  CallSession? get callSession => _session;
   CallSession? _previousSession;
 
   StatsOptions? _sfuStatsOptions;
@@ -847,22 +849,34 @@ class Call {
 
   /// Ends the call for all participants.
   Future<Result<None>> end({String? reason}) async {
-    final state = this.state.value;
-    _logger.d(() => '[end] status: ${state.status}');
+    _logger.d(() => '[end] status: ${state.value.status}');
 
-    if (state.status is! CallStatusActive) {
-      _logger.w(() => '[end] rejected (invalid status): ${state.status}');
-      return Result.error('invalid status: ${state.status}');
+    if (state.value.status is! CallStatusActive) {
+      _logger.w(() => '[end] rejected (invalid status): ${state.value.status}');
+      return Result.error('invalid status: ${state.value.status}');
     }
 
-    _session?.leave(reason: reason ?? 'user is ending the call');
-    await _clear('end');
+    try {
+      final didDisconnect = await _disconnect(
+        sfuLeaveReason: reason ?? 'user is ending the call',
+      );
 
-    final result = await _permissionsManager.endCall();
-    _stateManager.lifecycleCallEnded();
+      // If another disconnect already ran (or is running), don't fire the
+      // server-side endCall a second time and don't re-emit the lifecycle
+      // event.
+      if (!didDisconnect) {
+        _logger.v(() => '[end] disconnect short-circuited');
+        return const Result.success(none);
+      }
 
-    _logger.v(() => '[end] completed: $result');
-    return result;
+      final result = await _permissionsManager.endCall();
+      _stateManager.lifecycleCallEnded();
+
+      _logger.v(() => '[end] completed: $result');
+      return result;
+    } finally {
+      _leaveCallTriggered = false;
+    }
   }
 
   /// Joins the call.
@@ -1170,6 +1184,9 @@ class Call {
         networkMonitor: networkMonitor,
         streamVideo: _streamVideo,
         statsOptions: _sfuStatsOptions!,
+        audioConfigurationPolicy:
+            _stateManager.callState.preferences.audioConfigurationPolicy ??
+            _streamVideo.options.audioConfigurationPolicy,
         leftoverTraceRecords:
             _previousSession
                 ?.getTrace()
@@ -2041,41 +2058,58 @@ class Call {
   ///
   /// - [reason]: optional reason for leaving the call
   Future<Result<None>> leave({DisconnectReason? reason}) async {
-    try {
-      if (_leaveCallTriggered) {
-        _logger.i(() => '[leave] rejected (already leaving call)');
-        return const Result.success(none);
-      }
+    _logger.i(() => '[leave] reason: $reason');
 
-      _leaveCallTriggered = true;
+    try {
+      final didDisconnect = await _disconnect(
+        sfuLeaveReason: _sfuLeaveReason(reason),
+      );
 
-      // Complete the leave completer to cancel ongoing operations
-      if (!_callLifecycleCompleter.isCompleted) {
-        _callLifecycleCompleter.complete();
+      if (didDisconnect) {
+        _stateManager.lifecycleCallDisconnected(reason: reason);
       }
 
-      final state = this.state.value;
-      _logger.i(() => '[leave] state: $state');
+      _logger.v(() => '[leave] finished');
+      return const Result.success(none);
+    } finally {
+      _leaveCallTriggered = false;
+    }
+  }
 
-      if (state.status.isDisconnected) {
-        _logger.d(() => '[leave] rejected (state.status is disconnected)');
-        return const Result.success(none);
-      }
+  /// Shared cleanup sequence for [leave] and [end].
+  ///
+  /// Sets [_leaveCallTriggered], completes [_callLifecycleCompleter], sends
+  /// the SFU leave message, and runs [_clear]. Returns `true` when the
+  /// cleanup actually ran; `false` if it was short-circuited because a
+  /// concurrent disconnect was already in flight or the call was already
+  /// disconnected.
+  Future<bool> _disconnect({required String sfuLeaveReason}) async {
+    if (_leaveCallTriggered) {
+      _logger.i(() => '[disconnect] rejected (already disconnecting)');
+      return false;
+    }
 
-      try {
-        _session?.leave(reason: _sfuLeaveReason(reason));
-      } finally {
-        await _clear('leave');
-      }
+    _leaveCallTriggered = true;
 
-      _stateManager.lifecycleCallDisconnected(reason: reason);
+    // Complete the lifecycle completer to cancel ongoing operations awaiting
+    // it (e.g. _startSession). This must run regardless of whether the
+    // disconnect proceeds further so that nothing gets stuck waiting.
+    if (!_callLifecycleCompleter.isCompleted) {
+      _callLifecycleCompleter.complete();
+    }
 
-      _logger.v(() => '[leave] finished');
+    if (state.value.status.isDisconnected) {
+      _logger.d(() => '[disconnect] rejected (status is disconnected)');
+      return false;
+    }
 
-      return const Result.success(none);
+    try {
+      _session?.leave(reason: sfuLeaveReason);
     } finally {
-      _leaveCallTriggered = false;
+      await _clear('disconnect');
     }
+
+    return true;
   }
 
   String _sfuLeaveReason(DisconnectReason? reason) {
@@ -2109,6 +2143,7 @@ class Call {
     ]) {
       timer.cancel();
     }
+
     _videoModerationTimer?.cancel();
     _videoModerationTimer = null;
 
@@ -2132,7 +2167,9 @@ class Call {
 
     if (_session != null) {
       unawaited(
-        _session!.dispose().catchError((Object e) {
+        _session!.dispose().catchError((
+          Object e,
+        ) {
           _logger.w(() => '[clear] session dispose failed: $e');
         }),
       );
@@ -2288,7 +2325,9 @@ class Call {
       if (CurrentPlatform.isIos) {
         await _session?.rtcManager?.setAppleAudioConfiguration(
           speakerOn: _connectOptions.speakerDefaultOn,
-          policy: _streamVideo.options.audioConfigurationPolicy,
+          policy:
+              _stateManager.callState.preferences.audioConfigurationPolicy ??
+              _streamVideo.options.audioConfigurationPolicy,
         );
       }
     }
@@ -3295,9 +3334,23 @@ class Call {
     }
   }
 
