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

package/lib/domain/Phone/WebRTCPhone.js

@@ -125,7 +125,7 @@ class WebRTCPhone extends Emitter_1.default {
         return __awaiter(this, arguments, void 0, function* (callSession, constraints = null, conference = false, audioOnly = false, iceRestart = false) {
             const sipSession = this.findSipSession(callSession);
             logger.info('WebRTC phone - send reinvite', {
-                sessionId: sipSession ? sipSession.id : null,
+                sessionId: (0, sdp_1.getSipCallId)(sipSession),
                 constraints,
                 audioOnly,
                 conference,
@@ -283,7 +283,7 @@ class WebRTCPhone extends Emitter_1.default {
                 content: {
                     type: exports.ON_MESSAGE_TRACK_UPDATED,
                     update: isUpgrade ? 'upgrade' : 'downgrade',
-                    sipCallId: sipSession && (0, sdp_1.getSipSessionId)(sipSession),
+                    sipCallId: sipSession && (0, sdp_1.getSipCallId)(sipSession),
                     callId: callSession ? callSession.callId : null,
                     number: callSession ? callSession.number : null,
                 },
@@ -291,7 +291,7 @@ class WebRTCPhone extends Emitter_1.default {
         }, 2500);
     }
     _bindEvents(sipSession) {
-        const sipSessionId = (0, sdp_1.getSipSessionId)(sipSession);
+        const sipSessionId = (0, sdp_1.getSipCallId)(sipSession);
         if (sipSession instanceof api_1.Invitation) {
             // Monkey patch to know when canceled with the CANCEL message
             const onCancel = sipSession._onCancel.bind(sipSession);
@@ -303,7 +303,9 @@ class WebRTCPhone extends Emitter_1.default {
                 onCancel(message);
                 const elsewhere = message.data.indexOf('cause=26') !== -1 && message.data.indexOf('completed elsewhere') !== -1;
                 const callSession = this._createCallSession(sipSession);
-                delete this.callSessions[sipSessionId];
+                if (sipSessionId) {
+                    delete this.callSessions[sipSessionId];
+                }
                 this.eventEmitter.emit(exports.ON_CALL_CANCELED, callSession, elsewhere);
             };
         }
@@ -311,14 +313,14 @@ class WebRTCPhone extends Emitter_1.default {
             switch (newState) {
                 case session_state_1.SessionState.Terminating:
                     logger.info('WebRTC phone - call terminating', {
-                        sipId: sipSession.id,
+                        sipId: (0, sdp_1.getSipCallId)(sipSession),
                     });
                     this.eventEmitter.emit(exports.ON_CALL_ENDING, this._createCallSession(sipSession));
                     break;
                 case session_state_1.SessionState.Terminated:
                     {
                         logger.info('WebRTC phone - call terminated', {
-                            sipId: sipSession.id,
+                            sipId: (0, sdp_1.getSipCallId)(sipSession),
                         });
                         // Should be called before `_onCallTerminated` or the callCount will not decrement...
                         const callSession = this._createCallSession(sipSession);
@@ -345,7 +347,7 @@ 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: (0, sdp_1.getSipSessionId)(sipSession),
+                callId: (0, sdp_1.getSipCallId)(sipSession),
                 streamId: stream ? stream.id : null,
                 tracks: stream ? stream.getTracks() : null,
                 kind,
@@ -375,9 +377,9 @@ class WebRTCPhone extends Emitter_1.default {
             }
             const screenTrack = screenShareStream.getVideoTracks()[0];
             const sipSession = this.currentSipSession;
-            const pc = sipSession && this.client.getPeerConnection((0, sdp_1.getSipSessionId)(sipSession));
+            const pc = sipSession && this.client.getPeerConnection((0, sdp_1.getSipCallId)(sipSession));
