@wazo/sdk 0.37.27 → 0.37.31

package/esm/simple/room/__tests__/RemoteParticipant.test.js

@@ -3248,13 +3248,16 @@ var updateFrom = (instance     , from     ) => {
 
                              
                     
+                     
                             
               
                  
                        
                          
+                       
                                   
                       
+                  
                     
                  
                       
@@ -3271,6 +3274,7 @@ var updateFrom = (instance     , from     ) => {
                          
                      
                            
+                       
   
 
 class CallSession {
@@ -3290,10 +3294,14 @@ class CallSession {
 
                        
 
-                    
+                      
+
+                     // = creationTime
 
                     
 
+                 
+
                     
 
                             
@@ -3329,6 +3337,8 @@ class CallSession {
                      
 
                            
+
+                      
 
   constructor({
     answered,
@@ -3342,6 +3352,8 @@ class CallSession {
     paused,
     ringing,
     startTime,
+    creationTime,
+    endTime,
     cameraEnabled,
     dialedExtension,
     sipCallId,
@@ -3354,12 +3366,16 @@ class CallSession {
     recording,
     recordingPaused,
     videoRemotelyDowngraded,
+    sipSession,
+    answerTime,
   }                      ) {
     this.callId = callId;
     this.sipCallId = sipCallId;
     this.displayName = displayName;
     this.number = number;
+    this.creationTime = creationTime;
     this.startTime = startTime;
+    this.endTime = endTime;
     this.isCaller = isCaller;
     this.answered = answered;
     this.answeredBySystem = answeredBySystem;
@@ -3378,6 +3394,8 @@ class CallSession {
     this.recording = recording || false;
     this.recordingPaused = recordingPaused || false;
     this.videoRemotelyDowngraded = videoRemotelyDowngraded;
+    this.sipSession = sipSession;
+    this.answerTime = answerTime || this.answerTime;
 
     // Useful to compare instead of instanceof with minified code
     this.type = 'CallSession';
@@ -3408,7 +3426,7 @@ class CallSession {
   }
 
   answer() {
-    this.answered = true;
+    this.answerTime = new Date();
   }
 
   systemAnswer() {
@@ -3564,6 +3582,15 @@ class CallSession {
     return newFrom(callSession, CallSession);
   }
 
+  // Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
+  set answered(value         ) {
+    this.answerTime = value ? this.answerTime || new Date() : null;
+  }
+
+  get answered()          {
+    return !!this.answerTime;
+  }
+
   static parseCall(call      )              {
     return new CallSession({
       callId: call.id,
@@ -6610,6 +6637,7 @@ const MAX_REGISTER_TRIES = 10;
                       
                
                 
+                         
                              
                     
               
@@ -8226,6 +8254,13 @@ class WebRTCClient extends Emitter {
   }
 
   _createWebRTCConfiguration(configOverrides         = {}) {
+    let { host } = this.config;
+    let port = this.config.port || 443;
+
+    if (this.config.websocketSip) {
+      [host, port = 443] = this.config.websocketSip.split(':');
+    }
+
     const config         = {
       authorizationUsername: this.config.authorizationUser,
       authorizationPassword: this.config.password,
@@ -8261,7 +8296,7 @@ class WebRTCClient extends Emitter {
       },
       transportOptions: {
         traceSip: configOverrides.traceSip || false,
-        wsServers: `wss://${this.config.host}:${this.config.port || 443}/api/asterisk/ws`,
+        wsServers: `wss://${host}:${port}/api/asterisk/ws`,
       },
       sessionDescriptionHandlerFactoryOptions: {
         modifiers: [replaceLocalIpModifier],
@@ -8723,6 +8758,7 @@ const ON_CAMERA_DISABLED = 'onCameraDisabled';
 const ON_CAMERA_RESUMED = 'onCameraResumed';
 const ON_CALL_CANCELED = 'onCallCanceled';
 const ON_CALL_FAILED = 'onCallFailed';
+const ON_CALL_REJECTED = 'onCallRejected';
 const ON_CALL_ENDED = 'onCallEnded';
 const ON_CALL_ENDING = 'onCallEnding';
 const ON_MESSAGE = 'onMessage';
@@ -8767,6 +8803,7 @@ const events = [
   ON_CAMERA_DISABLED,
   ON_CALL_FAILED,
   ON_CALL_ENDED,
+  ON_CALL_REJECTED,
   ON_MESSAGE,
   ON_REINVITE,
   ON_TRACK,
@@ -9123,6 +9160,7 @@ class WebRTCPhone extends Emitter                  {
 