+  /// Enables or disables the microphone for this call.
+  ///
+  /// When [enabled] is `false`, [disableMode] controls how the local audio
+  /// track is muted. Defaults to [TrackDisableMode.stopTracks], which
+  /// releases the microphone hardware on mute so the system privacy
+  /// indicator turns off. Pass [TrackDisableMode.disableTracks] to keep
+  /// the capture session alive — this avoids the brief iOS
+  /// `AVAudioSession` teardown that otherwise ducks playback of other
+  /// participants for ~1–2 s during mute/unmute, at the cost of the
+  /// system microphone indicator remaining visible while muted.
+  /// Recommended for audio rooms and other playback-sensitive use cases.
+  ///
+  /// See [TrackDisableMode] for the full tradeoff.
   Future<Result<None>> setMicrophoneEnabled({
     required bool enabled,
     AudioConstraints? constraints,
+    TrackDisableMode? disableMode,
   }) async {
     if (enabled &&
         state.value.isVideoModerated &&
@@ -3313,6 +3366,7 @@ class Call {
         await _session?.setMicrophoneEnabled(
           enabled,
           constraints: constraints,
+          disableMode: disableMode,
         ) ??
         Result.error('Session is null');
 
@@ -3349,7 +3403,15 @@ class Call {
     return result.map((_) => none);
   }
 
-  Future<bool> requestScreenSharePermission() {
+  Future<bool> requestScreenSharePermission() async {
+    // Request screen share permission from the native factory if available
+    final nativeFactory = await _session?.rtcManager?.pcFactory
+        .ensureNativeFactory();
+
+    if (nativeFactory != null) {
+      return nativeFactory.requestCapturePermission();
+    }
+
     return Helper.requestCapturePermission();
   }
 
diff --git a/packages/stream_video/lib/src/call/session/call_session.dart b/packages/stream_video/lib/src/call/session/call_session.dart
index 7ba92a199..dc89bc55c 100644
--- a/packages/stream_video/lib/src/call/session/call_session.dart
+++ b/packages/stream_video/lib/src/call/session/call_session.dart
@@ -58,6 +58,7 @@ class CallSession extends Disposable {
     required Tracer tracer,
     this.clientPublishOptions,
     this.joinResponseTimeout = const Duration(seconds: 5),
+    AudioConfigurationPolicy? audioConfigurationPolicy,
   }) : _tracer = tracer,
        _streamVideo = streamVideo,
        sfuClient = SfuClient(
@@ -81,6 +82,7 @@ class CallSession extends Disposable {
          callCid: callCid,
          configuration: config.rtcConfig,
          sdpEditor: sdpEditor,
+         audioConfigurationPolicy: audioConfigurationPolicy,
        ) {
     _logger.i(() => '<init> callCid: $callCid, sessionId: $sessionId');
     _observeNetworkStatus();
@@ -520,7 +522,9 @@ class CallSession extends Disposable {
     StreamWebSocketCloseCode code, {
     String? closeReason,
   }) async {
-    _logger.d(() => '[close] code: $code, closeReason: $closeReason');
+    _logger.d(
+      () => '[close] code: $code, closeReason: $closeReason',
+    );
     _isLeavingOrClosed = true;
 
     await _eventsSubscription?.cancel();
@@ -546,7 +550,10 @@ class CallSession extends Disposable {
 
     if (rtcManager != null) {
       unawaited(
-        rtcManager!.dispose().catchError((Object e, StackTrace stk) {
+        rtcManager!.dispose().catchError((
+          Object e,
+          StackTrace stk,
+        ) {
           _logger.w(() => '[close] rtcManager.dispose failed: $e');
         }),
       );
@@ -1042,22 +1049,20 @@ class CallSession extends Disposable {
   Future<Result<RtcLocalTrack>> setMicrophoneEnabled(
     bool enabled, {
     AudioConstraints? constraints,
+    TrackDisableMode? disableMode,
   }) async {
     final rtcManager = this.rtcManager;
     if (rtcManager == null) {
       return Result.error('Unable to set microphone, Call not connected');
     }
 
-    final result = TracerZone.run(
-      _zonedTracer,
-      ++zonedTracerSeq,
-      () async {
-        return rtcManager.setMicrophoneEnabled(
-          enabled: enabled,
-          constraints: constraints,
-        );
-      },
-    );
+    final result = TracerZone.run(_zonedTracer, ++zonedTracerSeq, () async {
+      return rtcManager.setMicrophoneEnabled(
+        enabled: enabled,
+        constraints: constraints,
+        disableMode: disableMode,
+      );
+    });
 
     return result;
   }
diff --git a/packages/stream_video/lib/src/call/session/call_session_factory.dart b/packages/stream_video/lib/src/call/session/call_session_factory.dart
index c648c414b..fd13f61c6 100644
--- a/packages/stream_video/lib/src/call/session/call_session_factory.dart
+++ b/packages/stream_video/lib/src/call/session/call_session_factory.dart
@@ -51,6 +51,7 @@ class CallSessionFactory {
     required StreamVideo streamVideo,
     ClientPublishOptions? clientPublishOptions,
     List<TraceRecord> leftoverTraceRecords = const [],
+    AudioConfigurationPolicy? audioConfigurationPolicy,
   }) async {
     final finalSessionId = sessionId ?? const Uuid().v4();
     _logger.d(() => '[makeCallSession] sessionId: $finalSessionId($sessionId)');
@@ -98,6 +99,7 @@ class CallSessionFactory {
       statsOptions: statsOptions,
       streamVideo: streamVideo,
       tracer: tracer,
+      audioConfigurationPolicy: audioConfigurationPolicy,
     );
   }
 
diff --git a/packages/stream_video/lib/src/models/call_preferences.dart b/packages/stream_video/lib/src/models/call_preferences.dart
index 63fef252c..6365c9887 100644
--- a/packages/stream_video/lib/src/models/call_preferences.dart
+++ b/packages/stream_video/lib/src/models/call_preferences.dart
@@ -1,3 +1,4 @@
+import 'audio_configuration_policy.dart';
 import 'call_client_publish_options.dart';
 import 'moderation_blur_config.dart';
 
