@wazo/sdk 0.37.38 → 0.37.41

package/lib/simple/Auth.js

@@ -2270,12 +2270,14 @@ var authMethods = (client              , baseUrl        ) => ({
     backend        ,
     expiration        ,
     isMobile          = false,
+    tenantId         ,
   )                    => {
     const body         = {
       backend: backend || DEFAULT_BACKEND_USER,
       expiration: expiration || DETAULT_EXPIRATION,
       refresh_token: refreshToken,
       client_id: client.clientId,
+      tenant_id: tenantId,
     };
 
     const headers         = {
@@ -5427,6 +5429,7 @@ class ApiClient {
                             
                              
                           
+                           
                     
                        
 
@@ -5467,6 +5470,7 @@ class ApiClient {
     logger$a.info('refresh token callback called', {
       refreshToken: this.refreshToken,
       refreshBackend: this.refreshBackend,
+      refreshTenantId: this.refreshTenantId,
       refreshExpiration: this.refreshExpiration,
       isMobile: this.isMobile,
     });
@@ -5480,6 +5484,7 @@ class ApiClient {
       this.refreshBackend,
       this.refreshExpiration,
       this.isMobile,
+      this.refreshTenantId,
     );
 
     logger$a.info('token refreshed', { token: session.token });
@@ -5521,6 +5526,10 @@ class ApiClient {
     this.refreshBackend = refreshBackend;
   }
 
+  setRefreshTenantId(tenantId         ) {
+    this.refreshTenantId = tenantId;
+  }
+
   setIsMobile(isMobile         ) {
     this.isMobile = isMobile;
   }
@@ -8456,11 +8465,14 @@ class WebRTCClient extends Emitter {
   }
 
   async _disconnectTransport(force          = false) {
-    if (force) {
+    if (force && this.userAgent && this.userAgent.transport) {
       // Bypass sip.js state machine that prevent to close WS with the state `Connecting`
       this.userAgent.transport.disconnectResolve = () => {};
-      // eslint-disable-next-line
-      this.userAgent.transport._ws.close(1000);
+
+      if (this.userAgent.transport._ws) {
+        this.userAgent.transport._ws.close(1000);
+      }
+
       return;
     }
 
@@ -9112,7 +9124,7 @@ class WebRTCPhone extends Emitter                  {
         // When stopping screenshare and we had video before that, we have to re-upgrade
         // 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;
+        const screenshareStopped = this.currentScreenShare.hadVideo;
         const constraints = { audio: false, video: screenshareStopped };
 
         // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -9556,7 +9568,7 @@ class WebRTCPhone extends Emitter                  {
 
     logger$8.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
 
-    // Stop screenshre if needed
+    // Stop screenshare if needed
     if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
       this.lastScreenShare = this.currentScreenShare;
       await this.stopScreenSharing(false, callSession);
@@ -10318,7 +10330,7 @@ class WebRTCPhone extends Emitter                  {
       endTime: fromSession ? fromSession.endTime : null,
       answered: state === sessionState.SessionState.Established,
       paused: this.client.isCallHeld(sipSession),
-      isCaller: 'incoming' in extra ? !extra.incoming : false,
+      isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
       cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
       number,
       ringing: false,
@@ -13776,6 +13788,13 @@ class Auth {
       expiration: this.expiration,
       mobile: this.mobile,
     });
+
+    if (backend) {
+      getApiClient().setRefreshBackend(backend);
+    }
+
+    getApiClient().setRefreshTenantId(tenantId);
+
     return this._onAuthenticated(rawSession);
   }
 

package/lib/simple/Configuration.js

@@ -2268,12 +2268,14 @@ var authMethods = (client              , baseUrl        ) => ({
     backend        ,
     expiration        ,
     isMobile          = false,
+    tenantId         ,
   )                    => {
     const body         = {
       backend: backend || DEFAULT_BACKEND_USER,
       expiration: expiration || DETAULT_EXPIRATION,
       refresh_token: refreshToken,
       client_id: client.clientId,
+      tenant_id: tenantId,
     };
 
     const headers         = {
@@ -5425,6 +5427,7 @@ class ApiClient {
                             
                              
                           
+                           
                     
                        
 
