@wazo/sdk 0.37.23 → 0.37.27

package/lib/simple/room/SipRoom.js

@@ -116,7 +116,7 @@ const logger$c = IssueReporter$1 ? IssueReporter$1.loggerFor('api') : console;
 
 // Use a function here to be able to mock it in tests
 const realFetch = () => {
-  if (typeof document !== 'undefined') {
+  if (typeof document !== 'undefined' && window.fetch) {
     // Browser
     return window.fetch;
   }
@@ -1933,7 +1933,7 @@ class Contact {
       this[key] = old[key] || this[key];
     });
 
-    if (old.lineState && !this.lineState) {
+    if (old.lineState) {
       this.lineState = old.lineState;
     }
 
@@ -2640,8 +2640,9 @@ class ExternalApp {
                              
                
                
-                 
+                
                       
+                        
  
 
 class Meeting {
@@ -2652,9 +2653,10 @@ class Meeting {
                
                
                
-                 
+                    
                       
                             
+                     
 
   static parse(plain                         )          {
     return new Meeting({
@@ -2664,8 +2666,9 @@ class Meeting {
       ownerUuids: plain.owner_uuids,
       port: plain.port,
       uuid: plain.uuid,
-      number: plain.number,
+      extension: plain.exten,
       persistent: plain.persistent,
+      creationTime: moment__default['default'](plain.creation_time).toDate(),
     });
   }
 
@@ -2687,8 +2690,9 @@ class Meeting {
     ownerUuids,
     port,
     uri,
-    number,
+    extension,
     persistent,
+    creationTime,
   }         = {}) {
     this.guestSipAuthorization = guestSipAuthorization;
     this.uri = uri;
@@ -2696,8 +2700,9 @@ class Meeting {
     this.ownerUuids = ownerUuids;
     this.port = port;
     this.uuid = uuid;
-    this.number = number;
+    this.extension = extension;
     this.persistent = persistent;
+    this.creationTime = creationTime;
 
     // Useful to compare instead of instanceof with minified code
     this.type = 'Meeting';
@@ -6067,11 +6072,16 @@ class WazoSessionDescriptionHandler extends sessionDescriptionHandler.SessionDes
 
   // $FlowFixMe
   setRemoteSessionDescription(sessionDescription                           )                {
-    const result = super.setRemoteSessionDescription(sessionDescription);
+    try {
+      const result = super.setRemoteSessionDescription(sessionDescription);
 
-    this.eventEmitter.emit('setDescription', sessionDescription);
+      this.eventEmitter.emit('setDescription', sessionDescription);
 
-    return result;
+      return result;
+    } catch (error) {
+      this.logger.error('SessionDescriptionHandler.setRemoteSessionDescription error', error);
+      throw error;
+    }
   }
 
   // Overridden to avoid to fix ice-candidates missing in the SDP in react-native and chrome canary.
@@ -6976,7 +6986,7 @@ class WebRTCClient extends Emitter {
   }
 
   call(number        , enableVideo          , audioOnly          = false, conference          = false)          {
-    logger$8.info('sdk webrtc creating call', { number, enableVideo, audioOnly });
+    logger$8.info('sdk webrtc creating call', { number, enableVideo, audioOnly, conference });
     const inviterOptions         = {
       sessionDescriptionHandlerOptionsReInvite: {
         conference,
@@ -8052,6 +8062,7 @@ class WebRTCClient extends Emitter {
     const screenSharing = constraints && 'screen' in constraints ? constraints.screen : false;
     const isDesktop = constraints && 'desktop' in constraints ? constraints.desktop : false;
     const withAudio = constraints && 'audio' in constraints ? constraints.audio : true;
+    const mandatoryVideo = constraints && typeof constraints.video === 'object' ? constraints.video.mandatory : {};
 
     logger$8.info('Retrieving media a configuration', { enableVideo, screenSharing, isDesktop, withAudio, constraints });
 
@@ -8059,7 +8070,8 @@ class WebRTCClient extends Emitter {
       constraints: {
         // Exact constraint are not supported with `getDisplayMedia` and we must have a video=false in desktop screenshare
         audio: screenSharing ? !isDesktop : (withAudio ? this._getAudioConstraints() : false),
-        video: screenSharing ? (isDesktop ? ({ mandatory: { chromeMediaSource: 'desktop' } }) : { cursor: 'always' })
+        video: screenSharing ? (isDesktop
+          ? ({ mandatory: { chromeMediaSource: 'desktop', ...(mandatoryVideo || {}) } }) : { cursor: 'always' })
           : this._getVideoConstraints(enableVideo),
         screen: screenSharing,
         desktop: isDesktop,
@@ -8394,12 +8406,14 @@ class WebRTCClient extends Emitter {
 
     session.sessionDescriptionHandler.remoteMediaStream.onaddtrack = onTrack;
 
