@wazo/sdk 0.38.21 → 0.39.0-beta.1

package/lib/domain/Phone/WebRTCPhone.js

@@ -1,2733 +1,1464 @@
-'use strict';
-
-Object.defineProperty(exports, '__esModule', { value: true });
-
-var invitation = require('sip.js/lib/api/invitation');
-var sessionState = require('sip.js/lib/api/session-state');
-var moment = require('moment');
-require('webrtc-adapter');
-require('sip.js/lib/api/user-agent-state');
-require('sip.js/lib/core/messages/parser');
-require('sip.js/lib/core/messages/methods/constants');
-require('sip.js/lib/grammar/uri');
-require('sip.js/lib/api/user-agent');
-require('sip.js/lib/platform/web/modifiers/modifiers');
-require('sip.js/lib/api/registerer');
-require('sip.js/lib/api/inviter');
-require('sip.js/lib/api/messager');
-require('sip.js/lib/api/registerer-state');
-require('sip.js/lib/api/transport-state');
-require('sip.js/lib/platform/web/session-description-handler/peer-connection-configuration-default');
-require('getstats');
-var EventEmitter = require('events');
-require('sip.js/lib/platform/web/session-description-handler/session-description-handler');
-require('sip.js/lib/platform/web/session-description-handler/session-description-handler-options');
-require('js-base64');
-require('sdp-transform');
-var jsrsasign = require('jsrsasign');
-require('json-to-graphql-query/lib/jsonToGraphQLQuery');
-
-function _interopDefaultLegacy (e) { return e && typeof e === 'object' && 'default' in e ? e : { 'default': e }; }
-
-var moment__default = /*#__PURE__*/_interopDefaultLegacy(moment);
-var EventEmitter__default = /*#__PURE__*/_interopDefaultLegacy(EventEmitter);
-
-//      
-
-var newFrom = (instance     , ToClass     ) => {
-  const args = {};
-  Object.getOwnPropertyNames(instance).forEach(prop => {
-    args[prop] = instance[prop];
-  });
-
-  return new ToClass(args);
-};
-
-//      
-/* eslint-disable no-param-reassign */
-
-var updateFrom = (instance     , from     ) => {
-  Object.keys(from).forEach(key => {
-    // $FlowFixMe
-    if (
-      (!instance[key] && from[key])
-      || (typeof instance[key] !== 'undefined' && typeof from[key] !== 'undefined' && instance[key] !== from[key])
-    ) {
-      // $FlowFixMe
-      instance[key] = from[key];
-    }
-  });
-};
-
-//      
-
-                             
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-                            
-              
-                 
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-                         
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-                 
-                     
-                    
-                       
-                    
-                         
-                     
-                           
-                       
-                      
-  
-
-class CallSession {
-              
-
-  // Wazo's callId, like `1594062407.xxxx`
-                 
-
-                      
-
-  // Used to retrieve the real callee when doing indirect transfer
-                          
-
-                        
-
-                 
-
-                       
-
-                      
-
-                     // = creationTime
-
-                    
-
-                 
-
-                    
-
-                            
-
-                          
-
-                   
-
-  // Should be computed ?
-                  
-
-  // Asterisk callId, like `7aed6793-4405-466d-873e-92d21c2fef9f`
-                    
-
-                     
-
-                 
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-                      
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-                                    
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-                         
-
-                      
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-                   
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-                         
-
-               
-
-                     
-
-                           
-
-                      
-
-                      
-
-  constructor({
-    answered,
-    answeredBySystem,
-    isCaller,
-    displayName,
-    callId,
-    muted,
-    videoMuted,
-    number,
-    paused,
-    ringing,
-    startTime,
-    creationTime,
-    endTime,
-    cameraEnabled,
-    dialedExtension,
-    sipCallId,
-    sipStatus,
-    callerNumber,
-    call,
-    autoAnswer,
-    ignored,
-    screensharing,
-    recording,
-    recordingPaused,
-    videoRemotelyDowngraded,
-    sipSession,
-    answerTime,
-    conference,
-  }                      ) {
-    this.callId = callId;
-    this.sipCallId = sipCallId;
-    this.displayName = displayName;
-    this.number = number;
-    this.creationTime = creationTime;
-    this.startTime = startTime;
-    this.endTime = endTime;
-    this.isCaller = isCaller;
-    this.answered = answered;
-    this.answeredBySystem = answeredBySystem;
-    this.ringing = ringing;
-    this.paused = paused;
-    this.muted = muted;
-    this.videoMuted = videoMuted;
-    this.callerNumber = callerNumber;
-    this.cameraEnabled = cameraEnabled;
-    this.dialedExtension = dialedExtension || '';
-    this.call = call;
-    this.sipStatus = sipStatus;
-    this.autoAnswer = autoAnswer || false;
-    this.ignored = ignored || false;
-    this.screensharing = screensharing || false;
-    this.recording = recording || false;
-    this.recordingPaused = recordingPaused || false;
-    this.videoRemotelyDowngraded = videoRemotelyDowngraded;
-    this.sipSession = sipSession;
-    this.answerTime = answerTime || this.answerTime;
-    this.conference = conference;
-
-    // Useful to compare instead of instanceof with minified code
-    this.type = 'CallSession';
-  }
-
-  resume() {
-    this.paused = false;
-  }
-
-  hold() {
-    this.paused = true;
-  }
-
-  mute() {
-    this.muted = true;
-  }
-
-  unmute() {
-    this.muted = false;
-  }
-
-  muteVideo() {
-    this.videoMuted = true;
-  }
-
-  unmuteVideo() {
-    this.videoMuted = false;
-  }
-
-  answer() {
-    this.answerTime = new Date();
-  }
-
-  systemAnswer() {
-    this.answeredBySystem = true;
-  }
-
-  enableCamera() {
-    this.cameraEnabled = true;
-  }
-
-  disableCamera() {
-    this.cameraEnabled = false;
-  }
-
-  ignore() {
-    this.ignored = true;
-  }
-
-  startScreenSharing() {
-    this.screensharing = true;
-  }
-
-  stopScreenSharing() {
-    this.screensharing = false;
-  }
-
-  setIsConference(conference         ) {
-    this.conference = conference;
-  }
-
-  isIncoming()          {
-    return !this.isCaller && !this.answered;
-  }
-
-  isOutgoing()          {
-    return this.isCaller && !this.answered;
-  }
-
-  isActive()          {
-    return this.answered || this.isOutgoing();
-  }
-
-  isAnswered()          {
-    return this.answered;
-  }
-
-  isAnsweredBySystem()          {
-    return this.answeredBySystem;
-  }
-
-  isRinging()          {
-    return this.ringing;
-  }
-
-  isOnHold()          {
-    return this.paused;
-  }
-
-  isMuted()          {
-    return this.muted;
-  }
-
-  isVideoMuted()          {
-    return this.videoMuted;
-  }
-
-  isCameraEnabled()          {
-    return this.cameraEnabled;
-  }
-
-  isIgnored()          {
-    return this.ignored;
-  }
-
-  isScreenSharing()          {
-    return this.screensharing;
-  }
-
-  isRecording()          {
-    return this.recording;
-  }
-
-  recordingIsPaused()          {
-    return this.recordingPaused;
-  }
-
-  hasAnInitialInterceptionNumber()          {
-    return this.number.startsWith('*8');
-  }
-
-  isAnInterception()          {
-    return this.dialedExtension.startsWith('*8');
-  }
-
-  isEstablished()          {
-    return this.sipStatus === sessionState.SessionState.Established;
-  }
-
-  isTerminating()          {
-    return this.sipStatus === sessionState.SessionState.Terminating;
-  }
-
-  isTerminated()          {
-    return this.sipStatus === sessionState.SessionState.Terminated;
-  }
-
-  isConference()          {
-    return this.conference;
-  }
-
-  getElapsedTimeInSeconds() {
-    if (!this.startTime) {
-      return 0;
-    }
-    return (Date.now() - this.startTime) / 1000;
-  }
-
-  getId() {
-    return this.sipCallId || this.callId;
-  }
-
-  is(callSession              ) {
-    if (!callSession) {
-      return false;
-    }
-
-    return this.isId(callSession.sipCallId) || this.isId(callSession.callId);
-  }
-
-  isId(id        )          {
-    return (
-      this.getId() === id || (this.sipCallId && this.sipCallId === id) || (Boolean(this.callId) && this.callId === id)
-    );
-  }
-
-  updateFrom(session             ) {
-    updateFrom(this, session);
-  }
-
-  separateDisplayName()                                          {
-    const names = this.displayName.split(' ');
-    const firstName = names[0];
-    const lastName = names.slice(1).join(' ');
-
-    return { firstName, lastName };
-  }
-
-  getTalkingToIds() {
-    return this.call ? this.call.talkingToIds : [];
-  }
-
-  setVideoRemotelyDowngraded(value         ) {
-    this.videoRemotelyDowngraded = value;
-  }
-
-  isVideoRemotelyDowngraded() {
-    return this.videoRemotelyDowngraded;
-  }
-
-  static newFrom(callSession             ) {
-    return newFrom(callSession, CallSession);
-  }
-
-  // Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
-  set answered(value         ) {
-    this.answerTime = value ? this.answerTime || new Date() : null;
-  }
-
-  get answered()          {
-    return !!this.answerTime;
-  }
-
-  toJSON() {
-    const jsonObj = { ...this };
-    // $FlowFixMe
-    jsonObj.answered = this.answered;
-
-    return jsonObj;
-  }
-
-  static parseCall(call      )              {
-    return new CallSession({
-      callId: call.id,
-      sipCallId: call.sipCallId,
-      displayName: call.calleeName || call.calleeNumber,
-      number: call.calleeNumber,
-      callerNumber: call.callerNumber,
-      startTime: +call.startingTime,
-      paused: call.isOnHold(),
-      isCaller: call.isCaller,
-      muted: call.muted,
-      videoMuted: false,
-      screensharing: false,
-      recording: call.isRecording(),
-      recordingPaused: false, // @TODO
-      ringing: call.isRinging(),
-      answered: call.isUp(),
-      answeredBySystem: call.isUp() && call.talkingToIds.length === 0,
-      cameraEnabled: call.isVideo,
-      dialedExtension: call.dialedExtension,
-      call,
-      conference: false, // @FIXME?
+"use strict";
+var __awaiter = (this && this.__awaiter) || function (thisArg, _arguments, P, generator) {
+    function adopt(value) { return value instanceof P ? value : new P(function (resolve) { resolve(value); }); }
+    return new (P || (P = Promise))(function (resolve, reject) {
+        function fulfilled(value) { try { step(generator.next(value)); } catch (e) { reject(e); } }
+        function rejected(value) { try { step(generator["throw"](value)); } catch (e) { reject(e); } }
+        function step(result) { result.done ? resolve(result.value) : adopt(result.value).then(fulfilled, rejected); }
+        step((generator = generator.apply(thisArg, _arguments || [])).next());
     });
-  }
-}
-
-/*       */
-/* global navigator */
-const isMobile = () => typeof navigator !== 'undefined' && navigator.product === 'ReactNative';
-
-/*       */
-
-IssueReporter$1 ? IssueReporter$1.loggerFor('api') : console;
-
-// Use a function here to be able to mock it in tests
-const realFetch = () => {
-  if (typeof document !== 'undefined' && window.fetch) {
-    // Browser
-    return window.fetch;
-  }
-
-  if (isMobile()) {
-    // React native
-    return fetch;
-  }
-
-  // nodejs
-  // this package is disable for react-native in package.json because it requires nodejs modules
-  // Used to trick the optimizer to avoid requiring `node-fetch/lib/index` directly
-  // It causes to require it on browsers when delivered by a nodejs engine (cf: vitejs).
-  try {
-    // $FlowFixMe
-    return require(Math.random() >= 0 ? 'node-fetch/lib/index' : '');
-  } catch (e) {
-    return fetch;
-  }
 };
-
-/* global window */
-
-const TRACE = 'trace';
-const DEBUG = 'debug';
-const INFO = 'info';
-const LOG = 'log';
-const WARN = 'warn';
-const ERROR = 'error';
-const CONSOLE_METHODS = [INFO, LOG, WARN, ERROR];
-const LOG_LEVELS = [TRACE, DEBUG, INFO, LOG, WARN, ERROR];
-const CATEGORY_PREFIX = 'logger-category=';
-
-const MAX_REMOTE_RETRY = 100;
-
-const addLevelsTo = (instance        , withMethods = false) => {
-  instance.TRACE = TRACE;
-  instance.DEBUG = DEBUG;
-  instance.INFO = INFO;
-  instance.LOG = LOG;
-  instance.WARN = WARN;
-  instance.ERROR = ERROR;
-
-  if (withMethods) {
-    instance.trace = (...args) => instance.apply(null, [TRACE, ...args]);
-    instance.debug = (...args) => instance.apply(null, [DEBUG, ...args]);
-    instance.info = (...args) => instance.apply(null, [INFO, ...args]);
-    instance.log = (...args) => instance.apply(null, [LOG, ...args]);
-    instance.warn = (...args) => instance.apply(null, [WARN, ...args]);
-    instance.error = (...args) => instance.apply(null, [ERROR, ...args]);
-  }
-
-  return instance;
+var __importDefault = (this && this.__importDefault) || function (mod) {
+    return (mod && mod.__esModule) ? mod : { "default": mod };
 };
-
-const safeStringify = (object        ) => {
-  const result = '{"message": "Not parsable JSON"}';
-  try {
-    return JSON.stringify(object);
-  } catch (e) {
-    // Nothing to do
-  }
-
-  return result;
-};
-
-class IssueReporter {
-                
-               
-              
-               
-                
-
-                            
-                   
-                                     
-
-                   
-                            
-                                
-                                  
-                       
-
-  constructor() {
-    addLevelsTo(this);
-
-    this.oldConsoleMethods = null;
-    this.enabled = false;
-    this.remoteClientConfiguration = null;
-    this.buffer = [];
-    this.bufferTimeout = null;
-    this._boundProcessBuffer = this._processBuffer.bind(this);
-    this._boundParseLoggerBody = this._parseLoggerBody.bind(this);
-    this._callback = null;
-  }
-
-  init() {
-    this._catchConsole();
-  }
-
-  setCallback(cb          ) {
-    this._callback = cb;
-  }
-
-  configureRemoteClient(configuration         = { tag: 'wazo-sdk', host: null, port: null, level: null, extra: {} }) {
-    this.remoteClientConfiguration = configuration;
-  }
-
-  enable() {
-    if (!this.oldConsoleMethods) {
-      this.init();
-    }
-
-    this.enabled = true;
-  }
-
-  disable() {
-    this.enabled = false;
-  }
-
-  loggerFor(category        )      {
-    const logger = (level        , ...args     ) => {
-      this.log.apply(this, [level, this._makeCategory(category), ...args]);
-    };
-
-    return addLevelsTo(logger, true);
-  }
-
-  removeSlashes(str        ) {
-    return str.replace(/"/g, "'").replace(/\\/g, '');
-  }
-
-  log(level        , ...args     ) {
-    if (!this.enabled) {
-      return;
-    }
-
-    // Handle category label
-    let category = null;
-    let skipSendToRemote = false;
-    let extra = {};
-    if (args[0].indexOf(CATEGORY_PREFIX) === 0) {
-      category = args[0].split('=')[1];
-      // eslint-disable-next-line no-param-reassign
-      args.splice(0, 1);
-    }
-
-    // Handle extra data as object for the last argument
-    const lastArg = args[args.length - 1];
-    if (lastArg && ((typeof lastArg === 'object' && Object.keys(lastArg).length) || lastArg instanceof Error)) {
-      if (lastArg instanceof Error) {
-        extra = {
-          errorMessage: lastArg.message,
-          errorStack: lastArg.stack,
-          errorType: lastArg.constructor.name,
-          skipSendToRemote: lastArg.skipSendToRemote,
+Object.defineProperty(exports, "__esModule", { value: true });
+exports.events = exports.MESSAGE_TYPE_SIGNAL = exports.MESSAGE_TYPE_CHAT = exports.ON_EARLY_MEDIA = exports.ON_DISCONNECTED = exports.ON_SIGNAL = exports.ON_CHAT = exports.ON_NETWORK_STATS = exports.ON_MESSAGE_TRACK_UPDATED = exports.ON_CALL_ERROR = exports.ON_VIDEO_INPUT_CHANGE = exports.ON_PLAY_PROGRESS_SOUND = exports.ON_PLAY_HANGUP_SOUND = exports.ON_PLAY_INBOUND_CALL_SIGNAL_SOUND = exports.ON_PLAY_RING_SOUND = exports.ON_TERMINATE_SOUND = exports.ON_SHARE_SCREEN_ENDED = exports.ON_SHARE_SCREEN_ENDING = exports.ON_SHARE_SCREEN_STARTED = exports.ON_REMOVE_STREAM = exports.ON_VIDEO_STREAM = exports.ON_AUDIO_STREAM = exports.ON_TRACK = exports.ON_REINVITE = exports.ON_MESSAGE = exports.ON_CALL_ENDING = exports.ON_CALL_ENDED = exports.ON_CALL_REJECTED = exports.ON_CALL_FAILED = exports.ON_CALL_CANCELED = exports.ON_CAMERA_RESUMED = exports.ON_CAMERA_DISABLED = exports.ON_CALL_UNHELD = exports.ON_CALL_HELD = exports.ON_CALL_RESUMED = exports.ON_CALL_UNMUTED = exports.ON_CALL_MUTED = exports.ON_CALL_OUTGOING = exports.ON_CALL_INCOMING = exports.ON_CALL_ANSWERED = exports.ON_CALL_ACCEPTED = exports.ON_PROGRESS = exports.ON_UNREGISTERED = exports.ON_REGISTERED = exports.ON_USER_AGENT = void 0;