@@ -54,6 +55,11 @@ abstract class CallPreferences {
   /// Configuration for how the SDK handles call moderation events.
   /// Defaults to [VideoModerationConfig.disabled].
   VideoModerationConfig get videoModerationConfig;
+
+  /// Per-call audio configuration override. When non-null, the per-call
+  /// native peer-connection factory is built with this policy instead of
+  /// the client-level default `StreamVideoOptions.audioConfigurationPolicy`.
+  AudioConfigurationPolicy? get audioConfigurationPolicy;
 }
 
 class DefaultCallPreferences implements CallPreferences {
@@ -68,6 +74,7 @@ class DefaultCallPreferences implements CallPreferences {
     this.closedCaptionsVisibilityDurationMs = 2700,
     this.closedCaptionsVisibleCaptions = 2,
     this.videoModerationConfig = const VideoModerationConfig.disabled(),
+    this.audioConfigurationPolicy,
   });
 
   /// The maximum duration to wait when establishing a connection to the call.
@@ -149,4 +156,12 @@ class DefaultCallPreferences implements CallPreferences {
   /// Defaults to [VideoModerationConfig.disabled].
   @override
   final VideoModerationConfig videoModerationConfig;
+
+  /// Per-call audio configuration override. When non-null, the per-call
+  /// native peer-connection factory is built with this policy instead of
+  /// the client-level default `StreamVideoOptions.audioConfigurationPolicy`.
+  ///
+  /// Defaults to null (falls back to `StreamVideoOptions.audioConfigurationPolicy`).
+  @override
+  final AudioConfigurationPolicy? audioConfigurationPolicy;
 }
diff --git a/packages/stream_video/lib/src/webrtc/media/media_constraints.dart b/packages/stream_video/lib/src/webrtc/media/media_constraints.dart
index 0364ad025..16973bdcb 100644
--- a/packages/stream_video/lib/src/webrtc/media/media_constraints.dart
+++ b/packages/stream_video/lib/src/webrtc/media/media_constraints.dart
@@ -33,12 +33,21 @@ abstract class MediaConstraints {
 }
 
 extension MediaDevices on rtc_interface.MediaDevices {
-  Future<rtc.MediaStream> getMedia(MediaConstraints constraints) async {
+  /// Captures media. When [nativeFactory] is non-null, the resulting tracks
+  /// are pinned to that per-call native factory. When null, falls through to
+  /// the global browser/webrtc entrypoints (used on web, which has no
+  /// per-call factory concept).
+  Future<rtc.MediaStream> getMedia(
+    MediaConstraints constraints, {
+    rtc.NativePeerConnectionFactory? nativeFactory,
+  }) async {
     final constraintsMap = constraints.toMap();
     streamLog.i(
       'SV:MediaDevices',
       () =>
-          '[getMedia] #${constraints.runtimeType}; constraintsMap: $constraintsMap',
+          '[getMedia] #${constraints.runtimeType}; '
+          'nativeFactory: ${nativeFactory?.factoryId}, '
+          'constraintsMap: $constraintsMap',
     );
 
     final (tracer, sequence) = TracerZone.currentTracer;
@@ -51,16 +60,16 @@ extension MediaDevices on rtc_interface.MediaDevices {
         tag = 'navigator.mediaDevices.getDisplayMedia';
         tracer?.trace('$tag.$sequence', constraintsMap);
 
-        stream = await rtc.navigator.mediaDevices.getDisplayMedia(
-          constraintsMap,
-        );
+        stream = nativeFactory != null
+            ? await nativeFactory.getDisplayMedia(constraintsMap)
+            : await rtc.navigator.mediaDevices.getDisplayMedia(constraintsMap);
       } else {
         tag = 'navigator.mediaDevices.getUserMedia';
         tracer?.trace('$tag.$sequence', constraintsMap);
 
-        stream = await rtc.navigator.mediaDevices.getUserMedia(
-          constraintsMap,
-        );
+        stream = nativeFactory != null
+            ? await nativeFactory.getUserMedia(constraintsMap)
+            : await rtc.navigator.mediaDevices.getUserMedia(constraintsMap);
       }
     } catch (e) {
       tracer?.trace('$tag.failure.$sequence', e.toString());
diff --git a/packages/stream_video/lib/src/webrtc/model/track_disable_mode.dart b/packages/stream_video/lib/src/webrtc/model/track_disable_mode.dart
new file mode 100644
index 000000000..24d1d9b86
--- /dev/null
+++ b/packages/stream_video/lib/src/webrtc/model/track_disable_mode.dart
@@ -0,0 +1,29 @@
+/// Controls how a local media track is disabled when the user mutes it.
+enum TrackDisableMode {
+  /// Keep the local track and the underlying capture session alive, and only
+  /// flip `enabled = false` on the track so WebRTC stops encoding frames.
+  ///
+  /// Pros:
+  /// - On iOS the shared `AVAudioSession` is not torn down, so playback of
+  ///   other participants is not interrupted when the user mutes/unmutes.
+  ///
+  /// Cons:
+  /// - The microphone/camera capture hardware keeps running, so the system
+  ///   privacy indicator (orange mic dot / green camera dot on iOS, the
+  ///   equivalent indicators on Android 12+) remains visible while muted.
+  disableTracks,
+
+  /// Stop the local track and release the capture hardware on mute, then
+  /// recreate it via `getUserMedia` on unmute.
+  ///
+  /// Pros:
+  /// - The system privacy indicator turns off while muted, matching the
+  ///   behaviour users expect from native voice apps.
+  ///
+  /// Cons:
+  /// - On iOS, disposing the local `MediaStream` tears down the shared
+  ///   `AVAudioSession` via the native `streamDispose` path, which briefly
+  ///   ducks playback of the other participants for ~1–2 seconds while the
+  ///   audio graph restarts. This might be noticeable in audio rooms.
+  stopTracks,
+}
diff --git a/packages/stream_video/lib/src/webrtc/peer_connection_factory.dart b/packages/stream_video/lib/src/webrtc/peer_connection_factory.dart
index ace37e2e5..cd7fb0edc 100644
--- a/packages/stream_video/lib/src/webrtc/peer_connection_factory.dart
+++ b/packages/stream_video/lib/src/webrtc/peer_connection_factory.dart
@@ -5,18 +5,26 @@ import '../../protobuf/video/sfu/models/models.pb.dart';
 import '../call/session/call_session_config.dart';
 import '../call/stats/tracer.dart';
 import '../logger/impl/tagged_logger.dart';
+import '../models/audio_configuration_policy.dart';
 import '../models/call_cid.dart';
+import '../platform_detector/platform_detector.dart';
 import '../sfu/sfu_client.dart';
 import '../types/other.dart';
 import 'peer_type.dart';
 import 'sdp/editor/sdp_editor.dart';
 import 'traced_peer_connection.dart';
 
