@wazo/sdk 0.37.24 → 0.37.25

package/esm/web-rtc-client.js

@@ -100,7 +100,7 @@ const isMobile = () => typeof navigator !== 'undefined' && navigator.product ===
 
 const methods = ['head', 'get', 'post', 'put', 'delete'];
 
-const logger = IssueReporter$1 ? IssueReporter$1.loggerFor('api') : console;
+const logger$2 = IssueReporter$1 ? IssueReporter$1.loggerFor('api') : console;
 
 // Use a function here to be able to mock it in tests
 const realFetch = () => {
@@ -239,7 +239,7 @@ class ApiRequester {
     };
 
     if (this.refreshTokenPromise) {
-      logger.info('A token is already refreshing, waiting ...', { url });
+      logger$2.info('A token is already refreshing, waiting ...', { url });
       await this.refreshTokenPromise;
     }
 
@@ -259,7 +259,7 @@ class ApiRequester {
         return promise.then(async (err        ) => {
           // Check if the token is still valid
           if (firstCall && this._checkTokenExpired(response, err)) {
-            logger.warn('token expired', { error: err.reason });
+            logger$2.warn('token expired', { error: err.reason });
             // Replay the call after refreshing the token
             return this._replayWithNewToken(err, path, method, body, headers, parse);
           }
@@ -269,7 +269,7 @@ class ApiRequester {
             : exceptionClass.fromResponse(err, response.status);
 
           if (this.shouldLogErrors) {
-            logger.error('API error', error);
+            logger$2.error('API error', error);
           }
 
           throw error;
@@ -279,7 +279,7 @@ class ApiRequester {
       return newParse(response, isJson);
     }).catch(error => {
       if (this.shouldLogErrors) {
-        logger.error('Fetch failed', { url, options, message: error.message, stack: error.stack });
+        logger$2.error('Fetch failed', { url, options, message: error.message, stack: error.stack });
       }
 
       throw error;
@@ -307,13 +307,13 @@ class ApiRequester {
   ) {
     const isTokenNotFound = this._isTokenNotFound(err);
     let newPath = path;
-    logger.info('refreshing token', { inProgress: !!this.refreshTokenPromise });
+    logger$2.info('refreshing token', { inProgress: !!this.refreshTokenPromise });
 
     this.refreshTokenPromise = this.refreshTokenPromise || this.refreshTokenCallback();
 
     return this.refreshTokenPromise.then(() => {
       this.refreshTokenPromise = null;
-      logger.info('token refreshed', { isTokenNotFound });
+      logger$2.info('token refreshed', { isTokenNotFound });
       if (isTokenNotFound) {
         const pathParts = path.split('/');
         pathParts.pop();
@@ -4138,7 +4138,7 @@ const TRANSFER_FLOW_BLIND = 'blind';
 
 // const MINIMUM_WAZO_ENGINE_VERSION_FOR_CTI_HOLD = '20.11';
 
-const logger$1 = IssueReporter$1 ? IssueReporter$1.loggerFor('cti-phone') : console;
+IssueReporter$1 ? IssueReporter$1.loggerFor('cti-phone') : console;
 
 /*       */
 
@@ -5057,7 +5057,7 @@ const AGENTD_VERSION = '1.0';
 const WEBHOOKD_VERSION = '1.0';
 const AMID_VERSION = '1.0';
 
-const logger$2 = IssueReporter$1 ? IssueReporter$1.loggerFor('api') : console;
+const logger$1 = IssueReporter$1 ? IssueReporter$1.loggerFor('api') : console;
 
 class ApiClient {
                        
@@ -5109,12 +5109,12 @@ class ApiClient {
   }
 
   async forceRefreshToken() {
-    logger$2.info('forcing refresh token, calling callback');
+    logger$1.info('forcing refresh token, calling callback');
     return this.refreshTokenCallback();
   }
 
   async refreshTokenCallback() {
-    logger$2.info('refresh token callback called', {
+    logger$1.info('refresh token callback called', {
       refreshToken: this.refreshToken,
       refreshBackend: this.refreshBackend,
       refreshExpiration: this.refreshExpiration,
@@ -5132,7 +5132,7 @@ class ApiClient {
       this.isMobile,
     );
 
