@wazo/sdk 1.0.0-alpha.2 → 1.0.0-alpha.20

package/esm/__tests__/__voice__/call.test.js

@@ -2,7 +2,7 @@
 import { Inviter, UserAgent } from 'sip.js';
 import softphone from '../../voice/softphone';
 import Call from '../../voice/call';
-import { Actions, EstablishedStates, States } from '../../state-machine/call-state-machine';
+import { Actions, EstablishedStates, EstablishedSubStates, States } from '../../state-machine/call-state-machine';
 import { Actions as SoftphoneActions } from '../../state-machine/softphone-state-machine';
 import ApiCall from '../../domain/Call';
 import InvalidStateTransition from '../../domain/InvalidStateTransition';
@@ -27,6 +27,9 @@ jest.mock('../..', () => ({
 }));
 jest.mock('../../web-rtc-client', () => jest.fn().mockImplementation(() => ({
     answer: jest.fn(() => Promise.resolve()),
+    mute: jest.fn(() => Promise.resolve()),
+    hold: jest.fn(() => Promise.resolve()),
+    unhold: jest.fn(() => Promise.resolve()),
     hangup: jest.fn(() => Promise.resolve()),
     register: jest.fn(() => Promise.resolve()),
     onCallEnded: jest.fn(() => Promise.resolve()),
@@ -66,9 +69,17 @@ describe('Call', () => {
             expect(call.id).toBe(apiCall.sipCallId);
             expect(call.apiId).toBe(apiCall.id);
             expect(call.number).toBe(apiCall.calleeNumber);
-            expect(call.answerTime).toStrictEqual(apiCall.startingTime);
+            expect(call.answerTime).toStrictEqual(undefined);
+            expect(call.creationTime).toStrictEqual(apiCall.startingTime);
             expect(call.isMuted()).toBeTruthy();
-            expect(call.state).toStrictEqual({ [States.ESTABLISHED]: EstablishedStates.MUTED });
+            expect(call.state).toStrictEqual({
+                [States.ESTABLISHED]: {
+                    [EstablishedStates.ONGOING]: {},
+                    [EstablishedSubStates.MUTE]: EstablishedStates.MUTED,
+                    [EstablishedSubStates.HOLD]: EstablishedStates.UN_HELD,
+                },
+            });
+            expect(call.isEstablished()).toBeTruthy();
         });
     });
     describe('accept', () => {
@@ -111,4 +122,42 @@ describe('Call', () => {
             expect(softphone.client.hangup).toHaveBeenCalled();
         });
     });
+    describe('state propagation', () => {
+        it('Should update the call state in events', (done) => {
+            const call = new Call(sipCall, softphone);
+            // @ts-ignore: private method
+            call._sendAction(Actions.MAKE_CALL);
+            call.on('remotelyAccepted', () => {
+                expect(call.state).toStrictEqual({
+                    [States.ESTABLISHED]: {
+                        [EstablishedStates.ONGOING]: {},
+                        [EstablishedSubStates.MUTE]: EstablishedStates.UN_MUTED,
+                        [EstablishedSubStates.HOLD]: EstablishedStates.UN_HELD,
+                    },
+                });
+                done();
+            });
+            call.onAccepted();
+        });
+    });
+    describe('Multiple sub state', () => {
+        it('Should mute and hold a call', () => {
+            const call = new Call(sipCall, softphone);
+            // @ts-ignore: private method
+            call._sendAction(Actions.MAKE_CALL);
+            expect(call.isMuted()).toBeFalsy();
+            expect(call.isHeld()).toBeFalsy();
+            // @ts-ignore: private method
+            call._sendAction(Actions.REMOTLY_ACCEPTED);
+            call.mute();
+            expect(call.isMuted()).toBeTruthy();
+            expect(call.isHeld()).toBeFalsy();
+            call.hold();
+            expect(call.isMuted()).toBeTruthy();
+            expect(call.isHeld()).toBeTruthy();
+            call.resume();
+            expect(call.isMuted()).toBeTruthy();
+            expect(call.isHeld()).toBeFalsy();
+        });
+    });
 });