+/// Owns the per-call native peer connection factory.
+///
+/// The factory is built lazily on first [makePeerConnection] call and
+/// disposed via [dispose]. Callers must dispose the factory once every PC
+/// it created has been disposed.
 class StreamPeerConnectionFactory {
   StreamPeerConnectionFactory({
     required this.sessionId,
     required this.callCid,
     required this.sdpEditor,
+    this.audioConfigurationPolicy,
   });
 
   final _logger = taggedLogger(tag: 'SV:PeerConnectionFactory');
@@ -25,6 +33,66 @@ class StreamPeerConnectionFactory {
   final StreamCallCid callCid;
   final SdpEditor sdpEditor;
 
+  /// Audio policy applied to the per-call factory build. Falls back to
+  /// [BroadcasterAudioPolicy] when null.
+  final AudioConfigurationPolicy? audioConfigurationPolicy;
+
+  rtc.NativePeerConnectionFactory? _nativeFactory;
+
+  /// Whether the platform exposes the per-call native factory APIs.
+  /// Web / desktop fall through to the global webrtc entrypoints because
+  /// per-call factories aren't a concept there.
+  bool get _isPerCallFactorySupported =>
+      CurrentPlatform.isAndroid ||
+      CurrentPlatform.isIos ||
+      CurrentPlatform.isMacOS;
+
+  /// The per-call native factory, lazily built on first use. Returns null on
+  /// web (no per-call factory concept) so callers can fall back to the
+  /// global webrtc entrypoints.
+  Future<rtc.NativePeerConnectionFactory?> ensureNativeFactory() async {
+    if (!_isPerCallFactorySupported) {
+      return null;
+    }
+
+    if (_nativeFactory != null) {
+      return _nativeFactory;
+    }
+
+    final policy = audioConfigurationPolicy ?? const BroadcasterAudioPolicy();
+    final options = <String, dynamic>{
+      'bypassVoiceProcessing': policy.bypassVoiceProcessing,
+    };
+
+    if (CurrentPlatform.isAndroid) {
+      options['androidAudioConfiguration'] = policy
+          .getAndroidConfiguration()
+          .toMap();
+    }
+
+    if (CurrentPlatform.isIos || CurrentPlatform.isMacOS) {
+      options['appleAudioConfiguration'] = policy
+          .getAppleConfiguration()
+          .toMap();
+    }
+
+    _nativeFactory = await rtc.NativePeerConnectionFactory.create(
+      options: options,
+    );
+
+    _logger.i(
+      () =>
+          '[ensureNativeFactory] built per-call factory '
+          'id: ${_nativeFactory!.factoryId}, policy: ${policy.runtimeType}',
+    );
+    return _nativeFactory;
+  }
+
+  /// Synchronous accessor returning the cached factory if already built.
+  /// Returns null if [ensureNativeFactory] has not yet been awaited or on
+  /// platforms without per-call factory support.
+  rtc.NativePeerConnectionFactory? get nativeFactory => _nativeFactory;
+
   Future<TracedStreamPeerConnection> makeSubscriber(
     SfuClient sfuClient,
     RTCConfiguration configuration,
@@ -82,10 +150,17 @@ class StreamPeerConnectionFactory {
           '[createPeerConnection] #$type; configuration: '
           '${configuration.toMap()}, mediaConstraints: $mediaConstraints',
     );
-    final pc = await rtc.createPeerConnection(
-      configuration.toMap(),
-      mediaConstraints,
-    );
+
+    final nativeFactory = await ensureNativeFactory();
+    final pc = nativeFactory != null
+        ? await nativeFactory.createPeerConnection(
+            configuration.toMap(),
+            mediaConstraints,
+          )
+        : await rtc.createPeerConnection(
+            configuration.toMap(),
+            mediaConstraints,
+          );
 
     final tracer = Tracer(
       "$tracerIdPrefix-${type == StreamPeerType.publisher ? 'pub' : 'sub'}",
@@ -112,4 +187,20 @@ class StreamPeerConnectionFactory {
       tracer: tracer,
     );
   }
+
+  /// Tears down the per-call factory. Must be called only after every PC the
+  /// factory created has been disposed.
+  Future<void> dispose() async {
+    if (_nativeFactory == null) {
+      return;
+    }
+    final factory = _nativeFactory!;
+    _nativeFactory = null;
+    try {
+      await factory.dispose();
+      _logger.i(() => '[dispose] released factory id: ${factory.factoryId}');
+    } catch (e, stk) {
+      _logger.w(() => '[dispose] native factory dispose failed: $e\n$stk');
+    }
+  }
 }
diff --git a/packages/stream_video/lib/src/webrtc/rtc_manager.dart b/packages/stream_video/lib/src/webrtc/rtc_manager.dart
index 6dc56032d..1d47dca93 100644
--- a/packages/stream_video/lib/src/webrtc/rtc_manager.dart
+++ b/packages/stream_video/lib/src/webrtc/rtc_manager.dart
@@ -20,6 +20,7 @@ import 'codecs_helper.dart';
 import 'model/rtc_tracks_info.dart';
 import 'model/rtc_video_encoding.dart';
 import 'peer_connection.dart';
+import 'peer_connection_factory.dart';
 import 'rtc_audio_api/rtc_audio_api.dart'
     show checkIfAudioOutputChangeSupported;
 import 'rtc_parser.dart';
@@ -41,10 +42,7 @@ typedef OnLocalTrackPublished = void Function(RtcLocalTrack track);
 /// Called when a subscriber track is received.
 /// {@endtemplate}
 typedef OnRemoteTrackReceived =
-    void Function(
-      StreamPeerConnection pc,
-      RtcRemoteTrack track,
-    );
+    void Function(StreamPeerConnection pc, RtcRemoteTrack track);
 
 const _tag = 'SV:RtcManager';
 
@@ -58,6 +56,7 @@ class RtcManager extends Disposable {
     required this.publishOptions,
     required this.stateManager,
     required StreamVideo streamVideo,
+    required this.pcFactory,
   }) : _streamVideo = streamVideo {
     subscriber.onTrack = _onRemoteTrack;
   }
@@ -72,6 +71,8 @@ class RtcManager extends Disposable {
   final TracedStreamPeerConnection subscriber;
   final StreamVideo _streamVideo;
 