-    logger$2.info('token refreshed', { token: session.token });
+    logger$1.info('token refreshed', { token: session.token });
 
     if (this.onRefreshToken) {
       this.onRefreshToken(session.token, session);
@@ -5305,7 +5305,7 @@ const DEFAULT_ICE_TIMEOUT = 3000;
 const SEND_STATS_DELAY = 5000;
 
 const states = ['STATUS_NULL', 'STATUS_NEW', 'STATUS_CONNECTING', 'STATUS_CONNECTED', 'STATUS_COMPLETED'];
-const logger$3 = IssueReporter$1 ? IssueReporter$1.loggerFor('webrtc-client') : console;
+const logger = IssueReporter$1 ? IssueReporter$1.loggerFor('webrtc-client') : console;
 const statsLogger = IssueReporter$1 ? IssueReporter$1.loggerFor('webrtc-stats') : console;
 
 // events
@@ -5484,13 +5484,13 @@ class WebRTCClient extends Emitter {
 
   createUserAgent(configOverrides         )            {
     const webRTCConfiguration = this._createWebRTCConfiguration(configOverrides);
-    logger$3.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),
       onInvite: (invitation            ) => {
-        logger$3.info('sdk webrtc on invite', {
+        logger.info('sdk webrtc on invite', {
           method: 'delegate.onInvite',
           id: invitation.id,
           remoteURI: invitation.remoteURI,
@@ -5533,7 +5533,7 @@ class WebRTCClient extends Emitter {
   }
 
   onConnect() {
-    logger$3.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) {
       this.registerer.waitingToggle(false);
@@ -5542,7 +5542,7 @@ class WebRTCClient extends Emitter {
   }
 
   async onDisconnect(error        ) {
-    logger$3.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);
@@ -5567,37 +5567,37 @@ class WebRTCClient extends Emitter {
     };
 
     if (this.skipRegister) {
-      logger$3.info('sdk webrtc skip register...', logInfo);
+      logger.info('sdk webrtc skip register...', logInfo);
       return Promise.resolve();
     }
 
-    logger$3.info('sdk webrtc registering...', logInfo);
+    logger.info('sdk webrtc registering...', logInfo);
 
     if (!this.userAgent) {
-      logger$3.info('sdk webrtc recreating User Agent');
+      logger.info('sdk webrtc recreating User Agent');
       this.userAgent = this.createUserAgent(this.uaConfigOverrides);
     }
     if (!this.userAgent || this.isRegistered()) {
-      logger$3.info('sdk webrtc registering aborted, already registered or no UA can be created');
+      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)) {
-      logger$3.info('sdk webrtc registering aborted due to a registration in progress.');
+      logger.info('sdk webrtc registering aborted due to a registration in progress.');
       return Promise.resolve();
     }
 
     const registerOptions = this._isWeb() ? {} : { extraContactHeaderParams: ['mobility=mobile'] };
 
     const onRegisterFailed = () => {
-      logger$3.info('sdk webrtc registering failed', { tries, registerer: !!this.registerer });
+      logger.info('sdk webrtc registering failed', { tries, registerer: !!this.registerer });
       this.connectionPromise = null;
       if (this.registerer && this.registerer.waiting) {
         this.registerer.waitingToggle(false);
       }
 
       if (tries <= MAX_REGISTER_TRIES) {
-        logger$3.info('sdk webrtc registering, retrying...', { tries });
+        logger.info('sdk webrtc registering, retrying...', { tries });
         setTimeout(() => this.register(tries + 1), 300);
       }
     };
@@ -5606,18 +5606,18 @@ class WebRTCClient extends Emitter {
       // Avoid race condition with the close method called just before register and setting userAgent to null
       // during the resolution of the primise.
       if (!this.userAgent) {
-        logger$3.info('sdk webrtc recreating User Agent after connection');
+        logger.info('sdk webrtc recreating User Agent after connection');
         this.userAgent = this.createUserAgent(this.uaConfigOverrides);
       }
 
