@wazo/sdk 1.0.0-alpha.8 → 1.0.0-alpha.9

package/esm/utils/sdp.js

@@ -1,7 +1,7 @@
 /* eslint-disable no-underscore-dangle */
 import sdpParser from 'sdp-transform';
 import { UserAgent } from 'sip.js/lib/api/user-agent';
-import { Inviter } from 'sip.js/lib/api';
+import { Inviter, Invitation } from 'sip.js/lib/api';
 export const SIP_ID_MIN_LENGTH = 20;
 export const getCandidates = (rawSdp) => {
     if (!rawSdp) {
@@ -110,18 +110,19 @@ export const getSipCallId = (sipSession) => {
     if (!sipSession) {
         return '';
     }
-    const message = sipSession.message;
-    if (message?.callId) {
-        return message.callId.substring(0, SIP_ID_MIN_LENGTH);
-    }
     // For Inviter
     // @ts-ignore: private
     if (sipSession instanceof Inviter && sipSession.outgoingRequestMessage) {
         // @ts-ignore: private
-        return sipSession.outgoingRequestMessage.callId.substring(0, SIP_ID_MIN_LENGTH);
+        return sipSession.outgoingRequestMessage.callId;
     }
     // For Invitation
-    return (sipSession.id || '').substring(0, SIP_ID_MIN_LENGTH);
+    // @ts-ignore: private
+    if (sipSession instanceof Invitation && sipSession.incomingInviteRequest.message) {
+        // @ts-ignore: private
+        return sipSession.incomingInviteRequest.message.callId;
+    }
+    return sipSession.id;
 };
 // We need to replace 0.0.0.0 to 127.0.0.1 in the sdp to avoid MOH during a createOffer.
 export const replaceLocalIpModifier = (description) => Promise.resolve({

package/esm/voice/call.js

@@ -79,7 +79,6 @@ class Call extends EventEmitter {
             },
         };
         const call = new Call(sipCall, softphone);
-        // call._sendAction(Actions.ES);
         call.apiId = apiCall.id;
         call.answerTime = new Date(+apiCall.startingTime);
         // Update call state
@@ -109,7 +108,6 @@ class Call extends EventEmitter {
         this.sipCall = sipCall;
         this.number = getCallNumber(sipCall);
         this.displayName = getCallDisplayName(sipCall);
-        this._bindEvents();
     }
     async accept(options = {}) {
         assertCan(this.callActor, Actions.ACCEPT);
@@ -401,6 +399,7 @@ class Call extends EventEmitter {
         this._onScreenSharing(localStream, false, desktop);
     }
     onMakeCall() {
+        this._bindEvents();
         this._sendAction(Actions.MAKE_CALL);
     }
     onAccepted() {

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 { getSipCallId, makeURI, replaceLocalIpModifier, SIP_ID_MIN_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'];
@@ -173,7 +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);
@@ -456,7 +456,7 @@ 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,
                     });
@@ -486,7 +486,7 @@ 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) {
@@ -502,7 +502,7 @@ 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);
@@ -510,7 +510,7 @@ 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;
         });
     }
@@ -567,7 +567,7 @@ 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) {
@@ -621,13 +621,13 @@ 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);
     }
@@ -664,13 +664,13 @@ 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);
     }
@@ -844,7 +844,7 @@ export default class WebRTCClient extends Emitter {
             return;
         }
         logger.info('send WebRTC message', {
-            sipId: sipSession.id,
+            sipId: getSipCallId(sipSession),
             contentType,
         });
         try {
@@ -1218,7 +1218,7 @@ export default class WebRTCClient extends Emitter {
         this.sipSessions[id] = session;
     }
     getSipSession(id) {
-        return id in this.sipSessions ? this.sipSessions[id.substring(0, SIP_ID_MIN_LENGTH)] : null;
+        return id in this.sipSessions ? this.sipSessions[id] : null;
     }
     getSipCallIds() {
         return Object.keys(this.sipSessions);
@@ -1438,17 +1438,14 @@ export default class WebRTCClient extends Emitter {
     }
     async _onProgress(session, early = false) {
         await this._setupMedias(session);
-        const sessionId = getSipCallId(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,
         });
@@ -1638,18 +1635,18 @@ 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 {
@@ -1661,12 +1658,12 @@ 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 {
@@ -1678,7 +1675,7 @@ 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 {
@@ -1690,7 +1687,7 @@ 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/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/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/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,
@@ -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);
@@ -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);
@@ -553,7 +555,7 @@ class WebRTCPhone extends Emitter_1.default {
     _onCallTerminated(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,
         });
@@ -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 => {
@@ -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);
         }
@@ -804,7 +806,7 @@ class WebRTCPhone extends Emitter_1.default {
             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
@@ -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();
@@ -1061,7 +1063,7 @@ class WebRTCPhone extends Emitter_1.default {
             }
             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,
             });
@@ -1279,7 +1281,7 @@ 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

