Hi everyone, I’m building a React Native iOS app where I’m integrating Wazo (native WebRTC) and Jitsi (WebView / WebRTC). Use case: Wazo is used to maintain a background call session (mainly signaling + audio keep-alive). Jitsi is used in the foreground for video calls. Problem: When Jitsi starts, it takes control of the microphone and camera. The Wazo call disconnects after ~5 minutes (likely due to media / audio session conflict). Even if Wazo audio/video is muted or tracks are disabled, the session still drops. My questions: Is it officially supported or recommended to run two WebRTC stacks (Wazo + Jitsi) simultaneously on iOS? Can Wazo stay connected without active audio/video tracks while Jitsi uses mic/camera? Is there a way to release Wazo media streams temporarily (but keep signaling alive) while Jitsi is loading or active? Are there any AVAudioSession / background mode limitations on iOS that make this impossible by design? If this is not supported, what is the recommended architecture (single WebRTC pipeline, switching media ownership, etc.)? Environment: iOS (React Native) Wazo SDK (native WebRTC) Jitsi Meet (WebView) CallKit + PushKit enabled Any guidance, documentation, or real-world experience would be greatly appreciated. Thanks in advance 🙏

I’m facing a strange audio routing issue that seems specific to iPhone 14 Pro / Pro Max. I’m using LiveKit (WebRTC) in a React Native app, which uses AVAudioSession internally for audio capture (VoIP / call-style usage). 🔍 What’s happening: I’m using an external USB microphone. On these devices: iPhone 11 → ✅ USB mic works iPhone 13 → ✅ USB mic works iPhone 17 Pro → ✅ USB mic works iPhone 14 Pro Max → ❌ USB mic does NOT work On iPhone 14 Pro Max: The same USB mic: ✅ Works in Voice Memos ✅ Works in Instagram Live ❌ Does NOT appear as an input option in my app ❌ Does NOT work in WhatsApp / Instagram calls Also: In my app on iPhone 14 Pro Max, iOS does not show the audio input selector UI On iPhone 17 Pro, the same app and same build does show the selector and the USB mic works ⚙️ My audio session config ( LiveKit ): await AudioSession.setAppleAudioConfiguration({ audioCategory: 'playAndRecord', audioMode: 'default', audioCategoryOptions: ['allowBluetooth', 'defaultToSpeaker'], }); await AudioSession.startAudioSession(); ❓ My questions: Is this a known limitation or behavior specific to iPhone 14 Pro / Pro Max? Does iPhone 14 Pro have different audio routing rules for call / VoIP mode compared to other devices? Why does the same USB mic work in recording apps (Voice Memos, Instagram Live) but not in call-style apps (LiveKit, WhatsApp, Instagram call)? Is there any documented difference in AVAudioSession behavior on iPhone 14 Pro regarding external USB audio inputs?

In my application, I use CallKit and have supportsHolding = true set. During my phone call, another call comes in (e.g., GSM). I accept the incoming call and put the current call on hold. If I end the active call myself, everything is fine, and CallKit calls the method provider(_ provider: CXProvider, didActivate audioSession: AVAudioSession). However, if the other party ends the call, the second call remains on hold. In the application, the user clicks on unhold, and I notify CallKit that the hold has ended. But in this case, the didActivate method is not called at all. If I try to activate the audio myself after unhold, I receive the error: Domain=NSOSStatusErrorDomain Code=561017449 "Session activation failed" UserInfo={NSLocalizedDescription=Session activation failed} AVAudioSessionErrorInsufficientPriority == NSOSStatusErrorDomain Code: 561017449 What needs to be done for CallKit to activate my audio?

