@voicenter-team/opensips-js 1.0.76 → 1.0.78

package/README.md

@@ -23,37 +23,25 @@ const openSIPSJS = new OpenSIPSJS({
         extraHeaders: [ 'X-Bar: bar' ],
         pcConfig: {},
     },
+    modules: [ 'audio', 'video', 'msrp' ]
 })
 ```
 
-Then you will be able to call next methods on openSIPSJS instance:
+Then you can work with appropriate modules:
+```javascript
+openSIPSJS.audio
+openSIPSJS.video
+openSIPSJS.msrp
+```
 
-### Methods
+# OpensipsJS
+### OpensipsJS instance methods
 - `begin(): OpensipsInstance` - start opensips
-- `initCall(target: String, addToCurrentRoom: Boolean): void` - call to the target. If addToCurrentRoom is true then the call will be added to the user's current room
-- `holdCall(callId: String, automatic?: Boolean): Promise<void>` - put call on hold
-- `unholdCall(callId: String): Promise<void>` - unhold a call
-- `terminateCall(callId: String): void` - terminate call
-- `moveCall(callId: String, roomId: Number): Promise<void>` - Same as callChangeRoom. Move call to the specific room
-- `transferCall(callId: String, target: String): void` - transfer call to target
-- `mergeCall(roomId: Number): void` - merge calls in specific room. Works only for rooms with 2 calls inside
-- `answerCall(callId: String): void` - answer a call
-- `mute(): void` - mute ourself
-- `unmute(): void` - unmute ourself
-- `muteCaller(callId: String): void` - mute caller
-- `unmuteCaller(callId: String): void` - unmute caller
-- `setMicrophone(deviceId: String): Promise<void>` - set passed device as input device for calls
-- `setSpeaker(deviceId: String): Promise<void>` - set passed device as output device for calls
-- `setActiveRoom(roomId: Number): Promise<void>` - switch to the room
-- `setMicrophoneSensitivity(value: Number): void` - set sensitivity of microphone. Value should be in range from 0 to 1
-- `setSpeakerVolume(value: Number): void` - set volume of callers. Value should be in range from 0 to 1
-- `setDND(value: Boolean): void` - set the agent "Do not disturb" status
+- `on(event: OpensipsEvent, callback): void` - remove event listener
 - `subscribe({type: String, listener: function}): void` - subscribe to an event. Available events: `new_call`, `ended`, `progress`, `failed`, `confirmed`
 - `removeIListener(type: String): void` - remove event listener
-- `on(event: OpensipsEvent, callback): void` - remove event listener
-- `setMetricsConfig(config: WebrtcMetricsConfigType): void` - set the metric config (used for audio quality indicator)
 
-### Opensips Events
+### OpensipsJS events
 
 | Event      | Callback interface    | Description |
 |----------------|---------|---------------|
@@ -75,20 +63,47 @@ Then you will be able to call next methods on openSIPSJS instance:
 
 WebrtcMetricsConfigType
 
-| Parameter      | Type                   |
-|----------------|------------------------|
-| `refreshEvery` | `number \| undefined`  |
-| `startAfter`   | `number \| undefined`  |
-| `startAfter`   | `number \| undefined`  |
-| `verbose`      | `boolean \| undefined` |
-| `pname`        | `string \| undefined`  |
-| `cid`          | `string \| undefined`  |
-| `uid`          | `string \| undefined`  |
-| `record`       | `boolean \| undefined` |
-| `ticket`       | `boolean \| undefined` |
-
-Also there are next public fields on openSIPSJS instance:
-### Fields
+| Parameter      | Type      | Default |
