@voicenter-team/opensips-js 1.0.10

package/package.json

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+{
+  "name": "@voicenter-team/opensips-js",
+  "version": "1.0.10",
+  "description": "The JS package for opensips",
+  "default": "src/index.ts",
+  "main": "build/index.js",
+  "types": "build/index.d.ts",
+  "scripts": {
+    "build": "tsc --project tsconfig.json && tsc-alias -p tsconfig.json",
+    "dev": "vite",
+    "demo-build": "vite build",
+    "docs-build": "yarn run demo-build && cd docs && yarn && yarn run build",
+    "test": "echo \"Error: no test specified\" && exit 1",
+    "ts-check": "tsc --noEmit",
+    "lint": "eslint . --ext .js,.jsx,.cjs,.mjs,.ts --fix --ignore-path .gitignore"
+  },
+  "engines": {
+    "node": ">= 16",
+    "npm": "please-use-yarn",
+    "yarn": ">= 1.22.4"
+  },
+  "precommit": "ts-check, lint",
+  "repository": {
+    "type": "git",
+    "url": "git+ssh://git@github.com/VoicenterTeam/opensips-js.git"
+  },
+  "author": "Bohdan Konechnyi",
+  "license": "ISC",
+  "bugs": {
+    "url": "https://github.com/VoicenterTeam/opensips-js/issues"
+  },
+  "homepage": "https://github.com/VoicenterTeam/opensips-js#readme",
+  "devDependencies": {
+    "@types/node": "^18.14.2",
+    "@typescript-eslint/eslint-plugin": "^5.53.0",
+    "@typescript-eslint/parser": "^5.53.0",
+    "@vitejs/plugin-vue": "^4.2.3",
+    "dotenv": "^16.0.3",
+    "eslint": "^8.35.0",
+    "pre-commit": "^1.2.2",
+    "process": "^0.11.10",
+    "regenerator-runtime": "^0.13.11",
+    "tsc-alias": "^1.8.6",
+    "typescript": "^4.9.5",
+    "vite": "^4.3.8",
+    "vite-plugin-singlefile": "^0.13.5",
+    "vue": "3.2.25"
+  },
+  "dependencies": {
+    "@types/mime": "^3.0.1",
+    "generate-unique-id": "^2.0.1",
+    "jssip": "^3.10.0",
+    "loglevel": "^1.8.1",
+    "mime": "^3.0.0",
+    "p-iteration": "^1.1.8"
+  },
+  "files": [
+    "build",
+    "src/types"
+  ]
+}

package/src/types/declarations.d.ts

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+// export {}
+// declare global {
+//     interface Window {
+//         webkitAudioContext: typeof AudioContext
+//     }
+// }

package/src/types/generic.d.ts

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+export type Writeable<T> = { -readonly [P in keyof T]: T[P] }

package/src/types/listeners.d.ts

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+import { ICall, RoomChangeEmitType } from '@/types/rtc'
+import { UAEventMap } from 'jssip/lib/UA'
+
+export type readyListener = (value: boolean) => void
+export type changeActiveCallsListener = (event: { [key: string]: ICall }) => void
+export type TestEventListener = (event: { test: string }) => void
+export type ActiveRoomListener = (event: number | undefined) => void
+export type CallAddingProgressListener = (callId: string | undefined) => void
+export type RoomDeletedListener = (roomId: number) => void
+export type changeActiveInputMediaDeviceListener = (event: string) => void
+export type changeActiveOutputMediaDeviceListener = (event: string) => void
+export type changeAvailableDeviceListListener = (event: Array<MediaDeviceInfo>) => void
+export type changeMuteWhenJoinListener = (value: boolean) => void
+export type changeIsDNDListener = (value: boolean) => void
+export type changeIsMutedListener = (value: boolean) => void
+export type changeOriginalStreamListener = (value: MediaStream) => void
+export type addRoomListener = (value: RoomChangeEmitType) => void
+export type updateRoomListener = (value: RoomChangeEmitType) => void
+export type removeRoomListener = (value: RoomChangeEmitType) => void
+
+export interface OpenSIPSEventMap extends UAEventMap {
+    ready: readyListener
+    changeActiveCalls: changeActiveCallsListener
+    callConfirmed: TestEventListener
+    currentActiveRoomChanged: ActiveRoomListener
+    callAddingInProgressChanged: CallAddingProgressListener
+    roomDeleted: RoomDeletedListener
+    changeActiveInputMediaDevice: changeActiveInputMediaDeviceListener
+    changeActiveOutputMediaDevice: changeActiveOutputMediaDeviceListener
+    changeAvailableDeviceList: changeAvailableDeviceListListener
+    changeMuteWhenJoin: changeMuteWhenJoinListener
+    changeIsDND: changeIsDNDListener
+    changeIsMuted: changeIsMutedListener
+    changeOriginalStream: changeOriginalStreamListener
+    addRoom: addRoomListener
+    updateRoom: updateRoomListener
+    removeRoom: removeRoomListener
+}
+
+export type ListenersKeyType = keyof OpenSIPSEventMap
+export type ListenersCallbackFnType = OpenSIPSEventMap[ListenersKeyType]
+export type ListenerCallbackFnType<T extends ListenersKeyType> = OpenSIPSEventMap[T]

