@wazo/sdk 0.43.6 → 1.0.0-alpha.1

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

@@ -0,0 +1,86 @@
+"use strict";
+var __awaiter = (this && this.__awaiter) || function (thisArg, _arguments, P, generator) {
+    function adopt(value) { return value instanceof P ? value : new P(function (resolve) { resolve(value); }); }
+    return new (P || (P = Promise))(function (resolve, reject) {
+        function fulfilled(value) { try { step(generator.next(value)); } catch (e) { reject(e); } }
+        function rejected(value) { try { step(generator["throw"](value)); } catch (e) { reject(e); } }
+        function step(result) { result.done ? resolve(result.value) : adopt(result.value).then(fulfilled, rejected); }
+        step((generator = generator.apply(thisArg, _arguments || [])).next());
+    });
+};
+var __importDefault = (this && this.__importDefault) || function (mod) {
+    return (mod && mod.__esModule) ? mod : { "default": mod };
+};
+Object.defineProperty(exports, "__esModule", { value: true });
+/* eslint-disable no-underscore-dangle */
+const sip_js_1 = require("sip.js");
+const softphone_1 = __importDefault(require("../../voice/softphone"));
+const call_1 = __importDefault(require("../../voice/call"));
+const call_state_machine_1 = require("../../state-machine/call-state-machine");
+const softphone_state_machine_1 = require("../../state-machine/softphone-state-machine");
+const InvalidStateTransition_1 = __importDefault(require("../../domain/InvalidStateTransition"));
+const uaOptions = {
+    logBuiltinEnabled: false,
+    transportOptions: {
+        traceSip: false,
+        wsServers: 'ws://localhost',
+    },
+};
+jest.mock('../..', () => ({
+    __esModule: true,
+    default: {
+        Auth: {
+            getHost: jest.fn(() => 'my-stack.io'),
+            getSession: jest.fn(() => ({
+                displayName: () => 'John Doe',
+                primaryWebRtcLine: () => ({}),
+            })),
+        },
+    },
+}));
+jest.mock('../../web-rtc-client', () => jest.fn().mockImplementation(() => ({
+    answer: jest.fn(() => Promise.resolve()),
+    hangup: jest.fn(() => Promise.resolve()),
+    register: jest.fn(() => Promise.resolve()),
+    setOnHeartbeatTimeout: jest.fn(() => Promise.resolve()),
+    setOnHeartbeatCallback: jest.fn(() => Promise.resolve()),
+    on: jest.fn(),
+})));
+const ua = new sip_js_1.UserAgent(uaOptions);
+const sipCall = new sip_js_1.Inviter(ua, sip_js_1.UserAgent.makeURI('sip:1234@my.stack.io'));
+softphone_1.default.connect();
+softphone_1.default._sendAction(softphone_state_machine_1.Actions.REGISTER_DONE);
+const invalidStateTransition = (state, action) => new InvalidStateTransition_1.default(`Invalid state transition from ${state} with action ${action}`, state, action);
+describe('Call', () => {
+    describe('accept', () => {
+        it('Should throw an error when accepting a not ringing call', () => __awaiter(void 0, void 0, void 0, function* () {
+            const call = new call_1.default(sipCall, softphone_1.default);
+            yield expect(call.accept()).rejects.toThrowError(invalidStateTransition(call_state_machine_1.States.IDLE, call_state_machine_1.Actions.ACCEPT));
+        }));
+        it.only('Should throw an error when the softphone is not registered', () => __awaiter(void 0, void 0, void 0, function* () {
+            const call = new call_1.default(sipCall, softphone_1.default);
+            call._sendAction(call_state_machine_1.Actions.INCOMING_CALL);
+            softphone_1.default._sendAction(softphone_state_machine_1.Actions.TRANSPORT_CLOSED);
+            yield expect(call.accept()).rejects.toThrowError(invalidStateTransition(call_state_machine_1.States.RINGING, call_state_machine_1.Actions.ACCEPT));
+        }));
+        it('Should not throw an error when accepting a not ringing call', () => __awaiter(void 0, void 0, void 0, function* () {
+            const call = new call_1.default(sipCall, softphone_1.default);
+            call._sendAction(call_state_machine_1.Actions.INCOMING_CALL);
+            yield call.accept();
+            expect(softphone_1.default.client.answer).toHaveBeenCalled();
+        }));
+    });
+    describe('hangup', () => {
+        it('Should throw an error when terminating a non established call', () => __awaiter(void 0, void 0, void 0, function* () {
+            const call = new call_1.default(sipCall, softphone_1.default);
+            yield expect(call.hangup()).rejects.toThrowError(invalidStateTransition(call_state_machine_1.States.IDLE, call_state_machine_1.Actions.HANGUP));
+        }));
+        it('Should not throw an error when terminating an established call', () => __awaiter(void 0, void 0, void 0, function* () {
+            const call = new call_1.default(sipCall, softphone_1.default);
+            call._sendAction(call_state_machine_1.Actions.MAKE_CALL);
+            call._sendAction(call_state_machine_1.Actions.REMOTLY_ACCEPTED);
+            yield call.hangup();
+            expect(softphone_1.default.client.hangup).toHaveBeenCalled();
+        }));
+    });
+});

