Is there a way to configure the APNS notification sound volume to be louder? I am implementing some custom sounds(narrative sentences) for APNS, it does play the custom sound, but the volume of the custom sound is not loud enough even though I had set the device's volume and "RingTone and Alerts" volume to max. I tried to amplify the custom sound file, it does play louder but the result is minimum if I want to maintain the quality of the sound without it been distorted. I tried to use Notification Service Extension, AVAudioPlayer and AVAudioSession to play the sound, it does play louder in max volume compare with relying on default sound payload in APNS, but the problem is AVAudioPlayer and AVAudioSession do not seems to be usable when the application is in background or killed state, is there any other alternative I could use?

I'm working on adding CarPlay support to an audio app and am running into an issue. Occasionally, when a user opens the app from CarPlay while the main app scene is either not connected or is currently in the background, I will receive an error when attempting to activate the audio session. The code below mimics my setup: do { try AVAudioSession.sharedInstance().setCategory(.playback, mode: .spokenAudio) try AVAudioSession.sharedInstance().setActive(true) } catch { print(error) // NSOSStatusErrorDomain - 560557684: Session activation failed } That error code maps to AVAudioSession.ErrorCode.cannotInterruptOthers. Once in this state, all subsequent attempts to play different pieces of content will fail. However, things will start working normally if the user opens the app on their phone and tries again from CarPlay (while the app is in the foreground on their phone). I'm not sure why it would behave this way and want to note that I do have the audio background mode capability enabled. Has anyone else encountered this? Are there any workarounds or changes I could make to prevent this from happening?

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!

I did watch WWDC 2019 Session 716 and understand that an active audio session is key to unlocking low‑level networking on watchOS. I’m configuring my audio session and engine as follows: private func configureAudioSession(completion: @escaping (Bool) -> Void) { let audioSession = AVAudioSession.sharedInstance() do { try audioSession.setCategory(.playAndRecord, mode: .voiceChat, options: []) try audioSession.setActive(true, options: .notifyOthersOnDeactivation) // Retrieve sample rate and configure the audio format. let sampleRate = audioSession.sampleRate print("Active hardware sample rate: \(sampleRate)") audioFormat = AVAudioFormat(standardFormatWithSampleRate: sampleRate, channels: 1) // Configure the audio engine. audioInputNode = audioEngine.inputNode audioEngine.attach(audioPlayerNode) audioEngine.connect(audioPlayerNode, to: audioEngine.mainMixerNode, format: audioFormat) try audioEngine.start() completion(true) } catch { print("Error configuring audio session: \(error.localizedDescription)") completion(false) } } private func setupUDPConnection() { let parameters = NWParameters.udp parameters.includePeerToPeer = true connection = NWConnection(host: "***.***.xxxxx.***", port: 0000, using: parameters) setupNWConnectionHandlers() } private func setupTCPConnection() { let parameters = NWParameters.tcp connection = NWConnection(host: "***.***.xxxxx.***", port: 0000, using: parameters) setupNWConnectionHandlers() } private func setupWebSocketConnection() { guard let url = URL(string: "ws://***.***.xxxxx.***:0000") else { print("Invalid WebSocket URL") return } let session = URLSession(configuration: .default) webSocketTask = session.webSocketTask(with: url) webSocketTask?.resume() print("WebSocket connection initiated") sendAudioToServer() receiveDataFromServer() sendWebSocketPing(after: 0.6) } private func setupNWConnectionHandlers() { connection?.stateUpdateHandler = { [weak self] state in DispatchQueue.main.async { switch state { case .ready: print("Connected (NWConnection)") self?.isConnected = true self?.failToConnect = false self?.receiveDataFromServer() self?.sendAudioToServer() case .waiting(let error), .failed(let error): print("Connection error: \(error.localizedDescription)") DispatchQueue.main.asyncAfter(deadline: .now() + 2) { self?.setupNetwork() } case .cancelled: print("NWConnection cancelled") self?.isConnected = false default: break } } } connection?.start(queue: .main) } Duplex in this context refers to two-way audio transmission simultaneously recording and sending audio while also receiving and playing back incoming audio, similar to a VoIP/SIP call. The setup works fine on the simulator, which suggests that the core logic is correct. However, since the simulator doesn’t fully replicate WatchOS hardware behavior especially for audio sessions and networking issues might arise when running on a real device. The problem likely lies in either the Watch’s actual hardware limitations, permission constraints, or specific audio session configurations. I am reaching out to seek further assistance regarding the challenges I've been experiencing with establishing a UDP, TCP & web socket connection on watchOS using NWConnection for duplex audio streaming. Despite implementing the recommendations provided earlier, I am still encountering difficulties From what I can see, your implementation is focused on streaming audio playback with the server. In my case, I'm looking for a slightly different approach: I want to capture audio and send buffers of a specific size to the server while playing audio simultaneously, essentially achieving full duplex streaming similar to a VOIP call. Additionally, I’d like to ensure that if no external audio route is connected, the Apple Watch speaker is used by default. Any thoughts or insights on adapting this setup for those requirements would be very welcome.