@@ -5465,6 +5468,7 @@ class ApiClient {
     logger$a.info('refresh token callback called', {
       refreshToken: this.refreshToken,
       refreshBackend: this.refreshBackend,
+      refreshTenantId: this.refreshTenantId,
       refreshExpiration: this.refreshExpiration,
       isMobile: this.isMobile,
     });
@@ -5478,6 +5482,7 @@ class ApiClient {
       this.refreshBackend,
       this.refreshExpiration,
       this.isMobile,
+      this.refreshTenantId,
     );
 
     logger$a.info('token refreshed', { token: session.token });
@@ -5519,6 +5524,10 @@ class ApiClient {
     this.refreshBackend = refreshBackend;
   }
 
+  setRefreshTenantId(tenantId         ) {
+    this.refreshTenantId = tenantId;
+  }
+
   setIsMobile(isMobile         ) {
     this.isMobile = isMobile;
   }
@@ -8454,11 +8463,14 @@ class WebRTCClient extends Emitter {
   }
 
   async _disconnectTransport(force          = false) {
-    if (force) {
+    if (force && this.userAgent && this.userAgent.transport) {
       // Bypass sip.js state machine that prevent to close WS with the state `Connecting`
       this.userAgent.transport.disconnectResolve = () => {};
-      // eslint-disable-next-line
-      this.userAgent.transport._ws.close(1000);
+
+      if (this.userAgent.transport._ws) {
+        this.userAgent.transport._ws.close(1000);
+      }
+
       return;
     }
 
@@ -9110,7 +9122,7 @@ class WebRTCPhone extends Emitter                  {
         // When stopping screenshare and we had video before that, we have to re-upgrade
         // 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;
+        const screenshareStopped = this.currentScreenShare.hadVideo;
         const constraints = { audio: false, video: screenshareStopped };
 
         // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -9554,7 +9566,7 @@ class WebRTCPhone extends Emitter                  {
 
     logger$8.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
 
-    // Stop screenshre if needed
+    // Stop screenshare if needed
     if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
       this.lastScreenShare = this.currentScreenShare;
       await this.stopScreenSharing(false, callSession);
@@ -10316,7 +10328,7 @@ class WebRTCPhone extends Emitter                  {
       endTime: fromSession ? fromSession.endTime : null,
       answered: state === sessionState.SessionState.Established,
       paused: this.client.isCallHeld(sipSession),
-      isCaller: 'incoming' in extra ? !extra.incoming : false,
+      isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
       cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
       number,
       ringing: false,
@@ -13757,6 +13769,13 @@ class Auth {
       expiration: this.expiration,
       mobile: this.mobile,
     });
+
+    if (backend) {
+      getApiClient().setRefreshBackend(backend);
+    }
+
+    getApiClient().setRefreshTenantId(tenantId);
+
     return this._onAuthenticated(rawSession);
   }
 

package/lib/simple/Directory.js

@@ -2244,12 +2244,14 @@ var authMethods = (client              , baseUrl        ) => ({
     backend        ,
     expiration        ,
     isMobile          = false,
+    tenantId         ,
   )                    => {
     const body         = {
       backend: backend || DEFAULT_BACKEND_USER,
       expiration: expiration || DETAULT_EXPIRATION,
       refresh_token: refreshToken,
       client_id: client.clientId,
+      tenant_id: tenantId,
     };
 
     const headers         = {
@@ -4572,6 +4574,7 @@ class ApiClient {
                             
                              
                           
+                           
                     
                        
 
@@ -4612,6 +4615,7 @@ class ApiClient {
     logger.info('refresh token callback called', {
       refreshToken: this.refreshToken,
       refreshBackend: this.refreshBackend,
+      refreshTenantId: this.refreshTenantId,
       refreshExpiration: this.refreshExpiration,
       isMobile: this.isMobile,
     });
@@ -4625,6 +4629,7 @@ class ApiClient {
       this.refreshBackend,
       this.refreshExpiration,
       this.isMobile,
+      this.refreshTenantId,
     );
 
     logger.info('token refreshed', { token: session.token });
@@ -4666,6 +4671,10 @@ class ApiClient {
     this.refreshBackend = refreshBackend;
   }
 
+  setRefreshTenantId(tenantId         ) {
+    this.refreshTenantId = tenantId;
+  }
+
   setIsMobile(isMobile         ) {
     this.isMobile = isMobile;
   }

package/lib/simple/Phone.js

@@ -2871,12 +2871,14 @@ var authMethods = (client              , baseUrl        ) => ({
     backend        ,
     expiration        ,
     isMobile          = false,
+    tenantId         ,
   )                    => {
     const body         = {
       backend: backend || DEFAULT_BACKEND_USER,
       expiration: expiration || DETAULT_EXPIRATION,
       refresh_token: refreshToken,
       client_id: client.clientId,
+      tenant_id: tenantId,
     };
 
     const headers         = {
@@ -5425,6 +5427,7 @@ class ApiClient {
                             
                              
                           