package/lib/__tests__/__voice__/softphone.test.js

@@ -0,0 +1,80 @@
+"use strict";
+var __awaiter = (this && this.__awaiter) || function (thisArg, _arguments, P, generator) {
+    function adopt(value) { return value instanceof P ? value : new P(function (resolve) { resolve(value); }); }
+    return new (P || (P = Promise))(function (resolve, reject) {
+        function fulfilled(value) { try { step(generator.next(value)); } catch (e) { reject(e); } }
+        function rejected(value) { try { step(generator["throw"](value)); } catch (e) { reject(e); } }
+        function step(result) { result.done ? resolve(result.value) : adopt(result.value).then(fulfilled, rejected); }
+        step((generator = generator.apply(thisArg, _arguments || [])).next());
+    });
+};
+var __importDefault = (this && this.__importDefault) || function (mod) {
+    return (mod && mod.__esModule) ? mod : { "default": mod };
+};
+Object.defineProperty(exports, "__esModule", { value: true });
+const softphone_1 = require("../../voice/softphone");
+const softphone_state_machine_1 = require("../../state-machine/softphone-state-machine");
+const __1 = __importDefault(require("../.."));
+const InvalidStateTransition_1 = __importDefault(require("../../domain/InvalidStateTransition"));
+const server = 'my.stack.io';
+const displayName = 'John Doe';
+const webRtcLine = {};
+const session = {
+    uuid: '1234',
+    displayName: jest.fn(() => displayName),
+    primaryWebRtcLine: jest.fn(() => webRtcLine),
+};
+jest.mock('../..', () => ({
+    __esModule: true,
+    default: {
+        Auth: {
+            getHost: jest.fn(() => server),
+            getSession: jest.fn(() => session),
+        },
+    },
+}));
+jest.mock('../../web-rtc-client', () => jest.fn().mockImplementation(() => ({
+    register: jest.fn(() => Promise.resolve()),
+    unregister: jest.fn(() => Promise.resolve()),
+    setOnHeartbeatTimeout: jest.fn(() => Promise.resolve()),
+    setOnHeartbeatCallback: jest.fn(() => Promise.resolve()),
+    on: jest.fn(),
+    INVITE: 'invite',
+})));
+const invalidStateTransition = (state, action) => new InvalidStateTransition_1.default(`Invalid state transition from ${state} with action ${action}`, state, action);
+describe('Softphone', () => {
+    describe('connect', () => {
+        it('Should retrieve information from Wazo.Auth', () => {
+            const instance = new softphone_1.Softphone();
+            jest.spyOn(instance, 'connectWithCredentials');
+            instance.connect({});
+            expect(__1.default.Auth.getHost).toHaveBeenCalled();
+            expect(__1.default.Auth.getSession).toHaveBeenCalled();
+            expect(instance.connectWithCredentials).toHaveBeenCalledWith(server, webRtcLine, displayName, { userUuid: session.uuid });
+            expect(instance.client.on).toHaveBeenCalledWith('invite', expect.anything());
+        });
+        it('Should throw an error when already connected', () => __awaiter(void 0, void 0, void 0, function* () {
+            const instance = new softphone_1.Softphone();
+            instance.softphoneActor.send({ type: softphone_state_machine_1.Actions.REGISTER });
+            yield expect(instance.connect({})).rejects.toThrowError(invalidStateTransition(softphone_state_machine_1.States.REGISTERING, softphone_state_machine_1.Actions.REGISTER));
+        }));
+        it('Should not allow to register twice', () => __awaiter(void 0, void 0, void 0, function* () {
+            const instance = new softphone_1.Softphone();
+            instance.connect({});
+            yield expect(instance.connect({})).rejects.toThrowError(invalidStateTransition(softphone_state_machine_1.States.REGISTERING, softphone_state_machine_1.Actions.REGISTER));
+        }));
+    });
+    describe('disconnect', () => {
+        it('Should throw an error when not connected', () => __awaiter(void 0, void 0, void 0, function* () {
+            const instance = new softphone_1.Softphone();
+            yield expect(instance.disconnect()).rejects.toThrowError(invalidStateTransition(softphone_state_machine_1.States.UNREGISTERED, softphone_state_machine_1.Actions.UNREGISTER));
+        }));
+        it('Should unregister the client', () => __awaiter(void 0, void 0, void 0, function* () {
+            const instance = new softphone_1.Softphone();
+            instance.connect({});
+            instance.softphoneActor.send({ type: softphone_state_machine_1.Actions.REGISTER_DONE });
+            yield instance.disconnect();
+            expect(instance.client.unregister).toHaveBeenCalled();
+        }));
+    });
+});

