diff --git a/android/src/main/java/org/membraneframework/reactnative/MembraneWebRTC.kt b/android/src/main/java/org/membraneframework/reactnative/MembraneWebRTC.kt
index 9dc07b3..efce804 100644
--- a/android/src/main/java/org/membraneframework/reactnative/MembraneWebRTC.kt
+++ b/android/src/main/java/org/membraneframework/reactnative/MembraneWebRTC.kt
@@ -79,7 +79,12 @@ class MembraneWebRTC(val sendEvent: (name: String, data: Map<String, Any?>) -> U
     var onTracksUpdateListeners: MutableList<OnTrackUpdateListener> = mutableListOf()
   }
 
-  fun onDestroy() {
+  fun onModuleCreate(appContext: AppContext){
+    this.appContext = appContext
+    this.audioSwitchManager = AudioSwitchManager(appContext.reactContext!!)
+  }
+
+  fun onModuleDestroy() {
     audioSwitchManager?.stop()
   }
 
@@ -98,7 +103,7 @@ class MembraneWebRTC(val sendEvent: (name: String, data: Map<String, Any?>) -> U
     }
   }
 
-  private fun getSimulcastConfigFromOptions(simulcastConfigMap: com.reactnativemembrane.SimulcastConfig): SimulcastConfig {
+  private fun getSimulcastConfigFromOptions(simulcastConfigMap: org.membraneframework.reactnative.SimulcastConfig): SimulcastConfig {
     val simulcastEnabled = simulcastConfigMap.enabled
     val activeEncodings = simulcastConfigMap.activeEncodings.map { e -> e.toTrackEncoding() }
     return SimulcastConfig(
@@ -137,7 +142,6 @@ class MembraneWebRTC(val sendEvent: (name: String, data: Map<String, Any?>) -> U
   }
 
   fun create() {
-    audioSwitchManager = AudioSwitchManager(appContext?.reactContext!!)
     membraneRTC = MembraneRTC.create(
       appContext = appContext?.reactContext!!, listener = this
     )
diff --git a/android/src/main/java/org/membraneframework/reactnative/MembraneWebRTCModule.kt b/android/src/main/java/org/membraneframework/reactnative/MembraneWebRTCModule.kt
index a9179df..ba86fd2 100644
--- a/android/src/main/java/org/membraneframework/reactnative/MembraneWebRTCModule.kt
+++ b/android/src/main/java/org/membraneframework/reactnative/MembraneWebRTCModule.kt
@@ -91,7 +91,19 @@ class MembraneWebRTCModule : Module() {
     }
 
     OnCreate {
-      membraneWebRTC.appContext = appContext
+      membraneWebRTC.onModuleCreate(appContext)
+    }
+
+    OnDestroy {
+      membraneWebRTC.onModuleDestroy()
+    }
+
+    OnActivityDestroys {
+      membraneWebRTC.disconnect()
+    }
+
+    OnActivityResult { _, result ->
+      membraneWebRTC.onActivityResult(result.requestCode, result.resultCode, result.data)
     }
 
     AsyncFunction("create") Coroutine ({ ->
@@ -272,17 +284,5 @@ class MembraneWebRTCModule : Module() {
     AsyncFunction("getStatistics") { ->
       membraneWebRTC.getStatistics()
     }
-
-    OnActivityResult { _, result ->
-      membraneWebRTC.onActivityResult(result.requestCode, result.resultCode, result.data)
-    }
-
-    OnDestroy {
-      membraneWebRTC.onDestroy()
-    }
-
-    OnActivityDestroys {
-      membraneWebRTC.disconnect()
-    }
   }
 }
diff --git a/src/common/webRTC.ts b/src/common/webRTC.ts
index da2eb78..386e200 100644
--- a/src/common/webRTC.ts
+++ b/src/common/webRTC.ts
@@ -1,14 +1,6 @@
 import { LoggingSeverity, TrackEncoding } from '../MembraneWebRTC.types';
 import MembraneWebRTCModule from '../MembraneWebRTCModule';
 
-/**
- * This function initialize necessary native objects to properly handle sound and video.
- * Call it only once in your app before any other functionality, otherwise package will not work as intended.
- */
-export async function initializeWebRTC() {
-  await MembraneWebRTCModule.create();
-}
-
 /**
  * sets track encoding that server should send to the client library.
  * The encoding will be sent whenever it is available. If chooses encoding is
diff --git a/src/hooks/useWebRTC.ts b/src/hooks/useWebRTC.ts
index 94395fc..5381b2a 100644
--- a/src/hooks/useWebRTC.ts
+++ b/src/hooks/useWebRTC.ts
@@ -5,14 +5,6 @@ import { ConnectionOptions, Metadata } from '../MembraneWebRTC.types';
 import MembraneWebRTCModule from '../MembraneWebRTCModule';
 import { ReceivableEvents, eventEmitter } from '../common/eventEmitter';
 
-/**
- * This function initialize necessary native objects to properly handle sound and video.
- * Call it only once in your app before any other functionality, otherwise package will not work as intended.
- */
-export async function initializeWebRTC() {
-  await MembraneWebRTCModule.create();
-}
-
 /**
  * The hook used to manage a connection with membrane server.
  * @returns An object with functions to manage membrane server connection and `error` if connection failed.
@@ -113,6 +105,9 @@ export function useWebRTC() {
 
         socket.current = _socket;
         webrtcChannel.current = _webrtcChannel;
+
+        await MembraneWebRTCModule.create();
+
         await new Promise<void>((resolve, reject) => {
           _webrtcChannel
             .join()
diff --git a/src/index.tsx b/src/index.tsx
index 8ae067e..6bca02e 100644
--- a/src/index.tsx
+++ b/src/index.tsx
@@ -13,7 +13,6 @@ export {
   updateVideoTrackMetadata,
 } from './common/metadata';
 export {
-  initializeWebRTC,
   changeWebRTCLoggingSeverity,
   setTargetTrackEncoding,
 } from './common/webRTC';