@voicenter-team/opensips-js 1.0.20 → 1.0.22

package/build/index.d.ts

@@ -1,60 +1,45 @@
-import JsSIP from 'jssip';
 import UA from './helpers/UA';
-import { TempTimeData, ITimeData } from './helpers/time.helper';
-import { WebrtcMetricsConfigType, MediaDeviceType } from '@/types/webrtcmetrics';
+import { ITimeData } from './helpers/time.helper';
+import { WebrtcMetricsConfigType } from '@/types/webrtcmetrics';
 import { ListenersKeyType, ListenerCallbackFnType } from '@/types/listeners';
-import { RTCSessionExtended, ICall, IntervalType, ListenerEventType, IRoom, IDoCallParam, ICallStatus, IRoomUpdate, IOpenSIPSJSOptions } from '@/types/rtc';
-import { SendMessageOptions } from 'jssip/lib/Message';
-export interface InnerState {
-    isMuted: boolean;
-    muteWhenJoin: boolean;
-    isDND: boolean;
-    activeCalls: {
-        [key: string]: ICall;
-    };
-    extendedCalls: {
-        [key: string]: ICall;
-    };
-    activeRooms: {
-        [key: number]: IRoom;
-    };
-    callTime: {
-        [key: string]: TempTimeData;
-    };
-    callStatus: {
-        [key: string]: ICallStatus;
-    };
-    timeIntervals: {
-        [key: string]: IntervalType;
-    };
-    callMetrics: {
-        [key: string]: any;
-    };
-    availableMediaDevices: Array<MediaDeviceInfo>;
-    selectedMediaDevices: {
-        [key in MediaDeviceType]: string;
-    };
-    microphoneInputLevel: number;
-    speakerVolume: number;
-    originalStream: MediaStream | null;
-    listeners: {
-        [key: string]: Array<(call: RTCSessionExtended, event: ListenerEventType | undefined) => void>;
-    };
-    metricConfig: WebrtcMetricsConfigType;
-    isAutoAnswer: boolean;
-}
+import { RTCSessionExtended, ICall, IRoom, IDoCallParam, IRoomUpdate, IOpenSIPSJSOptions, CustomLoggerType } from '@/types/rtc';
+import { IMessage } from '@/types/msrp';
 declare class OpenSIPSJS extends UA {
     private initialized;
     private readonly options;
+    private logger;
     private readonly newRTCSessionEventName;
     private readonly registeredEventName;
     private readonly unregisteredEventName;
-    private readonly activeCalls;
-    private readonly extendedCalls;
-    private _currentActiveRoomId;
-    private _callAddingInProgress;
-    private state;
-    constructor(options: IOpenSIPSJSOptions);
+    private readonly disconnectedEventName;
+    private readonly connectedEventName;
+    private readonly newMSRPSessionEventName;
+    private muted;
+    private isAutoAnswer;
+    private isDNDEnabled;
+    private muteWhenJoinEnabled;
+    private activeRooms;
+    private activeCalls;
+    private extendedCalls;
+    private activeMessages;
+    private extendedMessages;
+    private msrpHistory;
+    private microphoneInputLevelValue;
+    private speakerVolumeValue;
+    private availableMediaDevices;
+    private selectedMediaDevices;
+    private callStatus;
+    private callTime;
+    private callMetrics;
+    private timeIntervals;
+    private metricConfig;
+    private originalStreamValue;
+    private currentActiveRoomIdValue;
+    private isCallAddingInProgress;
+    private isMSRPInitializingValue;
+    private isReconnecting;
+    private listenersList;
+    constructor(options: IOpenSIPSJSOptions, logger?: CustomLoggerType);
     on<T extends ListenersKeyType>(type: T, listener: ListenerCallbackFnType<T>): this;
     off<T extends ListenersKeyType>(type: T, listener: ListenerCallbackFnType<T>): this;
     emit(type: ListenersKeyType, args: any): boolean;
@@ -75,26 +60,24 @@ declare class OpenSIPSJS extends UA {
     get currentActiveRoomId(): number | undefined;
     private set currentActiveRoomId(value);
     get autoAnswer(): boolean;
-    set autoAnswer(value: boolean);
     get callAddingInProgress(): string | undefined;
     private set callAddingInProgress(value);
+    get isMSRPInitializing(): boolean | undefined;
     get muteWhenJoin(): boolean;