I can reproduce the bug that CallKit doesn't active audiosession after the outgoing call put on hold because of an incoming call. VoIP calling with CallKit Steps to reproduce: Download SpeakerBox example app from the link above and start it with XCode Start a new outgoing call Call your phone from other phone Hold and Accept the call After a few secs finish the call from the other phone The outgoing call will be still on hold Click on the call and click Toggle Hold The call won't be active again because the audiosession is activated. Logs: Received provider(_:didDeactivate:) Received provider(_:didDeactivate:) Received provider(_:didDeactivate:) Received provider(_:didDeactivate:) Received provider(_:didDeactivate:) Requested transaction successfully Starting audio Type: stdio AURemoteIO.cpp:1162 failed: 561017449 (enable 3, outf< 1 ch, 44100 Hz, Float32> inf< 1 ch, 44100 Hz, Float32>) Type: Error | Timestamp: 2024-08-15 12:20:29.949437+02:00 | Process: Speakerbox | Library: libEmbeddedSystemAUs.dylib | Subsystem: com.apple.coreaudio | Category: aurioc | TID: 0x19540d AVAEInternal.h:109 [AVAudioEngineGraph.mm:1344:Initialize: (err = PerformCommand(*outputNode, kAUInitialize, NULL, 0)): error 561017449 Type: Error | Timestamp: 2024-08-15 12:20:29.949619+02:00 | Process: Speakerbox | Library: AVFAudio | Subsystem: com.apple.avfaudio | Category: avae | TID: 0x19540d Couldn't start Apple Voice Processing IO: Error Domain=com.apple.coreaudio.avfaudio Code=561017449 "(null)" UserInfo={failed call=err = PerformCommand(*outputNode, kAUInitialize, NULL, 0)} Type: Notice | Timestamp: 2024-08-15 12:20:29.949730+02:00 | Process: Speakerbox | Library: Speakerbox | TID: 0x19540d Route change: Type: Notice | Timestamp: 2024-08-15 12:20:30.167498+02:00 | Process: Speakerbox | Library: Speakerbox | TID: 0x19540d ReasonUnknown Type: Notice | Timestamp: 2024-08-15 12:20:30.167549+02:00 | Process: Speakerbox | Library: Speakerbox | TID: 0x19540d Previous route: Type: Notice | Timestamp: 2024-08-15 12:20:30.167568+02:00 | Process: Speakerbox | Library: Speakerbox | TID: 0x19540d <AVAudioSessionRouteDescription: 0x302c00bc0, inputs = ( "<AVAudioSessionPortDescription: 0x302c01330, type = MicrophoneBuiltIn; name = iPhone Mikrofon; UID = Built-In Microphone; selectedDataSource = (null)>" ); outputs = ( "<AVAudioSessionPortDescription: 0x302c004d0, type = Receiver; name = Vev\U0151; UID = Built-In Receiver; selectedDataSource = (null)>" )> Type: Notice | Timestamp: 2024-08-15 12:20:30.167626+02:00 | Process: Speakerbox | Library: Speakerbox | TID: 0x19540d

Currently, I have successfully used ChannelMap to map hardware input channels and obtained audio data from the hardware device's MIC and OTG inputs. Additionally, I have used ChannelMap to map output channels to freely feed data for playback to each output channel. However, I now have a problem. I have a hardware device that only has output channels (no input channels), and the system has set this hardware device as the default playback device. In this case, how can I obtain the audio data being played to the output channels for modification?

I'm seeing crashes in _MPRemoteCommandEventDispatch on iOS 26.x devices in 3 apps. According to Bugsnag logs they are: NSInternalInconsistencyException: event dispatch <_MPRemoteCommandEventDispatch: <MPRemoteCommandEvent: 0x11c049500 commandID=THV0 command=<MPRemoteCommand: 0x109ad1ea0 type=Play (0) enabled=YES handlers=[0x109b6a310]> sourceID=(null) ([HostedRoutingSessionDataSource] handleControlSendingCommand<2W5E>)> state:201> deallocated without calling continuation I attached a log from Xcode organizer matching Bugsnag crash. mpr_remote_command_event.crash When I set the brakpoint on the -[_MPRemoteCommandEventDispatch dealloc] I can see it it's hit every time I tap play or pause on locked screen play button. Thread 0 Crashed: 0 libsystem_kernel.dylib 0x00000002370420cc __pthread_kill + 8 (:-1) 1 libsystem_pthread.dylib 0x00000001e975c810 pthread_kill + 268 (pthread.c:1721) 2 libsystem_c.dylib 0x0000000198f8ff64 abort + 124 (abort.c:122) 3 libc++abi.dylib 0x000000018a7cf808 __abort_message + 132 (abort_message.cpp:66) 4 libc++abi.dylib 0x000000018a7be484 demangling_terminate_handler() + 304 (cxa_default_handlers.cpp:76) 5 libobjc.A.dylib 0x000000018a6cff78 _objc_terminate() + 156 (objc-exception.mm:496) 6 xxxxxxxxxxxxxx 0x00000001003a7db8 CPPExceptionTerminate() + 416 (BSG_KSCrashSentry_CPPException.mm:156) 7 libc++abi.dylib 0x000000018a7cebdc std::__terminate(void (*)()) + 16 (cxa_handlers.cpp:59) 8 libc++abi.dylib 0x000000018a7ceb80 std::terminate() + 108 (cxa_handlers.cpp:88) 9 CoreFoundation 0x000000018d7341c4 __CFRunLoopPerCalloutARPEnd + 256 (CFRunLoop.c:769) 10 CoreFoundation 0x000000018d70bb5c __CFRunLoopRun + 1976 (CFRunLoop.c:3179) 11 CoreFoundation 0x000000018d70aa6c _CFRunLoopRunSpecificWithOptions + 532 (CFRunLoop.c:3462) 12 GraphicsServices 0x000000022e31c498 GSEventRunModal + 120 (GSEvent.c:2049) 13 UIKitCore 0x00000001930ceba4 -[UIApplication _run] + 792 (UIApplication.m:3902) 14 UIKitCore 0x0000000193077a78 UIApplicationMain + 336 (UIApplication.m:5577) 15 xxxxxxxxxxxxxx 0x00000001000c0134 main + 308 (main.swift:15) 16 dyld 0x000000018a722e28 start + 7116 (dyldMain.cpp:1477) Is the crash happening when the app is being terminated? Thank you!