-      logger$3.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$3.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) {
@@ -5630,7 +5630,7 @@ class WebRTCClient extends Emitter {
       const options = {
         requestDelegate: {
           onReject: response => {
-            logger$3.error('sdk webrtc registering, rejected', { response });
+            logger.error('sdk webrtc registering, rejected', { response });
 
             onRegisterFailed();
           },
@@ -5638,13 +5638,13 @@ class WebRTCClient extends Emitter {
       };
 
       return this.registerer.register(options).catch((e) => {
-        logger$3.error('sdk webrtc registering, error', e);
+        logger.error('sdk webrtc registering, error', e);
 
         this.eventEmitter.emit(REGISTRATION_FAILED);
         return e;
       });
     }).catch(error => {
-      logger$3.error('sdk webrtc registering, transport error', error);
+      logger.error('sdk webrtc registering, transport error', error);
       onRegisterFailed();
     });
   }
@@ -5673,21 +5673,21 @@ class WebRTCClient extends Emitter {
   }
 
   unregister() {
-    logger$3.info('sdk webrtc unregistering..', { userAgent: !!this.userAgent, registerer: !!this.registerer });
+    logger.info('sdk webrtc unregistering..', { userAgent: !!this.userAgent, registerer: !!this.registerer });
     if (!this.registerer) {
       return Promise.resolve();
     }
 
     try {
       return this.registerer.unregister().then(() => {
-        logger$3.info('sdk webrtc unregistered');
+        logger.info('sdk webrtc unregistered');
         this._cleanupRegister();
       }).catch(e => {
-        logger$3.error('sdk webrtc unregistering, promise error', e);
+        logger.error('sdk webrtc unregistering, promise error', e);
         this._cleanupRegister();
       });
     } catch (e) {
-      logger$3.error('sdk webrtc unregistering, error', e);
+      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();
@@ -5695,7 +5695,7 @@ class WebRTCClient extends Emitter {
   }
 
   stop()               {
-    logger$3.info('sdk webrtc stop', { userAgent: !!this.userAgent });
+    logger.info('sdk webrtc stop', { userAgent: !!this.userAgent });
     if (!this.userAgent) {
       return Promise.resolve();
     }
@@ -5703,12 +5703,12 @@ class WebRTCClient extends Emitter {
     return this.userAgent.stop().then(() => {
       return this._cleanupRegister();
     }).catch(e => {
-      logger$3.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        , enableVideo          , audioOnly          = false, conference          = false)          {
-    logger$3.info('sdk webrtc creating call', { number, enableVideo, audioOnly });
+    logger.info('sdk webrtc creating call', { number, enableVideo, audioOnly, conference });
     const inviterOptions         = {
       sessionDescriptionHandlerOptionsReInvite: {
         conference,
@@ -5739,7 +5739,7 @@ class WebRTCClient extends Emitter {
           this._onAccepted(session, response.session, true);
         },
         onReject: (response                  ) => {
-          logger$3.info('on call rejected', { id: session.id, fromTag: session.fromTag });
+          logger.info('on call rejected', { id: session.id, fromTag: session.fromTag });
           this._stopSendingStats(session);
           this.stopNetworkMonitoring(session);
 
