@wazo/sdk 0.37.27 → 0.37.31

package/src/domain/CallSession.js

@@ -1,5 +1,6 @@
 // @flow
 import { SessionState } from 'sip.js/lib/api/session-state';
+import type { Session } from 'sip.js/lib/api/session';
 
 import Call from './Call';
 import newFrom from '../utils/new-from';
@@ -7,13 +8,16 @@ import updateFrom from '../utils/update-from';
 
 type CallSessionArguments = {
   answered: boolean,
+  answerTime?: ?Date,
   answeredBySystem: boolean,
   call: ?Call,
   callId: string,
   callerNumber: string,
   cameraEnabled: boolean,
+  creationTime?: ?Date,
   dialedExtension?: string | null,
   displayName: string,
+  endTime?: ?Date,
   isCaller: boolean,
   muted: boolean,
   videoMuted: boolean,
@@ -30,6 +34,7 @@ type CallSessionArguments = {
   screensharing: boolean,
   recording: boolean,
   recordingPaused: boolean,
+  sipSession?: Session,
 };
 
 export default class CallSession {
@@ -49,11 +54,15 @@ export default class CallSession {
 
   callerNumber: string;
 
-  startTime: number;
+  creationTime: ?Date;
 
-  isCaller: boolean;
+  startTime: number; // = creationTime
+
+  answerTime: ?Date;
 
-  answered: boolean;
+  endTime: ?Date;
+
+  isCaller: boolean;
 
   answeredBySystem: boolean;
 
@@ -89,6 +98,8 @@ export default class CallSession {
 
   recordingPaused: boolean;
 
+  sipSession: Session;
+
   constructor({
     answered,
     answeredBySystem,
@@ -101,6 +112,8 @@ export default class CallSession {
     paused,
     ringing,
     startTime,
+    creationTime,
+    endTime,
     cameraEnabled,
     dialedExtension,
     sipCallId,
@@ -113,12 +126,16 @@ export default class CallSession {
     recording,
     recordingPaused,
     videoRemotelyDowngraded,
+    sipSession,
+    answerTime,
   }: CallSessionArguments) {
     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;
@@ -137,6 +154,8 @@ export default 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';
@@ -167,7 +186,7 @@ export default class CallSession {
   }
 
   answer() {
-    this.answered = true;
+    this.answerTime = new Date();
   }
 
   systemAnswer() {
@@ -323,6 +342,15 @@ export default 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: boolean) {
+    this.answerTime = value ? this.answerTime || new Date() : null;
+  }
+
+  get answered(): boolean {
+    return !!this.answerTime;
+  }
+
   static parseCall(call: Call): CallSession {
     return new CallSession({
       callId: call.id,

package/src/domain/Phone/WebRTCPhone.js

@@ -30,6 +30,7 @@ export const ON_CAMERA_DISABLED = 'onCameraDisabled';
 export const ON_CAMERA_RESUMED = 'onCameraResumed';
 export const ON_CALL_CANCELED = 'onCallCanceled';
 export const ON_CALL_FAILED = 'onCallFailed';
+export const ON_CALL_REJECTED = 'onCallRejected';
 export const ON_CALL_ENDED = 'onCallEnded';
 export const ON_CALL_ENDING = 'onCallEnding';
 export const ON_MESSAGE = 'onMessage';
@@ -74,6 +75,7 @@ export const events = [
   ON_CAMERA_DISABLED,
   ON_CALL_FAILED,
   ON_CALL_ENDED,
+  ON_CALL_REJECTED,
   ON_MESSAGE,
   ON_REINVITE,
   ON_TRACK,
@@ -430,6 +432,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
 
           // 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);
@@ -627,6 +630,8 @@ export default class WebRTCPhone extends Emitter implements Phone {
       this.removeIncomingSessions(sipSessionId);
     }
 
+    callSession.answerTime = new Date();
+    this._updateCallSession(callSession);
     this.eventEmitter.emit(ON_CALL_ACCEPTED, callSession, cameraEnabled);
 
     return callSession;
@@ -842,6 +847,8 @@ 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);
@@ -941,7 +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);
@@ -1151,6 +1158,8 @@ export default class WebRTCPhone extends Emitter implements Phone {
       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);
 
@@ -1429,6 +1438,9 @@ export default class WebRTCPhone extends Emitter implements Phone {
           callSession);
       }
 
