@wazo/sdk 0.39.0-beta.9 → 0.39.0

package/esm/web-rtc-client.js

@@ -55,6 +55,8 @@ export const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAG
 export class CanceledCallError extends Error {
 }
 const MAX_REGISTER_TRIES = 5;
+// setting a 24hr timeout and letting the backend define the actual value
+const NO_ANSWER_TIMEOUT = 60 * 60 * 24; // in seconds
 export default class WebRTCClient extends Emitter {
     clientId;
     config;
@@ -112,7 +114,7 @@ export default class WebRTCClient extends Emitter {
         }
         return [];
     }
-    constructor(config, session, uaConfigOverrides = undefined) {
+    constructor(config, session, uaConfigOverrides) {
         super();
         // For debug purpose
         this.clientId = Math.ceil(Math.random() * 1000);
@@ -173,10 +175,10 @@ export default class WebRTCClient extends Emitter {
         this.video = media.video;
         this.audio = media.audio;
     }
-    createUserAgent(configOverrides) {
-        const webRTCConfiguration = this._createWebRTCConfiguration(configOverrides);
-        logger.info('sdk webrtc, creating UA', { webRTCConfiguration, clientId: this.clientId });
-        webRTCConfiguration.delegate = {
+    createUserAgent(uaConfigOverrides) {
+        const uaOptions = this._createUaOptions(uaConfigOverrides);
+        logger.info('sdk webrtc, creating UA', { uaOptions, clientId: this.clientId });
+        uaOptions.delegate = {
             onConnect: this.onConnect.bind(this),
             onDisconnect: this.onDisconnect.bind(this),
             onInvite: (invitation) => {
@@ -192,7 +194,8 @@ export default class WebRTCClient extends Emitter {
                 this.eventEmitter.emit(INVITE, invitation, this.sessionWantsToDoVideo(invitation), shouldAutoAnswer);
             },
         };
-        const ua = new UserAgent(webRTCConfiguration);
+        const ua = new UserAgent(uaOptions);
+        ua.start();
         if (ua.transport && ua.transport.connectPromise) {
             ua.transport.connectPromise.catch((e) => {
                 logger.warn('Transport connect error', e);
@@ -372,16 +375,27 @@ export default class WebRTCClient extends Emitter {
             userAgent: !!this.userAgent,
             registerer: !!this.registerer,
         });
-        if (!this.registerer) {
-            return Promise.resolve();
-        }
         try {
-            return this.registerer.unregister().then(() => {
-                logger.info('sdk webrtc unregistered', { clientId: this.clientId });
-                this._cleanupRegister();
-            }).catch(e => {
-                logger.error('sdk webrtc unregistering, promise error', e);
-                this._cleanupRegister();
+            return new Promise((resolve, reject) => {
+                if (!this.registerer) {
+                    return resolve();
+                }
+                const onRegisterStateChange = (state) => {
+                    if (state === RegistererState.Unregistered) {
+                        logger.info('sdk webrtc unregistered', { clientId: this.clientId });
+                        if (this.registerer) {
+                            this.registerer.stateChange.addListener(onRegisterStateChange);
+                        }
+                        this._cleanupRegister();
+                        resolve();
+                    }
+                };
+                this.registerer.stateChange.addListener(onRegisterStateChange);
+                this.registerer.unregister().then().catch(e => {
+                    logger.error('sdk webrtc unregistering, promise error', e);
+                    this._cleanupRegister();
+                    reject();
+                });
             });
         }
         catch (e) {
@@ -441,9 +455,7 @@ export default class WebRTCClient extends Emitter {
         const inviteOptions = {
             requestDelegate: {
                 onAccept: (response) => {
-                    // @ts-ignore
                     if (session.sessionDescriptionHandler?.peerConnection) {
-                        // @ts-ignore
                         session.sessionDescriptionHandler.peerConnection.sfu = conference;
                     }
                     // @ts-ignore
@@ -712,9 +724,9 @@ export default class WebRTCClient extends Emitter {
             hasVideo,
             isConference,
         };
-        if (isConference) {
-            this.mute(session);
-        }
+        // We should also mute the call, because when holding a call during a voicemail, the audio is still sent with the
+        // `sendonly` direction
+        this.mute(session);
         session.sessionDescriptionHandlerOptionsReInvite = {
             // @ts-ignore
             hold: true,
@@ -754,9 +766,7 @@ export default class WebRTCClient extends Emitter {
         if (session.pendingReinvite) {
             return Promise.resolve();
         }
-        if (isConference) {
-            this.unmute(session);
-        }
+        this.unmute(session);
         delete this.heldSessions[this.getSipSessionId(session)];
         session.sessionDescriptionHandlerOptionsReInvite = {
             // @ts-ignore
@@ -1117,9 +1127,9 @@ export default class WebRTCClient extends Emitter {
         });
         // Release old video stream
         if (localStream) {
-            localStream.getVideoTracks().forEach(track => {
-                track.stop();
-            });
+            localStream.getVideoTracks()
+                .filter((track) => track.enabled)
+                .forEach((track) => track.stop());
         }
         const constraints = {
             video: id ? {
@@ -1439,7 +1449,7 @@ export default class WebRTCClient extends Emitter {
         if (!pc || !remoteStream) {
             return;
         }
-        remoteStream.getTracks().forEach(track => {
+        remoteStream.getVideoTracks().forEach(track => {
             remoteStream.removeTrack(track);
         });