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/types/api/auth.d.ts

@@ -23,6 +23,9 @@ 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>;

package/lib/types/api/auth.d.ts.map

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+{"version":3,"file":"auth.d.ts","sourceRoot":"","sources":["../../../src/api/auth.ts"],"names":[],"mappings":"AAAA,OAAO,YAAY,MAAM,wBAAwB,CAAC;AAClD,OAAO,KAAK,EAAE,IAAI,EAAE,MAAM,EAAE,KAAK,EAAE,IAAI,EAAE,cAAc,EAAE,mBAAmB,EAAE,iBAAiB,EAAE,kBAAkB,EAAE,oBAAoB,EAAE,iBAAiB,EAAE,eAAe,EAAE,MAAM,iBAAiB,CAAC;AACvM,OAAO,OAAqB,MAAM,mBAAmB,CAAC;AAEtD,eAAO,MAAM,oBAAoB,cAAc,CAAC;AAChD,eAAO,MAAM,iBAAiB,cAAc,CAAC;AAC7C,eAAO,MAAM,kBAAkB,OAAO,CAAC;AAEvC,KAAK,WAAW,GAAG;IACjB,QAAQ,EAAE,MAAM,CAAC;IACjB,QAAQ,EAAE,MAAM,CAAC;IACjB,OAAO,CAAC,EAAE,MAAM,CAAC;IACjB,UAAU,EAAE,MAAM,CAAC;IACnB,MAAM,CAAC,EAAE,OAAO,CAAC;IACjB,QAAQ,CAAC,EAAE,MAAM,CAAC;IAClB,UAAU,CAAC,EAAE,MAAM,CAAC;CACrB,CAAC;AAQF,KAAK,qBAAqB,GAAG;IAC3B,QAAQ,CAAC,EAAE,MAAM,CAAC;IAClB,KAAK,CAAC,EAAE,MAAM,CAAC;CAChB,CAAC;AAEF,MAAM,WAAW,KAAK;IACpB,UAAU,EAAE,CAAC,KAAK,EAAE,KAAK,KAAK,OAAO,CAAC,OAAO,CAAC,CAAC;IAC/C,YAAY,EAAE,CAAC,KAAK,EAAE,KAAK,KAAK,OAAO,CAAC,OAAO,GAAG,IAAI,GAAG,SAAS,CAAC,CAAC;IACpE,KAAK,CAAC,MAAM,EAAE,WAAW,GAAG,OAAO,CAAC,OAAO,GAAG,IAAI,GAAG,SAAS,CAAC,CAAE;IACjE,MAAM,EAAE,CAAC,KAAK,EAAE,KAAK,KAAK,OAAO,CAAC,cAAc,CAAC,CAAC;IAClD,SAAS,EAAE,CAAC,aAAa,EAAE,MAAM,KAAK,OAAO,CAAC,OAAO,GAAG,IAAI,GAAG,SAAS,CAAC,CAAC;IAC1E,UAAU,EAAE,MAAM,OAAO,CAAC,IAAI,GAAG;QAAE,QAAQ,EAAC,MAAM,CAAA;KAAE,CAAC,CAAC;IACtD,yBAAyB,CAAC,MAAM,EAAE,MAAM,EAAE,WAAW,EAAE,MAAM,GAAG,OAAO,CAAC,GAAG,CAAC,CAAC;IAC7E,YAAY,EAAE,CAAC,YAAY,EAAE,MAAM,EAAE,OAAO,EAAE,MAAM,EAAE,UAAU,EAAE,MAAM,EAAE,QAAQ,CAAC,EAAE,OAAO,EAAE,QAAQ,CAAC,EAAE,MAAM,EAAE,UAAU,CAAC,EAAE,MAAM,KAAK,OAAO,CAAC,OAAO,GAAG,IAAI,GAAG,SAAS,CAAC,CAAC;IAC7K,kBAAkB,EAAE,CAAC,QAAQ,EAAE,MAAM,KAAK,OAAO,CAAC,OAAO,CAAC,CAAC;IAC3D,cAAc,EAAE,CAAC,QAAQ,EAAE,IAAI,EAAE,WAAW,EAAE,MAAM,EAAE,WAAW,EAAE,MAAM,KAAK,OAAO,CAAC,OAAO,CAAC,CAAC;IAC/F,eAAe,EAAE,CAAC,QAAQ,EAAE,IAAI,EAAE,WAAW,EAAE,MAAM,EAAE,aAAa,EAAE,MAAM,GAAG,IAAI,GAAG,SAAS,EAAE,qBAAqB,EAAE,MAAM,GAAG,IAAI,GAAG,SAAS,KAAK,OAAO,CAAC,IAAI,CAAC,CAAC;IACpK,2BAA2B,EAAE,CAAC,QAAQ,EAAE,IAAI,KAAK,OAAO,CAAC,MAAM,CAAC,CAAC;IACjE,sBAAsB,EAAE,CAAC,MAAM,EAAE,qBAAqB,KAAK,OAAO,CAAC,OAAO,CAAC,CAAC;IAC5E,aAAa,EAAE,CAAC,QAAQ,EAAE,MAAM,EAAE,QAAQ,EAAE,MAAM,KAAK,OAAO,CAAC,OAAO,CAAC,CAAC;IACxE,iBAAiB,EAAE,CAAC,QAAQ,EAAE,IAAI,KAAK,OAAO,CAAC,IAAI,CAAC,CAAC;IACrD,UAAU,EAAE,CAAC,QAAQ,EAAE,MAAM,EAAE,QAAQ,EAAE,MAAM,EAAE,SAAS,EAAE,MAAM,EAAE,QAAQ,EAAE,MAAM,KAAK,OAAO,CAAC,IAAI,CAAC,CAAC;IACvG,YAAY,EAAE,CAAC,QAAQ,EAAE,IAAI,EAAE,KAAK,EAAE,MAAM,EAAE,IAAI,CAAC,EAAE,OAAO,KAAK,OAAO,CAAC,IAAI,CAAC,CAAC;IAC/E,aAAa,EAAE,CAAC,QAAQ,EAAE,IAAI,EAAE,UAAU,EAAE,IAAI,KAAK,OAAO,CAAC,IAAI,CAAC,CAAC;IACnE,sBAAsB,EAAE,CAAC,MAAM,EAAE,MAAM,EAAE,KAAK,OAAO,CAAC,GAAG,CAAC,CAAC;IAC3D,gBAAgB,EAAE,CAAC,QAAQ,EAAE,IAAI,EAAE,UAAU,EAAE,IAAI,KAAK,OAAO,CAAC,GAAG,CAAC,CAAC;IACrE,YAAY,EAAE,CAAC,QAAQ,EAAE,IAAI,EAAE,SAAS,EAAE,IAAI,KAAK,OAAO,CAAC,GAAG,CAAC,CAAC;IAChE,cAAc,EAAE,CAAC,SAAS,EAAE,IAAI,KAAK,OAAO,CAAC,GAAG,CAAC,CAAC;IAClD,eAAe,EAAE,CAAC,QAAQ,EAAE,IAAI,EAAE,SAAS,EAAE,IAAI,KAAK,OAAO,CAAC,IAAI,CAAC,CAAC;IACpE,OAAO,EAAE,CAAC,QAAQ,EAAE,IAAI,KAAK,OAAO,CAAC,eAAe,CAAC,CAAC;IACtD,cAAc,EAAE,CAAC,QAAQ,EAAE,IAAI,KAAK,OAAO,CAAC,GAAG,CAAC,CAAC;IACjD,iBAAiB,EAAE,CAAC,QAAQ,EAAE,IAAI,EAAE,YAAY,EAAE,IAAI,KAAK,OAAO,CAAC,IAAI,CAAC,CAAC;IACzE,SAAS,EAAE,MAAM,OAAO,CAAC,iBAAiB,CAAC,CAAC;IAC5C,UAAU,EAAE,CAAC,QAAQ,EAAE,IAAI,KAAK,OAAO,CAAC,OAAO,CAAC,CAAC;IACjD,WAAW,EAAE,MAAM,OAAO,CAAC,mBAAmB,CAAC,CAAC;IAChD,SAAS,EAAE,CAAC,UAAU,EAAE,IAAI,KAAK,OAAO,CAAC,iBAAiB,CAAC,CAAC;IAC5D,YAAY,EAAE,CAAC,IAAI,EAAE,MAAM,KAAK,OAAO,CAAC,MAAM,CAAC,CAAC;IAChD,YAAY,EAAE,CAAC,IAAI,EAAE,IAAI,EAAE,IAAI,EAAE,MAAM,EAAE,OAAO,EAAE,MAAM,EAAE,KAAK,EAAE,MAAM,EAAE,OAAO,EAAE,KAAK,CAAC,MAAM,CAAC,MAAM,EAAE,GAAG,CAAC,CAAC,KAAK,OAAO,CAAC,MAAM,CAAC,CAAC;IACjI,YAAY,EAAE,CAAC,IAAI,EAAE,IAAI,KAAK,OAAO,CAAC,OAAO,CAAC,CAAC;IAC/C,WAAW,EAAE,CAAC,IAAI,EAAE,MAAM,KAAK,OAAO,CAAC,IAAI,CAAC,CAAC;IAC7C,UAAU,EAAE,MAAM,OAAO,CAAC,kBAAkB,CAAC,CAAC;IAC9C,WAAW,EAAE,CAAC,IAAI,EAAE,IAAI,KAAK,OAAO,CAAC,OAAO,CAAC,CAAC;IAC9C,YAAY,EAAE,CAAC,IAAI,EAAE,MAAM,EAAE,WAAW,EAAE,MAAM,EAAE,YAAY,EAAE,KAAK,CAAC,MAAM,CAAC,MAAM,EAAE,GAAG,CAAC,CAAC,KAAK,OAAO,CAAC,IAAI,CAAC,CAAC;IAC7G,YAAY,EAAE,MAAM,OAAO,CAAC,oBAAoB,CAAC,CAAC;IAClD,YAAY,EAAE,CAAC,UAAU,EAAE,IAAI,KAAK,OAAO,CAAC,OAAO,CAAC,CAAC;IACrD,YAAY,EAAE,CAAC,QAAQ,EAAE,IAAI,KAAK,OAAO,CAAC,GAAG,CAAC,CAAC;IAC/C,gBAAgB,EAAE,CAAC,QAAQ,EAAE,IAAI,EAAE,QAAQ,EAAE,MAAM,KAAK,OAAO,CAAC,MAAM,CAAC,CAAC;IACxE,kBAAkB,EAAE,CAAC,QAAQ,EAAE,IAAI,EAAE,QAAQ,EAAE,MAAM,KAAK,OAAO,CAAC,MAAM,CAAC,CAAC;IAC1E,mBAAmB,EAAE,CAAC,QAAQ,EAAE,IAAI,EAAE,QAAQ,EAAE,MAAM,KAAK,OAAO,CAAC,IAAI,CAAC,CAAC;CAC1E;iCAEwB,YAAY,WAAW,MAAM,KAAG,KAAK;AAA9D,wBAqOI"}

