[英]Callkit and Webrtc no audio when accept call from locked screen
I'm trying to make callkit work with webrtc on incoming call, but when I receive call and accept it from locked screen there are no sound untill I run the app in the foreground mode.我试图让 callkit 在来电时与 webrtc 一起工作,但是当我接到电话并从锁定屏幕接受它时,直到我在前台模式下运行应用程序时才发出声音。 I've configured audiosession send notification to RTCAudioSession, but it doesn't work.我已经配置了 audiosession 向 RTCAudioSession 发送通知,但它不起作用。 Do you have some workarounds?你有一些解决方法吗?
func configureAudioSession() {
let sharedSession = AVAudioSession.sharedInstance()
do {
try sharedSession.setCategory(AVAudioSessionCategoryPlayAndRecord, mode: AVAudioSessionModeVideoChat, options: .mixWithOthers)
try sharedSession.setMode(AVAudioSessionModeVideoChat)
// try sharedSession.setAggregatedIOPreference(AVAudioSessionIOType.aggregated)
} catch {
debugPrint("Failed to configure `AVAudioSession`")
}
}
func handleIncomingCall(spaceName:String) {
if callUUID != nil {
oldCallUUID = callUUID
}
callUUID = UUID()
print("CallManager handle uuid = \(callUUID?.description)")
let update = CXCallUpdate()
update.hasVideo = true
update.remoteHandle = CXHandle(type: .generic, value: spaceName)
self.configureAudioSession()
provider?.reportNewIncomingCall(with: callUUID!, update: update, completion: { error in
print("CallManager report new incoming call completion")
})
}
func provider(_ provider: CXProvider, didActivate audioSession: AVAudioSession) {
print("CallManager didActivate")
RTCAudioSession.sharedInstance().audioSessionDidActivate(audioSession)
RTCAudioSession.sharedInstance().isAudioEnabled = true
self.callDelegate?.callIsAnswered()
}
func provider(_ provider: CXProvider, didDeactivate audioSession: AVAudioSession) {
print("CallManager didDeactivate")
RTCAudioSession.sharedInstance().audioSessionDidDeactivate(audioSession)
RTCAudioSession.sharedInstance().isAudioEnabled = false
}
Okay, I've found what the cause of an issue.好的,我找到了问题的原因。 In IOS 12 there is a problem with webrtc, when you start webrtc from locked screen and trying to get access to camera - the output volume breaks, so the solution is to check if the screen is Active or not, and if not - do not request and add local RTCVideoTrack into your RTCStream.在 IOS 12 中,webrtc 存在问题,当您从锁定屏幕启动 webrtc 并尝试访问相机时 - 输出音量中断,因此解决方案是检查屏幕是否处于活动状态,如果不是 - 不要请求并将本地 RTCVideoTrack 添加到您的 RTCStream 中。
您的测试 iPhone 的 iOS 版本是什么?
Please Note that I share my code and its about to my needs and I share for reference.请注意,我分享了我的代码及其关于我的需求,并分享以供参考。 you need to change it according to your need.你需要根据你的需要改变它。
when you receive voip notification create new incident of your webrtc handling class, and add this two lines to code block because enabling audio session from voip notification fails当您收到 voip 通知时,创建 webrtc 处理类的新事件,并将这两行添加到代码块中,因为从 voip 通知启用音频会话失败
RTCAudioSession.sharedInstance().useManualAudio = true
RTCAudioSession.sharedInstance().isAudioEnabled = false
didReceive method; didReceive 方法;
func pushRegistry(_ registry: PKPushRegistry, didReceiveIncomingPushWith payload: PKPushPayload, for type: PKPushType, completion: @escaping () -> Void) {
let state = UIApplication.shared.applicationState
if(payload.dictionaryPayload["hangup"] == nil && state != .active
){
Globals.voipPayload = payload.dictionaryPayload as! [String:Any] // I pass parameters to Webrtc handler via Global singleton to create answer according to sdp sent by payload.
RTCAudioSession.sharedInstance().useManualAudio = true
RTCAudioSession.sharedInstance().isAudioEnabled = false
Globals.sipGateway = SipGateway() // my Webrtc and Janus gateway handler class
Globals.sipGateway?.configureCredentials(true) // I check janus gateway credentials stored in Shared preferences and initiate websocket connection and create peerconnection
to my janus gateway which is signaling server for my environment
initProvider() //Crating callkit provider
self.update.remoteHandle = CXHandle(type: .generic, value:String(describing: payload.dictionaryPayload["caller_id"]!))