package/esm/api/auth.js

@@ -49,6 +49,9 @@ export default ((client, baseUrl) => ({
         }
         return client.post(`${baseUrl}/token`, body, headers).then(Session.parse);
     },
+    samlLogOut() {
+        return client.get(`${baseUrl}/saml/logout`, null, null);
+    },
     initiateIdpAuthentication: async (domain, redirectUrl) => {
         const body = {
             domain,

package/esm/checker/checks/webrtc.js

@@ -1,5 +1,5 @@
 /* global MediaStream */
-import { getSipSessionId } from '../../utils/sdp';
+import { getSipCallId } from '../../utils/sdp';
 import WebRTCClient from '../../web-rtc-client';
 export default {
     name: 'WebRTC',
@@ -29,7 +29,7 @@ export default {
         });
         client.on(client.REGISTERED, () => {
             const sipSession = client.call('*10');
-            if (!sipSession || !getSipSessionId(sipSession)) {
+            if (!sipSession || !getSipCallId(sipSession)) {
                 return handleError('Unable to make call through WebRTC');
             }
             setTimeout(() => {

package/esm/domain/CallLog.js

@@ -53,8 +53,7 @@ export default class CallLog {
     }
     static parseNew(plain, session) {
         const callLog = CallLog.parse(plain);
-        // @TODO: FIXME add verification declined vs missed call
-        callLog.newMissedCall = session && session.hasExtension((plain.requested_extension || plain.destination_extension)) && !plain.answered;
+        callLog.newMissedCall = session && plain.destination_user_uuid === session.uuid && !plain.answered;
         return callLog;
     }
     static newFrom(profile) {
@@ -112,7 +111,7 @@ export default class CallLog {
         return this.callDirection === 'inbound';
     }
     isAnOutgoingCall(session) {
-        console.warn(`@wazo/sdk 
+        console.warn(`@wazo/sdk
       CallLog.isAnOutgoingcall(session) method is obsolete.
       Please use CallLog.isOutgoing(session).
     `);

package/esm/domain/Phone/CTIPhone.js

@@ -105,7 +105,7 @@ export default class CTIPhone extends Emitter {
             number: callSession.number,
         });
         try {
-            await CallApi.cancelCall(callSession);
+            await CallApi.cancelCall({ apiId: callSession.getId() });
             if (this.currentCall && callSession.callId === this.currentCall.id) {
                 this.endCurrentCall(callSession);
             }
@@ -127,7 +127,7 @@ export default class CTIPhone extends Emitter {
             callId: callSession.getId(),
             number: callSession.number,
         });
-        await CallApi.cancelCall(callSession);
+        await CallApi.cancelCall({ id: callSession.getId() });
         this.eventEmitter.emit('onCallEnded', callSession);
     }
     async transfer(callSession, number) {