-    pc.oniceconnectionstatechange = () => {
-      logger$8.info('on ice connection state changed', { state: pc.iceConnectionState });
-      if (pc.iceConnectionState === 'disconnected') {
-        this.eventEmitter.emit(ON_DISCONNECTED$1);
-      }
-    };
+    if (pc) {
+      pc.oniceconnectionstatechange = () => {
+        logger$8.info('on ice connection state changed', { state: pc.iceConnectionState });
+        if (pc.iceConnectionState === 'disconnected') {
+          this.eventEmitter.emit(ON_DISCONNECTED$1);
+        }
+      };
+    }
 
     if (withEvent) {
       this.eventEmitter.emit(ACCEPTED, session, sessionDialog);
@@ -8609,13 +8623,17 @@ class WebRTCClient extends Emitter {
     const nbStats = this.sessionNetworkStats[sessionId].length;
     const lastNetworkStats = this.sessionNetworkStats[sessionId][nbStats - 1];
     const lastAudioSent = lastNetworkStats ? lastNetworkStats.audioBytesSent : 0;
+    const lastAudioContentSent = lastNetworkStats ? lastNetworkStats.audioContentSent : 0;
     const lastVideoSent = lastNetworkStats ? lastNetworkStats.videoBytesSent : 0;
     const lastAudioReceived = lastNetworkStats ? lastNetworkStats.audioBytesReceived : 0;
+    const lastAudioContentReceived = lastNetworkStats ? lastNetworkStats.audioContentReceived : 0;
     const lastVideoReceived = lastNetworkStats ? lastNetworkStats.videoBytesReceived : 0;
     const lastTransportSent = lastNetworkStats ? lastNetworkStats.transportSent : 0;
     const lastTransportReceived = lastNetworkStats ? lastNetworkStats.transportReceived : 0;
-    let audioBytesSent = 0;
-    let audioBytesReceived = 0;
+    let audioBytesSent = 0; // content + header
+    let audioBytesReceived = 0; // content + header
+    let audioContentSent = 0; // useful to detect blank call
+    let audioContentReceived = 0;
     let videoBytesSent = 0;
     let videoBytesReceived = 0;
     let transportSent = 0;
@@ -8625,6 +8643,7 @@ class WebRTCClient extends Emitter {
     stats.forEach(report => {
       if (report.type === 'outbound-rtp' && report.kind === 'audio') {
         audioBytesSent += Number(report.bytesSent) + Number(report.headerBytesSent);
+        audioContentSent += Number(report.bytesSent);
       }
       if (report.type === 'remote-outbound-rtp' && report.kind === 'audio') {
         audioBytesSent += Number(report.bytesSent);
@@ -8636,6 +8655,7 @@ class WebRTCClient extends Emitter {
         packetsLost += Number(report.packetsLost);
         networkStats.packetsReceived = Number(report.packetsReceived);
         audioBytesReceived += Number(report.bytesReceived) + Number(report.headerBytesReceived);
+        audioContentReceived += Number(report.bytesReceived);
       }
       if (report.type === 'inbound-rtp' && report.kind === 'video') {
         videoBytesReceived += Number(report.bytesReceived) + Number(report.headerBytesReceived);
@@ -8664,10 +8684,17 @@ class WebRTCClient extends Emitter {
     });
 
     networkStats.packetsLost = packetsLost;
+
     networkStats.totalAudioBytesSent = audioBytesSent;
     networkStats.audioBytesSent = audioBytesSent - lastAudioSent;
     networkStats.totalAudioBytesReceived = audioBytesReceived - lastAudioReceived;
     networkStats.audioBytesReceived = audioBytesReceived;
+
+    networkStats.totalAudioContentSent = audioContentSent;
+    networkStats.audioContentSent = audioContentSent - lastAudioContentSent;
+    networkStats.totalAudioBytesReceived = audioContentReceived - lastAudioContentReceived;
+    networkStats.audioContentReceived = audioContentReceived;
+
     networkStats.totalVideoBytesSent = videoBytesSent;
     networkStats.videoBytesSent = videoBytesSent - lastVideoSent;
     networkStats.totalVideoBytesReceived = videoBytesReceived;
@@ -8879,6 +8906,7 @@ class WebRTCPhone extends Emitter                  {
       constraints,
       audioOnly,
       conference,
+      iceRestart,
     });
 
     if (!sipSession) {
@@ -8982,6 +9010,11 @@ class WebRTCPhone extends Emitter                  {
 
     this.client.downgradeToAudio(sipSession);
 