+const session_state_1 = require("sip.js/lib/api/session-state");
+const api_1 = require("sip.js/lib/api");
+const CallSession_1 = __importDefault(require("../CallSession"));
+const Emitter_1 = __importDefault(require("../../utils/Emitter"));
+const IssueReporter_1 = __importDefault(require("../../service/IssueReporter"));
+exports.ON_USER_AGENT = 'onUserAgent';
+exports.ON_REGISTERED = 'onRegistered';
+exports.ON_UNREGISTERED = 'onUnRegistered';
+exports.ON_PROGRESS = 'onProgress';
+exports.ON_CALL_ACCEPTED = 'onCallAccepted';
+exports.ON_CALL_ANSWERED = 'onCallAnswered';
+exports.ON_CALL_INCOMING = 'onCallIncoming';
+exports.ON_CALL_OUTGOING = 'onCallOutgoing';
+exports.ON_CALL_MUTED = 'onCallMuted';
+exports.ON_CALL_UNMUTED = 'onCallUnmuted';
+exports.ON_CALL_RESUMED = 'onCallResumed';
+exports.ON_CALL_HELD = 'onCallHeld';
+exports.ON_CALL_UNHELD = 'onCallUnHeld';
+exports.ON_CAMERA_DISABLED = 'onCameraDisabled';
+exports.ON_CAMERA_RESUMED = 'onCameraResumed';
+exports.ON_CALL_CANCELED = 'onCallCanceled';
+exports.ON_CALL_FAILED = 'onCallFailed';
+exports.ON_CALL_REJECTED = 'onCallRejected';
+exports.ON_CALL_ENDED = 'onCallEnded';
+exports.ON_CALL_ENDING = 'onCallEnding';
+exports.ON_MESSAGE = 'onMessage';
+exports.ON_REINVITE = 'reinvite';
+exports.ON_TRACK = 'onTrack';
+exports.ON_AUDIO_STREAM = 'onAudioStream';
+exports.ON_VIDEO_STREAM = 'onVideoStream';
+exports.ON_REMOVE_STREAM = 'onRemoveStream';
+exports.ON_SHARE_SCREEN_STARTED = 'onScreenShareStarted';
+exports.ON_SHARE_SCREEN_ENDING = 'onScreenShareEnding';
+exports.ON_SHARE_SCREEN_ENDED = 'onScreenShareEnded';
+exports.ON_TERMINATE_SOUND = 'terminateSound';
+exports.ON_PLAY_RING_SOUND = 'playRingingSound';
+exports.ON_PLAY_INBOUND_CALL_SIGNAL_SOUND = 'playInboundCallSignalSound';
+exports.ON_PLAY_HANGUP_SOUND = 'playHangupSound';
+exports.ON_PLAY_PROGRESS_SOUND = 'playProgressSound';
+exports.ON_VIDEO_INPUT_CHANGE = 'videoInputChange';
+exports.ON_CALL_ERROR = 'onCallError';
+exports.ON_MESSAGE_TRACK_UPDATED = 'onTrackUpdated';
+exports.ON_NETWORK_STATS = 'onNetworkStats';
+exports.ON_CHAT = 'phone/ON_CHAT';
+exports.ON_SIGNAL = 'phone/ON_SIGNAL';
+exports.ON_DISCONNECTED = 'onDisconnected';
+exports.ON_EARLY_MEDIA = 'onEarlyMedia';
+exports.MESSAGE_TYPE_CHAT = 'message/TYPE_CHAT';
+exports.MESSAGE_TYPE_SIGNAL = 'message/TYPE_SIGNAL';
+exports.events = [exports.ON_USER_AGENT, exports.ON_REGISTERED, exports.ON_UNREGISTERED, exports.ON_PROGRESS, exports.ON_CALL_ACCEPTED, exports.ON_CALL_ANSWERED, exports.ON_CALL_INCOMING, exports.ON_CALL_OUTGOING, exports.ON_CALL_MUTED, exports.ON_CALL_UNMUTED, exports.ON_CALL_RESUMED, exports.ON_CALL_HELD, exports.ON_CALL_UNHELD, exports.ON_CAMERA_DISABLED, exports.ON_CALL_FAILED, exports.ON_CALL_ENDED, exports.ON_CALL_REJECTED, exports.ON_MESSAGE, exports.ON_REINVITE, exports.ON_TRACK, exports.ON_AUDIO_STREAM, exports.ON_VIDEO_STREAM, exports.ON_REMOVE_STREAM, exports.ON_SHARE_SCREEN_ENDED, exports.ON_TERMINATE_SOUND, exports.ON_PLAY_RING_SOUND, exports.ON_PLAY_INBOUND_CALL_SIGNAL_SOUND, exports.ON_PLAY_HANGUP_SOUND, exports.ON_PLAY_PROGRESS_SOUND, exports.ON_VIDEO_INPUT_CHANGE, exports.ON_SHARE_SCREEN_STARTED, exports.ON_CALL_ERROR, exports.ON_CHAT, exports.ON_SIGNAL, exports.ON_NETWORK_STATS, exports.ON_DISCONNECTED, exports.ON_EARLY_MEDIA];
+const logger = IssueReporter_1.default.loggerFor('webrtc-phone');
+class WebRTCPhone extends Emitter_1.default {
+    constructor(client, audioOutputDeviceId, allowVideo = false, audioRingDeviceId) {
+        super();
+        this.client = client;
+        this.allowVideo = allowVideo;
+        this.callSessions = {};
+        this.audioOutputDeviceId = audioOutputDeviceId;
+        this.audioRingDeviceId = audioRingDeviceId || audioOutputDeviceId;
+        this.audioOutputVolume = 1;
+        this.audioRingVolume = 1;
+        this.incomingSessions = [];
+        this.ringingEnabled = true;
+        this.shouldSendReinvite = false;
+        this.bindClientEvents();
+        this.acceptedSessions = {};
+        this.rejectedSessions = {};
+        this.ignoredSessions = {};
+    }
+    register() {
+        if (!this.client) {
+            return Promise.resolve();
+        }
+        return this.client.register().then(() => {
+            return this.bindClientEvents();
+        }).catch(error => {
+            // Avoid exception on `t.server.scheme` in sip transport when losing the webrtc socket connection
+            console.error('register error', error, error.message, error.stack);
+            logger.error('WebRTC register error', {
+                message: error.message,
+                stack: error.stack,
+            });
+        });
+    }
+    unregister() {
+        if (!this.client || !this.client.isRegistered()) {
+            return null;
+        }
+        return this.client.unregister();
+    }
+    stop() {
+        if (!this.client) {
+            return;
+        }
+        logger.info('WebRTC phone stop');
+        this.client.stop();
+    }
+    removeIncomingSessions(id) {
+        this.incomingSessions = this.incomingSessions.filter(sessionId => sessionId !== id);
+    }
+    isWebRTC() {
+        return true;
+    }
+    sendReinvite(callSession, constraints = null, conference = false, audioOnly = false, iceRestart = false) {
+        return __awaiter(this, void 0, void 0, function* () {
+            const sipSession = this.findSipSession(callSession);
+            logger.info('WebRTC phone - send reinvite', {
+                sessionId: sipSession ? sipSession.id : null,
+                constraints,
+                audioOnly,
+                conference,
+                iceRestart,
+            });
+            if (!sipSession) {
+                return;
+            }
+            const shouldScreenShare = constraints === null || constraints === void 0 ? void 0 : constraints.screen;
+            const isUpgrade = shouldScreenShare || (constraints === null || constraints === void 0 ? void 0 : constraints.video);
+            // Downgrade
+            if (constraints && !isUpgrade) {
+                // No reinvite needed
+                this._downgradeToAudio(callSession);
+                return;
+            }
+            if (isUpgrade) {
+                const shouldReinvite = yield this._upgradeToVideo(callSession, constraints, conference);
+                if (!shouldReinvite) {
+                    return;
+                }
+            }
+            return this.client.reinvite(sipSession, constraints, conference, audioOnly, iceRestart);
+        });
+    }
+    getUserAgent() {
+        var _a, _b;
+        return ((_b = (_a = this.client) === null || _a === void 0 ? void 0 : _a.config) === null || _b === void 0 ? void 0 : _b.userAgentString) || 'webrtc-phone';
+    }
+    startHeartbeat() {
+        logger.info('WebRTC phone - start heartbeat', {
+            client: !!this.client,
+            hasHeartbeat: this.client.hasHeartbeat(),
+        });
+        if (!this.client || this.client.hasHeartbeat()) {
+            return;
+        }
+        this.client.startHeartbeat();
+    }
+    stopHeartbeat() {
+        logger.info('WebRTC phone - stopHeartbeat', {
+            client: !!this.client,
+        });
+        if (!this.client) {
+            return;
+        }
+        this.client.stopHeartbeat();
+    }
+    setOnHeartbeatTimeout(cb) {
+        this.client.setOnHeartbeatTimeout(cb);
+    }
+    setOnHeartbeatCallback(cb) {
+        this.client.setOnHeartbeatCallback(cb);
+    }
+    reconnect() {
+        logger.info('WebRTC phone - reconnect', {
+            client: !!this.client,
+        });
+        if (!this.client) {
+            return;
+        }
+        this.client.attemptReconnection();
+    }
+    getOptions() {
+        return {
+            accept: true,
+            decline: true,
+            mute: true,
+            hold: true,
+            transfer: true,
+            sendKey: true,
+            addParticipant: false,
+            record: true,
+            merge: true,
         };
-      } else {
-        extra = lastArg;
-      }
-      // eslint-disable-next-line no-param-reassign
-      args.splice(1, 1);
-    }
-
-    if (extra.skipSendToRemote) {
-      skipSendToRemote = true;
-      delete extra.skipSendToRemote;
-    }
-
-    const date = moment__default['default']().format('YYYY-MM-DD HH:mm:ss.SSS');
-    const message = args.join(', ');
-    let consoleMessage = message;
-
-    if (Object.keys(extra).length) {
-      const parsedExtra = safeStringify(extra);
-      consoleMessage = `${consoleMessage} (${parsedExtra})`;
-    }
-
-    if (category) {
-      consoleMessage = `[${category}] ${consoleMessage}`;
-    }
-
-    // Log the message in the console anyway (but don't console.error on mobile)
-    const consoleLevel = isMobile() && level === 'error' ? WARN : level;
-
-    // eslint-disable-next-line
-    const oldMethod = this.oldConsoleMethods[consoleLevel] || this.oldConsoleMethods.log;
-    oldMethod.apply(oldMethod, [date, consoleMessage]);
-
-    if (this._callback) {
-      this._callback(level, consoleMessage);
-    }
-
-    if (!skipSendToRemote) {
-      this._sendToRemoteLogger(level, { date, message, category, ...extra });
-    }
-  }
-
-  logRequest(url        , options        , response        , start      ) {
-    if (!this.enabled) {
-      return;
-    }
-    const { status } = response;
-    const duration = new Date() - start;
-
-    let level = TRACE;
-    if (status >= 400 && status < 500) {
-      level = WARN;
-    } else if (status >= 500) {
-      level = ERROR;
-    }
-
-    this.log(level, this._makeCategory('http'), url, {
-      status,
-      body: this.removeSlashes(JSON.stringify(options.body)),
-      method: options.method,
-      headers: options.headers,
-      duration,
-    });
-  }
-
-  // Logs aren't stored anymore
-  getLogs() {
-    console.warn('IssueReporter\'s logs aren\'t stored anymore. Please use fluentd to store them');
-    return [];
-  }
-
-  getParsedLogs() {
-    console.warn('IssueReporter\'s logs aren\'t stored anymore. Please use fluentd to store them');
-    return [];
-  }
-
-  getReport() {
-    return this.getParsedLogs().join('\r\n');
-  }
-
-  _catchConsole() {
-    this.oldConsoleMethods = {};
-    CONSOLE_METHODS.forEach((methodName        ) => {
-      // eslint-disable-next-line
-      this.oldConsoleMethods[methodName] = console[methodName];
-      window.console[methodName] = (...args) => {
-        // Store message
-        try {
-          this.log(methodName, args.join(' '));
-          // Use old console method to log it normally
-          this.oldConsoleMethods[methodName].apply(null, args);
-        } catch (e) {
-          // Avoid circular structure issues
+    }
+    onConnect() {
+        if (!this.client) {
+            return Promise.reject(new Error('No webrtc client'));
         }
-      };
-    });
-  }
-
-  _sendToRemoteLogger(level        , payload         = {}) {
-    if (!this.remoteClientConfiguration) {
-      return;
-    }
-
-    const { level: minLevel, bufferSize } = this.remoteClientConfiguration;
-    if (!minLevel || this._isLevelAbove(minLevel, level)) {
-      return;
-    }
-
-    payload.level = level;
-
-    if (bufferSize > 0) {
-      return this._addToBuffer(payload);
-    }
-
-    this._sendDebugToGrafana(payload);
-  }
-
-  _parseLoggerBody(payload        ) {
-    const { level } = payload;
-    const { maxMessageSize, extra } = this.remoteClientConfiguration || {};
-
-    delete payload.level;
-
-    if (maxMessageSize && typeof payload.message === 'string' && payload.message.length > maxMessageSize) {
-      payload.message = `${payload.message.substr(0, maxMessageSize)}...`;
-    }
-
-    return safeStringify({
-      level,
-      ...payload,
-      ...extra,
-    });
-  }
-
-  _addToBuffer(payload        ) {
-    // Reset buffer timer
-    if (this.bufferTimeout) {
-      clearTimeout(this.bufferTimeout);
-      this.bufferTimeout = null;
-    }
-
-    this.buffer.push(payload);
-
-    // $FlowFixMe
-    const { bufferSize, bufferTimeout } = this.remoteClientConfiguration;
-
-    if (this.buffer.length > bufferSize) {
-      return this._processBuffer();
-    }
-
-    if (bufferTimeout > 0) {
-      this.bufferTimeout = setTimeout(this._boundProcessBuffer, bufferTimeout);
-    }
-  }
-
-  _processBuffer() {
-    this._sendDebugToGrafana(this.buffer);
-
-    this.buffer = [];
-    if (this.bufferTimeout) {
-      clearTimeout(this.bufferTimeout);
-      this.bufferTimeout = null;
-    }
-  }
-
-  _sendDebugToGrafana(payload                   , retry         = 0) {
-    if (!this.remoteClientConfiguration || retry >= MAX_REMOTE_RETRY) {
-      return;
-    }
-
-    const { tag, host, port } = this.remoteClientConfiguration;
-    const isArray = Array.isArray(payload);
-    const isSecure = +port === 443;
-    const url = `http${isSecure ? 's' : ''}://${host}${isSecure ? '' : `:${port}`}/${tag}`;
-    const body = isArray ? `[${payload.map(this._boundParseLoggerBody).join(',')}]` : this._parseLoggerBody(payload);
-
-    realFetch()(url, {
-      method: 'POST',
-      headers: {
-        Accept: 'application/json',
-        'Content-Type': 'application/json',
-      },
-      // $FlowFixMe
-      body,
-    }).catch((e       ) => {
-      // $FlowFixMe
-      e.skipSendToRemote = true;
-      this.log('error', this._makeCategory('grafana'), 'Sending log to grafana, error', e);
-
-      setTimeout(() => {
-        if (isArray) {
-          payload = payload.map(message => this._writeRetryCount(message, retry + 1));
-        } else if (payload && typeof payload === 'object') {
-          payload = this._writeRetryCount(payload, retry + 1);
+        return this.client.onConnect();
+    }
+    onDisconnect() {
+        if (!this.client) {
+            return Promise.reject(new Error('No webrtc client'));
         }
-
-        this._sendDebugToGrafana(payload, retry + 1);
-      }, 5000 + retry * 1000);
-    });
-  }
-
-  _writeRetryCount(message                 , count        )                  {
-    if (message && typeof message === 'object') {
-      // $FlowFixMe
-      message._retry = count;
-    }
-
-    return message;
-  }
-
-  _isLevelAbove(level1        , level2        ) {
-    const index1 = LOG_LEVELS.indexOf(level1);
-    const index2 = LOG_LEVELS.indexOf(level2);
-    if (index1 === -1 || index2 === -1) {
-      return false;
-    }
-
-    return index1 > index2;
-  }
-
-  _makeCategory(category        ) {
-    return `${CATEGORY_PREFIX}${category}`;
-  }
-}
-
-var IssueReporter$1 = new IssueReporter();
-
-/* global RTCSessionDescriptionInit, navigator */
-
-IssueReporter$1 ? IssueReporter$1.loggerFor('webrtc-sdh') : console;
-
-//      
-
-class Emitter {
-                             
-
-  constructor() {
-    this.eventEmitter = new EventEmitter__default['default']();
-  }
-
-  on(event        , callback          ) {
-    this.eventEmitter.on(event, callback);
-  }
-
-  once(event        , callback          ) {
-    this.eventEmitter.once(event, callback);
-  }
-
-  off(event        , callback          ) {
-    this.eventEmitter.removeListener(event, callback);
-  }
-
-  unbind() {
-    this.eventEmitter.removeAllListeners();
-  }
-}
-
-//      
-
-const swarmPublicKey  = `-----BEGIN PUBLIC KEY-----
-MIIBIjANBgkqhkiG9w0BAQEFAAOCAQ8AMIIBCgKCAQEAmkXOuNfY8u5xoTiIhkb8
-djnbIwG/Wrz3vpo8BZir8L5e1a1nSy740qBjP7ZINBQoALDhFfmdOfJnCyEGiHuz
-ZW6jbG6C3PryE3Bu6GKwqSmD6k3q4Zk27fpwYAnNl+rWhYYM563rJZBda/INyHNN
-pK7M1mixWi7gNdjjXwoEXSSBx+VpYMkY6LiAB2mvHXTY9M1qI14dvgGoQISZQoKi
-NMTRCg5UP2ic0Dd9nSz/XpcOxGfa+0fwIl1F7RC1tJXOqvkGGPTOV4LLfg/Yta3h
-nUPX9EZZDIX6vO/0IBV1LzjSl2A1bYFYAjJfowv3i1CpvONBOClHjSY5t9Y8MH6p
-BwIDAQAB
------END PUBLIC KEY-----`;
-
-//      
-
-jsrsasign.KEYUTIL.getKey(swarmPublicKey);
-
-/* eslint-disable dot-notation */
-
-// Can't use node cache mechanism here because when requiring lib/CallApi.js
-// this file will be merge with CallApi.js and the cache will be lost.
-// So we have to store values in global scope.
