@wazo/sdk 0.37.24 → 0.37.25

package/esm/domain/Phone/WebRTCPhone.js

@@ -401,7 +401,7 @@ const isMobile = () => typeof navigator !== 'undefined' && navigator.product ===
 
 /*       */
 
-const logger = IssueReporter$1 ? IssueReporter$1.loggerFor('api') : console;
+IssueReporter$1 ? IssueReporter$1.loggerFor('api') : console;
 
 // Use a function here to be able to mock it in tests
 const realFetch = () => {
@@ -786,7 +786,7 @@ var IssueReporter$1 = new IssueReporter();
 
 /* global RTCSessionDescriptionInit, navigator */
 
-const wazoLogger = IssueReporter$1 ? IssueReporter$1.loggerFor('webrtc-sdh') : console;
+IssueReporter$1 ? IssueReporter$1.loggerFor('webrtc-sdh') : console;
 
 //      
 
@@ -828,7 +828,7 @@ BwIDAQAB
 
 //      
 
-const swarmKey = KEYUTIL.getKey(swarmPublicKey);
+KEYUTIL.getKey(swarmPublicKey);
 
 /* eslint-disable dot-notation */
 
@@ -849,15 +849,15 @@ global.wazoFetchOptions = global.wazoFetchOptions || {};
 
 // 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;
 
 /*       */
 
-const logger$2 = IssueReporter$1 ? IssueReporter$1.loggerFor('api') : console;
+IssueReporter$1 ? IssueReporter$1.loggerFor('api') : console;
 
 //      
-const logger$3 = IssueReporter$1 ? IssueReporter$1.loggerFor('webrtc-client') : console;
-const statsLogger = IssueReporter$1 ? IssueReporter$1.loggerFor('webrtc-stats') : console;
+IssueReporter$1 ? IssueReporter$1.loggerFor('webrtc-client') : console;
+IssueReporter$1 ? IssueReporter$1.loggerFor('webrtc-stats') : console;
 
 /* eslint-disable no-param-reassign */
 
@@ -943,7 +943,7 @@ const events = [
   ON_DISCONNECTED,
 ];
 
-const logger$4 = IssueReporter$1.loggerFor('webrtc-phone');
+const logger = IssueReporter$1.loggerFor('webrtc-phone');
 
 class WebRTCPhone extends Emitter                  {
                            
@@ -1016,7 +1016,7 @@ class WebRTCPhone extends Emitter                  {
     }).catch(error => {
       // Avoid exception on `t.server.scheme` in sip transport when losing the webrtc socket connection
       console.error('register error', error, error.message, error.stack);
-      logger$4.error('WebRTC register error', { message: error.message, stack: error.stack });
+      logger.error('WebRTC register error', { message: error.message, stack: error.stack });
     });
   }
 
@@ -1031,7 +1031,7 @@ class WebRTCPhone extends Emitter                  {
     if (!this.client) {
       return;
     }
-    logger$4.info('WebRTC phone stop');
+    logger.info('WebRTC phone stop');
     this.client.stop();
   }
 
@@ -1046,11 +1046,12 @@ class WebRTCPhone extends Emitter                  {
   async sendReinvite(callSession              , constraints          = null, conference          = false,
     audioOnly          = false, iceRestart          = false) {
     const sipSession = this.findSipSession(callSession);
-    logger$4.info('WebRTC phone - send reinvite', {
+    logger.info('WebRTC phone - send reinvite', {
       sessionId: sipSession ? sipSession.id : null,
       constraints,
       audioOnly,
       conference,
+      iceRestart,
     });
 
     if (!sipSession) {
@@ -1083,7 +1084,7 @@ class WebRTCPhone extends Emitter                  {
   }
 