+    if (callSession) {
+      callSession.cameraEnabled = false;
+      this._updateCallSession(callSession);
+    }
+
     if (withMessage) {
       this._sendReinviteMessage(callSession, false);
     }
@@ -9009,6 +9042,11 @@ class WebRTCPhone extends Emitter                  {
 
     const newStream = await this.client.upgradeToVideo(sipSession, constraints, isConference);
 
+    if (callSession) {
+      callSession.cameraEnabled = true;
+      this._updateCallSession(callSession);
+    }
+
     // If no stream is returned, it means we have to reinvite
     if (!newStream) {
       return true;
@@ -9278,7 +9316,7 @@ class WebRTCPhone extends Emitter                  {
   }
 
   _onCallAccepted(sipSession         , cameraEnabled         )              {
-    logger$7.info('WebRTC phone -  on call accepted', { sipId: sipSession.id, cameraEnabled });
+    logger$7.info('WebRTC phone - on call accepted', { sipId: sipSession.id, cameraEnabled });
 
     const callSession = this._createAcceptedCallSession(sipSession, cameraEnabled);
     this.currentSipSession = sipSession;
@@ -9347,6 +9385,7 @@ class WebRTCPhone extends Emitter                  {
   _onCallTerminated(sipSession         ) {
     logger$7.info('WebRTC phone - on call terminated', { sipId: sipSession.id });
 
+    const sipSessionId = this.getSipSessionId(sipSession);
     const callSession = this._createCallSession(sipSession);
     const isCurrentSession = this.isCurrentCallSipSession(callSession);
     const isCurrentIncomingCall = callSession.is(this.getIncomingCallSession());
@@ -9364,7 +9403,6 @@ class WebRTCPhone extends Emitter                  {
       }, 5);
     }
 
-    const sipSessionId = this.getSipSessionId(sipSession);
     if (sipSessionId) {
       this.removeIncomingSessions(sipSessionId);
     }
@@ -9790,7 +9828,7 @@ 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$7.info('make WebRTC call', { number, lineId: line ? line.id : null, cameraEnabled, audioOnly, conference });
     if (!number) {
       return Promise.resolve(null);
     }
@@ -10009,8 +10047,12 @@ class WebRTCPhone extends Emitter                  {
     }
 
     const sessionId = this.incomingSessions[0];
+    const sipSession = this.client.getSipSession(sessionId);
+    if (!sipSession) {
+      return null;
+    }
 
-    return this._createCallSession(this.client.getSipSession(sessionId));
+    return this._createCallSession(sipSession);
   }
 
   sendMessage(sipSession          = null, body        , contentType         = 'text/plain') {
@@ -10151,9 +10193,13 @@ class WebRTCPhone extends Emitter                  {
       if (this.shouldSendReinvite && this.currentSipSession
         && this.currentSipSession.state === sessionState.SessionState.Established) {
         this.shouldSendReinvite = false;
+        const pc = this.currentSipSession.sessionDescriptionHandler.peerConnection;
+        const isConference = pc ? pc.sfu : false;
+        const hasVideo = this.currentCallSession && this.currentCallSession.cameraEnabled;
+
         try {
           // Send reinvite with iceRestart
-          this.sendReinvite(this.currentCallSession, null, false, false, true);
+          this.sendReinvite(this.currentCallSession, null, isConference, !hasVideo, true);
         } catch (e) {
           logger$7.error('WebRTC reinvite after register, error', { message: e.message, stack: e.stack });
         }
@@ -10237,6 +10283,15 @@ class WebRTCPhone extends Emitter                  {
     return this.client.getSipSessionId(sipSession);
   }
 
+  _updateCallSession(callSession             ) {
+    const sipSession = this.findSipSession(callSession);
+    const sipSessionId = sipSession ? this.getSipSessionId(sipSession) : null;
+
+    if (sipSessionId) {
+      this.callSessions[sipSessionId] = callSession;
+    }
+  }
+
   _onMessage(message         ) {
     if (!message || message.method !== 'MESSAGE') {
       return;

package/lib/simple/room/Stream.js

@@ -1246,7 +1246,7 @@ class Contact {
       this[key] = old[key] || this[key];
     });
 