         if (pc.getReceivers) {
@@ -1494,6 +1504,7 @@ export default class WebRTCClient extends Emitter {
     createAudioElementFor(sessionId) {
         const audio = document.createElement('audio');
         audio.setAttribute('id', `audio-${sessionId}`);
+        logger.info('creating audio element', { sessionId, audioOutputDeviceId: this.audioOutputDeviceId });
         if (audio.setSinkId && this.audioOutputDeviceId) {
             audio.setSinkId(this.audioOutputDeviceId);
         }
@@ -1620,7 +1631,7 @@ export default class WebRTCClient extends Emitter {
             ...config,
         }));
     }
-    _createWebRTCConfiguration(configOverrides = {}) {
+    _createUaOptions(uaOptionsOverrides = {}) {
         let { host, } = this.config;
         let port = this.config.port || 443;
         if (this.config.websocketSip) {
@@ -1628,11 +1639,12 @@ export default class WebRTCClient extends Emitter {
             [host] = webSocketSip;
             port = Number(webSocketSip[1]);
         }
-        const config = {
+        const { userUuid } = this.config;
+        const uaOptions = {
+            noAnswerTimeout: NO_ANSWER_TIMEOUT,
             authorizationUsername: this.config.authorizationUser,
             authorizationPassword: this.config.password,
             displayName: this.config.displayName,
-            autoStart: true,
             hackIpInContact: true,
             contactParams: {
                 transport: 'wss',
@@ -1658,8 +1670,8 @@ export default class WebRTCClient extends Emitter {
                 return new WazoSessionDescriptionHandler(uaLogger, wazoMediaStreamFactory, sdhOptions, isWeb, session);
             },
             transportOptions: {
-                traceSip: configOverrides?.traceSip || false,
-                wsServers: `wss://${host}:${port}/api/asterisk/ws`,
+                traceSip: uaOptionsOverrides?.traceSip || false,
+                wsServers: `wss://${host}:${port}/api/asterisk/ws${userUuid ? `?userUuid=${userUuid}` : ''}`,
             },
             sessionDescriptionHandlerFactoryOptions: {
                 modifiers: [replaceLocalIpModifier],
@@ -1675,19 +1687,21 @@ export default class WebRTCClient extends Emitter {
                         rtcpMuxPolicy: 'require',
                         iceServers: WebRTCClient.getIceServers(this.config.host),
                         ...this._getRtcOptions(),
-                        ...(configOverrides?.peerConnectionOptions || {}),
+                        ...(uaOptionsOverrides?.peerConnectionOptions || {}),
                     },
                 },
                 // Configuration used in SDH to create the PeerConnection
                 peerConnectionConfiguration: {
                     rtcpMuxPolicy: 'require',
                     iceServers: WebRTCClient.getIceServers(this.config.host),
-                    ...(configOverrides?.peerConnectionOptions || {}),
+                    ...(uaOptionsOverrides?.peerConnectionOptions || {}),
                 },
             },
         };
-        return { ...config,
-            ...configOverrides,
+        delete uaOptionsOverrides?.traceSip;
+        return {
+            ...uaOptions,
+            ...uaOptionsOverrides,
         };
     }
     // eslint-disable-next-line no-unused-vars
@@ -1846,6 +1860,19 @@ export default class WebRTCClient extends Emitter {
         // @ts-ignore
         const earlyStream = sipSession && sipSession.sessionDescriptionHandler ? sipSession.sessionDescriptionHandler.remoteMediaStream : null;
         const stream = newStream || removeStream || earlyStream;
+        const shouldPause = audioElement.currentTime > 0 && !audioElement.paused && !audioElement.ended && audioElement.readyState > 2;
+        logger.info('setting up media', {
+            sessionId,
+            streams: {
+                new: !!newStream,
+                remove: !!removeStream,
+                early: !!earlyStream,
+            },
+            isWeb: this._isWeb(),
+            hasAudio: this._hasAudio(),
+            isConference,
+            shouldPause,
+        });
         if (!this._isWeb() || !stream) {
             return;
         }
@@ -1853,7 +1880,7 @@ export default class WebRTCClient extends Emitter {
             // Conference local stream is handled in Room
             return;
         }
-        if (audioElement.currentTime > 0 && !audioElement.paused && !audioElement.ended && audioElement.readyState > 2) {
+        if (shouldPause) {
             audioElement.pause();
         }
         audioElement.srcObject = stream;
@@ -1888,7 +1915,7 @@ export default class WebRTCClient extends Emitter {
         }
     }
     _cleanupStream(stream) {
-        stream.getTracks().forEach(track => track.stop());
+        stream.getTracks().filter(track => track.enabled).forEach(track => track.stop());
     }
     _toggleAudio(session, muteAudio) {
         // @ts-ignore
@@ -2106,8 +2133,10 @@ export default class WebRTCClient extends Emitter {
         networkStats.totalTransportReceived = transportReceived;
         networkStats.transportReceived = transportReceived - lastTransportReceived;
         networkStats.bandwidth = networkStats.audioBytesSent + networkStats.audioBytesReceived + networkStats.videoBytesSent + networkStats.videoBytesReceived + networkStats.transportReceived + networkStats.transportSent;
-        this.eventEmitter.emit(ON_NETWORK_STATS, session, networkStats, this.sessionNetworkStats[sessionId]);
-        this.sessionNetworkStats[sessionId].push(networkStats);
+        if (this.sessionNetworkStats[sessionId]) {
+            this.eventEmitter.emit(ON_NETWORK_STATS, session, networkStats, this.sessionNetworkStats[sessionId]);
+            this.sessionNetworkStats[sessionId].push(networkStats);
+        }
     }
     async _accept(session, options = {}) {
         if (session.state === SessionState.Terminated || session.state === SessionState.Terminating) {

