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

package/esm/web-rtc-client.js

@@ -17,7 +17,7 @@ import Emitter from './utils/Emitter';
 import ApiClient from './api-client';
 import IssueReporter from './service/IssueReporter';
 import Heartbeat from './utils/Heartbeat';
-import { getSipSessionId, makeURI, replaceLocalIpModifier, SIP_ID_LENGTH } from './utils/sdp';
+import { getSipCallId, makeURI, replaceLocalIpModifier } from './utils/sdp';
 import { cleanupStream, createUaOptions, fetchNetworkStats, getAudioConstraints, getLocalStream, getLocalTracks, getLocalVideoStream, getMediaConfiguration, getPeerConnection, getRemoteStream, getRemoteTracks, getRemoteVideoStream, getRemoteVideoStreamFromPc, getVideoConstraints, hasALocalVideoTrack, hasARemoteVideoTrack, hasAVideoTrack, hasLocalVideo, hasRemoteVideo, hasVideoTrack, isAudioOnly, isVideoRemotelyHeld, isWeb, sessionWantsToDoVideo, setLocalMediaStream, toggleAudio, toggleVideo, updateRemoteStream } from './utils/webrtc';
 const SEND_STATS_DELAY = 5000;
 const states = ['STATUS_NULL', 'STATUS_NEW', 'STATUS_CONNECTING', 'STATUS_CONNECTED', 'STATUS_COMPLETED'];
@@ -25,6 +25,7 @@ const logger = IssueReporter ? IssueReporter.loggerFor('webrtc-client') : consol
 const statsLogger = IssueReporter ? IssueReporter.loggerFor('webrtc-stats') : console;
 // events
 const REGISTERED = 'registered';
+const REGISTERING = 'registering';
 const UNREGISTERED = 'unregistered';
 const REGISTRATION_FAILED = 'registrationFailed';
 const INVITE = 'invite';
@@ -42,7 +43,7 @@ const ON_ERROR = 'onError';
 const ON_SCREEN_SHARING_REINVITE = 'onScreenSharingReinvite';
 const ON_NETWORK_STATS = 'onNetworkStats';
 const ON_DISCONNECTED = 'onDisconnected';
-export const events = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
+export const events = [REGISTERED, REGISTERING, UNREGISTERED, REGISTRATION_FAILED, INVITE];
 export const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
 export class CanceledCallError extends Error {
 }
@@ -76,6 +77,7 @@ export default class WebRTCClient extends Emitter {
     // sugar
     ON_USER_AGENT;
     REGISTERED;
+    REGISTERING;
     UNREGISTERED;
     REGISTRATION_FAILED;
     INVITE;
@@ -103,7 +105,9 @@ export default class WebRTCClient extends Emitter {
         this.skipRegister = config.skipRegister;
         this._buildConfig(config, session).then((newConfig) => {
             this.config = newConfig;
-            this.userAgent = this.createUserAgent(uaConfigOverrides);
+            if (!this.userAgent) {
+                this.userAgent = this.createUserAgent(uaConfigOverrides);
+            }
         });
         this.audioOutputDeviceId = config.audioDeviceOutput;
         this.audioOutputVolume = config.audioOutputVolume || 1;
@@ -128,6 +132,7 @@ export default class WebRTCClient extends Emitter {
         this.heartbeat.setOnHeartbeatTimeout(this._onHeartbeatTimeout.bind(this));
         // sugar
         this.REGISTERED = REGISTERED;
+        this.REGISTERING = REGISTERING;
         this.UNREGISTERED = UNREGISTERED;
         this.REGISTRATION_FAILED = REGISTRATION_FAILED;
         this.INVITE = INVITE;
@@ -168,7 +173,7 @@ export default class WebRTCClient extends Emitter {
                 logger.info('sdk webrtc on invite', {
                     method: 'delegate.onInvite',
                     clientId: this.clientId,
-                    id: invitation.id,
+                    id: getSipCallId(invitation),
                     remoteURI: invitation.remoteURI,
                 });
                 this._setupSession(invitation);
@@ -230,7 +235,7 @@ export default class WebRTCClient extends Emitter {
                 // @ts-ignore: private
                 this.registerer.waitingToggle(false);
             }
-            this.eventEmitter.emit(UNREGISTERED);
+            this.eventEmitter.emit(UNREGISTERED, error);
         }
     }
     async register(tries = 0) {
@@ -251,6 +256,7 @@ export default class WebRTCClient extends Emitter {
             return Promise.resolve();
         }
         logger.info('sdk webrtc registering...', logInfo);
+        this.eventEmitter.emit(REGISTERING);