Globals.callId = UUID()
let state = UIApplication.shared.applicationState
Globals.provider.reportNewIncomingCall(with:Globals.callId , update: self.update, completion: { error in
})
}
}
func initProvider(){
let config = CXProviderConfiguration(localizedName: "ulakBEL")
config.iconTemplateImageData = UIImage(named: "ulakbel")!.pngData()
config.ringtoneSound = "ringtone.caf"
// config.includesCallsInRecents = false;
config.supportsVideo = false
Globals.provider = CXProvider(configuration:config )
Globals.provider.setDelegate(self, queue: nil)
update = CXCallUpdate()
update.hasVideo = false
update.supportsDTMF = true
}
modify your didActivate and didDeActive delegate functions like below,修改您的 didActivate 和 didDeActive 委托函数,如下所示,
func provider(_ provider: CXProvider, didActivate audioSession: AVAudioSession) {
print("CallManager didActivate")
RTCAudioSession.sharedInstance().audioSessionDidActivate(audioSession)
RTCAudioSession.sharedInstance().isAudioEnabled = true
// self.callDelegate?.callIsAnswered()
}
func provider(_ provider: CXProvider, didDeactivate audioSession: AVAudioSession) {
print("CallManager didDeactivate")
RTCAudioSession.sharedInstance().audioSessionDidDeactivate(audioSession)
RTCAudioSession.sharedInstance().isAudioEnabled = false
}
in Webrtc handler class configure media senders and audiosession在 Webrtc 处理程序类中配置媒体发送器和音频会话
private func createPeerConnection(webRTCCallbacks:PluginHandleWebRTCCallbacksDelegate) {
let rtcConfig = RTCConfiguration.init()
rtcConfig.iceServers = server.iceServers
rtcConfig.bundlePolicy = RTCBundlePolicy.maxBundle
rtcConfig.rtcpMuxPolicy = RTCRtcpMuxPolicy.require
rtcConfig.continualGatheringPolicy = .gatherContinually
rtcConfig.sdpSemantics = .planB
let constraints = RTCMediaConstraints(mandatoryConstraints: nil,
optionalConstraints: ["DtlsSrtpKeyAgreement":kRTCMediaConstraintsValueTrue])
pc = sessionFactory.peerConnection(with: rtcConfig, constraints: constraints, delegate: nil)
self.createMediaSenders()
self.configureAudioSession()
if webRTCCallbacks.getJsep() != nil{
handleRemoteJsep(webrtcCallbacks: webRTCCallbacks)
}
}
mediaSenders;媒体发送者;
private func createMediaSenders() {
let streamId = "stream"
// Audio
let audioTrack = self.createAudioTrack()
self.pc.add(audioTrack, streamIds: [streamId])
// Video
/* let videoTrack = self.createVideoTrack()
self.localVideoTrack = videoTrack
self.peerConnection.add(videoTrack, streamIds: [streamId])
self.remoteVideoTrack = self.peerConnection.transceivers.first { $0.mediaType == .video }?.receiver.track as? RTCVideoTrack
// Data
if let dataChannel = createDataChannel() {
dataChannel.delegate = self
self.localDataChannel = dataChannel
}*/
}
private func createAudioTrack() -> RTCAudioTrack {
let audioConstrains = RTCMediaConstraints(mandatoryConstraints: nil, optionalConstraints: nil)
let audioSource = sessionFactory.audioSource(with: audioConstrains)
let audioTrack = sessionFactory.audioTrack(with: audioSource, trackId: "audio0")
return audioTrack
}
audioSession ;音频会话;
private func configureAudioSession() {
self.rtcAudioSession.lockForConfiguration()
do {
try self.rtcAudioSession.setCategory(AVAudioSession.Category.playAndRecord.rawValue)
try self.rtcAudioSession.setMode(AVAudioSession.Mode.voiceChat.rawValue)
} catch let error {
debugPrint("Error changeing AVAudioSession category: \(error)")
}
self.rtcAudioSession.unlockForConfiguration()
}
Please consider that because I worked with callbacks and delegates code includes delegates and callback chunks.请考虑这一点,因为我使用回调和委托代码包括委托和回调块。 you can ignore them accordingly!!你可以相应地忽略它们!!
FOR REFERENCE You can also check the example at this link供参考您还可以在此链接中查看示例
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