@@ -5762,11 +5762,11 @@ class WebRTCClient extends Emitter {
   }
 
   answer(session            , enableVideo          ) {
-    logger$3.info('sdk webrtc answer call', { id: session.id, enableVideo });
+    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$3.warn(error);
+      logger.warn(error);
       return Promise.reject(new Error(error));
     }
     const options = {
@@ -5776,15 +5776,15 @@ class WebRTCClient extends Emitter {
     return session.accept(options).then(() => {
       if (session.isCanceled) {
         const message = 'accepted a canceled session (or was canceled during the accept phase).';
-        logger$3.error(message, { id: session.id });
+        logger.error(message, { id: session.id });
         this.onCallEnded(session);
 
         throw new CanceledCallError(message);
       }
-      logger$3.info('sdk webrtc answer, accepted.');
+      logger.info('sdk webrtc answer, accepted.');
       this._onAccepted(session);
     }).catch(e => {
-      logger$3.error(`answer call error for ${session ? session.id : 'n/a'}`, e);
+      logger.error(`answer call error for ${session ? session.id : 'n/a'}`, e);
 
       throw e;
     });
@@ -5792,7 +5792,7 @@ class WebRTCClient extends Emitter {
 
   hangup(session         ) {
     const { state, id } = session;
-    logger$3.info('sdk webrtc hangup call', { id, state });
+    logger.info('sdk webrtc hangup call', { id, state });
 
     try {
       this._stopSendingStats(session);
@@ -5821,7 +5821,7 @@ class WebRTCClient extends Emitter {
 
       return bye();
     } catch (error) {
-      logger$3.warn('sdk webrtc hangup, error', error);
+      logger.warn('sdk webrtc hangup, error', error);
     }
 
     return null;
@@ -5839,7 +5839,7 @@ class WebRTCClient extends Emitter {
   // Fetch and emit an event at `interval` with session network stats
   startNetworkMonitoring(session         , interval         = 1000)          {
     const sessionId = this.getSipSessionId(session);
-    logger$3.info('starting network inspection', { id: sessionId });
+    logger.info('starting network inspection', { id: sessionId });
 
     this.sessionNetworkStats[sessionId] = [];
 
@@ -5849,7 +5849,7 @@ class WebRTCClient extends Emitter {
   stopNetworkMonitoring(session         )          {
     const sessionId = this.getSipSessionId(session);
     const exists = sessionId in this.networkMonitoringInterval;
-    logger$3.info('stopping network inspection', { id: sessionId, exists });
+    logger.info('stopping network inspection', { id: sessionId, exists });
 
     if (exists) {
       clearInterval(this.networkMonitoringInterval[sessionId]);
@@ -5859,17 +5859,17 @@ class WebRTCClient extends Emitter {
   }
 
   reject(session         ) {
-    logger$3.info('sdk webrtc reject call', { id: session.id });
+    logger.info('sdk webrtc reject call', { id: session.id });
 
     try {
       return session.reject ? session.reject() : session.cancel();
     } catch (e) {
-      logger$3.warn('Error when rejecting call', e.message, e.stack);
+      logger.warn('Error when rejecting call', e.message, e.stack);
     }
   }
 
   async close() {
-    logger$3.info('sdk webrtc closing client', { userAgent: !!this.userAgent });
+    logger.info('sdk webrtc closing client', { userAgent: !!this.userAgent });
     this._cleanupMedia();
     this.connectionPromise = null;
 
@@ -5900,7 +5900,7 @@ class WebRTCClient extends Emitter {
       // eg: "INVITE not rejectable in state Completed"
     }
     this.userAgent = null;
-    logger$3.info('sdk webrtc client closed');
+    logger.info('sdk webrtc client closed');
   }
 
   getNumber(session         )          {
@@ -5913,13 +5913,13 @@ class WebRTCClient extends Emitter {
   }
 
   mute(session         ) {
-    logger$3.info('sdk webrtc mute', { id: session.id });
+    logger.info('sdk webrtc mute', { id: session.id });
 
     this._toggleAudio(session, true);
   }
 
   unmute(session         ) {
-    logger$3.info('sdk webrtc unmute', { id: session.id });
+    logger.info('sdk webrtc unmute', { id: session.id });
 
     this._toggleAudio(session, false);
   }
@@ -5959,13 +5959,13 @@ class WebRTCClient extends Emitter {
   }
 
