diff --git a/ios/RCTWebRTC/WebRTCModule+RTCMediaStream.m b/ios/RCTWebRTC/WebRTCModule+RTCMediaStream.m
index 49e922987..eafae4394 100644
--- a/ios/RCTWebRTC/WebRTCModule+RTCMediaStream.m
+++ b/ios/RCTWebRTC/WebRTCModule+RTCMediaStream.m
@@ -4,6 +4,7 @@
 #import <WebRTC/RTCMediaConstraints.h>
 #import <WebRTC/RTCMediaStreamTrack.h>
 #import <WebRTC/RTCVideoTrack.h>
+#import <WebRTC/WebRTC.h>
 
 #import "RTCMediaStreamTrack+React.h"
 #import "WebRTCModuleOptions.h"
@@ -214,6 +215,7 @@ - (RTCVideoTrack *)createScreenCaptureVideoTrack {
 
     if (constraints[@"audio"]) {
         audioTrack = [self createAudioTrack:constraints];
+        [self ensureAudioSessionWithRecording];
     }
     if (constraints[@"video"]) {
         videoTrack = [self createVideoTrack:constraints];
@@ -408,15 +410,32 @@ - (RTCVideoTrack *)createScreenCaptureVideoTrack {
         NSString *trackUUID = [[NSUUID UUID] UUIDString];
         if ([originalTrack.kind isEqualToString:@"audio"]) {
             RTCAudioTrack *audioTrack = [self.peerConnectionFactory audioTrackWithTrackId:trackUUID];
+            audioTrack.isEnabled = originalTrack.isEnabled;
             [self.localTracks setObject:audioTrack forKey:trackUUID];
-            return trackUUID;
+            for (NSString* streamId in self.localStreams) {
+                RTCMediaStream* stream = [self.localStreams objectForKey:streamId];
+                for (RTCAudioTrack* track in stream.audioTracks) {
+                    if ([trackID isEqualToString:track.trackId]) {
+                        [stream addAudioTrack:audioTrack];
+                    }
+                }
+            }
         } else {
             RTCVideoTrack *originalVideoTrack = (RTCVideoTrack *)originalTrack;
             RTCVideoSource *videoSource = originalVideoTrack.source;
             RTCVideoTrack *videoTrack = [self.peerConnectionFactory videoTrackWithSource:videoSource trackId:trackUUID];
+            videoTrack.isEnabled = originalTrack.isEnabled;
             [self.localTracks setObject:videoTrack forKey:trackUUID];
-            return trackUUID;
+            for (NSString* streamId in self.localStreams) {
+                RTCMediaStream* stream = [self.localStreams objectForKey:streamId];
+                for (RTCVideoTrack* track in stream.videoTracks) {
+                    if ([trackID isEqualToString:track.trackId]) {
+                        [stream addVideoTrack:videoTrack];
+                    }
+                }
+            }
         }
+        return trackUUID;
     }
     return @"";
 #endif
@@ -518,4 +537,33 @@ - (RTCMediaStreamTrack *)trackForId:(nonnull NSString *)trackId pcId:(nonnull NS
     return peerConnection.remoteTracks[trackId];
 }
 
+- (void)ensureAudioSessionWithRecording {
+  RTCAudioSession* session = [RTCAudioSession sharedInstance];
+
+  // we also need to set default WebRTC audio configuration, since it may be activated after
+  // this method is called
+  RTCAudioSessionConfiguration* config = [RTCAudioSessionConfiguration webRTCConfiguration];
+  // require audio session to be either PlayAndRecord or MultiRoute
+  if (session.category != AVAudioSessionCategoryPlayAndRecord) {
+    [session lockForConfiguration];
+    config.category = AVAudioSessionCategoryPlayAndRecord;
+    config.categoryOptions =
+             AVAudioSessionCategoryOptionAllowAirPlay|
+             AVAudioSessionCategoryOptionAllowBluetooth|
+             AVAudioSessionCategoryOptionAllowBluetoothA2DP|
+             AVAudioSessionCategoryOptionDefaultToSpeaker;
+    config.mode = AVAudioSessionModeVideoChat;
+    NSError* error = nil;
+    bool success = [session setCategory:config.category withOptions:config.categoryOptions error:&error];
+    if (!success) {
+      NSLog(@"ensureAudioSessionWithRecording: setCategory failed due to: %@", [error localizedDescription]);
+    }
+    success = [session setMode:config.mode error:&error];
+    if (!success) {
+      NSLog(@"ensureAudioSessionWithRecording: Error setting category: %@", [error localizedDescription]);
+    }
+    [session unlockForConfiguration];
+  }
+}
+
 @end
diff --git a/package-lock.json b/package-lock.json
index 6d0e61834..0638cea27 100644
--- a/package-lock.json
+++ b/package-lock.json
@@ -1,12 +1,12 @@
 {
   "name": "@stream-io/react-native-webrtc",
-  "version": "125.0.3",
+  "version": "125.0.4-rc.3",
   "lockfileVersion": 2,
   "requires": true,
   "packages": {
     "": {
       "name": "@stream-io/react-native-webrtc",
-      "version": "125.0.3",
+      "version": "125.0.4-rc.3",
       "license": "MIT",
       "dependencies": {
         "base64-js": "1.5.1",
diff --git a/package.json b/package.json
index eb4b396c9..7ad273cec 100644
--- a/package.json
+++ b/package.json
@@ -1,6 +1,6 @@
 {
   "name": "@stream-io/react-native-webrtc",
-  "version": "125.0.3",
+  "version": "125.0.4-rc.3",
   "repository": {
     "type": "git",
     "url": "git+https://github.com/GetStream/react-native-webrtc.git"
diff --git a/stream-react-native-webrtc.podspec b/stream-react-native-webrtc.podspec
index 4697f58ef..c6b495dd8 100644
--- a/stream-react-native-webrtc.podspec
+++ b/stream-react-native-webrtc.podspec
@@ -19,7 +19,7 @@ Pod::Spec.new do |s|
   s.libraries           = 'c', 'sqlite3', 'stdc++'
   s.framework           = 'AudioToolbox','AVFoundation', 'CoreAudio', 'CoreGraphics', 'CoreVideo', 'GLKit', 'VideoToolbox'
   s.dependency          'React-Core'
-  s.dependency          'WebRTC-SDK', '~>125.6422.05'
+  s.dependency          'WebRTC-SDK', '~>125.6422.07'
   # Swift/Objective-C compatibility #https://blog.cocoapods.org/CocoaPods-1.5.0/
   s.pod_target_xcconfig = {
     'DEFINES_MODULE' => 'YES'