           // Should be called before `_onCallTerminated` or the callCount will not decrement...
           const callSession = this._createCallSession(sipSession);
+          callSession.endTime = new Date();
           const wasCurrentSession = this._onCallTerminated(sipSession);
 
           return this.eventEmitter.emit(ON_CALL_ENDED, callSession, wasCurrentSession);
@@ -9318,6 +9356,8 @@ class WebRTCPhone extends Emitter                  {
       this.removeIncomingSessions(sipSessionId);
     }
 
+    callSession.answerTime = new Date();
+    this._updateCallSession(callSession);
     this.eventEmitter.emit(ON_CALL_ACCEPTED, callSession, cameraEnabled);
 
     return callSession;
@@ -9533,6 +9573,8 @@ class WebRTCPhone extends Emitter                  {
     }
 
     this.shouldSendReinvite = false;
+    callSession.answerTime = new Date();
+    this._updateCallSession(callSession);
     this.acceptedSessions[callSession.getId()] = true;
 
     this.eventEmitter.emit(ON_CALL_ANSWERED, callSession);
@@ -9632,7 +9674,7 @@ class WebRTCPhone extends Emitter                  {
     // 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 });
-      this.holdSipSession(this.currentSipSession, callSession);
+      this.holdSipSession(this.currentSipSession, this.currentCallSession);
     }
 
     const promise = this.unholdSipSession(sipSession, callSession, withEvent, isConference);
@@ -9842,6 +9884,8 @@ class WebRTCPhone extends Emitter                  {
       return Promise.resolve(null);
     }
     const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
+    callSession.creationTime = new Date();
+    this._updateCallSession(callSession);
 
     this.eventEmitter.emit(ON_PLAY_PROGRESS_SOUND, this.audioOutputDeviceId, this.audioOutputVolume);
 
@@ -10120,6 +10164,9 @@ class WebRTCPhone extends Emitter                  {
           callSession);
       }
 
+      callSession.creationTime = new Date();
+      this._updateCallSession(callSession);
+
       this.eventEmitter.emit(ON_CALL_INCOMING, callSession, wantsToDoVideo);
     });
 
@@ -10159,6 +10206,15 @@ class WebRTCPhone extends Emitter                  {
       this.eventEmitter.emit(ON_CALL_ERROR, e);
     });
 
+    this.client.on(this.client.REJECTED, session => {
+      logger$7.info('WebRTC call rejected', session.id);
+      const callSession = this._createCallSession(session);
+      callSession.endTime = new Date();
+      this._updateCallSession(callSession);
+
+      this.eventEmitter.emit(ON_CALL_REJECTED, callSession);
+    });
+
     this.client.on(this.client.UNREGISTERED, () => {
       logger$7.info('WebRTC unregistered');
       this.eventEmitter.emit(ON_UNREGISTERED);
@@ -10371,6 +10427,7 @@ class WebRTCPhone extends Emitter                  {
     const number = identity ? identity.uri._normal.user : null;
     const { state } = sipSession || {};
     const sessionId = this.getSipSessionId(sipSession);
+    fromSession = fromSession || this.callSessions[sessionId];
 
     const callSession = new CallSession({
       callId: fromSession && fromSession.callId,
@@ -10378,6 +10435,9 @@ class WebRTCPhone extends Emitter                  {
       sipStatus: state,
       displayName: identity ? identity.displayName || number : number,
       startTime: fromSession ? fromSession.startTime : new Date(),
+      creationTime: fromSession ? fromSession.creationTime : null,
+      answerTime: fromSession ? fromSession.answerTime : null,
+      endTime: fromSession ? fromSession.endTime : null,
       answered: state === SessionState.Established,
       paused: this.client.isCallHeld(sipSession),
       isCaller: 'incoming' in extra ? !extra.incoming : false,
@@ -10388,6 +10448,7 @@ class WebRTCPhone extends Emitter                  {
       videoMuted: fromSession ? fromSession.isVideoMuted() : false,
       recording: fromSession ? fromSession.isRecording() : false,
       recordingPaused: false, // @TODO
+      sipSession,
       ...extra,
     });
 