package/esm/domain/Phone/WebRTCPhone.js

@@ -3,7 +3,7 @@ import { Invitation } from 'sip.js/lib/api';
 import CallSession from '../CallSession';
 import Emitter from '../../utils/Emitter';
 import IssueReporter from '../../service/IssueReporter';
-import { getCallDisplayName, getCallNumber, getSipSessionId } from '../../utils/sdp';
+import { getCallDisplayName, getCallNumber, getSipCallId } from '../../utils/sdp';
 import { downgradeToAudio, getStreamFromConstraints, getWebRtcStats, setLocalMediaStream, upgradeToVideo } from '../../utils/webrtc';
 export const ON_USER_AGENT = 'onUserAgent';
 export const ON_REGISTERED = 'onRegistered';
@@ -126,7 +126,7 @@ export default class WebRTCPhone extends Emitter {
     async sendReinvite(callSession, constraints = null, conference = false, audioOnly = false, iceRestart = false) {
         const sipSession = this.findSipSession(callSession);
         logger.info('WebRTC phone - send reinvite', {
-            sessionId: sipSession ? sipSession.id : null,
+            sessionId: getSipCallId(sipSession),
             constraints,
             audioOnly,
             conference,
@@ -280,7 +280,7 @@ export default class WebRTCPhone extends Emitter {
                 content: {
                     type: ON_MESSAGE_TRACK_UPDATED,
                     update: isUpgrade ? 'upgrade' : 'downgrade',
-                    sipCallId: sipSession && getSipSessionId(sipSession),
+                    sipCallId: sipSession && getSipCallId(sipSession),
                     callId: callSession ? callSession.callId : null,
                     number: callSession ? callSession.number : null,
                 },
@@ -288,7 +288,7 @@ export default class WebRTCPhone extends Emitter {
         }, 2500);
     }
     _bindEvents(sipSession) {
-        const sipSessionId = getSipSessionId(sipSession);
+        const sipSessionId = getSipCallId(sipSession);
         if (sipSession instanceof Invitation) {
             // Monkey patch to know when canceled with the CANCEL message
             const onCancel = sipSession._onCancel.bind(sipSession);
@@ -300,7 +300,9 @@ export default class WebRTCPhone extends Emitter {
                 onCancel(message);
                 const elsewhere = message.data.indexOf('cause=26') !== -1 && message.data.indexOf('completed elsewhere') !== -1;
                 const callSession = this._createCallSession(sipSession);
-                delete this.callSessions[sipSessionId];
+                if (sipSessionId) {
+                    delete this.callSessions[sipSessionId];
+                }
                 this.eventEmitter.emit(ON_CALL_CANCELED, callSession, elsewhere);
             };
         }
@@ -308,14 +310,14 @@ export default class WebRTCPhone extends Emitter {
             switch (newState) {
                 case SessionState.Terminating:
                     logger.info('WebRTC phone - call terminating', {
-                        sipId: sipSession.id,
+                        sipId: getSipCallId(sipSession),
                     });
                     this.eventEmitter.emit(ON_CALL_ENDING, this._createCallSession(sipSession));
                     break;
                 case SessionState.Terminated:
                     {
                         logger.info('WebRTC phone - call terminated', {
-                            sipId: sipSession.id,
+                            sipId: getSipCallId(sipSession),
                         });
                         // Should be called before `_onCallTerminated` or the callCount will not decrement...
                         const callSession = this._createCallSession(sipSession);
@@ -342,7 +344,7 @@ export default class WebRTCPhone extends Emitter {
             const [stream] = event.streams;
             const kind = event && event.track && event.track.kind;
             logger.info('on track stream called on the peer connection', {
-                callId: getSipSessionId(sipSession),
+                callId: getSipCallId(sipSession),
                 streamId: stream ? stream.id : null,
                 tracks: stream ? stream.getTracks() : null,
                 kind,
@@ -371,9 +373,9 @@ export default class WebRTCPhone extends Emitter {
         }
         const screenTrack = screenShareStream.getVideoTracks()[0];
         const sipSession = this.currentSipSession;
-        const pc = sipSession && this.client.getPeerConnection(getSipSessionId(sipSession));
+        const pc = sipSession && this.client.getPeerConnection(getSipCallId(sipSession));
         const sender = pc && pc.getSenders && pc.getSenders().find((s) => s && s.track && s.track.kind === 'video');
