@voicenter-team/opensips-js 1.0.46 → 1.0.74

package/README.md

@@ -29,23 +29,25 @@ const openSIPSJS = new OpenSIPSJS({
 Then you will be able to call next methods on openSIPSJS instance:
 
 ### Methods
-- `async setMediaDevices(setDefaults: Boolean = false)` - will set up media devices
-- `async setMicrophone(deviceId: String)` - set passed device as input device for calls
-- `async setSpeaker(deviceId: String)` - set passed device as output device for calls
-- `async setCurrentActiveRoomId(roomId: Number)` - move to the room
-- `doCallHold({callId: Number, toHold: Boolean, automatic: Boolean})` - hold/unhold call by id
-- `doCall(target: String, addToCurrentRoom: Boolean)` - call to the target. If addToCurrentRoom is true then the call will be added to the user's current room
-- `callTerminate(callId: String)` - terminate call
-- `callTransfer({callId: String, target: String})` - transfer call to target
-- `callMerge(roomId: Number)` - merge calls in specific room
-- `callAnswer(callId: String)` - answer the call
-- `setMetricsConfig(config: WebrtcMetricsConfigType)` - set the metric config (used for audio quality indicator)
-- `doMute(muted: Boolean)` - set the agent muteness
-- `setDND(value: Boolean)` - set the agent "Do not disturb" status
-- `async callChangeRoom({callId: String, roomId: Number})` - move call to the room
-- `callMove({callId: String, roomId: Number})` - Same as callChangeRoom. Move call to the specific room
-- `subscribe({type: String, listener: function})` - subscribe to an event. Available events: `new_call`, `ended`, `progress`, `failed`, `confirmed`
-- `removeIListener(type: String)` - remove event listener
+- `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
+- `setDND(value: Boolean): void` - set the agent "Do not disturb" status
+- `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
+- `setMetricsConfig(config: WebrtcMetricsConfigType): void` - set the metric config (used for audio quality indicator)
 
 WebrtcMetricsConfigType
 
@@ -72,4 +74,4 @@ Also there are next public fields on openSIPSJS instance:
 - `selectedInputDevice: String` - returns current selected input device id
 - `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
+- `isMuted: Boolean` - returns if the agent is muted

package/dist/index.d.ts

@@ -63,6 +63,8 @@ declare type changeActiveMessagesListener = (event: { [key: string]: IMessage })
 
 declare type changeActiveOutputMediaDeviceListener = (event: string) => void
 
+declare type changeActiveStreamListener = (value: MediaStream) => void
+
 declare type changeAvailableDeviceListListener = (event: Array<MediaDeviceInfo>) => void
 
 declare type changeCallMetricsListener = (event: { [key: string]: any }) => void
@@ -71,13 +73,18 @@ declare type changeCallStatusListener = (event: { [key: string]: ICallStatus })
 
 declare type changeCallTimeListener = (event: { [key: string]: ITimeData }) => void
 
+declare type changeCallVolumeListener = (event: ChangeVolumeEventType) => void
+
 declare type changeIsDNDListener = (value: boolean) => void
 
 declare type changeIsMutedListener = (value: boolean) => void
 
 declare type changeMuteWhenJoinListener = (value: boolean) => void
 
-declare type changeOriginalStreamListener = (value: MediaStream) => void
+declare type ChangeVolumeEventType = {
+    callId: string
+    volume: number
+}
 