package/esm/websocket-client.js

@@ -84,6 +84,7 @@ const logger = IssueReporter.loggerFor('wazo-ws');
 const messageLogger = IssueReporter.loggerFor('wazo-ws-message');
 class WebSocketClient extends Emitter {
     initialized;
+    session;
     host;
     version;
     token;
@@ -103,14 +104,16 @@ class WebSocketClient extends Emitter {
      * @param events
      * @param version
      * @param heartbeat
+     * @param session
      * @param options @see https://github.com/pladaria/reconnecting-websocket#available-options
      */
-    constructor({ host, token, version, events, heartbeat, }, options = {
+    constructor({ host, token, version, events, heartbeat, session, }, options = {
         host,
         token,
         version: 1,
         events: [],
         heartbeat: {},
+        session: null,
     }) {
         super();
         this.initialized = false;
@@ -120,6 +123,7 @@ class WebSocketClient extends Emitter {
         this.events = events;
         this.options = options;
         this.version = version;
+        this.session = session;
         this._boundOnHeartbeat = this._onHeartbeat.bind(this);
         const { delay, timeout, max, } = heartbeat || {};
         this.heartbeat = new Heartbeat(delay, timeout, max);
@@ -346,7 +350,10 @@ class WebSocketClient extends Emitter {
         if (!this.host || !this.token) {
             this.close();
         }
-        const url = `wss://${this.host || ''}/api/websocketd/?token=${this.token || ''}&version=${this.version}`;
+        let url = `wss://${this.host || ''}/api/websocketd/?token=${this.token || ''}&version=${this.version}`;
+        if (this.session) {
+            url += `&userUuid=${this.session.uuid}`;
+        }
         logger.info('Wazo WS url computed to reconnect', {
             url,
         });