-        const localStream = sipSession && this.client.getLocalStream(getSipSessionId(sipSession));
+        const localStream = sipSession && this.client.getLocalStream(getSipCallId(sipSession));
         const videoTrack = localStream ? localStream.getTracks().find(track => track.kind === 'video') : null;
         const hadVideo = !!videoTrack;
         // Stop local video tracks
@@ -435,19 +437,19 @@ export default class WebRTCPhone extends Emitter {
     }
     _onScreenSharing(screenStream, sipSession, callSession, hadVideo, desktop) {
         const screenTrack = screenStream.getVideoTracks()[0];
-        const sipSessionId = getSipSessionId(sipSession);
+        const sipSessionId = getSipCallId(sipSession);
         const pc = this.client.getPeerConnection(sipSessionId);
         const sender = pc && pc.getSenders && pc.getSenders().find((s) => s && s.track && s.track.kind === 'video');
         logger.info('WebRTC phone - on screensharing', {
             hadVideo,
-            id: getSipSessionId(sipSession),
+            id: getSipCallId(sipSession),
             screenTrack,
         });
         if (screenTrack) {
             screenTrack.onended = () => {
                 logger.info('WebRTC phone - on screenshare ended', {
                     hadVideo,
-                    id: getSipSessionId(sipSession),
+                    id: getSipCallId(sipSession),
                     screenTrack,
                 });
                 this.eventEmitter.emit(ON_SHARE_SCREEN_ENDING, this._createCallSession(sipSession, callSession));
@@ -467,13 +469,13 @@ export default class WebRTCPhone extends Emitter {
     }
     _onCallAccepted(sipSession, cameraEnabled) {
         logger.info('WebRTC phone - on call accepted', {
-            sipId: sipSession.id,
+            sipId: getSipCallId(sipSession),
             cameraEnabled,
         });
         const callSession = this._createAcceptedCallSession(sipSession, cameraEnabled);
         if (this.currentSipSession && !this.isCurrentCallSipSession(callSession)) {
             logger.info('WebRTC phone - on call accepted with another ongoing call, holding it', {
-                sipId: sipSession.id,
+                sipId: getSipCallId(sipSession),
                 cameraEnabled,
             });
             this.holdSipSession(this.currentSipSession, this.currentCallSession, true);
@@ -481,7 +483,7 @@ export default class WebRTCPhone extends Emitter {
         this.currentSipSession = sipSession;
         this.currentCallSession = callSession;
         this.eventEmitter.emit(ON_TERMINATE_SOUND, callSession, 'call accepted');
-        const sipSessionId = getSipSessionId(sipSession);
+        const sipSessionId = getSipCallId(sipSession);
         if (sipSessionId) {
             this.removeIncomingSessions(sipSessionId);
         }
@@ -540,9 +542,9 @@ export default class WebRTCPhone extends Emitter {
         return this.currentSipSession && this.client.changeSessionVideoInputDevice(id, this.currentSipSession);
     }
     _onCallTerminated(sipSession) {
-        const sipSessionId = getSipSessionId(sipSession);
+        const sipSessionId = getSipCallId(sipSession);
         logger.info('WebRTC phone - on call terminated', {
-            sipId: sipSession.id,
+            sipId: getSipCallId(sipSession),
             sipSessionId,
             ids: Object.keys(this.callSessions),
         });
@@ -562,8 +564,8 @@ export default class WebRTCPhone extends Emitter {
         }
         if (sipSessionId) {
             this.removeIncomingSessions(sipSessionId);
+            delete this.callSessions[sipSessionId];
         }
-        delete this.callSessions[sipSessionId];
         if (isCurrentSession) {
             this.currentSipSession = undefined;
             this.currentCallSession = undefined;
@@ -571,7 +573,7 @@ export default class WebRTCPhone extends Emitter {
         const hasIncomingCallSession = this.hasIncomingCallSession();
         // Re-trigger incoming call event for remaining incoming calls
         logger.info('WebRTC phone - check to re-trigger incoming call', {
-            sipId: sipSession.id,
+            sipId: getSipCallId(sipSession),
             shouldRetrigger,
             hasIncomingCallSession,
         });
@@ -632,7 +634,7 @@ export default class WebRTCPhone extends Emitter {
         if (!this.currentSipSession) {
             return false;
         }
-        return this.currentSipSession && getSipSessionId(this.currentSipSession) === callSession.getId();
+        return this.currentSipSession && getSipCallId(this.currentSipSession) === callSession.getId();
