@wazo/sdk 0.37.29 → 0.37.33

package/src/domain/MeetingStatus.js

@@ -0,0 +1,27 @@
+// @flow
+
+export type MeetingStatusResponse = {
+  full: boolean
+};
+
+export default class MeetingStatus {
+  type: string;
+
+  full: boolean;
+
+  constructor({
+    full,
+  }: Object = {}) {
+    this.full = full;
+
+    // Useful to compare instead of instanceof with minified code
+    this.type = 'MeetingStatus';
+  }
+
+  static parse(plain: MeetingStatusResponse): MeetingStatus {
+    return new MeetingStatus({
+      full: plain.full,
+    });
+  }
+
+}

package/src/domain/Phone/WebRTCPhone.js

@@ -631,6 +631,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
 
     callSession.answerTime = new Date();
+    this._updateCallSession(callSession);
     this.eventEmitter.emit(ON_CALL_ACCEPTED, callSession, cameraEnabled);
 
     return callSession;
@@ -847,6 +848,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
 
     this.shouldSendReinvite = false;
     callSession.answerTime = new Date();
+    this._updateCallSession(callSession);
     this.acceptedSessions[callSession.getId()] = true;
 
     this.eventEmitter.emit(ON_CALL_ANSWERED, callSession);
@@ -946,7 +948,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     // 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.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);
@@ -1157,6 +1159,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
     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);
 
@@ -1298,7 +1301,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     return this.client && this.client.isConference(callSession.sipCallId);
   }
 
-  async close(): Promise<void> {
+  async close(force: boolean = false): Promise<void> {
     logger.info('WebRTC close');
 
     try {
@@ -1310,7 +1313,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       logger.error('WebRTC close, unregister error', e);
     }
 
-    this.client.close();
+    this.client.close(force);
     this.unbind();
 
     this.incomingSessions = [];
@@ -1436,6 +1439,8 @@ export default class WebRTCPhone extends Emitter implements Phone {
       }
 
       callSession.creationTime = new Date();
+      this._updateCallSession(callSession);
+
       this.eventEmitter.emit(ON_CALL_INCOMING, callSession, wantsToDoVideo);
     });
 
@@ -1479,6 +1484,8 @@ export default class WebRTCPhone extends Emitter implements Phone {
       logger.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);
     });
 
@@ -1705,6 +1712,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       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),

package/src/domain/__tests__/CallSession.test.js

@@ -1,5 +1,7 @@
 import CallSession from '../CallSession';
 
+const stringify = cs => JSON.parse(JSON.stringify(cs));
+
 describe('CallSession domain', () => {
   it('should update from another CallSession without data loss', () => {
     const callSession = new CallSession({
@@ -38,7 +40,7 @@ describe('CallSession domain', () => {
     expect(cs2.is(cs1)).toBeTruthy();
   });
 
-  it('should set answerTime when setting answered', () => {
+  it('should set answerTime when setting answered to true', () => {
     const cs1 = new CallSession({});
     cs1.answered = true;
 
@@ -47,11 +49,37 @@ describe('CallSession domain', () => {
 
     const cs3 = new CallSession({ answered: true });
 
-    expect(cs1.answered).toBeTruthy();
+    expect(cs1.answered).toEqual(true);
     expect(cs1.answerTime).not.toBeNull();
-    expect(cs2.answered).toBeTruthy();
+    expect(cs2.answered).toEqual(true);
     expect(cs2.answerTime).not.toBeNull();
-    expect(cs3.answered).toBeTruthy();
+    expect(cs3.answered).toEqual(true);
     expect(cs3.answerTime).not.toBeNull();
+
+    const cs4 = stringify(cs1);
+    const cs5 = stringify(cs2);
+    const cs6 = stringify(cs3);
+
+    expect(cs4.answered).toEqual(true);
+    expect(cs5.answered).toEqual(true);
+    expect(cs6.answered).toEqual(true);
+  });
+
+  it('should reset answerTime when setting answered to false', () => {
+    const cs1 = new CallSession({});
+    cs1.answered = false;
+
+    const cs2 = new CallSession({ answered: false });
+
+    expect(cs1.answered).toEqual(false);
+    expect(cs1.answerTime).toBeNull();
+    expect(cs2.answered).toEqual(false);
+    expect(cs2.answerTime).toBeNull();
+
+    const cs4 = stringify(cs1);
+    const cs5 = stringify(cs2);
+
+    expect(cs4.answered).toEqual(false);
+    expect(cs5.answered).toEqual(false);
   });
 });

package/src/web-rtc-client.js

@@ -87,6 +87,7 @@ type WebRtcConfig = {
   displayName: string,
   host: string,
   port?: number,
+  websocketSip?: ?string,
   authorizationUser: ?string,
   password: ?string,
   uri: string,
@@ -128,6 +129,7 @@ export default class WebRTCClient extends Emitter {
   skipRegister: boolean;
   networkMonitoringInterval: { [string]: IntervalID };
   sessionNetworkStats: { [string]: Object };
+  forceClosed: boolean;
 
   // sugar
   ON_USER_AGENT: string;
@@ -190,6 +192,7 @@ export default 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);
@@ -311,6 +314,8 @@ export default class WebRTCClient extends Emitter {
       skipRegister: this.skipRegister,
     };
 
+    this.forceClosed = false;
+
     if (this.skipRegister) {
       logger.info('sdk webrtc skip register...', logInfo);
       return Promise.resolve();
@@ -335,7 +340,15 @@ export default class WebRTCClient extends Emitter {
     const registerOptions = this._isWeb() ? {} : { extraContactHeaderParams: ['mobility=mobile'] };
 
     const onRegisterFailed = () => {
-      logger.info('sdk webrtc registering failed', { tries, registerer: !!this.registerer });
+      logger.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);
@@ -613,8 +626,9 @@ export default class WebRTCClient extends Emitter {
     }
   }
 
-  async close() {
-    logger.info('sdk webrtc closing client', { userAgent: !!this.userAgent });
+  async close(force: boolean = false) {
+    logger.info('sdk webrtc closing client', { userAgent: !!this.userAgent, force });
+    this.forceClosed = force;
     this._cleanupMedia();
     this.connectionPromise = null;
 
@@ -633,7 +647,7 @@ export default class WebRTCClient extends Emitter {
     this.userAgent.delegate = null;
     this.userAgent.stateChange.removeAllListeners();
 
-    await this._disconnectTransport();
+    await this._disconnectTransport(force);
 
     this._cleanupRegister();
 
@@ -1703,6 +1717,13 @@ export default class WebRTCClient extends Emitter {
   }
 
   _createWebRTCConfiguration(configOverrides: Object = {}) {
+    let { host } = this.config;
+    let port = this.config.port || 443;
+
+    if (this.config.websocketSip) {
+      [host, port = 443] = this.config.websocketSip.split(':');
+    }
+
     const config: Object = {
       authorizationUsername: this.config.authorizationUser,
       authorizationPassword: this.config.password,
@@ -1738,7 +1759,7 @@ export default 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],
@@ -2062,7 +2083,15 @@ export default class WebRTCClient extends Emitter {
     return UserAgent.makeURI(`sip:${target}@${this.config.host}`);
   }
 
-  async _disconnectTransport() {
+  async _disconnectTransport(force: boolean = 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 {