package/src/types/rtc.d.ts

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+import { MediaStreamConstraints } from 'lib.dom.d.ts'
+import { Partial } from 'lib.es5.d.ts'
+import {
+    AnswerOptions,
+    EndEvent,
+    IncomingAckEvent,
+    IncomingEvent,
+    OutgoingAckEvent,
+    OutgoingEvent,
+    RTCSession, RTCSessionEventMap,
+} from 'jssip/lib/RTCSession'
+import { UAConfiguration } from 'jssip/lib/UA'
+
+export type IntervalType = ReturnType<typeof setInterval>
+
+export type ListenerEventType = EndEvent | IncomingEvent | OutgoingEvent | IncomingAckEvent | OutgoingAckEvent
+
+export type RTCBundlePolicy = 'balanced' | 'max-bundle' | 'max-compat'
+export type RTCIceTransportPolicy = 'all' | 'relay'
+export type RTCRtcpMuxPolicy = 'require'
+
+export interface RTCIceServer {
+    credential?: string;
+    urls: string | string[];
+    username?: string;
+}
+
+export interface RTCConfiguration {
+    bundlePolicy?: RTCBundlePolicy;
+    certificates?: RTCCertificate[];
+    iceCandidatePoolSize?: number;
+    iceServers?: RTCIceServer[];
+    iceTransportPolicy?: RTCIceTransportPolicy;
+    rtcpMuxPolicy?: RTCRtcpMuxPolicy;
+}
+
+export interface StreamMediaType extends HTMLAudioElement {
+    className: string
+    setSinkId (id: string): Promise<void>
+}
+
+export interface AnswerOptionsExtended extends AnswerOptions {
+    mediaConstraints?: MediaStreamConstraints
+}
+
+export interface RTCSessionExtended extends RTCSession {
+    id: string
+    _automaticHold: boolean
+    _id: string
+    _localHold: boolean
+    _audioMuted: boolean
+    _cancel_reason: string
+    _contact: string
+    _end_time: Date
+    _eventsCount: number
+    _from_tag: string
+    _is_canceled: boolean
+    _is_confirmed: boolean
+    _late_sdp: string
+    _videoMuted: boolean
+    _status: number
+    _remote_identity: string
+    answer(options?: AnswerOptionsExtended): void
+}
+
+export interface ICall extends RTCSessionExtended {
+    roomId?: number
+    localMuted?: boolean
+    localHold?: boolean
+    audioTag?: StreamMediaType
+}
+
+export type RoomChangeEmitType = {
+    room: IRoom
+    roomList: { [key: number]: IRoom }
+}
+
+export interface MediaEvent extends Event {
+    stream: MediaStream
+}
+
+export interface IDoCallParam {
+    target: string
+    addToCurrentRoom: boolean
+}
+
+export interface IRoom {
+    started: Date
+    incomingInProgress: boolean
+    roomId: number
+}
+
+export interface ICallStatus {
+    isMoving: boolean
+    isTransferring: boolean
+    isMerging: boolean
+}
+
+export interface ICallStatusUpdate {
+    callId: string
+    isMoving?: boolean
+    isTransferring?: boolean
+    isMerging?: boolean
+}
+
+export type IRoomUpdate = Omit<IRoom, 'started'> & {
+    started?: Date
+}
+
+export interface IOpenSIPSJSOptions {
+    configuration: Omit<UAConfiguration, 'sockets'>,
+    socketInterfaces: [ string ]
+    sipDomain: string
+    sipOptions: {
+        session_timers: boolean
+        extraHeaders: [ string ]
+        pcConfig: RTCConfiguration
+    }
+}
+
+export interface TriggerListenerOptions {
+    listenerType: string
+    session: RTCSessionExtended
+    event?:  ListenerEventType
+}
+
+/* UA */
+export interface CallOptionsExtended extends AnswerOptionsExtended {
+    eventHandlers?: Partial<RTCSessionEventMap>;
+    anonymous?: boolean;
+    fromUserName?: string;
+    fromDisplayName?: string;
+}

package/src/types/webrtcmetrics.d.ts

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+export interface WebrtcMetricsConfigType {
+    refreshEvery?: number
+    startAfter?: number
+    stopAfter?: number
+    verbose?: boolean
+    pname?: string
+    cid?: string
+    uid?: string
+    record?: boolean
+    ticket?: boolean
+}
+
+export type ProbeDirectionType = 'inbound' | 'outbound'
+
+export interface ProbeMetricInType {
+    level_in: number
+    codec_id_in: string
+    codec_in: { mime_type: null | number, clock_rate: null | number, sdp_fmtp_line: null | number }
+    delta_jitter_ms_in: number
+    percent_packets_lost_in: number
+    delta_packets_in: number
+    delta_packets_lost_in: number
+    total_packets_in: number
+    total_packets_lost_in: number
+    total_KBytes_in: number
+    delta_KBytes_in: number
+    delta_kbs_in: number,
+    mos_in: number
+    mos_emodel_in: number
+    ssrc: string
+    direction: ProbeDirectionType
+}
+export interface ProbeMetricOutType {
+    level_out: number
+    codec_id_out: string
+    codec_out: { mime_type: null | number, clock_rate: null | number, sdp_fmtp_line: null | number }
+    delta_jitter_ms_out: number
+    delta_rtt_ms_out: null | number
+    total_rtt_ms_out: number
+    total_rtt_measure_out: number
+    percent_packets_lost_out: number
+    delta_packets_out: number
+    delta_packets_lost_out: number
+    total_packets_out: number
+    total_packets_lost_out: number
+    total_KBytes_out: number
+    delta_KBytes_out: number
+    delta_kbs_out: number
+    timestamp_out: null | number
+    mos_out: number
+    mos_emodel_out: number
+    ssrc: string
+    direction: ProbeDirectionType
+}
+
+export interface Probe {
+    audio: { [key: string]: ProbeMetricInType | ProbeMetricOutType }
+}
+
+interface MetricAudioData extends ProbeMetricInType {
+    callId?: string
+}
+
+export type MediaDeviceType = 'input' | 'output'