package/lib/checker/checks/webrtc.js

@@ -13,6 +13,7 @@ var __importDefault = (this && this.__importDefault) || function (mod) {
 };
 Object.defineProperty(exports, "__esModule", { value: true });
 /* global MediaStream */
+const sdp_1 = require("../../utils/sdp");
 const web_rtc_client_1 = __importDefault(require("../../web-rtc-client"));
 exports.default = {
     name: 'WebRTC',
@@ -42,7 +43,7 @@ exports.default = {
         });
         client.on(client.REGISTERED, () => {
             const sipSession = client.call('*10');
-            if (!sipSession || !client.getSipSessionId(sipSession)) {
+            if (!sipSession || !(0, sdp_1.getSipSessionId)(sipSession)) {
                 return handleError('Unable to make call through WebRTC');
             }
             setTimeout(() => {

package/lib/domain/InvalidState.js

@@ -0,0 +1,9 @@
+"use strict";
+Object.defineProperty(exports, "__esModule", { value: true });
+class InvalidState extends Error {
+    constructor(message, state) {
+        super(message);
+        this.state = state;
+    }
+}
+exports.default = InvalidState;

package/lib/domain/InvalidStateTransition.js

@@ -0,0 +1,10 @@
+"use strict";
+Object.defineProperty(exports, "__esModule", { value: true });
+class InvalidStateTransition extends Error {
+    constructor(message, action, state) {
+        super(message);
+        this.action = action;
+        this.state = state;
+    }
+}
+exports.default = InvalidStateTransition;