@@ -10416,6 +10477,7 @@ var PHONE_EVENTS = /*#__PURE__*/Object.freeze({
   ON_CAMERA_RESUMED: ON_CAMERA_RESUMED,
   ON_CALL_CANCELED: ON_CALL_CANCELED,
   ON_CALL_FAILED: ON_CALL_FAILED,
+  ON_CALL_REJECTED: ON_CALL_REJECTED,
   ON_CALL_ENDED: ON_CALL_ENDED,
   ON_CALL_ENDING: ON_CALL_ENDING,
   ON_MESSAGE: ON_MESSAGE,
@@ -10445,6 +10507,14 @@ var PHONE_EVENTS = /*#__PURE__*/Object.freeze({
   'default': WebRTCPhone
 });
 
+/*       */
+class SFUNotAvailableError extends Error {
+  constructor(message         = 'Sorry your user is not configured to support video conference') {
+    super(message);
+    this.name = 'SFUNotAvailableError';
+  }
+}
+
 //      
 
 const logger$6 = IssueReporter$1.loggerFor('simple-phone');
@@ -10829,7 +10899,7 @@ class Phone extends Emitter {
 
   checkSfu() {
     if (!this.hasSfu()) {
-      throw new Error('Sorry your user is not configured to support video conference');
+      throw new SFUNotAvailableError();
     }
   }
 
@@ -13742,6 +13812,7 @@ const Wazo = {
   InvalidSubscription,
   InvalidAuthorization,
   CanceledCallError,
+  SFUNotAvailableError,
 };
 
 if (window) {

package/esm/simple/utils.js

@@ -3248,13 +3248,16 @@ var updateFrom = (instance     , from     ) => {
 
                              
                     
+                     
                             
               
                  
                        
                          
+                       
                                   
                       
+                  
                     
                  
                       
@@ -3271,6 +3274,7 @@ var updateFrom = (instance     , from     ) => {
                          
                      
                            
+                       
   
 
 class CallSession {
@@ -3290,10 +3294,14 @@ class CallSession {
 
                        
 
-                    
+                      
+
+                     // = creationTime
 
                     
 
+                 
+
                     
 
                             
@@ -3329,6 +3337,8 @@ class CallSession {
                      
 
                            
+
+                      
 
   constructor({
     answered,
@@ -3342,6 +3352,8 @@ class CallSession {
     paused,
     ringing,
     startTime,
+    creationTime,
+    endTime,
     cameraEnabled,
     dialedExtension,
     sipCallId,
@@ -3354,12 +3366,16 @@ class CallSession {
     recording,
     recordingPaused,
     videoRemotelyDowngraded,
+    sipSession,
+    answerTime,
   }                      ) {
     this.callId = callId;
     this.sipCallId = sipCallId;
     this.displayName = displayName;
     this.number = number;
+    this.creationTime = creationTime;
     this.startTime = startTime;
+    this.endTime = endTime;
     this.isCaller = isCaller;
     this.answered = answered;
     this.answeredBySystem = answeredBySystem;
@@ -3378,6 +3394,8 @@ class CallSession {
     this.recording = recording || false;
     this.recordingPaused = recordingPaused || false;
     this.videoRemotelyDowngraded = videoRemotelyDowngraded;
+    this.sipSession = sipSession;
+    this.answerTime = answerTime || this.answerTime;
 
     // Useful to compare instead of instanceof with minified code
     this.type = 'CallSession';
@@ -3408,7 +3426,7 @@ class CallSession {
   }
 
   answer() {
-    this.answered = true;
+    this.answerTime = new Date();
   }
 
   systemAnswer() {
@@ -3564,6 +3582,15 @@ class CallSession {
     return newFrom(callSession, CallSession);
   }
 