-    set muteWhenJoin(value: boolean);
     get isDND(): boolean;
-    set isDND(value: boolean);
     get speakerVolume(): number;
-    set speakerVolume(value: number);
     get microphoneInputLevel(): number;
-    set microphoneInputLevel(value: number);
     get getActiveCalls(): {
         [key: string]: ICall;
     };
     get hasActiveCalls(): boolean;
+    get getActiveMessages(): {
+        [key: string]: IMessage;
+    };
     get getActiveRooms(): {
         [key: number]: IRoom;
     };
     get isMuted(): boolean;
-    set isMuted(value: boolean);
     get getInputDeviceList(): MediaDeviceInfo[];
     get getOutputDeviceList(): MediaDeviceInfo[];
     get getUserMediaConstraints(): {
@@ -108,9 +91,7 @@ declare class OpenSIPSJS extends UA {
     get getInputDefaultDevice(): MediaDeviceInfo | undefined;
     get getOutputDefaultDevice(): MediaDeviceInfo | undefined;
     get selectedInputDevice(): string;
-    set selectedInputDevice(deviceId: string);
     get selectedOutputDevice(): string;
-    set selectedOutputDevice(deviceId: string);
     get originalStream(): MediaStream | null;
     private setAvailableMediaDevices;
     updateDeviceList(): Promise<void>;
@@ -122,8 +103,8 @@ declare class OpenSIPSJS extends UA {
     private _stopCallTimer;
     setMetricsConfig(config: WebrtcMetricsConfigType): void;
     sendDTMF(callId: string, value: string): void;
+    private setIsMuted;
     doMute(value: boolean): void;
-    sendMessage(target: string | JsSIP.URI, body: string, options?: SendMessageOptions): import("jssip/lib/Message").Message;
     doCallHold({ callId, toHold, automatic }: {
         callId: string;
         toHold: boolean;
@@ -131,12 +112,16 @@ declare class OpenSIPSJS extends UA {
     }): void;
     private _cancelAllOutgoingUnanswered;
     callAnswer(callId: string): void;
+    msrpAnswer(callId: string): void;
     callMove(callId: string, roomId: number): Promise<void>;
     updateCall(value: ICall): void;
+    updateMSRPSession(value: IMessage): void;
     updateRoom(value: IRoomUpdate): void;
     private hasAutoAnswerHeaders;
     private _addCall;
     private _addCallStatus;
+    private _addMMSRPSession;
+    private _addMSRPMessage;
     private _updateCallStatus;
     private _removeCallStatus;
     private _addRoom;
@@ -152,6 +137,7 @@ declare class OpenSIPSJS extends UA {
     _muteReconfigure(call: ICall): void;
     muteCaller(callId: string, value: boolean): void;
     callTerminate(callId: string): void;
+    messageTerminate(callId: string): void;
     callTransfer(callId: string, target: string): void;
     callMerge(roomId: number): void;
     setDND(value: boolean): void;
@@ -161,18 +147,31 @@ declare class OpenSIPSJS extends UA {
     subscribe(type: string, listener: (c: RTCSessionExtended) => void): void;
     removeIListener(value: string): void;
     private addCall;
+    private addMessageSession;
     private _triggerListener;
+    private _triggerMSRPListener;
     private _removeCall;
+    private _removeMMSRPSession;
     private _activeCallListRemove;
+    private _activeMessageListRemove;
     private newRTCSessionCallback;
+    private newMSRPSessionCallback;
     private setInitialized;
-    start(): this;
+    begin(): this | undefined;
     setMuteWhenJoin(value: boolean): void;
+    setMicrophoneInputLevel(value: number): void;
+    setSpeakerVolume(value: number): void;
+    setAutoAnswer(value: boolean): void;
+    private setSelectedInputDevice;
+    private setSelectedOutputDevice;
+    private setIsMSRPInitializing;
     private _setCallMetrics;
     private _removeCallMetrics;
     private _getCallQuality;
     private _triggerAddStream;
     doCall({ target, addToCurrentRoom }: IDoCallParam): void;
+    initMSRP(target: string, body: string, options: any): void;
+    sendMSRP(msrpSessionId: string, body: string): void;
     callChangeRoom({ callId, roomId }: {
         callId: string;
         roomId: number;