iOS Audio not working during call answered when phone is locked. WebRTC used for calling
Asked Answered
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3

5

I am facing a problem with Audio When using Callkit with WebRTC for VOIP call, While answering the call from Lock Screen.

General Functionality :

My app activates the audioSession when it's launched. For an incoming call, SDP Offer & Answer are generated and exchanged. Peer Connection is set up. Both audio and video streams are generated, whether it's audio call or video call. Then Call is reported to callkit by using the following code:

    callProvider.reportNewIncomingCall(with: currentCallUUID!, update: update) { error in }

If app is in the foreground, it works fine.

But, when the phone is locked, and user answers the call from lock screen, the Streams are exchanged but no audio comes on either end until user enters into the app himself.

As the user enters into the App, audio becomes active on both the ends.

All the background settings and capabilities are set properly.

I have also referred to the following work around provided by Apple staff. But even it does not work.

https://forums.developer.apple.com/thread/64544

As I mentioned, I am using WebRTC for calling. If I exchange the media streams after the user answers the call( still on Lock Screen) and peer connection is set at that time. It works fine (But it adds the delay in making the call connection).

But if Peer Connection is made before displaying call (say before reporting call to callkit), the audio stops working.

Statfarad answered 28/7, 2017 at 6:25 Comment(0)
S
5

I am able to resolve this issue.

Steps that I followed -

  • I checked the code related to WebRTC here

  • I added RTCAudioSession header file which is actually a private class of Webrtc. So every time I receive a call event from signaling, I enable RTCAudiosession and on end of the call, I disable it.

  • I have to render the incoming streams to a dummy view (Although it is not displayed when the call is going and the app is not yet open, but it is required to make audio working).

I hope this will help if someone is facing the same issue.

Statfarad answered 17/8, 2017 at 5:3 Comment(0)
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2

@abhimanyu are you still facing the issue or you made it work. I am facing same issue with CallKit.

As per my understanding in WebRTC M60 release they have fixed on issue related to CallKit, which I think created a side effect and caused this issue.

The issue which they have fixed is related to System AudioSession, when ever CallKit presents incoming call UI and play ringer tone CallKit takes control of AudioSession and after user action (accept/ decline) it releases control. In WebRTC M60 release, now they have added observers for this control exchange. That's why it is working if app is in foreground, but if phone is locked and any incoming call is accepted then (I am assuming you are using CallKit UI for call and not redirecting user to App on accept from lock screen) due to Native UI of call it is not possible for WebRTC to activate its own AudioSession instance as call is going through CallKit Screen.

Link for bug which has been fixed on WebRTC M60: https://bugs.chromium.org/p/webrtc/issues/detail?id=7446

If you found any workaround for this issue please let me know.

Parapodium answered 10/8, 2017 at 7:8 Comment(0)
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0

Please Note that I share my code and its about to my needs and I share for reference. you need to change it according to your need.

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

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

didReceive method;

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,

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

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;

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 ;

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. you can ignore them accordingly!!

FOR REFERENCE You can also check the example at this link

Anthelion answered 9/6, 2021 at 11:15 Comment(0)

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