@wazo/sdk 0.37.0-rc.5 → 0.37.2

package/src/checker/checks/aor.js

@@ -5,6 +5,8 @@ export default {
   check: async (server, session) => {
     const apiClient = new WazoApiClient({ server });
     apiClient.setToken(session.token);
+    apiClient.disableErrorLogging();
+
     const line = session.primaryWebRtcLine();
     if (!line) {
       return 'No WebRTC line for this user';

package/src/checker/checks/api.js

@@ -6,6 +6,7 @@ export default {
   check: async (server, session) => {
     const client = new WazoApiClient({ server });
     client.setToken(session.token);
+    client.disableErrorLogging();
 
     const handleApiError = (apiName, error) => {
       const statusText = error instanceof ServerError ? 'server error' : 'api error';

package/src/domain/CallLog.js

@@ -1,20 +1,24 @@
 // @flow
 import moment from 'moment';
 
-import Session from './Session';
 import newFrom from '../utils/new-from';
 
+import Session from './Session';
+import type { RecordingResponse } from './Recording';
+import Recording from './Recording';
+
 type CallLogResponse = {
-  answer: string,
+  answer: ?string,
   answered: boolean,
   call_direction: string,
   destination_extension: string,
   destination_name: ?string,
   duration: number,
-  end: string,
+  end: ?string,
   id: number,
   source_extension: string,
   source_name: string,
+  recordings: RecordingResponse[],
   requested_extension: string,
   requested_name: string,
   start: string,
@@ -27,7 +31,7 @@ type Response = {
 };
 
 type CallLogArguments = {
-  answer: Date,
+  answer: ?Date,
   answered: boolean,
   newMissedCall?: boolean,
   callDirection: string,
@@ -42,13 +46,14 @@ type CallLogArguments = {
   id: number,
   duration: number,
   start: Date,
-  end: Date,
+  end: ?Date,
+  recordings: Recording[],
 };
 
 export default class CallLog {
   type: string;
 
-  answer: Date;
+  answer: ?Date;
   answered: boolean;
   newMissedCall: boolean;
   callDirection: string;
@@ -65,7 +70,7 @@ export default class CallLog {
   id: number;
   duration: number;
   start: Date;
-  end: Date;
+  end: ?Date;
 
   static merge(current: Array<CallLog>, toMerge: Array<CallLog>): Array<?CallLog> {
     const onlyUnique = (value, index, self) => self.indexOf(value) === index;
@@ -82,7 +87,7 @@ export default class CallLog {
 
   static parse(plain: CallLogResponse): CallLog {
     return new CallLog({
-      answer: moment(plain.answer).toDate(),
+      answer: plain.answer ? moment(plain.answer).toDate() : null,
       answered: plain.answered,
       callDirection: plain.call_direction,
       destination: {
@@ -96,13 +101,14 @@ export default class CallLog {
       id: plain.id,
       duration: (plain.duration || 0) * 1000, // duration is in seconds
       start: moment(plain.start).toDate(),
-      end: moment(plain.end).toDate(),
+      end: plain.end ? moment(plain.end).toDate() : null,
+      recordings: Recording.parseMany(plain.recordings),
     });
   }
 
   static parseNew(plain: CallLogResponse, session: Session): CallLog {
     return new CallLog({
-      answer: moment(plain.answer).toDate(),
+      answer: plain.answer ? moment(plain.answer).toDate() : null,
       answered: plain.answered,
       callDirection: plain.call_direction,
       destination: {
@@ -116,7 +122,8 @@ export default class CallLog {
       id: plain.id,
       duration: (plain.duration || 0) * 1000, // duration is in seconds
       start: moment(plain.start).toDate(),
-      end: moment(plain.end).toDate(),
+      end: plain.end ? moment(plain.end).toDate() : null,
+      recordings: Recording.parseMany(plain.recordings),
       // @TODO: FIXME add verification declined vs missed call
       newMissedCall: session
         && session.hasExtension(plain.requested_extension || plain.destination_extension) && !plain.answered,

package/src/domain/CallSession.js

@@ -2,7 +2,6 @@
 import { SessionState } from 'sip.js/lib/api/session-state';
 
 import Call from './Call';
-import Session from './Session';
 import newFrom from '../utils/new-from';
 import updateFrom from '../utils/update-from';
 
@@ -311,7 +310,7 @@ export default class CallSession {
     return newFrom(callSession, CallSession);
   }
 
