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

package/lib/web-rtc-client.js

@@ -51,6 +51,7 @@ const logger = IssueReporter_1.default ? IssueReporter_1.default.loggerFor('webr
 const statsLogger = IssueReporter_1.default ? IssueReporter_1.default.loggerFor('webrtc-stats') : console;
 // events
 const REGISTERED = 'registered';
+const REGISTERING = 'registering';
 const UNREGISTERED = 'unregistered';
 const REGISTRATION_FAILED = 'registrationFailed';
 const INVITE = 'invite';
@@ -68,7 +69,7 @@ const ON_ERROR = 'onError';
 const ON_SCREEN_SHARING_REINVITE = 'onScreenSharingReinvite';
 const ON_NETWORK_STATS = 'onNetworkStats';
 const ON_DISCONNECTED = 'onDisconnected';
-exports.events = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
+exports.events = [REGISTERED, REGISTERING, UNREGISTERED, REGISTRATION_FAILED, INVITE];
 exports.transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
 class CanceledCallError extends Error {
 }
@@ -85,7 +86,9 @@ class WebRTCClient extends Emitter_1.default {
         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;
@@ -110,6 +113,7 @@ class WebRTCClient extends Emitter_1.default {
         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;
@@ -151,7 +155,7 @@ class WebRTCClient extends Emitter_1.default {
                 logger.info('sdk webrtc on invite', {
                     method: 'delegate.onInvite',
                     clientId: this.clientId,
-                    id: invitation.id,
+                    id: (0, sdp_1.getSipCallId)(invitation),
                     remoteURI: invitation.remoteURI,
                 });
                 this._setupSession(invitation);
@@ -218,7 +222,7 @@ class WebRTCClient extends Emitter_1.default {
                     // @ts-ignore: private
                     this.registerer.waitingToggle(false);
                 }
-                this.eventEmitter.emit(UNREGISTERED);
+                this.eventEmitter.emit(UNREGISTERED, error);
             }
         });
     }
@@ -242,6 +246,7 @@ class WebRTCClient extends Emitter_1.default {
                 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);
@@ -346,11 +351,18 @@ class WebRTCClient extends Emitter_1.default {
     }
     unregister() {
         return __awaiter(this, void 0, void 0, function* () {
+            var _a;
             logger.info('sdk webrtc unregistering..', {
                 clientId: this.clientId,
                 userAgent: !!this.userAgent,
                 registerer: !!this.registerer,
             });
+            // @ts-ignore: private
+            if ((_a = this.registerer) === null || _a === void 0 ? void 0 : _a.waiting) {
+                logger.info('sdk webrtc unregistering, waiting for register to finish...');
+                // We have to wait for the register phase to finish
+                yield this.waitForRegister();
+            }
             try {
                 return new Promise((resolve, reject) => {
                     if (!this.registerer) {
@@ -425,7 +437,7 @@ class WebRTCClient extends Emitter_1.default {
             this._setupSession(session);
         }
         if (conference) {
-            this.conferences[(0, sdp_1.getSipSessionId)(session)] = true;
+            this.conferences[(0, sdp_1.getSipCallId)(session)] = true;
         }
         const inviteOptions = {
             requestDelegate: {
@@ -441,7 +453,7 @@ class WebRTCClient extends Emitter_1.default {
                 },
                 onReject: (response) => {
                     logger.info('on call rejected', {
-                        id: session === null || session === void 0 ? void 0 : session.id,
+                        id: (0, sdp_1.getSipCallId)(session),
                         // @ts-ignore: fromTag does not exist
                         fromTag: session === null || session === void 0 ? void 0 : session.fromTag,
                     });
@@ -471,7 +483,7 @@ class WebRTCClient extends Emitter_1.default {
     answer(session, enableVideo) {
         logger.info('sdk webrtc answer call', {
             clientId: this.clientId,