Hi, We've noticed that this issue occurs more frequently after upgrading to iOS 18.4.1 and can result in one-way audio. Our app uses CallKit with WebRTC to establish VoIP connections. However, on iOS 18.4.1, CallKit no longer triggers: func provider(_ provider: CXProvider, didActivate audioSession: AVAudioSession) We're currently comparing the occurrence rate across different iOS versions to better understand the impact. Could you please help analyze the root cause of this issue?

I am facing a severe audio routing instability issue when using CallKit and the Zego Express SDK on iOS. The problem is that the audio route immediately reverts from the Speaker back to the Earpiece, effectively disabling the Speaker button functionality.📝 Observed BehaviorWhen the user taps the native CallKit Speaker Button, the audio route is correctly changed to the Speaker, but then instantly flips back to the Receiver (Earpiece), as shown in the system log captured via AVAudioSession.routeChangeNotification monitoring.🧾 Log Evidence (Flapping Occurs in 0.4 seconds)The following log snippet clearly illustrates the system overriding the user's action (reason: 4) with an unexpected CategoryChange (reason: 3) event: TimestampComponentReason CodeDescriptionRouteIs Speaker16:31:18.009[CallKitManager]4Override (CallKit/ControlCenter)Loa ngoài (Speaker)true16:31:18.411[CallKitManager]3CategoryChangeThiết bị nhận (Receiver)false

I am having an issue with the code that I posted below. I capture voice in my CarPlay app, then allow the user to have it read back to them using AVSpeechUtterance. This works fine on some cars, but many of my beta testers report no audio being played. I have also experienced this in a rental car where the audio was either too quiet or the audio didn't play. Does anyone see any issue with the code that I posted? This is for CarPlay specifically. class CarPlayTextToSpeechService: NSObject, ObservableObject, AVSpeechSynthesizerDelegate { private var speechSynthesizer = AVSpeechSynthesizer() static let shared = CarPlayTextToSpeechService() /// Completion callback private var completionCallback: (() -> Void)? override init() { super.init() speechSynthesizer.delegate = self } func configureAudioSession() { do { try AVAudioSession.sharedInstance().setCategory(.playback, mode: .voicePrompt, options: [.duckOthers, .interruptSpokenAudioAndMixWithOthers, .allowBluetoothHFP]) } catch { print("Failed to set audio session category: \(error.localizedDescription)") } } public func speak(_ text: String, completion: (() -> Void)? = nil) { self.configureAudioSession() // Store the completion callback self.completionCallback = completion Task(priority: .high) { let speechUtterance = AVSpeechUtterance(string: text) let langCode = Locale.preferredLocalLanguageCountryCode if langCode == "en-US" { speechUtterance.voice = AVSpeechSynthesisVoice(identifier: AVSpeechSynthesisVoiceIdentifierAlex) } else { speechUtterance.voice = AVSpeechSynthesisVoice(language: langCode) } try AVAudioSession.sharedInstance().setActive(true, options: .notifyOthersOnDeactivation) speechSynthesizer.speak(speechUtterance) } } func speechSynthesizer(_ synthesizer: AVSpeechSynthesizer, didFinish utterance: AVSpeechUtterance) { Task { stopSpeech() try AVAudioSession.sharedInstance().setActive(false) } // Call completion callback if available self.completionCallback?() self.completionCallback = nil } func stopSpeech() { speechSynthesizer.stopSpeaking(at: .immediate) } }