-  static parseCall(session: Session, call: Call): CallSession {
+  static parseCall(call: Call): CallSession {
     return new CallSession({
       callId: call.id,
       sipCallId: call.sipCallId,

package/src/domain/Contact.js

@@ -420,7 +420,7 @@ export default class Contact {
 
     return new Contact({
       sourceId: single.id || '',
-      name: single.displayName,
+      name: single.displayName || '',
       number: numbers.length ? numbers[0].number : '',
       numbers,
       emails,
@@ -451,7 +451,7 @@ export default class Contact {
 
     return new Contact({
       sourceId: single.id || '',
-      name: single.name,
+      name: single.name || '',
       number: numbers.length ? numbers[0].number : '',
       numbers,
       emails,
@@ -552,7 +552,7 @@ export default class Contact {
   }: ContactArguments = {}) {
     this.id = id;
     this.uuid = uuid;
-    this.name = name;
+    this.name = name || '';
     this.number = number;
     this.numbers = numbers;
     this.email = email;

package/src/domain/Phone/CTIPhone.js

@@ -102,7 +102,7 @@ export default class CTIPhone extends Emitter implements Phone {
     if (!this.currentCall) {
       return null;
     }
-    const callSession = CallSession.parseCall(this.session, this.currentCall);
+    const callSession = CallSession.parseCall(this.currentCall);
     this.eventEmitter.emit('onCallOutgoing', callSession);
 
     return callSession;
@@ -286,6 +286,10 @@ export default class CTIPhone extends Emitter implements Phone {
     return null;
   }
 
+  getLocalVideoStream(): ?MediaStream {
+    return null;
+  }
+
   setActiveSipSession() {}
 
   isRegistered(): boolean {
@@ -301,7 +305,7 @@ export default class CTIPhone extends Emitter implements Phone {
   }
 
   getCurrentCallSession(): ?CallSession {
-    return this.currentCall ? CallSession.parseCall(this.session, this.currentCall) : null;
+    return this.currentCall ? CallSession.parseCall(this.currentCall) : null;
   }
 
   enableRinging() {}

package/src/domain/Phone/Phone.js

@@ -56,6 +56,8 @@ export interface Phone {
 
   getLocalStreamForCall(callSession: CallSession): ?MediaStream;
 
+  getLocalVideoStream(callSession: CallSession): ?MediaStream;
+
   getOptions(): AvailablePhoneOptions;
 
   getRemoteStreamForCall(callSession: CallSession): ?MediaStream;

package/src/domain/Phone/WebRTCPhone.js

@@ -190,9 +190,14 @@ export default class WebRTCPhone extends Emitter implements Phone {
     return true;
   }
 
-  async sendReinvite(callSession: ?CallSession, constraints: ?Object = null, conference: boolean = false) {
+  async sendReinvite(callSession: ?CallSession, constraints: ?Object = null, conference: boolean = false,
+    audioOnly: boolean = false) {
     const sipSession = this.findSipSession(callSession);
-    logger.info('WebRTC phone - send reinvite', { sessionId: sipSession ? sipSession.id : null, constraints });
+    logger.info('WebRTC phone - send reinvite', {
+      sessionId: sipSession ? sipSession.id : null,
+      constraints,
+      audioOnly,
+    });
 
     if (!sipSession) {
       return null;
@@ -218,7 +223,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     // Force reinvite in SDH
     sipSession.sessionDescriptionHandler.localMediaStreamConstraints = null;
 
-    return this.client.reinvite(sipSession, constraints, conference);
+    return this.client.reinvite(sipSession, constraints, conference, audioOnly);
   }
 
   getUserAgent() {
@@ -294,9 +299,9 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
 
     // Release local video stream when downgrading to audio
-    const localVideoStream = sipSession.sessionDescriptionHandler.localMediaStream;
+    const localStream = sipSession.sessionDescriptionHandler.localMediaStream;
     const pc = sipSession.sessionDescriptionHandler.peerConnection;
-    const videoTracks = localVideoStream.getVideoTracks();
+    const videoTracks = localStream.getVideoTracks();
 