-global.wazoClients = global.wazoClients || {};
-global.wazoClientId = global.wazoClientId || {};
-global.wazoClientToken = global.wazoClientToken || {};
-global.wazoRefreshToken = global.wazoRefreshToken || {};
-global.wazoRefreshTenantId = global.wazoRefreshTenantId || {};
-global.wazoRefreshDomainName = global.wazoRefreshDomainName || {};
-global.wazoOnRefreshToken = global.wazoOnRefreshToken || {};
-global.wazoOnRefreshTokenError = global.wazoOnRefreshTokenError || {};
-global.wazoRefreshExpiration = global.wazoRefreshExpiration || {};
-global.wazoRefreshBackend = global.wazoRefreshBackend || {};
-global.wazoIsMobile = global.wazoIsMobile || {};
-global.wazoFetchOptions = global.wazoFetchOptions || {};
-
-//      
-
-// const MINIMUM_WAZO_ENGINE_VERSION_FOR_CTI_HOLD = '20.11';
-
-IssueReporter$1 ? IssueReporter$1.loggerFor('cti-phone') : console;
-
-/*       */
-
-IssueReporter$1 ? IssueReporter$1.loggerFor('api') : console;
-
-//      
-IssueReporter$1 ? IssueReporter$1.loggerFor('webrtc-client') : console;
-IssueReporter$1 ? IssueReporter$1.loggerFor('webrtc-stats') : console;
-
-/* eslint-disable no-param-reassign */
-
-const ON_USER_AGENT = 'onUserAgent';
-const ON_REGISTERED = 'onRegistered';
-const ON_UNREGISTERED = 'onUnRegistered';
-const ON_PROGRESS = 'onProgress';
-const ON_CALL_ACCEPTED = 'onCallAccepted';
-const ON_CALL_ANSWERED = 'onCallAnswered';
-const ON_CALL_INCOMING = 'onCallIncoming';
-const ON_CALL_OUTGOING = 'onCallOutgoing';
-const ON_CALL_MUTED = 'onCallMuted';
-const ON_CALL_UNMUTED = 'onCallUnmuted';
-const ON_CALL_RESUMED = 'onCallResumed';
-const ON_CALL_HELD = 'onCallHeld';
-const ON_CALL_UNHELD = 'onCallUnHeld';
-const ON_CAMERA_DISABLED = 'onCameraDisabled';
-const ON_CAMERA_RESUMED = 'onCameraResumed';
-const ON_CALL_CANCELED = 'onCallCanceled';
-const ON_CALL_FAILED = 'onCallFailed';
-const ON_CALL_REJECTED = 'onCallRejected';
-const ON_CALL_ENDED = 'onCallEnded';
-const ON_CALL_ENDING = 'onCallEnding';
-const ON_MESSAGE = 'onMessage';
-const ON_REINVITE = 'reinvite';
-const ON_TRACK = 'onTrack';
-const ON_AUDIO_STREAM = 'onAudioStream';
-const ON_VIDEO_STREAM = 'onVideoStream';
-const ON_REMOVE_STREAM = 'onRemoveStream';
-const ON_SHARE_SCREEN_STARTED = 'onScreenShareStarted';
-const ON_SHARE_SCREEN_ENDING = 'onScreenShareEnding';
-const ON_SHARE_SCREEN_ENDED = 'onScreenShareEnded';
-const ON_TERMINATE_SOUND = 'terminateSound';
-const ON_PLAY_RING_SOUND = 'playRingingSound';
-const ON_PLAY_INBOUND_CALL_SIGNAL_SOUND = 'playInboundCallSignalSound';
-const ON_PLAY_HANGUP_SOUND = 'playHangupSound';
-const ON_PLAY_PROGRESS_SOUND = 'playProgressSound';
-const ON_VIDEO_INPUT_CHANGE = 'videoInputChange';
-const ON_CALL_ERROR = 'onCallError';
-const ON_MESSAGE_TRACK_UPDATED = 'onTrackUpdated';
-const ON_NETWORK_STATS = 'onNetworkStats';
-const ON_CHAT = 'phone/ON_CHAT';
-const ON_SIGNAL = 'phone/ON_SIGNAL';
-const ON_DISCONNECTED = 'onDisconnected';
-const ON_EARLY_MEDIA = 'onEarlyMedia';
-
-const MESSAGE_TYPE_CHAT = 'message/TYPE_CHAT';
-const MESSAGE_TYPE_SIGNAL = 'message/TYPE_SIGNAL';
-
-const events = [
-  ON_USER_AGENT,
-  ON_REGISTERED,
-  ON_UNREGISTERED,
-  ON_PROGRESS,
-  ON_CALL_ACCEPTED,
-  ON_CALL_ANSWERED,
-  ON_CALL_INCOMING,
-  ON_CALL_OUTGOING,
-  ON_CALL_MUTED,
-  ON_CALL_UNMUTED,
-  ON_CALL_RESUMED,
-  ON_CALL_HELD,
-  ON_CALL_UNHELD,
-  ON_CAMERA_DISABLED,
-  ON_CALL_FAILED,
-  ON_CALL_ENDED,
-  ON_CALL_REJECTED,
-  ON_MESSAGE,
-  ON_REINVITE,
-  ON_TRACK,
-  ON_AUDIO_STREAM,
-  ON_VIDEO_STREAM,
-  ON_REMOVE_STREAM,
-  ON_SHARE_SCREEN_ENDED,
-  ON_TERMINATE_SOUND,
-  ON_PLAY_RING_SOUND,
-  ON_PLAY_INBOUND_CALL_SIGNAL_SOUND,
-  ON_PLAY_HANGUP_SOUND,
-  ON_PLAY_PROGRESS_SOUND,
-  ON_VIDEO_INPUT_CHANGE,
-  ON_SHARE_SCREEN_STARTED,
-  ON_CALL_ERROR,
-  ON_CHAT,
-  ON_SIGNAL,
-  ON_NETWORK_STATS,
-  ON_DISCONNECTED,
-  ON_EARLY_MEDIA,
-];
-
-const logger = IssueReporter$1.loggerFor('webrtc-phone');
-
-class WebRTCPhone extends Emitter                  {
-                           
-
-                      
-
-                                          
-
-                             
-
-                             
-
-                                   
-
-                               
-
-                             
-
-                            
-
-                          
-
-                          
-
-                           
-
-                           
-
-                          
-
-                             
-
-                          
-
-                              
-
-  constructor(
-    client                  ,
-    audioOutputDeviceId         ,
-    allowVideo          = false,
-    audioRingDeviceId         ,
-  ) {
-    super();
-
-    this.client = client;
-    this.allowVideo = allowVideo;
-    this.callSessions = {};
-    this.audioOutputDeviceId = audioOutputDeviceId;
-    this.audioRingDeviceId = audioRingDeviceId || audioOutputDeviceId;
-    this.audioOutputVolume = 1;
-    this.audioRingVolume = 1;
-    this.incomingSessions = [];
-    this.ringingEnabled = true;
-    this.shouldSendReinvite = false;
-
-    this.bindClientEvents();
-
-    this.acceptedSessions = {};
-    this.rejectedSessions = {};
-    this.ignoredSessions = {};
-  }
-
-  register() {
-    if (!this.client) {
-      return Promise.resolve();
-    }
-
-    return this.client.register().then(() => {
-      return this.bindClientEvents();
-    }).catch(error => {
-      // Avoid exception on `t.server.scheme` in sip transport when losing the webrtc socket connection
-      console.error('register error', error, error.message, error.stack);
-      logger.error('WebRTC register error', { message: error.message, stack: error.stack });
-    });
-  }
-
-  unregister() {
-    if (!this.client || !this.client.isRegistered()) {
-      return null;
-    }
-    return this.client.unregister();
-  }
-
-  stop() {
-    if (!this.client) {
-      return;
-    }
-    logger.info('WebRTC phone stop');
-    this.client.stop();
-  }
-
-  removeIncomingSessions(id        ) {
-    this.incomingSessions = this.incomingSessions.filter(sessionId => sessionId !== id);
-  }
-
-  isWebRTC() {
-    return true;
-  }
-
-  async sendReinvite(callSession              , constraints          = null, conference          = false,
-    audioOnly          = false, iceRestart          = false) {
-    const sipSession = this.findSipSession(callSession);
-    logger.info('WebRTC phone - send reinvite', {
-      sessionId: sipSession ? sipSession.id : null,
-      constraints,
-      audioOnly,
-      conference,
-      iceRestart,
-    });
-
-    if (!sipSession) {
-      return false;
-    }
-
-    // $FlowFixMe
-    const shouldScreenShare = constraints && constraints.screen;
-    const isUpgrade = shouldScreenShare || (constraints && constraints.video);
-
-    // Downgrade
-    if (constraints && !isUpgrade) {
-      // No reinvite needed
-      this._downgradeToAudio(callSession);
-      return false;
-    }
-
-    if (isUpgrade) {
-      const shouldReinvite = await this._upgradeToVideo(callSession, constraints, conference);
-      if (!shouldReinvite) {
-        return false;
-      }
-    }
-
-    return this.client.reinvite(sipSession, constraints, conference, audioOnly, iceRestart);
-  }
-
-  getUserAgent() {
-    return (this.client && this.client.config && this.client.config.userAgentString) || 'webrtc-phone';
-  }
-
-  startHeartbeat() {
-    logger.info('WebRTC phone - start heartbeat', { client: !!this.client, hasHeartbeat: this.client.hasHeartbeat() });
-    if (!this.client || this.client.hasHeartbeat()) {
-      return;
-    }
-
-    this.client.startHeartbeat();
-  }
-
-  stopHeartbeat() {
-    logger.info('WebRTC phone - stopHeartbeat', { client: !!this.client });
-    if (!this.client) {
-      return;
-    }
-
-    this.client.stopHeartbeat();
-  }
-
-  setOnHeartbeatTimeout(cb          ) {
-    this.client.setOnHeartbeatTimeout(cb);
-  }
-
-  setOnHeartbeatCallback(cb          ) {
-    this.client.setOnHeartbeatCallback(cb);
-  }
-
-  reconnect() {
-    logger.info('WebRTC phone - reconnect', { client: !!this.client });
-    if (!this.client) {
-      return;
-    }
-    this.client.attemptReconnection();
-  }
-
-  getOptions()                        {
-    return {
-      accept: true,
-      decline: true,
-      mute: true,
-      hold: true,
-      transfer: true,
-      sendKey: true,
-      addParticipant: false,
-      record: true,
-      merge: true,
-    };
-  }
-
-  onConnect() {
-    if (!this.client) {
-      return Promise.reject(new Error('No webrtc client'));
-    }
-    return this.client.onConnect();
-  }
-
-  onDisconnect() {
-    if (!this.client) {
-      return Promise.reject(new Error('No webrtc client'));
-    }
-    return this.client.onDisconnect();
-  }
-
-  _downgradeToAudio(callSession              , withMessage          = true) {
-    const sipSession = this.findSipSession(callSession);
-    if (!sipSession) {
-      return false;
-    }
-
-    logger.info('Downgrade to video', { id: callSession ? callSession.getId() : null, withMessage });
-
-    this.client.downgradeToAudio(sipSession);
-
-    if (callSession) {
-      callSession.cameraEnabled = false;
-      this._updateCallSession(callSession);
-    }
-
-    if (withMessage) {
-      this._sendReinviteMessage(callSession, false);
-    }
-
-    return true;
-  }
-
-  // Returns true if we need to send a re-INVITE request
-  async _upgradeToVideo(callSession              , constraints         , isConference         )                   {
-    const sipSession = this.findSipSession(callSession);
-    if (!sipSession) {
-      return Promise.resolve(false);
-    }
-
-    logger.info('Upgrade to video', {
-      id: callSession ? callSession.getId() : null,
-      constraints,
-      isConference,
-    });
-
-    const shouldScreenShare = constraints && !!constraints.screen;
-    const desktop = constraints && constraints.desktop;
-    const options = sipSession.sessionDescriptionHandlerOptionsReInvite;
-    const wasAudioOnly = options && options.audioOnly;
-
-    const newStream = await this.client.upgradeToVideo(sipSession, constraints, isConference);
-
-    if (callSession) {
-      callSession.cameraEnabled = true;
-      this._updateCallSession(callSession);
-    }
-
-    // If no stream is returned, it means we have to reinvite
-    if (!newStream) {
-      return true;
-    }
-
-    this._sendReinviteMessage(callSession, true);
-
-    if (shouldScreenShare) {
-      this._onScreenSharing(newStream, sipSession, callSession, false, desktop);
-    }
-
-    // We have to reinvite to change the direction on the bundle when upgrading from an audioOnly conference
-    if (isConference && wasAudioOnly) {
-      return true;
-    }
-
-    // No reinvite needed here
-    return false;
-  }
-
-  _sendReinviteMessage(callSession              , isUpgrade         ) {
-    const sipSession = this.findSipSession(callSession);
-    logger.info('Sending reinvite message', {
-      id: callSession ? callSession.getId() : null,
-      isUpgrade,
-    });
-
-    // Have to send the message after a delay due to latency to update the remote peer
-    setTimeout(() => {
-      this.sendMessage(sipSession, JSON.stringify({
-        type: MESSAGE_TYPE_SIGNAL,
-        content: {
-          type: ON_MESSAGE_TRACK_UPDATED,
-          update: isUpgrade ? 'upgrade' : 'downgrade',
-          sipCallId: this.getSipSessionId(sipSession),
-          callId: callSession ? callSession.callId : null,
-          number: callSession ? callSession.number : null,
-          callerNumber: callSession ? callSession.callerNumber : null,
-        },
-      }));
-    }, 2500);
-  }
-
-  _bindEvents(sipSession         ) {
-    if (sipSession._onCancel) {
-      // Monkey patch to know when canceled with the CANCEL message
-      const onCancel = sipSession._onCancel.bind(sipSession);
-      sipSession._onCancel = (message                        ) => {
-        logger.trace('on sip session canceled', { callId: message.callId });
-
-        onCancel(message);
-        const elsewhere = message.data.indexOf('cause=26') !== -1 && message.data.indexOf('completed elsewhere') !== -1;
+        return this.client.onDisconnect();
+    }
+    _downgradeToAudio(callSession, withMessage = true) {
+        const sipSession = this.findSipSession(callSession);
+        if (!sipSession) {
+            return false;
+        }
+        logger.info('Downgrade to video', {
+            id: callSession ? callSession.getId() : null,
+            withMessage,
+        });
+        this.client.downgradeToAudio(sipSession);
+        if (callSession) {
+            callSession.cameraEnabled = false;
+            this._updateCallSession(callSession);
+        }
+        if (withMessage) {
+            this._sendReinviteMessage(callSession, false);
+        }
+        return true;
+    }
+    // Returns true if we need to send a re-INVITE request
+    _upgradeToVideo(callSession, constraints, isConference) {
+        return __awaiter(this, void 0, void 0, function* () {
+            const sipSession = this.findSipSession(callSession);
+            if (!sipSession) {
+                return Promise.resolve(false);
+            }
+            logger.info('Upgrade to video', {
+                id: callSession ? callSession.getId() : null,
+                constraints,
+                isConference,
+            });
+            const shouldScreenShare = constraints && !!constraints.screen;
+            const desktop = constraints && constraints.desktop;
+            const options = sipSession.sessionDescriptionHandlerOptionsReInvite;
+            // @ts-ignore
+            const wasAudioOnly = options === null || options === void 0 ? void 0 : options.audioOnly;
+            const newStream = yield this.client.upgradeToVideo(sipSession, constraints, isConference);
+            if (callSession) {
+                callSession.cameraEnabled = true;
+                this._updateCallSession(callSession);
+            }
+            // If no stream is returned, it means we have to reinvite
+            if (!newStream) {
+                return true;
+            }
+            this._sendReinviteMessage(callSession, true);
+            if (shouldScreenShare) {
+                this._onScreenSharing(newStream, sipSession, callSession, false, desktop);
+            }
+            // We have to reinvite to change the direction on the bundle when upgrading from an audioOnly conference
+            if (isConference && wasAudioOnly) {
+                return true;
+            }
+            // No reinvite needed here
+            return false;
+        });
+    }
+    _sendReinviteMessage(callSession, isUpgrade) {
+        const sipSession = this.findSipSession(callSession);
+        logger.info('Sending reinvite message', {
+            id: callSession ? callSession.getId() : null,
+            isUpgrade,
+        });
+        // Have to send the message after a delay due to latency to update the remote peer
+        setTimeout(() => {
+            this.sendMessage(sipSession, JSON.stringify({
+                type: exports.MESSAGE_TYPE_SIGNAL,
+                content: {
+                    type: exports.ON_MESSAGE_TRACK_UPDATED,
+                    update: isUpgrade ? 'upgrade' : 'downgrade',
+                    sipCallId: sipSession && this.getSipSessionId(sipSession),
+                    callId: callSession ? callSession.callId : null,
+                    number: callSession ? callSession.number : null,
+                    callerNumber: callSession ? callSession.callerNumber : null,
+                },
+            }));
+        }, 2500);
+    }
+    _bindEvents(sipSession) {
+        if (sipSession instanceof api_1.Invitation) {
+            // Monkey patch to know when canceled with the CANCEL message
+            const onCancel = sipSession._onCancel.bind(sipSession);
+            sipSession._onCancel = (message) => {
+                logger.trace('on sip session canceled', {
+                    callId: message.callId,
+                });
+                onCancel(message);
+                const elsewhere = message.data.indexOf('cause=26') !== -1 && message.data.indexOf('completed elsewhere') !== -1;
+                const callSession = this._createCallSession(sipSession);
+                delete this.callSessions[callSession.getId()];
+                this.eventEmitter.emit(exports.ON_CALL_CANCELED, callSession, elsewhere);
+            };
+        }
+        else if (sipSession instanceof api_1.Invitation) {
+            console.warn('sipSession._onCancel not found, please update the wazo SDK accordingly');
+        }
+        sipSession.stateChange.addListener((newState) => {
+            switch (newState) {
+                case session_state_1.SessionState.Establishing:
+                    if (sipSession instanceof api_1.Invitation) {
+                        // No need to trigger progress for an invitation (eg: when we answer the call).
+                        return;
+                    }
+                    // When receiving a progress event, we know we are the caller so we have to force incoming to false
+                    return this.eventEmitter.emit(exports.ON_PROGRESS, this._createCallSession(sipSession, null, {
+                        incoming: false,
+                        ringing: true,
+                    }), this.audioOutputDeviceId, this.audioOutputVolume);
+                case session_state_1.SessionState.Terminating:
+                    logger.info('WebRTC phone - call terminating', {
+                        sipId: sipSession.id,
+                    });
+                    this.eventEmitter.emit(exports.ON_CALL_ENDING, this._createCallSession(sipSession));
+                    break;
+                case session_state_1.SessionState.Terminated:
+                    {
+                        logger.info('WebRTC phone - call terminated', {
+                            sipId: sipSession.id,
+                        });
+                        // Should be called before `_onCallTerminated` or the callCount will not decrement...