-            id: session.id,
+            id: (0, sdp_1.getSipCallId)(session),
             enableVideo,
         });
         if (!session || !session.accept) {
@@ -487,7 +499,7 @@ class WebRTCClient extends Emitter_1.default {
             if (session.isCanceled) {
                 const message = 'accepted a canceled session (or was canceled during the accept phase).';
                 logger.error(message, {
-                    id: session.id,
+                    id: (0, sdp_1.getSipCallId)(session),
                 });
                 this.onCallEnded(session);
                 throw new CanceledCallError(message);
@@ -495,7 +507,7 @@ class WebRTCClient extends Emitter_1.default {
             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 ${(0, sdp_1.getSipCallId)(session)}`, e);
             throw e;
         });
     }
@@ -506,7 +518,7 @@ class WebRTCClient extends Emitter_1.default {
             try {
                 this._stopSendingStats(session);
                 this._cleanupMedia(session);
-                delete this.sipSessions[(0, sdp_1.getSipSessionId)(session)];
+                delete this.sipSessions[(0, sdp_1.getSipCallId)(session)];
                 // Check if Invitation or Inviter (Invitation = incoming call)
                 const isInviter = session instanceof api_1.Inviter;
                 // @see github.com/onsip/SIP.js/blob/f11dfd584bc9788ccfc94e03034020672b738975/src/platform/web/simple-user/simple-user.ts#L1004
@@ -529,7 +541,7 @@ class WebRTCClient extends Emitter_1.default {
     }
     // Fetch and emit an event at `interval` with session network stats
     startNetworkMonitoring(session, interval = 1000) {
-        const sessionId = (0, sdp_1.getSipSessionId)(session);
+        const sessionId = (0, sdp_1.getSipCallId)(session);
         logger.info('starting network inspection', {
             id: sessionId,
             clientId: this.clientId,
@@ -538,7 +550,7 @@ class WebRTCClient extends Emitter_1.default {
         this.networkMonitoringInterval[sessionId] = setInterval(() => this._fetchNetworkStats(sessionId), interval);
     }
     stopNetworkMonitoring(session) {
-        const sessionId = (0, sdp_1.getSipSessionId)(session);
+        const sessionId = (0, sdp_1.getSipCallId)(session);
         const exists = (sessionId in this.networkMonitoringInterval);
         logger.info('stopping network inspection', {
             clientId: this.clientId,
@@ -555,7 +567,7 @@ class WebRTCClient extends Emitter_1.default {
         return __awaiter(this, void 0, void 0, function* () {
             logger.info('sdk webrtc reject call', {
                 clientId: this.clientId,
-                id: session.id,
+                id: (0, sdp_1.getSipCallId)(session),
             });
             try {
                 if (session instanceof api_1.Invitation) {
@@ -612,13 +624,13 @@ class WebRTCClient extends Emitter_1.default {
     }
     mute(session) {
         logger.info('sdk webrtc mute', {
-            id: session.id,
+            id: (0, sdp_1.getSipCallId)(session),
         });
         (0, webrtc_1.toggleAudio)(session, true);
     }
     unmute(session) {
         logger.info('sdk webrtc unmute', {
-            id: session.id,
+            id: (0, sdp_1.getSipCallId)(session),
         });
         (0, webrtc_1.toggleAudio)(session, false);
     }
@@ -655,18 +667,18 @@ class WebRTCClient extends Emitter_1.default {
     }
     toggleCameraOn(session) {
         logger.info('sdk webrtc toggle camera on', {
-            id: session.id,
+            id: (0, sdp_1.getSipCallId)(session),
         });
         (0, webrtc_1.toggleVideo)(session, false);
     }
     toggleCameraOff(session) {
         logger.info('sdk webrtc toggle camera off', {
-            id: session.id,
+            id: (0, sdp_1.getSipCallId)(session),
         });
         (0, webrtc_1.toggleVideo)(session, true);
     }
     hold(session, isConference = false, hadVideo = false) {
-        const sessionId = (0, sdp_1.getSipSessionId)(session);
+        const sessionId = (0, sdp_1.getSipCallId)(session);
