@voicenter-team/opensips-js 1.0.10

package/build/helpers/webrtcmetrics/utils/score.js

@@ -0,0 +1,235 @@
+"use strict";
+Object.defineProperty(exports, "__esModule", { value: true });
+exports.computeMOSForOutgoing = exports.computeMOS = exports.computeEModelMOSForOutgoing = exports.computeEModelMOS = void 0;
+const models_1 = require("./models");
+const helper_1 = require("./helper");
+const computeScore = (r) => {
+    if (r < 0) {
+        return 1;
+    }
+    if (r > 100) {
+        return 4.5;
+    }
+    return 1 + 0.035 * r + (7.0 / 1000000) * r * (r - 60) * (100 - r);
+};
+const computeEModelMOS = (report, kind = models_1.VALUE.AUDIO, previousReport, beforeLastReport, ssrc) => {
+    const currentSSRCReport = (0, helper_1.getSSRCDataFromBunch)(ssrc, report, models_1.DIRECTION.INBOUND);
+    const previousSSRCReport = (0, helper_1.getSSRCDataFromBunch)(ssrc, previousReport, models_1.DIRECTION.INBOUND);
+    const beforeLastSSRCReport = (0, helper_1.getSSRCDataFromBunch)(ssrc, beforeLastReport, models_1.DIRECTION.INBOUND);
+    const rttValues = [];
+    const jitterValues = [];
+    const packetsLoss = currentSSRCReport[kind].percent_packets_lost_in;
+    const currentJitter = currentSSRCReport[kind].delta_jitter_ms_in;
+    const lastJitter = (previousSSRCReport && previousSSRCReport[kind].delta_jitter_ms_in) || null;
+    const beforeLastJitter = (beforeLastSSRCReport && beforeLastSSRCReport[kind].delta_jitter_ms_in) ||
+        null;
+    const currentRTTConnectivity = report.data.delta_rtt_connectivity_ms;
+    const lastRTTConnectivity = (previousReport && previousReport.data.delta_rtt_connectivity_ms) ||
+        null;
+    const beforeLastRTTConnectivity = (beforeLastReport && beforeLastReport.data.delta_rtt_connectivity_ms) ||
+        null;
+    if (currentRTTConnectivity) {
+        rttValues.push(currentRTTConnectivity);
+    }
+    if (lastRTTConnectivity) {
+        rttValues.push(lastRTTConnectivity);
+    }
+    if (beforeLastRTTConnectivity) {
+        rttValues.push(beforeLastRTTConnectivity);
+    }
+    // Put Jitter values
+    if (currentJitter) {
+        jitterValues.push(currentJitter);
+    }
+    if (previousReport && lastJitter) {
+        jitterValues.push(lastJitter);
+    }
+    if (beforeLastReport && beforeLastJitter) {
+        jitterValues.push(beforeLastJitter);
+    }
+    const rtt = rttValues.length > 0 ? (0, helper_1.average)(rttValues) : 100; // Default value if no value;
+    const jitter = jitterValues.length > 0 ? (0, helper_1.average)(jitterValues) : 10; // Default value if no value;
+    const rx = 93.2 - packetsLoss;
+    const ry = 0.18 * rx * rx - 27.9 * rx + 1126.62;
+    const d = (rtt + jitter) / 2;
+    const h = d - 177.3 < 0 ? 0 : 1;
+    const id = 0.024 * d + 0.11 * (d - 177.3) * h;
+    const r = ry - id;
+    return computeScore(r);
+};
+exports.computeEModelMOS = computeEModelMOS;
+const computeEModelMOSForOutgoing = (report, kind = models_1.VALUE.AUDIO, previousReport, beforeLastReport, ssrc) => {
+    const currentSSRCReport = (0, helper_1.getSSRCDataFromBunch)(ssrc, report, models_1.DIRECTION.OUTBOUND);
+    const previousSSRCReport = (0, helper_1.getSSRCDataFromBunch)(ssrc, previousReport, models_1.DIRECTION.OUTBOUND);
+    const beforeLastSSRCReport = (0, helper_1.getSSRCDataFromBunch)(ssrc, beforeLastReport, models_1.DIRECTION.OUTBOUND);
+    const rttValues = [];
+    const jitterValues = [];
+    const packetsLoss = currentSSRCReport[kind].percent_packets_lost_out;
+    const currentRtt = currentSSRCReport[kind].delta_rtt_ms_out;
+    const lastRtt = (previousSSRCReport && previousSSRCReport[kind].delta_rtt_ms_out) || null;
+    const beforeLastRtt = (beforeLastSSRCReport && beforeLastSSRCReport[kind].delta_rtt_ms_out) ||
+        null;