-    if (old.lineState && !this.lineState) {
+    if (old.lineState) {
       this.lineState = old.lineState;
     }
 
@@ -1961,7 +1961,7 @@ const logger$c = IssueReporter$1 ? IssueReporter$1.loggerFor('api') : console;
 
 // Use a function here to be able to mock it in tests
 const realFetch = () => {
-  if (typeof document !== 'undefined') {
+  if (typeof document !== 'undefined' && window.fetch) {
     // Browser
     return window.fetch;
   }
@@ -2640,8 +2640,9 @@ class ExternalApp {
                              
                
                
-                 
+                
                       
+                        
  
 
 class Meeting {
@@ -2652,9 +2653,10 @@ class Meeting {
                
                
                
-                 
+                    
                       
                             
+                     
 
   static parse(plain                         )          {
     return new Meeting({
@@ -2664,8 +2666,9 @@ class Meeting {
       ownerUuids: plain.owner_uuids,
       port: plain.port,
       uuid: plain.uuid,
-      number: plain.number,
+      extension: plain.exten,
       persistent: plain.persistent,
+      creationTime: moment__default['default'](plain.creation_time).toDate(),
     });
   }
 
@@ -2687,8 +2690,9 @@ class Meeting {
     ownerUuids,
     port,
     uri,
-    number,
+    extension,
     persistent,
+    creationTime,
   }         = {}) {
     this.guestSipAuthorization = guestSipAuthorization;
     this.uri = uri;
@@ -2696,8 +2700,9 @@ class Meeting {
     this.ownerUuids = ownerUuids;
     this.port = port;
     this.uuid = uuid;
-    this.number = number;
+    this.extension = extension;
     this.persistent = persistent;
+    this.creationTime = creationTime;
 
     // Useful to compare instead of instanceof with minified code
     this.type = 'Meeting';
@@ -6067,11 +6072,16 @@ class WazoSessionDescriptionHandler extends sessionDescriptionHandler.SessionDes
 
   // $FlowFixMe
   setRemoteSessionDescription(sessionDescription                           )                {
-    const result = super.setRemoteSessionDescription(sessionDescription);
+    try {
+      const result = super.setRemoteSessionDescription(sessionDescription);
 
-    this.eventEmitter.emit('setDescription', sessionDescription);
+      this.eventEmitter.emit('setDescription', sessionDescription);
 
-    return result;
+      return result;
+    } catch (error) {
+      this.logger.error('SessionDescriptionHandler.setRemoteSessionDescription error', error);
+      throw error;
+    }
   }
 
   // Overridden to avoid to fix ice-candidates missing in the SDP in react-native and chrome canary.
@@ -6976,7 +6986,7 @@ class WebRTCClient extends Emitter {
   }
 
   call(number        , enableVideo          , audioOnly          = false, conference          = false)          {
-    logger$8.info('sdk webrtc creating call', { number, enableVideo, audioOnly });
+    logger$8.info('sdk webrtc creating call', { number, enableVideo, audioOnly, conference });
     const inviterOptions         = {
       sessionDescriptionHandlerOptionsReInvite: {
         conference,
@@ -8052,6 +8062,7 @@ class WebRTCClient extends Emitter {
     const screenSharing = constraints && 'screen' in constraints ? constraints.screen : false;
     const isDesktop = constraints && 'desktop' in constraints ? constraints.desktop : false;
     const withAudio = constraints && 'audio' in constraints ? constraints.audio : true;
+    const mandatoryVideo = constraints && typeof constraints.video === 'object' ? constraints.video.mandatory : {};
 
     logger$8.info('Retrieving media a configuration', { enableVideo, screenSharing, isDesktop, withAudio, constraints });
 
@@ -8059,7 +8070,8 @@ class WebRTCClient extends Emitter {
       constraints: {
         // Exact constraint are not supported with `getDisplayMedia` and we must have a video=false in desktop screenshare
         audio: screenSharing ? !isDesktop : (withAudio ? this._getAudioConstraints() : false),
-        video: screenSharing ? (isDesktop ? ({ mandatory: { chromeMediaSource: 'desktop' } }) : { cursor: 'always' })
+        video: screenSharing ? (isDesktop
+          ? ({ mandatory: { chromeMediaSource: 'desktop', ...(mandatoryVideo || {}) } }) : { cursor: 'always' })
           : this._getVideoConstraints(enableVideo),
         screen: screenSharing,
         desktop: isDesktop,
@@ -8394,12 +8406,14 @@ class WebRTCClient extends Emitter {
 
     session.sessionDescriptionHandler.remoteMediaStream.onaddtrack = onTrack;
 