+                        const callSession = this._createCallSession(sipSession);
+                        callSession.endTime = new Date();
+                        const wasCurrentSession = this._onCallTerminated(sipSession);
+                        return this.eventEmitter.emit(exports.ON_CALL_ENDED, callSession, wasCurrentSession);
+                    }
+                default:
+                    break;
+            }
+        });
+        if (!sipSession.sessionDescriptionHandler) {
+            return;
+        }
+        // Video events
+        const { 
+        // @ts-ignore
+        peerConnection, } = sipSession.sessionDescriptionHandler;
+        peerConnection.ontrack = (rawEvent) => {
+            const event = rawEvent;
+            const [stream] = event.streams;
+            const kind = event && event.track && event.track.kind;
+            logger.info('on track stream called on the peer connection', {
+                callId: this.getSipSessionId(sipSession),
+                streamId: stream ? stream.id : null,
+                tracks: stream ? stream.getTracks() : null,
+                kind,
+            });
+            if (kind === 'audio') {
+                return this.eventEmitter.emit(exports.ON_AUDIO_STREAM, stream);
+            }
+            // not sure this does anything
+            if (kind === 'video') {
+                event.track.enabled = false;
+            }
+            return this.eventEmitter.emit(exports.ON_VIDEO_STREAM, stream, event.track.id, event, sipSession);
+        };
+        peerConnection.onremovestream = (event) => {
+            logger.info('on remove stream called on the peer connection', {
+                id: event.stream.id,
+                tracks: event.stream.getTracks(),
+            });
+            this.eventEmitter.emit(exports.ON_REMOVE_STREAM, event.stream);
+        };
+    }
+    startScreenSharing(constraints, callSession) {
+        return __awaiter(this, void 0, void 0, function* () {
+            logger.info('WebRTC - start screen sharing', {
+                constraints,
+                id: callSession ? callSession.getId() : null,
+            });
+            const screenShareStream = yield this.client.getStreamFromConstraints(constraints);
+            if (!screenShareStream) {
+                throw new Error(`Can't create media stream for screensharing with: ${JSON.stringify(constraints)}`);
+            }
+            const screenTrack = screenShareStream.getVideoTracks()[0];
+            const sipSession = this.currentSipSession;
+            const pc = sipSession && this.client.getPeerConnection(this.getSipSessionId(sipSession));
+            const sender = pc && pc.getSenders && pc.getSenders().find((s) => s && s.track && s.track.kind === 'video');
+            const localStream = sipSession && this.client.getLocalStream(this.getSipSessionId(sipSession));
+            const videoTrack = localStream ? localStream.getTracks().find(track => track.kind === 'video') : null;
+            const hadVideo = !!videoTrack;
+            // Stop local video tracks
+            if (videoTrack && localStream) {
+                videoTrack.enabled = false;
+                videoTrack.stop();
+                localStream.removeTrack(videoTrack);
+            }
+            if (localStream) {
+                localStream.addTrack(screenTrack);
+            }
+            if (sender) {
+                sender.replaceTrack(screenTrack);
+            }
+            this._onScreenSharing(screenShareStream, sipSession, callSession, hadVideo, constraints.desktop);
+            return screenShareStream;
+        });
+    }
+    stopScreenSharing(restoreLocalStream = true, callSession) {
+        return __awaiter(this, void 0, void 0, function* () {
+            if (!this.currentScreenShare) {
+                return;
+            }
+            logger.info('WebRTC phone - stop screen sharing', {
+                restoreLocalStream,
+            });
+            let reinvited;
+            try {
+                if (this.currentScreenShare.stream) {
+                    this.currentScreenShare.stream.getTracks().forEach((track) => track.stop());
+                }
+                if (restoreLocalStream) {
+                    const targetCallSession = callSession || this.currentCallSession;
+                    const conference = this.isConference(targetCallSession);
+                    // When stopping screenshare and we had video before that, we have to re-upgrade
+                    // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
+                    // We have to downgrade to audio.
+                    const screenshareStopped = this.currentScreenShare.hadVideo;
+                    const constraints = {
+                        audio: false,
+                        video: screenshareStopped,
+                    };
+                    // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
+                    this._downgradeToAudio(targetCallSession, false);
+                    // Wait for the track to be removed, unless the video sender won't have a null track in `_upgradeToVideo`.
+                    yield new Promise(resolve => setTimeout(resolve, 500));
+                    reinvited = yield this.sendReinvite(targetCallSession, constraints, conference);
+                }
+            }
+            catch (e) {
+                console.warn(e);
+            }
+            const sipSession = this.currentSipSession;
+            this.eventEmitter.emit(exports.ON_SHARE_SCREEN_ENDED, callSession && sipSession ? this._createCallSession(sipSession, callSession, {
+                screensharing: false,
+            }) : null);
+            this.currentScreenShare = null;
+            return reinvited;
+        });
+    }
+    _onScreenSharing(screenStream, sipSession, callSession, hadVideo, desktop) {
+        const screenTrack = screenStream.getVideoTracks()[0];
+        const sipSessionId = this.getSipSessionId(sipSession);
+        const pc = this.client.getPeerConnection(sipSessionId);
+        const sender = pc && pc.getSenders && pc.getSenders().find((s) => s && s.track && s.track.kind === 'video');
+        logger.info('WebRTC phone - on screensharing', {
+            hadVideo,
+            id: this.getSipSessionId(sipSession),
+            screenTrack,
+        });
+        if (screenTrack) {
+            screenTrack.onended = () => {
+                logger.info('WebRTC phone - on screenshare ended', {
+                    hadVideo,
+                    id: this.getSipSessionId(sipSession),
+                    screenTrack,
+                });
+                this.eventEmitter.emit(exports.ON_SHARE_SCREEN_ENDING, this._createCallSession(sipSession, callSession));
+            };
+        }
+        this.currentScreenShare = {
+            stream: screenStream,
+            hadVideo,
+            sender,
+            sipSessionId,
+            desktop,
+        };
+        this.client.setLocalMediaStream(this.getSipSessionId(sipSession), screenStream);
+        this.eventEmitter.emit(exports.ON_SHARE_SCREEN_STARTED, this._createCallSession(sipSession, callSession, {
+            screensharing: true,
+        }), screenStream);
+    }
+    _onCallAccepted(sipSession, cameraEnabled) {
+        logger.info('WebRTC phone - on call accepted', {
+            sipId: sipSession.id,
+            cameraEnabled,
+        });
+        const callSession = this._createAcceptedCallSession(sipSession, cameraEnabled);
+        this.currentSipSession = sipSession;
+        this.currentCallSession = callSession;
+        this.eventEmitter.emit(exports.ON_TERMINATE_SOUND, callSession, 'call accepted');
+        const sipSessionId = this.getSipSessionId(sipSession);
+        if (sipSessionId) {
+            this.removeIncomingSessions(sipSessionId);
+        }
+        callSession.answerTime = new Date();
+        this._updateCallSession(callSession);
+        this.eventEmitter.emit(exports.ON_CALL_ACCEPTED, callSession, cameraEnabled);
+        return callSession;
+    }
+    changeAudioDevice(id) {
+        logger.info('WebRTC phone - change audio device', {
+            deviceId: id,
+        });
+        this.audioOutputDeviceId = id;
+        this.client.changeAudioOutputDevice(id);
+    }
+    createAudioElementFor(sessionId) {
+        return this.client.createAudioElementFor(sessionId);
+    }
+    changeRingDevice(id) {
+        logger.info('WebRTC phone - changing ring device', {
+            id,
+        });
+        this.audioRingDeviceId = id;
+    }
+    // volume is a value between 0 and 1
+    changeAudioVolume(volume) {
+        logger.info('WebRTC phone - changing audio volume', {
+            volume,
+        });
+        this.audioOutputVolume = volume;
+        this.client.changeAudioOutputVolume(volume);
+    }
+    // volume is a value between 0 and 1
+    changeRingVolume(volume) {
+        logger.info('WebRTC phone - changing ring volume', {
+            volume,
+        });
+        this.audioRingVolume = volume;
+    }
+    changeAudioInputDevice(id, force) {
+        logger.info('WebRTC phone - changeAudio input device', {
+            deviceId: id,
+        });
+        return this.client.changeAudioInputDevice(id, this.currentSipSession, force);
+    }
+    changeVideoInputDevice(id) {
+        return __awaiter(this, void 0, void 0, function* () {
+            logger.info('WebRTC phone - change video input device', {
+                deviceId: id,
+            });
+            return this.client.changeVideoInputDevice(id, this.currentSipSession);
+        });
+    }
+    changeSessionVideoInputDevice(id) {
+        logger.info('WebRTC phone - change session video input device', {
+            deviceId: id,
+        });
+        return this.currentSipSession && this.client.changeSessionVideoInputDevice(id, this.currentSipSession);
+    }
+    _onCallTerminated(sipSession) {
+        logger.info('WebRTC phone - on call terminated', {
+            sipId: sipSession.id,
+        });
+        const sipSessionId = this.getSipSessionId(sipSession);
         const callSession = this._createCallSession(sipSession);
+        const isCurrentSession = this.isCurrentCallSipSession(callSession);
+        const isCurrentIncomingCall = callSession.is(this.getIncomingCallSession());
+        // If the terminated call was an incoming call, we have to re-trigger if it's was the first incoming call
+        // Otherwise, we have to re-trigger an incoming call event if another call is incoming
+        const shouldRetrigger = isCurrentSession ? this.incomingSessions.length : isCurrentIncomingCall;
+        // Terminate incoming sound only for the ringing call
+        // (so another terminated call won't stop the incoming call sound)
+        if (isCurrentIncomingCall) {
+            setTimeout(() => {
+                // Avoid race condition when the other is calling and hanging up immediately
+                this.eventEmitter.emit(exports.ON_TERMINATE_SOUND, callSession, 'call terminated');
+            }, 5);
+        }
+        if (sipSessionId) {
+            this.removeIncomingSessions(sipSessionId);
+        }
         delete this.callSessions[callSession.getId()];
-
-        this.eventEmitter.emit(ON_CALL_CANCELED, callSession, elsewhere);
-      };
-    } else if (sipSession instanceof invitation.Invitation) {
-      console.warn('sipSession._onCancel not found, please update the wazo SDK accordingly');
-    }
-
-    sipSession.stateChange.addListener((newState              ) => {
-      switch (newState) {
-        case sessionState.SessionState.Establishing:
-          if (sipSession instanceof invitation.Invitation) {
-            // No need to trigger progress for an invitation (eg: when we answer the call).
+        if (isCurrentSession) {
+            this.currentSipSession = undefined;
+            this.currentCallSession = undefined;
+        }
+        const hasIncomingCallSession = this.hasIncomingCallSession();
+        // Re-trigger incoming call event for remaining incoming calls
+        logger.info('WebRTC phone - check to re-trigger incoming call', {
+            sipId: sipSession.id,
+            shouldRetrigger,
+            hasIncomingCallSession,
+        });
+        this.client.onCallEnded(sipSession);
+        if (hasIncomingCallSession && shouldRetrigger) {
+            const nextCallSession = this.getIncomingCallSession();
+            // Avoid race condition
+            setTimeout(() => {
+                if (this.ringingEnabled) {
+                    if (!this.currentCallSession) {
+                        this.eventEmitter.emit(exports.ON_PLAY_RING_SOUND, this.audioRingDeviceId, this.audioRingVolume, nextCallSession);
+                    }
+                    else {
+                        this.eventEmitter.emit(exports.ON_PLAY_INBOUND_CALL_SIGNAL_SOUND, this.audioOutputDeviceId, this.audioOutputVolume, nextCallSession);
+                    }
+                }
+                this.eventEmitter.emit(exports.ON_CALL_INCOMING, nextCallSession, nextCallSession === null || nextCallSession === void 0 ? void 0 : nextCallSession.cameraEnabled);
+            }, 100);
+        }
+        if (callSession.getId() in this.ignoredSessions) {
+            return false;
+        }
+        // @ts-ignore
+        if (!sipSession.isCanceled) {
+            setTimeout(() => {
+                this.eventEmitter.emit(exports.ON_PLAY_HANGUP_SOUND, this.audioOutputDeviceId, this.audioOutputVolume, callSession);
+            }, 10);
+        }
+        return isCurrentSession;
+    }
+    setActiveSipSession(callSession) {
+        const sipSessionId = this.findSipSession(callSession);
+        if (!sipSessionId) {
             return;
-          }
-          // When receiving a progress event, we know we are the caller so we have to force incoming to false
-          return this.eventEmitter.emit(
-            ON_PROGRESS,
-            this._createCallSession(sipSession, null, { incoming: false, ringing: true }),
-            this.audioOutputDeviceId,
-            this.audioOutputVolume,
-          );
-        case sessionState.SessionState.Terminating:
-          logger.info('WebRTC phone - call terminating', { sipId: sipSession.id });
-
-          this.eventEmitter.emit(ON_CALL_ENDING, this._createCallSession(sipSession));
-
-          break;
-        case sessionState.SessionState.Terminated: {
-          logger.info('WebRTC phone - call terminated', { sipId: sipSession.id });
-
-          // Should be called before `_onCallTerminated` or the callCount will not decrement...
-          const callSession = this._createCallSession(sipSession);
-          callSession.endTime = new Date();
-          const wasCurrentSession = this._onCallTerminated(sipSession);
-
-          return this.eventEmitter.emit(ON_CALL_ENDED, callSession, wasCurrentSession);
         }
-      }
-    });
-
-    if (!sipSession.sessionDescriptionHandler) {
-      return;
-    }
-
-    // Video events
-    const { peerConnection } = sipSession.sessionDescriptionHandler;
-    peerConnection.ontrack = rawEvent => {
-      const event = rawEvent;
-      const [stream] = event.streams;
-      const kind = event && event.track && event.track.kind;
-
-      logger.info('on track stream called on the peer connection', {
-        callId: this.getSipSessionId(sipSession),
-        streamId: stream ? stream.id : null,
-        tracks: stream ? stream.getTracks() : null,
-        kind,
-      });
-
-      if (kind === 'audio') {
-        return this.eventEmitter.emit(ON_AUDIO_STREAM, stream);
-      }
-
-      // not sure this does anything
-      if (kind === 'video') {
-        event.track.enabled = false;
-      }
-
-      return this.eventEmitter.emit(ON_VIDEO_STREAM, stream, event.track.id, event, sipSession);
-    };
-
-    peerConnection.onremovestream = event => {
-      logger.info('on remove stream called on the peer connection', {
-        id: event.stream.id,
-        tracks: event.stream.getTracks(),
-      });
-
-      this.eventEmitter.emit(ON_REMOVE_STREAM, event.stream);
-    };
-  }
-
-  async startScreenSharing(constraints        , callSession              ) {
-    logger.info('WebRTC - start screen sharing', { constraints, id: callSession ? callSession.getId() : null });
-
-    const screenShareStream = await this.client.getStreamFromConstraints(constraints);
-    if (!screenShareStream) {
-      throw new Error(`Can't create media stream for screensharing with: ${JSON.stringify(constraints)}`);
-    }
-
-    const screenTrack = screenShareStream.getVideoTracks()[0];
-    const sipSession = this.currentSipSession;
-    const pc = this.client.getPeerConnection(this.getSipSessionId(sipSession));
-    const sender = pc && pc.getSenders && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
-    const localStream = this.client.getLocalStream(this.getSipSessionId(sipSession));
-    const videoTrack = localStream ? localStream.getTracks().find(track => track.kind === 'video') : null;
-    const hadVideo = !!videoTrack;
-
-    // Stop local video tracks
-    if (videoTrack && localStream) {
-      videoTrack.enabled = false;
-      videoTrack.stop();
-      localStream.removeTrack(videoTrack);
-    }
-
-    if (localStream) {
-      localStream.addTrack(screenTrack);
-    }
-
-    if (sender) {
-      sender.replaceTrack(screenTrack);
-    }
-
-    this._onScreenSharing(screenShareStream, sipSession, callSession, hadVideo, constraints.desktop);
-
-    return screenShareStream;
-  }
-
-  async stopScreenSharing(restoreLocalStream          = true, callSession               ) {
-    if (!this.currentScreenShare) {
-      return;
-    }
-
-    logger.info('WebRTC phone - stop screen sharing', {
-      restoreLocalStream,
-    });
-
-    let reinvited = false;
-
-    try {
-      if (this.currentScreenShare.stream) {
-        this.currentScreenShare.stream.getTracks().forEach(track => track.stop());
-      }
-
-      if (restoreLocalStream) {
-        const targetCallSession = callSession || this.currentCallSession;
-        const conference = this.isConference(targetCallSession);
-
-        // When stopping screenshare and we had video before that, we have to re-upgrade
-        // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
-        // We have to downgrade to audio.
-        const screenshareStopped = this.currentScreenShare.hadVideo;
-        const constraints = { audio: false, video: screenshareStopped };
-
-        // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
-        this._downgradeToAudio(targetCallSession, false);
-        // Wait for the track to be removed, unless the video sender won't have a null track in `_upgradeToVideo`.