   toggleCameraOn(session         ) {
-    logger$3.info('sdk webrtc toggle camera on', { id: session.id });
+    logger.info('sdk webrtc toggle camera on', { id: session.id });
 
     this._toggleVideo(session, false);
   }
 
   toggleCameraOff(session         ) {
-    logger$3.info('sdk webrtc toggle camera off', { id: session.id });
+    logger.info('sdk webrtc toggle camera off', { id: session.id });
 
     this._toggleVideo(session, true);
   }
@@ -5974,7 +5974,7 @@ class WebRTCClient extends Emitter {
     const sessionId = this.getSipSessionId(session);
     const hasVideo = hadVideo || this.hasLocalVideo(sessionId);
 
-    logger$3.info('sdk webrtc hold', {
+    logger.info('sdk webrtc hold', {
       sessionId,
       keys: Object.keys(this.heldSessions),
       pendingReinvite: !!session.pendingReinvite,
@@ -6023,7 +6023,7 @@ class WebRTCClient extends Emitter {
     const sessionId = this.getSipSessionId(session);
     const hasVideo = sessionId in this.heldSessions && this.heldSessions[sessionId].hasVideo;
 
-    logger$3.info('sdk webrtc unhold', {
+    logger.info('sdk webrtc unhold', {
       sessionId,
       keys: Object.keys(this.heldSessions),
       pendingReinvite: !!session.pendingReinvite,
@@ -6171,7 +6171,7 @@ class WebRTCClient extends Emitter {
     if (!session.sessionDescriptionHandler) {
       return;
     }
-    logger$3.info('Sending DTMF', {
+    logger.info('Sending DTMF', {
       id: this.getSipSessionId(session),
       tone,
     });
@@ -6187,7 +6187,7 @@ class WebRTCClient extends Emitter {
   transfer(session         , target        ) {
     this.hold(session);
 
-    logger$3.info('Transfering a session', {
+    logger.info('Transfering a session', {
       id: this.getSipSessionId(session),
       target,
     });
@@ -6209,17 +6209,17 @@ class WebRTCClient extends Emitter {
   atxfer(session         ) {
     this.hold(session);
 
-    logger$3.info('webrtc transfer started', { id: this.getSipSessionId(session) });
+    logger.info('webrtc transfer started', { id: this.getSipSessionId(session) });
 
     const result         = {
       newSession: null,
       init: async (target        ) => {
-        logger$3.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: () => {
-        logger$3.info('webrtc transfer completed', {
+        logger.info('webrtc transfer completed', {
           id: this.getSipSessionId(session),
           referId: this.getSipSessionId(result.newSession),
         });
@@ -6240,7 +6240,7 @@ class WebRTCClient extends Emitter {
       return;
     }
 
-    logger$3.info('send WebRTC message', { sipId: sipSession.id, contentType });
+    logger.info('send WebRTC message', { sipId: sipSession.id, contentType });
 
     try {
       sipSession.message({
@@ -6282,7 +6282,7 @@ class WebRTCClient extends Emitter {
   }
 
   changeAudioOutputVolume(volume        ) {
-    logger$3.info('Changing audio output volume', { volume });
+    logger.info('Changing audio output volume', { volume });
 
     Object.values(this.audioElements).forEach(audioElement => {
       if (audioElement instanceof HTMLAudioElement) {
@@ -6294,7 +6294,7 @@ class WebRTCClient extends Emitter {
   }
 
   changeAudioOutputDevice(id        ) {
-    logger$3.info('Changing audio output device', { id });
+    logger.info('Changing audio output device', { id });
 