     // Remove video senders
     pc.getSenders().filter(sender => sender.track && sender.track.kind === 'video').forEach(videoSender => {
@@ -311,7 +316,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     videoTracks.forEach(videoTrack => {
       videoTrack.enabled = false;
       videoTrack.stop();
-      localVideoStream.removeTrack(videoTrack);
+      localStream.removeTrack(videoTrack);
     });
 
     this._sendReinviteMessage(callSession, false);
@@ -328,12 +333,14 @@ export default class WebRTCPhone extends Emitter implements Phone {
 
     const pc = sipSession.sessionDescriptionHandler.peerConnection;
     const shouldScreenShare = constraints && constraints.screen;
+    const options = sipSession.sessionDescriptionHandlerOptionsReInvite;
+    const wasAudioOnly = options && options.audioOnly;
 
     // Check if a video sender already exists
     let videoSender;
     if (isConference) {
       const trans = pc.getTransceivers().find(transceiver => transceiver.mid === '1');
-      videoSender = trans.sender;
+      videoSender = trans ? trans.sender : null;
     } else {
       videoSender = pc.getSenders().find(sender => sender.track === null);
     }
@@ -370,6 +377,11 @@ export default class WebRTCPhone extends Emitter implements Phone {
       this._onScreenSharing(newStream, sipSession, callSession, false);
     }
 
+    // We have to reinvite to change the direction on the bundle when upgrading from an audioOnly conference
+    if (isConference && wasAudioOnly) {
+      return true;
+    }
+
     // No reinvite needed here
     return false;
   }
@@ -456,7 +468,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
         event.track.enabled = false;
       }
 
-      return this.eventEmitter.emit(ON_VIDEO_STREAM, stream, event.track.id, event);
+      return this.eventEmitter.emit(ON_VIDEO_STREAM, stream, event.track.id, event, sipSession);
     };
 
     peerConnection.onremovestream = event => {
@@ -512,14 +524,11 @@ export default class WebRTCPhone extends Emitter implements Phone {
         const targetCallSession = callSession || this.currentCallSession;
         const conference = this.isConference(targetCallSession);
 
-        if (this.currentScreenShare.sender && this.currentScreenShare.hadVideo) {
-          // When stopping screenshare and we had video before that, we have to re-upgrade
-          await this.sendReinvite(targetCallSession, { audio: false, video: true }, conference);
-        } else {
-          // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
-          // We have to downgrade to audio.
-          await this.sendReinvite(targetCallSession, { audio: false, video: false }, conference);
-        }
+        // When stopping screenshare and we had video before that, we have to re-upgrade
+        // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
+        // We have to downgrade to audio.
+        const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
+        await this.sendReinvite(targetCallSession, { audio: false, video: screenshareStopped }, conference);
       }
     } catch (e) {
       console.warn(e);
@@ -856,6 +865,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
   }
 
+  // @Deprecated
   unhold(callSession: CallSession, withEvent: boolean = true, isConference: boolean = false): void {
     logger.info('WebRTC unhold', { id: callSession ? callSession.getId() : null });
 

package/src/domain/Recording.js

@@ -0,0 +1,48 @@
+// @flow
+import moment from 'moment';
+
+export type RecordingResponse = {
+  deleted: boolean,
+  end_time: ?string,
+  filename: string,
+  start_time: string,
+  uuid: string,
+};
+
+type RecordingArguments = {
+  deleted: boolean,
+  end: ?Date,
+  fileName: string,
+  start: Date,
+  uuid: string,
+}
+
+export default class Recording {
+  deleted: boolean;
+  end: ?Date;
+  fileName: string;
+  start: Date;
+  uuid: string;
+
+  static parseMany(recordings: RecordingResponse[]): Recording[] {
+    return recordings.map(item => Recording.parse(item));
+  }
+
+  static parse(plain: RecordingResponse): Recording {
+    return new Recording({
+      deleted: plain.deleted,
+      fileName: plain.filename,
+      end: plain.end_time ? moment(plain.end_time).toDate() : null,
+      start: moment(plain.start_time).toDate(),
+      uuid: plain.uuid,
+    });
+  }
+
+  constructor({ deleted, end, fileName, start, uuid }: RecordingArguments) {
+    this.deleted = deleted;
+    this.end = end;
+    this.fileName = fileName;
+    this.start = start;
+    this.uuid = uuid;
+  }
+}

