@wazo/sdk 0.38.22 → 0.39.0-beta.2

package/src/{web-rtc-client.js → web-rtc-client.ts}

@@ -1,37 +1,32 @@
-// @flow
 /* eslint-disable class-methods-use-this, no-param-reassign, max-classes-per-file, no-underscore-dangle */
+
 /* global window, document, navigator */
 import 'webrtc-adapter';
-import type InviterInviteOptions from 'sip.js/lib/api/inviter-invite-options';
-import type Invitation from 'sip.js/lib/api/invitation';
+import type { InviterInviteOptions } from 'sip.js/lib/api/inviter-invite-options';
 import type { IncomingResponse } from 'sip.js/lib/core/messages/incoming-response';
-import type { Session } from 'sip.js/lib/core/session';
+
 import type { SessionDialog } from 'sip.js/lib/core/dialogs/session-dialog';
 import type { IncomingRequestMessage } from 'sip.js/lib/core/messages/incoming-request-message';
-import type { SessionDescriptionHandlerFactoryOptions }
-  from 'sip.js/lib/platform/web/session-description-handler/session-description-handler-factory-options';
-import type SessionDescriptionHandlerConfiguration
-  from 'sip.js/lib/platform/web/session-description-handler/session-description-handler-configuration';
-import type SessionDescriptionHandler
-  from 'sip.js/lib/platform/web/session-description-handler/session-description-handler';
+import type { SessionDescriptionHandlerFactoryOptions } from 'sip.js/lib/platform/web/session-description-handler/session-description-handler-factory-options';
+import type { SessionDescriptionHandlerConfiguration } from 'sip.js/lib/platform/web/session-description-handler/session-description-handler-configuration';
+import type { SessionDescriptionHandler } from 'sip.js/lib/platform/web/session-description-handler/session-description-handler';
 import { UserAgentState } from 'sip.js/lib/api/user-agent-state';
 import { Parser } from 'sip.js/lib/core/messages/parser';
 import { C } from 'sip.js/lib/core/messages/methods/constants';
 import { URI } from 'sip.js/lib/grammar/uri';
 import { UserAgent } from 'sip.js/lib/api/user-agent';
 import { holdModifier, stripVideo } from 'sip.js/lib/platform/web/modifiers/modifiers';
-import { Registerer } from 'sip.js/lib/api/registerer';
-import { Inviter } from 'sip.js/lib/api/inviter';
+
 import { Messager } from 'sip.js/lib/api/messager';
 import { RegistererState } from 'sip.js/lib/api/registerer-state';
 import { SessionState } from 'sip.js/lib/api/session-state';
 import { TransportState } from 'sip.js/lib/api/transport-state';
-import { defaultPeerConnectionConfiguration }
-  from 'sip.js/lib/platform/web/session-description-handler/peer-connection-configuration-default';
+import { defaultPeerConnectionConfiguration } from 'sip.js/lib/platform/web/session-description-handler/peer-connection-configuration-default';
 import getStats from 'getstats';
 
+import { OutgoingByeRequest, OutgoingInviteRequest, OutgoingRequest } from 'sip.js/lib/core';
+import { Inviter, Invitation, Registerer } from 'sip.js/lib/api';
 import WazoSessionDescriptionHandler, { wazoMediaStreamFactory } from './lib/WazoSessionDescriptionHandler';
-
 import Emitter from './utils/Emitter';
 import ApiClient from './api-client';
 import IssueReporter from './service/IssueReporter';
@@ -40,19 +35,16 @@ import { getVideoDirection, hasAnActiveVideo } from './utils/sdp';
 import { lastIndexOf } from './utils/array';
 
 // We need to replace 0.0.0.0 to 127.0.0.1 in the sdp to avoid MOH during a createOffer.
-export const replaceLocalIpModifier = (description: Object) => Promise.resolve({
-  // description is immutable... so we have to clone it or the `type` attribute won't be returned.
+export const replaceLocalIpModifier = (description: Record<string, any>) => Promise.resolve({ // description is immutable... so we have to clone it or the `type` attribute won't be returned.
   ...JSON.parse(JSON.stringify(description)),
   sdp: description.sdp.replace('c=IN IP4 0.0.0.0', 'c=IN IP4 127.0.0.1'),
 });
-
+const SIP_ID_LENGTH = 36;
 const DEFAULT_ICE_TIMEOUT = 3000;
 const SEND_STATS_DELAY = 5000;
-
 const states = ['STATUS_NULL', 'STATUS_NEW', 'STATUS_CONNECTING', 'STATUS_CONNECTED', 'STATUS_COMPLETED'];
 const logger = IssueReporter ? IssueReporter.loggerFor('webrtc-client') : console;
 const statsLogger = IssueReporter ? IssueReporter.loggerFor('webrtc-stats') : console;
-
 // events
 const REGISTERED = 'registered';
 const UNREGISTERED = 'unregistered';
@@ -71,84 +63,119 @@ const ON_ERROR = 'onError';
 const ON_SCREEN_SHARING_REINVITE = 'onScreenSharingReinvite';
 const ON_NETWORK_STATS = 'onNetworkStats';
 const ON_DISCONNECTED = 'onDisconnected';
-
 export const events = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
 export const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
 export class CanceledCallError extends Error {}
-
 const MAX_REGISTER_TRIES = 5;
-
 type MediaConfig = {
-  audio: Object & boolean,
-  video: Object & boolean,
-  localVideo?: Object & boolean,
+  audio: Record<string, any> | boolean;
+  video?: Record<string, any> | boolean;
+  localVideo?: Record<string, any> | boolean;
 };
-
 type WebRtcConfig = {
-  displayName: string,
-  host: string,
-  port?: number,
-  websocketSip?: ?string,
-  authorizationUser: ?string,
-  password: ?string,
-  uri: string,
-  media: MediaConfig,
-  iceCheckingTimeout: ?number,
-  log?: Object,
-  audioOutputDeviceId?: string,
+  displayName?: string;
+  host: string;
+  port?: number;
+  websocketSip?: string;
+  authorizationUser?: string;
+  password?: string;
+  uri?: string;
+  media?: MediaConfig;
+  iceCheckingTimeout?: number;
+  log?: Record<string, any>;
+  audioOutputDeviceId?: string;
   audioOutputVolume?: number;
-  userAgentString?: string,
-  heartbeatDelay: number,
-  heartbeatTimeout: number,
-  maxHeartbeats: number,
-  skipRegister: boolean,
-};
+  userAgentString?: string;
+  heartbeatDelay?: number;
+  heartbeatTimeout?: number;
+  maxHeartbeats?: number;
+  skipRegister?: boolean;
+}; // @see https://github.com/onsip/SIP.js/blob/master/src/Web/Simple.js
 
-// @see https://github.com/onsip/SIP.js/blob/master/src/Web/Simple.js
 export default class WebRTCClient extends Emitter {
   config: WebRtcConfig;
-  uaConfigOverrides: ?Object;
-  userAgent: UserAgent;
-  registerer: Registerer;
+
+  uaConfigOverrides: Record<string, any> | null | undefined;
+
+  userAgent: UserAgent | null;
+
+  registerer: Registerer | null;
+
   hasAudio: boolean;
-  audio: Object | boolean;
-  audioElements: { [string]: HTMLAudioElement };
-  video: Object | boolean;
-  audioStreams: Object;
-  audioOutputDeviceId: ?string;
+
+  audio: MediaTrackConstraints | boolean | undefined;
+
+  audioElements: Record<string, HTMLAudioElement>;
+
+  video: MediaTrackConstraints | boolean | undefined;
+
+  audioStreams: Record<string, any>;
+
+  audioOutputDeviceId: string | null | undefined;
+
   audioOutputVolume: number;
-  heldSessions: Object;
-  connectionPromise: ?Promise<void>;
-  _boundOnHeartbeat: Function;
+
+  heldSessions: Record<string, any>;
+
+  connectionPromise: Promise<void> | null | undefined;
+
+  _boundOnHeartbeat: (...args: Array<any>) => any;
+
   heartbeat: Heartbeat;
-  heartbeatTimeoutCb: ?Function;
-  heartbeatCb: ?Function;
-  statsIntervals: Object;
-  sipSessions: { [string]: Session };
-  conferences: { [string]: boolean };
+
+  heartbeatTimeoutCb: ((...args: Array<any>) => any) | null | undefined;
+
+  heartbeatCb: ((...args: Array<any>) => any) | null | undefined;
+
+  statsIntervals: Record<string, any>;
+
+  sipSessions: Record<string, Invitation | Inviter>;
+
+  conferences: Record<string, boolean>;
+
   skipRegister: boolean;
-  networkMonitoringInterval: { [string]: IntervalID };
-  sessionNetworkStats: { [string]: Object };
+
+  networkMonitoringInterval: Record<string, any>;
+
+  sessionNetworkStats: Record<string, Record<string, any>>;
+
   forceClosed: boolean;
 
   // sugar
   ON_USER_AGENT: string;
+
   REGISTERED: string;
+
   UNREGISTERED: string;
+
   REGISTRATION_FAILED: string;
+
   INVITE: string;
+
   CONNECTED: string;
+
   DISCONNECTED: string;
+
   TRANSPORT_ERROR: string;
+
   MESSAGE: string;
+
   ACCEPTED: string;
+
   REJECTED: string;
+
   ON_TRACK: string;
+
   ON_REINVITE: string;
+
   ON_ERROR: string;
+
   ON_SCREEN_SHARING_REINVITE: string;
+
   ON_NETWORK_STATS: string;
+
   ON_EARLY_MEDIA: string;
+
   ON_DISCONNECTED: string;
 
   static isAPrivateIp(ip: string): boolean {
@@ -156,22 +183,23 @@ export default class WebRTCClient extends Emitter {
     return regex.exec(ip) == null;
   }
 
-  static getIceServers(ip: string): Array<{ urls: Array<string> }> {
+  static getIceServers(ip: string): Array<{
+    urls: Array<string>;
+  }> {
     if (WebRTCClient.isAPrivateIp(ip)) {
-      return [
-        {
-          urls: ['stun:stun.l.google.com:19302', 'stun:stun4.l.google.com:19302'],
-        },
-      ];
+      return [{
+        urls: ['stun:stun.l.google.com:19302', 'stun:stun4.l.google.com:19302'],
+      }];
     }
+
     return [];
   }
 
-  constructor(config: WebRtcConfig, session: ?Session, uaConfigOverrides: ?Object) {
+  constructor(config: WebRtcConfig, session: Invitation | Inviter | null | undefined, uaConfigOverrides: Record<string, any> | null | undefined = undefined) {
     super();
     this.uaConfigOverrides = uaConfigOverrides;
     this.config = config;
-    this.skipRegister = config.skipRegister;
+    this.skipRegister = config.skipRegister as boolean;
 
     this._buildConfig(config, session).then((newConfig: WebRtcConfig) => {
       this.config = newConfig;
@@ -183,7 +211,10 @@ export default class WebRTCClient extends Emitter {
 
     if (config.media) {
       this.configureMedia(config.media);
-      this.setMediaConstraints({ audio: config.media.audio, video: config.media.video });
+      this.setMediaConstraints({
+        audio: config.media.audio,
+        video: config.media.video,
+      });
     }
 
     this.heldSessions = {};
@@ -194,12 +225,10 @@ export default class WebRTCClient extends Emitter {
     this.networkMonitoringInterval = {};
     this.sessionNetworkStats = {};
     this.forceClosed = false;
-
     this._boundOnHeartbeat = this._onHeartbeat.bind(this);
     this.heartbeat = new Heartbeat(config.heartbeatDelay, config.heartbeatTimeout, config.maxHeartbeats);
     this.heartbeat.setSendHeartbeat(this.pingServer.bind(this));
     this.heartbeat.setOnHeartbeatTimeout(this._onHeartbeatTimeout.bind(this));
-
     // sugar
     this.REGISTERED = REGISTERED;
     this.UNREGISTERED = UNREGISTERED;
@@ -231,10 +260,12 @@ export default class WebRTCClient extends Emitter {
     this.audio = media.audio;
   }
 
-  createUserAgent(configOverrides: ?Object): UserAgent {
+  createUserAgent(configOverrides: Record<string, any> | null | undefined): UserAgent {
     const webRTCConfiguration = this._createWebRTCConfiguration(configOverrides);
-    logger.info('sdk webrtc, creating UA', { webRTCConfiguration });
 
+    logger.info('sdk webrtc, creating UA', {
+      webRTCConfiguration,
+    });
     webRTCConfiguration.delegate = {
       onConnect: this.onConnect.bind(this),
       onDisconnect: this.onDisconnect.bind(this),
@@ -242,25 +273,26 @@ export default class WebRTCClient extends Emitter {
         logger.info('sdk webrtc on invite', {
           method: 'delegate.onInvite',
           id: invitation.id,
+          // @ts-ignore
           remoteURI: invitation.remoteURI,
         });
+
         this._setupSession(invitation);
-        const shouldAutoAnswer = !!invitation.request.getHeader('alert-info');
 