             const sender = pc && pc.getSenders && pc.getSenders().find((s) => s && s.track && s.track.kind === 'video');
-            const localStream = sipSession && this.client.getLocalStream((0, sdp_1.getSipSessionId)(sipSession));
+            const localStream = sipSession && this.client.getLocalStream((0, sdp_1.getSipCallId)(sipSession));
             const videoTrack = localStream ? localStream.getTracks().find(track => track.kind === 'video') : null;
             const hadVideo = !!videoTrack;
             // Stop local video tracks
@@ -442,19 +444,19 @@ class WebRTCPhone extends Emitter_1.default {
     }
     _onScreenSharing(screenStream, sipSession, callSession, hadVideo, desktop) {
         const screenTrack = screenStream.getVideoTracks()[0];
-        const sipSessionId = (0, sdp_1.getSipSessionId)(sipSession);
+        const sipSessionId = (0, sdp_1.getSipCallId)(sipSession);
         const pc = this.client.getPeerConnection(sipSessionId);
         const sender = pc && pc.getSenders && pc.getSenders().find((s) => s && s.track && s.track.kind === 'video');
         logger.info('WebRTC phone - on screensharing', {
             hadVideo,
-            id: (0, sdp_1.getSipSessionId)(sipSession),
+            id: (0, sdp_1.getSipCallId)(sipSession),
             screenTrack,
         });
         if (screenTrack) {
             screenTrack.onended = () => {
                 logger.info('WebRTC phone - on screenshare ended', {
                     hadVideo,
-                    id: (0, sdp_1.getSipSessionId)(sipSession),
+                    id: (0, sdp_1.getSipCallId)(sipSession),
                     screenTrack,
                 });
                 this.eventEmitter.emit(exports.ON_SHARE_SCREEN_ENDING, this._createCallSession(sipSession, callSession));
@@ -474,13 +476,13 @@ class WebRTCPhone extends Emitter_1.default {
     }
     _onCallAccepted(sipSession, cameraEnabled) {
         logger.info('WebRTC phone - on call accepted', {
-            sipId: sipSession.id,
+            sipId: (0, sdp_1.getSipCallId)(sipSession),
             cameraEnabled,
         });
         const callSession = this._createAcceptedCallSession(sipSession, cameraEnabled);
         if (this.currentSipSession && !this.isCurrentCallSipSession(callSession)) {
             logger.info('WebRTC phone - on call accepted with another ongoing call, holding it', {
-                sipId: sipSession.id,
+                sipId: (0, sdp_1.getSipCallId)(sipSession),
                 cameraEnabled,
             });
             this.holdSipSession(this.currentSipSession, this.currentCallSession, true);
@@ -488,7 +490,7 @@ class WebRTCPhone extends Emitter_1.default {
         this.currentSipSession = sipSession;
         this.currentCallSession = callSession;
         this.eventEmitter.emit(exports.ON_TERMINATE_SOUND, callSession, 'call accepted');
-        const sipSessionId = (0, sdp_1.getSipSessionId)(sipSession);
+        const sipSessionId = (0, sdp_1.getSipCallId)(sipSession);
         if (sipSessionId) {
             this.removeIncomingSessions(sipSessionId);
         }
@@ -551,9 +553,9 @@ class WebRTCPhone extends Emitter_1.default {
         return this.currentSipSession && this.client.changeSessionVideoInputDevice(id, this.currentSipSession);
     }
     _onCallTerminated(sipSession) {
-        const sipSessionId = (0, sdp_1.getSipSessionId)(sipSession);
+        const sipSessionId = (0, sdp_1.getSipCallId)(sipSession);
         logger.info('WebRTC phone - on call terminated', {
-            sipId: sipSession.id,
+            sipId: (0, sdp_1.getSipCallId)(sipSession),
             sipSessionId,
             ids: Object.keys(this.callSessions),
         });
@@ -573,8 +575,8 @@ class WebRTCPhone extends Emitter_1.default {
         }
         if (sipSessionId) {
             this.removeIncomingSessions(sipSessionId);
+            delete this.callSessions[sipSessionId];
         }
-        delete this.callSessions[sipSessionId];
         if (isCurrentSession) {
             this.currentSipSession = undefined;
             this.currentCallSession = undefined;