   startHeartbeat() {
-    logger$4.info('WebRTC phone - start heartbeat', { client: !!this.client, hasHeartbeat: this.client.hasHeartbeat() });
+    logger.info('WebRTC phone - start heartbeat', { client: !!this.client, hasHeartbeat: this.client.hasHeartbeat() });
     if (!this.client || this.client.hasHeartbeat()) {
       return;
     }
@@ -1092,7 +1093,7 @@ class WebRTCPhone extends Emitter                  {
   }
 
   stopHeartbeat() {
-    logger$4.info('WebRTC phone - stopHeartbeat', { client: !!this.client });
+    logger.info('WebRTC phone - stopHeartbeat', { client: !!this.client });
     if (!this.client) {
       return;
     }
@@ -1109,7 +1110,7 @@ class WebRTCPhone extends Emitter                  {
   }
 
   reconnect() {
-    logger$4.info('WebRTC phone - reconnect', { client: !!this.client });
+    logger.info('WebRTC phone - reconnect', { client: !!this.client });
     if (!this.client) {
       return;
     }
@@ -1150,7 +1151,7 @@ class WebRTCPhone extends Emitter                  {
       return false;
     }
 
-    logger$4.info('Downgrade to video', { id: callSession ? callSession.getId() : null, withMessage });
+    logger.info('Downgrade to video', { id: callSession ? callSession.getId() : null, withMessage });
 
     this.client.downgradeToAudio(sipSession);
 
@@ -1173,7 +1174,7 @@ class WebRTCPhone extends Emitter                  {
       return Promise.resolve(false);
     }
 
-    logger$4.info('Upgrade to video', {
+    logger.info('Upgrade to video', {
       id: callSession ? callSession.getId() : null,
       constraints,
       isConference,
@@ -1213,7 +1214,7 @@ class WebRTCPhone extends Emitter                  {
 
   _sendReinviteMessage(callSession              , isUpgrade         ) {
     const sipSession = this.findSipSession(callSession);
-    logger$4.info('Sending reinvite message', {
+    logger.info('Sending reinvite message', {
       id: callSession ? callSession.getId() : null,
       isUpgrade,
     });
@@ -1239,7 +1240,7 @@ class WebRTCPhone extends Emitter                  {
       // Monkey patch to know when canceled with the CANCEL message
       const onCancel = sipSession._onCancel.bind(sipSession);
       sipSession._onCancel = (message                        ) => {
-        logger$4.trace('on sip session canceled', { callId: message.callId });
+        logger.trace('on sip session canceled', { callId: message.callId });
 
         onCancel(message);
         const elsewhere = message.data.indexOf('cause=26') !== -1 && message.data.indexOf('completed elsewhere') !== -1;
@@ -1267,13 +1268,13 @@ class WebRTCPhone extends Emitter                  {
             this.audioOutputVolume,
           );
         case SessionState.Terminating:
-          logger$4.info('WebRTC phone - call terminating', { sipId: sipSession.id });
+          logger.info('WebRTC phone - call terminating', { sipId: sipSession.id });
 
           this.eventEmitter.emit(ON_CALL_ENDING, this._createCallSession(sipSession));
 
           break;
         case SessionState.Terminated: {
-          logger$4.info('WebRTC phone - call terminated', { sipId: sipSession.id });
+          logger.info('WebRTC phone - call terminated', { sipId: sipSession.id });
 
           // Should be called before `_onCallTerminated` or the callCount will not decrement...
           const callSession = this._createCallSession(sipSession);
@@ -1295,7 +1296,7 @@ class WebRTCPhone extends Emitter                  {
       const [stream] = event.streams;
       const kind = event && event.track && event.track.kind;
 
