@wazo/sdk 0.37.29 → 0.37.33

package/lib/simple/room/SipRoom.js

@@ -3594,13 +3594,21 @@ class CallSession {
 
   // Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
   set answered(value         ) {
-    this.answerTime = value ? new Date() : null;
+    this.answerTime = value ? this.answerTime || new Date() : null;
   }
 
   get answered()          {
     return !!this.answerTime;
   }
 
+  toJSON() {
+    const jsonObj = { ...this };
+    // $FlowFixMe
+    jsonObj.answered = this.answered;
+
+    return jsonObj;
+  }
+
   static parseCall(call      )              {
     return new CallSession({
       callId: call.id,
@@ -4852,6 +4860,34 @@ var chatdMethods = (client              , baseUrl        ) => ({
     client.get(`${baseUrl}/users/me/rooms/messages`, options),
 });
 
+//      
+
+                                     
+               
+  
+
+class MeetingStatus {
+               
+
+                
+
+  constructor({
+    full,
+  }         = {}) {
+    this.full = full;
+
+    // Useful to compare instead of instanceof with minified code
+    this.type = 'MeetingStatus';
+  }
+
+  static parse(plain                       )                {
+    return new MeetingStatus({
+      full: plain.full,
+    });
+  }
+
+}
+
 /*       */
 
                   
@@ -5046,6 +5082,8 @@ var calldMethods = (client              , baseUrl        ) => ({
     client.put(`${baseUrl}/users/me/calls/${callId}/record/resume`, null, null,
       ApiRequester.successResponseParser),
 
+  guestGetMeetingStatus: (meetingUuid        )                         =>
+    client.get(`${baseUrl}/guests/me/meetings/${meetingUuid}/status`).then(MeetingStatus.parse),
 });
 
 //      
@@ -6647,6 +6685,7 @@ const MAX_REGISTER_TRIES = 10;
                       
                
                 
+                         
                              
                     
               
@@ -6688,6 +6727,7 @@ class WebRTCClient extends Emitter {
                         
                                                       
                                             
+                       
 
   // sugar
                         
@@ -6750,6 +6790,7 @@ class WebRTCClient extends Emitter {
     this.conferences = {};
     this.networkMonitoringInterval = {};
     this.sessionNetworkStats = {};
+    this.forceClosed = false;
 
     this._boundOnHeartbeat = this._onHeartbeat.bind(this);
     this.heartbeat = new Heartbeat(config.heartbeatDelay, config.heartbeatTimeout, config.maxHeartbeats);
@@ -6871,6 +6912,8 @@ class WebRTCClient extends Emitter {
       skipRegister: this.skipRegister,
     };
 
+    this.forceClosed = false;
+
     if (this.skipRegister) {
       logger$8.info('sdk webrtc skip register...', logInfo);
       return Promise.resolve();
@@ -6895,7 +6938,15 @@ class WebRTCClient extends Emitter {
     const registerOptions = this._isWeb() ? {} : { extraContactHeaderParams: ['mobility=mobile'] };
 
     const onRegisterFailed = () => {
-      logger$8.info('sdk webrtc registering failed', { tries, registerer: !!this.registerer });
+      logger$8.info('sdk webrtc registering failed', {
+        tries,
+        registerer: !!this.registerer,
+        forceClosed: this.forceClosed,
+      });
+      if (this.forceClosed) {
+        return;
+      }
+
       this.connectionPromise = null;
       if (this.registerer && this.registerer.waiting) {
         this.registerer.waitingToggle(false);
@@ -7173,8 +7224,9 @@ class WebRTCClient extends Emitter {
     }
   }
 
-  async close() {
-    logger$8.info('sdk webrtc closing client', { userAgent: !!this.userAgent });
+  async close(force          = false) {
+    logger$8.info('sdk webrtc closing client', { userAgent: !!this.userAgent, force });
+    this.forceClosed = force;
     this._cleanupMedia();
     this.connectionPromise = null;
 
@@ -7193,7 +7245,7 @@ class WebRTCClient extends Emitter {
     this.userAgent.delegate = null;
     this.userAgent.stateChange.removeAllListeners();
 
-    await this._disconnectTransport();
+    await this._disconnectTransport(force);
 
     this._cleanupRegister();
 