+                           
                     
                        
 
@@ -5465,6 +5468,7 @@ class ApiClient {
     logger$a.info('refresh token callback called', {
       refreshToken: this.refreshToken,
       refreshBackend: this.refreshBackend,
+      refreshTenantId: this.refreshTenantId,
       refreshExpiration: this.refreshExpiration,
       isMobile: this.isMobile,
     });
@@ -5478,6 +5482,7 @@ class ApiClient {
       this.refreshBackend,
       this.refreshExpiration,
       this.isMobile,
+      this.refreshTenantId,
     );
 
     logger$a.info('token refreshed', { token: session.token });
@@ -5519,6 +5524,10 @@ class ApiClient {
     this.refreshBackend = refreshBackend;
   }
 
+  setRefreshTenantId(tenantId         ) {
+    this.refreshTenantId = tenantId;
+  }
+
   setIsMobile(isMobile         ) {
     this.isMobile = isMobile;
   }
@@ -8454,11 +8463,14 @@ class WebRTCClient extends Emitter {
   }
 
   async _disconnectTransport(force          = false) {
-    if (force) {
+    if (force && this.userAgent && this.userAgent.transport) {
       // Bypass sip.js state machine that prevent to close WS with the state `Connecting`
       this.userAgent.transport.disconnectResolve = () => {};
-      // eslint-disable-next-line
-      this.userAgent.transport._ws.close(1000);
+
+      if (this.userAgent.transport._ws) {
+        this.userAgent.transport._ws.close(1000);
+      }
+
       return;
     }
 
@@ -9110,7 +9122,7 @@ class WebRTCPhone extends Emitter                  {
         // When stopping screenshare and we had video before that, we have to re-upgrade
         // 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;
+        const screenshareStopped = this.currentScreenShare.hadVideo;
         const constraints = { audio: false, video: screenshareStopped };
 
         // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -9554,7 +9566,7 @@ class WebRTCPhone extends Emitter                  {
 
     logger$8.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
 
-    // Stop screenshre if needed
+    // Stop screenshare if needed
     if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
       this.lastScreenShare = this.currentScreenShare;
       await this.stopScreenSharing(false, callSession);
@@ -10316,7 +10328,7 @@ class WebRTCPhone extends Emitter                  {
       endTime: fromSession ? fromSession.endTime : null,
       answered: state === sessionState.SessionState.Established,
       paused: this.client.isCallHeld(sipSession),
-      isCaller: 'incoming' in extra ? !extra.incoming : false,
+      isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
       cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
       number,
       ringing: false,
@@ -10458,6 +10470,13 @@ class Auth {
       expiration: this.expiration,
       mobile: this.mobile,
     });
+
+    if (backend) {
+      getApiClient().setRefreshBackend(backend);
+    }
+
+    getApiClient().setRefreshTenantId(tenantId);
+
     return this._onAuthenticated(rawSession);
   }
 

package/lib/simple/index.js

@@ -2268,12 +2268,14 @@ var authMethods = (client              , baseUrl        ) => ({
     backend        ,
     expiration        ,
     isMobile          = false,
+    tenantId         ,
   )                    => {
     const body         = {
       backend: backend || DEFAULT_BACKEND_USER,
       expiration: expiration || DETAULT_EXPIRATION,
       refresh_token: refreshToken,
       client_id: client.clientId,
+      tenant_id: tenantId,
     };
 
     const headers         = {
@@ -5425,6 +5427,7 @@ class ApiClient {
                             
                              
                           
+                           
                     
                        
 
@@ -5465,6 +5468,7 @@ class ApiClient {
     logger$a.info('refresh token callback called', {
       refreshToken: this.refreshToken,
       refreshBackend: this.refreshBackend,
+      refreshTenantId: this.refreshTenantId,
       refreshExpiration: this.refreshExpiration,
       isMobile: this.isMobile,
     });
@@ -5478,6 +5482,7 @@ class ApiClient {
       this.refreshBackend,
       this.refreshExpiration,
       this.isMobile,
+      this.refreshTenantId,
     );
 
     logger$a.info('token refreshed', { token: session.token });
@@ -5519,6 +5524,10 @@ class ApiClient {
     this.refreshBackend = refreshBackend;
   }
 