 declare type CommonLogMethodType = (...args: unknown[]) => void
 
@@ -116,11 +123,6 @@ declare interface ICallStatus {
     isMerging: boolean
 }
 
-declare interface IDoCallParam {
-    target: string
-    addToCurrentRoom: boolean
-}
-
 declare interface IMessage extends MSRPSessionExtended {
     roomId?: number
     localMuted?: boolean
@@ -396,13 +398,14 @@ declare interface OpenSIPSEventMap extends UAEventMap {
     changeMuteWhenJoin: changeMuteWhenJoinListener
     changeIsDND: changeIsDNDListener
     changeIsMuted: changeIsMutedListener
-    changeOriginalStream: changeOriginalStreamListener
+    changeActiveStream: changeActiveStreamListener
     addRoom: addRoomListener
     updateRoom: updateRoomListener
     removeRoom: removeRoomListener
     changeCallStatus: changeCallStatusListener
     changeCallTime: changeCallTimeListener
     changeCallMetrics: changeCallMetricsListener
+    changeCallVolume: changeCallVolumeListener
     newMSRPMessage: MSRPMessageListener
     newMSRPSession: MSRPSessionListener
 }
@@ -411,6 +414,7 @@ declare class OpenSIPSJS extends UAExtended {
     private initialized;
     private readonly options;
     private logger;
+    private VUMeter;
     private readonly newRTCSessionEventName;
     private readonly registeredEventName;
     private readonly unregisteredEventName;
@@ -436,7 +440,8 @@ declare class OpenSIPSJS extends UAExtended {
     private callMetrics;
     private timeIntervals;
     private metricConfig;
-    private originalStreamValue;
+    private activeStreamValue;
+    private initialStreamValue;
     private currentActiveRoomIdValue;
     private isCallAddingInProgress;
     private isMSRPInitializingValue;
@@ -491,32 +496,31 @@ declare class OpenSIPSJS extends UAExtended {
         };
         video: boolean;
     };
-    get getInputDefaultDevice(): MediaDeviceInfo;
-    get getOutputDefaultDevice(): MediaDeviceInfo;
     get selectedInputDevice(): string;
     get selectedOutputDevice(): string;
-    get originalStream(): MediaStream;
+    get activeStream(): MediaStream;
     private setAvailableMediaDevices;
     updateDeviceList(): Promise<void>;
-    setMediaDevices(setDefaults?: boolean): 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;
-    doMute(value: boolean): void;
-    doCallHold({ callId, toHold, automatic }: {
-        callId: string;
-        toHold: boolean;
-        automatic?: boolean;
-    }): void;
+    private processMute;
+    mute(): void;
+    unmute(): void;
+    private processHold;
+    holdCall(callId: string, automatic?: boolean): Promise<void>;
+    unholdCall(callId: string): Promise<void>;
     private cancelAllOutgoingUnanswered;
-    callAnswer(callId: string): void;
+    answerCall(callId: string): void;
     msrpAnswer(callId: string): void;
-    callMove(callId: string, roomId: number): Promise<void>;
+    moveCall(callId: string, roomId: number): Promise<void>;
     updateCall(value: ICall): void;
     updateMSRPSession(value: IMessage): void;
     updateRoom(value: IRoomUpdate): void;
@@ -528,8 +532,9 @@ declare class OpenSIPSJS extends UAExtended {
     private updateCallStatus;
     private removeCallStatus;
     private addRoom;
+    private getActiveStream;
     setMicrophone(dId: string): Promise<void>;
-    private setOriginalStream;
+    private setActiveStream;
     setSpeaker(dId: string): Promise<void>;
     private removeRoom;
     private deleteRoomIfEmpty;
@@ -537,14 +542,16 @@ declare class OpenSIPSJS extends UAExtended {
     private muteReconfigure;
     private roomReconfigure;
     private doConference;
-    muteCaller(callId: string, value: boolean): void;
-    callTerminate(callId: string): void;
+    private processCallerMute;
+    muteCaller(callId: string): void;
+    unmuteCaller(callId: string): void;
+    terminateCall(callId: string): void;
     messageTerminate(callId: string): void;
-    callTransfer(callId: string, target: string): void;
-    callMerge(roomId: number): void;
+    transferCall(callId: string, target: string): Error;
+    mergeCall(roomId: number): void;
     setDND(value: boolean): void;
     private startCallTimer;
-    setCurrentActiveRoomId(roomId: number | undefined): Promise<void>;
+    setActiveRoom(roomId: number | undefined): Promise<void>;
     private getNewRoomId;
     subscribe(type: string, listener: (c: RTCSessionExtended) => void): void;
     removeIListener(value: string): void;
@@ -570,11 +577,14 @@ declare class OpenSIPSJS extends UAExtended {
     private setCallMetrics;
     private removeCallMetrics;
     private getCallQuality;
+    private setupVUMeter;
+    private stopVUMeter;
+    setupStream(): Promise<void>;
     private triggerAddStream;
-    doCall({ target, addToCurrentRoom }: IDoCallParam): void;
+    initCall(target: string, addToCurrentRoom: boolean): void;
     initMSRP(target: string, body: string, options: any): void;
     sendMSRP(msrpSessionId: string, body: string): void;
-    private callChangeRoom;
+    private processRoomChange;
 }
 export default OpenSIPSJS;