+    const currentJitter = currentSSRCReport[kind].delta_jitter_ms_out;
+    const lastJitter = (previousSSRCReport && previousSSRCReport[kind].delta_jitter_ms_out) || null;
+    const beforeLastJitter = (beforeLastSSRCReport && beforeLastSSRCReport[kind].delta_jitter_ms_out) ||
+        null;
+    const currentRTTConnectivity = report.data.delta_rtt_connectivity_ms;
+    const lastRTTConnectivity = (previousReport && previousReport.data.delta_rtt_connectivity_ms) ||
+        null;
+    const beforeLastRTTConnectivity = (beforeLastReport && beforeLastReport.data.delta_rtt_connectivity_ms) ||
+        null;
+    // Put RTT values when exist
+    if (currentRtt) {
+        rttValues.push(currentRtt);
+    }
+    else if (currentRTTConnectivity) {
+        rttValues.push(currentRTTConnectivity);
+    }
+    if (lastRtt) {
+        rttValues.push(lastRtt);
+    }
+    else if (lastRTTConnectivity) {
+        rttValues.push(lastRTTConnectivity);
+    }
+    if (beforeLastRtt) {
+        rttValues.push(beforeLastRtt);
+    }
+    else if (beforeLastRTTConnectivity) {
+        rttValues.push(beforeLastRTTConnectivity);
+    }
+    // Put Jitter values
+    if (currentJitter) {
+        jitterValues.push(currentJitter);
+    }
+    if (previousReport && lastJitter) {
+        jitterValues.push(lastJitter);
+    }
+    if (beforeLastReport && beforeLastJitter) {
+        jitterValues.push(beforeLastJitter);
+    }
+    const rtt = rttValues.length > 0 ? (0, helper_1.average)(rttValues) : 100; // Default value if no value;
+    const jitter = jitterValues.length > 0 ? (0, helper_1.average)(jitterValues) : 10; // Default value if no value;
+    const rx = 93.2 - packetsLoss;
+    const ry = 0.18 * rx * rx - 27.9 * rx + 1126.62;
+    const d = (rtt + jitter) / 2;
+    const h = d - 177.3 < 0 ? 0 : 1;
+    const id = 0.024 * d + 0.11 * (d - 177.3) * h;
+    const r = ry - id;
+    return computeScore(r);
+};
+exports.computeEModelMOSForOutgoing = computeEModelMOSForOutgoing;
+const computeMOS = (report, kind = models_1.VALUE.AUDIO, previousReport, beforeLastReport, ssrc) => {
+    const currentSSRCReport = (0, helper_1.getSSRCDataFromBunch)(ssrc, report, models_1.DIRECTION.INBOUND);
+    const previousSSRCReport = (0, helper_1.getSSRCDataFromBunch)(ssrc, previousReport, models_1.DIRECTION.INBOUND);
+    const beforeLastSSRCReport = (0, helper_1.getSSRCDataFromBunch)(ssrc, beforeLastReport, models_1.DIRECTION.INBOUND);
+    const rttValues = [];
+    const jitterValues = [];
+    const packetsLoss = currentSSRCReport[kind].percent_packets_lost_in / 100;
+    const currentJitter = currentSSRCReport[kind].delta_jitter_ms_in;
+    const lastJitter = (previousSSRCReport && previousSSRCReport[kind].delta_jitter_ms_in) || null;
+    const beforeLastJitter = (beforeLastSSRCReport && beforeLastSSRCReport[kind].delta_jitter_ms_in) ||
+        null;
+    const currentRTTConnectivity = report.data.delta_rtt_connectivity_ms;
+    const lastRTTConnectivity = (previousReport && previousReport.data.delta_rtt_connectivity_ms) ||
+        null;
+    const beforeLastRTTConnectivity = (beforeLastReport && beforeLastReport.data.delta_rtt_connectivity_ms) ||
+        null;
+    // Put RTT values when exist
+    if (currentRTTConnectivity) {
+        rttValues.push(currentRTTConnectivity);
+    }
+    if (lastRTTConnectivity) {
+        rttValues.push(lastRTTConnectivity);
+    }
+    if (beforeLastRTTConnectivity) {
+        rttValues.push(beforeLastRTTConnectivity);
+    }
+    // Put Jitter values
+    if (currentJitter) {
+        jitterValues.push(currentJitter);
+    }
+    if (previousSSRCReport && lastJitter) {
+        jitterValues.push(lastJitter);
+    }
+    if (beforeLastSSRCReport && beforeLastJitter) {
+        jitterValues.push(beforeLastJitter);
+    }