package/lib/domain/Phone/WebRTCPhone.js

@@ -18,6 +18,8 @@ const api_1 = require("sip.js/lib/api");
 const CallSession_1 = __importDefault(require("../CallSession"));
 const Emitter_1 = __importDefault(require("../../utils/Emitter"));
 const IssueReporter_1 = __importDefault(require("../../service/IssueReporter"));
+const sdp_1 = require("../../utils/sdp");
+const webrtc_1 = require("../../utils/webrtc");
 exports.ON_USER_AGENT = 'onUserAgent';
 exports.ON_REGISTERED = 'onRegistered';
 exports.ON_UNREGISTERED = 'onUnRegistered';
@@ -220,7 +222,7 @@ class WebRTCPhone extends Emitter_1.default {
             id: callSession ? callSession.getId() : null,
             withMessage,
         });
-        this.client.downgradeToAudio(sipSession);
+        (0, webrtc_1.downgradeToAudio)(sipSession);
         if (callSession) {
             callSession.cameraEnabled = false;
             this._updateCallSession(callSession);
@@ -246,7 +248,7 @@ class WebRTCPhone extends Emitter_1.default {
             const desktop = constraints && constraints.desktop;
             const options = sipSession.sessionDescriptionHandlerOptionsReInvite;
             const wasAudioOnly = options === null || options === void 0 ? void 0 : options.audioOnly;
-            const newStream = yield this.client.upgradeToVideo(sipSession, constraints, isConference);
+            const newStream = yield (0, webrtc_1.upgradeToVideo)(sipSession, constraints, isConference);
             if (callSession) {
                 callSession.cameraEnabled = true;
                 this._updateCallSession(callSession);
@@ -280,16 +282,15 @@ class WebRTCPhone extends Emitter_1.default {
                 content: {
                     type: exports.ON_MESSAGE_TRACK_UPDATED,
                     update: isUpgrade ? 'upgrade' : 'downgrade',
-                    sipCallId: sipSession && this.getSipSessionId(sipSession),
+                    sipCallId: sipSession && (0, sdp_1.getSipSessionId)(sipSession),
                     callId: callSession ? callSession.callId : null,
                     number: callSession ? callSession.number : null,
-                    callerNumber: callSession ? callSession.callerNumber : null,
                 },
             }));
         }, 2500);
     }
     _bindEvents(sipSession) {
-        const sipSessionId = this.getSipSessionId(sipSession);
+        const sipSessionId = (0, sdp_1.getSipSessionId)(sipSession);
         if (sipSession instanceof api_1.Invitation) {
             // Monkey patch to know when canceled with the CANCEL message
             const onCancel = sipSession._onCancel.bind(sipSession);
@@ -343,7 +344,7 @@ class WebRTCPhone extends Emitter_1.default {
             const [stream] = event.streams;
             const kind = event && event.track && event.track.kind;
             logger.info('on track stream called on the peer connection', {
-                callId: this.getSipSessionId(sipSession),
+                callId: (0, sdp_1.getSipSessionId)(sipSession),
                 streamId: stream ? stream.id : null,
                 tracks: stream ? stream.getTracks() : null,
                 kind,
@@ -367,15 +368,15 @@ class WebRTCPhone extends Emitter_1.default {
                 constraints,
                 id: callSession ? callSession.getId() : null,
             });
-            const screenShareStream = yield this.client.getStreamFromConstraints(constraints);
+            const screenShareStream = yield (0, webrtc_1.getStreamFromConstraints)(constraints);
             if (!screenShareStream) {
                 throw new Error(`Can't create media stream for screensharing with: ${JSON.stringify(constraints)}`);
             }
             const screenTrack = screenShareStream.getVideoTracks()[0];
             const sipSession = this.currentSipSession;
-            const pc = sipSession && this.client.getPeerConnection(this.getSipSessionId(sipSession));
+            const pc = sipSession && this.client.getPeerConnection((0, sdp_1.getSipSessionId)(sipSession));
             const sender = pc && pc.getSenders && pc.getSenders().find((s) => s && s.track && s.track.kind === 'video');
-            const localStream = sipSession && this.client.getLocalStream(this.getSipSessionId(sipSession));
+            const localStream = sipSession && this.client.getLocalStream((0, sdp_1.getSipSessionId)(sipSession));
             const videoTrack = localStream ? localStream.getTracks().find(track => track.kind === 'video') : null;
             const hadVideo = !!videoTrack;
             // Stop local video tracks
@@ -440,19 +441,19 @@ class WebRTCPhone extends Emitter_1.default {
     }
     _onScreenSharing(screenStream, sipSession, callSession, hadVideo, desktop) {
         const screenTrack = screenStream.getVideoTracks()[0];
-        const sipSessionId = this.getSipSessionId(sipSession);
+        const sipSessionId = (0, sdp_1.getSipSessionId)(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: this.getSipSessionId(sipSession),
+            id: (0, sdp_1.getSipSessionId)(sipSession),
             screenTrack,
         });
         if (screenTrack) {
             screenTrack.onended = () => {