         if (!this.userAgent) {
             logger.info('sdk webrtc recreating User Agent');
             this.userAgent = this.createUserAgent(this.uaConfigOverrides);
@@ -358,6 +364,12 @@ export default class WebRTCClient extends Emitter {
             userAgent: !!this.userAgent,
             registerer: !!this.registerer,
         });
+        // @ts-ignore: private
+        if (this.registerer?.waiting) {
+            logger.info('sdk webrtc unregistering, waiting for register to finish...');
+            // We have to wait for the register phase to finish
+            await this.waitForRegister();
+        }
         try {
             return new Promise((resolve, reject) => {
                 if (!this.registerer) {
@@ -435,7 +447,7 @@ export default class WebRTCClient extends Emitter {
             this._setupSession(session);
         }
         if (conference) {
-            this.conferences[getSipSessionId(session)] = true;
+            this.conferences[getSipCallId(session)] = true;
         }
         const inviteOptions = {
             requestDelegate: {
@@ -450,7 +462,7 @@ export default class WebRTCClient extends Emitter {
                 },
                 onReject: (response) => {
                     logger.info('on call rejected', {
-                        id: session?.id,
+                        id: getSipCallId(session),
                         // @ts-ignore: fromTag does not exist
                         fromTag: session?.fromTag,
                     });
@@ -480,7 +492,7 @@ export default class WebRTCClient extends Emitter {
     answer(session, enableVideo) {
         logger.info('sdk webrtc answer call', {
             clientId: this.clientId,
-            id: session.id,
+            id: getSipCallId(session),
             enableVideo,
         });
         if (!session || !session.accept) {
@@ -496,7 +508,7 @@ export default class WebRTCClient extends Emitter {
             if (session.isCanceled) {
                 const message = 'accepted a canceled session (or was canceled during the accept phase).';
                 logger.error(message, {
-                    id: session.id,
+                    id: getSipCallId(session),
                 });
                 this.onCallEnded(session);
                 throw new CanceledCallError(message);
@@ -504,7 +516,7 @@ export default class WebRTCClient extends Emitter {
             logger.info('sdk webrtc answer, accepted.');
             this._onAccepted(session);
         }).catch(e => {
-            logger.error(`answer call error for ${session ? session.id : 'n/a'}`, e);
+            logger.error(`answer call error for ${getSipCallId(session)}`, e);
             throw e;
         });
     }
@@ -514,7 +526,7 @@ export default class WebRTCClient extends Emitter {
         try {
             this._stopSendingStats(session);
             this._cleanupMedia(session);
-            delete this.sipSessions[getSipSessionId(session)];
+            delete this.sipSessions[getSipCallId(session)];
             // Check if Invitation or Inviter (Invitation = incoming call)
             const isInviter = session instanceof Inviter;
             // @see github.com/onsip/SIP.js/blob/f11dfd584bc9788ccfc94e03034020672b738975/src/platform/web/simple-user/simple-user.ts#L1004
@@ -536,7 +548,7 @@ export default class WebRTCClient extends Emitter {
     }
     // Fetch and emit an event at `interval` with session network stats
     startNetworkMonitoring(session, interval = 1000) {
-        const sessionId = getSipSessionId(session);
+        const sessionId = getSipCallId(session);
         logger.info('starting network inspection', {
             id: sessionId,
             clientId: this.clientId,
@@ -545,7 +557,7 @@ export default class WebRTCClient extends Emitter {
         this.networkMonitoringInterval[sessionId] = setInterval(() => this._fetchNetworkStats(sessionId), interval);
     }
     stopNetworkMonitoring(session) {
-        const sessionId = getSipSessionId(session);
+        const sessionId = getSipCallId(session);
         const exists = (sessionId in this.networkMonitoringInterval);
         logger.info('stopping network inspection', {