+        const shouldAutoAnswer = !!invitation.request.getHeader('alert-info');
         this.eventEmitter.emit(INVITE, invitation, this.sessionWantsToDoVideo(invitation), shouldAutoAnswer);
       },
     };
+    const ua: any = new UserAgent(webRTCConfiguration);
 
-    const ua = new UserAgent(webRTCConfiguration);
     if (ua.transport && ua.transport.connectPromise) {
-      ua.transport.connectPromise.catch((e => {
+      ua.transport.connectPromise.catch((e: any) => {
         logger.warn('Transport connect error', e);
-      }));
+      });
     }
 
     ua.transport.onMessage = (rawMessage: string) => {
       const message = Parser.parseMessage(rawMessage, ua.transport.logger);
-
       // We have to re-sent the message to the UA ...
       ua.onTransportMessage(rawMessage);
       // And now do what we want with the message
@@ -268,7 +300,9 @@ export default class WebRTCClient extends Emitter {
 
       if (message && message.method === C.MESSAGE) {
         // We have to manually reply to MESSAGE with a 200 OK or Asterisk will hangup.
-        ua.userAgentCore.replyStateless(message, { statusCode: 200 });
+        ua.userAgentCore.replyStateless(message, {
+          statusCode: 200,
+        });
       }
     };
 
@@ -276,51 +310,65 @@ export default class WebRTCClient extends Emitter {
   }
 
   isConnected(): boolean {
-    return this.userAgent && this.userAgent.isConnected();
+    return Boolean(this.userAgent?.isConnected());
   }
 
   isConnecting(): boolean {
-    return this.userAgent && this.userAgent.transport && this.userAgent.transport.state === TransportState.Connecting;
+    return this.userAgent?.transport?.state === TransportState.Connecting;
   }
 
   isRegistered(): boolean {
-    return this.registerer && this.registerer.state === RegistererState.Registered;
+    return Boolean(this.registerer && this.registerer.state === RegistererState.Registered);
   }
 
   onConnect() {
-    logger.info('sdk webrtc connected', { method: 'delegate.onConnect' });
+    logger.info('sdk webrtc connected', {
+      method: 'delegate.onConnect',
+    });
     this.eventEmitter.emit(CONNECTED);
-    if (!this.isRegistered() && this.registerer && this.registerer.waiting) {
+
+    // @ts-ignore
+    if (!this.isRegistered() && this.registerer?.waiting) {
+      // @ts-ignore
       this.registerer.waitingToggle(false);
     }
+
     return this.register();
   }
 
   async onDisconnect(error?: Error) {
-    logger.info('sdk webrtc disconnected', { method: 'delegate.onConnect', error });
+    logger.info('sdk webrtc disconnected', {
+      method: 'delegate.onConnect',
+      error,
+    });
     this.connectionPromise = null;
     // The UA will attempt to reconnect automatically when an error occurred
     this.eventEmitter.emit(DISCONNECTED, error);
+
     if (this.isRegistered()) {
       await this.unregister();
-      if (this.registerer && this.registerer.waiting) {
+
+      // @ts-ignore
+      if (this.registerer?.waiting) {
+        // @ts-ignore
         this.registerer.waitingToggle(false);
       }
+
       this.eventEmitter.emit(UNREGISTERED);
     }
   }
 
-  register(tries: number = 0): Promise<any> {
+  async register(tries = 0): Promise<void> {
     const logInfo = {
       userAgent: !!this.userAgent,
       registered: this.isRegistered(),
       connectionPromise: !!this.connectionPromise,
       registerer: !!this.registerer,
+      // @ts-ignore
       waiting: this.registerer && this.registerer.waiting,
       tries,
       skipRegister: this.skipRegister,
     };
-
     this.forceClosed = false;
 
     if (this.skipRegister) {
@@ -334,17 +382,21 @@ export default class WebRTCClient extends Emitter {
       logger.info('sdk webrtc recreating User Agent');
       this.userAgent = this.createUserAgent(this.uaConfigOverrides);
     }
+
     if (!this.userAgent || this.isRegistered()) {
       logger.info('sdk webrtc registering aborted, already registered or no UA can be created');
       return Promise.resolve();
     }
 
-    if (this.connectionPromise || (this.registerer && this.registerer.waiting)) {
+    // @ts-ignore
+    if (this.connectionPromise || this.registerer?.waiting) {
       logger.info('sdk webrtc registering aborted due to a registration in progress.');
       return Promise.resolve();
     }
 
-    const registerOptions = this._isWeb() ? {} : { extraContactHeaderParams: ['mobility=mobile'] };
+    const registerOptions = this._isWeb() ? {} : {
+      extraContactHeaderParams: ['mobility=mobile'],
+    };
 
     const onRegisterFailed = () => {
       logger.info('sdk webrtc registering failed', {
@@ -352,17 +404,23 @@ export default class WebRTCClient extends Emitter {
         registerer: !!this.registerer,
         forceClosed: this.forceClosed,
       });
+
       if (this.forceClosed) {
         return;
       }
 
       this.connectionPromise = null;
+
+      // @ts-ignore
       if (this.registerer && this.registerer.waiting) {
+        // @ts-ignore
         this.registerer.waitingToggle(false);
       }
 
       if (tries <= MAX_REGISTER_TRIES) {
-        logger.info('sdk webrtc registering, retrying...', { tries });
+        logger.info('sdk webrtc registering, retrying...', {
+          tries,
+        });
         setTimeout(() => this.register(tries + 1), 300);
       }
     };
@@ -375,36 +433,39 @@ export default class WebRTCClient extends Emitter {
         this.userAgent = this.createUserAgent(this.uaConfigOverrides);
       }
 
-      logger.info('sdk webrtc registering, transport connected', { registerOptions, ua: !!this.userAgent });
+      logger.info('sdk webrtc registering, transport connected', {
+        registerOptions,
+        ua: !!this.userAgent,
+      });
       this.registerer = new Registerer(this.userAgent, registerOptions);
       this.connectionPromise = null;
+
       this._monkeyPatchRegisterer(this.registerer);
 
       // Bind registerer events
       this.registerer.stateChange.addListener(newState => {
-        logger.info('sdk webrtc registering, state changed', { newState });
+        logger.info('sdk webrtc registering, state changed', {
+          newState,
+        });
 
-        if (newState === RegistererState.Registered && this.registerer
-          && this.registerer.state === RegistererState.Registered) {
+        if (newState === RegistererState.Registered && this.registerer && this.registerer.state === RegistererState.Registered) {
           this.eventEmitter.emit(REGISTERED);
         } else if (newState === RegistererState.Unregistered) {
           this.eventEmitter.emit(UNREGISTERED);
         }
       });
-
       const options = {
         requestDelegate: {
-          onReject: response => {
-            logger.error('sdk webrtc registering, rejected', { response });
-
+          onReject: (response: any) => {
+            logger.error('sdk webrtc registering, rejected', {
+              response,
+            });
             onRegisterFailed();
           },
         },
       };
-
-      return this.registerer.register(options).catch((e) => {
+      return this.registerer.register(options).catch(e => {
         logger.error('sdk webrtc registering, error', e);
-
         this.eventEmitter.emit(REGISTRATION_FAILED);
         return e;
       });
@@ -415,30 +476,42 @@ export default class WebRTCClient extends Emitter {
   }
 
   // Monkey patching sip.js to avoid issues during register.onReject
-  _monkeyPatchRegisterer(registerer: ?Registerer) {
+  _monkeyPatchRegisterer(registerer: Registerer | null | undefined): void {
     if (!registerer) {
       return;
     }
+
+    // @ts-ignore
     const oldWaitingToggle = registerer.waitingToggle.bind(registerer);
+    // @ts-ignore
     const oldUnregistered = registerer.unregistered.bind(registerer);
 
+    // @ts-ignore
     registerer.waitingToggle = (waiting: boolean) => {
+      // @ts-ignore
       if (!registerer || registerer.waiting === waiting) {
         return;
       }
+
       oldWaitingToggle(waiting);
     };
 
+    // @ts-ignore
     registerer.unregistered = () => {
       if (!registerer || registerer.state === RegistererState.Terminated) {
         return;
       }
+
       oldUnregistered();
     };
   }
 
-  unregister() {
-    logger.info('sdk webrtc unregistering..', { userAgent: !!this.userAgent, registerer: !!this.registerer });
+  async unregister(): Promise<void> {
+    logger.info('sdk webrtc unregistering..', {
+      userAgent: !!this.userAgent,
+      registerer: !!this.registerer,
+    });
+
     if (!this.registerer) {
       return Promise.resolve();
     }
@@ -446,21 +519,27 @@ export default class WebRTCClient extends Emitter {
     try {
       return this.registerer.unregister().then(() => {
         logger.info('sdk webrtc unregistered');
+
         this._cleanupRegister();
       }).catch(e => {
         logger.error('sdk webrtc unregistering, promise error', e);
+
         this._cleanupRegister();
       });
-    } catch (e) {
+    } catch (e: any) {
       logger.error('sdk webrtc unregistering, error', e);
+
       // Avoid issue with `undefined is not an object (evaluating 'new.target.prototype')` when triggering a new
       // error when the registerer is in a bad state
       this._cleanupRegister();
     }
   }
 
-  stop(): Promise<any> {
-    logger.info('sdk webrtc stop', { userAgent: !!this.userAgent });
+  stop(): Promise<void> {
+    logger.info('sdk webrtc stop', {
+      userAgent: !!this.userAgent,
+    });
+
     if (!this.userAgent) {
       return Promise.resolve();
     }
@@ -468,13 +547,21 @@ export default class WebRTCClient extends Emitter {
     return this.userAgent.stop().then(() => {
       return this._cleanupRegister();
     }).catch(e => {
-      logger.warn('sdk webrtc stop, error', { message: e.message, stack: e.stack });
+      logger.warn('sdk webrtc stop, error', {
+        message: e.message,
+        stack: e.stack,
+      });
     });
   }
 
-  call(number: string, enableVideo?: boolean, audioOnly: boolean = false, conference: boolean = false): Session {
-    logger.info('sdk webrtc creating call', { number, enableVideo, audioOnly, conference });
-    const inviterOptions: Object = {
+  call(number: string, enableVideo?: boolean, audioOnly = false, conference = false): Inviter | Invitation {
+    logger.info('sdk webrtc creating call', {
+      number,
+      enableVideo,
+      audioOnly,
+      conference,
+    });
+    const inviterOptions: Record<string, any> = {
       sessionDescriptionHandlerOptionsReInvite: {
         conference,
         audioOnly,
@@ -486,11 +573,20 @@ export default class WebRTCClient extends Emitter {
       inviterOptions.sessionDescriptionHandlerModifiersReInvite = [stripVideo];
     }
 
-    const session = new Inviter(this.userAgent, this._makeURI(number), inviterOptions);
+    const uri = this._makeURI(number);
+    let session: any;
 
-    this.storeSipSession(session);
+    if (uri) {
+      session = this.userAgent ? new Inviter(this.userAgent, uri, inviterOptions) : null;
+    } else {
+      logger.error('Null URI');
+    }
 
-    this._setupSession(session);
+    if (session) {
+      this.storeSipSession(session);
+
+      this._setupSession(session);
+    }
 
     if (conference) {
       this.conferences[this.getSipSessionId(session)] = true;
@@ -499,30 +595,42 @@ export default class WebRTCClient extends Emitter {
     const inviteOptions: InviterInviteOptions = {
       requestDelegate: {
         onAccept: (response: IncomingResponse) => {
-          if (session.sessionDescriptionHandler.peerConnection) {
+          // @ts-ignore
+          if (session.sessionDescriptionHandler?.peerConnection) {
+            // @ts-ignore
             session.sessionDescriptionHandler.peerConnection.sfu = conference;
           }
+          // @ts-ignore
           this._onAccepted(session, response.session, true);
         },
-        onProgress: (payload) => {
+        onProgress: payload => {
           if (payload.message.statusCode === 183) {
+            // @ts-ignore
             this._onEarlyProgress(payload.session);
           }
         },
         onReject: (response: IncomingResponse) => {
-          logger.info('on call rejected', { id: session.id, fromTag: session.fromTag });
+          logger.info('on call rejected', {
+            id: session.id,
+            // @ts-ignore
+            fromTag: session.fromTag,
+          });
+
           this._stopSendingStats(session);
-          this.stopNetworkMonitoring(session);
 
+          this.stopNetworkMonitoring(session);
           this.eventEmitter.emit(REJECTED, session, response);
         },
       },
       sessionDescriptionHandlerOptions: this.getMediaConfiguration(enableVideo || false, conference),
     };
 