         const hasVideo = hadVideo || this.hasLocalVideo(sessionId);
         logger.info('sdk webrtc hold', {
             sessionId,
@@ -714,7 +726,7 @@ class WebRTCClient extends Emitter_1.default {
         });
     }
     unhold(session, isConference = false) {
-        const sessionId = (0, sdp_1.getSipSessionId)(session);
+        const sessionId = (0, sdp_1.getSipCallId)(session);
         const hasVideo = sessionId in this.heldSessions && this.heldSessions[sessionId].hasVideo;
         logger.info('sdk webrtc unhold', {
             sessionId,
@@ -729,7 +741,7 @@ class WebRTCClient extends Emitter_1.default {
             return Promise.resolve();
         }
         this.unmute(session);
-        delete this.heldSessions[(0, sdp_1.getSipSessionId)(session)];
+        delete this.heldSessions[(0, sdp_1.getSipCallId)(session)];
         session.sessionDescriptionHandlerOptionsReInvite = {
             hold: false,
             conference: isConference,
@@ -754,7 +766,7 @@ class WebRTCClient extends Emitter_1.default {
         return this.heldSessions[sessionId];
     }
     isCallHeld(session) {
-        return (0, sdp_1.getSipSessionId)(session) in this.heldSessions;
+        return (0, sdp_1.getSipCallId)(session) in this.heldSessions;
     }
     isVideoRemotelyHeld(sessionId) {
         return (0, webrtc_1.isVideoRemotelyHeld)(this.heldSessions[sessionId]);
@@ -764,7 +776,7 @@ class WebRTCClient extends Emitter_1.default {
             return false;
         }
         logger.info('Sending DTMF', {
-            id: (0, sdp_1.getSipSessionId)(session),
+            id: (0, sdp_1.getSipCallId)(session),
             tone,
         });
         return session.sessionDescriptionHandler.sendDtmf(tone);
@@ -781,7 +793,7 @@ class WebRTCClient extends Emitter_1.default {
     transfer(session, target) {
         this.hold(session);
         logger.info('Transfering a session', {
-            id: (0, sdp_1.getSipSessionId)(session),
+            id: (0, sdp_1.getSipCallId)(session),
             target,
         });
         const options = {
@@ -804,21 +816,21 @@ class WebRTCClient extends Emitter_1.default {
     atxfer(session) {
         this.hold(session);
         logger.info('webrtc transfer started', {
-            id: (0, sdp_1.getSipSessionId)(session),
+            id: (0, sdp_1.getSipCallId)(session),
         });
         const result = {
             newSession: null,
             init: (target) => __awaiter(this, void 0, void 0, function* () {
                 logger.info('webrtc transfer initialized', {
-                    id: (0, sdp_1.getSipSessionId)(session),
+                    id: (0, sdp_1.getSipCallId)(session),
                     target,
                 });
                 result.newSession = yield this.call(target);
             }),
             complete: () => {
                 logger.info('webrtc transfer completed', {
-                    id: (0, sdp_1.getSipSessionId)(session),
-                    referId: (0, sdp_1.getSipSessionId)(result.newSession),
+                    id: (0, sdp_1.getSipCallId)(session),
+                    referId: (0, sdp_1.getSipCallId)(result.newSession),
                 });
                 session.refer(result.newSession);
             },
@@ -835,7 +847,7 @@ class WebRTCClient extends Emitter_1.default {
             return;
         }
         logger.info('send WebRTC message', {
-            sipId: sipSession.id,
+            sipId: (0, sdp_1.getSipCallId)(sipSession),
             contentType,
         });
         try {
@@ -990,7 +1002,7 @@ class WebRTCClient extends Emitter_1.default {
         }
         const sdh = session.sessionDescriptionHandler;
         const pc = sdh.peerConnection;
-        const sessionId = (0, sdp_1.getSipSessionId)(session);
+        const sessionId = (0, sdp_1.getSipCallId)(session);
         const localStream = this.getLocalStream(sessionId);
         logger.info('changing video input device', {
             id,
@@ -1053,7 +1065,7 @@ class WebRTCClient extends Emitter_1.default {
