Flutter WebRTC：如何在应用后台执行过程中维护 WebRTC 调用？

问：

我正在开发一个 Flutter 应用程序，该应用程序集成了用于音频/视频通话的 WebRTC。当应用程序在前台时，WebRTC 调用运行良好，但是当应用程序进入后台时，我面临着维护呼叫的挑战，因为操作系统终止了应用程序内的操作并终止了 voip 实例

即使应用程序进入后台或终止状态，我如何使其工作？

示例代码：

 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 {}
}