             clientId: this.clientId,
@@ -561,7 +573,7 @@ export default class WebRTCClient extends Emitter {
     async reject(session) {
         logger.info('sdk webrtc reject call', {
             clientId: this.clientId,
-            id: session.id,
+            id: getSipCallId(session),
         });
         try {
             if (session instanceof Invitation) {
@@ -615,13 +627,13 @@ export default class WebRTCClient extends Emitter {
     }
     mute(session) {
         logger.info('sdk webrtc mute', {
-            id: session.id,
+            id: getSipCallId(session),
         });
         toggleAudio(session, true);
     }
     unmute(session) {
         logger.info('sdk webrtc unmute', {
-            id: session.id,
+            id: getSipCallId(session),
         });
         toggleAudio(session, false);
     }
@@ -658,18 +670,18 @@ export default class WebRTCClient extends Emitter {
     }
     toggleCameraOn(session) {
         logger.info('sdk webrtc toggle camera on', {
-            id: session.id,
+            id: getSipCallId(session),
         });
         toggleVideo(session, false);
     }
     toggleCameraOff(session) {
         logger.info('sdk webrtc toggle camera off', {
-            id: session.id,
+            id: getSipCallId(session),
         });
         toggleVideo(session, true);
     }
     hold(session, isConference = false, hadVideo = false) {
-        const sessionId = getSipSessionId(session);
+        const sessionId = getSipCallId(session);
         const hasVideo = hadVideo || this.hasLocalVideo(sessionId);
         logger.info('sdk webrtc hold', {
             sessionId,
@@ -717,7 +729,7 @@ export default class WebRTCClient extends Emitter {
         });
     }
     unhold(session, isConference = false) {
-        const sessionId = getSipSessionId(session);
+        const sessionId = getSipCallId(session);
         const hasVideo = sessionId in this.heldSessions && this.heldSessions[sessionId].hasVideo;
         logger.info('sdk webrtc unhold', {
             sessionId,
@@ -732,7 +744,7 @@ export default class WebRTCClient extends Emitter {
             return Promise.resolve();
         }
         this.unmute(session);
-        delete this.heldSessions[getSipSessionId(session)];
+        delete this.heldSessions[getSipCallId(session)];
         session.sessionDescriptionHandlerOptionsReInvite = {
             hold: false,
             conference: isConference,
@@ -757,7 +769,7 @@ export default class WebRTCClient extends Emitter {
         return this.heldSessions[sessionId];
     }
     isCallHeld(session) {
-        return getSipSessionId(session) in this.heldSessions;
+        return getSipCallId(session) in this.heldSessions;
     }
     isVideoRemotelyHeld(sessionId) {
         return isVideoRemotelyHeld(this.heldSessions[sessionId]);
@@ -767,7 +779,7 @@ export default class WebRTCClient extends Emitter {
             return false;
         }
         logger.info('Sending DTMF', {
-            id: getSipSessionId(session),
+            id: getSipCallId(session),
             tone,
         });
         return session.sessionDescriptionHandler.sendDtmf(tone);
@@ -784,7 +796,7 @@ export default class WebRTCClient extends Emitter {
     transfer(session, target) {
         this.hold(session);
         logger.info('Transfering a session', {
-            id: getSipSessionId(session),
+            id: getSipCallId(session),
             target,
         });
         const options = {
@@ -807,21 +819,21 @@ export default class WebRTCClient extends Emitter {
     atxfer(session) {
         this.hold(session);
         logger.info('webrtc transfer started', {
-            id: getSipSessionId(session),
+            id: getSipCallId(session),
         });
         const result = {
             newSession: null,
             init: async (target) => {
                 logger.info('webrtc transfer initialized', {
-                    id: getSipSessionId(session),
+                    id: getSipCallId(session),
                     target,
                 });
                 result.newSession = await this.call(target);
             },