package/lib/__tests__/api-client.test.js

@@ -12,11 +12,13 @@ var __importDefault = (this && this.__importDefault) || function (mod) {
     return (mod && mod.__esModule) ? mod : { "default": mod };
 };
 Object.defineProperty(exports, "__esModule", { value: true });
+const node_abort_controller_1 = require("node-abort-controller");
 const js_base64_1 = require("js-base64");
 const api_client_1 = __importDefault(require("../api-client"));
 const ServerError_1 = __importDefault(require("../domain/ServerError"));
 const BadResponse_1 = __importDefault(require("../domain/BadResponse"));
 const Session_1 = __importDefault(require("../domain/Session"));
+const controller = new node_abort_controller_1.AbortController();
 const mockedResponse = {
     data: {
         token: 1,
@@ -79,7 +81,7 @@ const server = 'localhost';
 const authVersion = '0.1';
 const username = 'wazo';
 const password = 'zowa';
-jest.mock('node-fetch/lib/index', () => { });
+jest.mock('node-fetch/src/index', () => { });
 const token = '1234';
 const client = new api_client_1.default({
     server,
@@ -112,6 +114,7 @@ describe('With correct API results', () => {
             expect(global.fetch).toBeCalledWith(`https://${server}/api/auth/${authVersion}/token`, {
                 method: 'post',
                 body: JSON.stringify(data),
+                signal: controller.signal,
                 headers,
                 agent: null,
             });
@@ -125,6 +128,7 @@ describe('With correct API results', () => {
             expect(global.fetch).toBeCalledWith(`https://${server}/api/auth/${authVersion}/token/${oldToken}`, {
                 method: 'delete',
                 body: null,
+                signal: controller.signal,
                 headers: {},
                 agent: null,
             });
@@ -145,6 +149,7 @@ describe('With unAuthorized API results', () => {
             expect(global.fetch).toBeCalledWith(`https://${server}/api/auth/${authVersion}/token/${tokenToCheck}`, {
                 method: 'head',
                 body: null,
+                signal: controller.signal,
                 headers: {},
                 agent: null,
             });
@@ -173,6 +178,7 @@ describe('With not found API results', () => {
             expect(global.fetch).toBeCalledWith(`https://${server}/api/calld/1.0/users/me/voicemails`, {
                 method: 'get',
                 body: null,
+                signal: controller.signal,
                 headers: {
                     'X-Auth-Token': token,
                     'Content-Type': 'application/json',

package/lib/__tests__/web-rtc-client.test.js

@@ -14,6 +14,22 @@ var __importDefault = (this && this.__importDefault) || function (mod) {
 Object.defineProperty(exports, "__esModule", { value: true });
 const web_rtc_client_1 = __importDefault(require("../web-rtc-client"));
 jest.mock('sip.js/lib/platform/web/transport');
+jest.mock('sip.js/lib/api/user-agent', () => {
+    var _a;
+    return ({
+        start: () => { },
+        UserAgent: (_a = class UserAgent {
+                constructor() {
+                    // @ts-ignore
+                    this.start = () => { };
+                    this.transport = {};
+                }
+            },
+            // @ts-ignore
+            _a.makeURI = () => { },
+            _a),
+    });
+});
 // @ts-expect-error
 const client = new web_rtc_client_1.default({});
 describe('WebRTC client', () => {