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

@@ -2,6 +2,7 @@
 import moment from 'moment';
 
 import CallLog from '../CallLog';
+import Recording from '../Recording';
 
 describe('CallLog domain', () => {
   it('can parse a plain call log to domain', () => {
@@ -22,6 +23,15 @@ describe('CallLog domain', () => {
           requested_extension: '',
           requested_name: '',
           id: 215,
+          recordings: [
+            {
+              deleted: false,
+              end_time: '2021-08-11T13:22:50.519733+00:00',
+              filename: '2021-08-11T13_22_45UTC-61886-xxxx-5c72-4072-8503-e3ad9a41c0a0.wav',
+              start_time: '2021-08-11T13:22:45.161954+00:00',
+              uuid: '0394ad6f-5c72-4072-8503-e3ad9a41c0a0',
+            },
+          ],
         },
       ],
       total: 233,
@@ -46,6 +56,15 @@ describe('CallLog domain', () => {
         duration: 4000,
         start: moment('2017-08-06T13:50:55.819057+00:00').toDate(),
         end: moment('2017-08-08T13:51:23.822529+00:00').toDate(),
+        recordings: [
+          new Recording({
+            deleted: false,
+            end: moment('2021-08-11T13:22:50.519733+00:00').toDate(),
+            fileName: '2021-08-11T13_22_45UTC-61886-xxxx-5c72-4072-8503-e3ad9a41c0a0.wav',
+            start: moment('2021-08-11T13:22:45.161954+00:00').toDate(),
+            uuid: '0394ad6f-5c72-4072-8503-e3ad9a41c0a0',
+          }),
+        ],
       }),
     ]);
   });

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

@@ -394,7 +394,14 @@ describe('Session domain', () => {
     const acl = ['calld.lines.*.presences.read'];
     const session = new Session({ acl, expiresAt: new Date(), token: 'abc' });
 
+    const oldWarn = console.warn;
+    // $FlowFixMe
+    console.warn = () => {};
+
     expect(session.acls).toEqual(acl);
     expect(session.acl).toEqual(acl);
+
+    // $FlowFixMe
+    console.warn = oldWarn;
   });
 });

package/src/index.js

@@ -11,6 +11,7 @@ import BadResponse from './domain/BadResponse';
 import ServerError from './domain/ServerError';
 import Call from './domain/Call';
 import CallLog from './domain/CallLog';
+import Recording from './domain/Recording';
 import ChatMessage from './domain/ChatMessage';
 import ChatRoom from './domain/ChatRoom';
 import Contact from './domain/Contact';
@@ -27,6 +28,8 @@ import Room from './domain/Room';
 import CallSession from './domain/CallSession';
 import IndirectTransfer from './domain/IndirectTransfer';
 import SwitchboardCall from './domain/SwitchboardCall';
+import WebRTCPhone from './domain/Phone/WebRTCPhone';
+import CTIPhone from './domain/Phone/CTIPhone';
 import type {
   NewContact as NewContactType,
   ContactResponse as ContactResponseType,
@@ -93,10 +96,12 @@ export default {
   ServerError,
   Call,
   CallSession,
+  CTIPhone,
   Features,
   IndirectTransfer,
   SwitchboardCall,
   CallLog,
+  Recording,
   ChatMessage,
   ChatRoom,
   Contact,
@@ -116,4 +121,5 @@ export default {
   LINE_STATE,
   SOCKET_EVENTS,
   Wazo,
+  WebRTCPhone,
 };

package/src/lib/WazoSessionDescriptionHandler.js

@@ -84,6 +84,7 @@ class WazoSessionDescriptionHandler extends SessionDescriptionHandler {
     const iceRestart = options.offerOptions ? options.offerOptions.iceRestart : false;
 
     const isConference = options ? !!options.conference : false;
+    const audioOnly = options ? !!options.audioOnly : false;
 
     // We should wait for ice when iceRestart (reinvite) or for the first invite
     // We shouldn't wait for ice when holding or resuming the call
@@ -110,16 +111,16 @@ class WazoSessionDescriptionHandler extends SessionDescriptionHandler {
     };
 
     return this.getLocalMediaStream(options)
-      .then(() => this.updateDirection(options, isConference))
+      .then(() => this.updateDirection(options, isConference, audioOnly))
       .then(() => this.createDataChannel(options))
       .then(() => {
-        if (isConference && (options.constraints && !options.constraints.video) && !('hold' in options)) {
+        if (isConference && (options.constraints && !options.constraints.video) && !('hold' in options) && !audioOnly) {
           // Add a video an empty bundle to be able to replaceTrack when joining a conference without video
           this.peerConnection.addTransceiver('video', { streams: [this._localMediaStream], direction: 'sendrecv' });
         }
       })
       .then(() => this.createLocalOfferOrAnswer(options))
-      .then((sessionDescription) => this.setLocalSessionDescription(sessionDescription))
+      .then(sessionDescription => this.setLocalSessionDescription(sessionDescription))
       .then(() => this.waitForIceGatheringComplete(iceRestart, iceTimeout))
       .then(() => this.getLocalSessionDescription())
       .then(description => {
@@ -242,7 +243,8 @@ class WazoSessionDescriptionHandler extends SessionDescriptionHandler {
   }
 