+  // Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
+  set answered(value         ) {
+    this.answerTime = value ? this.answerTime || new Date() : null;
+  }
+
+  get answered()          {
+    return !!this.answerTime;
+  }
+
   static parseCall(call      )              {
     return new CallSession({
       callId: call.id,
@@ -6610,6 +6637,7 @@ const MAX_REGISTER_TRIES = 10;
                       
                
                 
+                         
                              
                     
               
@@ -8226,6 +8254,13 @@ class WebRTCClient extends Emitter {
   }
 
   _createWebRTCConfiguration(configOverrides         = {}) {
+    let { host } = this.config;
+    let port = this.config.port || 443;
+
+    if (this.config.websocketSip) {
+      [host, port = 443] = this.config.websocketSip.split(':');
+    }
+
     const config         = {
       authorizationUsername: this.config.authorizationUser,
       authorizationPassword: this.config.password,
@@ -8261,7 +8296,7 @@ class WebRTCClient extends Emitter {
       },
       transportOptions: {
         traceSip: configOverrides.traceSip || false,
-        wsServers: `wss://${this.config.host}:${this.config.port || 443}/api/asterisk/ws`,
+        wsServers: `wss://${host}:${port}/api/asterisk/ws`,
       },
       sessionDescriptionHandlerFactoryOptions: {
         modifiers: [replaceLocalIpModifier],
@@ -8723,6 +8758,7 @@ const ON_CAMERA_DISABLED = 'onCameraDisabled';
 const ON_CAMERA_RESUMED = 'onCameraResumed';
 const ON_CALL_CANCELED = 'onCallCanceled';
 const ON_CALL_FAILED = 'onCallFailed';
+const ON_CALL_REJECTED = 'onCallRejected';
 const ON_CALL_ENDED = 'onCallEnded';
 const ON_CALL_ENDING = 'onCallEnding';
 const ON_MESSAGE = 'onMessage';
@@ -8767,6 +8803,7 @@ const events = [
   ON_CAMERA_DISABLED,
   ON_CALL_FAILED,
   ON_CALL_ENDED,
+  ON_CALL_REJECTED,
   ON_MESSAGE,
   ON_REINVITE,
   ON_TRACK,
@@ -9123,6 +9160,7 @@ class WebRTCPhone extends Emitter                  {
 
           // Should be called before `_onCallTerminated` or the callCount will not decrement...
           const callSession = this._createCallSession(sipSession);
+          callSession.endTime = new Date();
           const wasCurrentSession = this._onCallTerminated(sipSession);
 
           return this.eventEmitter.emit(ON_CALL_ENDED, callSession, wasCurrentSession);
@@ -9318,6 +9356,8 @@ class WebRTCPhone extends Emitter                  {
       this.removeIncomingSessions(sipSessionId);
     }
 
+    callSession.answerTime = new Date();
+    this._updateCallSession(callSession);
     this.eventEmitter.emit(ON_CALL_ACCEPTED, callSession, cameraEnabled);
 
     return callSession;
@@ -9533,6 +9573,8 @@ class WebRTCPhone extends Emitter                  {
     }
 
     this.shouldSendReinvite = false;
+    callSession.answerTime = new Date();
+    this._updateCallSession(callSession);
     this.acceptedSessions[callSession.getId()] = true;
 
     this.eventEmitter.emit(ON_CALL_ANSWERED, callSession);
@@ -9632,7 +9674,7 @@ class WebRTCPhone extends Emitter                  {
     // 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 });
-      this.holdSipSession(this.currentSipSession, callSession);
+      this.holdSipSession(this.currentSipSession, this.currentCallSession);
     }
 
     const promise = this.unholdSipSession(sipSession, callSession, withEvent, isConference);
@@ -9842,6 +9884,8 @@ class WebRTCPhone extends Emitter                  {
       return Promise.resolve(null);
     }
     const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
+    callSession.creationTime = new Date();
+    this._updateCallSession(callSession);
 
     this.eventEmitter.emit(ON_PLAY_PROGRESS_SOUND, this.audioOutputDeviceId, this.audioOutputVolume);
 
@@ -10120,6 +10164,9 @@ class WebRTCPhone extends Emitter                  {
           callSession);
       }
 