-    inviteOptions.sessionDescriptionHandlerOptions.audioOnly = audioOnly;
-
+    if (inviteOptions.sessionDescriptionHandlerOptions) {
+      // @ts-ignore
+      inviteOptions.sessionDescriptionHandlerOptions.audioOnly = audioOnly;
+    }
     inviteOptions.sessionDescriptionHandlerModifiers = [replaceLocalIpModifier];
+
     if (audioOnly) {
       inviteOptions.sessionDescriptionHandlerModifiers.push(stripVideo);
     }
@@ -532,38 +640,50 @@ export default class WebRTCClient extends Emitter {
     return session;
   }
 
-  answer(session: Invitation, enableVideo?: boolean) {
-    logger.info('sdk webrtc answer call', { id: session.id, enableVideo });
+  answer(session: Invitation, enableVideo?: boolean): Promise<void> {
+    logger.info('sdk webrtc answer call', {
+      id: session.id,
+      enableVideo,
+    });
 
     if (!session || !session.accept) {
       const error = 'No session to answer, or not an invitation';
       logger.warn(error);
       return Promise.reject(new Error(error));
     }
+
     const options = {
       sessionDescriptionHandlerOptions: this.getMediaConfiguration(enableVideo || false),
     };
-
     return session.accept(options).then(() => {
+      // @ts-ignore
       if (session.isCanceled) {
         const message = 'accepted a canceled session (or was canceled during the accept phase).';
-        logger.error(message, { id: session.id });
+        logger.error(message, {
+          id: session.id,
+        });
         this.onCallEnded(session);
-
         throw new CanceledCallError(message);
       }
+
       logger.info('sdk webrtc answer, accepted.');
+
       this._onAccepted(session);
     }).catch(e => {
       logger.error(`answer call error for ${session ? session.id : 'n/a'}`, e);
-
       throw e;
     });
   }
 
-  hangup(session: Session) {
-    const { state, id } = session;
-    logger.info('sdk webrtc hangup call', { id, state });
+  async hangup(session: Invitation | Inviter): Promise<OutgoingByeRequest | null> {
+    const {
+      state,
+      id,
+    }: any = session;
+    logger.info('sdk webrtc hangup call', {
+      id,
+      state,
+    });
 
     try {
       this._stopSendingStats(session);
@@ -571,15 +691,17 @@ export default class WebRTCClient extends Emitter {
       this._cleanupMedia(session);
 
       delete this.sipSessions[this.getSipSessionId(session)];
-
       // Check if Invitation or Inviter (Invitation = incoming call)
       const isInviter = session instanceof Inviter;
-      const cancel = () => session.cancel();
-      const reject = () => session.reject();
+
+      const cancel = () => (<Inviter>session).cancel();
+
+      const reject = () => (<Invitation>session).reject();
+
       const bye = () => session.bye && session.bye();
 
       // @see github.com/onsip/SIP.js/blob/f11dfd584bc9788ccfc94e03034020672b738975/src/platform/web/simple-user/simple-user.ts#L1004
-      const actions = {
+      const actions: Record<string, any> = {
         [SessionState.Initial]: isInviter ? cancel : reject,
         [SessionState.Establishing]: isInviter ? cancel : reject,
         [SessionState.Established]: bye,
@@ -587,19 +709,21 @@ export default class WebRTCClient extends Emitter {
 
       // Handle different session status
       if (actions[state]) {
-        return actions[state]();
+        return await actions[state]();
       }
 
-      return bye();
-    } catch (error) {
+      return await bye();
+    } catch (error: any) {
       logger.warn('sdk webrtc hangup, error', error);
     }
 
-    return null;
+    return Promise.resolve(null);
   }
 
-  async getStats(session: Session): ?Object {
-    const pc = session.sessionDescriptionHandler.peerConnection;
+  async getStats(session: Invitation | Inviter): Promise<RTCStatsReport | null> {
+    // @ts-ignore
+    const pc: RTCPeerConnection = session.sessionDescriptionHandler.peerConnection;
+
     if (!pc) {
       return null;
     }
@@ -608,58 +732,75 @@ export default class WebRTCClient extends Emitter {
   }
 
   // Fetch and emit an event at `interval` with session network stats
-  startNetworkMonitoring(session: Session, interval: number = 1000): ?Object {
+  startNetworkMonitoring(session: Inviter | Invitation, interval = 1000): void {
     const sessionId = this.getSipSessionId(session);
-    logger.info('starting network inspection', { id: sessionId });
-
+    logger.info('starting network inspection', {
+      id: sessionId,
+    });
     this.sessionNetworkStats[sessionId] = [];
-
     this.networkMonitoringInterval[sessionId] = setInterval(() => this._fetchNetworkStats(sessionId), interval);
   }
 
-  stopNetworkMonitoring(session: Session): ?Object {
+  stopNetworkMonitoring(session: Inviter | Invitation): void {
     const sessionId = this.getSipSessionId(session);
-    const exists = sessionId in this.networkMonitoringInterval;
-    logger.info('stopping network inspection', { id: sessionId, exists });
+    const exists = (sessionId in this.networkMonitoringInterval);
+    logger.info('stopping network inspection', {
+      id: sessionId,
+      exists,
+    });
 
     if (exists) {
-      clearInterval(this.networkMonitoringInterval[sessionId]);
+      clearInterval(this.networkMonitoringInterval[sessionId] as string);
       delete this.networkMonitoringInterval[sessionId];
       delete this.sessionNetworkStats[sessionId];
     }
   }
 
-  reject(session: Inviter) {
-    logger.info('sdk webrtc reject call', { id: session.id });
+  async reject(session: Invitation | Inviter): Promise<void> {
+    logger.info('sdk webrtc reject call', {
+      id: session.id,
+    });
 
     try {
-      return session.reject ? session.reject() : session.cancel();
-    } catch (e) {
+      if (session instanceof Invitation) {
+        return session.reject();
+      }
+
+      if (session instanceof Inviter) {
+        return session.cancel();
+      }
+    } catch (e: any) {
       logger.warn('Error when rejecting call', e.message, e.stack);
     }
   }
 
-  async close(force: boolean = false) {
-    logger.info('sdk webrtc closing client', { userAgent: !!this.userAgent, force });
+  async close(force = false): Promise<void> {
+    logger.info('sdk webrtc closing client', {
+      userAgent: !!this.userAgent,
+      force,
+    });
     this.forceClosed = force;
+
     this._cleanupMedia();
-    this.connectionPromise = null;
 
-    (Object.values(this.audioElements): any).forEach((audioElement: HTMLAudioElement) => {
+    this.connectionPromise = null;
+    (Object.values(this.audioElements) as any).forEach((audioElement: HTMLAudioElement) => {
       // eslint-disable-next-line
       audioElement.srcObject = null;
       audioElement.pause();
     });
-
     this.audioElements = {};
+
     if (!this.userAgent) {
-      return null;
+      return;
     }
 
     this.stopHeartbeat();
-    this.userAgent.delegate = null;
+    if (this.userAgent) {
+      this.userAgent.delegate = undefined;
+    }
+    // @ts-ignore: this is odd, may want to validate
     this.userAgent.stateChange.removeAllListeners();
-
     await this._disconnectTransport(force);
 
     this._cleanupRegister();
@@ -667,42 +808,48 @@ export default class WebRTCClient extends Emitter {
     try {
       // Don't wait here, It can take ~30s to stop ...
       this.userAgent.stop().catch(console.error);
-    } catch (_) {
-      // Avoid to raise exception when trying to close with hanged-up sessions remaining
+    } catch (_) { // Avoid to raise exception when trying to close with hanged-up sessions remaining
       // eg: "INVITE not rejectable in state Completed"
     }
+
     this.userAgent = null;
     logger.info('sdk webrtc client closed');
   }
 
-  getNumber(session: Inviter): ?String {
+  getNumber(session: Inviter): string | null | undefined {
     if (!session) {
       return null;
     }
 
-    // eslint-disable-next-line
+    // @ts-ignore
     return session.remoteIdentity.uri._normal.user;
   }
 
-  mute(session: Inviter) {
-    logger.info('sdk webrtc mute', { id: session.id });
+  mute(session: Inviter | Invitation): void {
+    logger.info('sdk webrtc mute', {
+      id: session.id,
+    });
 
     this._toggleAudio(session, true);
   }
 
-  unmute(session: Inviter) {
-    logger.info('sdk webrtc unmute', { id: session.id });
+  unmute(session: Inviter | Invitation): void {
+    logger.info('sdk webrtc unmute', {
+      id: session.id,
+    });
 
     this._toggleAudio(session, false);
   }
 
-  isAudioMuted(session: Inviter): boolean {
+  isAudioMuted(session: Inviter | Invitation): boolean {
     if (!session || !session.sessionDescriptionHandler) {
       return false;
     }
 
     let muted = true;
+    // @ts-ignore
     const pc = session.sessionDescriptionHandler.peerConnection;
+
     if (!pc) {
       return false;
     }
@@ -711,7 +858,8 @@ export default class WebRTCClient extends Emitter {
       if (!pc.getSenders().length) {
         return false;
       }
-      pc.getSenders().forEach(sender => {
+
+      pc.getSenders().forEach((sender: any) => {
         if (sender && sender.track && sender.track.kind === 'audio') {
           muted = muted && !sender.track.enabled;
         }
@@ -720,7 +868,8 @@ export default class WebRTCClient extends Emitter {
       if (!pc.getLocalStreams().length) {
         return false;
       }
-      pc.getLocalStreams().forEach(stream => {
+
+      pc.getLocalStreams().forEach((stream: MediaStream) => {
         stream.getAudioTracks().forEach(track => {
           muted = muted && !track.enabled;
         });
@@ -730,25 +879,29 @@ export default class WebRTCClient extends Emitter {
     return muted;
   }
 
-  toggleCameraOn(session: Inviter) {
-    logger.info('sdk webrtc toggle camera on', { id: session.id });
+  toggleCameraOn(session: Inviter | Invitation): void {
+    logger.info('sdk webrtc toggle camera on', {
+      id: session.id,
+    });
 
     this._toggleVideo(session, false);
   }
 
-  toggleCameraOff(session: Inviter) {
-    logger.info('sdk webrtc toggle camera off', { id: session.id });
+  toggleCameraOff(session: Inviter | Invitation): void {
+    logger.info('sdk webrtc toggle camera off', {
+      id: session.id,
+    });
 
     this._toggleVideo(session, true);
   }
 
-  hold(session: Inviter, isConference: boolean = false, hadVideo: boolean = false) {
+  hold(session: Inviter | Invitation, isConference = false, hadVideo = false): Promise<OutgoingInviteRequest | void> {
     const sessionId = this.getSipSessionId(session);
     const hasVideo = hadVideo || this.hasLocalVideo(sessionId);
-
     logger.info('sdk webrtc hold', {
       sessionId,
       keys: Object.keys(this.heldSessions),
+      // @ts-ignore
       pendingReinvite: !!session.pendingReinvite,
       isConference,
       hasVideo,
@@ -757,9 +910,12 @@ export default class WebRTCClient extends Emitter {
     if (sessionId in this.heldSessions) {
       return Promise.resolve();
     }
+
+    // @ts-ignore
     if (session.pendingReinvite) {
       return Promise.resolve();
     }
+
     this.heldSessions[sessionId] = {
       hasVideo,
       isConference,
@@ -770,14 +926,16 @@ export default class WebRTCClient extends Emitter {
     }
 
     session.sessionDescriptionHandlerOptionsReInvite = {
+      // @ts-ignore
       hold: true,
       conference: isConference,
     };
-
     const options = this.getMediaConfiguration(false, isConference);
+
     if (!this._isWeb()) {
       options.sessionDescriptionHandlerModifiers = [holdModifier];
     }
+
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
       hold: true,
@@ -786,6 +944,7 @@ export default class WebRTCClient extends Emitter {
 
     // Avoid sdh to create a new stream
     if (session.sessionDescriptionHandler) {
+      // @ts-ignore
       session.sessionDescriptionHandler.localMediaStreamConstraints = options.constraints;
     }
 
@@ -793,18 +952,19 @@ export default class WebRTCClient extends Emitter {
     return session.invite(options);
   }
 
-  unhold(session: Inviter, isConference: boolean = false) {
+  unhold(session: Inviter | Invitation, isConference = false): Promise<OutgoingInviteRequest | void> {
     const sessionId = this.getSipSessionId(session);
     const hasVideo = sessionId in this.heldSessions && this.heldSessions[sessionId].hasVideo;
-
     logger.info('sdk webrtc unhold', {
       sessionId,
       keys: Object.keys(this.heldSessions),
+      // @ts-ignore
       pendingReinvite: !!session.pendingReinvite,
       isConference,
       hasVideo,
     });
 