package/lib/types/domain/CallLog.d.ts.map

@@ -1 +1 @@
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package/lib/types/domain/Phone/WebRTCPhone.d.ts

@@ -3,7 +3,7 @@ import { OutgoingInviteRequest } from 'sip.js/lib/core';
 import { Inviter } from 'sip.js/lib/api';
 import CallSession from '../CallSession';
 import type { Phone, AvailablePhoneOptions } from './Phone';
-import WazoWebRTCClient from '../../web-rtc-client';
+import WebRTCClient from '../../web-rtc-client';
 import Emitter from '../../utils/Emitter';
 import { WazoSession } from '../types';
 export declare const ON_USER_AGENT = "onUserAgent";
@@ -52,7 +52,7 @@ export declare const MESSAGE_TYPE_CHAT = "message/TYPE_CHAT";
 export declare const MESSAGE_TYPE_SIGNAL = "message/TYPE_SIGNAL";
 export declare const events: string[];
 export default class WebRTCPhone extends Emitter implements Phone {
-    client: WazoWebRTCClient;
+    client: WebRTCClient;
     allowVideo: boolean;
     callSessions: Record<string, CallSession>;
     incomingSessions: string[];
@@ -69,7 +69,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     currentScreenShare: Record<string, any> | null;
     lastScreenShare: Record<string, any> | null;
     shouldSendReinvite: boolean;
-    constructor(client: WazoWebRTCClient, audioOutputDeviceId: string | undefined, allowVideo?: boolean, audioRingDeviceId?: string);
+    constructor(client: WebRTCClient, audioOutputDeviceId: string | undefined, allowVideo?: boolean, audioRingDeviceId?: string);
     register(): Promise<void>;
     unregister(): Promise<void> | null;
     stop(): void;