I did watch WWDC 2019 Session 716 and understand that an active audio session is key to unlocking low‑level networking on watchOS. I’m configuring my audio session and engine as follows: private func configureAudioSession(completion: @escaping (Bool) -> Void) { let audioSession = AVAudioSession.sharedInstance() do { try audioSession.setCategory(.playAndRecord, mode: .voiceChat, options: []) try audioSession.setActive(true, options: .notifyOthersOnDeactivation) // Retrieve sample rate and configure the audio format. let sampleRate = audioSession.sampleRate print("Active hardware sample rate: \(sampleRate)") audioFormat = AVAudioFormat(standardFormatWithSampleRate: sampleRate, channels: 1) // Configure the audio engine. audioInputNode = audioEngine.inputNode audioEngine.attach(audioPlayerNode) audioEngine.connect(audioPlayerNode, to: audioEngine.mainMixerNode, format: audioFormat) try audioEngine.start() completion(true) } catch { print("Error configuring audio session: \(error.localizedDescription)") completion(false) } } private func setupUDPConnection() { let parameters = NWParameters.udp parameters.includePeerToPeer = true connection = NWConnection(host: "***.***.xxxxx.***", port: 0000, using: parameters) setupNWConnectionHandlers() } private func setupTCPConnection() { let parameters = NWParameters.tcp connection = NWConnection(host: "***.***.xxxxx.***", port: 0000, using: parameters) setupNWConnectionHandlers() } private func setupWebSocketConnection() { guard let url = URL(string: "ws://***.***.xxxxx.***:0000") else { print("Invalid WebSocket URL") return } let session = URLSession(configuration: .default) webSocketTask = session.webSocketTask(with: url) webSocketTask?.resume() print("WebSocket connection initiated") sendAudioToServer() receiveDataFromServer() sendWebSocketPing(after: 0.6) } private func setupNWConnectionHandlers() { connection?.stateUpdateHandler = { [weak self] state in DispatchQueue.main.async { switch state { case .ready: print("Connected (NWConnection)") self?.isConnected = true self?.failToConnect = false self?.receiveDataFromServer() self?.sendAudioToServer() case .waiting(let error), .failed(let error): print("Connection error: \(error.localizedDescription)") DispatchQueue.main.asyncAfter(deadline: .now() + 2) { self?.setupNetwork() } case .cancelled: print("NWConnection cancelled") self?.isConnected = false default: break } } } connection?.start(queue: .main) } I am reaching out to seek further assistance regarding the challenges I've been experiencing with establishing a UDP, TCP & web socket connection on watchOS using NWConnection for duplex audio streaming. Despite implementing the recommendations provided earlier, I am still encountering difficulties. Or duplex audio streaming not possible on apple watch?

Environment→ ・Device: iPad 10th generation ・OS：**iOS18.3.2 I'm using AVAudioSession to record sound in my application. But I recently came to realize that when the app starts a recording session on a tablet, OS automatically sets the tablet volume to 50% and when after recording ends, it doesn't change back to the previous volume level before starting the recording. So I would like to know whether this is an OS default behavior or a bug? If it's a default behavior, I much appreciate if I can get a link to the documentation.