     }
     hasVideo(callSession) {
         return this.client.hasVideo(callSession.getId());
@@ -666,7 +668,7 @@ export default class WebRTCPhone extends Emitter {
             logger.warn('no CallSession to accept.');
             return Promise.resolve(null);
         }
-        const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
+        const sipSessionId = this.getSipCallIdFromCallSession(callSession);
         if (sipSessionId && !(sipSessionId in this.callSessions)) {
             logger.warn('Call session already ended or not found, can\'t accept it.', {
                 sipSessionId,
@@ -686,7 +688,7 @@ export default class WebRTCPhone extends Emitter {
         this.eventEmitter.emit(ON_CALL_ANSWERED, callSession);
         const sipSession = this.client.getSipSession(callSession.getId());
         if (sipSession) {
-            logger.info('accept call, session found', { sipId: sipSession.id });
+            logger.info('accept call, session found', { sipId: getSipCallId(sipSession) });
             return this.client.answer(sipSession, this.allowVideo ? cameraEnabled : false).then(() => {
                 return callSession.sipCallId;
             }).catch(e => {
@@ -698,7 +700,7 @@ export default class WebRTCPhone extends Emitter {
         const error = 'No CallSession found to accept.';
         logger.warn(error, {
             id: callSession ? callSession.getId() : 'n/a',
-            ids: this.client.getSipSessionIds(),
+            ids: this.client.getSipCallIds(),
         });
         if (callSession) {
             this.eventEmitter.emit(ON_CALL_ERROR, new Error(error), callSession);
@@ -721,7 +723,7 @@ export default class WebRTCPhone extends Emitter {
         const sipSession = this.findSipSession(callSession);
         if (sipSession) {
             logger.info('WebRTC rejecting', {
-                sipId: sipSession.id,
+                sipId: getSipCallId(sipSession),
             });
             this.client.hangup(sipSession);
         }
@@ -739,7 +741,7 @@ export default class WebRTCPhone extends Emitter {
         const sipSession = this.findSipSession(callSession);
         if (sipSession) {
             logger.info('WebRTC atxfer', {
-                sipId: sipSession.id,
+                sipId: getSipCallId(sipSession),
             });
             return this.client.atxfer(sipSession);
         }
@@ -765,12 +767,12 @@ export default class WebRTCPhone extends Emitter {
             return new Promise((resolve, reject) => reject(new Error('No session to resume')));
         }
         logger.info('WebRTC resuming', {
-            sipId: sipSession.id,
+            sipId: getSipCallId(sipSession),
         });
         // 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) {
+        if (this.currentSipSession && getSipCallId(this.currentSipSession) !== getSipCallId(sipSession)) {
             logger.info('WebRTC hold call after resume', {
-                id: this.currentSipSession.id,
+                id: getSipCallId(this.currentSipSession),
             });
             this.holdSipSession(this.currentSipSession, this.currentCallSession, withEvent);
         }
@@ -785,10 +787,10 @@ export default class WebRTCPhone extends Emitter {
         if (!sipSession) {
             return new Promise((resolve, reject) => reject(new Error('No session to hold')));
         }
-        const sessionId = getSipSessionId(sipSession);
+        const sessionId = getSipCallId(sipSession);
         const hasVideo = this.client.hasLocalVideo(sessionId);
         logger.info('WebRTC hold sip session', {
-            sipId: sipSession.id,
+            sipId: getSipCallId(sipSession),
             hasVideo,
         });
         // Stop screenshare if needed
@@ -812,7 +814,7 @@ export default class WebRTCPhone extends Emitter {
             return new Promise((resolve, reject) => reject(new Error('No session to unhold')));
         }
         const isConference = !!callSession && callSession.isConference();
-        const sessionId = getSipSessionId(sipSession);
+        const sessionId = getSipCallId(sipSession);
         logger.info('WebRTC unhold sip session', {
             sessionId,
             isConference,
@@ -864,7 +866,7 @@ export default class WebRTCPhone extends Emitter {
             return;
         }
         logger.info('WebRTC muting', {
-            sipId: sipSession.id,
+            sipId: getSipCallId(sipSession),
         });
         this.client.mute(sipSession);
         const newCallSession = this._createCallSession(sipSession, callSession, {