+      callSession.creationTime = new Date();
+      this._updateCallSession(callSession);
+
       this.eventEmitter.emit(ON_CALL_INCOMING, callSession, wantsToDoVideo);
     });
 
@@ -10159,6 +10206,15 @@ class WebRTCPhone extends Emitter                  {
       this.eventEmitter.emit(ON_CALL_ERROR, e);
     });
 
+    this.client.on(this.client.REJECTED, session => {
+      logger$7.info('WebRTC call rejected', session.id);
+      const callSession = this._createCallSession(session);
+      callSession.endTime = new Date();
+      this._updateCallSession(callSession);
+
+      this.eventEmitter.emit(ON_CALL_REJECTED, callSession);
+    });
+
     this.client.on(this.client.UNREGISTERED, () => {
       logger$7.info('WebRTC unregistered');
       this.eventEmitter.emit(ON_UNREGISTERED);
@@ -10371,6 +10427,7 @@ class WebRTCPhone extends Emitter                  {
     const number = identity ? identity.uri._normal.user : null;
     const { state } = sipSession || {};
     const sessionId = this.getSipSessionId(sipSession);
+    fromSession = fromSession || this.callSessions[sessionId];
 
     const callSession = new CallSession({
       callId: fromSession && fromSession.callId,
@@ -10378,6 +10435,9 @@ class WebRTCPhone extends Emitter                  {
       sipStatus: state,
       displayName: identity ? identity.displayName || number : number,
       startTime: fromSession ? fromSession.startTime : new Date(),
+      creationTime: fromSession ? fromSession.creationTime : null,
+      answerTime: fromSession ? fromSession.answerTime : null,
+      endTime: fromSession ? fromSession.endTime : null,
       answered: state === SessionState.Established,
       paused: this.client.isCallHeld(sipSession),
       isCaller: 'incoming' in extra ? !extra.incoming : false,
@@ -10388,6 +10448,7 @@ class WebRTCPhone extends Emitter                  {
       videoMuted: fromSession ? fromSession.isVideoMuted() : false,
       recording: fromSession ? fromSession.isRecording() : false,
       recordingPaused: false, // @TODO
+      sipSession,
       ...extra,
     });
 
@@ -10416,6 +10477,7 @@ var PHONE_EVENTS = /*#__PURE__*/Object.freeze({
   ON_CAMERA_RESUMED: ON_CAMERA_RESUMED,
   ON_CALL_CANCELED: ON_CALL_CANCELED,
   ON_CALL_FAILED: ON_CALL_FAILED,
+  ON_CALL_REJECTED: ON_CALL_REJECTED,
   ON_CALL_ENDED: ON_CALL_ENDED,
   ON_CALL_ENDING: ON_CALL_ENDING,
   ON_MESSAGE: ON_MESSAGE,
@@ -10445,6 +10507,14 @@ var PHONE_EVENTS = /*#__PURE__*/Object.freeze({
   'default': WebRTCPhone
 });
 
+/*       */
+class SFUNotAvailableError extends Error {
+  constructor(message         = 'Sorry your user is not configured to support video conference') {
+    super(message);
+    this.name = 'SFUNotAvailableError';
+  }
+}
+
 //      
 
 const logger$6 = IssueReporter$1.loggerFor('simple-phone');
@@ -10829,7 +10899,7 @@ class Phone extends Emitter {
 
   checkSfu() {
     if (!this.hasSfu()) {
-      throw new Error('Sorry your user is not configured to support video conference');
+      throw new SFUNotAvailableError();
     }
   }
 
@@ -13726,6 +13796,7 @@ const Wazo = {
   InvalidSubscription,
   InvalidAuthorization,
   CanceledCallError,
+  SFUNotAvailableError,
 };
 
 if (window) {

package/esm/web-rtc-client.js

@@ -5347,6 +5347,7 @@ const MAX_REGISTER_TRIES = 10;
                       
                
                 
+                         
                              
                     
               