-      logger$4.info('on track stream called on the peer connection', {
+      logger.info('on track stream called on the peer connection', {
         callId: this.getSipSessionId(sipSession),
         streamId: stream ? stream.id : null,
         tracks: stream ? stream.getTracks() : null,
@@ -1315,7 +1316,7 @@ class WebRTCPhone extends Emitter                  {
     };
 
     peerConnection.onremovestream = event => {
-      logger$4.info('on remove stream called on the peer connection', {
+      logger.info('on remove stream called on the peer connection', {
         id: event.stream.id,
         tracks: event.stream.getTracks(),
       });
@@ -1325,7 +1326,7 @@ class WebRTCPhone extends Emitter                  {
   }
 
   async startScreenSharing(constraints        , callSession              ) {
-    logger$4.info('WebRTC - start screen sharing', { constraints, id: callSession ? callSession.getId() : null });
+    logger.info('WebRTC - start screen sharing', { constraints, id: callSession ? callSession.getId() : null });
 
     const screenShareStream = await this.client.getStreamFromConstraints(constraints);
     if (!screenShareStream) {
@@ -1365,7 +1366,7 @@ class WebRTCPhone extends Emitter                  {
       return;
     }
 
-    logger$4.info('WebRTC phone - stop screen sharing', {
+    logger.info('WebRTC phone - stop screen sharing', {
       restoreLocalStream,
     });
 
@@ -1421,7 +1422,7 @@ class WebRTCPhone extends Emitter                  {
     const pc = this.client.getPeerConnection(sipSessionId);
     const sender = pc && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
 
-    logger$4.info('WebRTC phone - on screensharing', {
+    logger.info('WebRTC phone - on screensharing', {
       hadVideo,
       id: this.getSipSessionId(sipSession),
       screenTrack,
@@ -1429,7 +1430,7 @@ class WebRTCPhone extends Emitter                  {
 
     if (screenTrack) {
       screenTrack.onended = () => {
-        logger$4.info('WebRTC phone - on screenshare ended', {
+        logger.info('WebRTC phone - on screenshare ended', {
           hadVideo,
           id: this.getSipSessionId(sipSession),
           screenTrack,
@@ -1460,7 +1461,7 @@ class WebRTCPhone extends Emitter                  {
   }
 
   _onCallAccepted(sipSession         , cameraEnabled         )              {
-    logger$4.info('WebRTC phone -  on call accepted', { sipId: sipSession.id, cameraEnabled });
+    logger.info('WebRTC phone - on call accepted', { sipId: sipSession.id, cameraEnabled });
 
     const callSession = this._createAcceptedCallSession(sipSession, cameraEnabled);
     this.currentSipSession = sipSession;
@@ -1478,7 +1479,7 @@ class WebRTCPhone extends Emitter                  {
   }
 
   changeAudioDevice(id        ) {
-    logger$4.info('WebRTC phone - change audio device', { deviceId: id });
+    logger.info('WebRTC phone - change audio device', { deviceId: id });
     this.audioOutputDeviceId = id;
     this.client.changeAudioOutputDevice(id);
   }
@@ -1488,14 +1489,14 @@ class WebRTCPhone extends Emitter                  {
   }
 
   changeRingDevice(id        ) {
-    logger$4.info('WebRTC phone - changing ring device', { id });
+    logger.info('WebRTC phone - changing ring device', { id });
 
     this.audioRingDeviceId = id;
   }
 
   // volume is a value between 0 and 1
   changeAudioVolume(volume        ) {
-    logger$4.info('WebRTC phone - changing audio volume', { volume });
+    logger.info('WebRTC phone - changing audio volume', { volume });
 
     this.audioOutputVolume = volume;
     this.client.changeAudioOutputVolume(volume);
@@ -1503,31 +1504,31 @@ class WebRTCPhone extends Emitter                  {
 
