Callkit和Webrtc没有音频时接受呼叫从锁定屏幕

Question

我试图让callkit在传入呼叫时与webrtc一起工作，但当我从锁定屏幕接收呼叫并接受它时，在我以前台模式运行应用程序之前，不会有声音。我已配置audiosession向RTCAudoSession发送通知，但它不起作用。你有解决办法吗？

      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
    }

Answer 1

请注意，我分享我的代码，它将满足我的需要，我分享以供参考。您需要根据需要进行更改。

当您收到voip通知时，创建webrtc处理类的新事件，并将这两行代码添加到代码块，因为从voip通知启用音频会话失败

RTCAudioSession.sharedInstance().useManualAudio = true
RTCAudioSession.sharedInstance().isAudioEnabled = false 

双接收法；

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
      
        }
    

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
    
 
   }

在Webrtc处理程序类中，配置媒体发送器和audiosession

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)
        }
      
    }

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
    }

音频会议；

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()
    }

请考虑，因为我工作的回调和委托代码包括委托和回调块。

供参考您也可以在此链接查看示例

Answer 2

您测试的iPhone的iOS版本是什么？

Answer 3

好的，我找到了问题的原因。在IOS 12中，webrtc存在一个问题，当您从锁定屏幕启动webrtc并尝试访问摄像头时，输出音量会中断，因此解决方案是检查屏幕是否处于活动状态，如果不是，则不要请求并将本地RTCDeoTrack添加到RTCStream中。