@@ -6963,6 +6964,13 @@ class WebRTCClient extends Emitter {
   }
 
   _createWebRTCConfiguration(configOverrides         = {}) {
+    let { host } = this.config;
+    let port = this.config.port || 443;
+
+    if (this.config.websocketSip) {
+      [host, port = 443] = this.config.websocketSip.split(':');
+    }
+
     const config         = {
       authorizationUsername: this.config.authorizationUser,
       authorizationPassword: this.config.password,
@@ -6998,7 +7006,7 @@ class WebRTCClient extends Emitter {
       },
       transportOptions: {
         traceSip: configOverrides.traceSip || false,
-        wsServers: `wss://${this.config.host}:${this.config.port || 443}/api/asterisk/ws`,
+        wsServers: `wss://${host}:${port}/api/asterisk/ws`,
       },
       sessionDescriptionHandlerFactoryOptions: {
         modifiers: [replaceLocalIpModifier],

package/lib/__tests__/web-rtc-client.test.js

@@ -5353,6 +5353,7 @@ const MAX_REGISTER_TRIES = 10;
                       
                
                 
+                         
                              
                     
               
@@ -6969,6 +6970,13 @@ class WebRTCClient extends Emitter {
   }
 
   _createWebRTCConfiguration(configOverrides         = {}) {
+    let { host } = this.config;
+    let port = this.config.port || 443;
+
+    if (this.config.websocketSip) {
+      [host, port = 443] = this.config.websocketSip.split(':');
+    }
+
     const config         = {
       authorizationUsername: this.config.authorizationUser,
       authorizationPassword: this.config.password,
@@ -7004,7 +7012,7 @@ class WebRTCClient extends Emitter {
       },
       transportOptions: {
         traceSip: configOverrides.traceSip || false,
-        wsServers: `wss://${this.config.host}:${this.config.port || 443}/api/asterisk/ws`,
+        wsServers: `wss://${host}:${port}/api/asterisk/ws`,
       },
       sessionDescriptionHandlerFactoryOptions: {
         modifiers: [replaceLocalIpModifier],

package/lib/checker/Checker.js

@@ -5939,6 +5939,7 @@ const MAX_REGISTER_TRIES = 10;
                       
                
                 
+                         
                              
                     
               
@@ -7555,6 +7556,13 @@ class WebRTCClient extends Emitter {
   }
 
   _createWebRTCConfiguration(configOverrides         = {}) {
+    let { host } = this.config;
+    let port = this.config.port || 443;
+
+    if (this.config.websocketSip) {
+      [host, port = 443] = this.config.websocketSip.split(':');
+    }
+
     const config         = {
       authorizationUsername: this.config.authorizationUser,
       authorizationPassword: this.config.password,
@@ -7590,7 +7598,7 @@ class WebRTCClient extends Emitter {
       },
       transportOptions: {
         traceSip: configOverrides.traceSip || false,
-        wsServers: `wss://${this.config.host}:${this.config.port || 443}/api/asterisk/ws`,
+        wsServers: `wss://${host}:${port}/api/asterisk/ws`,
       },
       sessionDescriptionHandlerFactoryOptions: {
         modifiers: [replaceLocalIpModifier],

package/lib/checker/checks/index.js

@@ -5939,6 +5939,7 @@ const MAX_REGISTER_TRIES = 10;
                       
                
                 
+                         
                              
                     
               
@@ -7555,6 +7556,13 @@ class WebRTCClient extends Emitter {
   }
 
   _createWebRTCConfiguration(configOverrides         = {}) {
+    let { host } = this.config;
+    let port = this.config.port || 443;
+
+    if (this.config.websocketSip) {
+      [host, port = 443] = this.config.websocketSip.split(':');
+    }
+
     const config         = {
       authorizationUsername: this.config.authorizationUser,
       authorizationPassword: this.config.password,
@@ -7590,7 +7598,7 @@ class WebRTCClient extends Emitter {
       },
       transportOptions: {
         traceSip: configOverrides.traceSip || false,
-        wsServers: `wss://${this.config.host}:${this.config.port || 443}/api/asterisk/ws`,
+        wsServers: `wss://${host}:${port}/api/asterisk/ws`,
       },
       sessionDescriptionHandlerFactoryOptions: {
         modifiers: [replaceLocalIpModifier],