   // volume is a value between 0 and 1
   changeRingVolume(volume        ) {
-    logger$4.info('WebRTC phone - changing ring volume', { volume });
+    logger.info('WebRTC phone - changing ring volume', { volume });
 
     this.audioRingVolume = volume;
   }
 
   changeAudioInputDevice(id        ) {
-    logger$4.info('WebRTC phone - changeAudio input device', { deviceId: id });
+    logger.info('WebRTC phone - changeAudio input device', { deviceId: id });
 
     return this.client.changeAudioInputDevice(id, this.currentSipSession);
   }
 
   changeVideoInputDevice(id         ) {
-    logger$4.info('WebRTC phone - change video input device', { deviceId: id });
+    logger.info('WebRTC phone - change video input device', { deviceId: id });
 
     return this.client.changeVideoInputDevice(id, this.currentSipSession);
   }
 
   changeSessionVideoInputDevice(id         ) {
-    logger$4.info('WebRTC phone - change session video input device', { deviceId: id });
+    logger.info('WebRTC phone - change session video input device', { deviceId: id });
 
     return this.client.changeSessionVideoInputDevice(id, this.currentSipSession);
   }
 
   _onCallTerminated(sipSession         ) {
-    logger$4.info('WebRTC phone - on call terminated', { sipId: sipSession.id });
+    logger.info('WebRTC phone - on call terminated', { sipId: sipSession.id });
 
     const callSession = this._createCallSession(sipSession);
     const isCurrentSession = this.isCurrentCallSipSession(callSession);
@@ -1561,7 +1562,7 @@ class WebRTCPhone extends Emitter                  {
     const hasIncomingCallSession = this.hasIncomingCallSession();
 
     // Re-trigger incoming call event for remaining incoming calls
-    logger$4.info('WebRTC phone - check to re-trigger incoming call', {
+    logger.info('WebRTC phone - check to re-trigger incoming call', {
       sipId: sipSession.id,
       shouldRetrigger,
       hasIncomingCallSession,
@@ -1652,37 +1653,37 @@ class WebRTCPhone extends Emitter                  {
 
   // Deprecated
   getLocalStreamForCall(callSession             )               {
-    logger$4.warn('WebRTCPhone.getLocalStreamForCall is deprecated, use WebRTCPhone.getLocalStream instead');
+    logger.warn('WebRTCPhone.getLocalStreamForCall is deprecated, use WebRTCPhone.getLocalStream instead');
 
     return this.getLocalStream(callSession);
   }
 
   // Deprecated
   getRemoteStreamForCall(callSession             )               {
-    logger$4.warn('WebRTCPhone.getRemoteStreamForCall is deprecated, use WebRTCPhone.getRemoteStream instead');
+    logger.warn('WebRTCPhone.getRemoteStreamForCall is deprecated, use WebRTCPhone.getRemoteStream instead');
 
     return this.getRemoteVideoStream(callSession);
   }
 
   getRemoteStreamsForCall(callSession             )               {
-    logger$4.warn('WebRTCPhone.getRemoteStreamsForCall is deprecated, use WebRTCPhone.getRemoteStreams instead');
+    logger.warn('WebRTCPhone.getRemoteStreamsForCall is deprecated, use WebRTCPhone.getRemoteStreams instead');
 
     return this.getRemoteStream(callSession);
   }
 
   accept(callSession             , cameraEnabled          )                         {
-    logger$4.info('WebRTC phone - accept call', { id: callSession ? callSession.getId() : 'n/a', cameraEnabled });
+    logger.info('WebRTC phone - accept call', { id: callSession ? callSession.getId() : 'n/a', cameraEnabled });
     if (this.currentSipSession) {
       this.holdSipSession(this.currentSipSession, this.currentCallSession, true);
     }
 
     if (!callSession) {
-      logger$4.warn('no CallSession to accept.');
+      logger.warn('no CallSession to accept.');
       return Promise.resolve(null);
     }
 
     if (callSession.getId() in this.acceptedSessions) {
-      logger$4.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
+      logger.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
       return Promise.resolve(callSession.sipCallId);
     }
 