             complete: () => {
                 logger.info('webrtc transfer completed', {
-                    id: getSipSessionId(session),
-                    referId: getSipSessionId(result.newSession),
+                    id: getSipCallId(session),
+                    referId: getSipCallId(result.newSession),
                 });
                 session.refer(result.newSession);
             },
@@ -838,7 +850,7 @@ export default class WebRTCClient extends Emitter {
             return;
         }
         logger.info('send WebRTC message', {
-            sipId: sipSession.id,
+            sipId: getSipCallId(sipSession),
             contentType,
         });
         try {
@@ -988,7 +1000,7 @@ export default class WebRTCClient extends Emitter {
         }
         const sdh = session.sessionDescriptionHandler;
         const pc = sdh.peerConnection;
-        const sessionId = getSipSessionId(session);
+        const sessionId = getSipCallId(session);
         const localStream = this.getLocalStream(sessionId);
         logger.info('changing video input device', {
             id,
@@ -1049,7 +1061,7 @@ export default class WebRTCClient extends Emitter {
         const desktop = newConstraints && newConstraints.desktop;
         logger.info('Sending reinvite', {
             clientId: this.clientId,
-            id: getSipSessionId(sipSession),
+            id: getSipCallId(sipSession),
             newConstraints,
             conference,
             audioOnly,
@@ -1083,7 +1095,7 @@ export default class WebRTCClient extends Emitter {
                         sipSession.incomingInviteRequest.message.body = response.message.body;
                     }
                     logger.info('on re-INVITE accepted', {
-                        id: getSipSessionId(sipSession),
+                        id: getSipCallId(sipSession),
                         wasMuted,
                         shouldDoScreenSharing,
                     });
@@ -1194,7 +1206,7 @@ export default class WebRTCClient extends Emitter {
     }
     onCallEnded(session) {
         this._cleanupMedia(session);
-        delete this.sipSessions[getSipSessionId(session)];
+        delete this.sipSessions[getSipCallId(session)];
         this._stopSendingStats(session);
         this.stopNetworkMonitoring(session);
     }
@@ -1207,14 +1219,14 @@ export default class WebRTCClient extends Emitter {
         this.userAgent.attemptReconnection();
     }
     storeSipSession(session) {
-        const id = getSipSessionId(session);
+        const id = getSipCallId(session);
         logger.info('storing sip session', { id, clientId: this.clientId });
         this.sipSessions[id] = session;
     }
     getSipSession(id) {
-        return id in this.sipSessions ? this.sipSessions[id.substring(0, SIP_ID_LENGTH)] : null;
+        return id in this.sipSessions ? this.sipSessions[id] : null;
     }
-    getSipSessionIds() {
+    getSipCallIds() {
         return Object.keys(this.sipSessions);
     }
     async updateLocalStream(sipSessionId, newStream) {
@@ -1253,8 +1265,8 @@ export default class WebRTCClient extends Emitter {
         audio.autoplay = true;
         return audio;
     }
-    _onTransportError() {
-        logger.error('on transport error');
+    _onTransportError(reason) {
+        logger.warn('on transport error', { reason });
         this.eventEmitter.emit(TRANSPORT_ERROR);
         this.attemptReconnection();
     }
@@ -1307,7 +1319,7 @@ export default class WebRTCClient extends Emitter {
                 }
             }
             // We can invoke disconnect() with an error that can be catcher by `onDisconnect`, so we have to trigger it here.
-            this._onTransportError();
+            this._onTransportError('heartbeat timeout');
         }
     }
     _hasAudio() {
@@ -1369,7 +1381,7 @@ export default class WebRTCClient extends Emitter {
     }
     // Invitation and Inviter extends Session
     _setupSession(session) {
-        const sipSessionId = getSipSessionId(session);
+        const sipSessionId = getSipCallId(session);
         // When receiving an Invitation, the delegate is not defined.