@@ -582,7 +584,7 @@ class WebRTCPhone extends Emitter_1.default {
         const hasIncomingCallSession = this.hasIncomingCallSession();
         // Re-trigger incoming call event for remaining incoming calls
         logger.info('WebRTC phone - check to re-trigger incoming call', {
-            sipId: sipSession.id,
+            sipId: (0, sdp_1.getSipCallId)(sipSession),
             shouldRetrigger,
             hasIncomingCallSession,
         });
@@ -643,7 +645,7 @@ class WebRTCPhone extends Emitter_1.default {
         if (!this.currentSipSession) {
             return false;
         }
-        return this.currentSipSession && (0, sdp_1.getSipSessionId)(this.currentSipSession) === callSession.getId();
+        return this.currentSipSession && (0, sdp_1.getSipCallId)(this.currentSipSession) === callSession.getId();
     }
     hasVideo(callSession) {
         return this.client.hasVideo(callSession.getId());
@@ -677,7 +679,7 @@ class WebRTCPhone extends Emitter_1.default {
             logger.warn('no CallSession to accept.');
             return Promise.resolve(null);
         }
-        const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
+        const sipSessionId = this.getSipCallIdFromCallSession(callSession);
         if (sipSessionId && !(sipSessionId in this.callSessions)) {
             logger.warn('Call session already ended or not found, can\'t accept it.', {
                 sipSessionId,
@@ -697,7 +699,7 @@ class WebRTCPhone extends Emitter_1.default {
         this.eventEmitter.emit(exports.ON_CALL_ANSWERED, callSession);
         const sipSession = this.client.getSipSession(callSession.getId());
         if (sipSession) {
-            logger.info('accept call, session found', { sipId: sipSession.id });
+            logger.info('accept call, session found', { sipId: (0, sdp_1.getSipCallId)(sipSession) });
             return this.client.answer(sipSession, this.allowVideo ? cameraEnabled : false).then(() => {
                 return callSession.sipCallId;
             }).catch(e => {
@@ -709,7 +711,7 @@ class WebRTCPhone extends Emitter_1.default {
         const error = 'No CallSession found to accept.';
         logger.warn(error, {
             id: callSession ? callSession.getId() : 'n/a',
-            ids: this.client.getSipSessionIds(),
+            ids: this.client.getSipCallIds(),
         });
         if (callSession) {
             this.eventEmitter.emit(exports.ON_CALL_ERROR, new Error(error), callSession);
@@ -733,7 +735,7 @@ class WebRTCPhone extends Emitter_1.default {
             const sipSession = this.findSipSession(callSession);
             if (sipSession) {
                 logger.info('WebRTC rejecting', {
-                    sipId: sipSession.id,
+                    sipId: (0, sdp_1.getSipCallId)(sipSession),
                 });
                 this.client.hangup(sipSession);
             }
@@ -754,7 +756,7 @@ class WebRTCPhone extends Emitter_1.default {
         const sipSession = this.findSipSession(callSession);
         if (sipSession) {
             logger.info('WebRTC atxfer', {
-                sipId: sipSession.id,
+                sipId: (0, sdp_1.getSipCallId)(sipSession),
             });
             return this.client.atxfer(sipSession);
         }
@@ -780,12 +782,12 @@ class WebRTCPhone extends Emitter_1.default {
             return new Promise((resolve, reject) => reject(new Error('No session to resume')));
         }
         logger.info('WebRTC resuming', {
-            sipId: sipSession.id,
+            sipId: (0, sdp_1.getSipCallId)(sipSession),
         });
         // Hold current session if different from the current one (we don't want 2 sessions active at the same time).
-        if (this.currentSipSession && this.currentSipSession.id !== sipSession.id) {
+        if (this.currentSipSession && (0, sdp_1.getSipCallId)(this.currentSipSession) !== (0, sdp_1.getSipCallId)(sipSession)) {
             logger.info('WebRTC hold call after resume', {
-                id: this.currentSipSession.id,
+                id: (0, sdp_1.getSipCallId)(this.currentSipSession),