package/lib/api/dird.js

@@ -20,9 +20,9 @@ const getContactPayload = (contact) => ({
     note: contact.note ? contact.note : '',
 });
 exports.default = ((client, baseUrl) => ({
-    search: (context, term) => client.get(`${baseUrl}/directories/lookup/${context}`, {
+    search: (context, term, offset = 0, limit = null) => client.get(`${baseUrl}/directories/lookup/${context}`, {
         term,
-    }).then(Contact_1.default.parseMany),
+    }).then((response) => Contact_1.default.parseMany(response, offset, limit)),
     listPersonalContacts: (queryParams = null) => client.get(`${baseUrl}/personal`, queryParams).then((response) => Contact_1.default.parseManyPersonal(response.items)),
     fetchPersonalContact: (contactUuid) => client.get(`${baseUrl}/personal/${contactUuid}`).then(Contact_1.default.parsePersonal),
     addContact: (contact) => client.post(`${baseUrl}/personal`, getContactPayload(contact)).then(Contact_1.default.parsePersonal),

package/lib/api-client.js

@@ -112,6 +112,7 @@ class ApiClient {
                     this.onRefreshTokenError(error);
                 }
             }
+            return null;
         });
     }
     setToken(token) {
@@ -123,6 +124,9 @@ class ApiClient {
     setRefreshToken(refreshToken) {
         this.refreshToken = refreshToken;
     }
+    setRequestTimeout(requestTimeout) {
+        this.client.setRequestTimeout(requestTimeout);
+    }
     setClientId(clientId) {
         this.client.clientId = clientId;
     }

package/lib/checker/checks/ice-ipv4.js

@@ -17,7 +17,8 @@ exports.default = {
     name: 'Non IP v4 ice',
     check: (server, session, externalAppConfig) => new Promise((resolve, reject) => {
         if (typeof MediaStream === 'undefined') {
-            return resolve('Skipped on node');
+            resolve('Skipped on node');
+            return;
         }
         const mobile = (0, isMobile_1.default)();
         const offerOptions = {

package/lib/domain/Contact.js

@@ -31,11 +31,12 @@ class Contact {
         }
         return aLastName.localeCompare(bLastName);
     }
-    static parseMany(response) {
-        if (!response || !response.results) {
+    static parseMany(response, offset = 0, limit = null) {
+        if (!response || !response.results || limit === 0) {
             return [];
         }
-        return response.results.map(r => Contact.parse(r, response.column_types));
+        const results = limit !== null && limit > 0 ? response.results.slice(offset, limit) : offset > 0 ? response.results.slice(offset) : response.results;
+        return results.map(r => Contact.parse(r, response.column_types));
     }
     static manyGraphQlWithNumbersParser(numbers) {
         return (response) => {
@@ -136,13 +137,19 @@ class Contact {
             const postalAddress = plain.postalAddresses[0];
             address = `${postalAddress.street} ${postalAddress.city} ${postalAddress.postCode} ${postalAddress.country}`;
         }
+        const firstName = plain.givenName || '';
+        const lastName = plain.familyName || '';
+        const companyName = plain.company || '';
+        const isCompanyAccount = !firstName && !lastName;
+        const name = isCompanyAccount ? companyName : `${firstName} ${lastName}`;
         return new Contact({
-            name: `${plain.givenName || ''} ${plain.familyName || ''}`,
+            name,
             number: plain.phoneNumbers.length ? plain.phoneNumbers[0].number : '',
-            numbers: plain.phoneNumbers.length ? [{
-                    label: 'primary',
-                    number: plain.phoneNumbers[0].number,
-                }] : [],
+            // Make numbers unique
+            numbers: [...new Map(plain.phoneNumbers.map(item => [item.number, item])).values()].map((item, idx) => ({
+                label: idx === 0 ? 'primary' : 'secondary',
+                number: item.number,
+            })),
             email: plain.emailAddresses.length ? plain.emailAddresses[0].email : '',
             emails: plain.emailAddresses.length ? [{
                     label: 'primary',