     Object.values(this.audioElements).forEach(audioElement => {
       // `setSinkId` method is not included in any flow type definitions for HTMLAudioElements but is a valid method
@@ -6311,7 +6311,7 @@ class WebRTCClient extends Emitter {
 
   changeAudioInputDevice(id        , session          ) {
     const currentId = this.getAudioDeviceId();
-    logger$3.info('setting audio input device', { id, currentId, session: !!session });
+    logger.info('setting audio input device', { id, currentId, session: !!session });
 
     if (id === currentId) {
       return null;
@@ -6357,7 +6357,7 @@ class WebRTCClient extends Emitter {
 
   setVideoInputDevice(id         ) {
     const currentId = this.getVideoDeviceId();
-    logger$3.info('setting video input device', { id, currentId });
+    logger.info('setting video input device', { id, currentId });
 
     if (id === currentId) {
       return null;
@@ -6382,7 +6382,7 @@ class WebRTCClient extends Emitter {
     const sessionId = this.getSipSessionId(session);
     const localStream = this.getLocalStream(sessionId);
 
-    logger$3.info('changing video input device', { id, sessionId });
+    logger.info('changing video input device', { id, sessionId });
 
     // Release old video stream
     if (localStream) {
@@ -6450,7 +6450,7 @@ class WebRTCClient extends Emitter {
     const shouldDoScreenSharing = newConstraints && newConstraints.screen;
     const desktop = newConstraints && newConstraints.desktop;
 
-    logger$3.info('Sending reinvite', {
+    logger.info('Sending reinvite', {
       id: this.getSipSessionId(sipSession),
       newConstraints,
       conference,
@@ -6488,7 +6488,7 @@ class WebRTCClient extends Emitter {
             sipSession.incomingInviteRequest.message.body = response.message.body;
           }
 
-          logger$3.info('on re-INVITE accepted', {
+          logger.info('on re-INVITE accepted', {
             id: this.getSipSessionId(sipSession),
             wasMuted,
             shouldDoScreenSharing,
@@ -6663,7 +6663,7 @@ class WebRTCClient extends Emitter {
   }
 
   startHeartbeat() {
-    logger$3.info('sdk webrtc start heartbeat', { userAgent: !!this.userAgent });
+    logger.info('sdk webrtc start heartbeat', { userAgent: !!this.userAgent });
     if (!this.userAgent) {
       this.heartbeat.stop();
       return;
@@ -6676,7 +6676,7 @@ class WebRTCClient extends Emitter {
   }
 
   stopHeartbeat() {
-    logger$3.info('sdk webrtc stop heartbeat');
+    logger.info('sdk webrtc stop heartbeat');
     this.heartbeat.stop();
   }
 
@@ -6697,7 +6697,7 @@ class WebRTCClient extends Emitter {
   }
 
   attemptReconnection()       {
-    logger$3.info('attempt reconnection', { userAgent: !!this.userAgent });
+    logger.info('attempt reconnection', { userAgent: !!this.userAgent });
     if (!this.userAgent) {
       return;
     }
@@ -6722,7 +6722,7 @@ class WebRTCClient extends Emitter {
       return;
     }
 
-    logger$3.info('setting local media stream', {
+    logger.info('setting local media stream', {
       sipSessionId,
       tracks: newStream ? newStream.getTracks() : null,
     });
@@ -6739,7 +6739,7 @@ class WebRTCClient extends Emitter {
 
     const oldStream = this.getLocalStream(sipSessionId);
 
-    logger$3.info('updating local stream', {
+    logger.info('updating local stream', {
       sipSessionId,
       oldStream: !!oldStream,
       tracks: newStream.getTracks(),
@@ -6759,7 +6759,7 @@ class WebRTCClient extends Emitter {
     const remoteStream = this.getRemoteStream(sessionId);
     const pc = this.getPeerConnection(sessionId);
 
-    logger$3.info('Updating remote stream', {
+    logger.info('Updating remote stream', {
       sessionId,
       tracks: remoteStream ? remoteStream.getTracks() : null,
       receiverTracks: pc && pc.getReceivers ? pc.getReceivers().map(receiver => receiver.track) : null,
@@ -6784,14 +6784,16 @@ class WebRTCClient extends Emitter {
     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$3.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' } }) : { cursor: 'always' })
+        video: screenSharing ? (isDesktop
+          ? ({ mandatory: { chromeMediaSource: 'desktop', ...(mandatoryVideo || {}) } }) : { cursor: 'always' })
           : this._getVideoConstraints(enableVideo),
         screen: screenSharing,
         desktop: isDesktop,
@@ -6834,7 +6836,7 @@ class WebRTCClient extends Emitter {
   }
 