-        await new Promise(resolve => setTimeout(resolve, 500));
-
-        reinvited = await this.sendReinvite(targetCallSession, constraints, conference);
-      }
-    } catch (e) {
-      console.warn(e);
-    }
-
-    const sipSession = this.currentSipSession;
-
-    this.eventEmitter.emit(
-      ON_SHARE_SCREEN_ENDED,
-      callSession ? this._createCallSession(sipSession, callSession, { screensharing: false }) : null,
-    );
-
-    this.currentScreenShare = null;
-
-    return reinvited;
-  }
-
-  _onScreenSharing(screenStream        ,
-    sipSession         ,
-    callSession              ,
-    hadVideo         ,
-    desktop            = false) {
-    const screenTrack = screenStream.getVideoTracks()[0];
-    const sipSessionId = this.getSipSessionId(sipSession);
-    const pc = this.client.getPeerConnection(sipSessionId);
-    const sender = pc && pc.getSenders && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
-
-    logger.info('WebRTC phone - on screensharing', {
-      hadVideo,
-      id: this.getSipSessionId(sipSession),
-      screenTrack,
-    });
-
-    if (screenTrack) {
-      screenTrack.onended = () => {
-        logger.info('WebRTC phone - on screenshare ended', {
-          hadVideo,
-          id: this.getSipSessionId(sipSession),
-          screenTrack,
-        });
-
-        this.eventEmitter.emit(
-          ON_SHARE_SCREEN_ENDING,
-          this._createCallSession(sipSession, callSession),
-        );
-      };
-    }
-
-    this.currentScreenShare = {
-      stream: screenStream,
-      hadVideo,
-      sender,
-      sipSessionId,
-      desktop,
-    };
-
-    this.client.setLocalMediaStream(this.getSipSessionId(sipSession), screenStream);
-
-    this.eventEmitter.emit(
-      ON_SHARE_SCREEN_STARTED,
-      this._createCallSession(sipSession, callSession, { screensharing: true }),
-      screenStream,
-    );
-  }
-
-  _onCallAccepted(sipSession         , cameraEnabled         )              {
-    logger.info('WebRTC phone - on call accepted', { sipId: sipSession.id, cameraEnabled });
-
-    const callSession = this._createAcceptedCallSession(sipSession, cameraEnabled);
-    this.currentSipSession = sipSession;
-    this.currentCallSession = callSession;
-
-    this.eventEmitter.emit(ON_TERMINATE_SOUND, callSession, 'call accepted');
-    const sipSessionId = this.getSipSessionId(sipSession);
-    if (sipSessionId) {
-      this.removeIncomingSessions(sipSessionId);
-    }
-
-    callSession.answerTime = new Date();
-    this._updateCallSession(callSession);
-    this.eventEmitter.emit(ON_CALL_ACCEPTED, callSession, cameraEnabled);
-
-    return callSession;
-  }
-
-  changeAudioDevice(id        ) {
-    logger.info('WebRTC phone - change audio device', { deviceId: id });
-    this.audioOutputDeviceId = id;
-    this.client.changeAudioOutputDevice(id);
-  }
-
-  createAudioElementFor(sessionId        ) {
-    return this.client.createAudioElementFor(sessionId);
-  }
-
-  changeRingDevice(id        ) {
-    logger.info('WebRTC phone - changing ring device', { id });
-
-    this.audioRingDeviceId = id;
-  }
-
-  // volume is a value between 0 and 1
-  changeAudioVolume(volume        ) {
-    logger.info('WebRTC phone - changing audio volume', { volume });
-
-    this.audioOutputVolume = volume;
-    this.client.changeAudioOutputVolume(volume);
-  }
-
-  // volume is a value between 0 and 1
-  changeRingVolume(volume        ) {
-    logger.info('WebRTC phone - changing ring volume', { volume });
-
-    this.audioRingVolume = volume;
-  }
-
-  changeAudioInputDevice(id        , force          ) {
-    logger.info('WebRTC phone - changeAudio input device', { deviceId: id });
-
-    return this.client.changeAudioInputDevice(id, this.currentSipSession, force);
-  }
-
-  changeVideoInputDevice(id         ) {
-    logger.info('WebRTC phone - change video input device', { deviceId: id });
-
-    return this.client.changeVideoInputDevice(id, this.currentSipSession);
-  }
-
-  changeSessionVideoInputDevice(id         ) {
-    logger.info('WebRTC phone - change session video input device', { deviceId: id });
-
-    return this.client.changeSessionVideoInputDevice(id, this.currentSipSession);
-  }
-
-  _onCallTerminated(sipSession         ) {
-    logger.info('WebRTC phone - on call terminated', { sipId: sipSession.id });
-
-    const sipSessionId = this.getSipSessionId(sipSession);
-    const callSession = this._createCallSession(sipSession);
-    const isCurrentSession = this.isCurrentCallSipSession(callSession);
-    const isCurrentIncomingCall = callSession.is(this.getIncomingCallSession());
-
-    // If the terminated call was an incoming call, we have to re-trigger if it's was the first incoming call
-    // Otherwise, we have to re-trigger an incoming call event if another call is incoming
-    const shouldRetrigger = isCurrentSession ? this.incomingSessions.length : isCurrentIncomingCall;
-
-    // Terminate incoming sound only for the ringing call
-    // (so another terminated call won't stop the incoming call sound)
-    if (isCurrentIncomingCall) {
-      setTimeout(() => {
-        // Avoid race condition when the other is calling and hanging up immediately
-        this.eventEmitter.emit(ON_TERMINATE_SOUND, callSession, 'call terminated');
-      }, 5);
-    }
-
-    if (sipSessionId) {
-      this.removeIncomingSessions(sipSessionId);
-    }
-
-    delete this.callSessions[callSession.getId()];
-
-    if (isCurrentSession) {
-      this.currentSipSession = undefined;
-      this.currentCallSession = undefined;
-    }
-
-    const hasIncomingCallSession = this.hasIncomingCallSession();
-
-    // Re-trigger incoming call event for remaining incoming calls
-    logger.info('WebRTC phone - check to re-trigger incoming call', {
-      sipId: sipSession.id,
-      shouldRetrigger,
-      hasIncomingCallSession,
-    });
-
-    this.client.onCallEnded(sipSession);
-
-    if (hasIncomingCallSession && shouldRetrigger) {
-      const nextCallSession = this.getIncomingCallSession();
-      // Avoid race condition
-      setTimeout(() => {
-        if (this.ringingEnabled) {
-          if (!this.currentCallSession) {
-            this.eventEmitter.emit(ON_PLAY_RING_SOUND, this.audioRingDeviceId, this.audioRingVolume, nextCallSession);
-          } else {
-            this.eventEmitter.emit(ON_PLAY_INBOUND_CALL_SIGNAL_SOUND,
-              this.audioOutputDeviceId,
-              this.audioOutputVolume,
-              nextCallSession);
-          }
+        this.currentSipSession = sipSessionId;
+        this.currentCallSession = callSession;
+    }
+    hasAnActiveCall() {
+        return !!this.currentSipSession;
+    }
+    // /!\ In some case with react-native webrtc the session will have only one audio stream set
+    // Maybe due to https://github.com/react-native-webrtc/react-native-webrtc/issues/401
+    hasActiveRemoteVideoStream() {
+        const sipSession = this.currentSipSession;
+        if (!sipSession) {
+            return false;
+        }
+        const { 
+        // @ts-ignore
+        peerConnection, } = sipSession.sessionDescriptionHandler;
+        const remoteStream = peerConnection.getRemoteStreams().find((stream) => !!stream.getVideoTracks().length);
+        return remoteStream && remoteStream.getVideoTracks().some(track => !track.muted);
+    }
+    callCount() {
+        return Object.keys(this.callSessions).length;
+    }
+    isCurrentCallSipSession(callSession) {
+        if (!this.currentSipSession) {
+            return false;
         }
-
-        // $FlowFixMe
-        this.eventEmitter.emit(ON_CALL_INCOMING, nextCallSession, nextCallSession.cameraEnabled);
-      }, 100);
-    }
-
-    if (callSession.getId() in this.ignoredSessions) {
-      return;
-    }
-
-    if (!sipSession.isCanceled) {
-      setTimeout(() => {
-        this.eventEmitter.emit(ON_PLAY_HANGUP_SOUND, this.audioOutputDeviceId, this.audioOutputVolume, callSession);
-      }, 10);
-    }
-
-    return isCurrentSession;
-  }
-
-  setActiveSipSession(callSession             ) {
-    const sipSessionId = this.findSipSession(callSession);
-    if (!sipSessionId) {
-      return;
-    }
-
-    this.currentSipSession = sipSessionId;
-    this.currentCallSession = callSession;
-  }
-
-  hasAnActiveCall() {
-    return !!this.currentSipSession;
-  }
-
-  // /!\ In some case with react-native webrtc the session will have only one audio stream set
-  // Maybe due to https://github.com/react-native-webrtc/react-native-webrtc/issues/401
-  hasActiveRemoteVideoStream() {
-    const sipSession = this.currentSipSession;
-    if (!sipSession) {
-      return false;
-    }
-
-    const { peerConnection } = sipSession.sessionDescriptionHandler;
-    const remoteStream = peerConnection.getRemoteStreams().find(stream => !!stream.getVideoTracks().length);
-
-    return remoteStream && remoteStream.getVideoTracks().some(track => !track.muted);
-  }
-
-  callCount() {
-    return Object.keys(this.callSessions).length;
-  }
-
-  isCurrentCallSipSession(callSession             )          {
-    if (!this.currentSipSession) {
-      return false;
-    }
-
-    return this.currentSipSession && this.getSipSessionId(this.currentSipSession) === callSession.getId();
-  }
-
-  hasVideo(callSession             )          {
-    return this.client.hasVideo(callSession.getId());
-  }
-
-  hasAVideoTrack(callSession             )          {
-    return this.client.hasAVideoTrack(callSession.getId());
-  }
-
-  // Deprecated
-  getLocalStreamForCall(callSession             )               {
-    logger.warn('WebRTCPhone.getLocalStreamForCall is deprecated, use WebRTCPhone.getLocalStream instead');
-
-    return this.getLocalStream(callSession);
-  }
-
-  // Deprecated
-  getRemoteStreamForCall(callSession             )               {
-    logger.warn('WebRTCPhone.getRemoteStreamForCall is deprecated, use WebRTCPhone.getRemoteStream instead');
-
-    return this.getRemoteVideoStream(callSession);
-  }
-
-  getRemoteStreamsForCall(callSession             )               {
-    logger.warn('WebRTCPhone.getRemoteStreamsForCall is deprecated, use WebRTCPhone.getRemoteStreams instead');
-
-    return this.getRemoteStream(callSession);
-  }
-
-  accept(callSession             , cameraEnabled          )                         {
-    logger.info('WebRTC phone - accept call', { id: callSession ? callSession.getId() : 'n/a', cameraEnabled });
-    if (this.currentSipSession) {
-      this.holdSipSession(
-        this.currentSipSession,
-        this.currentCallSession,
-        true,
-      );
-    }
-
-    if (!callSession) {
-      logger.warn('no CallSession to accept.');
-      return Promise.resolve(null);
-    }
-
-    const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
-    if (sipSessionId && !(sipSessionId in this.callSessions)) {
-      logger.warn('Call session already ended or not found, can\'t accept it.', { sipSessionId });
-      return Promise.resolve(null);
-    }
-
-    if (callSession.getId() in this.acceptedSessions) {
-      logger.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
-      return Promise.resolve(callSession.sipCallId);
-    }
-
-    this.shouldSendReinvite = false;
-    callSession.answerTime = new Date();
-    this._updateCallSession(callSession);
-    this.acceptedSessions[callSession.getId()] = true;
-
-    this.eventEmitter.emit(ON_CALL_ANSWERED, callSession);
-
-    const sipSession = this.client.getSipSession(callSession.getId());
-    if (sipSession) {
-      if (sipSession.state === sessionState.SessionState.Terminated || sipSession.state === sessionState.SessionState.Terminating) {
-        logger.warn('Trying to answer a terminated sipSession.');
-
+        return this.currentSipSession && this.getSipSessionId(this.currentSipSession) === callSession.getId();
+    }
+    hasVideo(callSession) {
+        return this.client.hasVideo(callSession.getId());
+    }
+    hasAVideoTrack(callSession) {
+        return this.client.hasAVideoTrack(callSession.getId());
+    }
+    // Deprecated
+    getLocalStreamForCall(callSession) {
+        logger.warn('WebRTCPhone.getLocalStreamForCall is deprecated, use WebRTCPhone.getLocalStream instead');
+        return this.getLocalStream(callSession);
+    }
+    // Deprecated
+    getRemoteStreamForCall(callSession) {
+        logger.warn('WebRTCPhone.getRemoteStreamForCall is deprecated, use WebRTCPhone.getRemoteStream instead');
+        return this.getRemoteVideoStream(callSession);
+    }
+    getRemoteStreamsForCall(callSession) {
+        logger.warn('WebRTCPhone.getRemoteStreamsForCall is deprecated, use WebRTCPhone.getRemoteStreams instead');
+        return this.getRemoteStream(callSession);
+    }
+    accept(callSession, cameraEnabled) {
+        logger.info('WebRTC phone - accept call', {
+            id: callSession ? callSession.getId() : 'n/a',
+            cameraEnabled,
+        });
+        if (this.currentSipSession) {
+            this.holdSipSession(this.currentSipSession, this.currentCallSession, true);
+        }
+        if (!callSession) {
+            logger.warn('no CallSession to accept.');
+            return Promise.resolve(null);
+        }
+        const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
+        if (sipSessionId && !(sipSessionId in this.callSessions)) {
+            logger.warn('Call session already ended or not found, can\'t accept it.', {
+                sipSessionId,
+            });
+            return Promise.resolve(null);
+        }
+        if (callSession.getId() in this.acceptedSessions) {
+            logger.warn('CallSession already accepted.', {
+                id: callSession ? callSession.getId() : 'n/a',
+            });
+            return Promise.resolve(callSession.sipCallId);
+        }
+        this.shouldSendReinvite = false;
+        callSession.answerTime = new Date();
+        this._updateCallSession(callSession);
+        this.acceptedSessions[callSession.getId()] = true;
+        this.eventEmitter.emit(exports.ON_CALL_ANSWERED, callSession);
+        const sipSession = this.client.getSipSession(callSession.getId());
+        if (sipSession) {
+            if (sipSession.state === session_state_1.SessionState.Terminated || sipSession.state === session_state_1.SessionState.Terminating) {
+                logger.warn('Trying to answer a terminated sipSession.');
+                return Promise.resolve(null);
+            }
+            logger.info('accept call, session found', {
+                sipId: sipSession.id,
+            });
+            return this.client.answer(sipSession, this.allowVideo ? cameraEnabled : false).then(() => {
+                return callSession.sipCallId;
+            }).catch(e => {
+                logger.error(e);
+                this.endCurrentCall(callSession);
+                throw e;
+            });
+        }
+        logger.warn('no CallSession found to accept.', {
+            id: callSession ? callSession.getId() : 'n/a',
+        });
         return Promise.resolve(null);
-      }
-
-      logger.info('accept call, session found', { sipId: sipSession.id });
-
-      // $FlowFixMe
-      return this.client.answer(sipSession, this.allowVideo ? cameraEnabled : false).then(() => {
-        return callSession.sipCallId;
-      }).catch(e => {
-        logger.error(e);
-        this.endCurrentCall(callSession);
-        throw e;
-      });
-    }
-
-    logger.warn('no CallSession found to accept.', { id: callSession ? callSession.getId() : 'n/a' });
-
-    return Promise.resolve(null);
-  }
-
-  isAccepted(callSession             )          {
-    return callSession && callSession.getId() in this.acceptedSessions;
-  }
-
-  async reject(callSession             )                {
-    logger.info('reject WebRTC called', { id: callSession.getId() });
-
-    this.eventEmitter.emit(ON_TERMINATE_SOUND, callSession, 'call rejected locally');
-    if (!callSession || callSession.getId() in this.rejectedSessions) {
-      return;
-    }
-
-    this.shouldSendReinvite = false;
-    this.rejectedSessions[callSession.getId()] = true;
-
-    const sipSession = this.findSipSession(callSession);
-    if (sipSession) {
-      logger.info('WebRTC rejecting', { sipId: sipSession.id });
-
-      this.client.hangup(sipSession);
-    }
-  }
-
-  async ignore(callSession             )                {
-    logger.info('WebRTC ignore', { id: callSession.getId() });
-
-    // kill the ring
-    this.eventEmitter.emit(ON_TERMINATE_SOUND, callSession, 'ignoring call');
-    this.ignoredSessions[callSession.getId()] = true;
-    callSession.ignore();
-  }
-
-  atxfer(callSession             )          {
-    const sipSession = this.findSipSession(callSession);
-
-    if (sipSession) {
-      logger.info('WebRTC atxfer', { sipId: sipSession.id });
-
-      return this.client.atxfer(sipSession);
-    }
-  }
-
-  hold(callSession             , withEvent          = true)                {
-    logger.info('WebRTC hold', { id: callSession.getId() });
-
-    const sipSession = this.findSipSession(callSession);
-
-    return sipSession ? this.holdSipSession(sipSession, callSession, withEvent) : null;
-  }
-
-  // @Deprecated
-  unhold(callSession             , withEvent          = true)                {
-    console.warn('Please note that `phone.unhold()` is being deprecated; `phone.resume()` is the preferred method');
-
-    return this.resume(callSession, withEvent);
-  }
-
-  resume(callSession              , withEvent          = true)               {
-    logger.info('WebRTC resume called', { id: callSession ? callSession.getId() : null });
-
-    const sipSession = this.findSipSession(callSession);
-    if (!sipSession) {
-      return new Promise((resolve, reject) => reject(new Error('No session to resume')));
-    }
-
-    logger.info('WebRTC resuming', { sipId: sipSession.id });
-
-    // Hold current session if different from the current one (we don't want 2 sessions active at the same time).
-    if (this.currentSipSession && this.currentSipSession.id !== sipSession.id) {
-      logger.info('WebRTC hold call after resume', { id: this.currentSipSession.id });
-      this.holdSipSession(
-        this.currentSipSession,
-        this.currentCallSession,
-        withEvent,
-      );
-    }
-
-    const promise = this.unholdSipSession(sipSession, callSession, withEvent);
-    this.currentSipSession = sipSession;
-    if (callSession) {
-      this.currentCallSession = callSession;
-    }
-
-    return promise;
-  }
-
-  async holdSipSession(
-    sipSession         ,
-    callSession              ,
-    withEvent          = true,
-  )               {
-    if (!sipSession) {
-      return new Promise((resolve, reject) => reject(new Error('No session to hold')));
-    }
-    const sessionId = this.getSipSessionId(sipSession);
-    const hasVideo = this.client.hasLocalVideo(sessionId);
-
-    logger.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
-
-    // Stop screenshare if needed
-    if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
-      this.lastScreenShare = this.currentScreenShare;
-      await this.stopScreenSharing(false, callSession);
-    }
-
-    // Downgrade to audio if needed
-    if (hasVideo) {
-      await this.client.downgradeToAudio(sipSession);
-    }
-
-    const isConference = !!callSession && callSession.isConference();
-
-    const promise = this.client.hold(sipSession, isConference, hasVideo);
-    if (withEvent) {
-      this.eventEmitter.emit(ON_CALL_HELD, this._createCallSession(sipSession, callSession));
-    }
-
-    return promise;
-  }
-
-  async unholdSipSession(
-    sipSession         ,
-    callSession              ,
-    withEvent          = true,
-  )               {
-    if (!sipSession) {
-      return new Promise((resolve, reject) => reject(new Error('No session to unhold')));
-    }
-
-    const isConference = !!callSession && callSession.isConference();
-
-    const sessionId = this.getSipSessionId(sipSession);
-    logger.info('WebRTC unhold sip session', { sessionId, isConference });
-
-    const { hasVideo } = this.client.getHeldSession(sessionId) || {};
-    const wasScreensharing = this.lastScreenShare && this.lastScreenShare.sipSessionId === sessionId;
-    const wasDesktop = this.lastScreenShare && this.lastScreenShare.desktop;
-    const promise = this.client.unhold(sipSession, isConference);
-
-    if (hasVideo) {
-      const constraints = {
-        audio: false,
-        video: true,
-        screen: wasScreensharing,
-        desktop: wasDesktop,
-      };
-
-      await this.client.upgradeToVideo(sipSession, constraints, isConference);
-    }
-
-    const onScreenSharing = stream => {
-      const hadVideo = this.lastScreenShare && this.lastScreenShare.hadVideo;
-      if (stream) {
-        this._onScreenSharing(stream, sipSession, callSession, hadVideo);
-      }
-      this.lastScreenShare = null;
-    };
-
-    if (withEvent) {
-      const updatedCallSession = this._createCallSession(sipSession, callSession);
-      // Deprecated event
-      this.eventEmitter.emit(ON_CALL_UNHELD, updatedCallSession);
-      this.eventEmitter.emit(ON_CALL_RESUMED, updatedCallSession);
-    }
-
-    return promise.then(() => {
-      const stream = callSession ? this.getLocalVideoStream(callSession) : null;
-      if (wasScreensharing) {
-        onScreenSharing(stream);
-      }
-
-      return stream;
-    });
-  }
-
-  mute(callSession              , withEvent          = true)       {
-    logger.info('WebRTC mute called', { id: callSession ? callSession.getId() : null });
-
-    const sipSession = this.findSipSession(callSession);
-    if (!sipSession) {
-      return;
-    }
-
-    logger.info('WebRTC muting', { sipId: sipSession.id });
-    this.client.mute(sipSession);
-    const newCallSession = this._createCallSession(sipSession, callSession, { muted: true });
-
-    if (this.currentCallSession && newCallSession.is(this.currentCallSession)) {
-      this.currentCallSession = newCallSession;
-    }
-
-    if (withEvent) {
-      this.eventEmitter.emit(ON_CALL_MUTED, newCallSession);
-    }
-  }
-
-  unmute(callSession              , withEvent          = true)       {
-    logger.info('WebRTC unmute called', { id: callSession ? callSession.getId() : null });
-
-    const sipSession = this.findSipSession(callSession);
-    if (!sipSession) {
-      return;
-    }
-
-    logger.info('WebRTC unmuting', { sipId: sipSession.id });
-    this.client.unmute(sipSession);
-    const newCallSession = this._createCallSession(sipSession, callSession, { muted: false });
-
-    if (this.currentCallSession && newCallSession.is(this.currentCallSession)) {
-      this.currentCallSession = newCallSession;
-    }
-
-    if (withEvent) {
-      this.eventEmitter.emit(ON_CALL_UNMUTED, newCallSession);
-    }
-  }
-
-  isAudioMuted(callSession             )          {
-    const sipSession = this.findSipSession(callSession);
-    if (!sipSession) {
-      return true;
-    }
-
-    return this.client.isAudioMuted(sipSession);
-  }
-
-  turnCameraOn(callSession              )       {
-    const sipSession = this.findSipSession(callSession);
-    if (!sipSession) {
-      return;
-    }
-    logger.info('WebRTC turn camera on', { sipId: sipSession.id });
-
-    this.client.toggleCameraOn(sipSession);
-
-    this.eventEmitter.emit(ON_CAMERA_RESUMED, this._createCameraResumedCallSession(sipSession, callSession));
-  }
-
-  turnCameraOff(callSession              )       {
-    const sipSession = this.findSipSession(callSession);
-    if (!sipSession) {
-      return;
-    }
-    logger.info('WebRTC turn camera off', { sipId: sipSession.id });
-
-    this.client.toggleCameraOff(sipSession);
-    this.eventEmitter.emit(ON_CAMERA_DISABLED, this._createCameraDisabledCallSession(sipSession, callSession));
-  }
-
-  sendKey(callSession              , tone        )       {
-    const sipSession = this.findSipSession(callSession);
-    if (!sipSession) {
-      return;
-    }
-
-    logger.info('WebRTC send key', { sipId: sipSession.id, tone });
-    this.client.sendDTMF(sipSession, tone);
-  }
-
-  // Should be async to match CTIPhone definition
-  // @TODO: line is not used here
-  async makeCall(number        , line     , cameraEnabled          ,
-    audioOnly          = false, conference          = false)                        {
-    logger.info('make WebRTC call', { number, lineId: line ? line.id : null, cameraEnabled, audioOnly, conference });
-    if (!number) {
-      return Promise.resolve(null);
-    }
-
-    if (!this.client.isRegistered()) {
-      await this.client.register();
-    }
-    if (this.currentSipSession) {
-      this.holdSipSession(
-        this.currentSipSession,
-        this.currentCallSession,
-        true,
-      ).catch(e => {
-        logger.warn('Unable to hold current session when making another call', e);
-      });
-    }
-
-    let sipSession         ;
-    try {
-      sipSession = this.client.call(number, this.allowVideo ? cameraEnabled : false, audioOnly, conference);
-      this._bindEvents(sipSession);
-    } catch (error) {
-      console.warn(error);
-      logger.warn('make WebRTC call, error', { message: error.message, stack: error.stack });
-      return Promise.resolve(null);
-    }
-    const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
-    callSession.setIsConference(conference);
-    callSession.creationTime = new Date();
-    this._updateCallSession(callSession);
-
-    this.eventEmitter.emit(ON_PLAY_PROGRESS_SOUND, this.audioOutputDeviceId, this.audioOutputVolume);
-
-    this.currentSipSession = sipSession;
-    this.currentCallSession = callSession;
-
-    this.eventEmitter.emit(ON_CALL_OUTGOING, callSession);
-
-    // If an invite promise exists, catch exceptions on it to trigger error like OverConstraintsError.