+|----------------|-----------|----------|
+| `refreshEvery` | `number` | `undefined`  |
+| `startAfter`   | `number` | `undefined`  |
+| `startAfter`   | `number` | `undefined`  |
+| `verbose`      | `boolean` | `undefined` |
+| `pname`        | `string` | `undefined`  |
+| `cid`          | `string` | `undefined`  |
+| `uid`          | `string` | `undefined`  |
+| `record`       | `boolean` | `undefined` |
+| `ticket`       | `boolean` | `undefined` |
+
+Also, there are next public fields on OpensipsJS instance:
+### OpensipsJS instance fields
+- `sipDomain: String` - returns sip domain
+
+# Audio
+
+### Audio methods
+- `initCall(target: String, addToCurrentRoom: Boolean): void` - call to the target. If addToCurrentRoom is true then the call will be added to the user's current room
+- `holdCall(callId: String, automatic?: Boolean): Promise<void>` - put call on hold
+- `unholdCall(callId: String): Promise<void>` - unhold a call
+- `terminateCall(callId: String): void` - terminate call
+- `moveCall(callId: String, roomId: Number): Promise<void>` - Same as callChangeRoom. Move call to the specific room
+- `transferCall(callId: String, target: String): void` - transfer call to target
+- `mergeCall(roomId: Number): void` - merge calls in specific room. Works only for rooms with 2 calls inside
+- `answerCall(callId: String): void` - answer a call
+- `mute(): void` - mute ourself
+- `unmute(): void` - unmute ourself
+- `muteCaller(callId: String): void` - mute caller
+- `unmuteCaller(callId: String): void` - unmute caller
+- `setMicrophone(deviceId: String): Promise<void>` - set passed device as input device for calls
+- `setSpeaker(deviceId: String): Promise<void>` - set passed device as output device for calls
+- `setActiveRoom(roomId: Number): Promise<void>` - switch to the room
+- `setMicrophoneSensitivity(value: Number): void` - set sensitivity of microphone. Value should be in range from 0 to 1
+- `setSpeakerVolume(value: Number): void` - set volume of callers. Value should be in range from 0 to 1
+- `setDND(value: Boolean): void` - set the agent "Do not disturb" status
+- `setMetricsConfig(config: WebrtcMetricsConfigType): void` - set the metric config (used for audio quality indicator)
+
+### Audio instance fields
+- `sipOptions: Object` - returns sip options
 - `getActiveRooms: { [key: number]: IRoom }` - returns an object of active rooms where key is room id and value is room data
 - `sipDomain: String` - returns sip domain
 - `sipOptions: Object` - returns sip options
@@ -99,3 +114,80 @@ Also there are next public fields on openSIPSJS instance:
 - `selectedOutputDevice: String` - returns current selected output device id
 - `isDND: Boolean` - returns if the agent is in "Do not disturb" status
 - `isMuted: Boolean` - returns if the agent is muted
+
+# MSRP
+
+### MSRP methods
+- `initMSRP(target: String, body: String): void` - initialize connection with target contact. Body is the initial message to this target.
+- `sendMSRP(sessionId: String, body: String): Promise<void>` - send message
+- `msrpAnswer(sessionId: String)` - accept MSRP session invitation
+- `messageTerminate(sessionId: String)` - terminate message session
+
+### MSRP instance fields
+- `getActiveMessages: { [key: string]: IMessage }` - returns an object of active message sessions where key is session id and value is message session data.