+    // @ts-ignore
     if (session.pendingReinvite) {
       return Promise.resolve();
     }
@@ -815,111 +975,117 @@ export default class WebRTCClient extends Emitter {
 
     delete this.heldSessions[this.getSipSessionId(session)];
     session.sessionDescriptionHandlerOptionsReInvite = {
+      // @ts-ignore
       hold: false,
       conference: isConference,
     };
-
     const options = this.getMediaConfiguration(false, isConference);
+
     if (!this._isWeb()) {
       // We should sent an empty `sessionDescriptionHandlerModifiers` or sip.js will take the last sent modifiers
       // (eg: holdModifier)
       options.sessionDescriptionHandlerModifiers = [];
     }
+
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
       hold: false,
       conference: isConference,
     };
-
     // Send re-INVITE
     return session.invite(options);
   }
 
   // Returns true if a re-INVITE is required
-  async upgradeToVideo(session: Session, constraints: ?Object, isConference: boolean): Promise<?Object> {
+  async upgradeToVideo(session: Inviter | Invitation, constraints: Record<string, any>, isConference: boolean): Promise<MediaStream | undefined> {
+    // @ts-ignore
     const pc = session.sessionDescriptionHandler.peerConnection;
-
     // Check if a video sender already exists
     let videoSender;
+
     if (isConference) {
       // We search for the last transceiver without `video-` in the mid (video- means remote transceiver)
       const transceivers = pc.getTransceivers();
-      const transceiverIdx = lastIndexOf(transceivers, transceiver =>
-        transceiver.sender.track === null && transceiver.mid && transceiver.mid.indexOf('video') === -1);
-
+      const transceiverIdx = lastIndexOf(transceivers, transceiver => transceiver.sender.track === null && transceiver.mid && transceiver.mid.indexOf('video') === -1);
       videoSender = transceiverIdx !== -1 ? transceivers[transceiverIdx].sender : null;
     } else {
-      videoSender = pc && pc.getSenders && pc.getSenders().find(sender => sender.track === null);
+      videoSender = pc && pc.getSenders && pc.getSenders().find((sender: any) => sender.track === null);
     }
 
     if (!videoSender) {
       // When no video sender found, it means that we're in the first video upgrade in 1:1
-      return null;
+      return;
     }
 
     // Reuse bidirectional video stream
     const newStream = await this.getStreamFromConstraints(constraints);
+
     if (!newStream) {
       console.warn(`Can't create media stream with: ${JSON.stringify(constraints || {})}`);
-      return false;
+      return;
     }
 
     // Add previous local audio track
     if (constraints && !constraints.audio) {
-      const localVideoStream = session.sessionDescriptionHandler.localMediaStream;
+      // @ts-ignore
+      const localVideoStream: MediaStream = session.sessionDescriptionHandler.localMediaStream;
       const localAudioTrack = localVideoStream.getTracks().find(track => track.kind === 'audio');
+
       if (localAudioTrack) {
         newStream.addTrack(localAudioTrack);
       }
     }
 
     const videoTrack = newStream.getVideoTracks()[0];
+
     if (videoTrack) {
       videoSender.replaceTrack(videoTrack);
     }
 
     this.setLocalMediaStream(this.getSipSessionId(session), newStream);
-
     return newStream;
   }
 
-  downgradeToAudio(session: Session) {
+  downgradeToAudio(session: Invitation | Inviter): void {
     // Release local video stream when downgrading to audio
+    // @ts-ignore
     const localStream = session.sessionDescriptionHandler.localMediaStream;
+    // @ts-ignore
     const pc = session.sessionDescriptionHandler.peerConnection;
     const videoTracks = localStream.getVideoTracks();
 
     // Remove video senders
     if (pc.getSenders) {
-      pc.getSenders().filter(sender => sender.track && sender.track.kind === 'video').forEach(videoSender => {
+      pc.getSenders().filter((sender: any) => sender.track && sender.track.kind === 'video').forEach((videoSender: any) => {
         videoSender.replaceTrack(null);
       });
     }
 
-    videoTracks.forEach(videoTrack => {
+    videoTracks.forEach((videoTrack: MediaStreamTrack) => {
       videoTrack.enabled = false;
       videoTrack.stop();
       localStream.removeTrack(videoTrack);
     });
   }
 
-  async getStreamFromConstraints(constraints: Object, conference: boolean = false): Promise<?MediaStream> {
+  async getStreamFromConstraints(constraints: Record<string, any>, conference = false): Promise<MediaStream | null | undefined> {
     const video = constraints && constraints.video;
-    // $FlowFixMe
-    const { constraints: newConstraints } = this.getMediaConfiguration(video, conference, constraints);
-
+    const {
+      constraints: newConstraints,
+    } = this.getMediaConfiguration(video, conference, constraints);
     let newStream;
+
     try {
       newStream = await wazoMediaStreamFactory(newConstraints);
-    } catch (e) {
-      // Nothing to do when the user cancel the screensharing prompt
+    } catch (e: any) { // Nothing to do when the user cancel the screensharing prompt
     }
+
     if (!newStream) {
       return null;
     }
-    // $FlowFixMe
-    newStream.local = true;
 
+    // @ts-ignore
+    newStream.local = true;
     return newStream;
   }
 
@@ -927,26 +1093,27 @@ export default class WebRTCClient extends Emitter {
     return this.heldSessions[sessionId];
   }
 
-  isCallHeld(session: Inviter) {
+  isCallHeld(session: Inviter | Invitation): boolean {
     return this.getSipSessionId(session) in this.heldSessions;
   }
 
-  isVideoRemotelyHeld(sessionId: string) {
+  isVideoRemotelyHeld(sessionId: string): boolean {
     const pc = this.getPeerConnection(sessionId);
     const sdp = pc && pc.remoteDescription ? pc.remoteDescription.sdp : null;
+
     if (!sdp) {
       return false;
     }
 
     const videoDirection = getVideoDirection(sdp);
-
     return videoDirection === 'sendonly';
   }
 
-  sendDTMF(session: Inviter, tone: string) {
+  sendDTMF(session: Inviter | Invitation, tone: string): boolean {
     if (!session.sessionDescriptionHandler) {
-      return;
+      return false;
     }
+
     logger.info('Sending DTMF', {
       id: this.getSipSessionId(session),
       tone,
@@ -954,20 +1121,22 @@ export default class WebRTCClient extends Emitter {
     return session.sessionDescriptionHandler.sendDtmf(tone);
   }
 
-  message(destination: string, message: string) {
-    const messager = new Messager(this.userAgent, this._makeURI(destination), message);
-
-    return messager.message();
+  message(destination: string, message: string): void {
+    const uri = this._makeURI(destination);
+    if (!this.userAgent || !uri) {
+      logger.warn('Null value on message', { uri, userAgent: this.userAgent });
+      return;
+    }
+    const messager = new Messager(this.userAgent, uri, message);
+    messager.message();
   }
 
-  transfer(session: Inviter, target: string) {
+  transfer(session: Inviter | Invitation, target: string): void {
     this.hold(session);
-
     logger.info('Transfering a session', {
       id: this.getSipSessionId(session),
       target,
     });
-
     const options = {
       requestDelegate: {
         onAccept: () => {
@@ -975,23 +1144,29 @@ export default class WebRTCClient extends Emitter {
         },
       },
     };
-
     setTimeout(() => {
-      session.refer(this._makeURI(target), options);
+      const uri = this._makeURI(target);
+      if (!uri) {
+        logger.warn('transfer timeout: null URI');
+        return;
+      }
+      session.refer(uri, options);
     }, 50);
   }
 
   // check https://sipjs.com/api/0.12.0/refer/referClientContext/
-  atxfer(session: Inviter) {
+  atxfer(session: Inviter | Invitation): Record<string, any> {
     this.hold(session);
-
-    logger.info('webrtc transfer started', { id: this.getSipSessionId(session) });
-
-    const result: Object = {
+    logger.info('webrtc transfer started', {
+      id: this.getSipSessionId(session),
+    });
+    const result: Record<string, any> = {
       newSession: null,
       init: async (target: string) => {
-        logger.info('webrtc transfer initialized', { id: this.getSipSessionId(session), target });
-
+        logger.info('webrtc transfer initialized', {
+          id: this.getSipSessionId(session),
+          target,
+        });
         result.newSession = await this.call(target);
       },
       complete: () => {
@@ -999,7 +1174,6 @@ export default class WebRTCClient extends Emitter {
           id: this.getSipSessionId(session),
           referId: this.getSipSessionId(result.newSession),
         });
-
         session.refer(result.newSession);
       },
       cancel: () => {
@@ -1007,16 +1181,19 @@ export default class WebRTCClient extends Emitter {
         this.unhold(session);
       },
     };
-
     return result;
   }
 
-  sendMessage(sipSession: Session = null, body: string, contentType: string = 'text/plain') {
+  // eslint-disable-next-line @typescript-eslint/default-param-last
+  sendMessage(sipSession: Inviter | Invitation | null = null, body: string, contentType = 'text/plain'): void {
     if (!sipSession) {
       return;
     }
 
-    logger.info('send WebRTC message', { sipId: sipSession.id, contentType });
+    logger.info('send WebRTC message', {
+      sipId: sipSession.id,
+      contentType,
+    });
 
     try {
       sipSession.message({
@@ -1024,32 +1201,44 @@ export default class WebRTCClient extends Emitter {
           body: {
             content: body,
             contentType,
+            // @HEADSUP: contentDisposition is a required string, setting it to '' arbitrarily
+            contentDisposition: '',
           },
         },
       });
-    } catch (e) {
+    } catch (e: any) {
       console.warn(e);
     }
   }
 
-  pingServer() {
+  pingServer(): OutgoingRequest | void {
     if (!this.isConnected()) {
       return;
     }
 
-    const core = this.userAgent.userAgentCore;
+    const core = this.userAgent?.userAgentCore;
+
     const fromURI = this._makeURI(this.config.authorizationUser || '');
+
     const toURI = new URI('sip', '', this.config.host);
-    const message = core.makeOutgoingRequestMessage('OPTIONS', toURI, fromURI, toURI, {});
+    if (fromURI) {
+      const message = core?.makeOutgoingRequestMessage('OPTIONS', toURI, fromURI, toURI, {});
+
+      if (message) {
+        return core?.request(message);
+      }
 
-    return core.request(message);
+      logger.warn('pingServer: null message');
+    }
+    logger.warn('pingServer: null fromURI');
   }
 
-  getState() {
+  getState(): UserAgentState {
+    // @ts-ignore
     return this.userAgent ? states[this.userAgent.state] : UserAgentState.Stopped;
   }
 
-  getContactIdentifier() {
+  getContactIdentifier(): string | null {
     return this.userAgent ? `${this.userAgent.configuration.contactName}/${this.userAgent.contact.uri}` : null;
   }
 
@@ -1057,9 +1246,10 @@ export default class WebRTCClient extends Emitter {
     return this._isWeb() && navigator.userAgent.toLowerCase().indexOf('firefox') > -1;
   }
 
-  changeAudioOutputVolume(volume: number) {
-    logger.info('Changing audio output volume', { volume });
-
+  changeAudioOutputVolume(volume: number): void {
+    logger.info('Changing audio output volume', {
+      volume,
+    });
     Object.values(this.audioElements).forEach(audioElement => {
       if (audioElement instanceof HTMLAudioElement) {
         // eslint-disable-next-line no-param-reassign
@@ -1070,25 +1260,30 @@ export default class WebRTCClient extends Emitter {
   }
 
   changeAudioOutputDevice(id: string) {
-    logger.info('Changing audio output device', { id });
+    logger.info('Changing audio output device', {
+      id,
+    });
     this.audioOutputDeviceId = id;
-
     Object.values(this.audioElements).forEach(audioElement => {
       // `setSinkId` method is not included in any flow type definitions for HTMLAudioElements but is a valid method
       // audioElement is an array of HTMLAudioElements, and HTMLAudioElement inherits the method from HTMLMediaElement
       // https://developer.mozilla.org/en-US/docs/Web/API/HTMLAudioElement
       // https://developer.mozilla.org/en-US/docs/Web/API/HTMLMediaElement/setSinkId
-
-      // $FlowFixMe
+      // @ts-ignore
       if (audioElement.setSinkId) {
+        // @ts-ignore
         audioElement.setSinkId(id);
       }
     });
   }
 
