Flutter WebRTC: How to Maintain WebRTC Call During App Background Execution?

Question

I'm working on a Flutter app that incorporates WebRTC for audio/video calls. The WebRTC calls work well when the app is in the foreground, but I'm facing challenges in maintaining the call when the app goes into the background as the OS terminates operation inside the app and kills the voip instance

How I can make it work even when the app goes background or terminated state?

sample code:

 class VoipApp with WidgetsBindingObserver implements WebRTCDelegate {
        bool background = false;
        final BuildContext context;
        late final VoIP voIP;
        bool _canHandleNewCall = true;
    
        VoipApp(this.context) {
            voIP = VoIP(locator<Client>(), this);
        }
        @override
      void didChangeAppLifecycleState(AppLifecycleState? state) {
        background = (state == AppLifecycleState.detached ||
            state == AppLifecycleState.paused);
      }
    
      @override
      bool get canHandleNewCall => _canHandleNewCall;
    
      @override
      Future<webrtc_impl.RTCPeerConnection> createPeerConnection(
          Map<String, dynamic> configuration,
          [Map<String, dynamic> constraints = const {}]) {
        return webrtc_impl.createPeerConnection(configuration, constraints);
      }
    
      @override
      webrtc_impl.VideoRenderer createRenderer() {
        return webrtc_impl.RTCVideoRenderer();
      }
    
      @override
      Future<void> handleCallEnded(CallSession session) async {
        _canHandleNewCall = true;
      }
    
      @override
      Future<void> handleGroupCallEnded(GroupCall groupCall) async {
        throw UnimplementedError();
      }
    
      @override
      Future<void> handleMissedCall(CallSession session) async {
        Logs().i("MISSED CALL");
      }
    
      @override
      Future<void> handleNewCall(CallSession session) async {
        _canHandleNewCall = true;
        switch (session.direction) {
          case CallDirection.kIncoming:
            Logs().i("[VOIP] INCOMING CALL");
            Navigator.of(context).push(
              MaterialPageRoute(
                builder: (context) => CallScreen(
                  callSession: session,
                ),
              ),
            );
            break;
          case CallDirection.kOutgoing:
            Logs().i("[VOIP] OUTGOING CALL");
            Navigator.of(context).push(
              MaterialPageRoute(
                builder: (context) => CallScreen(
                  callSession: session,
                ),
              ),
            );
            break;
        }
      }
    
      @override
      Future<void> handleNewGroupCall(GroupCall groupCall) {
        throw UnimplementedError();
      }
    
      @override
      bool get isWeb => kIsWeb;
    
      @override
      webrtc_interface.MediaDevices get mediaDevices =>
          webrtc_impl.navigator.mediaDevices;
    
      @override
      Future<void> playRingtone() async {
      }
    @override
    Future<void> stopRingtone() async {}
}