                 logger.info('WebRTC phone - on screenshare ended', {
                     hadVideo,
-                    id: this.getSipSessionId(sipSession),
+                    id: (0, sdp_1.getSipSessionId)(sipSession),
                     screenTrack,
                 });
                 this.eventEmitter.emit(exports.ON_SHARE_SCREEN_ENDING, this._createCallSession(sipSession, callSession));
@@ -465,7 +466,7 @@ class WebRTCPhone extends Emitter_1.default {
             sipSessionId,
             desktop,
         };
-        this.client.setLocalMediaStream(this.getSipSessionId(sipSession), screenStream);
+        (0, webrtc_1.setLocalMediaStream)(sipSession, screenStream);
         this.eventEmitter.emit(exports.ON_SHARE_SCREEN_STARTED, this._createCallSession(sipSession, callSession, {
             screensharing: true,
         }), screenStream);
@@ -486,7 +487,7 @@ class WebRTCPhone extends Emitter_1.default {
         this.currentSipSession = sipSession;
         this.currentCallSession = callSession;
         this.eventEmitter.emit(exports.ON_TERMINATE_SOUND, callSession, 'call accepted');
-        const sipSessionId = this.getSipSessionId(sipSession);
+        const sipSessionId = (0, sdp_1.getSipSessionId)(sipSession);
         if (sipSessionId) {
             this.removeIncomingSessions(sipSessionId);
         }
@@ -549,7 +550,7 @@ class WebRTCPhone extends Emitter_1.default {
         return this.currentSipSession && this.client.changeSessionVideoInputDevice(id, this.currentSipSession);
     }
     _onCallTerminated(sipSession) {
-        const sipSessionId = this.getSipSessionId(sipSession);
+        const sipSessionId = (0, sdp_1.getSipSessionId)(sipSession);
         logger.info('WebRTC phone - on call terminated', {
             sipId: sipSession.id,
             sipSessionId,
@@ -641,7 +642,7 @@ class WebRTCPhone extends Emitter_1.default {
         if (!this.currentSipSession) {
             return false;
         }
-        return this.currentSipSession && this.getSipSessionId(this.currentSipSession) === callSession.getId();
+        return this.currentSipSession && (0, sdp_1.getSipSessionId)(this.currentSipSession) === callSession.getId();
     }
     hasVideo(callSession) {
         return this.client.hasVideo(callSession.getId());
@@ -799,7 +800,7 @@ class WebRTCPhone extends Emitter_1.default {
             if (!sipSession) {
                 return new Promise((resolve, reject) => reject(new Error('No session to hold')));
             }
-            const sessionId = this.getSipSessionId(sipSession);
+            const sessionId = (0, sdp_1.getSipSessionId)(sipSession);
             const hasVideo = this.client.hasLocalVideo(sessionId);
             logger.info('WebRTC hold sip session', {
                 sipId: sipSession.id,
@@ -812,7 +813,7 @@ class WebRTCPhone extends Emitter_1.default {
             }
             // Downgrade to audio if needed
             if (hasVideo) {
-                yield this.client.downgradeToAudio(sipSession);
+                yield (0, webrtc_1.downgradeToAudio)(sipSession);
             }
             const isConference = !!callSession && callSession.isConference();
             const promise = this.client.hold(sipSession, isConference, hasVideo);
@@ -828,7 +829,7 @@ class WebRTCPhone extends Emitter_1.default {
                 return new Promise((resolve, reject) => reject(new Error('No session to unhold')));
             }
             const isConference = !!callSession && callSession.isConference();
-            const sessionId = this.getSipSessionId(sipSession);
+            const sessionId = (0, sdp_1.getSipSessionId)(sipSession);
             logger.info('WebRTC unhold sip session', {
                 sessionId,
                 isConference,
@@ -844,7 +845,7 @@ class WebRTCPhone extends Emitter_1.default {
                     screen: wasScreensharing,
                     desktop: wasDesktop,
                 };
-                yield this.client.upgradeToVideo(sipSession, constraints, isConference);
+                yield (0, webrtc_1.upgradeToVideo)(sipSession, constraints, isConference);
             }
             const onScreenSharing = (stream) => {
                 const hadVideo = this.lastScreenShare && this.lastScreenShare.hadVideo;
@@ -1057,7 +1058,7 @@ class WebRTCPhone extends Emitter_1.default {
                 console.error('Call is unknown to the WebRTC phone', callSession ? callSession.sipCallId : null, callSession ? callSession.callId : null);
                 return false;
             }
-            const sipSessionId = this.getSipSessionId(sipSession);
+            const sipSessionId = (0, sdp_1.getSipSessionId)(sipSession);
             logger.info('WebRTC hangup', {
                 sipId: sipSession.id,
                 sipSessionId,
@@ -1077,7 +1078,7 @@ class WebRTCPhone extends Emitter_1.default {
     getStats(callSession) {
         return __awaiter(this, void 0, void 0, function* () {
             const sipSession = this.findSipSession(callSession);
-            return sipSession ? this.client.getStats(sipSession) : null;