-  async changeAudioInputDevice(id: string, session: ?Inviter, force: ?boolean) {
+  async changeAudioInputDevice(id: string, session: Inviter | Invitation | null | undefined, force: boolean | null | undefined): Promise<MediaStream | null> {
     const currentId = this.getAudioDeviceId();
-    logger.info('setting audio input device', { id, currentId, session: !!session });
+    logger.info('setting audio input device', {
+      id,
+      currentId,
+      session: !!session,
+    });
 
     if (!force && id === currentId) {
       return null;
@@ -1097,14 +1292,18 @@ export default class WebRTCClient extends Emitter {
     // in order to handle an audio track change for default devices
     // we need the actual id of the device (not 'default')
     let deviceId = id;
+
     if (id === 'default' && navigator.mediaDevices) {
       const devices = await navigator.mediaDevices.enumerateDevices();
       const defaultDevice = devices.find(device => device.deviceId === 'default');
+
       if (defaultDevice) {
         const deviceLabel = defaultDevice.label.replace('Default - ', '');
         const targetDevice = devices.find(device => device.label === deviceLabel);
+
         if (targetDevice) {
           deviceId = targetDevice.deviceId;
+
           if (!force && deviceId === currentId) {
             return null;
           }
@@ -1123,26 +1322,34 @@ export default class WebRTCClient extends Emitter {
 
     if (session && navigator.mediaDevices) {
       const sdh = session.sessionDescriptionHandler;
+      // @ts-ignore
       const pc = sdh.peerConnection;
-
-      const constraints = { audio: { deviceId: { exact: deviceId } } };
+      const constraints = {
+        audio: {
+          deviceId: {
+            exact: deviceId,
+          },
+        },
+      };
       const stream = await navigator.mediaDevices.getUserMedia(constraints);
       const audioTrack = stream.getAudioTracks()[0];
-      const sender = pc && pc.getSenders
-        && pc.getSenders().find(s => audioTrack && s && s.track && s.track.kind === audioTrack.kind);
+      const sender = pc && pc.getSenders && pc.getSenders().find((s: any) => audioTrack && s && s.track && s.track.kind === audioTrack.kind);
 
       if (sender) {
         if (sender.track) {
           audioTrack.enabled = sender.track.enabled;
         }
+
         sender.replaceTrack(audioTrack);
       }
 
       return stream;
     }
+
+    return null;
   }
 
-  changeVideoInputDevice(id: ?string, session: ?Inviter) {
+  async changeVideoInputDevice(id: string, session?: Inviter | Invitation): Promise<MediaStream | void> {
     this.setVideoInputDevice(id);
 
     if (session) {
@@ -1150,9 +1357,12 @@ export default class WebRTCClient extends Emitter {
     }
   }
 
-  setVideoInputDevice(id: ?string) {
+  setVideoInputDevice(id: string) {
     const currentId = this.getVideoDeviceId();
-    logger.info('setting video input device', { id, currentId });
+    logger.info('setting video input device', {
+      id,
+      currentId,
+    });
 
     if (id === currentId) {
       return null;
@@ -1160,24 +1370,27 @@ export default class WebRTCClient extends Emitter {
 
     // let's make sure we don't lose other video constraints settings -- width, height, frameRate...
     const videoObject = typeof this.video === 'object' ? this.video : {};
-    this.video = {
-      ...videoObject,
+    this.video = { ...videoObject,
       deviceId: {
         exact: id,
       },
     };
   }
 
-  changeSessionVideoInputDevice(id: ?string, session: Inviter) {
+  changeSessionVideoInputDevice(id: string | null | undefined, session: Inviter | Invitation): Promise<MediaStream | void> {
     if (!this.sessionWantsToDoVideo(session)) {
-      return;
+      return Promise.resolve();
     }
+
     const sdh = session.sessionDescriptionHandler;
+    // @ts-ignore
     const pc = sdh.peerConnection;
     const sessionId = this.getSipSessionId(session);
     const localStream = this.getLocalStream(sessionId);
-
-    logger.info('changing video input device', { id, sessionId });
+    logger.info('changing video input device', {
+      id,
+      sessionId,
+    });
 
     // Release old video stream
     if (localStream) {
@@ -1187,16 +1400,19 @@ export default class WebRTCClient extends Emitter {
     }
 
     const constraints = {
-      video: id ? { deviceId: { exact: id } } : true,
+      video: id ? {
+        deviceId: {
+          exact: id,
+        },
+      } : true,
     };
-    // $FlowFixMe
     return navigator.mediaDevices.getUserMedia(constraints).then(async stream => {
       const videoTrack = stream.getVideoTracks()[0];
-      let sender = pc && pc.getSenders
-        && pc.getSenders().find(s => videoTrack && s && s.track && s.track.kind === videoTrack.kind);
+      let sender = pc && pc.getSenders && pc.getSenders().find((s: any) => videoTrack && s && s.track && s.track.kind === videoTrack.kind);
       let wasTrackEnabled = false;
+
       if (!sender) {
-        sender = pc && pc.getSenders && pc.getSenders().find(s => !s.track);
+        sender = pc && pc.getSenders && pc.getSenders().find((s: any) => !s.track);
       }
 
       if (sender) {
@@ -1214,38 +1430,34 @@ export default class WebRTCClient extends Emitter {
       // let's update the local stream
       this.eventEmitter.emit('onVideoInputChange', stream);
       this.setLocalMediaStream(sessionId, stream);
-
       return stream;
     });
   }
 
-  getAudioDeviceId(): ?string {
-    // $FlowFixMe
-    return this.audio && typeof this.audio === 'object' && 'deviceId' in this.audio
-      ? this.audio.deviceId.exact : undefined;
+  getAudioDeviceId(): string | null | undefined {
+    // @ts-ignore
+    return this.audio && typeof this.audio === 'object' && 'deviceId' in this.audio ? this.audio.deviceId.exact : undefined;
   }
 
-  getVideoDeviceId(): ?string {
-    // $FlowFixMe
-    return this.video && typeof this.video === 'object' && 'deviceId' in this.video
-      ? this.video.deviceId.exact : undefined;
+  getVideoDeviceId(): string | null {
+    // @ts-ignore
+    return this.video && typeof this.video === 'object' && 'deviceId' in this.video ? this.video.deviceId.exact : undefined;
   }
 
-  reinvite(sipSession: Session, newConstraints: ?Object = null, conference: boolean = false,
-    audioOnly: boolean = false, iceRestart: boolean = false) {
+  reinvite(sipSession: Inviter | Invitation, newConstraints: Record<string, any> | null | undefined = null, conference = false, audioOnly = false, iceRestart = false): Promise<OutgoingInviteRequest | void> {
     if (!sipSession) {
-      return false;
+      return Promise.resolve();
     }
 
+    // @ts-ignore
     if (sipSession.pendingReinvite) {
-      return false;
+      return Promise.resolve();
     }
 
     const wasMuted = this.isAudioMuted(sipSession);
     const shouldDoVideo = newConstraints ? newConstraints.video : this.sessionWantsToDoVideo(sipSession);
     const shouldDoScreenSharing = newConstraints && newConstraints.screen;
     const desktop = newConstraints && newConstraints.desktop;
-
     logger.info('Sending reinvite', {
       id: this.getSipSessionId(sipSession),
       newConstraints,
@@ -1260,27 +1472,28 @@ export default class WebRTCClient extends Emitter {
     // When upgrading to video, remove the `stripVideo` modifiers
     if (newConstraints && newConstraints.video) {
       const modifiers = sipSession.sessionDescriptionHandlerModifiersReInvite;
-      sipSession.sessionDescriptionHandlerModifiersReInvite = modifiers.filter(modifier =>
-        modifier !== stripVideo);
+      sipSession.sessionDescriptionHandlerModifiersReInvite = modifiers.filter(modifier => modifier !== stripVideo);
     }
 
-    sipSession.sessionDescriptionHandlerOptionsReInvite = {
-      ...sipSession.sessionDescriptionHandlerOptionsReInvite,
+    sipSession.sessionDescriptionHandlerOptionsReInvite = { ...sipSession.sessionDescriptionHandlerOptionsReInvite,
+      // @ts-ignore
       conference,
       audioOnly,
     };
-
-    // $FlowFixMe
-    const { constraints } = this.getMediaConfiguration(shouldDoVideo, conference, newConstraints);
-
+    const {
+      constraints,
+    } = this.getMediaConfiguration(shouldDoVideo, conference, newConstraints);
     return sipSession.invite({
       requestDelegate: {
         onAccept: (response: IncomingResponse) => {
           // Update the SDP body to be able to call sessionWantsToDoVideo correctly in `_setup[Local|Remote]Media`.
           // Can't set directly sipSession.body because it's a getter.
-          if (sipSession instanceof Inviter) {
+          // @ts-ignore
+          if (sipSession instanceof Inviter && sipSession.outgoingRequestMessage.body) {
+            // @ts-ignore
             sipSession.outgoingRequestMessage.body.body = response.message.body;
-          } else {
+          } else if (sipSession instanceof Invitation) {
+            // @ts-ignore
             sipSession.incomingInviteRequest.message.body = response.message.body;
           }
 
@@ -1289,13 +1502,13 @@ export default class WebRTCClient extends Emitter {
             wasMuted,
             shouldDoScreenSharing,
           });
-
           this.updateRemoteStream(this.getSipSessionId(sipSession));
 
           if (wasMuted) {
             this.mute(sipSession);
           }
 
+          // @ts-ignore
           this._onAccepted(sipSession, response.session, false);
 
           if (shouldDoScreenSharing) {
@@ -1311,6 +1524,7 @@ export default class WebRTCClient extends Emitter {
       },
       sessionDescriptionHandlerOptions: {
         constraints,
+        // @ts-ignore
         conference,
         audioOnly,
         offerOptions: {
@@ -1320,35 +1534,35 @@ export default class WebRTCClient extends Emitter {
     });
   }
 
-  async getUserMedia(constraints: Object) {
+  async getUserMedia(constraints: Record<string, any>): Promise<MediaStream> {
     const newConstraints = {
       audio: this._getAudioConstraints(),
       video: this._getVideoConstraints(constraints.video),
     };
-
-    // $FlowFixMe
     return navigator.mediaDevices.getUserMedia(newConstraints);
   }
 
   getPeerConnection(sessionId: string) {
     const sipSession = this.sipSessions[sessionId];
+
     if (!sipSession) {
       return null;
     }
 
+    // @ts-ignore
     return sipSession.sessionDescriptionHandler ? sipSession.sessionDescriptionHandler.peerConnection : null;
   }
 
   // Local streams
-  getLocalStream(sessionId: string): ?MediaStream {
+  getLocalStream(sessionId: string): MediaStream | null {
     const sipSession = this.sipSessions[sessionId];
-
-    return sipSession && sipSession.sessionDescriptionHandler
-      ? sipSession.sessionDescriptionHandler.localMediaStream : null;
+    // @ts-ignore
+    return sipSession?.sessionDescriptionHandler?.localMediaStream || null;
   }
 
   getLocalTracks(sessionId: string): MediaStreamTrack[] {
     const localStream = this.getLocalStream(sessionId);
+
     if (!localStream) {
       return [];
     }
@@ -1365,20 +1579,20 @@ export default class WebRTCClient extends Emitter {
     return this.getLocalTracks(sessionId).some(track => track.kind === 'video');
   }
 
-  getLocalVideoStream(sessionId: string): ?MediaStream {
+  getLocalVideoStream(sessionId: string): MediaStream | null | undefined {
     return this.hasLocalVideo(sessionId) ? this.getLocalStream(sessionId) : null;
   }
 
   // Remote streams
-  getRemoteStream(sessionId: string): ?MediaStream {
+  getRemoteStream(sessionId: string): MediaStream | null {
     const sipSession = this.sipSessions[sessionId];
-
-    return sipSession && sipSession.sessionDescriptionHandler
-      ? sipSession.sessionDescriptionHandler.remoteMediaStream : null;
+    // @ts-ignore
+    return sipSession?.sessionDescriptionHandler?.remoteMediaStream || null;
   }
 
   getRemoteTracks(sessionId: string): MediaStreamTrack[] {
     const remoteStream = this.getRemoteStream(sessionId);
+
     if (!remoteStream) {
       return [];
     }
@@ -1395,7 +1609,7 @@ export default class WebRTCClient extends Emitter {
     return this.getRemoteTracks(sessionId).some(track => track.kind === 'video');
   }
 
-  getRemoteVideoStream(sessionId: string): ?MediaStream {
+  getRemoteVideoStream(sessionId: string): MediaStream | null | undefined {
     if (this.isVideoRemotelyHeld(sessionId)) {
       return null;
     }
@@ -1404,13 +1618,14 @@ export default class WebRTCClient extends Emitter {
   }
 
   //  Useful in a react-native environment when remoteMediaStream is not updated
-  getRemoteVideoStreamFromPc(sessionId: string): ?MediaStream {
+  getRemoteVideoStreamFromPc(sessionId: string): MediaStream | null | undefined {
     const pc = this.getPeerConnection(sessionId);
+
     if (!pc) {
       return null;
     }
 
-    return pc.getRemoteStreams().find(stream => !!stream.getVideoTracks().length);
+    return pc.getRemoteStreams().find((stream: MediaStream) => !!stream.getVideoTracks().length);
   }
 
   hasVideo(sessionId: string): boolean {
@@ -1421,45 +1636,55 @@ export default class WebRTCClient extends Emitter {
     return this.hasALocalVideoTrack(sessionId) || this.hasARemoteVideoTrack(sessionId);
   }
 