+  final StreamPeerConnectionFactory pcFactory;
+
   final transceiversManager = TransceiverManager();
 
   List<SfuPublishOptions> publishOptions;
@@ -215,6 +216,18 @@ class RtcManager extends Disposable {
     _defaultAudioConstraints = constraints;
 
     final localAudioTracks = tracks.values.whereType<RtcLocalAudioTrack>();
+
+    // Capture each track's current stopTrackOnMute before forcing a
+    // stop-and-recreate cycle. The cycle must use stopTrackOnMute: true so
+    // that unmuteTrack enters the recreate branch and picks up the new
+    // constraints via getUserMedia. After recreation the original value is
+    // restored so that subsequent mute/unmute calls honour the integrator's
+    // chosen TrackDisableMode.
+    final originalModes = {
+      for (final track in localAudioTracks)
+        track.trackId: track.stopTrackOnMute,
+    };
+
     for (final track in localAudioTracks) {
       await muteTrack(trackId: track.trackId, stopTrackOnMute: true);
     }
@@ -222,9 +235,21 @@ class RtcManager extends Disposable {
     for (final track in localAudioTracks) {
       await unmuteTrack(trackId: track.trackId);
     }
+
+    // Restore the original stopTrackOnMute value on each recreated track.
+    for (final entry in originalModes.entries) {
+      final track = tracks[entry.key];
+      if (track is RtcLocalAudioTrack) {
+        tracks[entry.key] = track.copyWith(stopTrackOnMute: entry.value);
+      }
+    }
   }
 
-  Future<void> unpublishTrack({required String trackId}) async {
+  /// Stops the local track / clones / media stream for [trackId] and calls
+  /// `pc.removeTrack` on every sender that referenced it.
+  Future<void> unpublishTrack({
+    required String trackId,
+  }) async {
     final publishedTrack = tracks.remove(trackId);
 
     if (publishedTrack == null) {
@@ -337,10 +362,7 @@ class RtcManager extends Disposable {
         final localTrack = tracks[item.track.trackId] as RtcLocalTrack?;
         if (localTrack != null) {
           tracks[item.track.trackId] = localTrack.copyWith(
-            clonedTracks: [
-              ...localTrack.clonedTracks,
-              mediaTrackClone,
-            ],
+            clonedTracks: [...localTrack.clonedTracks, mediaTrackClone],
           );
         }
       } else {
@@ -381,10 +403,7 @@ class RtcManager extends Disposable {
     );
 
     final sender = transceiversManager
-        .getWith(
-          videoSender.trackType,
-          videoSender.publishOptionId,
-        )
+        .getWith(videoSender.trackType, videoSender.publishOptionId)
         ?.sender;
 
     if (sender == null) {
@@ -481,14 +500,13 @@ class RtcManager extends Disposable {
     final trackIds = [...tracks.keys];
     await Future.wait(
       trackIds.map(
-        (trackId) => unpublishTrack(trackId: trackId).catchError((
-          Object e,
-          StackTrace stk,
-        ) {
-          _logger.e(
-            () => '[dispose] unpublishTrack failed for $trackId: $e\n$stk',
-          );
-        }),
+        (trackId) => unpublishTrack(trackId: trackId).catchError(
+          (Object e, StackTrace stk) {
+            _logger.e(
+              () => '[dispose] unpublishTrack failed for $trackId: $e\n$stk',
+            );
+          },
+        ),
       ),
     );
 
@@ -500,18 +518,24 @@ class RtcManager extends Disposable {
 
     await Future.wait([
       if (publisher != null)
-        publisher!.dispose().catchError((Object e, StackTrace stk) {
-          _logger.e(
-            () => '[dispose] publisher.dispose failed: $e\n$stk',
-          );
+        publisher!.dispose().catchError((
+          Object e,
+          StackTrace stk,
+        ) {
+          _logger.e(() => '[dispose] publisher.dispose failed: $e\n$stk');
         }),
-      subscriber.dispose().catchError((Object e, StackTrace stk) {
-        _logger.e(
-          () => '[dispose] subscriber.dispose failed: $e\n$stk',
-        );
+      subscriber.dispose().catchError((
+        Object e,
+        StackTrace stk,
+      ) {
+        _logger.e(() => '[dispose] subscriber.dispose failed: $e\n$stk');
       }),
     ]);
 
+    await pcFactory.dispose().catchError((Object e, StackTrace stk) {
+      _logger.w(() => '[dispose] pcFactory.dispose failed: $e\n$stk');
+    });
+
     return super.dispose();
   }
 
@@ -579,9 +603,7 @@ extension PublisherRtcManager on RtcManager {
     return transceiverInitIndex.toString();
   }
 
-  Future<List<RtcTrackInfo>> getAnnouncedTracks({
-    String? sdp,
-  }) async {
+  Future<List<RtcTrackInfo>> getAnnouncedTracks({String? sdp}) async {
     final finalSdp = sdp ?? (await publisher?.pc.getLocalDescription())?.sdp;
     final infos = <RtcTrackInfo>[];
 
@@ -1121,9 +1143,11 @@ extension PublisherRtcManager on RtcManager {
     }
 
     try {
+      final nativeFactory = await pcFactory.ensureNativeFactory();
       final audioTrack = await RtcLocalTrack.audio(
         trackIdPrefix: publisherId!,
         constraints: constraints ?? _defaultAudioConstraints,
+        nativeFactory: nativeFactory,
       );
 
       return Result.success(audioTrack);
@@ -1145,9 +1169,11 @@ extension PublisherRtcManager on RtcManager {
     }
 
     try {
+      final nativeFactory = await pcFactory.ensureNativeFactory();
       final videoTrack = await RtcLocalTrack.camera(
         trackIdPrefix: publisherId!,
         constraints: constraints,
+        nativeFactory: nativeFactory,
       );
 
       return Result.success(videoTrack);
@@ -1171,9 +1197,11 @@ extension PublisherRtcManager on RtcManager {
     }
 
     try {
+      final nativeFactory = await pcFactory.ensureNativeFactory();
       final screenShareTrack = await RtcLocalTrack.screenShare(
         trackIdPrefix: publisherId!,
         constraints: constraints,
+        nativeFactory: nativeFactory,
       );
 