I'm implementing the PushToTalk framework and have encountered an issue where channelManager(_:didActivate:) is not called under specific circumstances. What works: App is in foreground, receives PTT push → didActivate is called ✅ App receives audio in foreground, then is backgrounded → subsequent pushes trigger didActivate ✅ What doesn't work: App is launched, user joins channel, then immediately backgrounds PTT push arrives while app is backgrounded incomingPushResult is called, I return .activeRemoteParticipant(participant) The system UI shows the speaker name correctly However, didActivate is never called Audio data arrives via WebSocket but cannot be played (no audio session) Setup: Channel joined successfully before backgrounding UIBackgroundModes includes push-to-talk No manual audio session activation (setActive) anywhere in my code AVAudioEngine setup only happens inside didActivate delegate method Issue persists even after channel restoration via channelDescriptor(restoredChannelUUID:) Question: Is this expected behavior or a bug? If expected, what's the correct approach to handle incoming PTT audio when the app is backgrounded and hasn't received audio while in the foreground yet?

Hi everyone, I'm running into an issue with AVAudioRecorder when handling interruptions such as phone calls or alarms. Problem: When the app is recording audio and an interruption occurs: I handle the interruption with audioRecorder?.pause() inside AVAudioSession.interruptionNotification (on .began). On .ended, I check for .shouldResume and call audioRecorder?.record() again. The recorder resumes successfully, but only the audio recorded after the interruption is saved. The audio recorded before the interruption is lost, even though I'm using the same file URL and not recreating the recorder. Repro: Start a recording with AVAudioRecorder Simulate a system interruption (e.g., incoming call) Resume recording after the interruption Stop and inspect the output audio file Expected: Full audio (before and after interruption) should be saved. Actual: Only the audio after interruption is saved; the earlier part is missing Notes: According to the documentation, calling .record() after .pause() should resume recording into the same file. I confirmed that the file URL does not change, and I do not recreate the recorder instance. No error is thrown by the system during this process. This behavior happens consistently when the app is interrupted and resumed. Question: Is this a known issue? Is there a recommended workaround for preserving the full recording when interruptions happen? Thanks in advance!

My app utilizes background audio to play music files. I have the audio background mode enabled and I initialize the AVAudioSession in playback mode with the mixWithOthers option. And it usually works great while the app is backgrounded. I listen for audio interruptions as well as route changes and I am able to handle them appropriately and I can usually resume my background audio no problem. I discovered an issue while connected to CarPlay though. Roughly 50% of the time when I disconnect from a phone call while connected to CarPlay I get the following error after calling the play() method of my AVAudioPlayer instance: "ATAudioSessionClientImpl.mm:281 activation failed. status = 561015905" If I instead try to start a new audio session I get a similar error: Error Domain=NSOSStatusErrorDomain Code=561015905 "Session activation failed" UserInfo={NSLocalizedDescription=Session activation failed} Like I said, this isn't reproducible 100% of the time and is so far only seen while connected to CarPlay. I don't think Im forgetting so additional capability or plist setting, but if anyone has any clues it would be greatly appreciated. Otherwise this is likely just a bug that I need to report to Apple. One very important note, and reason I believe it's just a bug, is that while I was testing I found that other music apps like Spotify will also fail to resume their audio at the same time my app fails. Another important detail is that when it works successfully I receive the audio session interruption ended notification, and when it doesn't work I only receive a route configuration change or route override notification. From there I am able to still successfully granted background time to execute code, but my call to resume audio fails with the above mentioned error codes.

Hello, I am wondering if it is possible to have audio from my AirPods be sent to my speech to text service and at the same time have the built in mic audio input be sent to recording a video? I ask because I want my users to be able to say "CAPTURE" and I start recording a video (with audio from the built in mic) and then when the user says "STOP" I stop the recording.

My workout watch app supports audio playback during exercise sessions. When users carry both Apple Watch, iPhone, and AirPods, with AirPods connected to the iPhone, I want to route audio from Apple Watch to AirPods for playback. I've implemented this functionality using the following code. try? session.setCategory(.playback, mode: .default, policy: .longFormAudio, options: []) try await session.activate() When users are playing music on iPhone and trigger my code in the watch app, Apple Watch correctly guides users to select AirPods, pauses the iPhone's music, and plays my audio. However, when playback finishes and I end the session using the code below: try session.setActive(false, options:[.notifyOthersOnDeactivation]) the iPhone doesn't automatically resume the previously interrupted music playback—it requires manual intervention. Is this expected behavior, or am I missing other important steps in my code?