+      callSession.creationTime = new Date();
+      this._updateCallSession(callSession);
+
       this.eventEmitter.emit(ON_CALL_INCOMING, callSession, wantsToDoVideo);
     });
 
@@ -1468,6 +1480,15 @@ export default class WebRTCPhone extends Emitter implements Phone {
       this.eventEmitter.emit(ON_CALL_ERROR, e);
     });
 
+    this.client.on(this.client.REJECTED, session => {
+      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);
+    });
+
     this.client.on(this.client.UNREGISTERED, () => {
       logger.info('WebRTC unregistered');
       this.eventEmitter.emit(ON_UNREGISTERED);
@@ -1682,6 +1703,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     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,
@@ -1689,6 +1711,9 @@ export default class WebRTCPhone extends Emitter implements Phone {
       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,
@@ -1699,6 +1724,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       videoMuted: fromSession ? fromSession.isVideoMuted() : false,
       recording: fromSession ? fromSession.isRecording() : false,
       recordingPaused: false, // @TODO
+      sipSession,
       ...extra,
     });
 

package/src/domain/SFUNotAvailableError.js

@@ -0,0 +1,7 @@
+/* @flow */
+export default class SFUNotAvailableError extends Error {
+  constructor(message: string = 'Sorry your user is not configured to support video conference') {
+    super(message);
+    this.name = 'SFUNotAvailableError';
+  }
+}

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

@@ -37,4 +37,21 @@ describe('CallSession domain', () => {
     expect(cs1.is(cs2)).toBeTruthy();
     expect(cs2.is(cs1)).toBeTruthy();
   });
+
+  it('should set answerTime when setting answered', () => {
+    const cs1 = new CallSession({});
+    cs1.answered = true;
+
+    const cs2 = new CallSession({});
+    cs2.answer();
+
+    const cs3 = new CallSession({ answered: true });
+
+    expect(cs1.answered).toBeTruthy();
+    expect(cs1.answerTime).not.toBeNull();
+    expect(cs2.answered).toBeTruthy();
+    expect(cs2.answerTime).not.toBeNull();
+    expect(cs3.answered).toBeTruthy();
+    expect(cs3.answerTime).not.toBeNull();
+  });
 });

package/src/index.js

@@ -9,6 +9,7 @@ import IssueReporter from './service/IssueReporter';
 // Domain
 import BadResponse from './domain/BadResponse';
 import ServerError from './domain/ServerError';
+import SFUNotAvailableError from './domain/SFUNotAvailableError';
 import Call from './domain/Call';
 import CallLog from './domain/CallLog';
 import Recording from './domain/Recording';
@@ -95,6 +96,7 @@ export default {
   WazoWebSocketClient: WebSocketClient,
   BadResponse,
   ServerError,
+  SFUNotAvailableError,
   Call,
   CallSession,
   CTIPhone,

package/src/simple/Phone.js

@@ -14,6 +14,7 @@ import IssueReporter from '../service/IssueReporter';
 import Emitter from '../utils/Emitter';
 
 import Wazo from './index';
+import SFUNotAvailableError from '../domain/SFUNotAvailableError';
 
 const logger = IssueReporter.loggerFor('simple-phone');
 const sipLogger = IssueReporter.loggerFor('sip.js');
@@ -397,7 +398,7 @@ class Phone extends Emitter {
 
   checkSfu() {
     if (!this.hasSfu()) {
-      throw new Error('Sorry your user is not configured to support video conference');
+      throw new SFUNotAvailableError();
     }
   }
 

package/src/simple/index.js

@@ -15,6 +15,7 @@ import { CanceledCallError } from '../web-rtc-client';
 
 import BadResponse from '../domain/BadResponse';
 import ServerError from '../domain/ServerError';
+import SFUNotAvailableError from '../domain/SFUNotAvailableError';
 import Call from '../domain/Call';
 import CallLog from '../domain/CallLog';
 import Recording from '../domain/Recording';
@@ -101,6 +102,7 @@ const Wazo = {
   InvalidSubscription,
   InvalidAuthorization,
   CanceledCallError,
+  SFUNotAvailableError,
 };
 
 if (window) {

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,
@@ -1703,6 +1704,13 @@ 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 +1746,7 @@ 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],