-  getSipSessionId(sipSession: ?Inviter): string {
+  getSipSessionId(sipSession: Inviter | Invitation | null | undefined): string {
     if (!sipSession) {
       return '';
     }
-    if (sipSession.message && sipSession.message.callId) {
-      return sipSession.message.callId;
+
+    // @ts-ignore
+    if (sipSession.message?.callId) {
+      // @ts-ignore
+      return sipSession.message.callId.substring(0, SIP_ID_LENGTH);
     }
 
     // For Inviter
-    if (sipSession.outgoingRequestMessage) {
-      return sipSession.outgoingRequestMessage.callId;
+    // @ts-ignore
+    if (sipSession instanceof Inviter && sipSession.outgoingRequestMessage) {
+      // @ts-ignore
+      return sipSession.outgoingRequestMessage.callId.substring(0, SIP_ID_LENGTH);
     }
 
     // For Invitation
-    return (sipSession.id || '').substr(0, 36);
+    return (sipSession.id || '').substring(0, SIP_ID_LENGTH);
   }
 
-  async waitForRegister() {
+  async waitForRegister(): Promise<void> {
     return new Promise(resolve => this.on(REGISTERED, resolve));
   }
 
   // eslint-disable-next-line no-unused-vars
-  sessionWantsToDoVideo(session: Inviter) {
+  sessionWantsToDoVideo(session: Inviter | Invitation): boolean {
     if (!session) {
       return false;
     }
-    const { body } = session.request || session;
+
+    const {
+      body,
+    } = session.request || session;
     // Sometimes with InviteClientContext the body is in the body attribute ...
     const sdp = typeof body === 'object' && body ? body.body : body;
-
     return hasAnActiveVideo(sdp);
   }
 
-  hasHeartbeat() {
+  hasHeartbeat(): boolean {
     return this.heartbeat.hasHeartbeat;
   }
 
-  startHeartbeat() {
-    logger.info('sdk webrtc start heartbeat', { userAgent: !!this.userAgent });
+  startHeartbeat(): void {
+    logger.info('sdk webrtc start heartbeat', {
+      userAgent: !!this.userAgent,
+    });
+
     if (!this.userAgent) {
       this.heartbeat.stop();
       return;
@@ -1467,53 +1692,60 @@ export default class WebRTCClient extends Emitter {
 
     this.eventEmitter.off(MESSAGE, this._boundOnHeartbeat);
     this.eventEmitter.on(MESSAGE, this._boundOnHeartbeat);
-
     this.heartbeat.start();
   }
 
-  stopHeartbeat() {
+  stopHeartbeat(): void {
     logger.info('sdk webrtc stop heartbeat');
     this.heartbeat.stop();
   }
 
-  setOnHeartbeatTimeout(cb: Function) {
+  setOnHeartbeatTimeout(cb: (...args: Array<any>) => any): void {
     this.heartbeatTimeoutCb = cb;
   }
 
-  setOnHeartbeatCallback(cb: Function) {
+  setOnHeartbeatCallback(cb: (...args: Array<any>) => any): void {
     this.heartbeatCb = cb;
   }
 
-  onCallEnded(session: Session) {
+  onCallEnded(session: Inviter | Invitation): void {
     this._cleanupMedia(session);
+
     delete this.sipSessions[this.getSipSessionId(session)];
 
     this._stopSendingStats(session);
+
     this.stopNetworkMonitoring(session);
   }
 
   attemptReconnection(): void {
-    logger.info('attempt reconnection', { userAgent: !!this.userAgent });
+    logger.info('attempt reconnection', {
+      userAgent: !!this.userAgent,
+    });
+
     if (!this.userAgent) {
       return;
     }
+
+    // @ts-ignore
     this.userAgent.attemptReconnection();
   }
 
-  storeSipSession(session: Session) {
+  storeSipSession(session: Invitation | Inviter): void {
     this.sipSessions[this.getSipSessionId(session)] = session;
   }
 
-  getSipSession(id: string): ?Session {
-    return id in this.sipSessions ? this.sipSessions[id] : null;
+  getSipSession(id: string): Invitation | Inviter | null | undefined {
+    return id in this.sipSessions ? this.sipSessions[id.substring(0, SIP_ID_LENGTH)] : null;
   }
 
   getSipSessionIds(): string[] {
     return Object.keys(this.sipSessions);
   }
 
-  setLocalMediaStream(sipSessionId: string, newStream: MediaStream) {
+  setLocalMediaStream(sipSessionId: string, newStream: MediaStream): void {
     const sipSession = this.sipSessions[sipSessionId];
+
     if (!sipSession) {
       return;
     }
@@ -1522,19 +1754,21 @@ export default class WebRTCClient extends Emitter {
       sipSessionId,
       tracks: newStream ? newStream.getTracks() : null,
     });
-
-    // eslint-disable-next-line no-underscore-dangle
-    sipSession.sessionDescriptionHandler._localMediaStream = newStream;
+    if (sipSession.sessionDescriptionHandler) {
+      // eslint-disable-next-line no-underscore-dangle
+      // @ts-ignore
+      sipSession.sessionDescriptionHandler._localMediaStream = newStream;
+    }
   }
 
-  updateLocalStream(sipSessionId: string, newStream: MediaStream) {
+  updateLocalStream(sipSessionId: string, newStream: MediaStream): void {
     const sipSession = this.sipSessions[sipSessionId];
+
     if (!sipSession) {
       return;
     }
 
     const oldStream = this.getLocalStream(sipSessionId);
-
     logger.info('updating local stream', {
       sipSessionId,
       oldStream: !!oldStream,
@@ -1551,14 +1785,13 @@ export default class WebRTCClient extends Emitter {
     this._setupMedias(sipSession, newStream);
   }
 
-  updateRemoteStream(sessionId: string) {
+  updateRemoteStream(sessionId: string): void {
     const remoteStream = this.getRemoteStream(sessionId);
     const pc = this.getPeerConnection(sessionId);
-
     logger.info('Updating remote stream', {
       sessionId,
       tracks: remoteStream ? remoteStream.getTracks() : null,
-      receiverTracks: pc && pc.getReceivers ? pc.getReceivers().map(receiver => receiver.track) : null,
+      receiverTracks: pc && pc.getReceivers ? pc.getReceivers().map((receiver: any) => receiver.track) : null,
     });
 
     if (!pc || !remoteStream) {
@@ -1570,27 +1803,36 @@ export default class WebRTCClient extends Emitter {
     });
 
     if (pc.getReceivers) {
-      pc.getReceivers().forEach(receiver => {
+      pc.getReceivers().forEach((receiver: any) => {
         remoteStream.addTrack(receiver.track);
       });
     }
   }
 
-  getMediaConfiguration(enableVideo: boolean, conference: boolean = false, constraints: ?Object = null): Object {
+  getMediaConfiguration(enableVideo: boolean, conference = false, constraints: Record<string, any> | null | undefined = null): Record<string, any> {
     const screenSharing = constraints && 'screen' in constraints ? constraints.screen : false;
     const isDesktop = constraints && 'desktop' in constraints ? constraints.desktop : false;
     const withAudio = constraints && 'audio' in constraints ? constraints.audio : true;
     const mandatoryVideo = constraints && typeof constraints.video === 'object' ? constraints.video.mandatory : {};
-
-    logger.info('Retrieving media a configuration', { enableVideo, screenSharing, isDesktop, withAudio, constraints });
-
+    logger.info('Retrieving media a configuration', {
+      enableVideo,
+      screenSharing,
+      isDesktop,
+      withAudio,
+      constraints,
+    });
     return {
       constraints: {
         // Exact constraint are not supported with `getDisplayMedia` and we must have a video=false in desktop screenshare
-        audio: screenSharing ? !isDesktop : (withAudio ? this._getAudioConstraints() : false),
-        video: screenSharing ? (isDesktop
-          ? ({ mandatory: { chromeMediaSource: 'desktop', ...(mandatoryVideo || {}) } }) : { cursor: 'always' })
-          : this._getVideoConstraints(enableVideo),
+        audio: screenSharing ? !isDesktop : withAudio ? this._getAudioConstraints() : false,
+        video: screenSharing ? isDesktop ? {
+          mandatory: {
+            chromeMediaSource: 'desktop',
+            ...(mandatoryVideo || {}),
+          },
+        } : {
+          cursor: 'always',
+        } : this._getVideoConstraints(enableVideo),
         screen: screenSharing,
         desktop: isDesktop,
         conference,
@@ -1608,42 +1850,44 @@ export default class WebRTCClient extends Emitter {
     };
   }
 
-  isConference(sessionId: string) {
+  isConference(sessionId: string): boolean {
     return sessionId in this.conferences;
   }
 
-  createAudioElementFor(sessionId: string) {
-    const audio: any = document.createElement('audio');
+  createAudioElementFor(sessionId: string): HTMLAudioElement {
+    const audio: HTMLAudioElement = document.createElement('audio');
     audio.setAttribute('id', `audio-${sessionId}`);
 
-    if (audio.setSinkId && this.audioOutputDeviceId) {
-      audio.setSinkId(this.audioOutputDeviceId);
+    if ((audio as any).setSinkId && this.audioOutputDeviceId) {
+      (audio as any).setSinkId(this.audioOutputDeviceId);
     }
 
     if (document.body) {
       document.body.appendChild(audio);
     }
-    this.audioElements[sessionId] = audio;
 
+    this.audioElements[sessionId] = audio;
     audio.volume = this.audioOutputVolume;
     audio.autoplay = true;
-
     return audio;
   }
 
-  _onTransportError() {
+  _onTransportError(): void {
     logger.error('on transport error');
     this.eventEmitter.emit(TRANSPORT_ERROR);
     this.attemptReconnection();
   }
 
-  _onHeartbeat(message: string | Object) {
+  _onHeartbeat(message: string | Record<string, any>): void {
     const body = message && typeof message === 'object' ? message.data : message;
+
     if (body && body.indexOf('200 OK') !== -1) {
       if (this.hasHeartbeat()) {
-        logger.info('on heartbeat received from Asterisk', { hasHeartbeat: this.hasHeartbeat() });
-
+        logger.info('on heartbeat received from Asterisk', {
+          hasHeartbeat: this.hasHeartbeat(),
+        });
         this.heartbeat.onHeartbeat();
+
         if (this.heartbeatCb) {
           this.heartbeatCb();
         }
@@ -1651,8 +1895,11 @@ export default class WebRTCClient extends Emitter {
     }
   }
 
-  async _onHeartbeatTimeout() {
-    logger.warn('sdk webrtc heartbeat timed out', { userAgent: !!this.userAgent, cb: !!this.heartbeatTimeoutCb });
+  async _onHeartbeatTimeout(): Promise<void> {
+    logger.warn('sdk webrtc heartbeat timed out', {
+      userAgent: !!this.userAgent,
+      cb: !!this.heartbeatTimeoutCb,
+    });
 
     if (this.heartbeatTimeoutCb) {
       this.heartbeatTimeoutCb();
@@ -1660,10 +1907,11 @@ export default class WebRTCClient extends Emitter {
 
     if (this.userAgent && this.userAgent.transport) {
       // Disconnect from WS and triggers events, but do not trigger disconnect if already disconnecting...
+      // @ts-ignore
       if (!this.userAgent.transport.transitioningState && !this.userAgent.transport.disconnectPromise) {
         try {
           await this.userAgent.transport.disconnect();
-        } catch (e) {
+        } catch (e: any) {
           logger.error('Transport disconnection after heartbeat timeout, error', e);
         }
       }
@@ -1673,31 +1921,37 @@ export default class WebRTCClient extends Emitter {
     }
   }
 
-  _isWeb() {
+  _isWeb(): boolean {
     return typeof window === 'object' && typeof document === 'object';
   }
 
-  _isVideoTrack(track: MediaStreamTrack) {
+  _isVideoTrack(track: MediaStreamTrack): boolean {
     return track.kind === 'video' && track.readyState === 'live';
   }
 
-  _hasAudio() {
+  _hasAudio(): boolean {
     return this.hasAudio;
   }
 
-  _getAudioConstraints() {
-    return this.audio && this.audio.deviceId && this.audio.deviceId.exact ? this.audio : true;
+  _getAudioConstraints(): MediaTrackConstraints | boolean {
+    // @ts-ignore
+    return this.audio?.deviceId?.exact ? this.audio : true;
   }
 
-  _getVideoConstraints(video: boolean = false) {
+  _getVideoConstraints(video = false): MediaTrackConstraints | boolean {
     if (!video) {
       return false;
     }
-    return this.video && this.video.deviceId && this.video.deviceId.exact ? this.video : true;
+
+    // @ts-ignore
+    return this.video?.deviceId?.exact ? this.video : true;
   }
 