+  setRefreshTenantId(tenantId         ) {
+    this.refreshTenantId = tenantId;
+  }
+
   setIsMobile(isMobile         ) {
     this.isMobile = isMobile;
   }
@@ -5690,6 +5699,13 @@ class Auth {
       expiration: this.expiration,
       mobile: this.mobile,
     });
+
+    if (backend) {
+      getApiClient().setRefreshBackend(backend);
+    }
+
+    getApiClient().setRefreshTenantId(tenantId);
+
     return this._onAuthenticated(rawSession);
   }
 
@@ -8729,11 +8745,14 @@ class WebRTCClient extends Emitter {
   }
 
   async _disconnectTransport(force          = false) {
-    if (force) {
+    if (force && this.userAgent && this.userAgent.transport) {
       // Bypass sip.js state machine that prevent to close WS with the state `Connecting`
       this.userAgent.transport.disconnectResolve = () => {};
-      // eslint-disable-next-line
-      this.userAgent.transport._ws.close(1000);
+
+      if (this.userAgent.transport._ws) {
+        this.userAgent.transport._ws.close(1000);
+      }
+
       return;
     }
 
@@ -9385,7 +9404,7 @@ class WebRTCPhone extends Emitter                  {
         // When stopping screenshare and we had video before that, we have to re-upgrade
         // 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;
+        const screenshareStopped = this.currentScreenShare.hadVideo;
         const constraints = { audio: false, video: screenshareStopped };
 
         // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -9829,7 +9848,7 @@ class WebRTCPhone extends Emitter                  {
 
     logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
 
-    // Stop screenshre if needed
+    // Stop screenshare if needed
     if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
       this.lastScreenShare = this.currentScreenShare;
       await this.stopScreenSharing(false, callSession);
@@ -10591,7 +10610,7 @@ class WebRTCPhone extends Emitter                  {
       endTime: fromSession ? fromSession.endTime : null,
       answered: state === sessionState.SessionState.Established,
       paused: this.client.isCallHeld(sipSession),
-      isCaller: 'incoming' in extra ? !extra.incoming : false,
+      isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
       cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
       number,
       ringing: false,

package/lib/simple/room/LocalParticipant.js

@@ -2294,12 +2294,14 @@ var authMethods = (client              , baseUrl        ) => ({
     backend        ,
     expiration        ,
     isMobile          = false,
+    tenantId         ,
   )                    => {
     const body         = {
       backend: backend || DEFAULT_BACKEND_USER,
       expiration: expiration || DETAULT_EXPIRATION,
       refresh_token: refreshToken,
       client_id: client.clientId,
+      tenant_id: tenantId,
     };
 
     const headers         = {
@@ -5425,6 +5427,7 @@ class ApiClient {
                             
                              
                           
+                           
                     
                        
 
@@ -5465,6 +5468,7 @@ class ApiClient {
     logger$a.info('refresh token callback called', {
       refreshToken: this.refreshToken,
       refreshBackend: this.refreshBackend,
+      refreshTenantId: this.refreshTenantId,
       refreshExpiration: this.refreshExpiration,
       isMobile: this.isMobile,
     });
@@ -5478,6 +5482,7 @@ class ApiClient {
       this.refreshBackend,
       this.refreshExpiration,
       this.isMobile,
+      this.refreshTenantId,
     );
 
     logger$a.info('token refreshed', { token: session.token });
@@ -5519,6 +5524,10 @@ class ApiClient {
     this.refreshBackend = refreshBackend;
   }
 
+  setRefreshTenantId(tenantId         ) {
+    this.refreshTenantId = tenantId;
+  }
+
   setIsMobile(isMobile         ) {
     this.isMobile = isMobile;
   }
@@ -5690,6 +5699,13 @@ class Auth {
       expiration: this.expiration,
       mobile: this.mobile,
     });
+
+    if (backend) {
+      getApiClient().setRefreshBackend(backend);
+    }
+
+    getApiClient().setRefreshTenantId(tenantId);
+
     return this._onAuthenticated(rawSession);
   }
 
@@ -8729,11 +8745,14 @@ class WebRTCClient extends Emitter {
   }
 
   async _disconnectTransport(force          = false) {
-    if (force) {
+    if (force && this.userAgent && this.userAgent.transport) {
       // Bypass sip.js state machine that prevent to close WS with the state `Connecting`
       this.userAgent.transport.disconnectResolve = () => {};
-      // eslint-disable-next-line
-      this.userAgent.transport._ws.close(1000);
+
+      if (this.userAgent.transport._ws) {
+        this.userAgent.transport._ws.close(1000);
+      }
+
       return;
     }
 