Hello! Thank you for bringing the new iPhone experience with the PushToTalk framework. I have a working walkie talkie app based on the PushToTalk framework. Everything works fine except for an intermittent bug that I face from time to time on different devices with different iOS versions, from iOS 18 to iOS 26.2 Beta. Sometimes the app goes into a state where the AVAudioInputNode input node tap returns buffers with a constant size that contain only silence. Leaving and rejoining a channel helps, but relaunching or reinstalling (from Xcode) the app does not. Rebooting the device or deleting and reinstalling the app also helps. I do not activate the audio session in my app. I only configure it on launch using setCategory(.playAndRecord, options: [.defaultToSpeaker, .allowBluetooth]) So the flow is: channelManager?.requestBeginTransmitting(channelUUID: globalChannelUUID) func channelManager( _ channelManager: PTChannelManager, channelUUID: UUID, didBeginTransmittingFrom source: PTChannelTransmitRequestSource ) func channelManager( _ channelManager: PTChannelManager, didActivate audioSession: AVAudioSession ) { /// ... installTapAndStart() } private func installTapAndStart() { let inputNode = audioEngine.inputNode let hardwareFormat = inputNode.outputFormat(forBus: 0) guard let targetFormat = AVAudioFormat( commonFormat: .pcmFormatFloat32, sampleRate: configuration.audioSampleRate, channels: configuration.audioChannelsCount, interleaved: true ) else { handleError(RecorderError.invalidAudioFormat) return } let converter = AVAudioConverter(from: hardwareFormat, to: targetFormat)! print("[QUICOpusRecorder]: installTap") inputNode.installTap(onBus: 0, bufferSize: tapBufferSize, format: hardwareFormat) { [weak self] buffer, _ in guard let self else { return } // Here I handle audio data and sometimes get silence } //... do { audioEngine.prepare() try audioEngine.start() } catch { print(" ⚠️ Audio engine start error: \(error)") handleError(error) } } Moreover, if the app is in the foreground and PushToTalk gets stuck in this “silence bug”, I can avoid relying on the PushToTalk flow and simulate audio session activation manually in code. In this case I do not request a transmission and do not use any PushToTalk related code, and the app captures audio data perfectly. Once I leave the channel and rejoin it again, the issue is fixed and I start to receive non silent buffers of varying size, as expected. It works for a while. It can work fine for a day or more, communicating without launching the app, or with the app in the foreground. But it can also go into the “silence” state 30 minutes after working normally. I have no clue why this happens. The only thing I notice is that when the app is in this “stuck silence bug” state, iOS does not play its “chirp” system sound when audio recording starts. P.S. Channel descriptor restoration code: extension PushToTalkEngine: PTChannelRestorationDelegate { public func channelDescriptor(restoredChannelUUID channelUUID: UUID) -> PTChannelDescriptor { print("☀️ \(#function) channelUUID: \(channelUUID)") Task { @MainActor in // Here I fetch more detailed channel data asynchronously do { await initChannelManagerIfNeeded(channelUUID: channelUUID) let channelDescriptor = currentChannelDescriptor() lastChannelDescriptorName = channelDescriptor.name try await channelManager?.setChannelDescriptor( channelDescriptor, channelUUID: channelUUID ) } catch { handleError(error) } } return PTChannelDescriptor(name: "Loading...", image: nil) } } private func initChannelManagerIfNeeded(channelUUID: UUID? = nil) async { guard let channelUUID = channelUUID ?? currentUser?.globalChannelUUID else { print("❌ No global channel uuid found") return } do { guard channelManager == nil else { try await channelManager?.setTransmissionMode(.halfDuplex, channelUUID: channelUUID) return } channelManager = try await PTChannelManager.channelManager( delegate: self, restorationDelegate: self ) try await channelManager?.setTransmissionMode(.halfDuplex, channelUUID: channelUUID) } catch { handleError(error) } }