@@ -886,7 +888,7 @@ export default class WebRTCPhone extends Emitter {
             return;
         }
         logger.info('WebRTC unmuting', {
-            sipId: sipSession.id,
+            sipId: getSipCallId(sipSession),
         });
         this.client.unmute(sipSession);
         const newCallSession = this._createCallSession(sipSession, callSession, {
@@ -912,7 +914,7 @@ export default class WebRTCPhone extends Emitter {
             return;
         }
         logger.info('WebRTC turn camera on', {
-            sipId: sipSession.id,
+            sipId: getSipCallId(sipSession),
         });
         this.client.toggleCameraOn(sipSession);
         this.eventEmitter.emit(ON_CAMERA_RESUMED, this._createCameraResumedCallSession(sipSession, callSession));
@@ -923,7 +925,7 @@ export default class WebRTCPhone extends Emitter {
             return;
         }
         logger.info('WebRTC turn camera off', {
-            sipId: sipSession.id,
+            sipId: getSipCallId(sipSession),
         });
         this.client.toggleCameraOff(sipSession);
         this.eventEmitter.emit(ON_CAMERA_DISABLED, this._createCameraDisabledCallSession(sipSession, callSession));
@@ -934,7 +936,7 @@ export default class WebRTCPhone extends Emitter {
             return;
         }
         logger.info('WebRTC send key', {
-            sipId: sipSession.id,
+            sipId: getSipCallId(sipSession),
             tone,
         });
         this.client.sendDTMF(sipSession, tone);
@@ -997,7 +999,7 @@ export default class WebRTCPhone extends Emitter {
             return;
         }
         logger.info('WebRTC transfer', {
-            sipId: sipSession.id,
+            sipId: getSipCallId(sipSession),
             target,
         });
         this.client.transfer(sipSession, target);
@@ -1006,8 +1008,8 @@ export default class WebRTCPhone extends Emitter {
         const sipSession = this.client.getSipSession(source.sipCallId);
         const sipSessionTarget = this.client.getSipSession(destination.sipCallId);
         logger.info('WebRTC indirect transfer', {
-            sipId: sipSession ? sipSession.id : null,
-            target: sipSessionTarget ? sipSessionTarget.id : null,
+            sipId: sipSession ? getSipCallId(sipSession) : null,
+            target: sipSessionTarget ? getSipCallId(sipSessionTarget) : null,
         });
         if (!sipSessionTarget) {
             return Promise.reject();
@@ -1035,9 +1037,9 @@ export default class WebRTCPhone extends Emitter {
             console.error('Call is unknown to the WebRTC phone', callSession ? callSession.sipCallId : null, callSession ? callSession.callId : null);
             return false;
         }
-        const sipSessionId = getSipSessionId(sipSession);
+        const sipSessionId = getSipCallId(sipSession);
         logger.info('WebRTC hangup', {
-            sipId: sipSession.id,
+            sipId: getSipCallId(sipSession),
             sipSessionId,
         });
         await this.client.hangup(sipSession);
@@ -1190,7 +1192,7 @@ export default class WebRTCPhone extends Emitter {
             const autoAnswer = sipSession.request.getHeader('Answer-Mode') === 'Auto';
             const withVideo = this.allowVideo ? wantsToDoVideo : false;
             logger.info('WebRTC invite received', {
-                sipId: sipSession.id,
+                sipId: getSipCallId(sipSession),
                 withVideo,
                 autoAnswer,
             });
@@ -1222,12 +1224,12 @@ export default class WebRTCPhone extends Emitter {
             // if `_createCallSession` is called too soon `hasVideo` will return false because the stream doesn't exists yet
             setTimeout(() => {
                 // Update callSession
-                this._createCallSession(sipSession, this.callSessions[getSipSessionId(sipSession)]);
+                this._createCallSession(sipSession, this.callSessions[getSipCallId(sipSession)]);
                 this.eventEmitter.emit.apply(this.eventEmitter, [this.client.ON_REINVITE, ...args]);
             }, 2000);
         });
         this.client.on(this.client.ACCEPTED, async (sipSession) => {
-            const sessionId = getSipSessionId(sipSession);
+            const sessionId = getSipCallId(sipSession);
             const hasSession = (sessionId in this.callSessions);
             logger.info('WebRTC call accepted', {
                 sipId: sessionId,
@@ -1250,11 +1252,11 @@ export default class WebRTCPhone extends Emitter {
             this.eventEmitter.emit(ON_CALL_ERROR, e);
         });
         this.client.on(this.client.REJECTED, session => {
-            logger.info('WebRTC call rejected', session.id);