       return Result.success(screenShareTrack);
@@ -1186,9 +1214,7 @@ extension PublisherRtcManager on RtcManager {
   Future<Result<RtcLocalCameraTrack>> setTrackFacingMode({
     required FacingMode facingMode,
   }) async {
-    _logger.d(
-      () => '[setTrackFacingMode] facingMode: $facingMode',
-    );
+    _logger.d(() => '[setTrackFacingMode] facingMode: $facingMode');
 
     final track = getPublisherTrackByType(SfuTrackType.video);
     if (track == null) return Result.error('Track not found');
@@ -1204,9 +1230,7 @@ extension PublisherRtcManager on RtcManager {
 
     final updatedTrack = await track.recreate(
       transceivers,
-      mediaConstraints: track.mediaConstraints.copyWith(
-        facingMode: facingMode,
-      ),
+      mediaConstraints: track.mediaConstraints.copyWith(facingMode: facingMode),
     );
 
     tracks[updatedTrack.trackId] = updatedTrack;
@@ -1216,9 +1240,7 @@ extension PublisherRtcManager on RtcManager {
   Future<Result<RtcLocalCameraTrack>> setCameraVideoParameters({
     required RtcVideoParameters params,
   }) async {
-    _logger.d(
-      () => '[setCameraVideoParameters] params: $params',
-    );
+    _logger.d(() => '[setCameraVideoParameters] params: $params');
 
     final track = getPublisherTrackByType(SfuTrackType.video);
 
@@ -1413,11 +1435,13 @@ extension RtcManagerTrackHelper on RtcManager {
   Future<Result<RtcLocalTrack>> setMicrophoneEnabled({
     bool enabled = true,
     AudioConstraints? constraints,
+    TrackDisableMode? disableMode,
   }) {
     return _setTrackEnabled(
       trackType: SfuTrackType.audio,
       enabled: enabled,
       constraints: constraints,
+      disableMode: disableMode,
     );
   }
 
@@ -1462,6 +1486,7 @@ extension RtcManagerTrackHelper on RtcManager {
     required SfuTrackType trackType,
     required bool enabled,
     MediaConstraints? constraints,
+    TrackDisableMode? disableMode,
   }) async {
     final track = getPublisherTrackByType(trackType);
 
@@ -1479,6 +1504,7 @@ extension RtcManagerTrackHelper on RtcManager {
       final toggledTrack = await _toggleTrackMuteState(
         track: track,
         muted: !enabled,
+        disableMode: disableMode,
       );
 
       return Result.success(toggledTrack);
@@ -1499,9 +1525,14 @@ extension RtcManagerTrackHelper on RtcManager {
   Future<RtcLocalTrack> _toggleTrackMuteState({
     required RtcLocalTrack track,
     required bool muted,
+    TrackDisableMode? disableMode,
   }) async {
     if (muted) {
-      await muteTrack(trackId: track.trackId);
+      final stopTrackOnMute = disableMode == null
+          ? null
+          : disableMode == TrackDisableMode.stopTracks;
+
+      await muteTrack(trackId: track.trackId, stopTrackOnMute: stopTrackOnMute);
 
       // If the track is a screen share track, mute the audio track as well.
       if (track.trackType == SfuTrackType.screenShare) {
@@ -1509,7 +1540,10 @@ extension RtcManagerTrackHelper on RtcManager {
           SfuTrackType.screenShareAudio,
         );
         if (screenShareAudioTrack != null) {
-          await muteTrack(trackId: screenShareAudioTrack.trackId);
+          await muteTrack(
+            trackId: screenShareAudioTrack.trackId,
+            stopTrackOnMute: stopTrackOnMute,
+          );
         }
       }
     } else {
@@ -1607,9 +1641,7 @@ extension RtcManagerTrackHelper on RtcManager {
         .listen((event) async {
           _logger.i(() => '[ScreenSharingStartedEvent] received: $event');
 
-          await publishVideoTrack(
-            track: track,
-          );
+          await publishVideoTrack(track: track);
         });
   }
 
diff --git a/packages/stream_video/lib/src/webrtc/rtc_manager_factory.dart b/packages/stream_video/lib/src/webrtc/rtc_manager_factory.dart
index 42899db0e..17b7cf671 100644
--- a/packages/stream_video/lib/src/webrtc/rtc_manager_factory.dart
+++ b/packages/stream_video/lib/src/webrtc/rtc_manager_factory.dart
@@ -15,10 +15,12 @@ class RtcManagerFactory {
     required this.configuration,
     required SdpEditor sdpEditor,
     this.mediaConstraints = const {},
+    AudioConfigurationPolicy? audioConfigurationPolicy,
   }) : pcFactory = StreamPeerConnectionFactory(
          sessionId: sessionId,
          callCid: callCid,
          sdpEditor: sdpEditor,
+         audioConfigurationPolicy: audioConfigurationPolicy,
        );
 
   final _logger = taggedLogger(tag: 'SV:RtcManagerFactory');
@@ -73,6 +75,7 @@ class RtcManagerFactory {
       publishOptions: publishOptions,
       stateManager: stateManager,
       streamVideo: streamVideo,
+      pcFactory: pcFactory,
     );
   }
 }
diff --git a/packages/stream_video/lib/src/webrtc/rtc_media_device/rtc_media_device_notifier.dart b/packages/stream_video/lib/src/webrtc/rtc_media_device/rtc_media_device_notifier.dart
index c3fcf8bc5..a25b286bd 100644
--- a/packages/stream_video/lib/src/webrtc/rtc_media_device/rtc_media_device_notifier.dart
+++ b/packages/stream_video/lib/src/webrtc/rtc_media_device/rtc_media_device_notifier.dart
@@ -246,18 +246,17 @@ class RtcMediaDeviceNotifier {
     return rtc.Helper.regainAndroidAudioFocus();
   }
 