I am working on an app which plays audio - https://youtu.be/VbAfUk_eYl0?si=nJg5ayy2faWE78-g - and one of the features is, on restart, if you had paused playback of a file at the time the app was previously shut down (or were playing one at the time of shutdown), the paused state and position in the file is restored exactly as it was, on restart. The functionality works. However, it seems impossible to get the "now playing" information in iOS into the right state to reflect that via the MediaPlayer API. On restart, handlers are attached to the play/pause/togglePlayPause actions on MPRemoteCommandCenter.shared(), and the map of media info is updated on MPNowPlayingInfoCenter.default().nowPlayingInfo. What happens is that iOS's media view shows the audio as playing and offers a pause button - even though the play action is enabled and the pause action is disabled. Once playback has been initiated (my workaround is to have the pause action toggle the play state, since otherwise you wouldn't be able to initiate playback from controls in a car without initiating it once from a device first). I've created a simplified white-noise-player demo to illustrate the problem - simply build and deploy it, and then start the app, lock your device and look at the playback controls on the lock screen. It will show a pause button - same behavior I've described. https://github.com/timboudreau/ios-play-pause-demo I've tried a few things to narrow down the source of the issue - for example, thinking that not MPNowPlayingInfoPropertyPlaybackProgress and MPMediaItemPropertyPlaybackDuration might be the culprit (since the system interpolates elapsed time and it's recommended to update those properties infrequently) on startup might do the trick, but the result is the same, just without a duration or progress shown. What governs this behavior, and is there some way to explicitly tell the media player API your current state is paused?

I have a SwiftUI app - (https://youtu.be/VbAfUk_eYl0?si=JxUBh0Bpb-vc1E1U) - which I thought was almost ready for release - a manager for airdropped audio files from Logic Pro or other music creation applications. It uses AVAudioEngine and AVAudioPlayerNode to play audio, and the MediaPlayer API to integrate with car audio and similar, all of which works well. It does not currently have an explicit CarPlay integration (and I'm slightly horrified at the amount of work that is going to require). I had the good or bad luck of getting a loaner car with carplay while mine is being repaired yesterday, and lo and behold, when connected to the vehicle via CarPlay, there is no audio output in the vehicle at all. The now playing panel correctly shows the information my app provides about the currently playing song; the player node believes it is playing, the AVAudioSession is configured as it should be. But there is no sound. Obviously I cannot ship it in this state. I've tried fiddling with the parameters the AVAudioSession is configured with, in case there was some parameter that was preventing audio output, to no avail - currently: var options = AVAudioSession.CategoryOptions() options.insert(.allowAirPlay) options.insert(.allowBluetooth) options.insert(.allowBluetoothA2DP) try session.setCategory(.playback, mode: .default, options: options) try? session.setPreferredIOBufferDuration(0.002) // ~96 samples at 44.1kHz try? session.setPrefersNoInterruptionsFromSystemAlerts(true) try? session.setPrefersInterruptionOnRouteDisconnect(false) try session.setActive(true, options: [.notifyOthersOnDeactivation]) All diagnostics within the app show the player operating correctly - files are played and flushed; AVAudioPlayerNodeCompletionCallbacks are called when they should be. But the output is not audible in the vehicle. I would much prefer to ship this app without full-blown CarPlay integration, but with working audio when connected via CarPlay, and work on full CarPlay integration for the next release. Is there some secret handshake I am just missing to make this work?

As I've mentioned before our app uses PTT Framework to record and send audio messages. In one of supported by app mode we are using WebRTC.org library for that purpose. Internally WebRTC.org library uses Voice-Processing I/O Unit (kAudioUnitSubType_VoiceProcessingIO subtype) to retrieve audio from mic. According to https://developer.apple.com/documentation/avfaudio/avaudiosession/mode-swift.struct/voicechat using Voice-Processing I/O Unit leads to implicit enabling .voiceChat AVAudioSession mode (i.e. it looks like it's not possible to use Voice-Processing I/O Unit without .voiceChat mode). And problem is following: when user starts outgoing PTT, PTT Framework plays audio notification, but in case of enabled .voiceChat mode that sound is playing distorted or not playing at all. Questions: Is it known issue? Is there any way to workaround it?