+            logger.info('WebRTC call rejected', { id: getSipCallId(session) });
             const callSession = this._createCallSession(session);
             // `_createCallSession` add the CallSession in `this.callSessions`.
             // so we have to remove it or `callCount` will always return 1 even if the call was rejected
-            delete this.callSessions[getSipSessionId(session)];
+            delete this.callSessions[getSipCallId(session)];
             callSession.endTime = new Date();
             this.eventEmitter.emit(ON_CALL_REJECTED, callSession);
         });
@@ -1271,7 +1273,7 @@ export default class WebRTCPhone extends Emitter {
             const callSession = this._createCallSession(session, null, {
                 incoming: false,
                 ringing: true,
-                callId: getSipSessionId(session),
+                callId: getSipCallId(session),
             });
             this.eventEmitter.emit(ON_PROGRESS, callSession, this.audioOutputDeviceId, this.audioOutputVolume);
         });
@@ -1280,7 +1282,7 @@ export default class WebRTCPhone extends Emitter {
             const callSession = this._createCallSession(session, null, {
                 incoming: false,
                 ringing: true,
-                callId: getSipSessionId(session),
+                callId: getSipCallId(session),
             });
             this.eventEmitter.emit(ON_EARLY_MEDIA, callSession, this.audioOutputDeviceId, this.audioOutputVolume);
         });
@@ -1336,12 +1338,12 @@ export default class WebRTCPhone extends Emitter {
             this.eventEmitter.emit(ON_MESSAGE, message);
         });
         this.client.on(this.client.ON_NETWORK_STATS, (session, stats, previousStats) => {
-            const callSession = this.getCallSession(getSipSessionId(session));
+            const callSession = this.getCallSession(getSipCallId(session));
             this.eventEmitter.emit(ON_NETWORK_STATS, callSession, stats, previousStats);
         });
         // Used when upgrading directly in screenshare mode
         this.client.on(this.client.ON_SCREEN_SHARING_REINVITE, (sipSession, response, desktop) => {
-            const sipSessionId = getSipSessionId(sipSession);
+            const sipSessionId = getSipCallId(sipSession);
             const localStream = this.client.getLocalStream(sipSessionId);
             const callSession = this.callSessions[sipSessionId];
             logger.info('Updrading directly in screensharing mode', {
@@ -1353,10 +1355,10 @@ export default class WebRTCPhone extends Emitter {
     }
     // Find a corresponding sipSession from a CallSession
     findSipSession(callSession) {
-        const keys = this.client.getSipSessionIds();
+        const keys = this.client.getSipCallIds();
         const keyIndex = keys.findIndex(sessionId => callSession && callSession.isId(sessionId));
         if (keyIndex === -1) {
-            const currentSipSessionId = this.currentSipSession ? getSipSessionId(this.currentSipSession) : this.client.getSipSessionIds()[0];
+            const currentSipSessionId = this.currentSipSession ? getSipCallId(this.currentSipSession) : this.client.getSipCallIds()[0];
             return currentSipSessionId ? this.client.getSipSession(currentSipSessionId) : null;
         }
         return this.client.getSipSession(keys[keyIndex]);
@@ -1364,12 +1366,12 @@ export default class WebRTCPhone extends Emitter {
     getCallSession(sipSessionId) {
         return this.callSessions[sipSessionId];
     }
-    getSipSessionIdFromCallSession(callSession) {
+    getSipCallIdFromCallSession(callSession) {
         const sipSession = this.findSipSession(callSession);
-        return sipSession ? getSipSessionId(sipSession) : null;
+        return sipSession ? getSipCallId(sipSession) : null;
     }
     _updateCallSession(callSession) {
-        const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
+        const sipSessionId = this.getSipCallIdFromCallSession(callSession);
         if (sipSessionId) {
             this.callSessions[sipSessionId] = callSession;
         }
@@ -1442,7 +1444,7 @@ export default class WebRTCPhone extends Emitter {
         const number = getCallNumber(sipSession);
         const displayName = getCallDisplayName(sipSession);
         const { state } = sipSession || {};
-        const sessionId = getSipSessionId(sipSession);
+        const sessionId = getSipCallId(sipSession);
         fromSession = fromSession || this.callSessions[sessionId];
         const callSession = new CallSession({
             callId: fromSession && fromSession.callId,