             });
             this.holdSipSession(this.currentSipSession, this.currentCallSession, withEvent);
         }
@@ -801,10 +803,10 @@ class WebRTCPhone extends Emitter_1.default {
             if (!sipSession) {
                 return new Promise((resolve, reject) => reject(new Error('No session to hold')));
             }
-            const sessionId = (0, sdp_1.getSipSessionId)(sipSession);
+            const sessionId = (0, sdp_1.getSipCallId)(sipSession);
             const hasVideo = this.client.hasLocalVideo(sessionId);
             logger.info('WebRTC hold sip session', {
-                sipId: sipSession.id,
+                sipId: (0, sdp_1.getSipCallId)(sipSession),
                 hasVideo,
             });
             // Stop screenshare if needed
@@ -830,7 +832,7 @@ 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 = (0, sdp_1.getSipSessionId)(sipSession);
+            const sessionId = (0, sdp_1.getSipCallId)(sipSession);
             logger.info('WebRTC unhold sip session', {
                 sessionId,
                 isConference,
@@ -883,7 +885,7 @@ class WebRTCPhone extends Emitter_1.default {
             return;
         }
         logger.info('WebRTC muting', {
-            sipId: sipSession.id,
+            sipId: (0, sdp_1.getSipCallId)(sipSession),
         });
         this.client.mute(sipSession);
         const newCallSession = this._createCallSession(sipSession, callSession, {
@@ -905,7 +907,7 @@ class WebRTCPhone extends Emitter_1.default {
             return;
         }
         logger.info('WebRTC unmuting', {
-            sipId: sipSession.id,
+            sipId: (0, sdp_1.getSipCallId)(sipSession),
         });
         this.client.unmute(sipSession);
         const newCallSession = this._createCallSession(sipSession, callSession, {
@@ -931,7 +933,7 @@ class WebRTCPhone extends Emitter_1.default {
             return;
         }
         logger.info('WebRTC turn camera on', {
-            sipId: sipSession.id,
+            sipId: (0, sdp_1.getSipCallId)(sipSession),
         });
         this.client.toggleCameraOn(sipSession);
         this.eventEmitter.emit(exports.ON_CAMERA_RESUMED, this._createCameraResumedCallSession(sipSession, callSession));
@@ -942,7 +944,7 @@ class WebRTCPhone extends Emitter_1.default {
             return;
         }
         logger.info('WebRTC turn camera off', {
-            sipId: sipSession.id,
+            sipId: (0, sdp_1.getSipCallId)(sipSession),
         });
         this.client.toggleCameraOff(sipSession);
         this.eventEmitter.emit(exports.ON_CAMERA_DISABLED, this._createCameraDisabledCallSession(sipSession, callSession));
@@ -953,7 +955,7 @@ class WebRTCPhone extends Emitter_1.default {
             return;
         }
         logger.info('WebRTC send key', {
-            sipId: sipSession.id,
+            sipId: (0, sdp_1.getSipCallId)(sipSession),
             tone,
         });
         this.client.sendDTMF(sipSession, tone);
@@ -1018,7 +1020,7 @@ class WebRTCPhone extends Emitter_1.default {
             return;
         }
         logger.info('WebRTC transfer', {
-            sipId: sipSession.id,
+            sipId: (0, sdp_1.getSipCallId)(sipSession),
             target,
         });
         this.client.transfer(sipSession, target);
@@ -1028,8 +1030,8 @@ class WebRTCPhone extends Emitter_1.default {
             const sipSession = this.client.getSipSession(source.sipCallId);
             const sipSessionTarget = this.client.getSipSession(destination.sipCallId);
             logger.info('WebRTC indirect transfer', {
-                sipId: sipSession ? sipSession.id : null,
-                target: sipSessionTarget ? sipSessionTarget.id : null,
+                sipId: sipSession ? (0, sdp_1.getSipCallId)(sipSession) : null,
+                target: sipSessionTarget ? (0, sdp_1.getSipCallId)(sipSessionTarget) : null,
             });
             if (!sipSessionTarget) {
                 return Promise.reject();