Hi, I am looking for a good way to play sounds at a high frequency. At the moment I am using the AVAudioEngine, and create a couple AVAudioPlayerNode and for each sound I need to play I create a AVAudioPCMBuffer. When the app needs to play a sound, I get the correct AVAudioPCMBuffer for the sound and use the first available AVAudioPlayerNode and feed it to the buffer. The timing for a metronome app has to be very precise because if it's of by about 16ms the user can hear that it is not playing had the right interval. For low speeds this is working without any problems, but at high speeds it is getting worse. Maybe anyone has an idea on how I can improve my method. Its a Plugin for Flutter. import AVFoundation class FastSoundPlayer { private var audioPlayers: [SoundPlayer?] = [] private var sounds: [String: Sound] = [:] private var engine = AVAudioEngine() let session = AVAudioSession.sharedInstance() init() { do { try session.setCategory(AVAudioSession.Category.playback, mode: AVAudioSession.Mode.default, options: [AVAudioSession.CategoryOptions.mixWithOthers]) try session.setActive(true) createSoundPlayers(count: 20) try engine.start() } catch { print("Error starting audio engine: \(error.localizedDescription)") } } // Selector method to handle applicationDidBecomeActiveNotification func applicationDidBecomeActive() { // Reinitialize AVAudioEngine and reattach all nodes do { engine.reset() objc_sync_enter(audioPlayers) audioPlayers.removeAll() createSoundPlayers(count: 20) objc_sync_exit(audioPlayers) try engine.start() } catch { print("Error starting audio engine: \(error.localizedDescription)") } } func createSoundPlayers(count: Int) { for _ in 0..<count { let player = SoundPlayer() engine.attach(player.player) engine.connect(player.player, to: engine.mainMixerNode, format: nil) audioPlayers.append(player) } } func load(sound: Data, name: String) { let sound = Sound(soundData: sound) sounds[name] = sound } func play(name: String) { if !engine.isRunning { applicationDidBecomeActive() } guard let sound = sounds[name] else { print("Sound not found") return } if let player = getAvailablePlayer() { player.play(sound: sound) } } func getAvailablePlayer() -> SoundPlayer? { for player in audioPlayers { if !player!.isPlaying { return player } } return nil } } class SoundPlayer { let player = AVAudioPlayerNode() var isPlaying = false init() { player.volume = 1.0 } func play(sound: Sound) { player.scheduleBuffer(sound.sound!, at: nil, options: .interrupts, completionCallbackType: .dataPlayedBack) { _ in self.complete() } if (player.engine != nil && player.engine!.isRunning) { player.play() isPlaying = true } } func complete() { isPlaying = false } } class Sound { var sound: AVAudioPCMBuffer? init(soundData: Data) { do { let temporaryURL = FileManager.default.temporaryDirectory.appendingPathComponent("tempSound.wav") try soundData.write(to: temporaryURL) // Create AVAudioFile from the temporary file URL let audioFile = try AVAudioFile(forReading: temporaryURL) // Define the format for the PCM buffer (44100Hz, stereo) let format = AVAudioFormat(commonFormat: .pcmFormatInt16, sampleRate: 44100, channels: 2, interleaved: false) // Create AVAudioPCMBuffer guard let pcmBuffer = AVAudioPCMBuffer(pcmFormat: format!, frameCapacity: AVAudioFrameCount(audioFile.length)) else { // Failed to create PCM buffer self.sound = nil return } // Read audio file into PCM buffer try audioFile.read(into: pcmBuffer) // Assign the created AVAudioPCMBuffer to the sound property self.sound = pcmBuffer } catch { print("Error loading sound file: \(error.localizedDescription)") self.sound = nil } } } Thanks!

Dear Apple Team, I am facing an issue with UDP networking in my watchOS app for duplex audio streaming using NWConnection. I have already added the necessary capabilities, including background mode for audio, to ensure smooth operation. Issue Details: The UDP connection works fine on the simulator since it uses macOS networking and allows low-level access. However, on a real Apple Watch (running watchOS 10), the connection remains in a "waiting" state and fails with Error 50. I am aware of Technical Note TN3135 regarding low-level networking on watchOS, but even after following these guidelines, the issue persists. Questions: Does watchOS impose additional restrictions on UDP networking compared to iOS/macOS? Are there any specific entitlements or configurations required to allow UDP connections on a real Apple Watch? Is there a workaround or debugging method to get more insights into why the connection fails? I would appreciate any guidance or recommendations on resolving this issue.