-  _connectIfNeeded(): Promise<void> {
-    logger.info('connect if needed, checking', { connected: this.isConnected() });
+  _connectIfNeeded(): Promise<any> {
+    logger.info('connect if needed, checking', {
+      connected: this.isConnected(),
+    });
+
     if (!this.userAgent) {
       logger.info('need to recreate User Agent');
       this.userAgent = this.createUserAgent(this.uaConfigOverrides);
@@ -1710,14 +1964,13 @@ export default class WebRTCClient extends Emitter {
 
     if (this.isConnecting()) {
       logger.info('webrtc sdk, already connecting...');
-
+      // @ts-ignore
       this.connectionPromise = this.userAgent.transport.connectPromise;
-      return this.connectionPromise;
+      return Promise.resolve(this.connectionPromise);
     }
 
     if (this.connectionPromise) {
       logger.info('webrtc sdk, connection promise connecting...');
-      // $FlowFixMe
       return this.connectionPromise;
     }
 
@@ -1725,26 +1978,34 @@ export default class WebRTCClient extends Emitter {
 
     // Force UA to reconnect
     if (this.userAgent && this.userAgent.state !== UserAgentState.Stopped) {
+      // @ts-ignore
       this.userAgent.transitionState(UserAgentState.Stopped);
     }
+
     if (this.userAgent.transport && this.userAgent.transport.state !== TransportState.Disconnected) {
+      // @ts-ignore
       this.userAgent.transport.transitionState(TransportState.Disconnected);
     }
-    this.connectionPromise = this.userAgent.start().catch(console.error);
 
+    this.connectionPromise = this.userAgent.start().catch(console.error);
     return this.connectionPromise;
   }
 
-  _buildConfig(config: WebRtcConfig, session: ?Session): Promise<WebRtcConfig> {
+  _buildConfig(config: WebRtcConfig, session: Invitation | Inviter | null | undefined): Promise<WebRtcConfig> {
     // If no session provided, return the configuration directly
     if (!session) {
       return new Promise(resolve => resolve(config));
     }
 
-    const client = new ApiClient({ server: config.host });
+    const client = new ApiClient({
+      server: config.host,
+    });
+    // @ts-ignore
     client.setToken(session.token);
+    // @ts-ignore
     client.setRefreshToken(session.refreshToken);
 
+    // @ts-ignore
     return client.confd.getUserLineSipFromToken(session.uuid).then(sipLine => ({
       authorizationUser: sipLine.username,
       password: sipLine.secret,
@@ -1753,15 +2014,19 @@ export default class WebRTCClient extends Emitter {
     }));
   }
 
-  _createWebRTCConfiguration(configOverrides: Object = {}) {
-    let { host } = this.config;
-    let port = this.config.port || 443;
+  _createWebRTCConfiguration(configOverrides: Record<string, any> | null = {}): Record<string, any> {
+    let {
+      host,
+    } = this.config;
+    let port: number = this.config.port || 443;
 
     if (this.config.websocketSip) {
-      [host, port = 443] = this.config.websocketSip.split(':');
+      const webSocketSip = this.config.websocketSip.split(':');
+      [host] = webSocketSip;
+      port = Number(webSocketSip[1]);
     }
 
-    const config: Object = {
+    const config: Record<string, any> = {
       authorizationUsername: this.config.authorizationUser,
       authorizationPassword: this.config.password,
       displayName: this.config.displayName,
@@ -1777,25 +2042,23 @@ export default class WebRTCClient extends Emitter {
       userAgentString: this.config.userAgentString || 'wazo-sdk',
       reconnectionAttempts: 10000,
       reconnectionDelay: 5,
-      sessionDescriptionHandlerFactory: (session: Session, options: SessionDescriptionHandlerFactoryOptions = {}) => {
+      sessionDescriptionHandlerFactory: (session: Inviter | Invitation, options: SessionDescriptionHandlerFactoryOptions = {}) => {
         const uaLogger = session.userAgent.getLogger('sip.WazoSessionDescriptionHandler');
+
         const isWeb = this._isWeb();
-        const iceGatheringTimeout = 'peerConnectionOptions' in options
-          ? options.peerConnectionOptions.iceGatheringTimeout || DEFAULT_ICE_TIMEOUT : DEFAULT_ICE_TIMEOUT;
 
-        const sdhOptions: SessionDescriptionHandlerConfiguration = {
-          ...options,
+        // @ts-ignore
+        const iceGatheringTimeout = 'peerConnectionOptions' in options ? options.peerConnectionOptions.iceGatheringTimeout || DEFAULT_ICE_TIMEOUT : DEFAULT_ICE_TIMEOUT;
+        const sdhOptions: SessionDescriptionHandlerConfiguration = { ...options,
           iceGatheringTimeout,
-          peerConnectionConfiguration: {
-            ...defaultPeerConnectionConfiguration(),
+          peerConnectionConfiguration: { ...defaultPeerConnectionConfiguration(),
             ...(options.peerConnectionConfiguration || {}),
           },
         };
-
         return new WazoSessionDescriptionHandler(uaLogger, wazoMediaStreamFactory, sdhOptions, isWeb, session);
       },
       transportOptions: {
-        traceSip: configOverrides.traceSip || false,
+        traceSip: configOverrides?.traceSip || false,
         wsServers: `wss://${host}:${port}/api/asterisk/ws`,
       },
       sessionDescriptionHandlerFactoryOptions: {
@@ -1812,19 +2075,20 @@ export default class WebRTCClient extends Emitter {
             rtcpMuxPolicy: 'require',
             iceServers: WebRTCClient.getIceServers(this.config.host),
             ...this._getRtcOptions(),
-            ...configOverrides.peerConnectionOptions || {},
+            ...(configOverrides?.peerConnectionOptions || {}),
           },
         },
         // Configuration used in SDH to create the PeerConnection
         peerConnectionConfiguration: {
           rtcpMuxPolicy: 'require',
           iceServers: WebRTCClient.getIceServers(this.config.host),
-          ...configOverrides.peerConnectionOptions || {},
+          ...(configOverrides?.peerConnectionOptions || {}),
         },
       },
     };
-
-    return { ...config, ...configOverrides };
+    return { ...config,
+      ...configOverrides,
+    };
   }
 
   // eslint-disable-next-line no-unused-vars
@@ -1838,7 +2102,7 @@ export default class WebRTCClient extends Emitter {
   }
 
   // Invitation and Inviter extends Session
-  _setupSession(session: Session) {
+  _setupSession(session: Inviter | Invitation): void {
     const sipSessionId = this.getSipSessionId(session);
 
     // When receiving an Invitation, the delegate is not defined.
@@ -1847,12 +2111,12 @@ export default class WebRTCClient extends Emitter {
     }
 
     session.delegate.onSessionDescriptionHandler = (sdh: SessionDescriptionHandler) => {
+      // @ts-ignore
       sdh.on('error', e => {
         this.eventEmitter.emit(ON_ERROR, e);
       });
-
       sdh.peerConnectionDelegate = {
-        onicecandidateerror: (error: Object) => {
+        onicecandidateerror: (error: Record<string, any>) => {
           logger.error('on icecandidate error', {
             address: error.address,
             port: error.port,
@@ -1864,33 +2128,43 @@ export default class WebRTCClient extends Emitter {
       };
     };
 
+    // @ts-ignore
     const oldInviteRequest = session.onInviteRequest.bind(session);
     let hadRemoteVideo = false;
 
     // Monkey patch `onInviteRequest` to be able to know if there was a remote video stream before `onInvite` is called
     // Because when `onInvite` is called we already got the video track
-    session.onInviteRequest = (request) => {
+    // @ts-ignore
+    session.onInviteRequest = request => {
       hadRemoteVideo = this.hasARemoteVideoTrack(sipSessionId);
-
       oldInviteRequest(request);
     };
 
     session.delegate.onInvite = (request: IncomingRequestMessage) => {
-      let updatedCalleeName = null;
+      let updatedCalleeName: string | null = null;
       let updatedNumber = null;
+
       if (session.assertedIdentity) {
+        // @ts-ignore
         updatedNumber = session.assertedIdentity.uri.normal.user;
         updatedCalleeName = session.assertedIdentity.displayName || updatedNumber;
       }
-      logger.info('re-invite received', { updatedCalleeName, updatedNumber, hadRemoteVideo });
 
-      // Useful to know if it's a video upgrade or an unhold from a remote peer with a video stream
+      logger.info('re-invite received', {
+        updatedCalleeName,
+        updatedNumber,
+        hadRemoteVideo,
+      });
 
+      // Useful to know if it's a video upgrade or an unhold from a remote peer with a video stream
       // Update SDP
       // Remote video is handled by the `track` event. Here we're dealing with video stream removal.
-      if (session.incomingInviteRequest) {
+      if (session instanceof Invitation) {
+        // @ts-ignore
         session.incomingInviteRequest.message.body = request.body;
-      } else {
+        // @ts-ignore
+      } else if (session instanceof Inviter && session.outgoingInviteRequest.message.body) {
+        // @ts-ignore
         session.outgoingInviteRequest.message.body.body = request.body;
       }
 
@@ -1902,31 +2176,49 @@ export default class WebRTCClient extends Emitter {
     };
   }
 
-  _onEarlyProgress(session: SessionDialog) {
+  _onEarlyProgress(session: any): void {
     this._setupMedias(session);
 
     const sessionId = this.getSipSessionId(session).substr(0, 20);
     this.updateRemoteStream(sessionId);
-
-    logger.info('Early media progress progress received', { sessionId });
-
+    logger.info('Early media progress progress received', {
+      sessionId,
+    });
     this.eventEmitter.emit(ON_EARLY_MEDIA, session);
   }
 
-  _onAccepted(session: Session, sessionDialog?: SessionDialog, withEvent: boolean = true) {
-    logger.info('on call accepted', { id: session.id, remoteTag: session.remoteTag });
-
+  _onAccepted(session: Inviter | Invitation, sessionDialog?: SessionDialog, withEvent = true): void {
+    logger.info('on call accepted', {
+      id: session.id,
+      // @ts-ignore
+      remoteTag: session.remoteTag,
+    });
     this.storeSipSession(session);
 
     this._setupMedias(session);
 
     this.updateRemoteStream(this.getSipSessionId(session));
+    // @ts-ignore
+    const pc = session.sessionDescriptionHandler?.peerConnection;
 
-    const pc = session.sessionDescriptionHandler.peerConnection;
     const onTrack = (event: any) => {
       const isAudioOnly = this._isAudioOnly(session);
-      const { kind, label, readyState, id, muted } = event.track;
-      logger.info('on track event', { isAudioOnly, kind, label, readyState, id, muted });
+
+      const {
+        kind,
+        label,
+        readyState,
+        id,
+        muted,
+      } = event.track;
+      logger.info('on track event', {
+        isAudioOnly,
+        kind,
+        label,
+        readyState,
+        id,
+        muted,
+      });
 
       // Stop video track in audio only mode
       if (isAudioOnly && kind === 'video') {
@@ -1936,15 +2228,21 @@ export default class WebRTCClient extends Emitter {
       this.eventEmitter.emit(ON_TRACK, session, event);
     };
 
+    // @ts-ignore
     if (session.sessionDescriptionHandler.peerConnection) {
+      // @ts-ignore
       session.sessionDescriptionHandler.peerConnection.addEventListener('track', onTrack);
     }
 
+    // @ts-ignore
     session.sessionDescriptionHandler.remoteMediaStream.onaddtrack = onTrack;
 
     if (pc) {
       pc.oniceconnectionstatechange = () => {
-        logger.info('on ice connection state changed', { state: pc.iceConnectionState });
+        logger.info('on ice connection state changed', {
+          state: pc.iceConnectionState,
+        });
+
         if (pc.iceConnectionState === 'disconnected') {
           this.eventEmitter.emit(ON_DISCONNECTED);
         }
@@ -1958,17 +2256,19 @@ export default class WebRTCClient extends Emitter {
     this._startSendingStats(session);
   }
 
-  _isAudioOnly(session: Session): boolean {
-    return session.sessionDescriptionHandlerModifiersReInvite.find(modifier => modifier === stripVideo);
+  _isAudioOnly(session: Inviter | Invitation): boolean {
+    return Boolean(session.sessionDescriptionHandlerModifiersReInvite.find(modifier => modifier === stripVideo));
   }
 
-  _setupMedias(session: Session, newStream: ?MediaStream) {
+  _setupMedias(session: Inviter | Invitation, newStream: MediaStream | null | undefined = null): void {
     // Safari hack, because you cannot call .play() from a non user action
     const sessionId = this.getSipSessionId(session);
     const isConference = this.isConference(sessionId);
 
     if (sessionId in this.audioElements) {
-      logger.info('html element already exists for session', { sessionId });
+      logger.info('html element already exists for session', {
+        sessionId,
+      });
       return;
     }
 