+
+
+# Video
+
+### Video methods
+- `joinRoom(roomId: String, displayName: String, mediaConstraints: Object): void` - join conference room
+- `hangup()` - exit room
+- `startVideo()` - turn on camera
+- `stopVideo()` - turn off camera
+- `startAudio()` - mute
+- `stopAudio()` - unmute
+- `startScreenShare()` - start screen sharing
+- `stopScreenShare()` - stop screen sharing
+- `enableScreenShareWhiteboard(enable: boolean, stream: MediaStream)` - enable screen share whiteboard. stream parameter is screen share stream
+- `enableBokehEffectMask(): Promise<MediaStream>` - enable bokeh mask effect
+- `enableBackgroundImgEffectMask(): Promise<MediaStream>` - enable background image mask effect
+- `disableMask(): Promise<MediaStream>` - turn off mask effect. Returns stream without masking
+- `restartMasking(): Promise<void>` - rerun mask effect
+- `setupMaskVisualizationConfig(config: VisualizationConfigType)` - setup mask config
+- `startNoiseFilter()` - start noise filter
+- `stopNoiseFilter()` - stop noise filter
+- `setBitrate(bitrate: number)` - set bitrate for video
+- `enableWhiteboard(mode: 'whiteboard' | 'imageWhiteboard', enable: boolean, base64Image?: string)` - enable whiteboard. if mode is 'imageWhiteboard' then third parameter base64Image is required
+- `setupDrawerOptions(options: KonvaDrawerOptions)` - setup option for drawer
+- `setupScreenShareDrawerOptions(options: KonvaScreenShareDrawerOptions)` - setup option for screen share drawer
+
+VisualizationConfigType
+
+| Parameter      | Type                   | Default |
+|----------------|------------------------|---------|
+| `foregroundThreshold` | `number` | `0.5`  |
+| `maskOpacity`   | `number`| `0.5`  |
+| `maskBlur`   | `number`| `0`  |
+| `pixelCellWidth`      | `number`| `10` |
+| `backgroundBlur`        | `number`| `15`  |
+| `edgeBlur`          | `number`| `3`  |
+
+KonvaDrawerOptions
+
+| Parameter      | Type      |
+|----------------|-----------|
+| `container` | `number` |
+| `width`   | `number` |
+| `height`   | `number` |
+
+KonvaScreenShareDrawerOptions
+
+| Parameter      | Type      |
+|----------------|-----------|
+| `strokeWidth` | `number` |
+| `strokeColor`   | `string` |
+
+### Video events
+
+| Event      | Callback interface    | Description |
+|----------------|---------|---------------|
+| `member:join` | `(data) => {}`  |   Emitted when new member is joined   |
+| `member:update`   | `(data) => {}`  |  Emitted when member data is changed  |
+| `member:hangup`   | `(data) => {}`  |  Emitted when member leaves the conference  |
+| `hangup`   | `() => {}`  |  Emitted when we leave the conference  |
+| `screenShare:start`   | `() => {}`  |  Emitted when we share a screen  |
+| `screenShare:stop`   | `() => {}`  |  Emitted when we stop a screen sharing  |
+| `reconnect`   | `() => {}`  |  Emitted when reconnecting  |
+| `mediaConstraintsChange`   | `() => {}`  |  Emitted when media constraints change |
+| `metrics:report`   | `() => {}`  |  Emitted on metric report |
+| `metrics:stop`   | `() => {}`  |  Emitted when metrics are stopped |

package/dist/index.d.ts

@@ -39,6 +39,145 @@ declare interface AnswerOptionsExtended_2 extends AnswerOptions {
     mediaConstraints?: MediaConstraints | ExactConstraints_2
 }
 
+declare class AudioModule {
+    private context;
+    private currentActiveRoomIdValue;
+    private isAutoAnswer;
+    private isCallAddingInProgress;
+    private muteWhenJoinEnabled;
+    private isDNDEnabled;
+    private muted;
+    private microphoneInputLevelValue;
+    private speakerVolumeValue;
+    private activeRooms;
+    private activeCalls;
+    private extendedCalls;
+    private availableMediaDevices;
+    private selectedMediaDevices;
+    private callStatus;
+    private callTime;
+    private callMetrics;