         const desktop = newConstraints && newConstraints.desktop;
         logger.info('Sending reinvite', {
             clientId: this.clientId,
-            id: (0, sdp_1.getSipSessionId)(sipSession),
+            id: (0, sdp_1.getSipCallId)(sipSession),
             newConstraints,
             conference,
             audioOnly,
@@ -1085,7 +1097,7 @@ class WebRTCClient extends Emitter_1.default {
                         sipSession.incomingInviteRequest.message.body = response.message.body;
                     }
                     logger.info('on re-INVITE accepted', {
-                        id: (0, sdp_1.getSipSessionId)(sipSession),
+                        id: (0, sdp_1.getSipCallId)(sipSession),
                         wasMuted,
                         shouldDoScreenSharing,
                     });
@@ -1200,7 +1212,7 @@ class WebRTCClient extends Emitter_1.default {
     }
     onCallEnded(session) {
         this._cleanupMedia(session);
-        delete this.sipSessions[(0, sdp_1.getSipSessionId)(session)];
+        delete this.sipSessions[(0, sdp_1.getSipCallId)(session)];
         this._stopSendingStats(session);
         this.stopNetworkMonitoring(session);
     }
@@ -1213,14 +1225,14 @@ class WebRTCClient extends Emitter_1.default {
         this.userAgent.attemptReconnection();
     }
     storeSipSession(session) {
-        const id = (0, sdp_1.getSipSessionId)(session);
+        const id = (0, sdp_1.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, sdp_1.SIP_ID_LENGTH)] : null;
+        return id in this.sipSessions ? this.sipSessions[id] : null;
     }
-    getSipSessionIds() {
+    getSipCallIds() {
         return Object.keys(this.sipSessions);
     }
     updateLocalStream(sipSessionId, newStream) {
@@ -1263,8 +1275,8 @@ class WebRTCClient extends Emitter_1.default {
             return audio;
         });
     }
-    _onTransportError() {
-        logger.error('on transport error');
+    _onTransportError(reason) {
+        logger.warn('on transport error', { reason });
         this.eventEmitter.emit(TRANSPORT_ERROR);
         this.attemptReconnection();
     }
@@ -1319,7 +1331,7 @@ class WebRTCClient extends Emitter_1.default {
                     }
                 }
                 // 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');
             }
         });
     }
@@ -1377,7 +1389,7 @@ class WebRTCClient extends Emitter_1.default {
     }
     // Invitation and Inviter extends Session
     _setupSession(session) {
-        const sipSessionId = (0, sdp_1.getSipSessionId)(session);
+        const sipSessionId = (0, sdp_1.getSipCallId)(session);
         // When receiving an Invitation, the delegate is not defined.
         if (!session.delegate) {
             session.delegate = {};
@@ -1441,19 +1453,16 @@ class WebRTCClient extends Emitter_1.default {
     _onProgress(session_1) {
         return __awaiter(this, arguments, void 0, function* (session, early = false) {
             yield this._setupMedias(session);
-            const sessionId = (0, sdp_1.getSipSessionId)(session).substr(0, 20);
+            const sessionId = (0, sdp_1.getSipCallId)(session);
             (0, webrtc_1.updateRemoteStream)(session);
-            logger.info('progress received', {
-                sessionId,
-                early,
-            });
+            logger.info('progress received', { sessionId, early });
             this.eventEmitter.emit(early ? ON_EARLY_MEDIA : ON_PROGRESS, session);
         });
     }
     _onAccepted(session_1, sessionDialog_1) {
         return __awaiter(this, arguments, void 0, function* (session, sessionDialog, withEvent = true, initAllTracks = true) {
             logger.info('on call accepted', {
-                id: session.id,
+                id: (0, sdp_1.getSipCallId)(session),
                 clientId: this.clientId,
                 remoteTag: session.remoteTag,
             });
@@ -1499,7 +1508,7 @@ class WebRTCClient extends Emitter_1.default {
                 return;
             }
             // Safari hack, because you cannot call .play() from a non user action