-    if (sipSession.invitePromise) {
-      sipSession.invitePromise.catch(error => {
-        this.eventEmitter.emit(ON_CALL_ERROR, error, callSession);
-      });
-    }
-
-    return Promise.resolve(callSession);
-  }
-
-  transfer(callSession              , target        )       {
-    const sipSession = this.findSipSession(callSession);
-    if (!sipSession) {
-      return;
-    }
-
-    logger.info('WebRTC transfer', { sipId: sipSession.id, target });
-
-    this.client.transfer(sipSession, target);
-  }
-
-  async indirectTransfer(source             , destination             )                   {
-    const sipSession = this.client.getSipSession(source.sipCallId);
-    const sipSessionTarget = this.client.getSipSession(destination.sipCallId);
-
-    logger.info('WebRTC indirect transfer', {
-      sipId: sipSession ? sipSession.id : null,
-      target: sipSessionTarget ? sipSessionTarget.id : null,
-    });
-
-    if (!sipSessionTarget) {
-      return Promise.reject();
-    }
-
-    return new Promise(resolve => {
-      const options = {
-        requestDelegate: {
-          onAccept: () => {
-            // Wait a little so the REFER will take effect
+    }
+    isAccepted(callSession) {
+        return callSession && callSession.getId() in this.acceptedSessions;
+    }
+    reject(callSession) {
+        return __awaiter(this, void 0, void 0, function* () {
+            logger.info('reject WebRTC called', {
+                id: callSession.getId(),
+            });
+            this.eventEmitter.emit(exports.ON_TERMINATE_SOUND, callSession, 'call rejected locally');
+            if (!callSession || callSession.getId() in this.rejectedSessions) {
+                return;
+            }
+            this.shouldSendReinvite = false;
+            this.rejectedSessions[callSession.getId()] = true;
+            const sipSession = this.findSipSession(callSession);
+            if (sipSession) {
+                logger.info('WebRTC rejecting', {
+                    sipId: sipSession.id,
+                });
+                this.client.hangup(sipSession);
+            }
+        });
+    }
+    ignore(callSession) {
+        return __awaiter(this, void 0, void 0, function* () {
+            logger.info('WebRTC ignore', {
+                id: callSession.getId(),
+            });
+            // kill the ring
+            this.eventEmitter.emit(exports.ON_TERMINATE_SOUND, callSession, 'ignoring call');
+            this.ignoredSessions[callSession.getId()] = true;
+            callSession.ignore();
+        });
+    }
+    atxfer(callSession) {
+        const sipSession = this.findSipSession(callSession);
+        if (sipSession) {
+            logger.info('WebRTC atxfer', {
+                sipId: sipSession.id,
+            });
+            return this.client.atxfer(sipSession);
+        }
+    }
+    hold(callSession, withEvent = true) {
+        logger.info('WebRTC hold', {
+            id: callSession.getId(),
+        });
+        const sipSession = this.findSipSession(callSession);
+        return sipSession ? this.holdSipSession(sipSession, callSession, withEvent) : null;
+    }
+    // @Deprecated
+    unhold(callSession, withEvent = true) {
+        console.warn('Please note that `phone.unhold()` is being deprecated; `phone.resume()` is the preferred method');
+        return this.resume(callSession, withEvent);
+    }
+    resume(callSession, withEvent = true) {
+        logger.info('WebRTC resume called', {
+            id: callSession ? callSession.getId() : null,
+        });
+        const sipSession = this.findSipSession(callSession);
+        if (!sipSession) {
+            return new Promise((resolve, reject) => reject(new Error('No session to resume')));
+        }
+        logger.info('WebRTC resuming', {
+            sipId: sipSession.id,
+        });
+        // Hold current session if different from the current one (we don't want 2 sessions active at the same time).
+        if (this.currentSipSession && this.currentSipSession.id !== sipSession.id) {
+            logger.info('WebRTC hold call after resume', {
+                id: this.currentSipSession.id,
+            });
+            this.holdSipSession(this.currentSipSession, this.currentCallSession, withEvent);
+        }
+        const promise = this.unholdSipSession(sipSession, callSession, withEvent);
+        this.currentSipSession = sipSession;
+        if (callSession) {
+            this.currentCallSession = callSession;
+        }
+        return promise;
+    }
+    holdSipSession(sipSession, callSession, withEvent = true) {
+        return __awaiter(this, void 0, void 0, function* () {
+            if (!sipSession) {
+                return new Promise((resolve, reject) => reject(new Error('No session to hold')));
+            }
+            const sessionId = this.getSipSessionId(sipSession);
+            const hasVideo = this.client.hasLocalVideo(sessionId);
+            logger.info('WebRTC hold sip session', {
+                sipId: sipSession.id,
+                hasVideo,
+            });
+            // Stop screenshare if needed
+            if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
+                this.lastScreenShare = this.currentScreenShare;
+                yield this.stopScreenSharing(false, callSession);
+            }
+            // Downgrade to audio if needed
+            if (hasVideo) {
+                yield this.client.downgradeToAudio(sipSession);
+            }
+            const isConference = !!callSession && callSession.isConference();
+            const promise = this.client.hold(sipSession, isConference, hasVideo);
+            if (withEvent) {
+                this.eventEmitter.emit(exports.ON_CALL_HELD, this._createCallSession(sipSession, callSession));
+            }
+            return promise;
+        });
+    }
+    unholdSipSession(sipSession, callSession, withEvent = true) {
+        return __awaiter(this, void 0, void 0, function* () {
+            if (!sipSession) {
+                return new Promise((resolve, reject) => reject(new Error('No session to unhold')));
+            }
+            const isConference = !!callSession && callSession.isConference();
+            const sessionId = this.getSipSessionId(sipSession);
+            logger.info('WebRTC unhold sip session', {
+                sessionId,
+                isConference,
+            });
+            const { hasVideo, } = this.client.getHeldSession(sessionId) || {};
+            const wasScreensharing = this.lastScreenShare && this.lastScreenShare.sipSessionId === sessionId;
+            const wasDesktop = this.lastScreenShare && this.lastScreenShare.desktop;
+            const promise = this.client.unhold(sipSession, isConference);
+            if (hasVideo) {
+                const constraints = {
+                    audio: false,
+                    video: true,
+                    screen: wasScreensharing,
+                    desktop: wasDesktop,
+                };
+                yield this.client.upgradeToVideo(sipSession, constraints, isConference);
+            }
+            const onScreenSharing = (stream) => {
+                const hadVideo = this.lastScreenShare && this.lastScreenShare.hadVideo;
+                if (stream) {
+                    this._onScreenSharing(stream, sipSession, callSession, hadVideo);
+                }
+                this.lastScreenShare = null;
+            };
+            if (withEvent) {
+                const updatedCallSession = this._createCallSession(sipSession, callSession);
+                // Deprecated event
+                this.eventEmitter.emit(exports.ON_CALL_UNHELD, updatedCallSession);
+                this.eventEmitter.emit(exports.ON_CALL_RESUMED, updatedCallSession);
+            }
+            return promise.then(() => {
+                const stream = callSession ? this.getLocalVideoStream(callSession) : null;
+                if (wasScreensharing) {
+                    onScreenSharing(stream);
+                }
+                return stream;
+            });
+        });
+    }
+    mute(callSession, withEvent = true) {
+        logger.info('WebRTC mute called', {
+            id: callSession ? callSession.getId() : null,
+        });
+        const sipSession = this.findSipSession(callSession);
+        if (!sipSession) {
+            return;
+        }
+        logger.info('WebRTC muting', {
+            sipId: sipSession.id,
+        });
+        this.client.mute(sipSession);
+        const newCallSession = this._createCallSession(sipSession, callSession, {
+            muted: true,
+        });
+        if (this.currentCallSession && newCallSession.is(this.currentCallSession)) {
+            this.currentCallSession = newCallSession;
+        }
+        if (withEvent) {
+            this.eventEmitter.emit(exports.ON_CALL_MUTED, newCallSession);
+        }
+    }
+    unmute(callSession, withEvent = true) {
+        logger.info('WebRTC unmute called', {
+            id: callSession ? callSession.getId() : null,
+        });
+        const sipSession = this.findSipSession(callSession);
+        if (!sipSession) {
+            return;
+        }
+        logger.info('WebRTC unmuting', {
+            sipId: sipSession.id,
+        });
+        this.client.unmute(sipSession);
+        const newCallSession = this._createCallSession(sipSession, callSession, {
+            muted: false,
+        });
+        if (this.currentCallSession && newCallSession.is(this.currentCallSession)) {
+            this.currentCallSession = newCallSession;
+        }
+        if (withEvent) {
+            this.eventEmitter.emit(exports.ON_CALL_UNMUTED, newCallSession);
+        }
+    }
+    isAudioMuted(callSession) {
+        const sipSession = this.findSipSession(callSession);
+        if (!sipSession) {
+            return true;
+        }
+        return this.client.isAudioMuted(sipSession);
+    }
+    turnCameraOn(callSession) {
+        const sipSession = this.findSipSession(callSession);
+        if (!sipSession) {
+            return;
+        }
+        logger.info('WebRTC turn camera on', {
+            sipId: sipSession.id,
+        });
+        this.client.toggleCameraOn(sipSession);
+        this.eventEmitter.emit(exports.ON_CAMERA_RESUMED, this._createCameraResumedCallSession(sipSession, callSession));
+    }
+    turnCameraOff(callSession) {
+        const sipSession = this.findSipSession(callSession);
+        if (!sipSession) {
+            return;
+        }
+        logger.info('WebRTC turn camera off', {
+            sipId: sipSession.id,
+        });
+        this.client.toggleCameraOff(sipSession);
+        this.eventEmitter.emit(exports.ON_CAMERA_DISABLED, this._createCameraDisabledCallSession(sipSession, callSession));
+    }
+    sendKey(callSession, tone) {
+        const sipSession = this.findSipSession(callSession);
+        if (!sipSession) {
+            return;
+        }
+        logger.info('WebRTC send key', {
+            sipId: sipSession.id,
+            tone,
+        });
+        this.client.sendDTMF(sipSession, tone);
+    }
+    // Should be async to match CTIPhone definition
+    // @TODO: line is not used here
+    makeCall(number, line, cameraEnabled, audioOnly = false, conference = false) {
+        return __awaiter(this, void 0, void 0, function* () {
+            logger.info('make WebRTC call', {
+                number,
+                lineId: line ? line.id : null,
+                cameraEnabled,
+                audioOnly,
+                conference,
+            });
+            if (!number) {
+                return Promise.resolve(null);
+            }
+            if (!this.client.isRegistered()) {
+                yield this.client.register();
+            }
+            if (this.currentSipSession) {
+                this.holdSipSession(this.currentSipSession, this.currentCallSession, true).catch(e => {
+                    logger.warn('Unable to hold current session when making another call', e);
+                });
+            }
+            let sipSession;
+            try {
+                sipSession = this.client.call(number, this.allowVideo ? cameraEnabled : false, audioOnly, conference);
+                this._bindEvents(sipSession);
+            }
+            catch (error) {
+                console.warn(error);
+                logger.warn('make WebRTC call, error', {
+                    message: error.message,
+                    stack: error.stack,
+                });
+                return Promise.resolve(null);
+            }
+            const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
+            callSession.setIsConference(conference);
+            callSession.creationTime = new Date();
+            this._updateCallSession(callSession);
+            this.eventEmitter.emit(exports.ON_PLAY_PROGRESS_SOUND, this.audioOutputDeviceId, this.audioOutputVolume);
+            this.currentSipSession = sipSession;
+            this.currentCallSession = callSession;
+            this.eventEmitter.emit(exports.ON_CALL_OUTGOING, callSession);
+            // If an invite promise exists, catch exceptions on it to trigger error like OverConstraintsError.
+            // @ts-ignore
+            if (sipSession.invitePromise) {
+                // @ts-ignore
+                sipSession.invitePromise.catch(error => {
+                    this.eventEmitter.emit(exports.ON_CALL_ERROR, error, callSession);
+                });
+            }
+            return Promise.resolve(callSession);
+        });
+    }
+    transfer(callSession, target) {
+        const sipSession = this.findSipSession(callSession);
+        if (!sipSession) {
+            return;
+        }
+        logger.info('WebRTC transfer', {
+            sipId: sipSession.id,
+            target,
+        });
+        this.client.transfer(sipSession, target);
+    }
+    indirectTransfer(source, destination) {
+        return __awaiter(this, void 0, void 0, function* () {
+            const sipSession = this.client.getSipSession(source.sipCallId);
+            const sipSessionTarget = this.client.getSipSession(destination.sipCallId);
+            logger.info('WebRTC indirect transfer', {
+                sipId: sipSession ? sipSession.id : null,
+                target: sipSessionTarget ? sipSessionTarget.id : null,
+            });
+            if (!sipSessionTarget) {
+                return Promise.reject();
+            }
+            return new Promise(resolve => {
+                const options = {
+                    requestDelegate: {
+                        onAccept: () => {
+                            // Wait a little so the REFER will take effect
+                            setTimeout(() => {
+                                this.hangup(destination).then(resolve);
+                            }, 200);
+                        },
+                    },
+                };
+                return sipSessionTarget.refer(sipSession, options);
+            });
+        });
+    }
+    initiateCTIIndirectTransfer() { }
+    cancelCTIIndirectTransfer() { }
+    confirmCTIIndirectTransfer() { }
+    hangup(callSession) {
+        return __awaiter(this, void 0, void 0, function* () {
+            const sipSession = this.findSipSession(callSession);
+            if (!sipSession) {
+                console.error('Call is unknown to the WebRTC phone', callSession ? callSession.sipCallId : null, callSession ? callSession.callId : null);
+                return false;
+            }
+            logger.info('WebRTC hangup', {
+                sipId: sipSession.id,
+            });
+            const sipSessionId = this.getSipSessionId(sipSession);
+            yield this.client.hangup(sipSession);
+            if (callSession) {
+                // Removal of `this.callSessions` will be done in `_onCallTerminated`.