@@ -1059,9 +1061,9 @@ 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 = (0, sdp_1.getSipSessionId)(sipSession);
+            const sipSessionId = (0, sdp_1.getSipCallId)(sipSession);
             logger.info('WebRTC hangup', {
-                sipId: sipSession.id,
+                sipId: (0, sdp_1.getSipCallId)(sipSession),
                 sipSessionId,
             });
             yield this.client.hangup(sipSession);
@@ -1219,7 +1221,7 @@ class WebRTCPhone extends Emitter_1.default {
             const autoAnswer = sipSession.request.getHeader('Answer-Mode') === 'Auto';
             const withVideo = this.allowVideo ? wantsToDoVideo : false;
             logger.info('WebRTC invite received', {
-                sipId: sipSession.id,
+                sipId: (0, sdp_1.getSipCallId)(sipSession),
                 withVideo,
                 autoAnswer,
             });
@@ -1251,12 +1253,12 @@ 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[(0, sdp_1.getSipSessionId)(sipSession)]);
+                this._createCallSession(sipSession, this.callSessions[(0, sdp_1.getSipCallId)(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 = (0, sdp_1.getSipSessionId)(sipSession);
+            const sessionId = (0, sdp_1.getSipCallId)(sipSession);
             const hasSession = (sessionId in this.callSessions);
             logger.info('WebRTC call accepted', {
                 sipId: sessionId,
@@ -1279,11 +1281,11 @@ class WebRTCPhone extends Emitter_1.default {
             this.eventEmitter.emit(exports.ON_CALL_ERROR, e);
         });
         this.client.on(this.client.REJECTED, session => {
-            logger.info('WebRTC call rejected', session.id);
+            logger.info('WebRTC call rejected', { id: (0, sdp_1.getSipCallId)(session) });
             const callSession = this._createCallSession(session);
             // `_createCallSession` add the CallSession in `this.callSessions`.
             // so we have to remove it or `callCount` will always return 1 even if the call was rejected
-            delete this.callSessions[(0, sdp_1.getSipSessionId)(session)];
+            delete this.callSessions[(0, sdp_1.getSipCallId)(session)];
             callSession.endTime = new Date();
             this.eventEmitter.emit(exports.ON_CALL_REJECTED, callSession);
         });
@@ -1300,7 +1302,7 @@ class WebRTCPhone extends Emitter_1.default {
             const callSession = this._createCallSession(session, null, {
                 incoming: false,
                 ringing: true,
-                callId: (0, sdp_1.getSipSessionId)(session),
+                callId: (0, sdp_1.getSipCallId)(session),
             });
             this.eventEmitter.emit(exports.ON_PROGRESS, callSession, this.audioOutputDeviceId, this.audioOutputVolume);
         });
@@ -1309,7 +1311,7 @@ class WebRTCPhone extends Emitter_1.default {
             const callSession = this._createCallSession(session, null, {
                 incoming: false,
                 ringing: true,
-                callId: (0, sdp_1.getSipSessionId)(session),
+                callId: (0, sdp_1.getSipCallId)(session),
             });
             this.eventEmitter.emit(exports.ON_EARLY_MEDIA, callSession, this.audioOutputDeviceId, this.audioOutputVolume);
         });
@@ -1366,12 +1368,12 @@ 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((0, sdp_1.getSipSessionId)(session));
+            const callSession = this.getCallSession((0, sdp_1.getSipCallId)(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 = (0, sdp_1.getSipSessionId)(sipSession);
+            const sipSessionId = (0, sdp_1.getSipCallId)(sipSession);
             const localStream = this.client.getLocalStream(sipSessionId);
             const callSession = this.callSessions[sipSessionId];
             logger.info('Updrading directly in screensharing mode', {
@@ -1383,10 +1385,10 @@ class WebRTCPhone extends Emitter_1.default {
     }
     // Find a corresponding sipSession from a CallSession
     findSipSession(callSession) {
-        const keys = this.client.getSipSessionIds();
+        const keys = this.client.getSipCallIds();