@@ -8263,6 +8315,13 @@ 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,
@@ -8298,7 +8357,7 @@ 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],
@@ -8622,7 +8681,15 @@ class WebRTCClient extends Emitter {
     return userAgent.UserAgent.makeURI(`sip:${target}@${this.config.host}`);
   }
 
-  async _disconnectTransport() {
+  async _disconnectTransport(force          = false) {
+    if (force) {
+      // 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);
+      return;
+    }
+
     // Check if `disconnectPromise` is not present to avoid `Disconnect promise must not be defined` errors.
     if (this.userAgent && this.userAgent.transport && !this.userAgent.transport.disconnectPromise) {
       try {
@@ -9359,6 +9426,7 @@ class WebRTCPhone extends Emitter                  {
     }
 
     callSession.answerTime = new Date();
+    this._updateCallSession(callSession);
     this.eventEmitter.emit(ON_CALL_ACCEPTED, callSession, cameraEnabled);
 
     return callSession;
@@ -9575,6 +9643,7 @@ 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);
@@ -9674,7 +9743,7 @@ 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);
@@ -9885,6 +9954,7 @@ class WebRTCPhone extends Emitter                  {
     }
     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);
 
@@ -10026,7 +10096,7 @@ class WebRTCPhone extends Emitter                  {
     return this.client && this.client.isConference(callSession.sipCallId);
   }
 
-  async close()                {
+  async close(force          = false)                {
     logger$7.info('WebRTC close');
 
     try {
@@ -10038,7 +10108,7 @@ class WebRTCPhone extends Emitter                  {
       logger$7.error('WebRTC close, unregister error', e);
     }
 
-    this.client.close();
+    this.client.close(force);
     this.unbind();
 
     this.incomingSessions = [];
@@ -10164,6 +10234,8 @@ class WebRTCPhone extends Emitter                  {
       }
 
       callSession.creationTime = new Date();
+      this._updateCallSession(callSession);
+
       this.eventEmitter.emit(ON_CALL_INCOMING, callSession, wantsToDoVideo);
     });
 
@@ -10207,6 +10279,8 @@ class WebRTCPhone extends Emitter                  {
       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);
     });
 
@@ -10431,6 +10505,7 @@ class WebRTCPhone extends Emitter                  {
       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.SessionState.Established,
       paused: this.client.isCallHeld(sipSession),

package/lib/simple/room/Stream.js

@@ -3594,13 +3594,21 @@ class CallSession {
 
   // Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
   set answered(value         ) {
-    this.answerTime = value ? new Date() : null;
+    this.answerTime = value ? this.answerTime || new Date() : null;
   }
 
   get answered()          {
     return !!this.answerTime;
   }
 
+  toJSON() {
+    const jsonObj = { ...this };
+    // $FlowFixMe
+    jsonObj.answered = this.answered;
+
+    return jsonObj;
+  }
+
   static parseCall(call      )              {
     return new CallSession({
       callId: call.id,
@@ -4852,6 +4860,34 @@ var chatdMethods = (client              , baseUrl        ) => ({
     client.get(`${baseUrl}/users/me/rooms/messages`, options),
 });
 
+//      
+
+                                     
+               
+  
+
+class MeetingStatus {
+               
+
+                
+
+  constructor({
+    full,
+  }         = {}) {
+    this.full = full;
+
+    // Useful to compare instead of instanceof with minified code
+    this.type = 'MeetingStatus';
+  }
+
+  static parse(plain                       )                {
+    return new MeetingStatus({
+      full: plain.full,
+    });
+  }
+
+}
+
 /*       */
 
                   
@@ -5046,6 +5082,8 @@ var calldMethods = (client              , baseUrl        ) => ({
     client.put(`${baseUrl}/users/me/calls/${callId}/record/resume`, null, null,
       ApiRequester.successResponseParser),
 
+  guestGetMeetingStatus: (meetingUuid        )                         =>
+    client.get(`${baseUrl}/guests/me/meetings/${meetingUuid}/status`).then(MeetingStatus.parse),
 });
 
 //      
@@ -6647,6 +6685,7 @@ const MAX_REGISTER_TRIES = 10;
                       
                
                 
+                         
                              
                     
               
@@ -6688,6 +6727,7 @@ class WebRTCClient extends Emitter {
                         
                                                       
                                             