-    pc.oniceconnectionstatechange = () => {
-      logger$8.info('on ice connection state changed', { state: pc.iceConnectionState });
-      if (pc.iceConnectionState === 'disconnected') {
-        this.eventEmitter.emit(ON_DISCONNECTED$1);
-      }
-    };
+    if (pc) {
+      pc.oniceconnectionstatechange = () => {
+        logger$8.info('on ice connection state changed', { state: pc.iceConnectionState });
+        if (pc.iceConnectionState === 'disconnected') {
+          this.eventEmitter.emit(ON_DISCONNECTED$1);
+        }
+      };
+    }
 
     if (withEvent) {
       this.eventEmitter.emit(ACCEPTED, session, sessionDialog);
@@ -8609,13 +8623,17 @@ class WebRTCClient extends Emitter {
     const nbStats = this.sessionNetworkStats[sessionId].length;
     const lastNetworkStats = this.sessionNetworkStats[sessionId][nbStats - 1];
     const lastAudioSent = lastNetworkStats ? lastNetworkStats.audioBytesSent : 0;
+    const lastAudioContentSent = lastNetworkStats ? lastNetworkStats.audioContentSent : 0;
     const lastVideoSent = lastNetworkStats ? lastNetworkStats.videoBytesSent : 0;
     const lastAudioReceived = lastNetworkStats ? lastNetworkStats.audioBytesReceived : 0;
+    const lastAudioContentReceived = lastNetworkStats ? lastNetworkStats.audioContentReceived : 0;
     const lastVideoReceived = lastNetworkStats ? lastNetworkStats.videoBytesReceived : 0;
     const lastTransportSent = lastNetworkStats ? lastNetworkStats.transportSent : 0;
     const lastTransportReceived = lastNetworkStats ? lastNetworkStats.transportReceived : 0;
-    let audioBytesSent = 0;
-    let audioBytesReceived = 0;
+    let audioBytesSent = 0; // content + header
+    let audioBytesReceived = 0; // content + header
+    let audioContentSent = 0; // useful to detect blank call
+    let audioContentReceived = 0;
     let videoBytesSent = 0;
     let videoBytesReceived = 0;
     let transportSent = 0;
@@ -8625,6 +8643,7 @@ class WebRTCClient extends Emitter {
     stats.forEach(report => {
       if (report.type === 'outbound-rtp' && report.kind === 'audio') {
         audioBytesSent += Number(report.bytesSent) + Number(report.headerBytesSent);
+        audioContentSent += Number(report.bytesSent);
       }
       if (report.type === 'remote-outbound-rtp' && report.kind === 'audio') {
         audioBytesSent += Number(report.bytesSent);
@@ -8636,6 +8655,7 @@ class WebRTCClient extends Emitter {
         packetsLost += Number(report.packetsLost);
         networkStats.packetsReceived = Number(report.packetsReceived);
         audioBytesReceived += Number(report.bytesReceived) + Number(report.headerBytesReceived);
+        audioContentReceived += Number(report.bytesReceived);
       }
       if (report.type === 'inbound-rtp' && report.kind === 'video') {
         videoBytesReceived += Number(report.bytesReceived) + Number(report.headerBytesReceived);
@@ -8664,10 +8684,17 @@ class WebRTCClient extends Emitter {
     });
 
     networkStats.packetsLost = packetsLost;
+
     networkStats.totalAudioBytesSent = audioBytesSent;
     networkStats.audioBytesSent = audioBytesSent - lastAudioSent;
     networkStats.totalAudioBytesReceived = audioBytesReceived - lastAudioReceived;
     networkStats.audioBytesReceived = audioBytesReceived;
+
+    networkStats.totalAudioContentSent = audioContentSent;
+    networkStats.audioContentSent = audioContentSent - lastAudioContentSent;
+    networkStats.totalAudioBytesReceived = audioContentReceived - lastAudioContentReceived;
+    networkStats.audioContentReceived = audioContentReceived;
+
     networkStats.totalVideoBytesSent = videoBytesSent;
     networkStats.videoBytesSent = videoBytesSent - lastVideoSent;
     networkStats.totalVideoBytesReceived = videoBytesReceived;
@@ -8879,6 +8906,7 @@ class WebRTCPhone extends Emitter                  {
       constraints,
       audioOnly,
       conference,
+      iceRestart,
     });
 
     if (!sipSession) {
@@ -8982,6 +9010,11 @@ class WebRTCPhone extends Emitter                  {
 
     this.client.downgradeToAudio(sipSession);
 