+            return sipSession ? (0, webrtc_1.getWebRtcStats)(sipSession) : null;
         });
     }
     startNetworkMonitoring(callSession, interval = 1000) {
@@ -1249,12 +1250,12 @@ class WebRTCPhone extends Emitter_1.default {
             // 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[this.getSipSessionId(sipSession)]);
+                this._createCallSession(sipSession, this.callSessions[(0, sdp_1.getSipSessionId)(sipSession)]);
                 this.eventEmitter.emit.apply(this.eventEmitter, [this.client.ON_REINVITE, ...args]);
             }, 2000);
         });
         this.client.on(this.client.ACCEPTED, (sipSession) => __awaiter(this, void 0, void 0, function* () {
-            const sessionId = this.getSipSessionId(sipSession);
+            const sessionId = (0, sdp_1.getSipSessionId)(sipSession);
             const hasSession = (sessionId in this.callSessions);
             logger.info('WebRTC call accepted', {
                 sipId: sessionId,
@@ -1281,7 +1282,7 @@ class WebRTCPhone extends Emitter_1.default {
             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[this.getSipSessionId(session)];
+            delete this.callSessions[(0, sdp_1.getSipSessionId)(session)];
             callSession.endTime = new Date();
             this.eventEmitter.emit(exports.ON_CALL_REJECTED, callSession);
         });
@@ -1298,7 +1299,7 @@ class WebRTCPhone extends Emitter_1.default {
             const callSession = this._createCallSession(session, null, {
                 incoming: false,
                 ringing: true,
-                callId: this.getSipSessionId(session),
+                callId: (0, sdp_1.getSipSessionId)(session),
             });
             this.eventEmitter.emit(exports.ON_PROGRESS, callSession, this.audioOutputDeviceId, this.audioOutputVolume);
         });
@@ -1307,7 +1308,7 @@ class WebRTCPhone extends Emitter_1.default {
             const callSession = this._createCallSession(session, null, {
                 incoming: false,
                 ringing: true,
-                callId: this.getSipSessionId(session),
+                callId: (0, sdp_1.getSipSessionId)(session),
             });
             this.eventEmitter.emit(exports.ON_EARLY_MEDIA, callSession, this.audioOutputDeviceId, this.audioOutputVolume);
         });
@@ -1364,12 +1365,12 @@ class WebRTCPhone extends Emitter_1.default {
             this.eventEmitter.emit(exports.ON_MESSAGE, message);
         });
         this.client.on(this.client.ON_NETWORK_STATS, (session, stats, previousStats) => {
-            const callSession = this.getCallSession(this.getSipSessionId(session));
+            const callSession = this.getCallSession((0, sdp_1.getSipSessionId)(session));
             this.eventEmitter.emit(exports.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 = this.getSipSessionId(sipSession);
+            const sipSessionId = (0, sdp_1.getSipSessionId)(sipSession);
             const localStream = this.client.getLocalStream(sipSessionId);
             const callSession = this.callSessions[sipSessionId];
             logger.info('Updrading directly in screensharing mode', {
@@ -1384,7 +1385,7 @@ class WebRTCPhone extends Emitter_1.default {
         const keys = this.client.getSipSessionIds();
         const keyIndex = keys.findIndex(sessionId => callSession && callSession.isId(sessionId));
         if (keyIndex === -1) {
-            const currentSipSessionId = this.currentSipSession ? this.getSipSessionId(this.currentSipSession) : this.client.getSipSessionIds()[0];
+            const currentSipSessionId = this.currentSipSession ? (0, sdp_1.getSipSessionId)(this.currentSipSession) : this.client.getSipSessionIds()[0];
             return currentSipSessionId ? this.client.getSipSession(currentSipSessionId) : null;
         }
         return this.client.getSipSession(keys[keyIndex]);
@@ -1392,12 +1393,9 @@ class WebRTCPhone extends Emitter_1.default {
     getCallSession(sipSessionId) {
         return this.callSessions[sipSessionId];
     }
-    getSipSessionId(sipSession) {
-        return this.client.getSipSessionId(sipSession);
-    }
     getSipSessionIdFromCallSession(callSession) {
         const sipSession = this.findSipSession(callSession);
-        return sipSession ? this.getSipSessionId(sipSession) : null;
+        return sipSession ? (0, sdp_1.getSipSessionId)(sipSession) : null;
     }
     _updateCallSession(callSession) {
         const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
@@ -1475,7 +1473,7 @@ class WebRTCPhone extends Emitter_1.default {
         // @ts-ignore: private
         const number = identity ? identity.uri._normal.user : null;
         const { state, } = sipSession || {};
-        const sessionId = this.getSipSessionId(sipSession);
+        const sessionId = (0, sdp_1.getSipSessionId)(sipSession);
         fromSession = fromSession || this.callSessions[sessionId];
         const callSession = new CallSession_1.default(Object.assign({ callId: fromSession && fromSession.callId, sipCallId: sessionId, sipStatus: state, displayName: identity ? identity.displayName || number : number, 
             // @ts-ignore: date / number