   // Overridden to send `inactive` in conference
-  updateDirection(options?: SessionDescriptionHandlerOptions, isConference: boolean = false): Promise<void> {
+  updateDirection(options?: SessionDescriptionHandlerOptions, isConference: boolean = false,
+    audioOnly: boolean = false): Promise<void> {
     if (this._peerConnection === undefined) {
       return Promise.reject(new Error('Peer connection closed.'));
     }
@@ -257,14 +259,17 @@ class WazoSessionDescriptionHandler extends SessionDescriptionHandler {
         // if we are stable, assume we are creating a local offer
         this.logger.debug('SessionDescriptionHandler.updateDirection - setting offer direction');
         // determine the direction to offer given the current direction and hold state
-        const directionToOffer = (currentDirection: Object): Object => {
+        const directionToOffer = (currentDirection: string, transceiver: Object /* RTCRtpTransceiver */): Object => {
           if (isConference) {
+            if (audioOnly && transceiver.receiver.track && transceiver.receiver.track.kind === 'video') {
+              return 'sendonly';
+            }
             return options && options.hold ? 'sendonly' : 'sendrecv';
           }
 
           switch (currentDirection) {
             case 'inactive':
-              return options && options.hold ? 'inactive' : 'recvonly';
+              return options && options.hold ? 'inactive' : (isConference && !audioOnly ? 'sendrecv' : 'recvonly');
             case 'recvonly':
               return options && options.hold ? 'inactive' : 'recvonly';
             case 'sendonly':
@@ -280,7 +285,7 @@ class WazoSessionDescriptionHandler extends SessionDescriptionHandler {
         // set the transceiver direction to the offer direction
         this._peerConnection.getTransceivers().forEach((transceiver) => {
           if (transceiver.direction /* guarding, but should always be true */) {
-            const offerDirection = directionToOffer(transceiver.direction);
+            const offerDirection = directionToOffer(transceiver.direction, transceiver);
             if (transceiver.direction !== offerDirection) {
               // eslint-disable-next-line no-param-reassign
               transceiver.direction = offerDirection;
@@ -339,6 +344,13 @@ class WazoSessionDescriptionHandler extends SessionDescriptionHandler {
         // set the transceiver direction to the answer direction
         this._peerConnection.getTransceivers().forEach((transceiver) => {
           if (transceiver.direction /* guarding, but should always be true */) {
+            const { receiver } = transceiver;
+            if (isConference && audioOnly && receiver.track && receiver.track.kind === 'video') {
+              // eslint-disable-next-line no-param-reassign
+              transceiver.direction = 'inactive';
+              return;
+            }
+
             if (transceiver.direction !== 'stopped' && transceiver.direction !== answerDirection) {
               // eslint-disable-next-line no-param-reassign
               transceiver.direction = answerDirection;
@@ -377,8 +389,8 @@ class WazoSessionDescriptionHandler extends SessionDescriptionHandler {
       }
       const sender = pc.getSenders().find((otherSender) => otherSender.track && otherSender.track.kind === kind);
 
-      // Do not reuse sender tracks in SFU
-      if (sender && !sfu) {
+      // Do not reuse sender video tracks in SFU
+      if (sender && (!sfu || newTrack.kind === 'audio')) {
         trackUpdates.push(
           new Promise((resolve) => {
             this.logger.debug(`SessionDescriptionHandler.setLocalMediaStream - replacing sender ${kind} track`);

package/src/service/IssueReporter.js

@@ -243,7 +243,13 @@ class IssueReporter {
   }
 
   _isLevelAbove(level1: string, level2: string) {
-    return LOG_LEVELS.indexOf(level1) > LOG_LEVELS.indexOf(level2);
+    const index1 = LOG_LEVELS.indexOf(level1);
+    const index2 = LOG_LEVELS.indexOf(level2);
+    if (index1 === -1 || index2 === -1) {
+      return false;
+    }
+
+    return index1 > index2;
   }
 