   _onTransportError() {
-    logger$3.error('on transport error');
+    logger.error('on transport error');
     this.eventEmitter.emit(TRANSPORT_ERROR);
     this.attemptReconnection();
   }
@@ -6843,7 +6845,7 @@ class WebRTCClient extends Emitter {
     const body = message && typeof message === 'object' ? message.data : message;
     if (body && body.indexOf('200 OK') !== -1) {
       if (this.hasHeartbeat()) {
-        logger$3.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) {
@@ -6854,7 +6856,7 @@ class WebRTCClient extends Emitter {
   }
 
   async _onHeartbeatTimeout() {
-    logger$3.info('sdk webrtc heartbeat timed out', { userAgent: !!this.userAgent, cb: !!this.heartbeatTimeoutCb });
+    logger.info('sdk webrtc heartbeat timed out', { userAgent: !!this.userAgent, cb: !!this.heartbeatTimeoutCb });
 
     if (this.heartbeatTimeoutCb) {
       this.heartbeatTimeoutCb();
@@ -6866,7 +6868,7 @@ class WebRTCClient extends Emitter {
         try {
           await this.userAgent.transport.disconnect();
         } catch (e) {
-          logger$3.error('Transport disconnection after heartbeat timeout, error', e);
+          logger.error('Transport disconnection after heartbeat timeout, error', e);
         }
       }
 
@@ -6899,31 +6901,31 @@ class WebRTCClient extends Emitter {
   }
 
   _connectIfNeeded()                {
-    logger$3.info('connect if needed, checking', { connected: this.isConnected() });
+    logger.info('connect if needed, checking', { connected: this.isConnected() });
     if (!this.userAgent) {
-      logger$3.info('need to recreate User Agent');
+      logger.info('need to recreate User Agent');
       this.userAgent = this.createUserAgent(this.uaConfigOverrides);
     }
 
     if (this.isConnected()) {
-      logger$3.info('webrtc sdk, already connected');
+      logger.info('webrtc sdk, already connected');
       return Promise.resolve();
     }
 
     if (this.isConnecting()) {
-      logger$3.info('webrtc sdk, already connecting...');
+      logger.info('webrtc sdk, already connecting...');
 
       this.connectionPromise = this.userAgent.transport.connectPromise;
       return this.connectionPromise;
     }
 
     if (this.connectionPromise) {
-      logger$3.info('webrtc sdk, connection promise connecting...');
+      logger.info('webrtc sdk, connection promise connecting...');
       // $FlowFixMe
       return this.connectionPromise;
     }
 
-    logger$3.info('WebRTC UA needs to connect');
+    logger.info('WebRTC UA needs to connect');
 
     // Force UA to reconnect
     if (this.userAgent && this.userAgent.state !== UserAgentState.Stopped) {
@@ -7048,7 +7050,7 @@ class WebRTCClient extends Emitter {
 
       sdh.peerConnectionDelegate = {
         onicecandidateerror: (error        ) => {
-          logger$3.error('on icecandidate error', {
+          logger.error('on icecandidate error', {
             address: error.address,
             port: error.port,
             errorCode: error.errorCode,
@@ -7077,7 +7079,7 @@ class WebRTCClient extends Emitter {
         updatedNumber = session.assertedIdentity.uri.normal.user;
         updatedCalleeName = session.assertedIdentity.displayName || updatedNumber;
       }
-      logger$3.info('re-invite received', { updatedCalleeName, updatedNumber, hadRemoteVideo });
+      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
 