@@ -9385,7 +9404,7 @@ class WebRTCPhone extends Emitter                  {
         // When stopping screenshare and we had video before that, we have to re-upgrade
         // 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;
+        const screenshareStopped = this.currentScreenShare.hadVideo;
         const constraints = { audio: false, video: screenshareStopped };
 
         // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -9829,7 +9848,7 @@ class WebRTCPhone extends Emitter                  {
 
     logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
 
-    // Stop screenshre if needed
+    // Stop screenshare if needed
     if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
       this.lastScreenShare = this.currentScreenShare;
       await this.stopScreenSharing(false, callSession);
@@ -10591,7 +10610,7 @@ class WebRTCPhone extends Emitter                  {
       endTime: fromSession ? fromSession.endTime : null,
       answered: state === sessionState.SessionState.Established,
       paused: this.client.isCallHeld(sipSession),
-      isCaller: 'incoming' in extra ? !extra.incoming : false,
+      isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
       cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
       number,
       ringing: false,

package/lib/simple/room/Participant.js

@@ -2294,12 +2294,14 @@ var authMethods = (client              , baseUrl        ) => ({
     backend        ,
     expiration        ,
     isMobile          = false,
+    tenantId         ,
   )                    => {
     const body         = {
       backend: backend || DEFAULT_BACKEND_USER,
       expiration: expiration || DETAULT_EXPIRATION,
       refresh_token: refreshToken,
       client_id: client.clientId,
+      tenant_id: tenantId,
     };
 
     const headers         = {
@@ -5425,6 +5427,7 @@ class ApiClient {
                             
                              
                           
+                           
                     
                        
 
@@ -5465,6 +5468,7 @@ class ApiClient {
     logger$a.info('refresh token callback called', {
       refreshToken: this.refreshToken,
       refreshBackend: this.refreshBackend,
+      refreshTenantId: this.refreshTenantId,
       refreshExpiration: this.refreshExpiration,
       isMobile: this.isMobile,
     });
@@ -5478,6 +5482,7 @@ class ApiClient {
       this.refreshBackend,
       this.refreshExpiration,
       this.isMobile,
+      this.refreshTenantId,
     );
 
     logger$a.info('token refreshed', { token: session.token });
@@ -5519,6 +5524,10 @@ class ApiClient {
     this.refreshBackend = refreshBackend;
   }
 
+  setRefreshTenantId(tenantId         ) {
+    this.refreshTenantId = tenantId;
+  }
+
   setIsMobile(isMobile         ) {
     this.isMobile = isMobile;
   }
@@ -5690,6 +5699,13 @@ class Auth {
       expiration: this.expiration,
       mobile: this.mobile,
     });
+
+    if (backend) {
+      getApiClient().setRefreshBackend(backend);
+    }
+
+    getApiClient().setRefreshTenantId(tenantId);
+
     return this._onAuthenticated(rawSession);
   }
 
@@ -8729,11 +8745,14 @@ class WebRTCClient extends Emitter {
   }
 
   async _disconnectTransport(force          = false) {
-    if (force) {
+    if (force && this.userAgent && this.userAgent.transport) {
       // Bypass sip.js state machine that prevent to close WS with the state `Connecting`
       this.userAgent.transport.disconnectResolve = () => {};
-      // eslint-disable-next-line
-      this.userAgent.transport._ws.close(1000);
+
+      if (this.userAgent.transport._ws) {
+        this.userAgent.transport._ws.close(1000);
+      }
+
       return;
     }
 
@@ -9385,7 +9404,7 @@ class WebRTCPhone extends Emitter                  {
         // When stopping screenshare and we had video before that, we have to re-upgrade
         // 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;
+        const screenshareStopped = this.currentScreenShare.hadVideo;
         const constraints = { audio: false, video: screenshareStopped };
 
         // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -9829,7 +9848,7 @@ class WebRTCPhone extends Emitter                  {
 
     logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
 
-    // Stop screenshre if needed
+    // Stop screenshare if needed
     if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
       this.lastScreenShare = this.currentScreenShare;
       await this.stopScreenSharing(false, callSession);
@@ -10591,7 +10610,7 @@ class WebRTCPhone extends Emitter                  {
       endTime: fromSession ? fromSession.endTime : null,
       answered: state === sessionState.SessionState.Established,
       paused: this.client.isCallHeld(sipSession),
-      isCaller: 'incoming' in extra ? !extra.incoming : false,
+      isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
       cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
       number,
       ringing: false,