I have an iPadOS M-processor application with two different running configurations. In config1, the shared AVAudioSession is configured for .videoChat mode using the built-in microphone. The input/output nodes of the AVAudioEngine are configured with voice processing enabled. The built-in mic is formatted for 1 channel at 48KHz. In config2, the shared AVAudioSession is configured for .measurement mode using an external USB microphone. The input/output nodes of the AVAudioEngine are configured with voice processing disabled. The external mic is formatted for 2 channels at 44.1KHz I've written a configuration manager designed to safely switch between these two configurations. It works by stopping AVAudioEngine and detaching all but the input and output nodes, updating the shared audio session for the desired mic and sample-rates, and setting the appropriate state for voice processing to either true or false as required by the configuration. Finally the new audio graph is constructed by attaching appropriate nodes, connecting them, and re-starting AVAudioEngine I'm experiencing what I believe is a race-condition between switching voice processing on or off and then trying to re-build and start the new audio graph. Even though notifications, which are dumped to the console indicate that my requested input and sample-rate settings are in place, I crash when trying to start the audio engine because the sample-rate is wrong. Investigating further it looks like the switch from remote I/O to voice-processing I/O or vice-versa has not yet actually completed. I introduced a 100ms second delay and that seems to help but is obviously not a reliable way to build software that must work consistently. How can I make sure that what are apparently asynchronous configuration changes to the shared audio session and the input/output nodes have completed before I go on? I tried using route change notifications from the shared AVAudioSession but these lie. They say my preferred mic input and sample-rate setting is in place but when I dump the AVAudioEngine graph to the debugger console, I still see the wrong sample rate assigned to the input/output nodes. Also these are the wrong AU nodes. That is, VPIO is still in place when RIO should be, or vice-versa. How can I make the switch reliable without arbitrary time delays? Is my configuration manager approach appropriate (question for Apple engineers)?

We require assistance in resolving a critical audio design conflict within our Push-to-Talk (PTT) application. Our current volume amplification strategy—which relies on applying a GAIN factor to PCM samples in conjunction with setting the AVAudioSession category to Playback—is working successfully when PTT is used independently. However, upon integrating and reporting the same PTT call through the CallKit framework, this amplification effect is lost. The CallKit integration appears to be forcing a different, non-amplifying audio session category or configuration, negatively impacting the user's perceived call volume. We need guidance on how to maintain the AVAudioSessionCategoryPlayback setting, or an equivalent high-volume configuration, while operating under the control of CallKit.

Hi all, I have been quite stumped on this behavior for a little bit now, so thought it best to share here and see if someone more experience with AVAudioEngine / AVAudioSession can weigh in. Right now I have a AVAudioEngine that I am using to perform some voice chat with and give buffers to play. This works perfectly until route changes start to occur, which causes the AVAudioEngine to reset itself, which then causes all players attached to this engine to be stopped. Once a AVPlayerNode gets stopped due to this (but also any other time), all samples that were scheduled to be played then get purged. Where this becomes confusing for me is the completion handler gets called every time regardless of the sound actually being played. Is there a reliable way to know if a sample needs to be rescheduled after a player has been reset? I am not quite sure in my case what my observer of AVAudioEngineConfigurationChange needs to be doing, as this engine only handles output. All input is through a separate engine for simplicity. Currently I am storing a queue of samples as they get sent to the AVPlayerNode for playback, and after that completion checking if the player isPlaying or not. If it's playing I assume that the sound actually was played- and if not then I leave it in the queue and assume that an observer on the route change or the configuration change will realize there are samples in the queue and reset them Thanks for any feedback!

I'm building a React Native call application using the following combination of libraries: https://github.com/react-native-webrtc/react-native-callkeep https://github.com/react-native-webrtc/react-native-webrtc https://github.com/react-native-webrtc/react-native-voip-push-notification When I press the speaker button on the call screen displayed by CallKit and change it to ON, the speaker button display on the call screen reverts back to OFF after a few seconds. However, when the speaker button display reverts to OFF, the actual audio output route does not return to the earpiece - the audio continues to output from the speaker without any change. Could you please advise on what cases might cause the speaker button display to revert, and if there are any potential solutions?

I want the audio session to always use the built-in microphone. However, when using the setPreferredInput() method like in this example private func enableBuiltInMic() { // Get the shared audio session. let session = AVAudioSession.sharedInstance() // Find the built-in microphone input. guard let availableInputs = session.availableInputs, let builtInMicInput = availableInputs.first(where: { $0.portType == .builtInMic }) else { print("The device must have a built-in microphone.") return } // Make the built-in microphone input the preferred input. do { try session.setPreferredInput(builtInMicInput) } catch { print("Unable to set the built-in mic as the preferred input.") } } and calling that function once in the initializer, the audio session still switches to the external microphone once one is plugged in. The session's preferredInput is nil again at that point, even if the built-in microphone is still listed in availableInputs. So, why is the preferredInput suddenly reset? when would be the appropriate time to set the preferredInput again? Observing the session’s availableInputs did not work and setting the preferredInput again in the routeChangeNotification handler seems a bad choice as it’s already a bit too late then.