package/esm/index.js

@@ -6,7 +6,7 @@ export { default as ServerError } from './domain/ServerError';
 export { default as InvalidStateTransition } from './domain/InvalidStateTransition';
 export { default as InvalidState } from './domain/InvalidState';
 export { default as SFUNotAvailableError } from './domain/SFUNotAvailableError';
-export { default as Call } from './domain/Call';
+export { default as ApiCall } from './domain/Call';
 export { default as CallLog } from './domain/CallLog';
 export { default as Recording } from './domain/Recording';
 export { default as ChatMessage } from './domain/ChatMessage';
@@ -45,6 +45,8 @@ export { default as WazoApiClient } from './api-client';
 export { default as Agent } from './domain/Agent';
 export { default as ExternalApp } from './domain/ExternalApp';
 export { default as SessionDescriptionHandler } from './lib/WazoSessionDescriptionHandler';
+export { default as Call } from './voice/call';
+export { Softphone } from './voice/softphone';
 export * from './types/WebSocketMessage';
 export * from './utils/PhoneNumberUtil';
 export * from './utils/string';

package/esm/service/CallApi.js

@@ -1,4 +1,4 @@
-import Call from '../domain/Call';
+import ApiCall from '../domain/Call';
 import getApiClient from './getApiClient';
 export default class CallApi {
     static async fetchCallLogs(offset, limit) {
@@ -23,13 +23,13 @@ export default class CallApi {
         const lineToUse = line || session.primaryLine();
         return getApiClient().confd.getUserLineSip(session.uuid, lineToUse ? lineToUse.id : '');
     }
-    static async cancelCall(callSession) {
-        return getApiClient().calld.cancelCall(callSession.callId);
+    static async cancelCall(call) {
+        return getApiClient().calld.cancelCall(call.apiId || '');
     }
     static async makeCall(callFromLine, extension, isMobile = false, callbackAllLines = false) {
         const lineId = callFromLine ? callFromLine.id : null;
         const response = await getApiClient().calld.makeCall(extension, isMobile, lineId, callbackAllLines);
-        return Call.parse(response);
+        return ApiCall.parse(response);
     }
     static async relocateCall(callId, line, contactIdentifier) {
         return getApiClient().calld.relocateCall(callId, 'line', line, contactIdentifier);