   _makeCategory(category: string) {

package/src/service/__tests__/IssueReporter.test.js

@@ -1,15 +1,27 @@
 /* eslint-disable no-underscore-dangle */
 /* eslint-disable no-useless-escape */
+/* eslint-disable no-console */
 import IssueReporter from '../IssueReporter';
 import { realFetch } from '../../utils/api-requester';
 
 jest.mock('../../utils/api-requester');
 
-IssueReporter.enable();
+let oldLog;
+let oldError;
 
 class MyError extends Error {}
 
 describe('IssueReporter', () => {
+  beforeAll(() => {
+    oldLog = console.log;
+    oldError = console.error;
+
+    console.log = () => {};
+    console.error = () => {};
+
+    IssueReporter.enable();
+  });
+
   beforeEach(() => {
     jest.resetAllMocks();
 
@@ -18,12 +30,19 @@ describe('IssueReporter', () => {
     }));
   });
 
+  afterAll(() => {
+    console.log = oldLog;
+    console.error = oldError;
+  });
+
   it('should compute level order', () => {
     expect(IssueReporter._isLevelAbove('info', 'trace')).toBeTruthy();
     expect(IssueReporter._isLevelAbove('trace', 'warn')).toBeFalsy();
     expect(IssueReporter._isLevelAbove('error', 'warn')).toBeTruthy();
     expect(IssueReporter._isLevelAbove('info', 'warn')).toBeFalsy();
     expect(IssueReporter._isLevelAbove('trace', 'trace')).toBeFalsy();
+    expect(IssueReporter._isLevelAbove('trace', 'something')).toBeFalsy();
+    expect(IssueReporter._isLevelAbove('something', 'trace')).toBeFalsy();
   });
 
   it('should send if verbosity is higher than required', () => {

package/src/simple/Phone.js

@@ -283,6 +283,7 @@ class Phone extends Emitter {
     return this.phone && this.phone.hasLocalVideo(callSession);
   }
 
+  // @Deprecated
   getLocalMediaStream(callSession: CallSession) {
     logger.warn('Phone.getLocalMediaStream is deprecated, use Phone.getLocalStream instead');
 
@@ -301,17 +302,20 @@ class Phone extends Emitter {
     return this.phone ? this.phone.getRemoteVideoStream(callSession) : null;
   }
 
+  // @Deprecated
   getRemoteStreamForCall(callSession: CallSession) {
     logger.warn('Phone.getRemoteStreamForCall is deprecated, use Phone.getRemoteStream instead');
     return this.getRemoteStream(callSession);
   }
 
   // Returns remote streams directly from the peerConnection
+  // @Deprecated
   getRemoteStreamsForCall(callSession: CallSession) {
-    logger.warn('Phone.getRe is deprecated, use Phone.getLocalStream instead');
+    logger.warn('Phone.getRemoteStreamsForCall is deprecated, use Phone.getLocalStream instead');
     return this.getLocalStream(callSession);
   }
 
+  // @Deprecated
   getRemoteVideoStreamForCall(callSession: CallSession) {
     logger.warn('Phone.getRemoteVideoStreamForCall is deprecated, use Phone.getRemoteVideoStream instead');
 

package/src/simple/index.js

@@ -16,6 +16,7 @@ import BadResponse from '../domain/BadResponse';
 import ServerError from '../domain/ServerError';
 import Call from '../domain/Call';
 import CallLog from '../domain/CallLog';
+import Recording from '../domain/Recording';
 import ChatMessage from '../domain/ChatMessage';
 import ChatRoom from '../domain/ChatRoom';
 import Contact from '../domain/Contact';
@@ -28,6 +29,8 @@ import Voicemail from '../domain/Voicemail';
 import Relocation from '../domain/Relocation';
 import ConferenceRoom from '../domain/Room';
 import CallSession from '../domain/CallSession';
+import WebRTCPhone from '../domain/Phone/WebRTCPhone';
+import CTIPhone from '../domain/Phone/CTIPhone';
 import IndirectTransfer from '../domain/IndirectTransfer';
 import SwitchboardCall from '../domain/SwitchboardCall';
 import IssueReporter from '../service/IssueReporter';
@@ -58,6 +61,8 @@ const Wazo = {
     ServerError,
     Call,
     CallLog,
+    CTIPhone,
+    Recording,
     ChatMessage,
     ChatRoom,
     Contact,
@@ -72,6 +77,7 @@ const Wazo = {
     CallSession,
     IndirectTransfer,
     SwitchboardCall,
+    WebRTCPhone,
   },
 
   // Api