+    private timeIntervals;
+    private metricConfig;
+    private activeStreamValue;
+    private initialStreamValue;
+    private VUMeter;
+    constructor(context: OpenSIPSJS);
+    get sipOptions(): {
+        mediaConstraints: {
+            audio: {
+                deviceId: {
+                    exact: string;
+                };
+            };
+            video: boolean;
+        };
+        session_timers: boolean;
+        extraHeaders: [string];
+        pcConfig: RTCConfiguration_2;
+    };
+    get currentActiveRoomId(): number | undefined;
+    private set currentActiveRoomId(value);
+    get autoAnswer(): boolean;
+    get callAddingInProgress(): string | undefined;
+    private set callAddingInProgress(value);
+    get muteWhenJoin(): boolean;
+    get isDND(): boolean;
+    get speakerVolume(): number;
+    get microphoneInputLevel(): number;
+    get getActiveCalls(): {
+        [key: string]: ICall;
+    };
+    get hasActiveCalls(): boolean;
+    get getActiveRooms(): {
+        [key: number]: IRoom;
+    };
+    get isMuted(): boolean;
+    get getInputDeviceList(): MediaDeviceInfo[];
+    get getOutputDeviceList(): MediaDeviceInfo[];
+    get getUserMediaConstraints(): {
+        audio: {
+            deviceId: {
+                exact: string;
+            };
+        };
+        video: boolean;
+    };
+    get selectedInputDevice(): string;
+    get selectedOutputDevice(): string;
+    get activeStream(): MediaStream;
+    private setAvailableMediaDevices;
+    updateDeviceList(): Promise<void>;
+    private initializeMediaDevices;
+    setCallTime(value: ITimeData): void;
+    removeCallTime(callId: string): void;
+    private setTimeInterval;
+    private removeTimeInterval;
+    private stopCallTimer;
+    private emitVolumeChange;
+    setMetricsConfig(config: WebrtcMetricsConfigType): void;
+    sendDTMF(callId: string, value: string): void;
+    private setIsMuted;
+    private processMute;
+    mute(): void;
+    unmute(): void;
+    private processHold;
+    holdCall(callId: string, automatic?: boolean): Promise<void>;
+    unholdCall(callId: string): Promise<void>;
+    private cancelAllOutgoingUnanswered;
+    answerCall(callId: string): void;
+    moveCall(callId: string, roomId: number): Promise<void>;
+    updateCall(value: ICall): void;
+    updateRoom(value: IRoomUpdate): void;
+    private hasAutoAnswerHeaders;
+    private addCall;
+    private addCallStatus;
+    private updateCallStatus;
+    private removeCallStatus;
+    private addRoom;
+    private getActiveStream;
+    setMicrophone(dId: string): Promise<void>;
+    private setActiveStream;
+    setSpeaker(dId: string): Promise<void>;
+    private removeRoom;
+    private deleteRoomIfEmpty;
+    private checkInitialized;
+    private muteReconfigure;
+    private roomReconfigure;
+    private doConference;
+    private processCallerMute;
+    muteCaller(callId: string): void;
+    unmuteCaller(callId: string): void;
+    terminateCall(callId: string): void;
+    transferCall(callId: string, target: string): Error;
+    mergeCall(roomId: number): void;
+    setDND(value: boolean): void;
+    private startCallTimer;
+    setActiveRoom(roomId: number | undefined): Promise<void>;
+    private getNewRoomId;
+    private setupCall;
+    private removeCall;
+    private activeCallListRemove;
+    private newRTCSessionCallback;
+    setMuteWhenJoin(value: boolean): void;
+    setMicrophoneSensitivity(value: number): void;
+    setSpeakerVolume(value: number): void;
+    setAutoAnswer(value: boolean): void;
+    private setSelectedInputDevice;
+    private setSelectedOutputDevice;
+    private setCallMetrics;
+    private removeCallMetrics;
+    private getCallQuality;
+    private setupVUMeter;
+    private stopVUMeter;