-  /// Reinitializes the audio configuration for the WebRTC instance.
+  /// Refreshes the snapshot the implicit native peer-connection factory will
+  /// use the next time it is built.
   ///
-  /// This is used to reinitialize the audio configuration when the audio configuration policy changes.
-  /// When called after initial setup, it will automatically
-  /// dispose all existing peer connections, tracks, and streams, then recreate
-  /// the audio device module and peer connection factory with the new parameters.
+  /// Already-built factories keep their original configuration: the new
+  /// snapshot only takes effect on subsequent factory builds.
   Future<void> reinitializeAudioConfiguration(
     AudioConfigurationPolicy policy,
   ) async {
     await rtc.WebRTC.initialize(
+      refresh: true,
       options: {
-        'reinitialize': true,
         'bypassVoiceProcessing': policy.bypassVoiceProcessing,
         if (CurrentPlatform.isAndroid)
           'androidAudioConfiguration': policy.getAndroidConfiguration().toMap(),
diff --git a/packages/stream_video/lib/src/webrtc/rtc_track/rtc_local_track.dart b/packages/stream_video/lib/src/webrtc/rtc_track/rtc_local_track.dart
index 167f08c6a..d1f3d75e1 100644
--- a/packages/stream_video/lib/src/webrtc/rtc_track/rtc_local_track.dart
+++ b/packages/stream_video/lib/src/webrtc/rtc_track/rtc_local_track.dart
@@ -30,16 +30,26 @@ class RtcLocalTrack<T extends MediaConstraints> extends RtcTrack {
     required this.mediaConstraints,
     this.stopTrackOnMute = true,
     this.clonedTracks = const [],
+    this.nativeFactory,
     super.videoDimension,
   });
 
+  /// Per-call native factory the track is pinned to. Carried on the track so
+  /// [recreate] can target the same factory. Null on web, which has no
+  /// per-call factory concept.
+  final rtc.NativePeerConnectionFactory? nativeFactory;
+
   static Future<RtcLocalAudioTrack> audio({
     String trackIdPrefix = kLocalTrackIdPrefix,
     AudioConstraints constraints = const AudioConstraints(),
+    rtc.NativePeerConnectionFactory? nativeFactory,
   }) async {
     streamLog.i(_tag, () => 'Creating audio track');
 
-    final stream = await rtc.navigator.mediaDevices.getMedia(constraints);
+    final stream = await rtc.navigator.mediaDevices.getMedia(
+      constraints,
+      nativeFactory: nativeFactory,
+    );
     final audioTrack = stream.getAudioTracks().firstOrNull;
 
     if (audioTrack == null) {
@@ -53,6 +63,7 @@ class RtcLocalTrack<T extends MediaConstraints> extends RtcTrack {
       mediaStream: stream,
       mediaTrack: audioTrack,
       mediaConstraints: constraints,
+      nativeFactory: nativeFactory,
     );
 
     return track;
@@ -61,9 +72,13 @@ class RtcLocalTrack<T extends MediaConstraints> extends RtcTrack {
   static Future<RtcLocalCameraTrack> camera({
     String trackIdPrefix = kLocalTrackIdPrefix,
     CameraConstraints constraints = const CameraConstraints(),
+    rtc.NativePeerConnectionFactory? nativeFactory,
   }) async {
     streamLog.i(_tag, () => 'Creating camera track');
-    final stream = await rtc.navigator.mediaDevices.getMedia(constraints);
+    final stream = await rtc.navigator.mediaDevices.getMedia(
+      constraints,
+      nativeFactory: nativeFactory,
+    );
     final videoTrack = stream.getVideoTracks().firstOrNull;
 
     if (videoTrack == null) {
@@ -86,6 +101,7 @@ class RtcLocalTrack<T extends MediaConstraints> extends RtcTrack {
       mediaStream: stream,
       mediaTrack: videoTrack,
       mediaConstraints: updatedConstraints,
+      nativeFactory: nativeFactory,
     );
 
     return track;
@@ -94,10 +110,14 @@ class RtcLocalTrack<T extends MediaConstraints> extends RtcTrack {
   static Future<RtcLocalScreenShareTrack> screenShare({
     String trackIdPrefix = kLocalTrackIdPrefix,
     ScreenShareConstraints constraints = const ScreenShareConstraints(),
+    rtc.NativePeerConnectionFactory? nativeFactory,
   }) async {
     streamLog.i(_tag, () => 'Creating screen share track');
 
-    final stream = await rtc.navigator.mediaDevices.getMedia(constraints);
+    final stream = await rtc.navigator.mediaDevices.getMedia(
+      constraints,
+      nativeFactory: nativeFactory,
+    );
     final videoTrack = stream.getVideoTracks().firstOrNull;
 
     if (videoTrack == null) {
@@ -111,6 +131,7 @@ class RtcLocalTrack<T extends MediaConstraints> extends RtcTrack {
       mediaStream: stream,
       mediaTrack: videoTrack,
       mediaConstraints: constraints,
+      nativeFactory: nativeFactory,
     );
 
     return track;
@@ -202,6 +223,7 @@ class RtcLocalTrack<T extends MediaConstraints> extends RtcTrack {
     bool? stopTrackOnMute,
     RtcVideoDimension? videoDimension,
     List<rtc.MediaStreamTrack>? clonedTracks,
+    rtc.NativePeerConnectionFactory? nativeFactory,
   }) {
     return RtcLocalTrack(
       trackIdPrefix: trackIdPrefix ?? this.trackIdPrefix,
@@ -212,6 +234,7 @@ class RtcLocalTrack<T extends MediaConstraints> extends RtcTrack {
       stopTrackOnMute: stopTrackOnMute ?? this.stopTrackOnMute,
       videoDimension: videoDimension ?? this.videoDimension,
       clonedTracks: clonedTracks ?? this.clonedTracks,
+      nativeFactory: nativeFactory ?? this.nativeFactory,
     );
   }
 
@@ -228,8 +251,12 @@ class RtcLocalTrack<T extends MediaConstraints> extends RtcTrack {
     // Use the current constraints if none are provided.
     final constraints = mediaConstraints ?? this.mediaConstraints;
 