Let's consider the following code. I've created an actor that loads a list of .mp3 files from a Bundle and then makes it available for audio reproduction. Unfortunately, I'm experiencing a memory leak. At the play method. player.play() From Instruments I get _malloc_type_malloc_outlined libsystem_malloc.dylib start_wqthread libsystem_pthread.dylib private actor AudioActor { enum Failure: Error { case soundsNotLoaded([AudioPlayerClient.Sound: Error]) } enum Player { case music(AVAudioPlayer) } var players: [Sound: Player] = [:] let bundles: [Bundle] init(bundles: UncheckedSendable<[Bundle]>) { self.bundles = bundles.wrappedValue } func load(sounds: [Sound]) throws { try AVAudioSession.sharedInstance().setActive(true, options: []) var errors: [Sound: Error] = [:] for sound in sounds { guard let url = bundle.url(forResource: sound.name, withExtension: "mp3") else { continue } do { self.players[sound] = try .music(AVAudioPlayer(contentsOf: url)) } catch { errors[sound] = error } } guard errors.isEmpty else { throw Failure.soundsNotLoaded(errors) } } func play(sound: Sound, loops: Int?) throws { guard let player = self.players[sound] else { return } switch player { case let .music(player): player.numberOfLoops = loops ?? -1 player.play() } } func stop(sound: Sound) throws { guard let player = self.players[sound] else { throw Failure.soundsNotLoaded([:]) } switch player { case let .music(player): player.stop() } } }

I have an app under development - demo here - https://youtu.be/VbAfUk_eYl0?si=s6EDBx-4G6P_QbZO - which is sort of an audio player for airdropped files - something useful to musicians who dump work in progress to their phone, make notes, revise and update. I've been testing my handling of audio session interruption notifications, but seems to be a lot of inconsistency in how, when and why iOS delivers them, and I'm wondering if there is some rhyme or reason to it that I'm just not detecting. For example, I am playing a song in my app. Switch to Apple Music and start playing a song there. My app gets an interruption began notification - this is consistent. Switch back to my app, and about half the time, I will get an interruption ended notification (coupled often with a blast of the tail of whatever audio buffer was partially played when the interruption started, even though the engine was stopped - and followed by call to my AVAudioPlayerNodeCompletionCallback - is there some way to avoid this?). Half the time I don't get an interruption ended notification; my app can (as expected) end the interruption by activating the AVAudioSession and playing something. I have not been able to determine any pattern to this behavior, other than that if my app started playing using AVAudioPlayerNode.scheduleSegment rather than scheduleFile I think the notification will be consistently delivered on app activation rather than when I activate the session programmatically. I would like my app to behave deterministically, and would appreciate any help in deciphering what causes the inconsistent behavior in notifications from iOS.

We have the necessary background recording entitlements, and for many users... do not run into any issues. However, there is a subset of users that routinely get recordings ending.. we have narrowed this down and believe it to be the work of the watch dog. First we removed the entire view hierarchy when app is backgrounded. There is just 'Text("Recording")' This got the CPU usage in profiler down to 0%. We saw massive improvements to recording success rate. We walked away assuming that was enough. However we are still seeing the same sort of crashes. All in the background. We're using Observation to drive audio state changes to a Live Activity. Are those Observations causing the problem? Why doesn't apple provide a better API to background audio? The internet is full of weird issues https://stackoverflow.com/questions/76010213/why-is-my-react-native-app-sometimes-terminated-in-the-background-while-tracking https://stackoverflow.com/questions/71656047/why-is-my-react-native-app-terminating-in-the-background-while-recording-ios-r https://github.com/expo/expo/issues/16807 This is such a terrible user experience. And we have very little visibility into what is happening and why. No where in apple documentation states that in order for background recording to work, the app can only be 'Text("Recording")' It does not outline a CPU or memory threshold. It just kills us.