-            const sessionId = (0, sdp_1.getSipSessionId)(session);
+            const sessionId = (0, sdp_1.getSipCallId)(session);
             const isConference = this.isConference(sessionId);
             if (sessionId in this.audioElements) {
                 logger.info('html element already exists for session', {
@@ -1545,7 +1554,7 @@ class WebRTCClient extends Emitter_1.default {
         });
     }
     _cleanupMedia(session) {
-        const sessionId = (0, sdp_1.getSipSessionId)(session);
+        const sessionId = (0, sdp_1.getSipCallId)(session);
         const localStream = this.getLocalStream(sessionId);
         if (localStream) {
             (0, webrtc_1.cleanupStream)(localStream);
@@ -1564,7 +1573,7 @@ class WebRTCClient extends Emitter_1.default {
         };
         if (this._hasAudio() && (0, webrtc_1.isWeb)()) {
             if (session) {
-                cleanLocalElement((0, sdp_1.getSipSessionId)(session));
+                cleanLocalElement((0, sdp_1.getSipCallId)(session));
             }
             else {
                 Object.keys(this.audioElements).forEach(id => cleanLocalElement(id));
@@ -1584,7 +1593,7 @@ class WebRTCClient extends Emitter_1.default {
         if (!pc) {
             return;
         }
-        const sessionId = (0, sdp_1.getSipSessionId)(session);
+        const sessionId = (0, sdp_1.getSipCallId)(session);
         (0, getstats_1.default)(pc, (result) => {
             const { results, internal, nomore } = result, stats = __rest(result, ["results", "internal", "nomore"]);
             this.statsIntervals[sessionId] = nomore;
@@ -1592,7 +1601,7 @@ class WebRTCClient extends Emitter_1.default {
         }, SEND_STATS_DELAY);
     }
     _stopSendingStats(session) {
-        const sessionId = (0, sdp_1.getSipSessionId)(session);
+        const sessionId = (0, sdp_1.getSipCallId)(session);
         logger.trace('Check for stopping stats', {
             sessionId,
             ids: Object.keys(this.statsIntervals),
@@ -1649,18 +1658,18 @@ class WebRTCClient extends Emitter_1.default {
         return __awaiter(this, arguments, void 0, function* (session, options = {}) {
             if (session.state === session_state_1.SessionState.Terminated || session.state === session_state_1.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: (0, sdp_1.getSipCallId)(session) });
                 throw new Error(error);
             }
             if (session.state !== session_state_1.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: (0, sdp_1.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: (0, sdp_1.getSipCallId)(session) });
                 throw new Error(error);
             }
             try {
@@ -1674,12 +1683,12 @@ class WebRTCClient extends Emitter_1.default {
     _reject(session_1) {
         return __awaiter(this, arguments, void 0, function* (session, options = {}) {
             if (session.state !== session_state_1.SessionState.Initial && session.state !== session_state_1.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: (0, sdp_1.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: (0, sdp_1.getSipCallId)(session) });
                 return;
             }
             try {
@@ -1693,7 +1702,7 @@ class WebRTCClient extends Emitter_1.default {
     _cancel(session_1) {
         return __awaiter(this, arguments, void 0, function* (session, options = {}) {
             if (session.state !== session_state_1.SessionState.Initial && session.state !== session_state_1.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: (0, sdp_1.getSipCallId)(session) });
                 return;
             }
             try {
@@ -1707,7 +1716,7 @@ class WebRTCClient extends Emitter_1.default {
     _bye(session_1) {
         return __awaiter(this, arguments, void 0, function* (session, options = {}) {
             if (session.state !== session_state_1.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: (0, sdp_1.getSipCallId)(session) });
                 return null;
             }
             try {

