@wazo/sdk 0.37.30 → 0.37.34

package/src/api/calld.js

@@ -7,6 +7,7 @@ import ChatMessage from '../domain/ChatMessage';
 import Voicemail from '../domain/Voicemail';
 import Call from '../domain/Call';
 import IndirectTransfer from '../domain/IndirectTransfer';
+import MeetingStatus from '../domain/MeetingStatus';
 
 type CallQuery = {
   from_mobile: boolean,
@@ -200,4 +201,6 @@ export default (client: ApiRequester, baseUrl: string) => ({
     client.put(`${baseUrl}/users/me/calls/${callId}/record/resume`, null, null,
       ApiRequester.successResponseParser),
 
+  guestGetMeetingStatus: (meetingUuid: string): Promise<MeetingStatus> =>
+    client.get(`${baseUrl}/guests/me/meetings/${meetingUuid}/status`).then(MeetingStatus.parse),
 });

package/src/domain/CallSession.js

@@ -344,13 +344,21 @@ export default class CallSession {
 
   // Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
   set answered(value: boolean) {
-    this.answerTime = value ? new Date() : null;
+    this.answerTime = value ? this.answerTime || new Date() : null;
   }
 
   get answered(): boolean {
     return !!this.answerTime;
   }
 
+  toJSON() {
+    const jsonObj = { ...this };
+    // $FlowFixMe
+    jsonObj.answered = this.answered;
+
+    return jsonObj;
+  }
+
   static parseCall(call: Call): CallSession {
     return new CallSession({
       callId: call.id,

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;
@@ -840,6 +841,12 @@ export default class WebRTCPhone extends Emitter implements Phone {
       return Promise.resolve(null);
     }
 
+    const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
+    if (sipSessionId && !(sipSessionId in this.callSessions)) {
+      logger.warn('Call session already ended or not found, can\'t accept it.', { sipSessionId });
+      return Promise.resolve(null);
+    }
+
     if (callSession.getId() in this.acceptedSessions) {
       logger.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
       return Promise.resolve(callSession.sipCallId);
@@ -847,6 +854,7 @@ 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 +954,7 @@ 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 +1165,7 @@ 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 +1307,7 @@ 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 +1319,7 @@ 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 +1445,8 @@ export default class WebRTCPhone extends Emitter implements Phone {
       }
 
       callSession.creationTime = new Date();
+      this._updateCallSession(callSession);
+
       this.eventEmitter.emit(ON_CALL_INCOMING, callSession, wantsToDoVideo);
     });
 
@@ -1460,9 +1471,17 @@ export default class WebRTCPhone extends Emitter implements Phone {
     });
 
     this.client.on(this.client.ACCEPTED, (sipSession: Session) => {
-      logger.info('WebRTC call accepted', { sipId: sipSession.id });
+      const sessionId = this.getSipSessionId(sipSession);
+      const hasSession = sessionId in this.callSessions;
+
+      logger.info('WebRTC call accepted', { sipId: sessionId, hasSession });
+
+      if (!hasSession) {
+        logger.warn('Call accepted ignored, session is no longer present in the WebRtcPhone', { sessionId });
+        return;
+      }
 
-      this._onCallAccepted(sipSession, this.client.hasVideo(this.getSipSessionId(sipSession)));
+      this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
 
       if (this.audioOutputDeviceId) {
         this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
@@ -1479,6 +1498,8 @@ 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);
     });
 
@@ -1596,9 +1617,14 @@ export default class WebRTCPhone extends Emitter implements Phone {
     return this.client.getSipSessionId(sipSession);
   }
 
-  _updateCallSession(callSession: CallSession) {
+  getSipSessionIdFromCallSession(callSession: CallSession) {
     const sipSession = this.findSipSession(callSession);
-    const sipSessionId = sipSession ? this.getSipSessionId(sipSession) : null;
+
+    return sipSession ? this.getSipSessionId(sipSession) : null;
+  }
+
+  _updateCallSession(callSession: CallSession) {
+    const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
 
     if (sipSessionId) {
       this.callSessions[sipSessionId] = callSession;
@@ -1705,6 +1731,7 @@ 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

@@ -75,7 +75,7 @@ export const events = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
 export const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
 export class CanceledCallError extends Error {}
 
-const MAX_REGISTER_TRIES = 10;
+const MAX_REGISTER_TRIES = 5;
 
 type MediaConfig = {
   audio: Object & boolean,
@@ -129,6 +129,7 @@ export default class WebRTCClient extends Emitter {
   skipRegister: boolean;
   networkMonitoringInterval: { [string]: IntervalID };
   sessionNetworkStats: { [string]: Object };
+  forceClosed: boolean;
 
   // sugar
   ON_USER_AGENT: string;
@@ -191,6 +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);
@@ -312,6 +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();
@@ -336,7 +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);
@@ -614,8 +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;
 
@@ -634,7 +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();
 
@@ -2070,7 +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 {