         const keyIndex = keys.findIndex(sessionId => callSession && callSession.isId(sessionId));
         if (keyIndex === -1) {
-            const currentSipSessionId = this.currentSipSession ? (0, sdp_1.getSipSessionId)(this.currentSipSession) : this.client.getSipSessionIds()[0];
+            const currentSipSessionId = this.currentSipSession ? (0, sdp_1.getSipCallId)(this.currentSipSession) : this.client.getSipCallIds()[0];
             return currentSipSessionId ? this.client.getSipSession(currentSipSessionId) : null;
         }
         return this.client.getSipSession(keys[keyIndex]);
@@ -1394,12 +1396,12 @@ class WebRTCPhone extends Emitter_1.default {
     getCallSession(sipSessionId) {
         return this.callSessions[sipSessionId];
     }
-    getSipSessionIdFromCallSession(callSession) {
+    getSipCallIdFromCallSession(callSession) {
         const sipSession = this.findSipSession(callSession);
-        return sipSession ? (0, sdp_1.getSipSessionId)(sipSession) : null;
+        return sipSession ? (0, sdp_1.getSipCallId)(sipSession) : null;
     }
     _updateCallSession(callSession) {
-        const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
+        const sipSessionId = this.getSipCallIdFromCallSession(callSession);
         if (sipSessionId) {
             this.callSessions[sipSessionId] = callSession;
         }
@@ -1472,7 +1474,7 @@ class WebRTCPhone extends Emitter_1.default {
         const number = (0, sdp_1.getCallNumber)(sipSession);
         const displayName = (0, sdp_1.getCallDisplayName)(sipSession);
         const { state } = sipSession || {};
-        const sessionId = (0, sdp_1.getSipSessionId)(sipSession);
+        const sessionId = (0, sdp_1.getSipCallId)(sipSession);
         fromSession = fromSession || this.callSessions[sessionId];
         const callSession = new CallSession_1.default(Object.assign({ callId: fromSession && fromSession.callId, sipCallId: sessionId, sipStatus: state, displayName, 
             // @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.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;
+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.ApiCall = exports.SFUNotAvailableError = exports.InvalidState = exports.InvalidStateTransition = exports.ServerError = exports.BadResponse = exports.IssueReporter = exports.Emitter = void 0;
+exports.Softphone = exports.Call = 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; } });
@@ -35,7 +35,7 @@ Object.defineProperty(exports, "InvalidState", { enumerable: true, get: function
 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");
-Object.defineProperty(exports, "Call", { enumerable: true, get: function () { return __importDefault(Call_1).default; } });
+Object.defineProperty(exports, "ApiCall", { enumerable: true, get: function () { return __importDefault(Call_1).default; } });
 var CallLog_1 = require("./domain/CallLog");
 Object.defineProperty(exports, "CallLog", { enumerable: true, get: function () { return __importDefault(CallLog_1).default; } });
 var Recording_1 = require("./domain/Recording");
@@ -121,6 +121,10 @@ var ExternalApp_1 = require("./domain/ExternalApp");
 Object.defineProperty(exports, "ExternalApp", { enumerable: true, get: function () { return __importDefault(ExternalApp_1).default; } });
 var WazoSessionDescriptionHandler_1 = require("./lib/WazoSessionDescriptionHandler");
 Object.defineProperty(exports, "SessionDescriptionHandler", { enumerable: true, get: function () { return __importDefault(WazoSessionDescriptionHandler_1).default; } });
+var call_1 = require("./voice/call");
+Object.defineProperty(exports, "Call", { enumerable: true, get: function () { return __importDefault(call_1).default; } });
+var softphone_1 = require("./voice/softphone");
+Object.defineProperty(exports, "Softphone", { enumerable: true, get: function () { return softphone_1.Softphone; } });
 __exportStar(require("./types/WebSocketMessage"), exports);
 __exportStar(require("./utils/PhoneNumberUtil"), exports);
 __exportStar(require("./utils/string"), exports);