         if (!session.delegate) {
             session.delegate = {};
@@ -1432,17 +1444,14 @@ export default class WebRTCClient extends Emitter {
     }
     async _onProgress(session, early = false) {
         await this._setupMedias(session);
-        const sessionId = getSipSessionId(session).substr(0, 20);
+        const sessionId = getSipCallId(session);
         updateRemoteStream(session);
-        logger.info('progress received', {
-            sessionId,
-            early,
-        });
+        logger.info('progress received', { sessionId, early });
         this.eventEmitter.emit(early ? ON_EARLY_MEDIA : ON_PROGRESS, session);
     }
     async _onAccepted(session, sessionDialog, withEvent = true, initAllTracks = true) {
         logger.info('on call accepted', {
-            id: session.id,
+            id: getSipCallId(session),
             clientId: this.clientId,
             remoteTag: session.remoteTag,
         });
@@ -1486,7 +1495,7 @@ export default class WebRTCClient extends Emitter {
             return;
         }
         // Safari hack, because you cannot call .play() from a non user action
-        const sessionId = getSipSessionId(session);
+        const sessionId = getSipCallId(session);
         const isConference = this.isConference(sessionId);
         if (sessionId in this.audioElements) {
             logger.info('html element already exists for session', {
@@ -1531,7 +1540,7 @@ export default class WebRTCClient extends Emitter {
         audioElement.play().catch(() => { });
     }
     _cleanupMedia(session) {
-        const sessionId = getSipSessionId(session);
+        const sessionId = getSipCallId(session);
         const localStream = this.getLocalStream(sessionId);
         if (localStream) {
             cleanupStream(localStream);
@@ -1550,7 +1559,7 @@ export default class WebRTCClient extends Emitter {
         };
         if (this._hasAudio() && isWeb()) {
             if (session) {
-                cleanLocalElement(getSipSessionId(session));
+                cleanLocalElement(getSipCallId(session));
             }
             else {
                 Object.keys(this.audioElements).forEach(id => cleanLocalElement(id));
@@ -1570,7 +1579,7 @@ export default class WebRTCClient extends Emitter {
         if (!pc) {
             return;
         }
-        const sessionId = getSipSessionId(session);
+        const sessionId = getSipCallId(session);
         getStats(pc, (result) => {
             const { results, internal, nomore, ...stats } = result;
             this.statsIntervals[sessionId] = nomore;
@@ -1581,7 +1590,7 @@ export default class WebRTCClient extends Emitter {
         }, SEND_STATS_DELAY);
     }
     _stopSendingStats(session) {
-        const sessionId = getSipSessionId(session);
+        const sessionId = getSipCallId(session);
         logger.trace('Check for stopping stats', {
             sessionId,
             ids: Object.keys(this.statsIntervals),
@@ -1632,18 +1641,18 @@ export default class WebRTCClient extends Emitter {
     async _accept(session, options = {}) {
         if (session.state === SessionState.Terminated || session.state === SessionState.Terminating) {
             const error = 'Trying to accept a terminated sipSession.';
-            logger.warn(error, { state: session.state, sessionId: session.id });
+            logger.warn(error, { state: session.state, sessionId: getSipCallId(session) });
             throw new Error(error);
         }
         if (session.state !== SessionState.Initial) {
             const error = 'Trying to accept a non Initial sipSession.';
-            logger.warn(error, { state: session.state, sessionId: session.id });
+            logger.warn(error, { state: session.state, sessionId: getSipCallId(session) });
             throw new Error(error);
         }
         // @ts-ignore: private
         if (!session.incomingInviteRequest.acceptable) {
             const error = 'Trying to reject a non `acceptable` session';
-            logger.warn(error, { state: session.state, sessionId: session.id });
+            logger.warn(error, { state: session.state, sessionId: getSipCallId(session) });
             throw new Error(error);
         }
         try {
@@ -1655,12 +1664,12 @@ export default class WebRTCClient extends Emitter {
     }
     async _reject(session, options = {}) {
         if (session.state !== SessionState.Initial && session.state !== SessionState.Establishing) {
-            logger.warn('Trying to reject a session in a wrong state', { state: session.state, sessionId: session.id });
+            logger.warn('Trying to reject a session in a wrong state', { state: session.state, sessionId: getSipCallId(session) });
             return;
         }
         // @ts-ignore: private
         if (!session.incomingInviteRequest.rejectable) {
-            logger.warn('Trying to reject a non `rejectable` session', { state: session.state, sessionId: session.id });
+            logger.warn('Trying to reject a non `rejectable` session', { state: session.state, sessionId: getSipCallId(session) });
             return;
         }
         try {
@@ -1672,7 +1681,7 @@ export default class WebRTCClient extends Emitter {
     }
     async _cancel(session, options = {}) {