@@ -1694,24 +1695,24 @@ class WebRTCPhone extends Emitter                  {
     const sipSession = this.client.getSipSession(callSession.getId());
     if (sipSession) {
       if (sipSession.state === SessionState.Terminated || sipSession.state === SessionState.Terminating) {
-        logger$4.warn('Trying to answer a terminated sipSession.');
+        logger.warn('Trying to answer a terminated sipSession.');
 
         return Promise.resolve(null);
       }
 
-      logger$4.info('accept call, session found', { sipId: sipSession.id });
+      logger.info('accept call, session found', { sipId: sipSession.id });
 
       // $FlowFixMe
       return this.client.answer(sipSession, this.allowVideo ? cameraEnabled : false).then(() => {
         return callSession.sipCallId;
       }).catch(e => {
-        logger$4.error(e);
+        logger.error(e);
         this.endCurrentCall(callSession);
         throw e;
       });
     }
 
-    logger$4.warn('no CallSession found to accept.', { id: callSession ? callSession.getId() : 'n/a' });
+    logger.warn('no CallSession found to accept.', { id: callSession ? callSession.getId() : 'n/a' });
 
     return Promise.resolve(null);
   }
@@ -1721,7 +1722,7 @@ class WebRTCPhone extends Emitter                  {
   }
 
   async reject(callSession             )                {
-    logger$4.info('reject WebRTC called', { id: callSession.getId() });
+    logger.info('reject WebRTC called', { id: callSession.getId() });
 
     this.eventEmitter.emit(ON_TERMINATE_SOUND, callSession, 'call rejected locally');
     if (!callSession || callSession.getId() in this.rejectedSessions) {
@@ -1733,14 +1734,14 @@ class WebRTCPhone extends Emitter                  {
 
     const sipSession = this.findSipSession(callSession);
     if (sipSession) {
-      logger$4.info('WebRTC rejecting', { sipId: sipSession.id });
+      logger.info('WebRTC rejecting', { sipId: sipSession.id });
 
       this.client.hangup(sipSession);
     }
   }
 
   async ignore(callSession             )                {
-    logger$4.info('WebRTC ignore', { id: callSession.getId() });
+    logger.info('WebRTC ignore', { id: callSession.getId() });
 
     // kill the ring
     this.eventEmitter.emit(ON_TERMINATE_SOUND, callSession, 'ignoring call');
@@ -1752,14 +1753,14 @@ class WebRTCPhone extends Emitter                  {
     const sipSession = this.findSipSession(callSession);
 
     if (sipSession) {
-      logger$4.info('WebRTC atxfer', { sipId: sipSession.id });
+      logger.info('WebRTC atxfer', { sipId: sipSession.id });
 
       return this.client.atxfer(sipSession);
     }
   }
 
   hold(callSession             , withEvent          = true, isConference          = false)                {
-    logger$4.info('WebRTC hold', { id: callSession.getId() });
+    logger.info('WebRTC hold', { id: callSession.getId() });
 
     const sipSession = this.findSipSession(callSession);
 
@@ -1774,18 +1775,18 @@ class WebRTCPhone extends Emitter                  {
   }
 
   resume(callSession              , isConference          = false, withEvent          = true)               {
-    logger$4.info('WebRTC resume called', { id: callSession ? callSession.getId() : null });
+    logger.info('WebRTC resume called', { id: callSession ? callSession.getId() : null });
 
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return new Promise((resolve, reject) => reject(new Error('No session to resume')));
     }
 
-    logger$4.info('WebRTC resuming', { sipId: sipSession.id });
+    logger.info('WebRTC resuming', { sipId: sipSession.id });
 
     // Hold current session if different from the current one (we don't want 2 sessions active at the same time).
     if (this.currentSipSession && this.currentSipSession.id !== sipSession.id) {
-      logger$4.info('WebRTC hold call after resume', { id: this.currentSipSession.id });
+      logger.info('WebRTC hold call after resume', { id: this.currentSipSession.id });
       this.holdSipSession(this.currentSipSession, callSession);
     }
 
@@ -1810,7 +1811,7 @@ class WebRTCPhone extends Emitter                  {
     const sessionId = this.getSipSessionId(sipSession);
     const hasVideo = this.client.hasLocalVideo(sessionId);
 