package/lib/service/CallApi.js

@@ -51,9 +51,9 @@ class CallApi {
             return (0, getApiClient_1.default)().confd.getUserLineSip(session.uuid, lineToUse ? lineToUse.id : '');
         });
     }
-    static cancelCall(callSession) {
+    static cancelCall(call) {
         return __awaiter(this, void 0, void 0, function* () {
-            return (0, getApiClient_1.default)().calld.cancelCall(callSession.callId);
+            return (0, getApiClient_1.default)().calld.cancelCall(call.apiId || '');
         });
     }
     static makeCall(callFromLine_1, extension_1) {

package/lib/simple/Auth.js

@@ -154,6 +154,11 @@ class Auth {
             return this._onAuthenticated(rawSession);
         });
     }
+    samlLogOut() {
+        return __awaiter(this, void 0, void 0, function* () {
+            return yield this.logout(false, true);
+        });
+    }
     initiateIdpAuthentication(domain, redirectUrl) {
         return __awaiter(this, void 0, void 0, function* () {
             try {
@@ -206,20 +211,26 @@ class Auth {
         });
     }
     logout() {
-        return __awaiter(this, arguments, void 0, function* (deleteRefreshToken = true) {
+        return __awaiter(this, arguments, void 0, function* (deleteRefreshToken = true, saml = false) {
             var _a;
             try {
                 index_1.default.Websocket.close(true);
-                if (this.clientId && deleteRefreshToken) {
+                if (!saml && this.clientId && deleteRefreshToken) {
                     yield (0, getApiClient_1.default)().auth.deleteRefreshToken(this.clientId);
                 }
             }
             catch (e) {
                 // Nothing to
             }
+            let response;
             try {
                 if ((_a = this.session) === null || _a === void 0 ? void 0 : _a.token) {
-                    yield (0, getApiClient_1.default)().auth.logOut(this.session.token);
+                    if (saml) {
+                        response = yield (0, getApiClient_1.default)().auth.samlLogOut();
+                    }
+                    else {
+                        yield (0, getApiClient_1.default)().auth.logOut(this.session.token);
+                    }
                 }
             }
             catch (e) {
@@ -231,6 +242,7 @@ class Auth {
             this.authenticated = false;
             this.usingEdgeServer = undefined;
             (0, getApiClient_1.setFetchOptions)({});
+            return response;
         });
     }
     setOnHostFromHeaders(callback) {

package/lib/simple/Phone.js

@@ -237,11 +237,11 @@ class Phone extends Emitter_1.default {
     stopNetworkMonitoring(callSession) {
         return this.phone ? this.phone.stopNetworkMonitoring(callSession) : null;
     }
-    getSipSessionId(sipSession) {
+    getSipCallId(sipSession) {
         if (!sipSession || !this.phone) {
             return null;
         }
-        return (0, sdp_1.getSipSessionId)(sipSession);
+        return (0, sdp_1.getSipCallId)(sipSession);
     }
     sendMessage(body, sipSession, contentType = 'text/plain') {
         const toSipSession = sipSession || this.getCurrentSipSession();

package/lib/simple/index.js

@@ -68,7 +68,7 @@ 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 softphone_1 = __importStar(require("../voice/softphone"));
 const call_1 = __importDefault(require("../voice/call"));
 const Wazo = {
     Auth: Auth_1.default,
@@ -117,6 +117,7 @@ const Wazo = {
         SwitchboardCall: SwitchboardCall_1.default,
         WebRTCPhone: WebRTCPhone_1.default,
         Meeting: Meeting_1.default,
+        SoftphoneClass: softphone_1.Softphone,
     },
     // Api
     get api() {

package/lib/simple/room/Room.js

@@ -372,7 +372,7 @@ class Room extends Emitter_1.default {
             logger.info('on room audio stream');
             this.audioStream = stream;
             if (!this.roomAudioElement) {
-                const sessionId = (0, sdp_1.getSipSessionId)((_b = index_1.default.Phone.phone) === null || _b === void 0 ? void 0 : _b.currentSipSession);
+                const sessionId = (0, sdp_1.getSipCallId)((_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;
             }

package/lib/simple/room/SipRoom.js

@@ -136,7 +136,7 @@ class SipRoom extends Room_1.default {
     }
     _getCurrentSipCallIs() {
         var _a;
-        return (0, sdp_1.getSipSessionId)((_a = index_1.default.Phone.phone) === null || _a === void 0 ? void 0 : _a.currentSipSession);
+        return (0, sdp_1.getSipCallId)((_a = index_1.default.Phone.phone) === null || _a === void 0 ? void 0 : _a.currentSipSession);
     }
 }
 exports.default = SipRoom;