+                       
 
   // sugar
                         
@@ -6750,6 +6790,7 @@ class WebRTCClient extends Emitter {
     this.conferences = {};
     this.networkMonitoringInterval = {};
     this.sessionNetworkStats = {};
+    this.forceClosed = false;
 
     this._boundOnHeartbeat = this._onHeartbeat.bind(this);
     this.heartbeat = new Heartbeat(config.heartbeatDelay, config.heartbeatTimeout, config.maxHeartbeats);
@@ -6871,6 +6912,8 @@ class WebRTCClient extends Emitter {
       skipRegister: this.skipRegister,
     };
 
+    this.forceClosed = false;
+
     if (this.skipRegister) {
       logger$8.info('sdk webrtc skip register...', logInfo);
       return Promise.resolve();
@@ -6895,7 +6938,15 @@ class WebRTCClient extends Emitter {
     const registerOptions = this._isWeb() ? {} : { extraContactHeaderParams: ['mobility=mobile'] };
 
     const onRegisterFailed = () => {
-      logger$8.info('sdk webrtc registering failed', { tries, registerer: !!this.registerer });
+      logger$8.info('sdk webrtc registering failed', {
+        tries,
+        registerer: !!this.registerer,
+        forceClosed: this.forceClosed,
+      });
+      if (this.forceClosed) {
+        return;
+      }
+
       this.connectionPromise = null;
       if (this.registerer && this.registerer.waiting) {
         this.registerer.waitingToggle(false);
@@ -7173,8 +7224,9 @@ class WebRTCClient extends Emitter {
     }
   }
 
-  async close() {
-    logger$8.info('sdk webrtc closing client', { userAgent: !!this.userAgent });
+  async close(force          = false) {
+    logger$8.info('sdk webrtc closing client', { userAgent: !!this.userAgent, force });
+    this.forceClosed = force;
     this._cleanupMedia();
     this.connectionPromise = null;
 
@@ -7193,7 +7245,7 @@ class WebRTCClient extends Emitter {
     this.userAgent.delegate = null;
     this.userAgent.stateChange.removeAllListeners();
 
-    await this._disconnectTransport();
+    await this._disconnectTransport(force);
 
     this._cleanupRegister();
 
@@ -8263,6 +8315,13 @@ 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,
@@ -8298,7 +8357,7 @@ 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],
@@ -8622,7 +8681,15 @@ class WebRTCClient extends Emitter {
     return userAgent.UserAgent.makeURI(`sip:${target}@${this.config.host}`);
   }
 
-  async _disconnectTransport() {
+  async _disconnectTransport(force          = false) {
+    if (force) {
+      // 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);
+      return;
+    }
+
     // Check if `disconnectPromise` is not present to avoid `Disconnect promise must not be defined` errors.
     if (this.userAgent && this.userAgent.transport && !this.userAgent.transport.disconnectPromise) {
       try {
@@ -9359,6 +9426,7 @@ class WebRTCPhone extends Emitter                  {
     }
 
     callSession.answerTime = new Date();
+    this._updateCallSession(callSession);
     this.eventEmitter.emit(ON_CALL_ACCEPTED, callSession, cameraEnabled);
 
     return callSession;
@@ -9575,6 +9643,7 @@ 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);
@@ -9674,7 +9743,7 @@ 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);
@@ -9885,6 +9954,7 @@ class WebRTCPhone extends Emitter                  {
     }
     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);
 
@@ -10026,7 +10096,7 @@ class WebRTCPhone extends Emitter                  {
     return this.client && this.client.isConference(callSession.sipCallId);
   }
 
-  async close()                {
+  async close(force          = false)                {
     logger$7.info('WebRTC close');
 
     try {
@@ -10038,7 +10108,7 @@ class WebRTCPhone extends Emitter                  {
       logger$7.error('WebRTC close, unregister error', e);
     }
 
-    this.client.close();
+    this.client.close(force);
     this.unbind();
 
     this.incomingSessions = [];
@@ -10164,6 +10234,8 @@ class WebRTCPhone extends Emitter                  {
       }
 
       callSession.creationTime = new Date();
+      this._updateCallSession(callSession);
+
       this.eventEmitter.emit(ON_CALL_INCOMING, callSession, wantsToDoVideo);
     });
 
@@ -10207,6 +10279,8 @@ class WebRTCPhone extends Emitter                  {
       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);
     });
 
@@ -10431,6 +10505,7 @@ class WebRTCPhone extends Emitter                  {
       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.SessionState.Established,
       paused: this.client.isCallHeld(sipSession),