-    logger$4.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
+    logger.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
 
     // Stop screenshre if needed
     if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
@@ -1841,7 +1842,7 @@ class WebRTCPhone extends Emitter                  {
       return new Promise((resolve, reject) => reject(new Error('No session to unhold')));
     }
     const sessionId = this.getSipSessionId(sipSession);
-    logger$4.info('WebRTC unhold sip session', { sessionId, isConference });
+    logger.info('WebRTC unhold sip session', { sessionId, isConference });
 
     const { hasVideo } = this.client.getHeldSession(sessionId) || {};
     const wasScreensharing = this.lastScreenShare && this.lastScreenShare.sipSessionId === sessionId;
@@ -1885,14 +1886,14 @@ class WebRTCPhone extends Emitter                  {
   }
 
   mute(callSession              , withEvent          = true)       {
-    logger$4.info('WebRTC mute called', { id: callSession ? callSession.getId() : null });
+    logger.info('WebRTC mute called', { id: callSession ? callSession.getId() : null });
 
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return;
     }
 
-    logger$4.info('WebRTC muting', { sipId: sipSession.id });
+    logger.info('WebRTC muting', { sipId: sipSession.id });
     this.client.mute(sipSession);
     const newCallSession = this._createCallSession(sipSession, callSession, { muted: true });
 
@@ -1906,14 +1907,14 @@ class WebRTCPhone extends Emitter                  {
   }
 
   unmute(callSession              , withEvent          = true)       {
-    logger$4.info('WebRTC unmute called', { id: callSession ? callSession.getId() : null });
+    logger.info('WebRTC unmute called', { id: callSession ? callSession.getId() : null });
 
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return;
     }
 
-    logger$4.info('WebRTC unmuting', { sipId: sipSession.id });
+    logger.info('WebRTC unmuting', { sipId: sipSession.id });
     this.client.unmute(sipSession);
     const newCallSession = this._createCallSession(sipSession, callSession, { muted: false });
 
@@ -1940,7 +1941,7 @@ class WebRTCPhone extends Emitter                  {
     if (!sipSession) {
       return;
     }
-    logger$4.info('WebRTC turn camera on', { sipId: sipSession.id });
+    logger.info('WebRTC turn camera on', { sipId: sipSession.id });
 
     this.client.toggleCameraOn(sipSession);
 
@@ -1952,7 +1953,7 @@ class WebRTCPhone extends Emitter                  {
     if (!sipSession) {
       return;
     }
-    logger$4.info('WebRTC turn camera off', { sipId: sipSession.id });
+    logger.info('WebRTC turn camera off', { sipId: sipSession.id });
 
     this.client.toggleCameraOff(sipSession);
     this.eventEmitter.emit(ON_CAMERA_DISABLED, this._createCameraDisabledCallSession(sipSession, callSession));
@@ -1964,7 +1965,7 @@ class WebRTCPhone extends Emitter                  {
       return;
     }
 
-    logger$4.info('WebRTC send key', { sipId: sipSession.id, tone });
+    logger.info('WebRTC send key', { sipId: sipSession.id, tone });
     this.client.sendDTMF(sipSession, tone);
   }
 
@@ -1972,7 +1973,7 @@ class WebRTCPhone extends Emitter                  {
   // @TODO: line is not used here
   async makeCall(number        , line     , cameraEnabled          ,
     audioOnly          = false, conference          = false)                        {
-    logger$4.info('make WebRTC call', { number, lineId: line ? line.id : null, cameraEnabled });
+    logger.info('make WebRTC call', { number, lineId: line ? line.id : null, cameraEnabled, audioOnly, conference });
     if (!number) {
       return Promise.resolve(null);
     }
@@ -1982,7 +1983,7 @@ class WebRTCPhone extends Emitter                  {
     }
     if (this.currentSipSession) {
       this.holdSipSession(this.currentSipSession, this.currentCallSession, true).catch(e => {
-        logger$4.warn('Unable to hold current session when making another call', e);
+        logger.warn('Unable to hold current session when making another call', e);
       });
     }
 