+    setupStream(): Promise<void>;
+    private triggerAddStream;
+    initCall(target: string, addToCurrentRoom: boolean): void;
+    private processRoomChange;
+}
+
+declare type AudioModuleName = typeof MODULES.AUDIO
+
 declare type CallAddingProgressListener = (callId: string | undefined) => void
 
 declare interface CallOptionsExtended extends AnswerOptionsExtended {
@@ -147,7 +286,8 @@ declare interface IOpenSIPSJSOptions {
         session_timers: boolean
         extraHeaders: [ string ]
         pcConfig: RTCConfiguration_2
-    }
+    },
+    modules: Array<Modules>
 }
 
 declare interface IRoom {
@@ -168,14 +308,62 @@ declare interface ITimeData {
     formatted: string
 }
 
+declare interface JanusOptions extends AnswerOptions {
+    eventHandlers?: Partial<JanusSessionEventMap>
+    anonymous?: boolean;
+    fromUserName?: string;
+    fromDisplayName?: string;
+}
+
+declare interface JanusSessionEventMap {
+    'peerconnection': PeerConnectionListener;
+    'connecting': ConnectingListener;
+    'sending': SendingListener;
+    'progress': CallListener;
+    'accepted': CallListener;
+    'confirmed': ConfirmedListener;
+    'ended': EndListener;
+    'failed': EndListener;
+    'newDTMF': DTMFListener;
+    'newInfo': InfoListener;
+    'hold': HoldListener;
+    'unhold': HoldListener;
+    'muted': MuteListener;
+    'unmuted': MuteListener;
+    'reinvite': ReInviteListener;
+    'update': UpdateListener;
+    'refer': ReferListener;
+    'replaces': ReferListener;
+    'sdp': SDPListener;
+    'icecandidate': IceCandidateListener;
+    'getusermediafailed': Listener_2;
+    'active' : Listener_2;
+    'msgHistoryUpdate' : Listener_2;
+    'newMessage' : Listener_2;
+    'peerconnection:createofferfailed': Listener_2;
+    'peerconnection:createanswerfailed': Listener_2;
+    'peerconnection:setlocaldescriptionfailed': Listener_2;
+    'peerconnection:setremotedescriptionfailed': Listener_2;
+}
+
 declare type Listener = (event: unknown) => void
 
 declare type Listener_2 = (event: unknown) => void
 
+declare type Listener_3 = (event: unknown) => void
+
 declare type ListenerCallbackFnType<T extends ListenersKeyType> = OpenSIPSEventMap[T]
 
 declare type ListenersKeyType = keyof OpenSIPSEventMap
 
+declare const MODULES: {
+    readonly AUDIO: "audio";
+    readonly VIDEO: "video";
+    readonly MSRP: "msrp";
+};
+
+declare type Modules = AudioModuleName | VideoModuleName | MSRPModuleName
+
 declare type MSRPInitializingListener = (sessionId: string | undefined) => void
 
 declare class MSRPMessage {
@@ -200,6 +388,34 @@ declare type MSRPMessageEventType = {
 
 declare type MSRPMessageListener = (event: MSRPMessageEventType) => void;
 
+declare class MSRPModule {
+    private context;
+    private activeMessages;
+    private extendedMessages;
+    private msrpHistory;
+    private isMSRPInitializingValue;
+    constructor(context: any);
+    get isMSRPInitializing(): boolean;
+    get getActiveMessages(): {
+        [key: string]: IMessage;
+    };
+    msrpAnswer(callId: string): void;
+    updateMSRPSession(value: IMessage): void;
+    private addMMSRPSession;
+    private addMSRPMessage;
+    messageTerminate(callId: string): void;
+    private addMessageSession;
+    private triggerMSRPListener;
+    private removeMMSRPSession;
+    private activeMessageListRemove;
+    private newMSRPSessionCallback;
+    private setIsMSRPInitializing;
+    initMSRP(target: string, body: string, options: any): void;
+    sendMSRP(msrpSessionId: string, body: string): void;
+}
+
+declare type MSRPModuleName = typeof MODULES.MSRP
+
 declare interface MSRPOptions extends AnswerOptions {
     eventHandlers?: Partial<MSRPSessionEventMap>
     anonymous?: boolean;
@@ -348,14 +564,14 @@ declare interface MSRPSessionEventMap_2 {