         if (session.state !== SessionState.Initial && session.state !== SessionState.Establishing) {
-            logger.warn('Trying to cancel a session in a wrong state', { state: session.state, sessionId: session.id });
+            logger.warn('Trying to cancel a session in a wrong state', { state: session.state, sessionId: getSipCallId(session) });
             return;
         }
         try {
@@ -1684,7 +1693,7 @@ export default class WebRTCClient extends Emitter {
     }
     async _bye(session, options = {}) {
         if (session.state !== SessionState.Established) {
-            logger.warn('Trying to end a session in a wrong state', { state: session.state, sessionId: session.id });
+            logger.warn('Trying to end a session in a wrong state', { state: session.state, sessionId: getSipCallId(session) });
             return null;
         }
         try {

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

@@ -41,6 +41,9 @@ 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()),
@@ -80,9 +83,17 @@ 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({ [call_state_machine_1.States.ESTABLISHED]: call_state_machine_1.EstablishedStates.MUTED });
+            expect(call.state).toStrictEqual({
+                [call_state_machine_1.States.ESTABLISHED]: {
+                    [call_state_machine_1.EstablishedStates.ONGOING]: {},
+                    [call_state_machine_1.EstablishedSubStates.MUTE]: call_state_machine_1.EstablishedStates.MUTED,
+                    [call_state_machine_1.EstablishedSubStates.HOLD]: call_state_machine_1.EstablishedStates.UN_HELD,
+                },
+            });
+            expect(call.isEstablished()).toBeTruthy();
         }));
     });
     describe('accept', () => {
@@ -125,4 +136,42 @@ describe('Call', () => {
             expect(softphone_1.default.client.hangup).toHaveBeenCalled();
         }));
     });
+    describe('state propagation', () => {
+        it('Should update the call state in events', (done) => {
+            const call = new call_1.default(sipCall, softphone_1.default);
+            // @ts-ignore: private method
+            call._sendAction(call_state_machine_1.Actions.MAKE_CALL);
+            call.on('remotelyAccepted', () => {
+                expect(call.state).toStrictEqual({
+                    [call_state_machine_1.States.ESTABLISHED]: {
+                        [call_state_machine_1.EstablishedStates.ONGOING]: {},
+                        [call_state_machine_1.EstablishedSubStates.MUTE]: call_state_machine_1.EstablishedStates.UN_MUTED,
+                        [call_state_machine_1.EstablishedSubStates.HOLD]: call_state_machine_1.EstablishedStates.UN_HELD,
+                    },
+                });
+                done();
+            });
+            call.onAccepted();
+        });
+    });
+    describe('Multiple sub state', () => {
+        it('Should mute and hold a call', () => {
+            const call = new call_1.default(sipCall, softphone_1.default);
+            // @ts-ignore: private method
+            call._sendAction(call_state_machine_1.Actions.MAKE_CALL);
+            expect(call.isMuted()).toBeFalsy();
+            expect(call.isHeld()).toBeFalsy();
+            // @ts-ignore: private method
+            call._sendAction(call_state_machine_1.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/lib/api/auth.js

@@ -64,6 +64,9 @@ exports.default = ((client, baseUrl) => ({
         }
         return client.post(`${baseUrl}/token`, body, headers).then(Session_1.default.parse);
     }),
+    samlLogOut() {
+        return client.get(`${baseUrl}/saml/logout`, null, null);
+    },
     initiateIdpAuthentication: (domain, redirectUrl) => __awaiter(void 0, void 0, void 0, function* () {
         const body = {
             domain,

package/lib/checker/checks/webrtc.js

@@ -43,7 +43,7 @@ exports.default = {
         });
         client.on(client.REGISTERED, () => {
             const sipSession = client.call('*10');
-            if (!sipSession || !(0, sdp_1.getSipSessionId)(sipSession)) {
+            if (!sipSession || !(0, sdp_1.getSipCallId)(sipSession)) {
                 return handleError('Unable to make call through WebRTC');
             }
             setTimeout(() => {

package/lib/domain/CallLog.js

@@ -46,8 +46,7 @@ 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) {
@@ -105,7 +104,7 @@ 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/lib/domain/Phone/CTIPhone.js

@@ -119,7 +119,7 @@ class CTIPhone extends Emitter_1.default {
                 number: callSession.number,
             });
             try {
-                yield CallApi_1.default.cancelCall(callSession);
+                yield CallApi_1.default.cancelCall({ apiId: callSession.getId() });
                 if (this.currentCall && callSession.callId === this.currentCall.id) {
                     this.endCurrentCall(callSession);
                 }
@@ -143,7 +143,7 @@ class CTIPhone extends Emitter_1.default {
                 callId: callSession.getId(),
                 number: callSession.number,
             });
-            yield CallApi_1.default.cancelCall(callSession);
+            yield CallApi_1.default.cancelCall({ id: callSession.getId() });
             this.eventEmitter.emit('onCallEnded', callSession);
         });
     }