@@ -1992,7 +1993,7 @@ class WebRTCPhone extends Emitter                  {
       this._bindEvents(sipSession);
     } catch (error) {
       console.warn(error);
-      logger$4.warn('make WebRTC call, error', { message: error.message, stack: error.stack });
+      logger.warn('make WebRTC call, error', { message: error.message, stack: error.stack });
       return Promise.resolve(null);
     }
     const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
@@ -2020,7 +2021,7 @@ class WebRTCPhone extends Emitter                  {
       return;
     }
 
-    logger$4.info('WebRTC transfer', { sipId: sipSession.id, target });
+    logger.info('WebRTC transfer', { sipId: sipSession.id, target });
 
     this.client.transfer(sipSession, target);
   }
@@ -2029,7 +2030,7 @@ class WebRTCPhone extends Emitter                  {
     const sipSession = this.client.getSipSession(source.sipCallId);
     const sipSessionTarget = this.client.getSipSession(destination.sipCallId);
 
-    logger$4.info('WebRTC indirect transfer', {
+    logger.info('WebRTC indirect transfer', {
       sipId: sipSession ? sipSession.id : null,
       target: sipSessionTarget ? sipSessionTarget.id : null,
     });
@@ -2068,7 +2069,7 @@ class WebRTCPhone extends Emitter                  {
       return false;
     }
 
-    logger$4.info('WebRTC hangup', { sipId: sipSession.id });
+    logger.info('WebRTC hangup', { sipId: sipSession.id });
 
     const sipSessionId = this.getSipSessionId(sipSession);
 
@@ -2114,7 +2115,7 @@ class WebRTCPhone extends Emitter                  {
   endCurrentCall(callSession             )       {
     const isCurrent = this.isCurrentCallSipSession(callSession);
 
-    logger$4.info('Ending current call', {
+    logger.info('Ending current call', {
       id: callSession ? callSession.getId() : null,
       isCurrent,
     });
@@ -2138,7 +2139,7 @@ class WebRTCPhone extends Emitter                  {
   }
 
   async close()                {
-    logger$4.info('WebRTC close');
+    logger.info('WebRTC close');
 
     try {
       await Promise.race([
@@ -2146,7 +2147,7 @@ class WebRTCPhone extends Emitter                  {
         new Promise((resolve, reject) => setTimeout(() => reject(new Error('Unregister, timed out')), 3000)),
       ]);
     } catch (e) {
-      logger$4.error('WebRTC close, unregister error', e);
+      logger.error('WebRTC close, unregister error', e);
     }
 
     this.client.close();
@@ -2164,12 +2165,12 @@ class WebRTCPhone extends Emitter                  {
   }
 
   enableRinging()                       {
-    logger$4.info('WebRTC enable ringing');
+    logger.info('WebRTC enable ringing');
     this.ringingEnabled = true;
   }
 
   disableRinging()                       {
-    logger$4.info('WebRTC disable ringing');
+    logger.info('WebRTC disable ringing');
     this.ringingEnabled = false;
   }
 
@@ -2209,7 +2210,7 @@ class WebRTCPhone extends Emitter                  {
 
   // Deprecated
   getLocalMediaStream(callSession             ) {
-    logger$4.warn('WebRTCPhone.getLocalMediaStream is deprecated, use WebRTCPhone.getLocalStream instead');
+    logger.warn('WebRTCPhone.getLocalMediaStream is deprecated, use WebRTCPhone.getLocalStream instead');
     return this.getLocalStream(callSession);
   }
 
@@ -2251,7 +2252,7 @@ class WebRTCPhone extends Emitter                  {
     this.client.on(this.client.INVITE, (sipSession         , wantsToDoVideo         ) => {
       const autoAnswer = sipSession.request.getHeader('Answer-Mode') === 'Auto';
       const withVideo = this.allowVideo ? wantsToDoVideo : false;
-      logger$4.info('WebRTC invite received', { sipId: sipSession.id, withVideo, autoAnswer });
+      logger.info('WebRTC invite received', { sipId: sipSession.id, withVideo, autoAnswer });
 
       const callSession = this._createIncomingCallSession(sipSession, withVideo, null, autoAnswer);
       this.incomingSessions.push(callSession.getId());
@@ -2274,7 +2275,7 @@ class WebRTCPhone extends Emitter                  {
     });
 
     this.client.on(this.client.ON_REINVITE, (...args) => {
-      logger$4.info('WebRTC on reinvite', {
+      logger.info('WebRTC on reinvite', {
         sessionId: args[0].id,
         inviteId: args[1].id,
         updatedCalleeName: args[2],
@@ -2294,7 +2295,7 @@ class WebRTCPhone extends Emitter                  {
     });
 
     this.client.on(this.client.ACCEPTED, (sipSession         ) => {
-      logger$4.info('WebRTC call accepted', { sipId: sipSession.id });
+      logger.info('WebRTC call accepted', { sipId: sipSession.id });
 
       this._onCallAccepted(sipSession, this.client.hasVideo(this.getSipSessionId(sipSession)));
 
@@ -2305,22 +2306,22 @@ class WebRTCPhone extends Emitter                  {
     this.client.on('ended', () => {});
 
     this.client.on(this.client.ON_ERROR, e => {
-      logger$4.error('WebRTC error', e);
+      logger.error('WebRTC error', e);
       this.eventEmitter.emit(ON_CALL_ERROR, e);
     });
 
     this.client.on(this.client.UNREGISTERED, () => {
-      logger$4.info('WebRTC unregistered');
+      logger.info('WebRTC unregistered');
       this.eventEmitter.emit(ON_UNREGISTERED);
     });
 
     this.client.on(this.client.ON_DISCONNECTED, () => {
-      logger$4.info('WebRTC disconnected');
+      logger.info('WebRTC disconnected');
       this.eventEmitter.emit(ON_DISCONNECTED);
     });
 
     this.client.on(this.client.REGISTERED, () => {
-      logger$4.info('WebRTC registered', {
+      logger.info('WebRTC registered', {
         shouldSendReinvite: this.shouldSendReinvite,
         state: this.currentSipSession ? this.currentSipSession.state : null,
         currentSipSession: !!this.currentSipSession,
@@ -2341,19 +2342,19 @@ class WebRTCPhone extends Emitter                  {
           // Send reinvite with iceRestart
           this.sendReinvite(this.currentCallSession, null, isConference, !hasVideo, true);
         } catch (e) {
-          logger$4.error('WebRTC reinvite after register, error', { message: e.message, stack: e.stack });
+          logger.error('WebRTC reinvite after register, error', { message: e.message, stack: e.stack });
         }
       }
     });
 
     this.client.on(this.client.CONNECTED, () => {
-      logger$4.info('WebRTC client connected');
+      logger.info('WebRTC client connected');
 
       this.stopHeartbeat();
     });
 
     this.client.on(this.client.DISCONNECTED, () => {
-      logger$4.info('WebRTC client disconnected');
+      logger.info('WebRTC client disconnected');
 
       this.eventEmitter.emit(ON_UNREGISTERED);
 
@@ -2391,7 +2392,7 @@ class WebRTCPhone extends Emitter                  {
       const localStream = this.client.getLocalStream(sipSessionId);
       const callSession = this.callSessions[sipSessionId];
 
-      logger$4.info('Updrading directly in screensharing mode', {
+      logger.info('Updrading directly in screensharing mode', {
         sipSessionId,
         tracks: localStream ? localStream.getTracks() : null,
       });