package/lib/index.js

@@ -17,8 +17,8 @@ var __importDefault = (this && this.__importDefault) || function (mod) {
     return (mod && mod.__esModule) ? mod : { "default": mod };
 };
 Object.defineProperty(exports, "__esModule", { value: true });
-exports.WazoApiClient = exports.WebRTCClient = exports.SOCKET_EVENTS = exports.WebSocketClient = exports.AdHocAPIConference = exports.CallApi = exports.setIsMobile = exports.setOnRefreshTokenError = exports.setOnRefreshToken = exports.setCurrentServer = exports.setApiToken = exports.getApiClient = exports.SipLine = exports.MeetingAuthorization = exports.MeetingStatus = exports.ApiRequester = exports.Checker = exports.DebugDevice = exports.Meeting = exports.CTIPhone = exports.WebRTCPhone = exports.CallerID = exports.SwitchboardCall = exports.IndirectTransfer = exports.CallSession = exports.Room = exports.Relocation = exports.Voicemail = exports.Session = exports.LINE_STATE = exports.PROFILE_STATE = exports.Profile = exports.NotificationOptions = exports.Line = exports.Incall = exports.FORWARD_KEYS = exports.ForwardOption = exports.Features = exports.COUNTRIES = exports.Contact = exports.ChatRoom = exports.ChatMessage = exports.Recording = exports.CallLog = exports.Call = exports.SFUNotAvailableError = exports.ServerError = exports.BadResponse = exports.IssueReporter = exports.Emitter = void 0;
-exports.SessionDescriptionHandler = exports.ExternalApp = exports.Agent = void 0;
+exports.SOCKET_EVENTS = exports.WebSocketClient = exports.AdHocAPIConference = exports.CallApi = exports.setIsMobile = exports.setOnRefreshTokenError = exports.setOnRefreshToken = exports.setCurrentServer = exports.setApiToken = exports.getApiClient = exports.SipLine = exports.MeetingAuthorization = exports.MeetingStatus = exports.ApiRequester = exports.Checker = exports.DebugDevice = exports.Meeting = exports.CTIPhone = exports.WebRTCPhone = exports.CallerID = exports.SwitchboardCall = exports.IndirectTransfer = exports.CallSession = exports.Room = exports.Relocation = exports.Voicemail = exports.Session = exports.LINE_STATE = exports.PROFILE_STATE = exports.Profile = exports.NotificationOptions = exports.Line = exports.Incall = exports.FORWARD_KEYS = exports.ForwardOption = exports.Features = exports.COUNTRIES = exports.Contact = exports.ChatRoom = exports.ChatMessage = exports.Recording = exports.CallLog = exports.Call = exports.SFUNotAvailableError = exports.InvalidState = exports.InvalidStateTransition = exports.ServerError = exports.BadResponse = exports.IssueReporter = exports.Emitter = void 0;
+exports.SessionDescriptionHandler = exports.ExternalApp = exports.Agent = exports.WazoApiClient = exports.WebRTCClient = void 0;
 const index_1 = __importDefault(require("./simple/index"));
 var Emitter_1 = require("./utils/Emitter");
 Object.defineProperty(exports, "Emitter", { enumerable: true, get: function () { return __importDefault(Emitter_1).default; } });
@@ -28,6 +28,10 @@ var BadResponse_1 = require("./domain/BadResponse");
 Object.defineProperty(exports, "BadResponse", { enumerable: true, get: function () { return __importDefault(BadResponse_1).default; } });
 var ServerError_1 = require("./domain/ServerError");
 Object.defineProperty(exports, "ServerError", { enumerable: true, get: function () { return __importDefault(ServerError_1).default; } });
+var InvalidStateTransition_1 = require("./domain/InvalidStateTransition");
+Object.defineProperty(exports, "InvalidStateTransition", { enumerable: true, get: function () { return __importDefault(InvalidStateTransition_1).default; } });
+var InvalidState_1 = require("./domain/InvalidState");
+Object.defineProperty(exports, "InvalidState", { enumerable: true, get: function () { return __importDefault(InvalidState_1).default; } });
 var SFUNotAvailableError_1 = require("./domain/SFUNotAvailableError");
 Object.defineProperty(exports, "SFUNotAvailableError", { enumerable: true, get: function () { return __importDefault(SFUNotAvailableError_1).default; } });
 var Call_1 = require("./domain/Call");

package/lib/simple/Phone.js

@@ -45,6 +45,7 @@ const IssueReporter_1 = __importDefault(require("../service/IssueReporter"));
 const Emitter_1 = __importDefault(require("../utils/Emitter"));
 const index_1 = __importDefault(require("./index"));
 const SFUNotAvailableError_1 = __importDefault(require("../domain/SFUNotAvailableError"));
+const sdp_1 = require("../utils/sdp");
 const logger = IssueReporter_1.default.loggerFor('simple-phone');
 const sipLogger = IssueReporter_1.default.loggerFor('sip.js');
 const protocolLogger = IssueReporter_1.default.loggerFor('sip');
@@ -239,7 +240,7 @@ class Phone extends Emitter_1.default {
         if (!sipSession || !this.phone) {
             return null;
         }
-        return this.phone.getSipSessionId(sipSession);
+        return (0, sdp_1.getSipSessionId)(sipSession);
     }
     sendMessage(body, sipSession, contentType = 'text/plain') {
         const toSipSession = sipSession || this.getCurrentSipSession();