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@wazo/sdk",
-  "version": "1.0.0-alpha.2",
+  "version": "1.0.0-alpha.20",
   "description": "Wazo's JavaScript Software Development Kit.",
   "main": "index.js",
   "types": "lib/types/index.d.ts",

package/src/__tests__/__voice__/call.test.ts

@@ -3,7 +3,7 @@ import { Inviter, UserAgent, type URI } 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';
 
@@ -32,6 +32,9 @@ 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()),
@@ -76,9 +79,17 @@ 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();
     });
   });
 
@@ -132,4 +143,51 @@ 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/src/api/auth.ts

@@ -33,6 +33,7 @@ export interface AuthD {
   logIn(params: LoginParams): Promise<Session | null | undefined> ;
   logOut: (token: Token) => Promise<LogoutResponse>;
   samlLogIn: (samlSessionId: string) => Promise<Session | null | undefined>;
+  samlLogOut: () => Promise<void | { location:string }>;
   initiateIdpAuthentication(domain: string, redirectUrl: string): Promise<any>;
   refreshToken: (refreshToken: string, backend: string, expiration: number, isMobile?: boolean, tenantId?: string, domainName?: string) => Promise<Session | null | undefined>;
   deleteRefreshToken: (clientId: string) => Promise<boolean>;
@@ -139,6 +140,10 @@ export default ((client: ApiRequester, baseUrl: string): AuthD => ({
     return client.post(`${baseUrl}/token`, body, headers).then(Session.parse);
   },
 
+  samlLogOut() {
+    return client.get(`${baseUrl}/saml/logout`, null, null);
+  },
+
   initiateIdpAuthentication: async (domain: string, redirectUrl: string) => {
     const body = {
       domain,

package/src/checker/checks/webrtc-transport.ts

@@ -1,9 +1,9 @@
 import WebRTCClient from '../../web-rtc-client';
-import { WazoSession } from '../../domain/types';
+import { SipCall } from '../../domain/types';
 
 export default {
   name: 'WebRTC Transport (WS) ~30s',
-  check: (server: string, session: WazoSession): Promise<void> => new Promise((resolve, reject) => {
+  check: (server: string, session: SipCall): Promise<void> => new Promise((resolve, reject) => {
     const [host, port = 443] = server.split(':');
 
     const client = new WebRTCClient({

package/src/checker/checks/webrtc.ts

@@ -1,5 +1,5 @@
 /* global MediaStream */
-import { getSipSessionId } from '../../utils/sdp';
+import { getSipCallId } from '../../utils/sdp';
 import WebRTCClient from '../../web-rtc-client';
 
 export default {
@@ -36,7 +36,7 @@ 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');
       }
 

package/src/domain/CallLog.ts

@@ -118,8 +118,7 @@ export default class CallLog {
 
   static parseNew(plain: CallLogResponse, session: Session): CallLog {
     const callLog: CallLog = CallLog.parse(plain);
-    // @TODO: FIXME add verification declined vs missed call
-    callLog.newMissedCall = session && session.hasExtension((plain.requested_extension || plain.destination_extension) as string) && !plain.answered;
+    callLog.newMissedCall = session && plain.destination_user_uuid === session.uuid && !plain.answered;
     return callLog;
   }
 
@@ -203,7 +202,7 @@ export default class CallLog {
   }
 
   isAnOutgoingCall(session: Session): boolean {
-    console.warn(`@wazo/sdk 
+    console.warn(`@wazo/sdk
       CallLog.isAnOutgoingcall(session) method is obsolete.
       Please use CallLog.isOutgoing(session).
     `);

package/src/domain/CallSession.ts

@@ -2,7 +2,7 @@ import { SessionState } from 'sip.js/lib/api/session-state';
 import Call from './Call';
 import newFrom from '../utils/new-from';
 import updateFrom from '../utils/update-from';
-import { WazoSession } from './types';
+import { SipCall } from './types';
 
 type CallSessionArguments = {
   answered: boolean;
@@ -32,7 +32,7 @@ type CallSessionArguments = {
   screensharing: boolean;
   recording: boolean;
   recordingPaused: boolean;
-  sipSession?: WazoSession;
+  sipSession?: SipCall;
   conference: boolean;
 };
 export default class CallSession {
@@ -96,7 +96,7 @@ export default class CallSession {
 
   recordingPaused: boolean;
 
-  sipSession: WazoSession | undefined;
+  sipSession: SipCall | undefined;
 
   conference: boolean;