+                this.endCurrentCall(callSession);
+            }
+            else if (sipSessionId) {
+                if (callSession) {
+                    delete this.callSessions[sipSessionId];
+                }
+            }
+            this.shouldSendReinvite = false;
+            return true;
+        });
+    }
+    getStats(callSession) {
+        return __awaiter(this, void 0, void 0, function* () {
+            const sipSession = this.findSipSession(callSession);
+            return sipSession ? this.client.getStats(sipSession) : null;
+        });
+    }
+    startNetworkMonitoring(callSession, interval = 1000) {
+        const sipSession = this.findSipSession(callSession);
+        return sipSession ? this.client.startNetworkMonitoring(sipSession, interval) : null;
+    }
+    stopNetworkMonitoring(callSession) {
+        const sipSession = this.findSipSession(callSession);
+        return sipSession ? this.client.stopNetworkMonitoring(sipSession) : null;
+    }
+    forceCancel(sipSession) {
+        // @ts-ignore
+        if (!sipSession || !sipSession.outgoingInviteRequest) {
+            return;
+        }
+        // @ts-ignore
+        sipSession.outgoingInviteRequest.cancel();
+    }
+    endCurrentCall(callSession) {
+        const isCurrent = this.isCurrentCallSipSession(callSession);
+        logger.info('Ending current call', {
+            id: callSession ? callSession.getId() : null,
+            isCurrent,
+        });
+        if (isCurrent) {
+            this.currentSipSession = undefined;
+            this.currentCallSession = null;
+        }
+        this.eventEmitter.emit(exports.ON_TERMINATE_SOUND, callSession, 'locally ended');
+    }
+    onConnectionMade() { }
+    isConference(callSession) {
+        if (!callSession) {
+            return false;
+        }
+        return this.client && this.client.isConference(callSession.sipCallId);
+    }
+    close(force = false) {
+        return __awaiter(this, void 0, void 0, function* () {
+            logger.info('WebRTC close');
+            try {
+                yield Promise.race([this.unregister(), new Promise((resolve, reject) => setTimeout(() => reject(new Error('Unregister, timed out')), 3000))]);
+            }
+            catch (e) {
+                logger.error('WebRTC close, unregister error', e);
+            }
+            this.client.close(force);
+            this.unbind();
+            this.incomingSessions = [];
+            this.currentSipSession = undefined;
+            this.currentCallSession = null;
+            this.shouldSendReinvite = false;
+            this.rejectedSessions = {};
+        });
+    }
+    isRegistered() {
+        return this.client && this.client.isRegistered();
+    }
+    enableRinging() {
+        logger.info('WebRTC enable ringing');
+        this.ringingEnabled = true;
+    }
+    disableRinging() {
+        logger.info('WebRTC disable ringing');
+        this.ringingEnabled = false;
+    }
+    getCurrentCallSession() {
+        if (!this.currentSipSession) {
+            return null;
+        }
+        return this._createCallSession(this.currentSipSession, this.currentCallSession);
+    }
+    hasIncomingCallSession() {
+        return this.incomingSessions.length > 0;
+    }
+    getIncomingCallSession() {
+        if (!this.hasIncomingCallSession()) {
+            return null;
+        }
+        const sessionId = this.incomingSessions[0];
+        const sipSession = this.client.getSipSession(sessionId);
+        if (!sipSession) {
+            return null;
+        }
+        return this._createCallSession(sipSession);
+    }
+    // eslint-disable-next-line @typescript-eslint/default-param-last
+    sendMessage(sipSession = null, body, contentType = 'text/plain') {
+        return this.client.sendMessage(sipSession, body, contentType);
+    }
+    getLocalStream(callSession) {
+        return callSession ? this.client.getLocalStream(callSession.sipCallId) : null;
+    }
+    getLocalVideoStream(callSession) {
+        return callSession ? this.client.getLocalVideoStream(callSession.sipCallId) : null;
+    }
+    // Deprecated
+    getLocalMediaStream(callSession) {
+        logger.warn('WebRTCPhone.getLocalMediaStream is deprecated, use WebRTCPhone.getLocalStream instead');
+        return this.getLocalStream(callSession);
+    }
+    hasLocalVideo(callSession) {
+        return callSession ? this.client.hasLocalVideo(callSession.sipCallId) : false;
+    }
+    hasALocalVideoTrack(callSession) {
+        return callSession ? this.client.hasALocalVideoTrack(callSession.sipCallId) : false;
+    }
+    getRemoteStream(callSession) {
+        return callSession ? this.client.getRemoteStream(callSession.sipCallId) : null;
+    }
+    getRemoteVideoStream(callSession) {
+        return callSession ? this.client.getRemoteVideoStream(callSession.sipCallId) : null;
+    }
+    getRemoteVideoStreamFromPc(callSession) {
+        return callSession ? this.client.getRemoteVideoStreamFromPc(callSession.sipCallId) : null;
+    }
+    hasRemoteVideo(callSession) {
+        return callSession ? this.client.hasRemoteVideo(callSession.sipCallId) : false;
+    }
+    setMediaConstraints(media) {
+        this.client.setMediaConstraints(media);
+    }
+    isVideoRemotelyHeld(callSession) {
+        return callSession ? this.client.isVideoRemotelyHeld(callSession.sipCallId) : false;
+    }
+    bindClientEvents() {
+        this.client.unbind();
+        this.client.on(this.client.INVITE, (sipSession, wantsToDoVideo) => {
+            const autoAnswer = sipSession.request.getHeader('Answer-Mode') === 'Auto';
+            const withVideo = this.allowVideo ? wantsToDoVideo : false;
+            logger.info('WebRTC invite received', {
+                sipId: sipSession.id,
+                withVideo,
+                autoAnswer,
+            });
+            const callSession = this._createIncomingCallSession(sipSession, withVideo, null, autoAnswer);
+            this.incomingSessions.push(callSession.getId());
+            this._bindEvents(sipSession);
+            this.client.storeSipSession(sipSession);
+            if (!this.currentSipSession) {
+                if (this.ringingEnabled) {
+                    this.eventEmitter.emit(exports.ON_PLAY_RING_SOUND, this.audioRingDeviceId, this.audioRingVolume, callSession);
+                }
+            }
+            else {
+                this.eventEmitter.emit(exports.ON_PLAY_INBOUND_CALL_SIGNAL_SOUND, this.audioOutputDeviceId, this.audioOutputVolume, callSession);
+            }
+            callSession.creationTime = new Date();
+            this._updateCallSession(callSession);
+            this.eventEmitter.emit(exports.ON_CALL_INCOMING, callSession, wantsToDoVideo);
+        });
+        this.client.on(this.client.ON_REINVITE, (...args) => {
+            logger.info('WebRTC on reinvite', {
+                sessionId: args[0].id,
+                inviteId: args[1].id,
+                updatedCalleeName: args[2],
+                updatedNumber: args[3],
+            });
+            const sipSession = args[0];
+            // Give some time for the stream to be mounted.
+            // if `_createCallSession` is called too soon `hasVideo` will return false because the stream doesn't exists yet
             setTimeout(() => {
-              this.hangup(destination).then(resolve);
-            }, 200);
-          },
-        },
-      };
-
-      return sipSessionTarget.refer(sipSession, options);
-    });
-  }
-
-  initiateCTIIndirectTransfer() {}
-
-  cancelCTIIndirectTransfer() {}
-
-  confirmCTIIndirectTransfer() {}
-
-  async hangup(callSession              )                   {
-    const sipSession = this.findSipSession(callSession);
-    if (!sipSession) {
-      console.error('Call is unknown to the WebRTC phone', callSession ? callSession.sipCallId : null,
-        callSession ? callSession.callId : null);
-      return false;
-    }
-
-    logger.info('WebRTC hangup', { sipId: sipSession.id });
-
-    const sipSessionId = this.getSipSessionId(sipSession);
-
-    await this.client.hangup(sipSession);
-    if (callSession) {
-      // Removal of `this.callSessions` will be done in `_onCallTerminated`.
-      this.endCurrentCall(callSession);
-    } else if (sipSessionId) {
-      if (callSession) {
-        delete this.callSessions[callSession.getId()];
-      }
-    }
-
-    this.shouldSendReinvite = false;
-    return true;
-  }
-
-  async getStats(callSession              )                   {
-    const sipSession = this.findSipSession(callSession);
-
-    return sipSession ? this.client.getStats(sipSession) : null;
-  }
-
-  startNetworkMonitoring(callSession              , interval         = 1000) {
-    const sipSession = this.findSipSession(callSession);
-
-    return sipSession ? this.client.startNetworkMonitoring(sipSession, interval) : null;
-  }
-
-  stopNetworkMonitoring(callSession              ) {
-    const sipSession = this.findSipSession(callSession);
-
-    return sipSession ? this.client.stopNetworkMonitoring(sipSession) : null;
-  }
-
-  forceCancel(sipSession         )       {
-    if (!sipSession || !sipSession.outgoingInviteRequest) {
-      return;
-    }
-    sipSession.outgoingInviteRequest.cancel();
-  }
-
-  endCurrentCall(callSession             )       {
-    const isCurrent = this.isCurrentCallSipSession(callSession);
-
-    logger.info('Ending current call', {
-      id: callSession ? callSession.getId() : null,
-      isCurrent,
-    });
-
-    if (isCurrent) {
-      this.currentSipSession = undefined;
-      this.currentCallSession = null;
-    }
-
-    this.eventEmitter.emit(ON_TERMINATE_SOUND, callSession, 'locally ended');
-  }
-
-  onConnectionMade()       {}
-
-  isConference(callSession              )          {
-    if (!callSession) {
-      return false;
-    }
-
-    return this.client && this.client.isConference(callSession.sipCallId);
-  }
-
-  async close(force          = false)                {
-    logger.info('WebRTC close');
-
-    try {
-      await Promise.race([
-        this.unregister(),
-        new Promise((resolve, reject) => setTimeout(() => reject(new Error('Unregister, timed out')), 3000)),
-      ]);
-    } catch (e) {
-      logger.error('WebRTC close, unregister error', e);
-    }
-
-    this.client.close(force);
-    this.unbind();
-
-    this.incomingSessions = [];
-    this.currentSipSession = null;
-    this.currentCallSession = null;
-    this.shouldSendReinvite = false;
-    this.rejectedSessions = {};
-  }
-
-  isRegistered()          {
-    return this.client && this.client.isRegistered();
-  }
-
-  enableRinging()                       {
-    logger.info('WebRTC enable ringing');
-    this.ringingEnabled = true;
-  }
-
-  disableRinging()                       {
-    logger.info('WebRTC disable ringing');
-    this.ringingEnabled = false;
-  }
-
-  getCurrentCallSession()               {
-    if (!this.currentSipSession) {
-      return null;
-    }
-
-    return this._createCallSession(this.currentSipSession, this.currentCallSession);
-  }
-
-  hasIncomingCallSession()          {
-    return this.incomingSessions.length > 0;
-  }
-
-  getIncomingCallSession()               {
-    if (!this.hasIncomingCallSession()) {
-      return null;
-    }
-
-    const sessionId = this.incomingSessions[0];
-    const sipSession = this.client.getSipSession(sessionId);
-    if (!sipSession) {
-      return null;
-    }
-
-    return this._createCallSession(sipSession);
-  }
-
-  sendMessage(sipSession          = null, body        , contentType         = 'text/plain') {
-    return this.client.sendMessage(sipSession, body, contentType);
-  }
-
-  getLocalStream(callSession             )               {
-    return callSession ? this.client.getLocalStream(callSession.sipCallId) : null;
-  }
-
-  getLocalVideoStream(callSession             )               {
-    return callSession ? this.client.getLocalVideoStream(callSession.sipCallId) : null;
-  }
-
-  // Deprecated
-  getLocalMediaStream(callSession             ) {
-    logger.warn('WebRTCPhone.getLocalMediaStream is deprecated, use WebRTCPhone.getLocalStream instead');
-    return this.getLocalStream(callSession);
-  }
-
-  hasLocalVideo(callSession             ) {
-    return callSession ? this.client.hasLocalVideo(callSession.sipCallId) : null;
-  }
-
-  hasALocalVideoTrack(callSession             ) {
-    return callSession ? this.client.hasALocalVideoTrack(callSession.sipCallId) : null;
-  }
-
-  getRemoteStream(callSession             )               {
-    return callSession ? this.client.getRemoteStream(callSession.sipCallId) : null;
-  }
-
-  getRemoteVideoStream(callSession             )               {
-    return callSession ? this.client.getRemoteVideoStream(callSession.sipCallId) : null;
-  }
-
-  getRemoteVideoStreamFromPc(callSession             )               {
-    return callSession ? this.client.getRemoteVideoStreamFromPc(callSession.sipCallId) : null;
-  }
-
-  hasRemoteVideo(callSession             )          {
-    return callSession ? this.client.hasRemoteVideo(callSession.sipCallId) : false;
-  }
-
-  setMediaConstraints(media                        ) {
-    this.client.setMediaConstraints(media);
-  }
-
-  isVideoRemotelyHeld(callSession             ) {
-    return callSession ? this.client.isVideoRemotelyHeld(callSession.sipCallId) : null;
-  }
-
-  bindClientEvents() {
-    this.client.unbind();
-
-    this.client.on(this.client.INVITE, (sipSession         , wantsToDoVideo         ) => {
-      const autoAnswer = sipSession.request.getHeader('Answer-Mode') === 'Auto';
-      const withVideo = this.allowVideo ? wantsToDoVideo : false;
-      logger.info('WebRTC invite received', { sipId: sipSession.id, withVideo, autoAnswer });
-
-      const callSession = this._createIncomingCallSession(sipSession, withVideo, null, autoAnswer);
-      this.incomingSessions.push(callSession.getId());
-      this._bindEvents(sipSession);
-
-      this.client.storeSipSession(sipSession);
-
-      if (!this.currentSipSession) {
-        if (this.ringingEnabled) {
-          this.eventEmitter.emit(ON_PLAY_RING_SOUND, this.audioRingDeviceId, this.audioRingVolume, callSession);
+                // Update callSession
+                this._createCallSession(sipSession, this.callSessions[this.getSipSessionId(sipSession)]);
+                this.eventEmitter.emit.apply(this.eventEmitter, [this.client.ON_REINVITE, ...args]);
+            }, 2000);
+        });
+        this.client.on(this.client.ACCEPTED, (sipSession) => {
+            const sessionId = this.getSipSessionId(sipSession);
+            const hasSession = (sessionId in this.callSessions);
+            logger.info('WebRTC call accepted', {
+                sipId: sessionId,
+                hasSession,
+            });
+            if (!hasSession) {
+                logger.warn('Call accepted ignored, session is no longer present in the WebRtcPhone', {
+                    sessionId,
+                });
+                return;
+            }
+            this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
+            if (this.audioOutputDeviceId) {
+                this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
+            }
+        });
+        this.client.on('ended', () => { });
+        this.client.on(this.client.ON_ERROR, e => {
+            logger.error('WebRTC error', e);
+            this.eventEmitter.emit(exports.ON_CALL_ERROR, e);
+        });
+        this.client.on(this.client.REJECTED, session => {
+            logger.info('WebRTC call rejected', session.id);
+            const callSession = this._createCallSession(session);
+            callSession.endTime = new Date();
+            this._updateCallSession(callSession);
+            this.eventEmitter.emit(exports.ON_CALL_REJECTED, callSession);
+        });
+        this.client.on(this.client.UNREGISTERED, () => {
+            logger.info('WebRTC unregistered');
+            this.eventEmitter.emit(exports.ON_UNREGISTERED);
+        });
+        this.client.on(this.client.ON_DISCONNECTED, () => {
+            logger.info('WebRTC disconnected');
+            this.eventEmitter.emit(exports.ON_DISCONNECTED);
+        });
+        this.client.on(this.client.ON_EARLY_MEDIA, session => {
+            logger.info('WebRTC early media');
+            const callSession = this._createCallSession(session);
+            this.eventEmitter.emit(exports.ON_EARLY_MEDIA, callSession);
+        });
+        this.client.on(this.client.REGISTERED, () => {
+            var _a, _b;
+            logger.info('WebRTC registered', {
+                shouldSendReinvite: this.shouldSendReinvite,
+                state: this.currentSipSession ? this.currentSipSession.state : null,
+                currentSipSession: !!this.currentSipSession,
+            });
+            this.stopHeartbeat();
+            this.eventEmitter.emit(exports.ON_REGISTERED);
+            // If the phone registered with a current callSession (eg: when switching network):
+            // send a reinvite to renegociate ICE with new IP
+            if (this.shouldSendReinvite && this.currentSipSession && this.currentSipSession.state === session_state_1.SessionState.Established) {
+                this.shouldSendReinvite = false;
+                // @ts-ignore
+                const pc = (_b = (_a = this.currentSipSession) === null || _a === void 0 ? void 0 : _a.sessionDescriptionHandler) === null || _b === void 0 ? void 0 : _b.peerConnection;
+                const isConference = pc ? pc.sfu : false;
+                const hasVideo = this.currentCallSession && this.currentCallSession.cameraEnabled;
+                try {
+                    // Send reinvite with iceRestart
+                    this.sendReinvite(this.currentCallSession, null, isConference, !hasVideo, true);
+                }
+                catch (e) {
+                    logger.error('WebRTC reinvite after register, error', {
+                        message: e.message,
+                        stack: e.stack,
+                    });
+                }
+            }
+        });
+        this.client.on(this.client.CONNECTED, () => {
+            logger.info('WebRTC client connected');
+            this.stopHeartbeat();
+        });
+        this.client.on(this.client.DISCONNECTED, () => {
+            logger.info('WebRTC client disconnected');
+            this.eventEmitter.emit(exports.ON_UNREGISTERED);
+            // Do not trigger heartbeat if already running
+            if (!this.client.hasHeartbeat()) {
+                this.startHeartbeat();
+            }
+            // Tell to send reinvite when reconnecting
+            this.shouldSendReinvite = true;
+        });
+        this.client.on(this.client.ON_TRACK, (session, event) => {
+            this.eventEmitter.emit(exports.ON_TRACK, session, event);
+        });
+        this.client.on('onVideoInputChange', stream => {
+            this.eventEmitter.emit(exports.ON_VIDEO_INPUT_CHANGE, stream);
+        });
+        this.client.on(this.client.MESSAGE, (message) => {
+            this._onMessage(message);
+            this.eventEmitter.emit(exports.ON_MESSAGE, message);
+        });
+        this.client.on(this.client.ON_NETWORK_STATS, (session, stats, previousStats) => {
+            const callSession = this.getCallSession(this.getSipSessionId(session));
+            this.eventEmitter.emit(exports.ON_NETWORK_STATS, callSession, stats, previousStats);
+        });
+        // Used when upgrading directly in screenshare mode
+        this.client.on(this.client.ON_SCREEN_SHARING_REINVITE, (sipSession, response, desktop) => {
+            const sipSessionId = this.getSipSessionId(sipSession);
+            const localStream = this.client.getLocalStream(sipSessionId);
+            const callSession = this.callSessions[sipSessionId];
+            logger.info('Updrading directly in screensharing mode', {
+                sipSessionId,
+                tracks: (localStream === null || localStream === void 0 ? void 0 : localStream.getTracks()) || null,
+            });
+            this._onScreenSharing(localStream, sipSession, callSession, false, desktop);
+        });
+    }
+    // Find a corresponding sipSession from a CallSession
+    findSipSession(callSession) {
+        const keys = this.client.getSipSessionIds();
+        const keyIndex = keys.findIndex(sessionId => callSession && callSession.isId(sessionId));
+        if (keyIndex === -1) {
+            const currentSipSessionId = this.currentSipSession ? this.getSipSessionId(this.currentSipSession) : this.client.getSipSessionIds()[0];
+            return currentSipSessionId ? this.client.getSipSession(currentSipSessionId) : null;
         }
-      } else {
-        this.eventEmitter.emit(ON_PLAY_INBOUND_CALL_SIGNAL_SOUND,
-          this.audioOutputDeviceId,
-          this.audioOutputVolume,
-          callSession);
-      }
-
-      callSession.creationTime = new Date();
-      this._updateCallSession(callSession);
-
-      this.eventEmitter.emit(ON_CALL_INCOMING, callSession, wantsToDoVideo);
-    });
-
-    this.client.on(this.client.ON_REINVITE, (...args) => {
-      logger.info('WebRTC on reinvite', {
-        sessionId: args[0].id,
-        inviteId: args[1].id,
-        updatedCalleeName: args[2],
-        updatedNumber: args[3],
-      });
-
-      const sipSession = args[0];
-
-      // Give some time for the stream to be mounted.