package/lib/simple/index.js

@@ -68,6 +68,8 @@ const IssueReporter_1 = __importDefault(require("../service/IssueReporter"));
 const Features_1 = __importDefault(require("../domain/Features"));
 const Meeting_1 = __importDefault(require("../domain/Meeting"));
 const Checker_1 = __importDefault(require("../checker/Checker"));
+const softphone_1 = __importDefault(require("../voice/softphone"));
+const call_1 = __importDefault(require("../voice/call"));
 const Wazo = {
     Auth: Auth_1.default,
     Phone: Phone_1.default,
@@ -87,6 +89,10 @@ const Wazo = {
     loggerFor: IssueReporter_1.default.loggerFor.bind(IssueReporter_1.default),
     Features: Features_1.default,
     Checker: Checker_1.default,
+    Voice: {
+        Softphone: softphone_1.default,
+        Call: call_1.default,
+    },
     // Domain
     domain: {
         BadResponse: BadResponse_1.default,

package/lib/simple/room/Room.js

@@ -19,6 +19,7 @@ const Emitter_1 = __importDefault(require("../../utils/Emitter"));
 const index_1 = __importDefault(require("../index"));
 const RemoteParticipant_1 = __importDefault(require("./RemoteParticipant"));
 const IssueReporter_1 = __importDefault(require("../../service/IssueReporter"));
+const sdp_1 = require("../../utils/sdp");
 exports.SIGNAL_TYPE_PARTICIPANT_UPDATE = 'signal/PARTICIPANT_UPDATE';
 exports.SIGNAL_TYPE_PARTICIPANT_REQUEST = 'signal/PARTICIPANT_REQUEST';
 const logger = IssueReporter_1.default.loggerFor('sdk-room');
@@ -367,12 +368,12 @@ class Room extends Emitter_1.default {
         // This allow to map msid with the non parsed (eg without the `stripVideo` modifier) SDP
         index_1.default.Phone.phone.on(index_1.default.Phone.phone.client.ON_REINVITE, this._boundOnReinvite);
         this.on(this.ON_AUDIO_STREAM, (stream) => __awaiter(this, void 0, void 0, function* () {
-            var _b, _c, _d;
+            var _b, _c;
             logger.info('on room audio stream');
             this.audioStream = stream;
             if (!this.roomAudioElement) {
-                const sessionId = (_b = index_1.default.Phone.phone) === null || _b === void 0 ? void 0 : _b.getSipSessionId((_c = index_1.default.Phone.phone) === null || _c === void 0 ? void 0 : _c.currentSipSession);
-                this.roomAudioElement = (yield ((_d = index_1.default.Phone.phone) === null || _d === void 0 ? void 0 : _d.createAudioElementFor(sessionId)));
+                const sessionId = (0, sdp_1.getSipSessionId)((_b = index_1.default.Phone.phone) === null || _b === void 0 ? void 0 : _b.currentSipSession);
+                this.roomAudioElement = (yield ((_c = index_1.default.Phone.phone) === null || _c === void 0 ? void 0 : _c.createAudioElementFor(sessionId)));
                 this.roomAudioElement.srcObject = stream;
             }
             else {

package/lib/simple/room/SipRoom.js

@@ -15,6 +15,7 @@ Object.defineProperty(exports, "__esModule", { value: true });
 const IssueReporter_1 = __importDefault(require("../../service/IssueReporter"));
 const index_1 = __importDefault(require("../index"));
 const Room_1 = __importDefault(require("./Room"));
+const sdp_1 = require("../../utils/sdp");
 const logger = IssueReporter_1.default.loggerFor('sdk-sip-room');
 class SipRoom extends Room_1.default {
     static connect(_a) {
@@ -135,7 +136,7 @@ class SipRoom extends Room_1.default {
     }
     _getCurrentSipCallIs() {
         var _a;
-        return index_1.default.Phone.getSipSessionId((_a = index_1.default.Phone.phone) === null || _a === void 0 ? void 0 : _a.currentSipSession);
+        return (0, sdp_1.getSipSessionId)((_a = index_1.default.Phone.phone) === null || _a === void 0 ? void 0 : _a.currentSipSession);
     }
 }
 exports.default = SipRoom;