     'replaces': ReferListener;
     'sdp': SDPListener;
     'icecandidate': IceCandidateListener;
-    'getusermediafailed': Listener_2;
-    'active' : Listener_2;
-    'msgHistoryUpdate' : Listener_2;
-    'newMessage' : Listener_2;
-    'peerconnection:createofferfailed': Listener_2;
-    'peerconnection:createanswerfailed': Listener_2;
-    'peerconnection:setlocaldescriptionfailed': Listener_2;
-    'peerconnection:setremotedescriptionfailed': Listener_2;
+    'getusermediafailed': Listener_3;
+    'active' : Listener_3;
+    'msgHistoryUpdate' : Listener_3;
+    'newMessage' : Listener_3;
+    'peerconnection:createofferfailed': Listener_3;
+    'peerconnection:createanswerfailed': Listener_3;
+    'peerconnection:setlocaldescriptionfailed': Listener_3;
+    'peerconnection:setremotedescriptionfailed': Listener_3;
 }
 
 declare interface MSRPSessionExtended extends MSRPSession_2 {
@@ -412,179 +628,31 @@ declare interface OpenSIPSEventMap extends UAEventMap {
 
 declare class OpenSIPSJS extends UAExtended {
     private initialized;
-    private readonly options;
+    readonly options: IOpenSIPSJSOptions;
     private logger;
-    private VUMeter;
-    private readonly newRTCSessionEventName;
+    readonly newRTCSessionEventName: ListenersKeyType;
     private readonly registeredEventName;
     private readonly unregisteredEventName;
     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 activeStreamValue;
-    private initialStreamValue;
-    private currentActiveRoomIdValue;
-    private isCallAddingInProgress;
     private isMSRPInitializingValue;
     private isReconnecting;
+    audio: AudioModule;
+    msrp: MSRPModule;
+    video: VideoModule;
     private listenersList;
+    private modules;
     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;
     get sipDomain(): string;
-    get sipOptions(): {
-        mediaConstraints: {
-            audio: {
-                deviceId: {
-                    exact: string;
-                };
-            };
-            video: boolean;
-        };
-        session_timers: boolean;
-        extraHeaders: [string];
-        pcConfig: RTCConfiguration_2;
-    };
-    get currentActiveRoomId(): number | undefined;
-    private set currentActiveRoomId(value);
-    get autoAnswer(): boolean;
-    get callAddingInProgress(): string | undefined;
-    private set callAddingInProgress(value);
-    get isMSRPInitializing(): boolean;
-    get muteWhenJoin(): boolean;
-    get isDND(): boolean;
-    get speakerVolume(): number;
-    get microphoneInputLevel(): number;
-    get getActiveCalls(): {
-        [key: string]: ICall;
-    };
-    get hasActiveCalls(): boolean;
-    get getActiveMessages(): {
-        [key: string]: IMessage;
-    };
-    get getActiveRooms(): {
-        [key: number]: IRoom;
-    };
-    get isMuted(): boolean;
-    get getInputDeviceList(): MediaDeviceInfo[];
-    get getOutputDeviceList(): MediaDeviceInfo[];
-    get getUserMediaConstraints(): {
-        audio: {
-            deviceId: {
-                exact: string;
-            };
-        };
-        video: boolean;
-    };
-    get selectedInputDevice(): string;
-    get selectedOutputDevice(): string;
-    get activeStream(): MediaStream;
-    private setAvailableMediaDevices;
-    updateDeviceList(): Promise<void>;
-    private initializeMediaDevices;
-    setCallTime(value: ITimeData): void;
-    removeCallTime(callId: string): void;
-    private setTimeInterval;
-    private removeTimeInterval;
-    private stopCallTimer;
-    private emitVolumeChange;
-    setMetricsConfig(config: WebrtcMetricsConfigType): void;
-    sendDTMF(callId: string, value: string): void;
-    private setIsMuted;
-    private processMute;
-    mute(): void;
-    unmute(): void;