We have application using PTT Framework to record audio messages when app is backgrounded. Right now we are using AVAudioRecorder for that purpose. And problem is one specific user has frequent issue - recorded audio contains only silence. I've checked almost everything I can imagine but didn't find any possible reason of issue. Conditions: AVAudioRecorder uses following configuration: [ AVEncoderAudioQualityKey: AVAudioQuality.low.rawValue, AVFormatIDKey : kAudioFormatMPEG4AAC, AVNumberOfChannelsKey: 1, AVSampleRateKey: 16000.0 ] App waits both didBeginTransmitting and didActivate audioSession from PTChannelManager (audio session has playback category at that moment) App does AVAudioSession category change to playAndRecord App gets routeChangeNotification with categoryChange and category = playAndRecord There is no any interruption notifications from AVAudioSession during recording There is no any error notification from AVAudioRecorder Any idea what exactly I do wrong? Is there anything else I should check? Thanks in advance. P.S. it looks like recording audio with AudioUnit has the same issue, but let's exclude it from question atm for simplicity.

Hello! I'm use AVFoundation for preview video and audio from selected device, and I try use AVAudioEngine for preview audio in real-time, but I can't or I don't understand how select input device? I can hear only my microphone in real-time So far, I'm using AVCaptureAudioPreviewOutput for in real-time hear audio, but I think has delay. On iOS works easy with AVAudioEngine, but on macOS bruh...

Issue: Under certain conditions, using CallKit does not automatically enable the microphone. Steps to Reproduce: 1.Start an outgoing call, then the user manually mutes the audio. 2.Receive a native incoming call, end the current call, then answer the new incoming call.(This order is important.) 3.End the incoming call. 4.Start another outgoing call and observe the microphone; do not manually mute or unmute. Actual Behavior: The audio icon indicates that the audio is unmuted, but the microphone remains off, and the small yellow dot in the top status bar (which represents the microphone) does not appear. Expected Behavior: The microphone should be on, consistent with the audio icon display, and the small yellow dot should appear in the top status bar. Device: iPhone 16 pro & iPhone 15 pro, iOS 18.0+ Can it be reproduced using speakerbox(CallKit Demo)? YES

I am developing an iOS app that needs to play spoken audio on demand from a server, while ducking the audio of background music from another app (e.g., SoundtrackYourBrand or Apple Music). This must work even when the app is in the background, and the server dictates when and what audio is played. Ideally, the message should be played within a minute of the server requesting it. Current Attempt & Observations I initially tried using Firebase Cloud Messaging (FCM) silent notifications to send a URL to an audio file, which the app would then play using AVPlayer. This works consistently when the app is active, but in the background, it only works about 60% of the time. In cases where it fails, iOS ducks the background music (e.g., from SoundtrackYourBrand) but never plays the spoken audio. Interestingly, when I play the audio without enabling audio ducking, it seems to work 100% of the time from my limited testing, even in the background. The app has background modes enabled for Audio, Background Fetch, and Remote Notifications. Best Approach to Achieve This? I’d like guidance on the best Apple-compliant approach to reliably play audio on command from the server, even when the app is in the background. Some possible paths: Ensuring the app remains active in the background – Are there recommended ways to prevent the app from getting suspended, such as background tasks, a special background mode, or a persistent connection to the server? Alternative triggering mechanisms – Would something like VoIP, Push-to-Talk, or another background service be better suited for this use case? Built-in iOS speech synthesis (AVSpeechSynthesizer) – If playing external audio is unreliable, would generating speech dynamically from text be a more robust approach? Streaming audio instead of sending a URL – Could continuous streaming from the server keep the app active and allow playback at the right moment? I want to ensure the solution is reliable and works 100% of the time when needed. Any recommendations on the best approach for this would be greatly appreciated. Thank you for your time and guidance.

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?

Getting this error in iPhone Portrait Mode with notch. Currrently using AVQueuePlayer to play more than 30 mp3 files one by one. All constraint properties are correct but error occures only in Apple iPhone Portrait Mode with notch series. But same code works on same iPhone in Landscape mode. **But I get this error: ** LoudnessManager.mm:709 unable to open stream for LoudnessManager plist Type: Error | Timestamp: 2025-02-07 | Process: | Library: AudioToolbox | Subsystem: com.apple.coreaudio | Category: aqme | TID: 0x42754 LoudnessManager.mm:709 unable to open stream for LoudnessManager plist LoudnessManager.mm:709 unable to open stream for LoudnessManager plist Timestamp: 2025-02-07 | Library: AudioToolbox | Subsystem: com.apple.coreaudio | Category: aqme