@@ -7098,7 +7100,7 @@ class WebRTCClient extends Emitter {
   }
 
   _onAccepted(session         , sessionDialog                , withEvent          = true) {
-    logger$3.info('on call accepted', { id: session.id, remoteTag: session.remoteTag });
+    logger.info('on call accepted', { id: session.id, remoteTag: session.remoteTag });
 
     this.storeSipSession(session);
 
@@ -7110,7 +7112,7 @@ class WebRTCClient extends Emitter {
     const onTrack = (event     ) => {
       const isAudioOnly = this._isAudioOnly(session);
       const { kind, label, readyState, id, muted } = event.track;
-      logger$3.info('on track event', { isAudioOnly, kind, label, readyState, id, muted });
+      logger.info('on track event', { isAudioOnly, kind, label, readyState, id, muted });
 
       // Stop video track in audio only mode
       if (isAudioOnly && kind === 'video') {
@@ -7126,12 +7128,14 @@ class WebRTCClient extends Emitter {
 
     session.sessionDescriptionHandler.remoteMediaStream.onaddtrack = onTrack;
 
-    pc.oniceconnectionstatechange = () => {
-      logger$3.info('on ice connection state changed', { state: pc.iceConnectionState });
-      if (pc.iceConnectionState === 'disconnected') {
-        this.eventEmitter.emit(ON_DISCONNECTED);
-      }
-    };
+    if (pc) {
+      pc.oniceconnectionstatechange = () => {
+        logger.info('on ice connection state changed', { state: pc.iceConnectionState });
+        if (pc.iceConnectionState === 'disconnected') {
+          this.eventEmitter.emit(ON_DISCONNECTED);
+        }
+      };
+    }
 
     if (withEvent) {
       this.eventEmitter.emit(ACCEPTED, session, sessionDialog);
@@ -7300,10 +7304,10 @@ class WebRTCClient extends Emitter {
 
   _stopSendingStats(session         ) {
     const sessionId = this.getSipSessionId(session);
-    logger$3.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$3.trace('Stop sending stats for call', { sessionId });
+      logger.trace('Stop sending stats for call', { sessionId });
       this.statsIntervals[sessionId]();
       delete this.statsIntervals[sessionId];
     }
@@ -7319,7 +7323,7 @@ class WebRTCClient extends Emitter {
       try {
         await this.userAgent.transport.disconnect();
       } catch (e) {
-        logger$3.error('WebRTC transport disconnect, error', e);
+        logger.error('WebRTC transport disconnect, error', e);
       }
     }
   }
@@ -7341,13 +7345,17 @@ class WebRTCClient extends Emitter {
     const nbStats = this.sessionNetworkStats[sessionId].length;
     const lastNetworkStats = this.sessionNetworkStats[sessionId][nbStats - 1];
     const lastAudioSent = lastNetworkStats ? lastNetworkStats.audioBytesSent : 0;
+    const lastAudioContentSent = lastNetworkStats ? lastNetworkStats.audioContentSent : 0;
     const lastVideoSent = lastNetworkStats ? lastNetworkStats.videoBytesSent : 0;
     const lastAudioReceived = lastNetworkStats ? lastNetworkStats.audioBytesReceived : 0;
+    const lastAudioContentReceived = lastNetworkStats ? lastNetworkStats.audioContentReceived : 0;
     const lastVideoReceived = lastNetworkStats ? lastNetworkStats.videoBytesReceived : 0;
     const lastTransportSent = lastNetworkStats ? lastNetworkStats.transportSent : 0;
     const lastTransportReceived = lastNetworkStats ? lastNetworkStats.transportReceived : 0;
-    let audioBytesSent = 0;
-    let audioBytesReceived = 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;
@@ -7357,6 +7365,7 @@ class WebRTCClient extends Emitter {
     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);
@@ -7368,6 +7377,7 @@ class WebRTCClient extends Emitter {
         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);
@@ -7396,10 +7406,17 @@ class WebRTCClient extends Emitter {
     });
 
     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;