-    private processHold;
-    holdCall(callId: string, automatic?: boolean): Promise<void>;
-    unholdCall(callId: string): Promise<void>;
-    private cancelAllOutgoingUnanswered;
-    answerCall(callId: string): void;
-    msrpAnswer(callId: string): void;
-    moveCall(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;
-    private getActiveStream;
-    setMicrophone(dId: string): Promise<void>;
-    private setActiveStream;
-    setSpeaker(dId: string): Promise<void>;
-    private removeRoom;
-    private deleteRoomIfEmpty;
-    private checkInitialized;
-    private muteReconfigure;
-    private roomReconfigure;
-    private doConference;
-    private processCallerMute;
-    muteCaller(callId: string): void;
-    unmuteCaller(callId: string): void;
-    terminateCall(callId: string): void;
-    messageTerminate(callId: string): void;
-    transferCall(callId: string, target: string): Error;
-    mergeCall(roomId: number): void;
-    setDND(value: boolean): void;
-    private startCallTimer;
-    setActiveRoom(roomId: number | undefined): Promise<void>;
-    private getNewRoomId;
+    begin(): this;
     subscribe(type: string, listener: (c: RTCSessionExtended) => void): void;
     removeIListener(value: string): void;
-    private setupCall;
-    private addMessageSession;
     private triggerListener;
-    private triggerMSRPListener;
-    private removeCall;
-    private removeMMSRPSession;
-    private activeCallListRemove;
-    private activeMessageListRemove;
-    private newRTCSessionCallback;
-    private newMSRPSessionCallback;
     private setInitialized;
-    begin(): this;
-    setMuteWhenJoin(value: boolean): void;
-    setMicrophoneSensitivity(value: number): void;
-    setSpeakerVolume(value: number): void;
-    setAutoAnswer(value: boolean): void;
-    private setSelectedInputDevice;
-    private setSelectedOutputDevice;
-    private setIsMSRPInitializing;
-    private setCallMetrics;
-    private removeCallMetrics;
-    private getCallQuality;
-    private setupVUMeter;
-    private stopVUMeter;
-    setupStream(): Promise<void>;
-    private triggerAddStream;
-    initCall(target: string, addToCurrentRoom: boolean): void;
-    initMSRP(target: string, body: string, options: any): void;
-    sendMSRP(msrpSessionId: string, body: string): void;
-    private processRoomChange;
 }
 export default OpenSIPSJS;
 
@@ -673,19 +741,25 @@ declare class UAExtended extends UAConstructor implements UAExtendedInterface {
         ist: {};
         ict: {};
     };
+    _janus_sessions: any[];
     constructor(configuration: UAConfiguration);
     call(target: string, options?: CallOptionsExtended): RTCSession;
+    joinVideoCall(target: any, options: any): any;
     /**
      * new MSRPSession
      */
     newMSRPSession(session: MSRPSession, data: object): void;
+    newJanusSession(session: any, data: any): void;
     /**
      * MSRPSession destroyed.
      */
     destroyMSRPSession(session: MSRPSession): void;
+    destroyJanusSession(session: any): void;
     receiveRequest(request: any): void;
     startMSRP(target: string, options: MSRPOptions): MSRPSession;
+    startJanus(target: string, options: JanusOptions): MSRPSession;
     terminateMSRPSessions(options: object): void;
+    terminateJanusSessions(options: any): void;
     stop(): void;
 }
 
@@ -727,6 +801,15 @@ declare interface UAExtendedInterface_2 extends UA {
 
 declare type updateRoomListener = (value: RoomChangeEmitType) => void
 
+declare class VideoModule {
+    private context;
+    constructor(context: any);
+    get sipOptions(): any;
+    initCall(target: string): void;
+}
+
+declare type VideoModuleName = typeof MODULES.VIDEO
+
 declare interface WebrtcMetricsConfigType {
     refreshEvery?: number
     startAfter?: number