-    // Create a new track with the new constraints.
-    final newStream = await rtc.navigator.mediaDevices.getMedia(constraints);
+    // Create a new track with the new constraints, pinned to the same
+    // per-call factory the original was attached to.
+    final newStream = await rtc.navigator.mediaDevices.getMedia(
+      constraints,
+      nativeFactory: nativeFactory,
+    );
     final newTrack = newStream.getTracks().first;
     final clonedTracks = <rtc.MediaStreamTrack>[];
 
diff --git a/packages/stream_video/lib/stream_video.dart b/packages/stream_video/lib/stream_video.dart
index 6b920b847..3fe92ce7c 100644
--- a/packages/stream_video/lib/stream_video.dart
+++ b/packages/stream_video/lib/stream_video.dart
@@ -51,6 +51,7 @@ export 'src/webrtc/media/media_constraints.dart';
 export 'src/webrtc/model/rtc_video_dimension.dart';
 export 'src/webrtc/model/rtc_video_parameters.dart';
 export 'src/webrtc/model/stats/rtc_stats_models.dart';
+export 'src/webrtc/model/track_disable_mode.dart';
 export 'src/webrtc/peer_type.dart';
 export 'src/webrtc/rtc_media_device/rtc_media_device.dart';
 export 'src/webrtc/rtc_media_device/rtc_media_device_notifier.dart';
diff --git a/packages/stream_video/pubspec.yaml b/packages/stream_video/pubspec.yaml
index 29e79c54b..d94e7e77f 100644
--- a/packages/stream_video/pubspec.yaml
+++ b/packages/stream_video/pubspec.yaml
@@ -31,7 +31,10 @@ dependencies:
   rxdart: ^0.28.0
   sdp_transform: ^0.3.2
   state_notifier: ^1.0.0
-  stream_webrtc_flutter: ^2.2.6
+  stream_webrtc_flutter:
+    git:
+      url: https://github.com/GetStream/webrtc-flutter.git
+      ref: chore/per-call-pc-factory
   synchronized: ^3.1.0
   system_info2: ^4.0.0
   tart: ^0.6.0
diff --git a/packages/stream_video/test/src/call/fixtures/call_test_helpers.dart b/packages/stream_video/test/src/call/fixtures/call_test_helpers.dart
index 4a7655c02..7720a4690 100644
--- a/packages/stream_video/test/src/call/fixtures/call_test_helpers.dart
+++ b/packages/stream_video/test/src/call/fixtures/call_test_helpers.dart
@@ -171,6 +171,7 @@ MockStreamVideo setupMockStreamVideo({ClientState? clientState}) {
     () => streamVideo.currentUser,
   ).thenReturn(SampleCallData.defaultUserInfo);
   when(streamVideo.isAudioProcessorConfigured).thenReturn(false);
+  when(() => streamVideo.activeCalls).thenReturn(const <Call>[]);
 
   return streamVideo;
 }
@@ -359,6 +360,8 @@ MockSessionFactory setupMockSessionFactory({MockCallSession? callSession}) {
       onReconnectionNeeded: any(named: 'onReconnectionNeeded'),
       clientPublishOptions: any(named: 'clientPublishOptions'),
       streamVideo: any(named: 'streamVideo'),
+      leftoverTraceRecords: any(named: 'leftoverTraceRecords'),
+      audioConfigurationPolicy: any(named: 'audioConfigurationPolicy'),
     ),
   ).thenAnswer(
     (_) => Future.value(callSession ?? setupMockCallSession()),
diff --git a/packages/stream_video_filters/pubspec.yaml b/packages/stream_video_filters/pubspec.yaml
index 6a1fbb6dc..dbf35670e 100644
--- a/packages/stream_video_filters/pubspec.yaml
+++ b/packages/stream_video_filters/pubspec.yaml
@@ -15,7 +15,10 @@ dependencies:
     sdk: flutter
   plugin_platform_interface: ^2.0.2
   stream_video: ^1.3.3
-  stream_webrtc_flutter: ^2.2.6
+  stream_webrtc_flutter:
+    git:
+      url: https://github.com/GetStream/webrtc-flutter.git
+      ref: chore/per-call-pc-factory
 
 dev_dependencies:
   flutter_lints: ^6.0.0
diff --git a/packages/stream_video_flutter/example/pubspec.yaml b/packages/stream_video_flutter/example/pubspec.yaml
index a2f50cdda..f30d04a8a 100644
--- a/packages/stream_video_flutter/example/pubspec.yaml
+++ b/packages/stream_video_flutter/example/pubspec.yaml
@@ -31,7 +31,10 @@ dependencies:
   stream_video: ^1.3.3
   stream_video_flutter: ^1.3.3
   stream_video_push_notification: ^1.3.3
-  stream_webrtc_flutter: ^2.2.6
+  stream_webrtc_flutter:
+    git:
+      url: https://github.com/GetStream/webrtc-flutter.git
+      ref: chore/per-call-pc-factory
 
 dependency_overrides:
   stream_video:
diff --git a/packages/stream_video_flutter/pubspec.yaml b/packages/stream_video_flutter/pubspec.yaml
index 0035bb222..c8f89bd93 100644
--- a/packages/stream_video_flutter/pubspec.yaml
+++ b/packages/stream_video_flutter/pubspec.yaml
@@ -25,7 +25,10 @@ dependencies:
   rate_limiter: ^1.0.0
   rxdart: ^0.28.0
   stream_video: ^1.3.3
-  stream_webrtc_flutter: ^2.2.6
+  stream_webrtc_flutter:
+    git:
+      url: https://github.com/GetStream/webrtc-flutter.git
+      ref: chore/per-call-pc-factory
   visibility_detector: ^0.4.0+2
 
 dev_dependencies:
diff --git a/packages/stream_video_noise_cancellation/pubspec.yaml b/packages/stream_video_noise_cancellation/pubspec.yaml
index a032ebf80..cacbdcc76 100644
--- a/packages/stream_video_noise_cancellation/pubspec.yaml
+++ b/packages/stream_video_noise_cancellation/pubspec.yaml
@@ -15,7 +15,10 @@ dependencies:
     sdk: flutter
   plugin_platform_interface: ^2.0.2
   stream_video: ^1.3.3
-  stream_webrtc_flutter: ^2.2.6
+  stream_webrtc_flutter:
+    git:
+      url: https://github.com/GetStream/webrtc-flutter.git
+      ref: chore/per-call-pc-factory
 
 dev_dependencies:
   flutter_lints: ^6.0.0
diff --git a/packages/stream_video_push_notification/pubspec.yaml b/packages/stream_video_push_notification/pubspec.yaml
index da2341e52..0c2d078ae 100644
--- a/packages/stream_video_push_notification/pubspec.yaml
+++ b/packages/stream_video_push_notification/pubspec.yaml
@@ -23,7 +23,10 @@ dependencies:
   shared_preferences: ^2.5.3
   stream_video: ^1.3.3
   stream_video_flutter: ^1.3.3
-  stream_webrtc_flutter: ^2.2.6
+  stream_webrtc_flutter:
+    git:
+      url: https://github.com/GetStream/webrtc-flutter.git
+      ref: chore/per-call-pc-factory
   uuid: ^4.5.1
 
 dev_dependencies: