@wazo/sdk 0.37.2 → 0.37.6

package/lib/simple/Phone.js

@@ -916,7 +916,7 @@ var IssueReporter$1 = new IssueReporter();
 
 const methods = ['head', 'get', 'post', 'put', 'delete'];
 
-const logger$b = IssueReporter$1 ? IssueReporter$1.loggerFor('api') : console;
+const logger$c = IssueReporter$1 ? IssueReporter$1.loggerFor('api') : console;
 
 // Use a function here to be able to mock it in tests
 const realFetch = () => {
@@ -1055,7 +1055,7 @@ class ApiRequester {
     };
 
     if (this.refreshTokenPromise) {
-      logger$b.info('A token is already refreshing, waiting ...', { url });
+      logger$c.info('A token is already refreshing, waiting ...', { url });
       await this.refreshTokenPromise;
     }
 
@@ -1075,7 +1075,7 @@ class ApiRequester {
         return promise.then(async (err        ) => {
           // Check if the token is still valid
           if (firstCall && this._checkTokenExpired(response, err)) {
-            logger$b.warn('token expired', { error: err.reason });
+            logger$c.warn('token expired', { error: err.reason });
             // Replay the call after refreshing the token
             return this._replayWithNewToken(err, path, method, body, headers, parse);
           }
@@ -1085,7 +1085,7 @@ class ApiRequester {
             : exceptionClass.fromResponse(err, response.status);
 
           if (this.shouldLogErrors) {
-            logger$b.error('API error', error);
+            logger$c.error('API error', error);
           }
 
           throw error;
@@ -1095,7 +1095,7 @@ class ApiRequester {
       return newParse(response, isJson);
     }).catch(error => {
       if (this.shouldLogErrors) {
-        logger$b.error('Fetch failed', { url, options, message: error.message, stack: error.stack });
+        logger$c.error('Fetch failed', { url, options, message: error.message, stack: error.stack });
       }
 
       throw error;
@@ -1123,13 +1123,13 @@ class ApiRequester {
   ) {
     const isTokenNotFound = this._isTokenNotFound(err);
     let newPath = path;
-    logger$b.info('refreshing token', { inProgress: !!this.refreshTokenPromise });
+    logger$c.info('refreshing token', { inProgress: !!this.refreshTokenPromise });
 
     this.refreshTokenPromise = this.refreshTokenPromise || this.refreshTokenCallback();
 
     return this.refreshTokenPromise.then(() => {
       this.refreshTokenPromise = null;
-      logger$b.info('token refreshed', { isTokenNotFound });
+      logger$c.info('token refreshed', { isTokenNotFound });
       if (isTokenNotFound) {
         const pathParts = path.split('/');
         pathParts.pop();
@@ -3060,6 +3060,65 @@ class ExternalApp {
   }
 }
 
+//      
+
+                                       
+                                  
+                   
+               
+                             
+               
+               
+                
+ 
+
+class Meeting {
+               
+
+                                
+                   
+               
+               
+               
+                 
+                            
+
+  static parse(plain                         )          {
+    return new Meeting({
+      guestSipAuthorization: plain.guest_sip_authorization,
+      hostname: plain.hostname,
+      name: plain.name,
+      ownerUuids: plain.owner_uuids,
+      port: plain.port,
+      uuid: plain.uuid,
+      number: plain.number,
+    });
+  }
+
+  static newFrom(meeting         ) {
+    return newFrom(meeting, Meeting);
+  }
+
+  constructor({ uuid, name, guestSipAuthorization, ownerUuids, port, hostname, number }         = {}) {
+    this.guestSipAuthorization = guestSipAuthorization;
+    this.hostname = hostname;
+    this.name = name;
+    this.ownerUuids = ownerUuids;
+    this.port = port;
+    this.uuid = uuid;
+    this.number = number;
+
+    // Useful to compare instead of instanceof with minified code
+    this.type = 'Meeting';
+  }
+
+  getGuestSipCredentials() {
+    // eslint-disable-next-line no-undef
+    const [username, secret] = atob(this.guestSipAuthorization).split(':');
+    return { username, secret };
+  }
+}
+
 /*       */
 
 var confdMethods = (client              , baseUrl        ) => ({
@@ -3124,6 +3183,18 @@ var confdMethods = (client              , baseUrl        ) => ({
       return null;
     }
   },
+
+  createMyMeeting: (name        )                   =>
+    client.post(`${baseUrl}/users/me/meetings`, { name }).then(Meeting.parse),
+
+  deleteMyMeeting: (meetingUuid        )                   =>
+    client.delete(`${baseUrl}/users/me/meetings/${meetingUuid}`, null),
+
+  getMeeting: (meetingUuid        )                   =>
+    client.get(`${baseUrl}/meetings/${meetingUuid}`, null).then(Meeting.parse),
+
+  guestGetMeeting: (meetingUuid        )                   =>
+    client.get(`${baseUrl}/guests/me/meetings/${meetingUuid}`, null).then(Meeting.parse),
 });
 
 //      
@@ -3528,7 +3599,11 @@ class CallLog {
       answered: plain.answered,
       callDirection: plain.call_direction,
       destination: {
-        extension: plain.requested_extension || plain.destination_extension,
+        // @TEMP: Temporarily assuming empty numbers are meetings
+        // which is admittedly a very dangerous assumption. Did i mention it was temporary?
+        extension: plain.requested_extension
+          || plain.destination_extension
+          || `meeting-${plain.requested_name || plain.destination_name || ''}`,
         name: plain.requested_name || plain.destination_name || '',
       },
       source: {
@@ -3544,27 +3619,11 @@ class CallLog {
   }
 
   static parseNew(plain                 , session         )          {
-    return new CallLog({
-      answer: plain.answer ? moment__default['default'](plain.answer).toDate() : null,
-      answered: plain.answered,
-      callDirection: plain.call_direction,
-      destination: {
-        extension: plain.requested_extension || plain.destination_extension,
-        name: plain.requested_name || plain.destination_name || '',
-      },
-      source: {
-        extension: plain.source_extension,
-        name: plain.source_name,
-      },
-      id: plain.id,
-      duration: (plain.duration || 0) * 1000, // duration is in seconds
-      start: moment__default['default'](plain.start).toDate(),
-      end: plain.end ? moment__default['default'](plain.end).toDate() : null,
-      recordings: Recording.parseMany(plain.recordings),
-      // @TODO: FIXME add verification declined vs missed call
-      newMissedCall: session
-        && session.hasExtension(plain.requested_extension || plain.destination_extension) && !plain.answered,
-    });
+    const callLog          = CallLog.parse(plain);
+    // @TODO: FIXME add verification declined vs missed call
+    callLog.newMissedCall = session
+      && session.hasExtension(plain.requested_extension || plain.destination_extension) && !plain.answered;
+    return callLog;
   }
 
   static newFrom(profile         ) {
@@ -3829,7 +3888,7 @@ const TRANSFER_FLOW_BLIND = 'blind';
 
 // const MINIMUM_WAZO_ENGINE_VERSION_FOR_CTI_HOLD = '20.11';
 
-const logger$a = IssueReporter$1 ? IssueReporter$1.loggerFor('cti-phone') : console;
+const logger$b = IssueReporter$1 ? IssueReporter$1.loggerFor('cti-phone') : console;
 
 class CTIPhone extends Emitter                  {
                    
@@ -3842,7 +3901,7 @@ class CTIPhone extends Emitter                  {
 
   constructor(session         , isMobile          = false, callbackAllLines          = false) {
     super();
-    logger$a.info('CTI Phone created');
+    logger$b.info('CTI Phone created');
     this.session = session;
     this.isMobile = isMobile;
     this.callbackAllLines = callbackAllLines;
@@ -3903,7 +3962,7 @@ class CTIPhone extends Emitter                  {
   }
 
   async makeCall(number        , line      )                        {
-    logger$a.info('make CTI call', { number });
+    logger$b.info('make CTI call', { number });
 
     if (!number) {
       return null;
@@ -3926,7 +3985,7 @@ class CTIPhone extends Emitter                  {
     if (!callSession) {
       return Promise.resolve(null);
     }
-    logger$a.info('accept CTI call', { callId: callSession.getId(), number: callSession.number });
+    logger$b.info('accept CTI call', { callId: callSession.getId(), number: callSession.number });
 
     if (!this.currentCall) {
       this.currentCall = callSession.call;
@@ -3939,7 +3998,7 @@ class CTIPhone extends Emitter                  {
     if (!callSession) {
       return;
     }
-    logger$a.info('end current CTI call', { callId: callSession.getId(), number: callSession.number });
+    logger$b.info('end current CTI call', { callId: callSession.getId(), number: callSession.number });
 
     this.currentCall = undefined;
     this.eventEmitter.emit('onCallEnded', callSession);
@@ -3949,7 +4008,7 @@ class CTIPhone extends Emitter                  {
     if (!callSession) {
       return Promise.resolve(false);
     }
-    logger$a.info('hangup CTI call', { callId: callSession.getId(), number: callSession.number });
+    logger$b.info('hangup CTI call', { callId: callSession.getId(), number: callSession.number });
 
     try {
       await CallApi.cancelCall(callSession);
@@ -3960,7 +4019,7 @@ class CTIPhone extends Emitter                  {
       this.eventEmitter.emit('onCallEnded', callSession);
       return true;
     } catch (e) {
-      logger$a.error('hangup CTI call, error', e);
+      logger$b.error('hangup CTI call, error', e);
 
       this.eventEmitter.emit('onCallFailed', callSession);
       return false;
@@ -3973,7 +4032,7 @@ class CTIPhone extends Emitter                  {
     if (!callSession) {
       return;
     }
-    logger$a.info('reject CTI call', { callId: callSession.getId(), number: callSession.number });
+    logger$b.info('reject CTI call', { callId: callSession.getId(), number: callSession.number });
 
     await CallApi.cancelCall(callSession);
     this.eventEmitter.emit('onCallEnded', callSession);
@@ -3983,7 +4042,7 @@ class CTIPhone extends Emitter                  {
     if (!callSession) {
       return;
     }
-    logger$a.info('transfer CTI call', { callId: callSession.getId(), number: callSession.number, to: number });
+    logger$b.info('transfer CTI call', { callId: callSession.getId(), number: callSession.number, to: number });
 
     await CallApi.transferCall(callSession.callId, number, TRANSFER_FLOW_BLIND);
   }
@@ -3994,19 +4053,19 @@ class CTIPhone extends Emitter                  {
     if (!callSession) {
       return;
     }
-    logger$a.info('indirect CTI transfer', { callId: callSession.getId(), number: callSession.number, to: number });
+    logger$b.info('indirect CTI transfer', { callId: callSession.getId(), number: callSession.number, to: number });
 
     return CallApi.transferCall(callSession.callId, number, TRANSFER_FLOW_ATTENDED);
   }
 
   async cancelCTIIndirectTransfer(transferId        )               {
-    logger$a.info('cancel CTI transfer', { transferId });
+    logger$b.info('cancel CTI transfer', { transferId });
 
     return CallApi.cancelCallTransfer(transferId);
   }
 
   async confirmCTIIndirectTransfer(transferId        )               {
-    logger$a.info('confirm CTI transfer', { transferId });
+    logger$b.info('confirm CTI transfer', { transferId });
 
     return CallApi.confirmCallTransfer(transferId);
   }
@@ -4015,19 +4074,19 @@ class CTIPhone extends Emitter                  {
     if (!callSession) {
       return;
     }
-    logger$a.info('send CTI key', { callId: callSession.getId(), number: callSession.number, digits });
+    logger$b.info('send CTI key', { callId: callSession.getId(), number: callSession.number, digits });
 
     return CallApi.sendDTMF(callSession.callId, digits);
   }
 
   onConnectionMade() {
-    logger$a.info('on CTI connection made');
+    logger$b.info('on CTI connection made');
 
     this.eventEmitter.emit('onCallAccepted');
   }
 
   async close() {
-    logger$a.info('CTI close');
+    logger$b.info('CTI close');
 
     return Promise.resolve();
   }
@@ -4036,7 +4095,7 @@ class CTIPhone extends Emitter                  {
     if (!callSession) {
       return;
     }
-    logger$a.info('CTI hold', { callId: callSession.getId(), number: callSession.number });
+    logger$b.info('CTI hold', { callId: callSession.getId(), number: callSession.number });
 
     return CallApi.hold(callSession.callId);
   }
@@ -4045,7 +4104,7 @@ class CTIPhone extends Emitter                  {
     if (!callSession) {
       return;
     }
-    logger$a.info('CTI resume', { callId: callSession.getId(), number: callSession.number });
+    logger$b.info('CTI resume', { callId: callSession.getId(), number: callSession.number });
 
     return CallApi.resume(callSession.callId);
   }
@@ -4054,7 +4113,7 @@ class CTIPhone extends Emitter                  {
     if (!callSession) {
       return;
     }
-    logger$a.info('CTI mute', { callId: callSession.getId(), number: callSession.number });
+    logger$b.info('CTI mute', { callId: callSession.getId(), number: callSession.number });
 
     return CallApi.mute(callSession.callId);
   }
@@ -4063,7 +4122,7 @@ class CTIPhone extends Emitter                  {
     if (!callSession) {
       return;
     }
-    logger$a.info('CTI unmute', { callId: callSession.getId(), number: callSession.number });
+    logger$b.info('CTI unmute', { callId: callSession.getId(), number: callSession.number });
 
     return CallApi.unmute(callSession.callId);
   }
@@ -4709,6 +4768,9 @@ var calldMethods = (client              , baseUrl        ) => ({
   getConferenceParticipantsAsUser: async (conferenceId        )                  =>
     client.get(`${baseUrl}/users/me/conferences/${conferenceId}/participants`),
 
+  getMeetingParticipantsAsUser: async (meetingUuid        )                  =>
+    client.get(`${baseUrl}/users/me/meetings/${meetingUuid}/participants`),
+
   listTrunks: () => client.get(`${baseUrl}/trunks`),
 
   mute: (callId        ) =>
@@ -5062,7 +5124,7 @@ const AGENTD_VERSION = '1.0';
 const WEBHOOKD_VERSION = '1.0';
 const AMID_VERSION = '1.0';
 
-const logger$9 = IssueReporter$1 ? IssueReporter$1.loggerFor('api') : console;
+const logger$a = IssueReporter$1 ? IssueReporter$1.loggerFor('api') : console;
 
 class ApiClient {
                        
@@ -5114,12 +5176,12 @@ class ApiClient {
   }
 
   async forceRefreshToken() {
-    logger$9.info('forcing refresh token, calling callback');
+    logger$a.info('forcing refresh token, calling callback');
     return this.refreshTokenCallback();
   }
 
   async refreshTokenCallback() {
-    logger$9.info('refresh token callback called', {
+    logger$a.info('refresh token callback called', {
       refreshToken: this.refreshToken,
       refreshBackend: this.refreshBackend,
       refreshExpiration: this.refreshExpiration,
@@ -5137,7 +5199,7 @@ class ApiClient {
       this.isMobile,
     );
 
-    logger$9.info('token refreshed', { token: session.token });
+    logger$a.info('token refreshed', { token: session.token });
 
     if (this.onRefreshToken) {
       this.onRefreshToken(session.token, session);
@@ -5610,7 +5672,9 @@ class WazoSessionDescriptionHandler extends sessionDescriptionHandler.SessionDes
       .then(() => {
         if (isConference && (options.constraints && !options.constraints.video) && !('hold' in options) && !audioOnly) {
           // Add a video an empty bundle to be able to replaceTrack when joining a conference without video
-          this.peerConnection.addTransceiver('video', { streams: [this._localMediaStream], direction: 'sendrecv' });
+          if (this.peerConnection.addTransceiver) {
+            this.peerConnection.addTransceiver('video', { streams: [this._localMediaStream], direction: 'sendrecv' });
+          }
         }
       })
       .then(() => this.createLocalOfferOrAnswer(options))
@@ -5637,12 +5701,6 @@ class WazoSessionDescriptionHandler extends sessionDescriptionHandler.SessionDes
         };
       })
       .then((sessionDescription) => this.applyModifiers(sessionDescription, modifiers))
-      .then((sessionDescription) => {
-        // Set new constraints to avoid the constraints check issue in `sdh.getLocalMediaStream` later.
-        this.localMediaStreamConstraints = options.constraints;
-
-        return sessionDescription;
-      })
       .then((sessionDescription) => ({ body: sessionDescription.sdp, contentType: 'application/sdp' }))
       .catch((error) => {
         wazoLogger.error('error when creating media', error);
@@ -5837,6 +5895,9 @@ class WazoSessionDescriptionHandler extends sessionDescriptionHandler.SessionDes
 
         // set the transceiver direction to the answer direction
         this._peerConnection.getTransceivers().forEach((transceiver) => {
+          if (transceiver.stopped) {
+            return;
+          }
           if (transceiver.direction /* guarding, but should always be true */) {
             const { receiver } = transceiver;
             if (isConference && audioOnly && receiver.track && receiver.track.kind === 'video') {
@@ -5881,10 +5942,15 @@ class WazoSessionDescriptionHandler extends sessionDescriptionHandler.SessionDes
       if (kind !== 'audio' && kind !== 'video') {
         throw new Error(`Unknown new track kind ${kind}.`);
       }
-      const sender = pc.getSenders().find((otherSender) => otherSender.track && otherSender.track.kind === kind);
+      const sender = pc.getSenders && pc.getSenders().find((otherSender) =>
+        otherSender.track && otherSender.track.kind === kind);
 
       // Do not reuse sender video tracks in SFU
       if (sender && (!sfu || newTrack.kind === 'audio')) {
+        if (sender.track) {
+          // eslint-disable-next-line no-param-reassign
+          newTrack.enabled = sender.track.enabled;
+        }
         trackUpdates.push(
           new Promise((resolve) => {
             this.logger.debug(`SessionDescriptionHandler.setLocalMediaStream - replacing sender ${kind} track`);
@@ -5918,7 +5984,11 @@ class WazoSessionDescriptionHandler extends sessionDescriptionHandler.SessionDes
             // Review: could make streamless tracks a configurable option?
             // https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addTrack#Usage_notes
             try {
-              pc.addTrack(newTrack, localStream);
+              if (pc.addTrack) {
+                pc.addTrack(newTrack, localStream);
+              } else {
+                pc.addStream(localStream);
+              }
             } catch (error) {
               this.logger.error(`SessionDescriptionHandler.setLocalMediaStream - failed to add sender ${kind} track`);
               throw error;
@@ -6054,7 +6124,7 @@ const DEFAULT_ICE_TIMEOUT = 3000;
 const SEND_STATS_DELAY = 5000;
 
 const states = ['STATUS_NULL', 'STATUS_NEW', 'STATUS_CONNECTING', 'STATUS_CONNECTED', 'STATUS_COMPLETED'];
-const logger$8 = IssueReporter$1 ? IssueReporter$1.loggerFor('webrtc-client') : console;
+const logger$9 = IssueReporter$1 ? IssueReporter$1.loggerFor('webrtc-client') : console;
 const statsLogger = IssueReporter$1 ? IssueReporter$1.loggerFor('webrtc-stats') : console;
 
 // events
@@ -6100,6 +6170,7 @@ const MAX_REGISTER_TRIES = 10;
                            
                          
                            
+                        
                         
   
 
@@ -6126,6 +6197,7 @@ class WebRTCClient extends Emitter {
                          
                                      
                                      
+                        
 
   // sugar
                         
@@ -6164,6 +6236,8 @@ class WebRTCClient extends Emitter {
     super();
     this.uaConfigOverrides = uaConfigOverrides;
     this.config = config;
+    this.skipRegister = config.skipRegister;
+
     this._buildConfig(config, session).then((newConfig              ) => {
       this.config = newConfig;
       this.userAgent = this.createUserAgent(uaConfigOverrides);
@@ -6219,13 +6293,13 @@ class WebRTCClient extends Emitter {
 
   createUserAgent(configOverrides         )            {
     const webRTCConfiguration = this._createWebRTCConfiguration(configOverrides);
-    logger$8.info('sdk webrtc, creating UA', { webRTCConfiguration });
+    logger$9.info('sdk webrtc, creating UA', { webRTCConfiguration });
 
     webRTCConfiguration.delegate = {
       onConnect: this.onConnect.bind(this),
       onDisconnect: this.onDisconnect.bind(this),
       onInvite: (invitation            ) => {
-        logger$8.info('sdk webrtc on invite', {
+        logger$9.info('sdk webrtc on invite', {
           method: 'delegate.onInvite',
           id: invitation.id,
           remoteURI: invitation.remoteURI,
@@ -6268,7 +6342,7 @@ class WebRTCClient extends Emitter {
   }
 
   onConnect() {
-    logger$8.info('sdk webrtc connected', { method: 'delegate.onConnect' });
+    logger$9.info('sdk webrtc connected', { method: 'delegate.onConnect' });
     this.eventEmitter.emit(CONNECTED);
     if (!this.isRegistered() && this.registerer && this.registerer.waiting) {
       this.registerer.waitingToggle(false);
@@ -6277,7 +6351,7 @@ class WebRTCClient extends Emitter {
   }
 
   async onDisconnect(error        ) {
-    logger$8.info('sdk webrtc disconnected', { method: 'delegate.onConnect', error });
+    logger$9.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);
@@ -6291,40 +6365,48 @@ class WebRTCClient extends Emitter {
   }
 
   register(tries         = 0)               {
-    logger$8.info('sdk webrtc registering...', {
+    const logInfo = {
       userAgent: !!this.userAgent,
       registered: this.isRegistered(),
       connectionPromise: !!this.connectionPromise,
       registerer: !!this.registerer,
       waiting: this.registerer && this.registerer.waiting,
       tries,
-    });
+      skipRegister: this.skipRegister,
+    };
+
+    if (this.skipRegister) {
+      logger$9.info('sdk webrtc skip register...', logInfo);
+      return Promise.resolve();
+    }
+
+    logger$9.info('sdk webrtc registering...', logInfo);
 
     if (!this.userAgent) {
-      logger$8.info('sdk webrtc recreating User Agent');
+      logger$9.info('sdk webrtc recreating User Agent');
       this.userAgent = this.createUserAgent(this.uaConfigOverrides);
     }
     if (!this.userAgent || this.isRegistered()) {
-      logger$8.info('sdk webrtc registering aborted, already registered or no UA can be created');
+      logger$9.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$8.info('sdk webrtc registering aborted due to a registration in progress.');
+      logger$9.info('sdk webrtc registering aborted due to a registration in progress.');
       return Promise.resolve();
     }
 
     const registerOptions = this._isWeb() ? {} : { extraContactHeaderParams: ['mobility=mobile'] };
 
     const onRegisterFailed = () => {
-      logger$8.info('sdk webrtc registering failed', { tries, registerer: !!this.registerer });
+      logger$9.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$8.info('sdk webrtc registering, retrying...', { tries });
+        logger$9.info('sdk webrtc registering, retrying...', { tries });
         setTimeout(() => this.register(tries + 1), 300);
       }
     };
@@ -6333,18 +6415,18 @@ 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$8.info('sdk webrtc recreating User Agent after connection');
+        logger$9.info('sdk webrtc recreating User Agent after connection');
         this.userAgent = this.createUserAgent(this.uaConfigOverrides);
       }
 
-      logger$8.info('sdk webrtc registering, transport connected', { registerOptions, ua: !!this.userAgent });
+      logger$9.info('sdk webrtc registering, transport connected', { registerOptions, ua: !!this.userAgent });
       this.registerer = new registerer.Registerer(this.userAgent, registerOptions);
       this.connectionPromise = null;
       this._monkeyPatchRegisterer(this.registerer);
 
       // Bind registerer events
       this.registerer.stateChange.addListener(newState => {
-        logger$8.info('sdk webrtc registering, state changed', { newState });
+        logger$9.info('sdk webrtc registering, state changed', { newState });
 
         if (newState === registererState.RegistererState.Registered && this.registerer
           && this.registerer.state === registererState.RegistererState.Registered) {
@@ -6357,7 +6439,7 @@ class WebRTCClient extends Emitter {
       const options = {
         requestDelegate: {
           onReject: response => {
-            logger$8.error('sdk webrtc registering, rejected', { response });
+            logger$9.error('sdk webrtc registering, rejected', { response });
 
             onRegisterFailed();
           },
@@ -6365,13 +6447,13 @@ class WebRTCClient extends Emitter {
       };
 
       return this.registerer.register(options).catch((e) => {
-        logger$8.error('sdk webrtc registering, error', e);
+        logger$9.error('sdk webrtc registering, error', e);
 
         this.eventEmitter.emit(REGISTRATION_FAILED);
         return e;
       });
     }).catch(error => {
-      logger$8.error('sdk webrtc registering, transport error', error);
+      logger$9.error('sdk webrtc registering, transport error', error);
       onRegisterFailed();
     });
   }
@@ -6400,21 +6482,21 @@ class WebRTCClient extends Emitter {
   }
 
   unregister() {
-    logger$8.info('sdk webrtc unregistering..', { userAgent: !!this.userAgent, registerer: !!this.registerer });
+    logger$9.info('sdk webrtc unregistering..', { userAgent: !!this.userAgent, registerer: !!this.registerer });
     if (!this.registerer) {
       return Promise.resolve();
     }
 
     try {
       return this.registerer.unregister().then(() => {
-        logger$8.info('sdk webrtc unregistered');
+        logger$9.info('sdk webrtc unregistered');
         this._cleanupRegister();
       }).catch(e => {
-        logger$8.error('sdk webrtc unregistering, promise error', e);
+        logger$9.error('sdk webrtc unregistering, promise error', e);
         this._cleanupRegister();
       });
     } catch (e) {
-      logger$8.error('sdk webrtc unregistering, error', e);
+      logger$9.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();
@@ -6422,7 +6504,7 @@ class WebRTCClient extends Emitter {
   }
 
   stop()               {
-    logger$8.info('sdk webrtc stop', { userAgent: !!this.userAgent });
+    logger$9.info('sdk webrtc stop', { userAgent: !!this.userAgent });
     if (!this.userAgent) {
       return Promise.resolve();
     }
@@ -6430,12 +6512,12 @@ class WebRTCClient extends Emitter {
     return this.userAgent.stop().then(() => {
       return this._cleanupRegister();
     }).catch(e => {
-      logger$8.warn('sdk webrtc stop, error', { message: e.message, stack: e.stack });
+      logger$9.warn('sdk webrtc stop, error', { message: e.message, stack: e.stack });
     });
   }
 
   call(number        , enableVideo          , audioOnly          = false, conference          = false)          {
-    logger$8.info('sdk webrtc creating call', { number, enableVideo, audioOnly });
+    logger$9.info('sdk webrtc creating call', { number, enableVideo, audioOnly });
     const inviterOptions         = {
       sessionDescriptionHandlerOptionsReInvite: {
         conference,
@@ -6464,7 +6546,7 @@ class WebRTCClient extends Emitter {
           this._onAccepted(session, response.session, true);
         },
         onReject: (response                  ) => {
-          logger$8.info('on call rejected', { id: session.id, fromTag: session.fromTag });
+          logger$9.info('on call rejected', { id: session.id, fromTag: session.fromTag });
           this._stopSendingStats(session);
 
           this.eventEmitter.emit(REJECTED, session, response);
@@ -6486,11 +6568,11 @@ class WebRTCClient extends Emitter {
   }
 
   answer(session            , enableVideo          ) {
-    logger$8.info('sdk webrtc answer call', { id: session.id, enableVideo });
+    logger$9.info('sdk webrtc answer call', { id: session.id, enableVideo });
 
     if (!session || !session.accept) {
       const error = 'No session to answer, or not an invitation';
-      logger$8.warn(error);
+      logger$9.warn(error);
       return Promise.reject(new Error(error));
     }
     const options = {
@@ -6500,15 +6582,15 @@ 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$8.error(message, { id: session.id });
+        logger$9.error(message, { id: session.id });
         this.onCallEnded(session);
 
         throw new CanceledCallError(message);
       }
-      logger$8.info('sdk webrtc answer, accepted.');
+      logger$9.info('sdk webrtc answer, accepted.');
       this._onAccepted(session);
     }).catch(e => {
-      logger$8.error(`answer call error for ${session ? session.id : 'n/a'}`, e);
+      logger$9.error(`answer call error for ${session ? session.id : 'n/a'}`, e);
 
       throw e;
     });
@@ -6516,7 +6598,7 @@ class WebRTCClient extends Emitter {
 
   hangup(session         ) {
     const { state, id } = session;
-    logger$8.info('sdk webrtc hangup call', { id, state });
+    logger$9.info('sdk webrtc hangup call', { id, state });
 
     try {
       this._stopSendingStats(session);
@@ -6545,24 +6627,24 @@ class WebRTCClient extends Emitter {
 
       return bye();
     } catch (error) {
-      logger$8.warn('sdk webrtc hangup, error', error);
+      logger$9.warn('sdk webrtc hangup, error', error);
     }
 
     return null;
   }
 
   reject(session         ) {
-    logger$8.info('sdk webrtc reject call', { id: session.id });
+    logger$9.info('sdk webrtc reject call', { id: session.id });
 
     try {
       return session.reject ? session.reject() : session.cancel();
     } catch (e) {
-      logger$8.warn('Error when rejecting call', e.message, e.stack);
+      logger$9.warn('Error when rejecting call', e.message, e.stack);
     }
   }
 
   async close() {
-    logger$8.info('sdk webrtc closing client', { userAgent: !!this.userAgent });
+    logger$9.info('sdk webrtc closing client', { userAgent: !!this.userAgent });
     this._cleanupMedia();
     this.connectionPromise = null;
 
@@ -6593,7 +6675,7 @@ class WebRTCClient extends Emitter {
       // eg: "INVITE not rejectable in state Completed"
     }
     this.userAgent = null;
-    logger$8.info('sdk webrtc client closed');
+    logger$9.info('sdk webrtc client closed');
   }
 
   getNumber(session         )          {
@@ -6606,13 +6688,13 @@ class WebRTCClient extends Emitter {
   }
 
   mute(session         ) {
-    logger$8.info('sdk webrtc mute', { id: session.id });
+    logger$9.info('sdk webrtc mute', { id: session.id });
 
     this._toggleAudio(session, true);
   }
 
   unmute(session         ) {
-    logger$8.info('sdk webrtc unmute', { id: session.id });
+    logger$9.info('sdk webrtc unmute', { id: session.id });
 
     this._toggleAudio(session, false);
   }
@@ -6629,12 +6711,18 @@ class WebRTCClient extends Emitter {
     }
 
     if (pc.getSenders) {
+      if (!pc.getSenders().length) {
+        return false;
+      }
       pc.getSenders().forEach(sender => {
         if (sender && sender.track && sender.track.kind === 'audio') {
           muted = muted && !sender.track.enabled;
         }
       });
     } else {
+      if (!pc.getLocalStreams().length) {
+        return false;
+      }
       pc.getLocalStreams().forEach(stream => {
         stream.getAudioTracks().forEach(track => {
           muted = muted && !track.enabled;
@@ -6646,20 +6734,20 @@ class WebRTCClient extends Emitter {
   }
 
   toggleCameraOn(session         ) {
-    logger$8.info('sdk webrtc toggle camera on', { id: session.id });
+    logger$9.info('sdk webrtc toggle camera on', { id: session.id });
 
     this._toggleVideo(session, false);
   }
 
   toggleCameraOff(session         ) {
-    logger$8.info('sdk webrtc toggle camera off', { id: session.id });
+    logger$9.info('sdk webrtc toggle camera off', { id: session.id });
 
     this._toggleVideo(session, true);
   }
 
   hold(session         , isConference          = false) {
     const sessionId = this.getSipSessionId(session);
-    logger$8.info('sdk webrtc hold', {
+    logger$9.info('sdk webrtc hold', {
       sessionId,
       keys: Object.keys(this.heldSessions),
       pendingReinvite: !!session.pendingReinvite,
@@ -6693,6 +6781,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -6702,7 +6792,7 @@ class WebRTCClient extends Emitter {
   unhold(session         , isConference          = false) {
     const sessionId = this.getSipSessionId(session);
 
-    logger$8.info('sdk webrtc unhold', {
+    logger$9.info('sdk webrtc unhold', {
       sessionId,
       keys: Object.keys(this.heldSessions),
       pendingReinvite: !!session.pendingReinvite,
@@ -6735,6 +6825,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -6793,7 +6885,7 @@ class WebRTCClient extends Emitter {
       return;
     }
 
-    logger$8.info('send WebRTC message', { sipId: sipSession.id, contentType });
+    logger$9.info('send WebRTC message', { sipId: sipSession.id, contentType });
 
     try {
       sipSession.message({
@@ -6918,7 +7010,8 @@ class WebRTCClient extends Emitter {
     }
     const sdh = session.sessionDescriptionHandler;
     const pc = sdh.peerConnection;
-    const localStream = this.getLocalStream(this.getSipSessionId(session));
+    const sessionId = this.getSipSessionId(session);
+    const localStream = this.getLocalStream(sessionId);
 
     // Release old video stream
     if (localStream) {
@@ -6944,11 +7037,8 @@ class WebRTCClient extends Emitter {
 
       // let's update the local stream
       this.eventEmitter.emit('onVideoInputChange', stream);
-      try {
-        await sdh.setLocalMediaStream(stream);
-      } catch (_) {
-        // Nothing to do
-      }
+      this.setLocalMediaStream(sessionId, stream);
+
       return stream;
     });
   }
@@ -7160,7 +7250,7 @@ class WebRTCClient extends Emitter {
   }
 
   startHeartbeat() {
-    logger$8.info('sdk webrtc start heartbeat', { userAgent: !!this.userAgent });
+    logger$9.info('sdk webrtc start heartbeat', { userAgent: !!this.userAgent });
     if (!this.userAgent) {
       this.heartbeat.stop();
       return;
@@ -7173,7 +7263,7 @@ class WebRTCClient extends Emitter {
   }
 
   stopHeartbeat() {
-    logger$8.info('sdk webrtc stop heartbeat');
+    logger$9.info('sdk webrtc stop heartbeat');
     this.heartbeat.stop();
   }
 
@@ -7193,7 +7283,7 @@ class WebRTCClient extends Emitter {
   }
 
   attemptReconnection()       {
-    logger$8.info('attempt reconnection', { userAgent: !!this.userAgent });
+    logger$9.info('attempt reconnection', { userAgent: !!this.userAgent });
     if (!this.userAgent) {
       return;
     }
@@ -7250,9 +7340,11 @@ class WebRTCClient extends Emitter {
       remoteStream.removeTrack(track);
     });
 
-    pc.getReceivers().forEach(receiver => {
-      remoteStream.addTrack(receiver.track);
-    });
+    if (pc.getReceivers) {
+      pc.getReceivers().forEach(receiver => {
+        remoteStream.addTrack(receiver.track);
+      });
+    }
   }
 
   getMediaConfiguration(enableVideo         , conference          = false, constraints          = null)         {
@@ -7287,6 +7379,25 @@ class WebRTCClient extends Emitter {
     return sessionId in this.conferences;
   }
 
+  createAudioElementFor(sessionId        ) {
+    const audio      = document.createElement('audio');
+    audio.setAttribute('id', `audio-${sessionId}`);
+
+    if (audio.setSinkId && this.audioOutputDeviceId) {
+      audio.setSinkId(this.audioOutputDeviceId);
+    }
+
+    if (document.body) {
+      document.body.appendChild(audio);
+    }
+    this.audioElements[sessionId] = audio;
+
+    audio.volume = this.audioOutputVolume;
+    audio.autoplay = true;
+
+    return audio;
+  }
+
   _onTransportError() {
     this.eventEmitter.emit(TRANSPORT_ERROR);
     this.attemptReconnection();
@@ -7296,7 +7407,7 @@ class WebRTCClient extends Emitter {
     const body = message && typeof message === 'object' ? message.data : message;
     if (body && body.indexOf('200 OK') !== -1) {
       if (this.hasHeartbeat()) {
-        logger$8.info('on heartbeat received from Asterisk', { hasHeartbeat: this.hasHeartbeat() });
+        logger$9.info('on heartbeat received from Asterisk', { hasHeartbeat: this.hasHeartbeat() });
 
         this.heartbeat.onHeartbeat();
         if (this.heartbeatCb) {
@@ -7307,7 +7418,7 @@ class WebRTCClient extends Emitter {
   }
 
   async _onHeartbeatTimeout() {
-    logger$8.info('sdk webrtc heartbeat timed out', { userAgent: !!this.userAgent, cb: !!this.heartbeatTimeoutCb });
+    logger$9.info('sdk webrtc heartbeat timed out', { userAgent: !!this.userAgent, cb: !!this.heartbeatTimeoutCb });
 
     if (this.heartbeatTimeoutCb) {
       this.heartbeatTimeoutCb();
@@ -7319,7 +7430,7 @@ class WebRTCClient extends Emitter {
         try {
           await this.userAgent.transport.disconnect();
         } catch (e) {
-          logger$8.error('Transport disconnection after heartbeat timeout, error', e);
+          logger$9.error('Transport disconnection after heartbeat timeout, error', e);
         }
       }
 
@@ -7352,31 +7463,31 @@ class WebRTCClient extends Emitter {
   }
 
   _connectIfNeeded()                {
-    logger$8.info('connect if needed, checking', { connected: this.isConnected() });
+    logger$9.info('connect if needed, checking', { connected: this.isConnected() });
     if (!this.userAgent) {
-      logger$8.info('need to recreate User Agent');
+      logger$9.info('need to recreate User Agent');
       this.userAgent = this.createUserAgent(this.uaConfigOverrides);
     }
 
     if (this.isConnected()) {
-      logger$8.info('webrtc sdk, already connected');
+      logger$9.info('webrtc sdk, already connected');
       return Promise.resolve();
     }
 
     if (this.isConnecting()) {
-      logger$8.info('webrtc sdk, already connecting...');
+      logger$9.info('webrtc sdk, already connecting...');
 
       this.connectionPromise = this.userAgent.transport.connectPromise;
       return this.connectionPromise;
     }
 
     if (this.connectionPromise) {
-      logger$8.info('webrtc sdk, connection promise connecting...');
+      logger$9.info('webrtc sdk, connection promise connecting...');
       // $FlowFixMe
       return this.connectionPromise;
     }
 
-    logger$8.info('WebRTC UA needs to connect');
+    logger$9.info('WebRTC UA needs to connect');
 
     // Force UA to reconnect
     if (this.userAgent && this.userAgent.state !== userAgentState.UserAgentState.Stopped) {
@@ -7501,7 +7612,7 @@ class WebRTCClient extends Emitter {
 
       sdh.peerConnectionDelegate = {
         onicecandidateerror: (error        ) => {
-          logger$8.error('on icecandidate error', {
+          logger$9.error('on icecandidate error', {
             address: error.address,
             port: error.port,
             errorCode: error.errorCode,
@@ -7516,7 +7627,7 @@ class WebRTCClient extends Emitter {
     let hadRemoteVideo = false;
 
     // Monkey patch `onInviteRequest` to be able to know if there was a remote video stream before `onInvite` is called
-    // Becase when `onInvite` is called we already got the video track
+    // Because when `onInvite` is called we already got the video track
     session.onInviteRequest = (request) => {
       hadRemoteVideo = this.hasARemoteVideoTrack(sipSessionId);
 
@@ -7530,7 +7641,7 @@ class WebRTCClient extends Emitter {
         updatedNumber = session.assertedIdentity.uri.normal.user;
         updatedCalleeName = session.assertedIdentity.displayName || updatedNumber;
       }
-      logger$8.info('re-invite received', { updatedCalleeName, updatedNumber });
+      logger$9.info('re-invite received', { updatedCalleeName, updatedNumber });
 
       // Useful to know if it's a video upgrade or an unhold from a remote peer with a video stream
 
@@ -7551,7 +7662,7 @@ class WebRTCClient extends Emitter {
   }
 
   _onAccepted(session         , sessionDialog                , withEvent          = true) {
-    logger$8.info('on call accepted', { id: session.id, remoteTag: session.remoteTag });
+    logger$9.info('on call accepted', { id: session.id, remoteTag: session.remoteTag });
 
     this.storeSipSession(session);
 
@@ -7591,20 +7702,10 @@ class WebRTCClient extends Emitter {
     const isConference = this.isConference(sessionId);
 
     if (this._hasAudio() && this._isWeb() && !(sessionId in this.audioElements)) {
-      const audio      = document.createElement('audio');
-      audio.setAttribute('id', `audio-${sessionId}`);
-
-      if (audio.setSinkId && this.audioOutputDeviceId) {
-        audio.setSinkId(this.audioOutputDeviceId);
-      }
-
-      if (document.body) {
-        document.body.appendChild(audio);
-      }
-      this.audioElements[sessionId] = audio;
+      this.createAudioElementFor(sessionId);
     }
 
-    const audioElement = this.audioElements[this.getSipSessionId(session)];
+    const audioElement = this.audioElements[sessionId];
     const stream = newStream || this.getRemoteStream(sessionId);
 
     if (!this._isWeb() || !stream) {
@@ -7751,10 +7852,10 @@ class WebRTCClient extends Emitter {
 
   _stopSendingStats(session         ) {
     const sessionId = this.getSipSessionId(session);
-    logger$8.trace('Check for stopping stats', { sessionId, ids: Object.keys(this.statsIntervals) });
+    logger$9.trace('Check for stopping stats', { sessionId, ids: Object.keys(this.statsIntervals) });
 
     if (sessionId in this.statsIntervals) {
-      logger$8.trace('Stop sending stats for call', { sessionId });
+      logger$9.trace('Stop sending stats for call', { sessionId });
       this.statsIntervals[sessionId]();
       delete this.statsIntervals[sessionId];
     }
@@ -7770,7 +7871,7 @@ class WebRTCClient extends Emitter {
       try {
         await this.userAgent.transport.disconnect();
       } catch (e) {
-        logger$8.error('WebRTC transport disconnect, error', e);
+        logger$9.error('WebRTC transport disconnect, error', e);
       }
     }
   }
@@ -7857,7 +7958,7 @@ const events = [
   ON_SIGNAL,
 ];
 
-const logger$7 = IssueReporter$1.loggerFor('webrtc-phone');
+const logger$8 = IssueReporter$1.loggerFor('webrtc-phone');
 
 class WebRTCPhone extends Emitter                  {
                            
@@ -7928,7 +8029,7 @@ 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$7.error('WebRTC register error', { message: error.message, stack: error.stack });
+      logger$8.error('WebRTC register error', { message: error.message, stack: error.stack });
     });
   }
 
@@ -7943,7 +8044,7 @@ class WebRTCPhone extends Emitter                  {
     if (!this.client) {
       return;
     }
-    logger$7.info('WebRTC phone stop');
+    logger$8.info('WebRTC phone stop');
     this.client.stop();
   }
 
@@ -7958,14 +8059,14 @@ class WebRTCPhone extends Emitter                  {
   async sendReinvite(callSession              , constraints          = null, conference          = false,
     audioOnly          = false) {
     const sipSession = this.findSipSession(callSession);
-    logger$7.info('WebRTC phone - send reinvite', {
+    logger$8.info('WebRTC phone - send reinvite', {
       sessionId: sipSession ? sipSession.id : null,
       constraints,
       audioOnly,
     });
 
     if (!sipSession) {
-      return null;
+      return false;
     }
 
     // $FlowFixMe
@@ -7975,19 +8076,17 @@ class WebRTCPhone extends Emitter                  {
     // Downgrade
     if (constraints && !isUpgrade) {
       // No reinvite needed
-      return this._downgradeToAudio(callSession);
+      this._downgradeToAudio(callSession);
+      return false;
     }
 
     if (isUpgrade) {
       const shouldReinvite = await this._upgradeToVideo(callSession, constraints, conference);
       if (!shouldReinvite) {
-        return;
+        return false;
       }
     }
 
-    // Force reinvite in SDH
-    sipSession.sessionDescriptionHandler.localMediaStreamConstraints = null;
-
     return this.client.reinvite(sipSession, constraints, conference, audioOnly);
   }
 
@@ -7996,7 +8095,7 @@ class WebRTCPhone extends Emitter                  {
   }
 
   startHeartbeat() {
-    logger$7.info('WebRTC phone - start heartbeat', { client: !!this.client, hasHeartbeat: this.client.hasHeartbeat() });
+    logger$8.info('WebRTC phone - start heartbeat', { client: !!this.client, hasHeartbeat: this.client.hasHeartbeat() });
     if (!this.client || this.client.hasHeartbeat()) {
       return;
     }
@@ -8005,7 +8104,7 @@ class WebRTCPhone extends Emitter                  {
   }
 
   stopHeartbeat() {
-    logger$7.info('WebRTC phone - stopHeartbeat', { client: !!this.client });
+    logger$8.info('WebRTC phone - stopHeartbeat', { client: !!this.client });
     if (!this.client) {
       return;
     }
@@ -8022,7 +8121,7 @@ class WebRTCPhone extends Emitter                  {
   }
 
   reconnect() {
-    logger$7.info('WebRTC phone - reconnect', { client: !!this.client });
+    logger$8.info('WebRTC phone - reconnect', { client: !!this.client });
     if (!this.client) {
       return;
     }
@@ -8060,7 +8159,7 @@ class WebRTCPhone extends Emitter                  {
   _downgradeToAudio(callSession              ) {
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
-      return;
+      return false;
     }
 
     // Release local video stream when downgrading to audio
@@ -8173,7 +8272,7 @@ class WebRTCPhone extends Emitter                  {
       // Monkey patch to know when canceled with the CANCEL message
       const onCancel = sipSession._onCancel.bind(sipSession);
       sipSession._onCancel = (message                        ) => {
-        logger$7.trace('on sip session canceled', { callId: message.callId });
+        logger$8.trace('on sip session canceled', { callId: message.callId });
 
         onCancel(message);
         const elsewhere = message.data.indexOf('cause=26') !== -1 && message.data.indexOf('completed elsewhere') !== -1;
@@ -8198,13 +8297,13 @@ class WebRTCPhone extends Emitter                  {
             this.audioOutputVolume,
           );
         case sessionState.SessionState.Terminating:
-          logger$7.info('WebRTC phone - call terminating', { sipId: sipSession.id });
+          logger$8.info('WebRTC phone - call terminating', { sipId: sipSession.id });
 
           this.eventEmitter.emit(ON_CALL_ENDING, this._createCallSession(sipSession));
 
           break;
         case sessionState.SessionState.Terminated:
-          logger$7.info('WebRTC phone - call terminated', { sipId: sipSession.id });
+          logger$8.info('WebRTC phone - call terminated', { sipId: sipSession.id });
 
           this._onCallTerminated(sipSession);
 
@@ -8240,7 +8339,7 @@ class WebRTCPhone extends Emitter                  {
   }
 
   async startScreenSharing(constraints        , callSession              ) {
-    logger$7.info('WebRTC - start screen sharing', { constraints, id: callSession ? callSession.getId() : null });
+    logger$8.info('WebRTC - start screen sharing', { constraints, id: callSession ? callSession.getId() : null });
 
     const screenShareStream = await this._getStreamFromConstraints(constraints);
     if (!screenShareStream) {
@@ -8262,6 +8361,10 @@ class WebRTCPhone extends Emitter                  {
       localStream.removeTrack(videoTrack);
     }
 
+    if (localStream) {
+      localStream.addTrack(screenTrack);
+    }
+
     if (sender) {
       sender.replaceTrack(screenTrack);
     }
@@ -8276,7 +8379,8 @@ class WebRTCPhone extends Emitter                  {
       return;
     }
 
-    logger$7.info('WebRTC phone - stop screen sharing');
+    logger$8.info('WebRTC phone - stop screen sharing');
+    let reinvited = false;
 
     try {
       if (this.currentScreenShare.stream) {
@@ -8291,7 +8395,8 @@ class WebRTCPhone extends Emitter                  {
         // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
         // We have to downgrade to audio.
         const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
-        await this.sendReinvite(targetCallSession, { audio: false, video: screenshareStopped }, conference);
+        const constraints = { audio: false, video: screenshareStopped };
+        reinvited = await this.sendReinvite(targetCallSession, constraints, conference);
       }
     } catch (e) {
       console.warn(e);
@@ -8305,6 +8410,8 @@ class WebRTCPhone extends Emitter                  {
     );
 
     this.currentScreenShare = null;
+
+    return reinvited;
   }
 
   async _getStreamFromConstraints(constraints        , conference          = false)                        {
@@ -8342,7 +8449,7 @@ class WebRTCPhone extends Emitter                  {
   }
 
   _onCallAccepted(sipSession         , cameraEnabled         )              {
-    logger$7.info('WebRTC phone -  on call accepted', { sipId: sipSession.id, cameraEnabled });
+    logger$8.info('WebRTC phone -  on call accepted', { sipId: sipSession.id, cameraEnabled });
 
     const callSession = this._createAcceptedCallSession(sipSession, cameraEnabled);
     this.currentSipSession = sipSession;
@@ -8360,11 +8467,15 @@ class WebRTCPhone extends Emitter                  {
   }
 
   changeAudioDevice(id        ) {
-    logger$7.info('WebRTC phone - change audio device', { deviceId: id });
+    logger$8.info('WebRTC phone - change audio device', { deviceId: id });
     this.audioOutputDeviceId = id;
     this.client.changeAudioOutputDevice(id);
   }
 
+  createAudioElementFor(sessionId        ) {
+    return this.client.createAudioElementFor(sessionId);
+  }
+
   changeRingDevice(id        ) {
     this.audioRingDeviceId = id;
   }
@@ -8381,25 +8492,25 @@ class WebRTCPhone extends Emitter                  {
   }
 
   changeAudioInputDevice(id        ) {
-    logger$7.info('WebRTC phone - changeAudio input device', { deviceId: id });
+    logger$8.info('WebRTC phone - changeAudio input device', { deviceId: id });
 
     return this.client.changeAudioInputDevice(id, this.currentSipSession);
   }
 
   changeVideoInputDevice(id         ) {
-    logger$7.info('WebRTC phone - change video input device', { deviceId: id });
+    logger$8.info('WebRTC phone - change video input device', { deviceId: id });
 
     return this.client.changeVideoInputDevice(id, this.currentSipSession);
   }
 
   changeSessionVideoInputDevice(id         ) {
-    logger$7.info('WebRTC phone - change session video input device', { deviceId: id });
+    logger$8.info('WebRTC phone - change session video input device', { deviceId: id });
 
     return this.client.changeSessionVideoInputDevice(id, this.currentSipSession);
   }
 
   _onCallTerminated(sipSession         ) {
-    logger$7.info('WebRTC phone - on call terminated', { sipId: sipSession.id });
+    logger$8.info('WebRTC phone - on call terminated', { sipId: sipSession.id });
 
     const callSession = this._createCallSession(sipSession);
     const isCurrentSession = this.isCurrentCallSipSession(callSession);
@@ -8433,7 +8544,7 @@ class WebRTCPhone extends Emitter                  {
     const hasIncomingCallSession = this.hasIncomingCallSession();
 
     // Re-trigger incoming call event for remaining incoming calls
-    logger$7.info('WebRTC phone - check to re-trigger incoming call', {
+    logger$8.info('WebRTC phone - check to re-trigger incoming call', {
       sipId: sipSession.id,
       shouldRetrigger,
       hasIncomingCallSession,
@@ -8522,37 +8633,37 @@ class WebRTCPhone extends Emitter                  {
 
   // Deprecated
   getLocalStreamForCall(callSession             )               {
-    logger$7.warn('WebRTCPhone.getLocalStreamForCall is deprecated, use WebRTCPhone.getLocalStream instead');
+    logger$8.warn('WebRTCPhone.getLocalStreamForCall is deprecated, use WebRTCPhone.getLocalStream instead');
 
     return this.getLocalStream(callSession);
   }
 
   // Deprecated
   getRemoteStreamForCall(callSession             )               {
-    logger$7.warn('WebRTCPhone.getRemoteStreamForCall is deprecated, use WebRTCPhone.getRemoteStream instead');
+    logger$8.warn('WebRTCPhone.getRemoteStreamForCall is deprecated, use WebRTCPhone.getRemoteStream instead');
 
     return this.getRemoteVideoStream(callSession);
   }
 
   getRemoteStreamsForCall(callSession             )               {
-    logger$7.warn('WebRTCPhone.getRemoteStreamsForCall is deprecated, use WebRTCPhone.getRemoteStreams instead');
+    logger$8.warn('WebRTCPhone.getRemoteStreamsForCall is deprecated, use WebRTCPhone.getRemoteStreams instead');
 
     return this.getRemoteStream(callSession);
   }
 
   accept(callSession             , cameraEnabled          )                         {
-    logger$7.info('WebRTC phone - accept call', { id: callSession ? callSession.getId() : 'n/a', cameraEnabled });
+    logger$8.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$7.warn('no CallSession to accept.');
+      logger$8.warn('no CallSession to accept.');
       return Promise.resolve(null);
     }
 
     if (callSession.getId() in this.acceptedSessions) {
-      logger$7.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
+      logger$8.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
       return Promise.resolve(callSession.sipCallId);
     }
 
@@ -8564,24 +8675,24 @@ class WebRTCPhone extends Emitter                  {
     const sipSession = this.client.getSipSession(callSession.getId());
     if (sipSession) {
       if (sipSession.state === sessionState.SessionState.Terminated || sipSession.state === sessionState.SessionState.Terminating) {
-        logger$7.warn('Trying to answer a terminated sipSession.');
+        logger$8.warn('Trying to answer a terminated sipSession.');
 
         return Promise.resolve(null);
       }
 
-      logger$7.info('accept call, session found', { sipId: sipSession.id });
+      logger$8.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$7.error(e);
+        logger$8.error(e);
         this.endCurrentCall(callSession);
         throw e;
       });
     }
 
-    logger$7.warn('no CallSession found to accept.', { id: callSession ? callSession.getId() : 'n/a' });
+    logger$8.warn('no CallSession found to accept.', { id: callSession ? callSession.getId() : 'n/a' });
 
     return Promise.resolve(null);
   }
@@ -8591,7 +8702,7 @@ class WebRTCPhone extends Emitter                  {
   }
 
   async reject(callSession             )                {
-    logger$7.info('reject WebRTC called', { id: callSession.getId() });
+    logger$8.info('reject WebRTC called', { id: callSession.getId() });
 
     this.eventEmitter.emit(ON_TERMINATE_SOUND, callSession, 'call rejected locally');
     if (!callSession || callSession.getId() in this.rejectedSessions) {
@@ -8603,14 +8714,14 @@ class WebRTCPhone extends Emitter                  {
 
     const sipSession = this.findSipSession(callSession);
     if (sipSession) {
-      logger$7.info('WebRTC rejecting', { sipId: sipSession.id });
+      logger$8.info('WebRTC rejecting', { sipId: sipSession.id });
 
       this.client.hangup(sipSession);
     }
   }
 
   async ignore(callSession             )                {
-    logger$7.info('WebRTC ignore', { id: callSession.getId() });
+    logger$8.info('WebRTC ignore', { id: callSession.getId() });
 
     // kill the ring
     this.eventEmitter.emit(ON_TERMINATE_SOUND, callSession, 'ignoring call');
@@ -8619,7 +8730,7 @@ class WebRTCPhone extends Emitter                  {
   }
 
   hold(callSession             , withEvent          = true, isConference          = false)       {
-    logger$7.info('WebRTC hold', { id: callSession.getId() });
+    logger$8.info('WebRTC hold', { id: callSession.getId() });
 
     const sipSession = this.findSipSession(callSession);
 
@@ -8630,7 +8741,7 @@ class WebRTCPhone extends Emitter                  {
 
   // @Deprecated
   unhold(callSession             , withEvent          = true, isConference          = false)       {
-    logger$7.info('WebRTC unhold', { id: callSession ? callSession.getId() : null });
+    logger$8.info('WebRTC unhold', { id: callSession ? callSession.getId() : null });
 
     console.warn('Please note that `phone.unhold()` is being deprecated; `phone.resume()` is the preferred method');
 
@@ -8645,7 +8756,7 @@ class WebRTCPhone extends Emitter                  {
     const sipSession = this.findSipSession(callSession);
 
     if (sipSession) {
-      logger$7.info('WebRTC atxfer', { sipId: sipSession.id });
+      logger$8.info('WebRTC atxfer', { sipId: sipSession.id });
 
       return this.client.atxfer(sipSession);
     }
@@ -8660,7 +8771,7 @@ class WebRTCPhone extends Emitter                  {
     if (!sipSession) {
       return new Promise((resolve, reject) => reject(new Error('No session to hold')));
     }
-    logger$7.info('WebRTC hold sip session', { sipId: sipSession.id });
+    logger$8.info('WebRTC hold sip session', { sipId: sipSession.id });
 
     const promise = this.client.hold(sipSession, isConference);
     if (withEvent) {
@@ -8679,7 +8790,7 @@ class WebRTCPhone extends Emitter                  {
     if (!sipSession) {
       return new Promise((resolve, reject) => reject(new Error('No session to unhold')));
     }
-    logger$7.info('WebRTC unhold', { sipId: sipSession.id });
+    logger$8.info('WebRTC unhold', { sipId: sipSession.id });
 
     const promise = this.client.unhold(sipSession, isConference);
     if (withEvent) {
@@ -8690,18 +8801,18 @@ class WebRTCPhone extends Emitter                  {
   }
 
   resume(callSession              , isConference          = false)               {
-    logger$7.info('WebRTC resume called', { id: callSession ? callSession.getId() : null });
+    logger$8.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$7.info('WebRTC resuming', { sipId: sipSession.id });
+    logger$8.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$7.info('WebRTC hold call after resume', { id: this.currentSipSession.id });
+      logger$8.info('WebRTC hold call after resume', { id: this.currentSipSession.id });
       this.holdSipSession(this.currentSipSession, callSession);
     }
 
@@ -8716,14 +8827,14 @@ class WebRTCPhone extends Emitter                  {
   }
 
   mute(callSession              , withEvent          = true)       {
-    logger$7.info('WebRTC mute called', { id: callSession ? callSession.getId() : null });
+    logger$8.info('WebRTC mute called', { id: callSession ? callSession.getId() : null });
 
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return;
     }
 
-    logger$7.info('WebRTC muting', { sipId: sipSession.id });
+    logger$8.info('WebRTC muting', { sipId: sipSession.id });
     this.client.mute(sipSession);
     const newCallSession = this._createCallSession(sipSession, callSession, { muted: true });
 
@@ -8737,14 +8848,14 @@ class WebRTCPhone extends Emitter                  {
   }
 
   unmute(callSession              , withEvent          = true)       {
-    logger$7.info('WebRTC unmute called', { id: callSession ? callSession.getId() : null });
+    logger$8.info('WebRTC unmute called', { id: callSession ? callSession.getId() : null });
 
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
       return;
     }
 
-    logger$7.info('WebRTC unmuting', { sipId: sipSession.id });
+    logger$8.info('WebRTC unmuting', { sipId: sipSession.id });
     this.client.unmute(sipSession);
     const newCallSession = this._createCallSession(sipSession, callSession, { muted: false });
 
@@ -8771,7 +8882,7 @@ class WebRTCPhone extends Emitter                  {
     if (!sipSession) {
       return;
     }
-    logger$7.info('WebRTC turn camera on', { sipId: sipSession.id });
+    logger$8.info('WebRTC turn camera on', { sipId: sipSession.id });
 
     this.client.toggleCameraOn(sipSession);
 
@@ -8783,7 +8894,7 @@ class WebRTCPhone extends Emitter                  {
     if (!sipSession) {
       return;
     }
-    logger$7.info('WebRTC turn camera off', { sipId: sipSession.id });
+    logger$8.info('WebRTC turn camera off', { sipId: sipSession.id });
 
     this.client.toggleCameraOff(sipSession);
     this.eventEmitter.emit(ON_CAMERA_DISABLED, this._createCameraDisabledCallSession(sipSession, callSession));
@@ -8795,7 +8906,7 @@ class WebRTCPhone extends Emitter                  {
       return;
     }
 
-    logger$7.info('WebRTC send key', { sipId: sipSession.id, tone });
+    logger$8.info('WebRTC send key', { sipId: sipSession.id, tone });
     this.client.sendDTMF(sipSession, tone);
   }
 
@@ -8803,7 +8914,7 @@ class WebRTCPhone extends Emitter                  {
   // @TODO: line is not used here
   async makeCall(number        , line     , cameraEnabled          ,
     audioOnly          = false, conference          = false)                        {
-    logger$7.info('make WebRTC call', { number, lineId: line ? line.id : null, cameraEnabled });
+    logger$8.info('make WebRTC call', { number, lineId: line ? line.id : null, cameraEnabled });
     if (!number) {
       return Promise.resolve(null);
     }
@@ -8813,7 +8924,7 @@ class WebRTCPhone extends Emitter                  {
     }
     if (this.currentSipSession) {
       this.holdSipSession(this.currentSipSession, this.currentCallSession, true).catch(e => {
-        logger$7.warn('Unable to hold current session when making another call', e);
+        logger$8.warn('Unable to hold current session when making another call', e);
       });
     }
 
@@ -8823,7 +8934,7 @@ class WebRTCPhone extends Emitter                  {
       this._bindEvents(sipSession);
     } catch (error) {
       console.warn(error);
-      logger$7.warn('make WebRTC call, error', { message: error.message, stack: error.stack });
+      logger$8.warn('make WebRTC call, error', { message: error.message, stack: error.stack });
       return Promise.resolve(null);
     }
     const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
@@ -8851,7 +8962,7 @@ class WebRTCPhone extends Emitter                  {
       return;
     }
 
-    logger$7.info('WebRTC transfer', { sipId: sipSession.id, target });
+    logger$8.info('WebRTC transfer', { sipId: sipSession.id, target });
 
     this.client.transfer(sipSession, target);
   }
@@ -8860,7 +8971,7 @@ class WebRTCPhone extends Emitter                  {
     const sipSession = this.client.getSipSession(source.sipCallId);
     const sipSessionTarget = this.client.getSipSession(destination.sipCallId);
 
-    logger$7.info('WebRTC indirect transfer', {
+    logger$8.info('WebRTC indirect transfer', {
       sipId: sipSession ? sipSession.id : null,
       target: sipSessionTarget ? sipSessionTarget.id : null,
     });
@@ -8888,7 +8999,7 @@ class WebRTCPhone extends Emitter                  {
       return false;
     }
 
-    logger$7.info('WebRTC hangup', { sipId: sipSession.id });
+    logger$8.info('WebRTC hangup', { sipId: sipSession.id });
 
     const sipSessionId = this.getSipSessionId(sipSession);
 
@@ -8933,7 +9044,7 @@ class WebRTCPhone extends Emitter                  {
   }
 
   async close()                {
-    logger$7.info('WebRTC close');
+    logger$8.info('WebRTC close');
 
     try {
       await Promise.race([
@@ -8941,7 +9052,7 @@ class WebRTCPhone extends Emitter                  {
         new Promise((resolve, reject) => setTimeout(() => reject(new Error('Unregister, timed out')), 3000)),
       ]);
     } catch (e) {
-      logger$7.error('WebRTC close, unregister error', e);
+      logger$8.error('WebRTC close, unregister error', e);
     }
 
     this.client.close();
@@ -8959,12 +9070,12 @@ class WebRTCPhone extends Emitter                  {
   }
 
   enableRinging()                       {
-    logger$7.info('WebRTC enable ringing');
+    logger$8.info('WebRTC enable ringing');
     this.ringingEnabled = true;
   }
 
   disableRinging()                       {
-    logger$7.info('WebRTC disable ringing');
+    logger$8.info('WebRTC disable ringing');
     this.ringingEnabled = false;
   }
 
@@ -9004,7 +9115,7 @@ class WebRTCPhone extends Emitter                  {
 
   // Deprecated
   getLocalMediaStream(callSession             ) {
-    logger$7.warn('WebRTCPhone.getLocalMediaStream is deprecated, use WebRTCPhone.getLocalStream instead');
+    logger$8.warn('WebRTCPhone.getLocalMediaStream is deprecated, use WebRTCPhone.getLocalStream instead');
     return this.getLocalStream(callSession);
   }
 
@@ -9012,6 +9123,10 @@ class WebRTCPhone extends Emitter                  {
     return callSession ? this.client.hasLocalVideo(callSession.sipCallId) : null;
   }
 
+  hasALocalVideoTrack(callSession             ) {
+    return callSession ? this.client.hasALocalVideoTrack(callSession.sipCallId) : null;
+  }
+
   getRemoteStream(callSession             )               {
     return callSession ? this.client.getRemoteStream(callSession.sipCallId) : null;
   }
@@ -9034,7 +9149,7 @@ 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$7.info('WebRTC invite received', { sipId: sipSession.id, withVideo, autoAnswer });
+      logger$8.info('WebRTC invite received', { sipId: sipSession.id, withVideo, autoAnswer });
 
       const callSession = this._createIncomingCallSession(sipSession, withVideo, null, autoAnswer);
       this.incomingSessions.push(callSession.getId());
@@ -9057,7 +9172,7 @@ class WebRTCPhone extends Emitter                  {
     });
 
     this.client.on(this.client.ON_REINVITE, (...args) => {
-      logger$7.info('WebRTC on reinvite', {
+      logger$8.info('WebRTC on reinvite', {
         sessionId: args[0].id,
         inviteId: args[1].id,
         updatedCalleeName: args[2],
@@ -9077,7 +9192,7 @@ class WebRTCPhone extends Emitter                  {
     });
 
     this.client.on(this.client.ACCEPTED, (sipSession         ) => {
-      logger$7.info('WebRTC call accepted', { sipId: sipSession.id });
+      logger$8.info('WebRTC call accepted', { sipId: sipSession.id });
 
       this._onCallAccepted(sipSession, this.client.hasVideo(this.getSipSessionId(sipSession)));
 
@@ -9088,17 +9203,17 @@ class WebRTCPhone extends Emitter                  {
     this.client.on('ended', () => {});
 
     this.client.on(this.client.ON_ERROR, e => {
-      logger$7.error('WebRTC error', e);
+      logger$8.error('WebRTC error', e);
       this.eventEmitter.emit(ON_CALL_ERROR, e);
     });
 
     this.client.on(this.client.UNREGISTERED, () => {
-      logger$7.info('WebRTC unregistered');
+      logger$8.info('WebRTC unregistered');
       this.eventEmitter.emit(ON_UNREGISTERED);
     });
 
     this.client.on(this.client.REGISTERED, () => {
-      logger$7.info('WebRTC registered', {
+      logger$8.info('WebRTC registered', {
         shouldSendReinvite: this.shouldSendReinvite,
         state: this.currentSipSession ? this.currentSipSession.state : null,
         currentSipSession: !!this.currentSipSession,
@@ -9114,19 +9229,19 @@ class WebRTCPhone extends Emitter                  {
         try {
           this.sendReinvite(this.currentCallSession);
         } catch (e) {
-          logger$7.error('WebRTC reinvite after register, error', { message: e.message, stack: e.stack });
+          logger$8.error('WebRTC reinvite after register, error', { message: e.message, stack: e.stack });
         }
       }
     });
 
     this.client.on(this.client.CONNECTED, () => {
-      logger$7.info('WebRTC client connected');
+      logger$8.info('WebRTC client connected');
 
       this.stopHeartbeat();
     });
 
     this.client.on(this.client.DISCONNECTED, () => {
-      logger$7.info('WebRTC client disconnected');
+      logger$8.info('WebRTC client disconnected');
 
       this.eventEmitter.emit(ON_UNREGISTERED);
 
@@ -9361,7 +9476,7 @@ var PHONE_EVENTS = /*#__PURE__*/Object.freeze({
 class InvalidSubscription extends Error {}
 class InvalidAuthorization extends Error {}
 
-const logger$6 = IssueReporter$1.loggerFor('simple-auth');
+const logger$7 = IssueReporter$1.loggerFor('simple-auth');
 
 class Auth {
                    
@@ -9396,7 +9511,7 @@ class Auth {
     setApiClientId(this.clientId);
     setRefreshExpiration(this.expiration);
     setOnRefreshToken((token        , session         ) => {
-      logger$6.info('on refresh token done', { token });
+      logger$7.info('on refresh token done', { token });
       setApiToken(token);
       Wazo.Websocket.updateToken(token);
 
@@ -9672,6 +9787,8 @@ const CONFERENCE_USER_PARTICIPANT_JOINED = 'conference_user_participant_joined';
 const CONFERENCE_USER_PARTICIPANT_LEFT = 'conference_user_participant_left';
 const CONFERENCE_USER_PARTICIPANT_TALK_STARTED = 'conference_user_participant_talk_started';
 const CONFERENCE_USER_PARTICIPANT_TALK_STOPPED = 'conference_user_participant_talk_stopped';
+const MEETING_USER_PARTICIPANT_JOINED = 'meeting_user_participant_joined';
+const MEETING_USER_PARTICIPANT_LEFT = 'meeting_user_participant_left';
 const SWITCHBOARD_QUEUED_CALLS_UPDATED = 'switchboard_queued_calls_updated';
 const SWITCHBOARD_QUEUED_CALL_ANSWERED = 'switchboard_queued_call_answered';
 const SWITCHBOARD_HELD_CALLS_UPDATED = 'switchboard_held_calls_updated';
@@ -9713,7 +9830,7 @@ const BLACKLIST_EVENTS = [
 ];
 
 const HEARTBEAT_ENGINE_VERSION = '20.09';
-const logger$5 = IssueReporter$1.loggerFor('wazo-ws');
+const logger$6 = IssueReporter$1.loggerFor('wazo-ws');
 const messageLogger = IssueReporter$1.loggerFor('wazo-ws-message');
 
 class WebSocketClient extends Emitter {
@@ -9762,19 +9879,19 @@ class WebSocketClient extends Emitter {
   }
 
   connect() {
-    logger$5.info('connect method started', { host: this.host, token: this.token });
+    logger$6.info('connect method started', { host: this.host, token: this.token });
     this.socket = new ReconnectingWebSocket__default['default'](this._getUrl.bind(this), [], this.options);
     if (this.options.binaryType) {
       this.socket.binaryType = this.options.binaryType;
     }
 
     this.socket.onopen = () => {
-      logger$5.info('on Wazo WS open', { method: 'connect', host: this.host });
+      logger$6.info('on Wazo WS open', { method: 'connect', host: this.host });
       this.eventEmitter.emit(SOCKET_EVENTS$1.ON_OPEN);
     };
 
     this.socket.onerror = event => {
-      logger$5.error('on Wazo WS error', event.target);
+      logger$6.error('on Wazo WS error', event.target);
       this.eventEmitter.emit(SOCKET_EVENTS$1.ON_ERROR, event);
     };
 
@@ -9804,7 +9921,7 @@ class WebSocketClient extends Emitter {
 
     this.socket.onclose = event => {
       // Can't be converted to JSON (circular structure)
-      logger$5.info('on Wazo WS close', {
+      logger$6.info('on Wazo WS close', {
         reason: event.reason,
         code: event.code,
         readyState: event.target.readyState,
@@ -9822,12 +9939,12 @@ class WebSocketClient extends Emitter {
     };
 
     this.socket.onerror = event => {
-      logger$5.info('Wazo WS error', { message: event.message, code: event.code, readyState: event.target.readyState });
+      logger$6.info('Wazo WS error', { message: event.message, code: event.code, readyState: event.target.readyState });
     };
   }
 
   close(force          = false)       {
-    logger$5.info('Wazo WS close', { socket: !!this.socket, host: this.host, token: this.token });
+    logger$6.info('Wazo WS close', { socket: !!this.socket, host: this.host, token: this.token });
 
     if (!this.socket) {
       return;
@@ -9845,7 +9962,7 @@ class WebSocketClient extends Emitter {
 
   updateToken(token        ) {
     this.token = token;
-    logger$5.info('Wazo WS updating token', { url: this._getUrl(), token, socket: !!this.socket });
+    logger$6.info('Wazo WS updating token', { url: this._getUrl(), token, socket: !!this.socket });
 
     if (this.socket) {
       // If still connected, send the token to the WS
@@ -9863,7 +9980,7 @@ class WebSocketClient extends Emitter {
   }
 
   startHeartbeat() {
-    logger$5.info('Wazo WS start heartbeat', { host: this.host, token: this.token });
+    logger$6.info('Wazo WS start heartbeat', { host: this.host, token: this.token });
 
     if (!this.socket) {
       this.heartbeat.stop();
@@ -9877,7 +9994,7 @@ class WebSocketClient extends Emitter {
   }
 
   stopHeartbeat() {
-    logger$5.info('Wazo WS stop heartbeat', { host: this.host, token: this.token });
+    logger$6.info('Wazo WS stop heartbeat', { host: this.host, token: this.token });
 
     this.heartbeat.stop();
   }
@@ -9907,7 +10024,7 @@ class WebSocketClient extends Emitter {
   }
 
   reconnect(reason        ) {
-    logger$5.info('Wazo WS reconnect', { reason, socket: !!this.socket, host: this.host, token: this.token });
+    logger$6.info('Wazo WS reconnect', { reason, socket: !!this.socket, host: this.host, token: this.token });
     if (!this.socket) {
       return;
     }
@@ -9963,7 +10080,7 @@ class WebSocketClient extends Emitter {
       this.close();
     }
     const url = `wss://${this.host || ''}/api/websocketd/?token=${this.token || ''}&version=${this.version}`;
-    logger$5.log('Wazo WS url computed to reconnect', { url });
+    logger$6.log('Wazo WS url computed to reconnect', { url });
 
     return url;
   }
@@ -9971,12 +10088,12 @@ class WebSocketClient extends Emitter {
   _onHeartbeat(message        ) {
     if (message.payload === 'pong') {
       this.heartbeat.onHeartbeat();
-      logger$5.log('on heartbeat received from Wazo WS');
+      logger$6.log('on heartbeat received from Wazo WS');
     }
   }
 
   async _onHeartbeatTimeout() {
-    logger$5.log('heartbeat timed out', { host: this.host, token: this.token });
+    logger$6.log('heartbeat timed out', { host: this.host, token: this.token });
 
     this.close();
     this.eventEmitter.emit(SOCKET_EVENTS$1.ON_CLOSE, new Error('Websocket ping failure.'));
@@ -10018,6 +10135,8 @@ WebSocketClient.eventLists = [
   CHATD_USER_ROOM_CREATED,
   CONFERENCE_USER_PARTICIPANT_JOINED,
   CONFERENCE_USER_PARTICIPANT_LEFT,
+  MEETING_USER_PARTICIPANT_JOINED,
+  MEETING_USER_PARTICIPANT_LEFT,
   CONFERENCE_USER_PARTICIPANT_TALK_STARTED,
   CONFERENCE_USER_PARTICIPANT_TALK_STOPPED,
   SWITCHBOARD_QUEUED_CALLS_UPDATED,
@@ -10088,6 +10207,8 @@ var WebSocketClient$1 = /*#__PURE__*/Object.freeze({
   CONFERENCE_USER_PARTICIPANT_LEFT: CONFERENCE_USER_PARTICIPANT_LEFT,
   CONFERENCE_USER_PARTICIPANT_TALK_STARTED: CONFERENCE_USER_PARTICIPANT_TALK_STARTED,
   CONFERENCE_USER_PARTICIPANT_TALK_STOPPED: CONFERENCE_USER_PARTICIPANT_TALK_STOPPED,
+  MEETING_USER_PARTICIPANT_JOINED: MEETING_USER_PARTICIPANT_JOINED,
+  MEETING_USER_PARTICIPANT_LEFT: MEETING_USER_PARTICIPANT_LEFT,
   SWITCHBOARD_QUEUED_CALLS_UPDATED: SWITCHBOARD_QUEUED_CALLS_UPDATED,
   SWITCHBOARD_QUEUED_CALL_ANSWERED: SWITCHBOARD_QUEUED_CALL_ANSWERED,
   SWITCHBOARD_HELD_CALLS_UPDATED: SWITCHBOARD_HELD_CALLS_UPDATED,
@@ -10129,7 +10250,7 @@ var WebSocketClient$1 = /*#__PURE__*/Object.freeze({
 const { SOCKET_EVENTS, ...OTHER_EVENTS } = WebSocketClient$1;
 const ALL_EVENTS = [...Object.values(SOCKET_EVENTS), ...Object.values(OTHER_EVENTS)];
 
-const logger$4 = IssueReporter$1.loggerFor('simple-ws-client');
+const logger$5 = IssueReporter$1.loggerFor('simple-ws-client');
 
 class Websocket extends Emitter {
                            
@@ -10153,7 +10274,7 @@ class Websocket extends Emitter {
   }
 
   open(host        , session         ) {
-    logger$4.info('open simple WebSocket', { host, token: session.token });
+    logger$5.info('open simple WebSocket', { host, token: session.token });
     this.ws = new WebSocketClient(
       {
         host,
@@ -10181,7 +10302,7 @@ class Websocket extends Emitter {
   }
 
   updateToken(token        ) {
-    logger$4.info('update token via simple Websocket', { token, ws: !!this.ws });
+    logger$5.info('update token via simple Websocket', { token, ws: !!this.ws });
 
     if (!this.ws) {
       return;
@@ -10209,7 +10330,7 @@ var Websocket$1 = global.wazoWebsocketInstance;
 
 //      
 
-const logger$3 = IssueReporter$1.loggerFor('room');
+const logger$4 = IssueReporter$1.loggerFor('room');
 
 class Participant extends Emitter {
              
@@ -10243,6 +10364,7 @@ class Participant extends Emitter {
                     
 
   /**
+   * @param room Room The room where the participant is
    * @param rawParticipant string Participant sent via the Wazo WS
    * @param extra Object extra status of the participant
    */
@@ -10330,7 +10452,7 @@ class Participant extends Emitter {
   }
 
   onTalking(isTalking         ) {
-    logger$3.info('on participant talking', { name: this.name, isTalking, callId: this.callId });
+    logger$4.info('on participant talking', { name: this.name, isTalking, callId: this.callId });
     this.isTalking = isTalking;
     // you may notice we're not broadcasting: since all participants are getting this info
     // directly from asterisk, there's no need to do so
@@ -10433,7 +10555,7 @@ class Participant extends Emitter {
   }
 
   updateStatus(status        , broadcast          = true) {
-    logger$3.info('updating participant status', { name: this.name, status });
+    logger$4.info('updating participant status', { name: this.name, status });
 
     if (typeof status.audioMuted !== 'undefined' && status.audioMuted !== this.audioMuted) {
       if (status.audioMuted) {
@@ -10470,13 +10592,18 @@ class Participant extends Emitter {
     // Poor man's object comparison
     if (typeof status.extra !== 'undefined' && JSON.stringify(this.extra) !== JSON.stringify(status.extra)) {
       this.extra = { ...this.extra, ...status.extra };
+
+      if (this.extra.contact && !(this.extra.contact instanceof Contact)) {
+        this.extra.contact = new Contact(this.extra.contact);
+      }
+
       this.triggerUpdate(this.ON_EXTRA_CHANGE, broadcast);
     }
   }
 
   broadcastStatus(inboundStatus         = null) {
     const status = inboundStatus || this.getStatus();
-    logger$3.info('broadcasting participant status', { callId: this.callId, status });
+    logger$4.info('broadcasting participant status', { callId: this.callId, status });
 
     this.room.sendSignal({
       type: SIGNAL_TYPE_PARTICIPANT_UPDATE,
@@ -10497,18 +10624,24 @@ class RemoteParticipant extends Participant {
   }
 }
 
-/* global document */
+//      
+
+class LocalParticipant extends Participant {
+}
+
+//      
 
 const SIGNAL_TYPE_PARTICIPANT_UPDATE = 'signal/PARTICIPANT_UPDATE';
 const SIGNAL_TYPE_PARTICIPANT_REQUEST = 'signal/PARTICIPANT_REQUEST';
 
-const logger$2 = IssueReporter$1.loggerFor('sdk-room');
+const logger$3 = IssueReporter$1.loggerFor('sdk-room');
 
 class Room$1 extends Emitter {
                             
                
                     
                     
+                       
                               
                   
                      
@@ -10530,6 +10663,8 @@ class Room$1 extends Emitter {
 
                                              
                                            
+                                          
+                                        
                                 
                      
                   
@@ -10547,6 +10682,7 @@ class Room$1 extends Emitter {
    * @param extension string
    * @param sourceId number
    * @param callId string
+   * @param meetingUuid string
    * @param extra Object
    */
   constructor(
@@ -10554,15 +10690,17 @@ class Room$1 extends Emitter {
     extension        ,
     sourceId         ,
     callId         ,
+    meetingUuid         ,
     extra         = {},
   ) {
     super();
-    logger$2.info('room initialized', { callId, extension, sourceId });
+    logger$3.info('room initialized', { callId, extension, sourceId, meetingUuid });
 
     // Represents the room callSession
     this.callSession = callSession;
     this.extension = extension;
     this.sourceId = sourceId;
+    this.meetingUuid = meetingUuid;
     this.callId = callId;
     this.participants = [];
     this.connected = false;
@@ -10582,6 +10720,8 @@ class Room$1 extends Emitter {
     // Sugar syntax for `room.EVENT_NAME`
     this.CONFERENCE_USER_PARTICIPANT_JOINED = Wazo.Websocket.CONFERENCE_USER_PARTICIPANT_JOINED;
     this.CONFERENCE_USER_PARTICIPANT_LEFT = Wazo.Websocket.CONFERENCE_USER_PARTICIPANT_LEFT;
+    this.MEETING_USER_PARTICIPANT_JOINED = Wazo.Websocket.MEETING_USER_PARTICIPANT_JOINED;
+    this.MEETING_USER_PARTICIPANT_LEFT = Wazo.Websocket.MEETING_USER_PARTICIPANT_LEFT;
     this.ON_SHARE_SCREEN_ENDED = Wazo.Phone.ON_SHARE_SCREEN_ENDED;
     this.ON_MESSAGE = Wazo.Phone.ON_MESSAGE;
     this.ON_CHAT = Wazo.Phone.ON_CHAT;
@@ -10616,10 +10756,11 @@ class Room$1 extends Emitter {
    * @param audioOnly boolean
    * @param extra Object
    * @param room ?Room
+   * @param meeting ?Meeting
    * @returns {Promise<Room>}
    */
-  static async connect({ extension, constraints, audioOnly = false, extra, room }        ) {
-    logger$2.info('connecting to room', { extension, audioOnly, room: !!room });
+  static async connect({ extension, constraints, audioOnly = false, extra, room, meeting }        ) {
+    logger$3.info('connecting to room', { extension, audioOnly, room: !!room });
 
     if (!room) {
       await Wazo.Phone.connect();
@@ -10632,7 +10773,7 @@ class Room$1 extends Emitter {
 
       // Call_created is triggered before call_accepted, so we have to listen for it here.
       Wazo.Websocket.once(Wazo.Websocket.CALL_CREATED, ({ data }) => {
-        logger$2.info('room call received via WS', { callId: data.call_id });
+        logger$3.info('room call received via WS', { callId: data.call_id });
         if (room) {
           room.setCallId(data.call_id);
         }
@@ -10653,34 +10794,44 @@ class Room$1 extends Emitter {
       room.setCallId(room.callSession.call.id);
     }
 
-    // Fetch conference source
-    const sources = await getApiClient().dird.fetchConferenceSource('default');
-    // Retrieve conference sources
-    const contacts = await getApiClient().dird.fetchConferenceContacts(sources.items[0]);
-    // Retrieve conference
-    const conference = contacts.find(contact => contact.numbers.find(number => number.number === extension));
+    if (!meeting) {
+      // Fetch conference source
+      const sources = await getApiClient().dird.fetchConferenceSource('default');
+      // Retrieve conference sources
+      const contacts = await getApiClient().dird.fetchConferenceContacts(sources.items[0]);
+      // Retrieve conference
+      const conference = contacts.find(contact => contact.numbers.find(number => number.number === extension));
 
-    logger$2.info('connected to room', {
-      sourceId: conference ? conference.sourceId : null,
-      name: conference ? conference.name : null,
-    });
+      logger$3.info('connected to room', {
+        sourceId: conference ? conference.sourceId : null,
+        name: conference ? conference.name : null,
+      });
 
-    if (conference) {
-      room.setSourceId(conference.sourceId);
-      room.setName(conference.name);
+      if (conference) {
+        room.setSourceId(conference.sourceId);
+        room.setName(conference.name);
+      }
+    } else if (meeting) {
+      logger$3.info('Already connected to meeting', {
+        uuid: meeting.uuid,
+        name: meeting.name,
+      });
+
+      room.setMeetingUuid(meeting.uuid);
+      room.setName(meeting.name);
     }
 
     return room;
   }
 
   static disconnect() {
-    logger$2.info('static disconnection to room');
+    logger$3.info('static disconnection to room');
 
     Wazo.Phone.disconnect();
   }
 
   async disconnect() {
-    logger$2.info('disconnection to room called');
+    logger$3.info('disconnection to room called');
 
     await Wazo.Phone.hangup(this.callSession);
     this.callSession = null;
@@ -10695,20 +10846,24 @@ class Room$1 extends Emitter {
     Wazo.Phone.phone.off(Wazo.Phone.phone.client.ON_REINVITE, this._boundOnReinvite);
     Wazo.Websocket.off(this.CONFERENCE_USER_PARTICIPANT_JOINED, this._boundOnParticipantJoined);
     Wazo.Websocket.off(this.CONFERENCE_USER_PARTICIPANT_LEFT, this._boundOnParticipantLeft);
-
-    if (this.roomAudioElement && document.body) {
-      document.body.removeChild(this.roomAudioElement);
-    }
+    Wazo.Websocket.off(this.MEETING_USER_PARTICIPANT_JOINED, this._boundOnParticipantJoined);
+    Wazo.Websocket.off(this.MEETING_USER_PARTICIPANT_LEFT, this._boundOnParticipantLeft);
   }
 
   setSourceId(sourceId        ) {
-    logger$2.info('set room source id', { sourceId });
+    logger$3.info('set room source id', { sourceId });
 
     this.sourceId = sourceId;
   }
 
+  setMeetingUuid(meetingUuid        ) {
+    logger$3.info('set meeting uuid', { meetingUuid });
+
+    this.meetingUuid = meetingUuid;
+  }
+
   setCallId(callId        ) {
-    logger$2.info('set room call id', { callId });
+    logger$3.info('set room call id', { callId });
 
     if (callId) {
       this.callId = callId;
@@ -10716,7 +10871,7 @@ class Room$1 extends Emitter {
   }
 
   setName(name        ) {
-    logger$2.info('set room name', { name });
+    logger$3.info('set room name', { name });
 
     this.name = name;
   }
@@ -10735,7 +10890,7 @@ class Room$1 extends Emitter {
   }
 
   async startScreenSharing(constraints        ) {
-    logger$2.info('start room screen sharing', { constraints });
+    logger$3.info('start room screen sharing', { constraints });
 
     const screensharingStream = await Wazo.Phone.startScreenSharing(constraints, this.callSession);
     if (!screensharingStream) {
@@ -10748,18 +10903,26 @@ class Room$1 extends Emitter {
     return screensharingStream;
   }
 
-  stopScreenSharing(restoreLocalStream          = true) {
-    logger$2.info('stop room screen sharing');
+  async stopScreenSharing(restoreLocalStream          = true) {
+    logger$3.info('stop room screen sharing');
 
-    Wazo.Phone.stopScreenSharing(this.callSession, restoreLocalStream);
+    const reinvited = await Wazo.Phone.stopScreenSharing(this.callSession, restoreLocalStream);
 
     if (this.localParticipant) {
       this.localParticipant.onStopScreensharing();
     }
+
+    if (!reinvited && this.localParticipant) {
+      // When we had video and we made a screenshare we don't reinvite when stopping the screensahre
+      // So not REINVITE even is fired, we have to restore the video stream manually
+      const localVideoStream = Wazo.Phone.getLocalVideoStream(this.callSession);
+      // $FlowFixMe
+      this._associateStreamTo(localVideoStream, this.localParticipant);
+    }
   }
 
   turnCameraOff() {
-    logger$2.info('turn room camera off');
+    logger$3.info('turn room camera off');
 
     Wazo.Phone.turnCameraOff(this.callSession);
 
@@ -10769,7 +10932,7 @@ class Room$1 extends Emitter {
   }
 
   turnCameraOn() {
-    logger$2.info('turn room camera on');
+    logger$3.info('turn room camera on');
 
     Wazo.Phone.turnCameraOn(this.callSession);
 
@@ -10779,7 +10942,7 @@ class Room$1 extends Emitter {
   }
 
   mute() {
-    logger$2.info('mute room');
+    logger$3.info('mute room');
 
     Wazo.Phone.mute(this.callSession);
 
@@ -10787,7 +10950,7 @@ class Room$1 extends Emitter {
   }
 
   unmute() {
-    logger$2.info('unmute room');
+    logger$3.info('unmute room');
 
     Wazo.Phone.unmute(this.callSession);
 
@@ -10807,7 +10970,7 @@ class Room$1 extends Emitter {
   }
 
   hold() {
-    logger$2.info('hold room');
+    logger$3.info('hold room');
 
     Wazo.Phone.hold(this.callSession, true);
 
@@ -10817,7 +10980,7 @@ class Room$1 extends Emitter {
   }
 
   resume() {
-    logger$2.info('resume room');
+    logger$3.info('resume room');
 
     Wazo.Phone.resume(this.callSession, true);
 
@@ -10827,13 +10990,13 @@ class Room$1 extends Emitter {
   }
 
   sendDTMF(tone        ) {
-    logger$2.info('send room DTMF', { tone });
+    logger$3.info('send room DTMF', { tone });
 
     Wazo.Phone.sendDTMF(tone, this.callSession);
   }
 
   async sendReinvite(newConstraints         = null) {
-    logger$2.info('send room reinvite', { callId: this.callSession ? this.callSession.getId() : null, newConstraints });
+    logger$3.info('send room reinvite', { callId: this.callSession ? this.callSession.getId() : null, newConstraints });
     const wasScreensharing = this.localParticipant && this.localParticipant.screensharing;
 
     Wazo.Phone.on(Wazo.Phone.ON_SHARE_SCREEN_STARTED, () => {
@@ -10858,6 +11021,22 @@ class Room$1 extends Emitter {
     return response;
   }
 
+  hasALocalVideoTrack() {
+    return Wazo.Phone.hasALocalVideoTrack(this.callSession);
+  }
+
+  getLocalStream() {
+    return Wazo.Phone.getLocalStream(this.callSession);
+  }
+
+  getRemoteStream() {
+    return Wazo.Phone.getRemoteStream(this.callSession);
+  }
+
+  getRemoteVideoStream() {
+    return Wazo.Phone.getRemoteVideoStream(this.callSession);
+  }
+
   _bindEvents() {
     if (!Wazo.Phone.phone || !Wazo.Phone.phone.currentSipSession) {
       return;
@@ -10876,35 +11055,18 @@ class Room$1 extends Emitter {
     Wazo.Phone.phone.on(Wazo.Phone.phone.client.ON_REINVITE, this._boundOnReinvite);
 
     this.on(this.ON_AUDIO_STREAM, stream => {
-      logger$2.info('on room audio stream');
+      logger$3.info('on room audio stream');
 
       this.audioStream = stream;
-      if (document.createElement) {
-        // $FlowFixMe
-        let roomAudioElement                   = document.getElementById('audio-room');
-        if (!this.roomAudioElement) {
-          roomAudioElement = document.createElement('audio');
-          // $FlowFixMe
-          document.body.appendChild(roomAudioElement);
-        }
-        if (!roomAudioElement) {
-          return;
-        }
-        this.roomAudioElement = roomAudioElement;
+      if (!this.roomAudioElement) {
+        const sessionId = Wazo.Phone.phone.getSipSessionId(Wazo.Phone.phone.currentSipSession);
+        this.roomAudioElement = Wazo.Phone.phone.createAudioElementFor(sessionId);
         this.roomAudioElement.srcObject = stream;
-        this.roomAudioElement.autoplay = true;
-        this.roomAudioElement.id = 'audio-room';
-
-        // $FlowFixMe
-        if (this.roomAudioElement.setSinkId) {
-          // $FlowFixMe
-          this.roomAudioElement.setSinkId(Wazo.Phone.getOutputDevice());
-        }
       }
     });
 
     this.on(this.ON_VIDEO_STREAM, (stream, streamId, event, sipSession) => {
-      logger$2.info('on room video stream', { streamId });
+      logger$3.info('on room video stream', { streamId });
 
       this._mapMsid(sipSession.body.body);
 
@@ -10919,7 +11081,7 @@ class Room$1 extends Emitter {
     });
 
     this.on(this.ON_REMOVE_STREAM, stream => {
-      logger$2.info('on room remove stream');
+      logger$3.info('on room remove stream');
 
       const participant = this.participants.find(someParticipant =>
         someParticipant.streams.find(someStream => someStream && someStream.id === stream.id));
@@ -10974,6 +11136,9 @@ class Room$1 extends Emitter {
     Wazo.Websocket.on(this.CONFERENCE_USER_PARTICIPANT_JOINED, this._boundOnParticipantJoined);
     Wazo.Websocket.on(this.CONFERENCE_USER_PARTICIPANT_LEFT, this._boundOnParticipantLeft);
 
+    Wazo.Websocket.on(this.MEETING_USER_PARTICIPANT_JOINED, this._boundOnParticipantJoined);
+    Wazo.Websocket.on(this.MEETING_USER_PARTICIPANT_LEFT, this._boundOnParticipantLeft);
+
     // Phone events
     Wazo.Phone.on(this.ON_MESSAGE, this._boundOnMessage);
     Wazo.Phone.on(this.ON_CHAT, this._boundOnChat);
@@ -10984,15 +11149,15 @@ class Room$1 extends Emitter {
       Wazo.Phone.on(event, (...args) => this.eventEmitter.emit.apply(this.eventEmitter, [event, ...args])));
   }
 
-  _onMessage(message         ) {
+  _onMessage(message         )          {
     if (message.method !== 'MESSAGE') {
-      return;
+      return null;
     }
     let body;
     try {
       body = JSON.parse(message.body);
     } catch (e) {
-      return;
+      return null;
     }
 
     switch (body.type) {
@@ -11011,6 +11176,8 @@ class Room$1 extends Emitter {
     }
 
     this.eventEmitter.emit(this.ON_MESSAGE, body);
+
+    return body;
   }
 
   _onChat(content        ) {
@@ -11046,7 +11213,7 @@ class Room$1 extends Emitter {
         const requester               = this._getParticipantFromCallId(origin.callId);
         if (requester) {
           // @FIXME?: when need to trigger an update on join-in; this is a bit of a hack
-          logger$2.info('trigger room requester status', { origin });
+          logger$3.info('trigger room requester status', { origin });
           requester.triggerUpdate('REQUESTER_UPDATE');
         }
         break;
@@ -11075,25 +11242,26 @@ class Room$1 extends Emitter {
     const session = Wazo.Auth.getSession();
     let participants = [];
 
-    // @TODO: we could use a better function name here
-    const isJoining = part => {
-      this.__associateStreams(part);
-      // @VALIDATE: no need to publicize ourselves, no?
-      if (part instanceof RemoteParticipant) {
-        this.eventEmitter.emit(this.CONFERENCE_USER_PARTICIPANT_JOINED, part);
-      }
-    };
-
     // When we join the room, we can call `getConferenceParticipantsAsUser`, not before.
     if (participant.user_uuid === session.uuid) {
-      logger$2.info('room current user joined');
+      logger$3.info('room current user joined');
 
       // Retrieve participants via an API calls
       const conferenceId = this.sourceId || payload.data.conference_id;
-      const response = await getApiClient().calld.getConferenceParticipantsAsUser(conferenceId);
-      logger$2.info('fetching conference participants', { conferenceId });
+      let response;
+      try {
+        if (this.meetingUuid) {
+          logger$3.info('fetching meeting participants', { meetingUuid: this.meetingUuid });
+          response = await getApiClient().calld.getMeetingParticipantsAsUser(this.meetingUuid);
+        } else {
+          logger$3.info('fetching conference participants', { conferenceId });
+          response = await getApiClient().calld.getConferenceParticipantsAsUser(conferenceId);
+        }
+      } catch (e) {
+        logger$3.error('room participants fetching, error', e);
+      }
       if (response) {
-        logger$2.info('conference participants fetched', { nb: response.items.length });
+        logger$3.info('conference participants fetched', { nb: response.items.length });
         participants = response.items.map(item => {
           const isMe = item.call_id === this.callId;
 
@@ -11106,25 +11274,10 @@ class Room$1 extends Emitter {
 
         const localParticipant = participants.find(someParticipant => someParticipant instanceof Wazo.LocalParticipant);
         if (!this.localParticipant && localParticipant) {
-          this.localParticipant = localParticipant;
-
-          const localVideoStream = this._getLocalVideoStream();
-          if (localVideoStream) {
-            this._saveLocalVideoStream(localVideoStream);
-          }
-
-          this.connected = true;
-
-          // we're in the room, now let's request everyone's status
-          if (this.localParticipant) {
-            this.sendSignal({
-              type: SIGNAL_TYPE_PARTICIPANT_REQUEST,
-              origin: this.localParticipant.getStatus(),
-            });
-          }
+          this._onLocalParticipantJoined(localParticipant);
         }
 
-        participants.forEach(someParticipant => isJoining(someParticipant));
+        participants.forEach(someParticipant => this._isParticipantJoining(someParticipant));
         this.eventEmitter.emit(this.ON_JOINED, localParticipant, participants);
       }
 
@@ -11135,16 +11288,45 @@ class Room$1 extends Emitter {
       ? new Wazo.RemoteParticipant(this, participant)
       : null;
 
-    logger$2.info('other room user joined', { callId: participant.call_id, remoteParticipant: !!remoteParticipant });
+    logger$3.info('other room user joined', { callId: participant.call_id, remoteParticipant: !!remoteParticipant });
 
     if (remoteParticipant) {
       this.participants.push(remoteParticipant);
-      isJoining(remoteParticipant);
+      this._isParticipantJoining(remoteParticipant);
     }
 
     return remoteParticipant;
   }
 
+  _onLocalParticipantJoined(localParticipant                  ) {
+    this.localParticipant = localParticipant;
+
+    const localVideoStream = this._getLocalVideoStream();
+    if (localVideoStream) {
+      this._saveLocalVideoStream(localVideoStream);
+    }
+
+    this.connected = true;
+
+    localParticipant.broadcastStatus();
+
+    // we're in the room, now let's request everyone's status
+    if (this.localParticipant) {
+      this.sendSignal({
+        type: SIGNAL_TYPE_PARTICIPANT_REQUEST,
+        origin: this.localParticipant.getStatus(),
+      });
+    }
+  }
+
+  _isParticipantJoining(participant             ) {
+    this.__associateStreams(participant);
+    // @VALIDATE: no need to publicize ourselves, no?
+    if (participant instanceof RemoteParticipant) {
+      this.eventEmitter.emit(this.CONFERENCE_USER_PARTICIPANT_JOINED, participant);
+    }
+  }
+
   _saveLocalVideoStream(stream             ) {
     const { localParticipant } = this;
 
@@ -11190,9 +11372,11 @@ class Room$1 extends Emitter {
   _onParticipantTrackUpdate(oldParticipant             , update        )              {
     const newParticipant = oldParticipant;
 
-    const { trackId } = this._callIdStreamIdMap[newParticipant.callId] || {};
+    const { trackId, streamId } = this._callIdStreamIdMap[newParticipant.callId] || {};
     const pc = Wazo.Phone.phone.currentSipSession.sessionDescriptionHandler.peerConnection;
-    const stream = pc.getRemoteStreams().find(someStream => someStream.getTracks().some(track => track.id === trackId));
+    // Can't use `getReceivers` here because on FF we make the mapping based on the streamId
+    const stream = pc.getRemoteStreams().find(someStream =>
+      someStream.id === streamId || someStream.getTracks().some(track => track.id === trackId));
 
     if (update === 'downgrade') {
       newParticipant.streams = [];
@@ -11222,15 +11406,42 @@ class Room$1 extends Emitter {
       return;
     }
 
-    if (this._unassociatedVideoStreams[trackId]) {
+    const streamId = this._getStreamIdFrTrackId(trackId);
+    const key = this._getUnassociatedMapIdFromTrackIdOrStreamId(trackId, streamId);
+    const stream = this._unassociatedVideoStreams[key];
+
+    if (stream) {
       // Try to associate stream
-      this._associateStreamTo(this._unassociatedVideoStreams[trackId], participant);
+      this._associateStreamTo(stream, participant);
 
-      delete this._unassociatedVideoStreams[trackId];
+      delete this._unassociatedVideoStreams[key];
       delete this._unassociatedParticipants[participant.callId];
     }
   }
 
+  _getUnassociatedMapIdFromTrackIdOrStreamId(trackId        , streamId         ) {
+    // Find by trackId
+    if (trackId in this._unassociatedVideoStreams) {
+      return trackId;
+    }
+
+    // Find by streamId
+    if (streamId && streamId in this._unassociatedVideoStreams) {
+      return streamId;
+    }
+
+    // Find in all the streams by streamId (used on FF where we can't map by trackId)
+    const idx = Object.values(this._unassociatedVideoStreams).findIndex((stream        ) =>
+      stream && stream.id === streamId);
+    return idx === -1 ? null : Object.keys(this._unassociatedVideoStreams)[idx];
+  }
+
+  _getStreamIdFrTrackId(trackId        ) {
+    const mapping         = Object.values(this._callIdStreamIdMap).find((map        ) => map.trackId === trackId);
+
+    return mapping ? mapping.streamId : null;
+  }
+
   _associateStreamTo(rawStream     , participant             ) {
     const stream = new Wazo.Stream(rawStream, participant);
     participant.streams.push(stream);
@@ -11254,7 +11465,141 @@ class Room$1 extends Emitter {
 
 //      
 
-class LocalParticipant extends Participant {
+const logger$2 = IssueReporter$1.loggerFor('sdk-sip-room');
+
+class SipRoom extends Room$1 {
+
+  static async connect({ extension, constraints, audioOnly = false, extra, room }        ) {
+    logger$2.info('connecting to sip room', { extension, audioOnly, room: !!room });
+
+    if (!room) {
+      await Wazo.Phone.connect();
+
+      const withCamera = constraints && !!constraints.video;
+
+      const callSession = await Wazo.Phone.call(extension, withCamera, null, audioOnly, true);
+      // eslint-disable-next-line no-param-reassign
+      room = new SipRoom(callSession, extension, null, null, extra);
+    }
+
+    if (room && room.callSession && room.callSession.call) {
+      room.setCallId(room.callSession.call.id);
+    }
+
+    logger$2.info('connected to room', {
+      extension: room.extension,
+    });
+
+    return room;
+  }
+
+  mute() {
+    logger$2.info('mute sip room');
+
+    Wazo.Phone.mute(this.callSession, false);
+
+    this.sendMuteStatus();
+  }
+
+  unmute() {
+    logger$2.info('unmute sip room');
+
+    Wazo.Phone.unmute(this.callSession, false);
+
+    this.sendUnMuteStatus();
+  }
+
+  getLocalGuestName() {
+    return Wazo.Phone.phone.client.userAgent.options.displayName;
+  }
+
+  // Overridden to not listen to websocket messages
+  _transferEvents() {
+    // Phone events
+    Wazo.Phone.on(this.ON_MESSAGE, this._boundOnMessage);
+    Wazo.Phone.on(this.ON_CHAT, this._boundOnChat);
+    Wazo.Phone.on(this.ON_SIGNAL, this._boundOnSignal);
+    Wazo.Phone.on(this.ON_VIDEO_INPUT_CHANGE, this._boundSaveLocalVideoStream);
+
+    [this.ON_AUDIO_STREAM, this.ON_VIDEO_STREAM, this.ON_REMOVE_STREAM].forEach(event =>
+      Wazo.Phone.on(event, (...args) => this.eventEmitter.emit.apply(this.eventEmitter, [event, ...args])));
+  }
+
+  _onMessage(message         ) {
+    // eslint-disable-next-line no-underscore-dangle
+    const body = super._onMessage(message);
+    if (!body) {
+      return;
+    }
+
+    const getChannel = () => body.channels[0];
+
+    switch (body.type) {
+      case 'ConfbridgeWelcome':
+        body.channels.forEach(channel => {
+          this._onParticipantJoined(channel);
+        });
+
+        break;
+      case 'ConfbridgeJoin': {
+        const channel = getChannel();
+        this._onParticipantJoined(channel);
+        break;
+      }
+      case 'ConfbridgeLeave': {
+        const channel = getChannel();
+        this._onParticipantLeft({
+          data: {
+            call_id: channel.id,
+          },
+        });
+        break;
+      }
+    }
+  }
+
+  async _onParticipantJoined(channel        ) {
+    const isLocal = channel.channelvars.WAZO_SIP_CALL_ID === this._getCurrentSipCallIs();
+    const callId = channel.id;
+    const ParticipantClass = isLocal ? Wazo.LocalParticipant : Wazo.RemoteParticipant;
+    const name = channel.caller ? channel.caller.name : null;
+    const extra = isLocal ? { guestName: this.getLocalGuestName() } : {};
+    const participant = new ParticipantClass(this, {
+      caller_id_name: name,
+      call_id: callId,
+    }, extra);
+
+    const participantIdx = this.participants.findIndex(other => other.callId === participant.callId);
+    if (participantIdx !== -1 && name) {
+      this.participants[participantIdx].name = name;
+      return;
+    }
+
+    if (isLocal) {
+      // Should be updated by the Wazo WS but we don't have it for now
+      if (this.callSession) {
+        this.callSession.ringing = false;
+      }
+
+      this._onLocalParticipantJoined(participant);
+
+      // Give some time for the stream to be updated
+      setTimeout(() => {
+        this.eventEmitter.emit(this.ON_JOINED, participant, this.participants);
+      }, 1000);
+    }
+
+    this.participants.push(participant);
+
+    this._isParticipantJoining(participant);
+
+    return participant;
+  }
+
+  _getCurrentSipCallIs() {
+    return Wazo.Phone.getSipSessionId(Wazo.Phone.phone.currentSipSession);
+  }
+
 }
 
 /* global document */
@@ -11307,9 +11652,7 @@ class Stream {
       return false;
     }
 
-    // With canvas track (used to fake upgrade in conference), we don't have a width
-    return this.htmlStream.getTracks().find(track => track.kind === 'video'
-      && track.readyState !== 'ended' && 'width' in track.getSettings());
+    return this.htmlStream.getTracks().some(track => track.kind === 'video' && track.readyState !== 'ended');
   }
 
   get id() {
@@ -12210,6 +12553,7 @@ const Wazo = {
   Phone: Phone$1,
   Websocket: Websocket$1,
   Room: Room$1,
+  SipRoom,
   RemoteParticipant,
   LocalParticipant,
   Stream,
@@ -12246,6 +12590,7 @@ const Wazo = {
     IndirectTransfer,
     SwitchboardCall,
     WebRTCPhone,
+    Meeting,
   },
 
   // Api
@@ -12432,13 +12777,18 @@ class Phone extends Emitter {
     return adHocConference.start();
   }
 
-  mute(callSession             ) {
-    this.muteViaAPI(callSession);
+  mute(callSession             , withApi          = true) {
+    if (withApi) {
+      this.muteViaAPI(callSession);
+    }
+
     return this.phone && this.phone.mute(callSession);
   }
 
-  unmute(callSession             ) {
-    this.unmuteViaAPI(callSession);
+  unmute(callSession             , withApi          = true) {
+    if (withApi) {
+      this.unmuteViaAPI(callSession);
+    }
 
     return this.phone && this.phone.unmute(callSession);
   }
@@ -12544,6 +12894,10 @@ class Phone extends Emitter {
     return this.phone && this.phone.hasLocalVideo(callSession);
   }
 
+  hasALocalVideoTrack(callSession             ) {
+    return this.phone && this.phone.hasALocalVideoTrack(callSession);
+  }
+
   // @Deprecated
   getLocalMediaStream(callSession             ) {
     logger.warn('Phone.getLocalMediaStream is deprecated, use Phone.getLocalStream instead');