-      // if `_createCallSession` is called too soon `hasVideo` will return false because the stream doesn't exists yet
-      setTimeout(() => {
-        // Update callSession
-        this._createCallSession(sipSession, this.callSessions[this.getSipSessionId(sipSession)]);
-
-        this.eventEmitter.emit.apply(this.eventEmitter, [this.client.ON_REINVITE, ...args]);
-      }, 2000);
-    });
-
-    this.client.on(this.client.ACCEPTED, (sipSession         ) => {
-      const sessionId = this.getSipSessionId(sipSession);
-      const hasSession = sessionId in this.callSessions;
-
-      logger.info('WebRTC call accepted', { sipId: sessionId, hasSession });
-
-      if (!hasSession) {
-        logger.warn('Call accepted ignored, session is no longer present in the WebRtcPhone', { sessionId });
-        return;
-      }
-
-      this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
-
-      if (this.audioOutputDeviceId) {
-        this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
-      }
-    });
-    this.client.on('ended', () => {});
-
-    this.client.on(this.client.ON_ERROR, e => {
-      logger.error('WebRTC error', e);
-      this.eventEmitter.emit(ON_CALL_ERROR, e);
-    });
-
-    this.client.on(this.client.REJECTED, session => {
-      logger.info('WebRTC call rejected', session.id);
-      const callSession = this._createCallSession(session);
-      callSession.endTime = new Date();
-      this._updateCallSession(callSession);
-
-      this.eventEmitter.emit(ON_CALL_REJECTED, callSession);
-    });
-
-    this.client.on(this.client.UNREGISTERED, () => {
-      logger.info('WebRTC unregistered');
-      this.eventEmitter.emit(ON_UNREGISTERED);
-    });
-
-    this.client.on(this.client.ON_DISCONNECTED, () => {
-      logger.info('WebRTC disconnected');
-      this.eventEmitter.emit(ON_DISCONNECTED);
-    });
-    this.client.on(this.client.ON_EARLY_MEDIA, session => {
-      logger.info('WebRTC early media');
-      const callSession = this._createCallSession(session);
-      this.eventEmitter.emit(ON_EARLY_MEDIA, callSession);
-    });
-
-    this.client.on(this.client.REGISTERED, () => {
-      logger.info('WebRTC registered', {
-        shouldSendReinvite: this.shouldSendReinvite,
-        state: this.currentSipSession ? this.currentSipSession.state : null,
-        currentSipSession: !!this.currentSipSession,
-      });
-      this.stopHeartbeat();
-      this.eventEmitter.emit(ON_REGISTERED);
-
-      // If the phone registered with a current callSession (eg: when switching network):
-      // send a reinvite to renegociate ICE with new IP
-      if (this.shouldSendReinvite && this.currentSipSession
-        && this.currentSipSession.state === sessionState.SessionState.Established) {
-        this.shouldSendReinvite = false;
-        const pc = this.currentSipSession.sessionDescriptionHandler.peerConnection;
-        const isConference = pc ? pc.sfu : false;
-        const hasVideo = this.currentCallSession && this.currentCallSession.cameraEnabled;
-
+        return this.client.getSipSession(keys[keyIndex]);
+    }
+    getCallSession(sipSessionId) {
+        return this.callSessions[sipSessionId];
+    }
+    getSipSessionId(sipSession) {
+        return this.client.getSipSessionId(sipSession);
+    }
+    getSipSessionIdFromCallSession(callSession) {
+        const sipSession = this.findSipSession(callSession);
+        return sipSession ? this.getSipSessionId(sipSession) : null;
+    }
+    _updateCallSession(callSession) {
+        const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
+        if (sipSessionId) {
+            this.callSessions[sipSessionId] = callSession;
+        }
+    }
+    _onMessage(message) {
+        // @ts-ignore
+        if (!message || message.method !== 'MESSAGE') {
+            return;
+        }
+        let body;
         try {
-          // Send reinvite with iceRestart
-          this.sendReinvite(this.currentCallSession, null, isConference, !hasVideo, true);
-        } catch (e) {
-          logger.error('WebRTC reinvite after register, error', { message: e.message, stack: e.stack });
+            // @ts-ignore
+            body = JSON.parse(message.body);
         }
-      }
-    });
-
-    this.client.on(this.client.CONNECTED, () => {
-      logger.info('WebRTC client connected');
-
-      this.stopHeartbeat();
-    });
-
-    this.client.on(this.client.DISCONNECTED, () => {
-      logger.info('WebRTC client disconnected');
-
-      this.eventEmitter.emit(ON_UNREGISTERED);
-
-      // Do not trigger heartbeat if already running
-      if (!this.client.hasHeartbeat()) {
-        this.startHeartbeat();
-      }
-
-      // Tell to send reinvite when reconnecting
-      this.shouldSendReinvite = true;
-    });
-
-    this.client.on(this.client.ON_TRACK, (session, event) => {
-      this.eventEmitter.emit(ON_TRACK, session, event);
-    });
-
-    this.client.on('onVideoInputChange', stream => {
-      this.eventEmitter.emit(ON_VIDEO_INPUT_CHANGE, stream);
-    });
-
-    this.client.on(this.client.MESSAGE, (message         ) => {
-      this._onMessage(message);
-      this.eventEmitter.emit(ON_MESSAGE, message);
-    });
-
-    this.client.on(this.client.ON_NETWORK_STATS, (session, stats, previousStats) => {
-      const callSession = this.getCallSession(this.getSipSessionId(session));
-
-      this.eventEmitter.emit(ON_NETWORK_STATS, callSession, stats, previousStats);
-    });
-
-    // Used when upgrading directly in screenshare mode
-    this.client.on(this.client.ON_SCREEN_SHARING_REINVITE, (sipSession         , response     , desktop         ) => {
-      const sipSessionId = this.getSipSessionId(sipSession);
-      const localStream = this.client.getLocalStream(sipSessionId);
-      const callSession = this.callSessions[sipSessionId];
-
-      logger.info('Updrading directly in screensharing mode', {
-        sipSessionId,
-        tracks: localStream ? localStream.getTracks() : null,
-      });
-
-      this._onScreenSharing(localStream, sipSession, callSession, false, desktop);
-    });
-  }
-
-  // Find a corresponding sipSession from a CallSession
-  findSipSession(callSession              )           {
-    const keys = this.client.getSipSessionIds();
-    const keyIndex = keys.findIndex(sessionId => callSession && callSession.isId(sessionId));
-
-    if (keyIndex === -1) {
-      const currentSipSessionId = this.currentSipSession
-        ? this.getSipSessionId(this.currentSipSession)
-        : this.client.getSipSessionIds()[0];
-      return currentSipSessionId ? this.client.getSipSession(currentSipSessionId) : null;
-    }
-
-    return this.client.getSipSession(keys[keyIndex]);
-  }
-
-  getCallSession(sipSessionId        )               {
-    return this.callSessions[sipSessionId];
-  }
-
-  getSipSessionId(sipSession         ) {
-    return this.client.getSipSessionId(sipSession);
-  }
-
-  getSipSessionIdFromCallSession(callSession             ) {
-    const sipSession = this.findSipSession(callSession);
-
-    return sipSession ? this.getSipSessionId(sipSession) : null;
-  }
-
-  _updateCallSession(callSession             ) {
-    const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
-
-    if (sipSessionId) {
-      this.callSessions[sipSessionId] = callSession;
-    }
-  }
-
-  _onMessage(message         ) {
-    if (!message || message.method !== 'MESSAGE') {
-      return;
-    }
-
-    let body;
-
-    try {
-      body = JSON.parse(message.body);
-    } catch (e) {
-      return;
-    }
-
-    const { type, content } = body;
-
-    switch (type) {
-      case MESSAGE_TYPE_CHAT:
-        this.eventEmitter.emit(ON_CHAT, content);
-        break;
-
-      case MESSAGE_TYPE_SIGNAL: {
-        this.eventEmitter.emit(ON_SIGNAL, content);
-        break;
-      }
-    }
-  }
-
-  _createIncomingCallSession(
-    sipSession         ,
-    cameraEnabled         ,
-    fromSession               ,
-    autoAnswer          = false,
-  )              {
-    return this._createCallSession(sipSession, fromSession, {
-      incoming: true,
-      ringing: true,
-      cameraEnabled,
-      autoAnswer,
-    });
-  }
-
-  _createOutgoingCallSession(
-    sipSession         ,
-    cameraEnabled         ,
-    fromSession              ,
-  )              {
-    return this._createCallSession(sipSession, fromSession, { incoming: false, ringing: true, cameraEnabled });
-  }
-
-  _createAcceptedCallSession(
-    sipSession         ,
-    cameraEnabled          ,
-    fromSession              ,
-  )              {
-    return this._createCallSession(sipSession, fromSession, {
-      cameraEnabled: cameraEnabled !== undefined ? cameraEnabled : false,
-    });
-  }
-
-  _createMutedCallSession(sipSession         , fromSession              )              {
-    return this._createCallSession(sipSession, fromSession, {
-      muted: true,
-    });
-  }
-
-  _createUnmutedCallSession(sipSession         , fromSession              )              {
-    return this._createCallSession(sipSession, fromSession, {
-      muted: false,
-    });
-  }
-
-  _createCameraResumedCallSession(sipSession         , fromSession              )              {
-    return this._createCallSession(sipSession, fromSession, {
-      videoMuted: false,
-    });
-  }
-
-  _createCameraDisabledCallSession(sipSession         , fromSession              )              {
-    return this._createCallSession(sipSession, fromSession, {
-      videoMuted: true,
-    });
-  }
-
-  _createCallSession(sipSession         , fromSession               , extra         = {})              {
-    // eslint-disable-next-line
-    const identity = sipSession ? sipSession.remoteIdentity || sipSession.assertedIdentity : null;
-    const number = identity ? identity.uri._normal.user : null;
-    const { state } = sipSession || {};
-    const sessionId = this.getSipSessionId(sipSession);
-    fromSession = fromSession || this.callSessions[sessionId];
-
-    const callSession = new CallSession({
-      callId: fromSession && fromSession.callId,
-      sipCallId: sessionId,
-      sipStatus: state,
-      displayName: identity ? identity.displayName || number : number,
-      startTime: fromSession ? fromSession.startTime : new Date(),
-      creationTime: fromSession ? fromSession.creationTime : null,
-      answerTime: fromSession ? fromSession.answerTime : null,
-      endTime: fromSession ? fromSession.endTime : null,
-      answered: state === sessionState.SessionState.Established,
-      paused: this.client.isCallHeld(sipSession),
-      isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
-      cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
-      number,
-      ringing: false,
-      muted: fromSession ? fromSession.isMuted() : this.client.isAudioMuted(sipSession),
-      videoMuted: fromSession ? fromSession.isVideoMuted() : false,
-      recording: fromSession ? fromSession.isRecording() : false,
-      recordingPaused: false, // @TODO
-      sipSession,
-      conference: !!fromSession && fromSession.isConference(),
-      ...extra,
-    });
-
-    this.callSessions[callSession.getId()] = callSession;
-
-    return callSession;
-  }
+        catch (e) {
+            return;
+        }
+        const { type, content, } = body;
+        switch (type) {
+            case exports.MESSAGE_TYPE_CHAT:
+                this.eventEmitter.emit(exports.ON_CHAT, content);
+                break;
+            case exports.MESSAGE_TYPE_SIGNAL:
+                {
+                    this.eventEmitter.emit(exports.ON_SIGNAL, content);
+                    break;
+                }
+            default:
+        }
+    }
+    _createIncomingCallSession(sipSession, cameraEnabled, fromSession, autoAnswer = false) {
+        return this._createCallSession(sipSession, fromSession, {
+            incoming: true,
+            ringing: true,
+            cameraEnabled,
+            autoAnswer,
+        });
+    }
+    _createOutgoingCallSession(sipSession, cameraEnabled, fromSession) {
+        return this._createCallSession(sipSession, fromSession, {
+            incoming: false,
+            ringing: true,
+            cameraEnabled,
+        });
+    }
+    _createAcceptedCallSession(sipSession, cameraEnabled, fromSession) {
+        return this._createCallSession(sipSession, fromSession, {
+            cameraEnabled: cameraEnabled !== undefined ? cameraEnabled : false,
+        });
+    }
+    _createMutedCallSession(sipSession, fromSession) {
+        return this._createCallSession(sipSession, fromSession, {
+            muted: true,
+        });
+    }
+    _createUnmutedCallSession(sipSession, fromSession) {
+        return this._createCallSession(sipSession, fromSession, {
+            muted: false,
+        });
+    }
+    _createCameraResumedCallSession(sipSession, fromSession) {
+        return this._createCallSession(sipSession, fromSession, {
+            videoMuted: false,
+        });
+    }
+    _createCameraDisabledCallSession(sipSession, fromSession) {
+        return this._createCallSession(sipSession, fromSession, {
+            videoMuted: true,
+        });
+    }
+    _createCallSession(sipSession, fromSession, extra = {}) {
+        // eslint-disable-next-line
+        const identity = sipSession ? sipSession.remoteIdentity || sipSession.assertedIdentity : null;
+        // @ts-ignore
+        const number = identity ? identity.uri._normal.user : null;
+        const { state, } = sipSession || {};
+        const sessionId = this.getSipSessionId(sipSession);
+        fromSession = fromSession || this.callSessions[sessionId];
+        const callSession = new CallSession_1.default(Object.assign({ callId: fromSession && fromSession.callId, sipCallId: sessionId, sipStatus: state, displayName: identity ? identity.displayName || number : number, 
+            // @ts-ignore: @HEADSUP: startTime should be a number, not a date
+            startTime: fromSession ? fromSession.startTime : new Date(), creationTime: fromSession ? fromSession.creationTime : null, answerTime: fromSession ? fromSession.answerTime : null, endTime: fromSession ? fromSession.endTime : null, answered: state === session_state_1.SessionState.Established, paused: this.client.isCallHeld(sipSession), isCaller: fromSession ? fromSession.isCaller : 'incoming' in extra ? !extra.incoming : false, cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId), number, ringing: false, muted: fromSession ? fromSession.isMuted() : this.client.isAudioMuted(sipSession), videoMuted: fromSession ? fromSession.isVideoMuted() : false, recording: fromSession ? fromSession.isRecording() : false, recordingPaused: false, sipSession, conference: !!fromSession && fromSession.isConference() }, extra));
+        this.callSessions[callSession.getId()] = callSession;
+        return callSession;
+    }
 }
-
-exports.MESSAGE_TYPE_CHAT = MESSAGE_TYPE_CHAT;
-exports.MESSAGE_TYPE_SIGNAL = MESSAGE_TYPE_SIGNAL;
-exports.ON_AUDIO_STREAM = ON_AUDIO_STREAM;
-exports.ON_CALL_ACCEPTED = ON_CALL_ACCEPTED;
-exports.ON_CALL_ANSWERED = ON_CALL_ANSWERED;
-exports.ON_CALL_CANCELED = ON_CALL_CANCELED;
-exports.ON_CALL_ENDED = ON_CALL_ENDED;
-exports.ON_CALL_ENDING = ON_CALL_ENDING;
-exports.ON_CALL_ERROR = ON_CALL_ERROR;
-exports.ON_CALL_FAILED = ON_CALL_FAILED;
-exports.ON_CALL_HELD = ON_CALL_HELD;
-exports.ON_CALL_INCOMING = ON_CALL_INCOMING;
-exports.ON_CALL_MUTED = ON_CALL_MUTED;
-exports.ON_CALL_OUTGOING = ON_CALL_OUTGOING;
-exports.ON_CALL_REJECTED = ON_CALL_REJECTED;
-exports.ON_CALL_RESUMED = ON_CALL_RESUMED;
-exports.ON_CALL_UNHELD = ON_CALL_UNHELD;
-exports.ON_CALL_UNMUTED = ON_CALL_UNMUTED;
-exports.ON_CAMERA_DISABLED = ON_CAMERA_DISABLED;
-exports.ON_CAMERA_RESUMED = ON_CAMERA_RESUMED;
-exports.ON_CHAT = ON_CHAT;
-exports.ON_DISCONNECTED = ON_DISCONNECTED;
-exports.ON_EARLY_MEDIA = ON_EARLY_MEDIA;
-exports.ON_MESSAGE = ON_MESSAGE;
-exports.ON_MESSAGE_TRACK_UPDATED = ON_MESSAGE_TRACK_UPDATED;
-exports.ON_NETWORK_STATS = ON_NETWORK_STATS;
-exports.ON_PLAY_HANGUP_SOUND = ON_PLAY_HANGUP_SOUND;
-exports.ON_PLAY_INBOUND_CALL_SIGNAL_SOUND = ON_PLAY_INBOUND_CALL_SIGNAL_SOUND;
-exports.ON_PLAY_PROGRESS_SOUND = ON_PLAY_PROGRESS_SOUND;
-exports.ON_PLAY_RING_SOUND = ON_PLAY_RING_SOUND;
-exports.ON_PROGRESS = ON_PROGRESS;
-exports.ON_REGISTERED = ON_REGISTERED;
-exports.ON_REINVITE = ON_REINVITE;
-exports.ON_REMOVE_STREAM = ON_REMOVE_STREAM;
-exports.ON_SHARE_SCREEN_ENDED = ON_SHARE_SCREEN_ENDED;
-exports.ON_SHARE_SCREEN_ENDING = ON_SHARE_SCREEN_ENDING;
-exports.ON_SHARE_SCREEN_STARTED = ON_SHARE_SCREEN_STARTED;
-exports.ON_SIGNAL = ON_SIGNAL;
-exports.ON_TERMINATE_SOUND = ON_TERMINATE_SOUND;
-exports.ON_TRACK = ON_TRACK;
-exports.ON_UNREGISTERED = ON_UNREGISTERED;
-exports.ON_USER_AGENT = ON_USER_AGENT;
-exports.ON_VIDEO_INPUT_CHANGE = ON_VIDEO_INPUT_CHANGE;
-exports.ON_VIDEO_STREAM = ON_VIDEO_STREAM;
 exports.default = WebRTCPhone;
-exports.events = events;