+    const rtt = rttValues.length > 0 ? (0, helper_1.average)(rttValues) : 100; // Default value if no value;
+    const jitter = jitterValues.length > 0 ? (0, helper_1.average)(jitterValues) : 10; // Default value if no value;
+    const codecFittingParameterA = 0;
+    const codecFittingParameterB = 19.8;
+    const codecFittingParameterC = 29.7;
+    const ld = 30;
+    const d = (rtt + jitter) / 2 + ld;
+    const h = d - 177.3 < 0 ? 0 : 1;
+    const id = 0.024 * d + 0.11 * (d - 177.3) * h;
+    const ie = codecFittingParameterA +
+        codecFittingParameterB * Math.log(1 + codecFittingParameterC * packetsLoss);
+    const r = 93.2 - (ie + id);
+    return computeScore(r);
+};
+exports.computeMOS = computeMOS;
+const computeMOSForOutgoing = (report, kind = models_1.VALUE.AUDIO, previousReport, beforeLastReport, ssrc) => {
+    const currentSSRCReport = (0, helper_1.getSSRCDataFromBunch)(ssrc, report, models_1.DIRECTION.OUTBOUND);
+    const previousSSRCReport = (0, helper_1.getSSRCDataFromBunch)(ssrc, previousReport, models_1.DIRECTION.OUTBOUND);
+    const beforeLastSSRCReport = (0, helper_1.getSSRCDataFromBunch)(ssrc, beforeLastReport, models_1.DIRECTION.OUTBOUND);
+    const rttValues = [];
+    const jitterValues = [];
+    const packetsLoss = currentSSRCReport[kind].percent_packets_lost_out / 100;
+    const currentRtt = currentSSRCReport[kind].delta_rtt_ms_out;
+    const lastRtt = (previousSSRCReport && previousSSRCReport[kind].delta_rtt_ms_out) || null;
+    const beforeLastRtt = (beforeLastSSRCReport && beforeLastSSRCReport[kind].delta_rtt_ms_out) ||
+        null;
+    const currentJitter = currentSSRCReport[kind].delta_jitter_ms_out;
+    const lastJitter = (previousSSRCReport && previousSSRCReport[kind].delta_jitter_ms_out) || null;
+    const beforeLastJitter = (beforeLastSSRCReport && beforeLastSSRCReport[kind].delta_jitter_ms_out) ||
+        null;
+    const currentRTTConnectivity = report.data.delta_rtt_connectivity_ms;
+    const lastRTTConnectivity = (previousReport && previousReport.data.delta_rtt_connectivity_ms) ||
+        null;
+    const beforeLastRTTConnectivity = (beforeLastReport && beforeLastReport.data.delta_rtt_connectivity_ms) ||
+        null;
+    // Put RTT values when exist
+    if (currentRtt) {
+        rttValues.push(currentRtt);
+    }
+    else if (currentRTTConnectivity) {
+        rttValues.push(currentRTTConnectivity);
+    }
+    if (lastRtt) {
+        rttValues.push(lastRtt);
+    }
+    else if (lastRTTConnectivity) {
+        rttValues.push(lastRTTConnectivity);
+    }
+    if (beforeLastRtt) {
+        rttValues.push(beforeLastRtt);
+    }
+    else if (beforeLastRTTConnectivity) {
+        rttValues.push(beforeLastRTTConnectivity);
+    }
+    // Put Jitter values
+    if (currentJitter) {
+        jitterValues.push(currentJitter);
+    }
+    if (previousSSRCReport && lastJitter) {
+        jitterValues.push(lastJitter);
+    }
+    if (beforeLastSSRCReport && beforeLastJitter) {
+        jitterValues.push(beforeLastJitter);
+    }
+    const rtt = rttValues.length > 0 ? (0, helper_1.average)(rttValues) : 100; // Default value if no value;
+    const jitter = jitterValues.length > 0 ? (0, helper_1.average)(jitterValues) : 10; // Default value if no value;
+    const codecFittingParameterA = 0;
+    const codecFittingParameterB = 19.8;
+    const codecFittingParameterC = 29.7;
+    const ld = 30;
+    const d = (rtt + jitter) / 2 + ld;
+    const h = d - 177.3 < 0 ? 0 : 1;
+    const id = 0.024 * d + 0.11 * (d - 177.3) * h;
+    const ie = codecFittingParameterA +
+        codecFittingParameterB * Math.log(1 + codecFittingParameterC * packetsLoss);
+    const r = 93.2 - (ie + id);
+    return computeScore(r);
+};
+exports.computeMOSForOutgoing = computeMOSForOutgoing;