+    if (callSession) {
+      callSession.cameraEnabled = false;
+      this._updateCallSession(callSession);
+    }
+
     if (withMessage) {
       this._sendReinviteMessage(callSession, false);
     }
@@ -9009,6 +9042,11 @@ class WebRTCPhone extends Emitter                  {
 
     const newStream = await this.client.upgradeToVideo(sipSession, constraints, isConference);
 
+    if (callSession) {
+      callSession.cameraEnabled = true;
+      this._updateCallSession(callSession);
+    }
+
     // If no stream is returned, it means we have to reinvite
     if (!newStream) {
       return true;
@@ -9278,7 +9316,7 @@ class WebRTCPhone extends Emitter                  {
   }
 
   _onCallAccepted(sipSession         , cameraEnabled         )              {
-    logger$7.info('WebRTC phone -  on call accepted', { sipId: sipSession.id, cameraEnabled });
+    logger$7.info('WebRTC phone - on call accepted', { sipId: sipSession.id, cameraEnabled });
 
     const callSession = this._createAcceptedCallSession(sipSession, cameraEnabled);
     this.currentSipSession = sipSession;
@@ -9347,6 +9385,7 @@ class WebRTCPhone extends Emitter                  {
   _onCallTerminated(sipSession         ) {
     logger$7.info('WebRTC phone - on call terminated', { sipId: sipSession.id });
 
+    const sipSessionId = this.getSipSessionId(sipSession);
     const callSession = this._createCallSession(sipSession);
     const isCurrentSession = this.isCurrentCallSipSession(callSession);
     const isCurrentIncomingCall = callSession.is(this.getIncomingCallSession());
@@ -9364,7 +9403,6 @@ class WebRTCPhone extends Emitter                  {
       }, 5);
     }
 
-    const sipSessionId = this.getSipSessionId(sipSession);
     if (sipSessionId) {
       this.removeIncomingSessions(sipSessionId);
     }
@@ -9790,7 +9828,7 @@ 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$7.info('make WebRTC call', { number, lineId: line ? line.id : null, cameraEnabled, audioOnly, conference });
     if (!number) {
       return Promise.resolve(null);
     }
@@ -10009,8 +10047,12 @@ class WebRTCPhone extends Emitter                  {
     }
 
     const sessionId = this.incomingSessions[0];
+    const sipSession = this.client.getSipSession(sessionId);
+    if (!sipSession) {
+      return null;
+    }
 
-    return this._createCallSession(this.client.getSipSession(sessionId));
+    return this._createCallSession(sipSession);
   }
 
   sendMessage(sipSession          = null, body        , contentType         = 'text/plain') {
@@ -10151,9 +10193,13 @@ class WebRTCPhone extends Emitter                  {
       if (this.shouldSendReinvite && this.currentSipSession
         && this.currentSipSession.state === sessionState.SessionState.Established) {
         this.shouldSendReinvite = false;
+        const pc = this.currentSipSession.sessionDescriptionHandler.peerConnection;
+        const isConference = pc ? pc.sfu : false;
+        const hasVideo = this.currentCallSession && this.currentCallSession.cameraEnabled;
+
         try {
           // Send reinvite with iceRestart
-          this.sendReinvite(this.currentCallSession, null, false, false, true);
+          this.sendReinvite(this.currentCallSession, null, isConference, !hasVideo, true);
         } catch (e) {
           logger$7.error('WebRTC reinvite after register, error', { message: e.message, stack: e.stack });
         }
@@ -10237,6 +10283,15 @@ class WebRTCPhone extends Emitter                  {
     return this.client.getSipSessionId(sipSession);
   }
 
+  _updateCallSession(callSession             ) {
+    const sipSession = this.findSipSession(callSession);
+    const sipSessionId = sipSession ? this.getSipSessionId(sipSession) : null;
+
+    if (sipSessionId) {
+      this.callSessions[sipSessionId] = callSession;
+    }
+  }
+
   _onMessage(message         ) {
     if (!message || message.method !== 'MESSAGE') {
       return;