@@ -1977,10 +2277,11 @@ export default class WebRTCClient extends Emitter {
     }
 
     const audioElement = this.audioElements[sessionId];
+    // @ts-ignore
     const sipSession = this.sipSessions[session.callId];
     const removeStream = this.getRemoteStream(sessionId);
-    const earlyStream = sipSession && sipSession.sessionDescriptionHandler
-      ? sipSession.sessionDescriptionHandler.remoteMediaStream : null;
+    // @ts-ignore
+    const earlyStream = sipSession && sipSession.sessionDescriptionHandler ? sipSession.sessionDescriptionHandler.remoteMediaStream : null;
     const stream = newStream || removeStream || earlyStream;
 
     if (!this._isWeb() || !stream) {
@@ -1995,30 +2296,34 @@ export default class WebRTCClient extends Emitter {
     if (audioElement.currentTime > 0 && !audioElement.paused && !audioElement.ended && audioElement.readyState > 2) {
       audioElement.pause();
     }
+
     audioElement.srcObject = stream;
     audioElement.volume = this.audioOutputVolume;
     audioElement.play().catch(() => {});
   }
 
-  _cleanupMedia(session: Session) {
+  _cleanupMedia(session?: Inviter | Invitation): void {
     const sessionId = this.getSipSessionId(session);
     const localStream = this.getLocalStream(sessionId);
+
     if (localStream) {
       this._cleanupStream(localStream);
     }
 
-    const cleanLocalElement = id => {
+    const cleanLocalElement = (id: string) => {
       const element = this.audioElements[id];
+
       if (!element) {
         return;
       }
 
       element.pause();
+
       if (element.parentNode) {
         element.parentNode.removeChild(element);
       }
-      element.srcObject = null;
 
+      element.srcObject = null;
       delete this.audioElements[id];
     };
 
@@ -2031,25 +2336,27 @@ export default class WebRTCClient extends Emitter {
     }
   }
 
-  _cleanupStream(stream: MediaStream) {
+  _cleanupStream(stream: MediaStream): void {
     stream.getTracks().forEach(track => track.stop());
   }
 
-  _toggleAudio(session: Inviter, muteAudio: boolean) {
+  _toggleAudio(session: Inviter | Invitation, muteAudio: boolean): void {
+    // @ts-ignore
     const pc = session.sessionDescriptionHandler ? session.sessionDescriptionHandler.peerConnection : null;
+
     if (!pc) {
       return;
     }
 
     if (pc.getSenders) {
-      pc.getSenders().forEach(sender => {
+      pc.getSenders().forEach((sender: any) => {
         if (sender && sender.track && sender.track.kind === 'audio') {
           // eslint-disable-next-line
           sender.track.enabled = !muteAudio;
         }
       });
     } else {
-      pc.getLocalStreams().forEach(stream => {
+      pc.getLocalStreams().forEach((stream: MediaStream) => {
         stream.getAudioTracks().forEach(track => {
           // eslint-disable-next-line
           track.enabled = !muteAudio;
@@ -2058,18 +2365,19 @@ export default class WebRTCClient extends Emitter {
     }
   }
 
-  _toggleVideo(session: Inviter, muteCamera: boolean) {
-    const pc = session.sessionDescriptionHandler.peerConnection;
+  _toggleVideo(session: Inviter | Invitation, muteCamera: boolean): void {
+    // @ts-ignore
+    const pc = session.sessionDescriptionHandler?.peerConnection;
 
     if (pc.getSenders) {
-      pc.getSenders().forEach(sender => {
+      pc.getSenders().forEach((sender: any) => {
         if (sender && sender.track && sender.track.kind === 'video') {
           // eslint-disable-next-line
           sender.track.enabled = !muteCamera;
         }
       });
     } else {
-      pc.getLocalStreams().forEach(stream => {
+      pc.getLocalStreams().forEach((stream: MediaStream) => {
         stream.getVideoTracks().forEach(track => {
           // eslint-disable-next-line
           track.enabled = !muteCamera;
@@ -2081,15 +2389,18 @@ export default class WebRTCClient extends Emitter {
   /**
    * @param pc RTCPeerConnection
    */
-  _getRemoteStream(pc: any) {
-    let remoteStream;
+  _getRemoteStream(pc: any): MediaStream | null {
+    let remoteStream: MediaStream | null = null;
 
     if (pc && pc.getReceivers) {
-      remoteStream = typeof global !== 'undefined' && global.window && global.window.MediaStream
-        ? new global.window.MediaStream() : new window.MediaStream();
-      pc.getReceivers().forEach(receiver => {
-        const { track } = receiver;
-        if (track) {
+      remoteStream = typeof global !== 'undefined' && global.window && global.window.MediaStream ? new global.window.MediaStream() : new window.MediaStream();
+
+      pc.getReceivers().forEach((receiver: any) => {
+        const {
+          track,
+        } = receiver;
+
+        if (track && remoteStream) {
           remoteStream.addTrack(track);
         }
       });
@@ -2100,24 +2411,31 @@ export default class WebRTCClient extends Emitter {
     return remoteStream;
   }
 
-  _cleanupRegister() {
+  _cleanupRegister(): void {
     if (this.registerer) {
+      // @ts-ignore
       this.registerer.stateChange.removeAllListeners();
       this.registerer = null;
     }
   }
 
-  _startSendingStats(session: Session) {
+  _startSendingStats(session: Inviter | Invitation): void {
+    // @ts-ignore
     const pc = session.sessionDescriptionHandler.peerConnection;
+
     if (!pc) {
       return;
     }
-    const sessionId = this.getSipSessionId(session);
 
-    getStats(pc, (result: Object) => {
-      const { results, internal, nomore, ...stats } = result;
+    const sessionId = this.getSipSessionId(session);
+    getStats(pc, (result: Record<string, any>) => {
+      const {
+        results,
+        internal,
+        nomore,
+        ...stats
+      } = result;
       this.statsIntervals[sessionId] = nomore;
-
       statsLogger.trace('stats', {
         sessionId,
         ...stats,
@@ -2125,27 +2443,35 @@ export default class WebRTCClient extends Emitter {
     }, SEND_STATS_DELAY);
   }
 
-  _stopSendingStats(session: Session) {
+  _stopSendingStats(session: Invitation | Inviter): void {
     const sessionId = this.getSipSessionId(session);
-    logger.trace('Check for stopping stats', { sessionId, ids: Object.keys(this.statsIntervals) });
+    logger.trace('Check for stopping stats', {
+      sessionId,
+      ids: Object.keys(this.statsIntervals),
+    });
 
     if (sessionId in this.statsIntervals) {
-      logger.trace('Stop sending stats for call', { sessionId });
+      logger.trace('Stop sending stats for call', {
+        sessionId,
+      });
       this.statsIntervals[sessionId]();
       delete this.statsIntervals[sessionId];
     }
   }
 
-  _makeURI(target: string): URI {
+  _makeURI(target: string): URI | undefined {
     return UserAgent.makeURI(`sip:${target}@${this.config.host}`);
   }
 
-  async _disconnectTransport(force: boolean = false) {
+  async _disconnectTransport(force = false) {
     if (force && this.userAgent && this.userAgent.transport) {
       // Bypass sip.js state machine that prevent to close WS with the state `Connecting`
+      // @ts-ignore
       this.userAgent.transport.disconnectResolve = () => {};
 
+      // @ts-ignore
       if (this.userAgent.transport._ws) {
+        // @ts-ignore
         this.userAgent.transport._ws.close(1000);
       }
 
@@ -2153,24 +2479,25 @@ export default class WebRTCClient extends Emitter {
     }
 
     // Check if `disconnectPromise` is not present to avoid `Disconnect promise must not be defined` errors.
+    // @ts-ignore
     if (this.userAgent && this.userAgent.transport && !this.userAgent.transport.disconnectPromise) {
       try {
         await this.userAgent.transport.disconnect();
-      } catch (e) {
+      } catch (e: any) {
         logger.error('WebRTC transport disconnect, error', e);
       }
     }
   }
 
-  async _fetchNetworkStats(sessionId: string) {
+  async _fetchNetworkStats(sessionId: string): Promise<Record<string, any> | null | undefined> {
     const session = this.getSipSession(sessionId);
-
     const stats = session ? await this.getStats(session) : null;
+
     if (!stats || !(sessionId in this.sessionNetworkStats)) {
-      return;
+      return Promise.resolve(null);
     }
 
-    const networkStats: Object = {
+    const networkStats: Record<string, any> = {
       audioBytesSent: 0,
       videoBytesSent: 0,
       videoBytesReceived: 0,
@@ -2187,70 +2514,77 @@ export default class WebRTCClient extends Emitter {
     const lastTransportSent = lastNetworkStats ? lastNetworkStats.transportSent : 0;
     const lastTransportReceived = lastNetworkStats ? lastNetworkStats.transportReceived : 0;
     let audioBytesSent = 0; // content + header
+
     let audioBytesReceived = 0; // content + header
+
     let audioContentSent = 0; // useful to detect blank call
+
     let audioContentReceived = 0;
     let videoBytesSent = 0;
     let videoBytesReceived = 0;
     let transportSent = 0;
     let transportReceived = 0;
     let packetsLost = 0;
-
     stats.forEach(report => {
       if (report.type === 'outbound-rtp' && report.kind === 'audio') {
         audioBytesSent += Number(report.bytesSent) + Number(report.headerBytesSent);
         audioContentSent += Number(report.bytesSent);
       }
+
       if (report.type === 'remote-outbound-rtp' && report.kind === 'audio') {
         audioBytesSent += Number(report.bytesSent);
       }
+
       if (report.type === 'outbound-rtp' && report.kind === 'video') {
         videoBytesSent += Number(report.bytesSent) + Number(report.headerBytesSent);
       }
+
       if (report.type === 'inbound-rtp' && report.kind === 'audio') {
         packetsLost += Number(report.packetsLost);
         networkStats.packetsReceived = Number(report.packetsReceived);
         audioBytesReceived += Number(report.bytesReceived) + Number(report.headerBytesReceived);
         audioContentReceived += Number(report.bytesReceived);
       }
+
       if (report.type === 'inbound-rtp' && report.kind === 'video') {
         videoBytesReceived += Number(report.bytesReceived) + Number(report.headerBytesReceived);
       }
+
       if (report.type === 'outbound-rtp' && report.kind === 'video') {
         if ('framesPerSecond' in report) {
           networkStats.framesPerSecond = Number(report.framesPerSecond);
         }
+
         if ('framerateMean' in report) {
           // framerateMean is only available in FF
           networkStats.framesPerSecond = Math.round(report.framerateMean);
         }
       }
+
       if (report.type === 'remote-inbound-rtp' && report.kind === 'audio') {
         packetsLost += Number(report.packetsLost);
         networkStats.roundTripTime = Number(report.roundTripTime);
         networkStats.jitter = Number(report.jitter);
       }
+
       if (report.type === 'remote-inbound-rtp' && report.kind === 'video') {
         packetsLost += Number(report.packetsLost);
       }
+
       if (report.type === 'transport') {
         transportSent += Number(report.bytesSent);
         transportReceived += Number(report.bytesReceived);
       }
     });
-
     networkStats.packetsLost = packetsLost;
-
     networkStats.totalAudioBytesSent = audioBytesSent;
     networkStats.audioBytesSent = audioBytesSent - lastAudioSent;
     networkStats.totalAudioBytesReceived = audioBytesReceived - lastAudioReceived;
     networkStats.audioBytesReceived = audioBytesReceived;
-
     networkStats.totalAudioContentSent = audioContentSent;
     networkStats.audioContentSent = audioContentSent - lastAudioContentSent;
     networkStats.totalAudioBytesReceived = audioContentReceived - lastAudioContentReceived;
     networkStats.audioContentReceived = audioContentReceived;
-
     networkStats.totalVideoBytesSent = videoBytesSent;
     networkStats.videoBytesSent = videoBytesSent - lastVideoSent;
     networkStats.totalVideoBytesReceived = videoBytesReceived;
@@ -2259,11 +2593,7 @@ export default class WebRTCClient extends Emitter {
     networkStats.transportSent = transportSent;
     networkStats.totalTransportReceived = transportReceived;
     networkStats.transportReceived = transportReceived - lastTransportReceived;
-
-    networkStats.bandwidth = networkStats.audioBytesSent + networkStats.audioBytesReceived
-      + networkStats.videoBytesSent + networkStats.videoBytesReceived + networkStats.transportReceived
-      + networkStats.transportSent;
-
+    networkStats.bandwidth = networkStats.audioBytesSent + networkStats.audioBytesReceived + networkStats.videoBytesSent + networkStats.videoBytesReceived + networkStats.transportReceived + networkStats.transportSent;
     this.eventEmitter.emit(ON_NETWORK_STATS, session, networkStats, this.sessionNetworkStats[sessionId]);
     this.sessionNetworkStats[sessionId].push(networkStats);
   }