package/lib/domain/Phone/WebRTCPhone.js

@@ -306,9 +306,6 @@ class WebRTCPhone extends Emitter_1.default {
                 this.eventEmitter.emit(exports.ON_CALL_CANCELED, callSession, elsewhere);
             };
         }
-        else if (sipSession instanceof api_1.Invitation) {
-            console.warn('sipSession._onCancel not found, please update the wazo SDK accordingly');
-        }
         sipSession.stateChange.addListener((newState) => {
             switch (newState) {
                 case session_state_1.SessionState.Establishing:
@@ -362,10 +359,6 @@ class WebRTCPhone extends Emitter_1.default {
             if (kind === 'audio') {
                 return this.eventEmitter.emit(exports.ON_AUDIO_STREAM, stream);
             }
-            // not sure this does anything
-            if (kind === 'video') {
-                event.track.enabled = false;
-            }
             return this.eventEmitter.emit(exports.ON_VIDEO_STREAM, stream, event.track.id, event, sipSession);
         };
         peerConnection.onremovestream = (event) => {
@@ -417,10 +410,12 @@ class WebRTCPhone extends Emitter_1.default {
             logger.info('WebRTC phone - stop screen sharing', {
                 restoreLocalStream,
             });
-            let reinvited;
+            let reinvited = null;
             try {
                 if (this.currentScreenShare.stream) {
-                    this.currentScreenShare.stream.getTracks().forEach((track) => track.stop());
+                    this.currentScreenShare.stream.getTracks()
+                        .filter((track) => track.enabled)
+                        .forEach((track) => track.stop());
                 }
                 if (restoreLocalStream) {
                     const targetCallSession = callSession || this.currentCallSession;
@@ -1124,12 +1119,15 @@ class WebRTCPhone extends Emitter_1.default {
         return __awaiter(this, void 0, void 0, function* () {
             logger.info('WebRTC close');
             try {
-                yield Promise.race([this.unregister(), new Promise((resolve, reject) => setTimeout(() => reject(new Error('Unregister, timed out')), 3000))]);
+                yield Promise.race([
+                    this.unregister(),
+                    new Promise((resolve, reject) => setTimeout(() => reject(new Error('Unregister, timed out')), 3000)),
+                ]);
             }
             catch (e) {
                 logger.error('WebRTC close, unregister error', e);
             }
-            this.client.close(force);
+            yield this.client.close(force);
             this.unbind();
             this.incomingSessions = [];
             this.currentSipSession = undefined;

package/lib/domain/Session.js

@@ -6,6 +6,7 @@ Object.defineProperty(exports, "__esModule", { value: true });
 /* eslint-disable no-nested-ternary */
 const jsrsasign_1 = require("jsrsasign");
 const pubkey_1 = __importDefault(require("../pubkey"));
+const Contact_1 = __importDefault(require("./Contact"));
 const new_from_1 = __importDefault(require("../utils/new-from"));
 const compare_version_1 = __importDefault(require("../utils/compare-version"));
 const { jws } = jsrsasign_1.KJUR;
@@ -89,6 +90,28 @@ class Session {
     primaryCtiLine() {
         return this.profile && (this.profile.sipLines || []).find(sipLine => sipLine && !sipLine.isWebRtc());
     }
+    getContact() {
+        if (this.contact) {
+            return this.contact;
+        }
+        const { email, firstName, lastName, mobileNumber, state, status } = this.profile || {};
+        const number = this.primaryNumber() || '';
+        const numbers = [{ label: 'primary', number }];
+        if (mobileNumber) {
+            numbers.push({ label: 'mobile', number: mobileNumber });
+        }
+        return new Contact_1.default({
+            uuid: this.uuid || '',
+            email,
+            name: `${firstName} ${lastName}`,
+            number,
+            numbers,
+            state: state || '',
+            status,
+            backend: 'wazo',
+            source: 'default',
+        });
+    }
     primaryContext() {
         if (this.engineVersion) {
             if (this.hasEngineVersionGte(MINIMUM_WAZO_ENGINE_VERSION_FOR_DEFAULT_CONTEXT)) {