package/build/helpers/webrtcmetrics/utils/shortUUId.d.ts

@@ -0,0 +1 @@
+export default function shortUUID(): string;

package/build/helpers/webrtcmetrics/utils/shortUUId.js

@@ -0,0 +1,7 @@
+"use strict";
+Object.defineProperty(exports, "__esModule", { value: true });
+function shortUUID() {
+    const uuid = +new Date();
+    return `${uuid}`;
+}
+exports.default = shortUUID;

package/build/index.d.ts

@@ -0,0 +1,170 @@
+import JsSIP from 'jssip';
+import UA from './helpers/UA';
+import { TempTimeData, ITimeData } from './helpers/time.helper';
+import { WebrtcMetricsConfigType, MediaDeviceType } 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;
+    };
+    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;
+}
+declare class OpenSIPSJS extends UA {
+    private initialized;
+    private readonly options;
+    private readonly newRTCSessionEventName;
+    private readonly activeCalls;
+    private _currentActiveRoomId;
+    private _callAddingInProgress;
+    private state;
+    constructor(options: IOpenSIPSJSOptions);
+    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: import("@/types/rtc").RTCConfiguration;
+    };
+    get currentActiveRoomId(): number | undefined;
+    private set currentActiveRoomId(value);
+    get callAddingInProgress(): string | undefined;
+    private set callAddingInProgress(value);
+    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 getActiveRooms(): {
+        [key: number]: IRoom;
+    };
+    get isMuted(): boolean;
+    set isMuted(value: boolean);
+    get getInputDeviceList(): MediaDeviceInfo[];
+    get getOutputDeviceList(): MediaDeviceInfo[];
+    get getUserMediaConstraints(): {
+        audio: {
+            deviceId: {
+                exact: string;
+            };
+        };
+        video: boolean;
+    };
+    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>;
+    setMediaDevices(setDefaults?: boolean): Promise<void>;
+    setCallTime(value: ITimeData): void;
+    removeCallTime(callId: string): void;
+    private setTimeInterval;
+    private removeTimeInterval;
+    private _stopCallTimer;
+    setMetricsConfig(config: WebrtcMetricsConfigType): void;
+    sendDTMF(callId: string, value: string): void;
+    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;
+        automatic?: boolean;
+    }): void;
+    private _cancelAllOutgoingUnanswered;
+    callAnswer(callId: string): void;
+    callMove(callId: string, roomId: number): Promise<void>;
+    updateCall(value: ICall): void;
+    updateRoom(value: IRoomUpdate): void;
+    private _addCall;
+    private _addCallStatus;
+    private _updateCallStatus;
+    private _removeCallStatus;
+    private _addRoom;
+    setMicrophone(dId: string): Promise<void>;
+    private _setOriginalStream;
+    setSpeaker(dId: string): Promise<void>;
+    private removeRoom;
+    private deleteRoomIfEmpty;
+    private checkInitialized;
+    private muteReconfigure;
+    private roomReconfigure;
+    private _doConference;
+    _muteReconfigure(call: ICall): void;
+    muteCaller(callId: string, value: boolean): void;
+    callTerminate(callId: string): void;
+    callTransfer(callId: string, target: string): void;
+    callMerge(roomId: number): void;
+    setDND(value: boolean): void;
+    private _startCallTimer;
+    setCurrentActiveRoomId(roomId: number | undefined): Promise<void>;
+    private getNewRoomId;
+    subscribe(type: string, listener: (c: RTCSessionExtended) => void): void;
+    removeIListener(value: string): void;
+    private addCall;
+    private _triggerListener;
+    private _removeCall;
+    private _activeCallListRemove;
+    private newRTCSessionCallback;
+    private setInitialized;
+    start(): this;
+    setMuteWhenJoin(value: boolean): void;
+    private _setCallMetrics;
+    private _removeCallMetrics;
+    private _getCallQuality;
+    private _triggerAddStream;
+    doCall({ target, addToCurrentRoom }: IDoCallParam): void;
+    callChangeRoom({ callId, roomId }: {
+        callId: string;
+        roomId: number;
+    }): Promise<void>;
+}
+export default OpenSIPSJS;