@wazo/sdk 0.48.2 → 0.49.0

package/dist/index.js

@@ -1743,6 +1743,7 @@ class CallSession {
     answeredBySystem;
     dialedExtension;
     diversion;
+    assertedIdentity;
     ringing;
     paused;
     sipCallId;
@@ -1760,7 +1761,7 @@ class CallSession {
     recordingState;
     sipSession;
     conference;
-    constructor({ answered, answeredBySystem, isCaller, displayName, callId, muted, videoMuted, number, paused, ringing, startTime, creationTime, endTime, cameraEnabled, dialedExtension, diversion, sipCallId, sipStatus, callerNumber, call, autoAnswer, ignored, screensharing, recording, recordingPaused, recordingState, videoRemotelyDowngraded, sipSession, answerTime, conference }){
+    constructor({ answered, answeredBySystem, isCaller, displayName, callId, muted, videoMuted, number, paused, ringing, startTime, creationTime, endTime, cameraEnabled, dialedExtension, diversion, assertedIdentity, sipCallId, sipStatus, callerNumber, call, autoAnswer, ignored, screensharing, recording, recordingPaused, recordingState, videoRemotelyDowngraded, sipSession, answerTime, conference }){
         this.callId = callId;
         this.sipCallId = sipCallId;
         this.displayName = displayName;
@@ -1779,6 +1780,7 @@ class CallSession {
         this.cameraEnabled = cameraEnabled;
         this.dialedExtension = dialedExtension || '';
         this.diversion = diversion;
+        this.assertedIdentity = assertedIdentity || void 0;
         this.call = call;
         this.sipStatus = sipStatus;
         this.autoAnswer = autoAnswer || false;
@@ -5228,10 +5230,24 @@ const transportEvents = [
 ];
 class CanceledCallError extends Error {
 }
+const toAssertedIdentity = (identity)=>{
+    if (!identity) return null;
+    const number = identity.uri?.user || void 0;
+    const displayName = identity.displayName || number;
+    if (!displayName) return null;
+    return {
+        displayName,
+        number
+    };
+};
 const MAX_REGISTER_TRIES = 5;
 const NO_ANSWER_TIMEOUT = 86400;
+let liveClientCount = 0;
 class WebRTCClient extends Emitter {
     clientId;
+    _countedAsLive;
+    _closed;
+    lastTransportMessageAt;
     config;
     uaConfigOverrides;
     userAgent;
@@ -5302,11 +5318,32 @@ class WebRTCClient extends Emitter {
         web_rtc_client_logger.info('sdk webrtc constructor', {
             clientId: this.clientId
         });
+        liveClientCount += 1;
+        this._countedAsLive = true;
+        this._closed = false;
+        this.lastTransportMessageAt = null;
+        if (liveClientCount > 1) web_rtc_client_logger.info('sdk webrtc multiple clients alive', {
+            liveClientCount,
+            clientId: this.clientId
+        });
         this.uaConfigOverrides = uaConfigOverrides;
         this.config = config;
         this.skipRegister = config.skipRegister;
         this._buildConfig(config, session).then((newConfig)=>{
             this.config = newConfig;
+            if (this._closed) {
+                web_rtc_client_logger.info('sdk webrtc constructor, client closed during config build, skipping UA creation', {
+                    clientId: this.clientId
+                });
+                return;
+            }
+            if (this.userAgent && !session) {
+                web_rtc_client_logger.info('sdk webrtc constructor, userAgent already created, skipping', {
+                    clientId: this.clientId
+                });
+                return;
+            }
+            if (this.registerer) this._cleanupRegister();
             this.userAgent = this.createUserAgent(uaConfigOverrides);
         });
         this.audioOutputDeviceId = config.audioDeviceOutput;
@@ -5365,6 +5402,19 @@ class WebRTCClient extends Emitter {
         this.audio = media.audio;
     }
     createUserAgent(uaConfigOverrides) {
+        this.lastTransportMessageAt = null;
+        if (this.userAgent) {
+            web_rtc_client_logger.warn('sdk webrtc, stopping existing UA before creating a new one', {
+                clientId: this.clientId
+            });
+            this.userAgent.stop().catch((error)=>{
+                web_rtc_client_logger.error('sdk webrtc, error stopping previous UA', {
+                    error: error?.message,
+                    clientId: this.clientId
+                });
+            });
+            this.userAgent = null;
+        }
         const uaOptions = this._createUaOptions(uaConfigOverrides);
         web_rtc_client_logger.info('sdk webrtc, creating UA', {
             uaOptions: {
@@ -5394,6 +5444,7 @@ class WebRTCClient extends Emitter {
             web_rtc_client_logger.warn('Transport connect error', e);
         });
         ua.transport.onMessage = (rawMessage)=>{
+            this.lastTransportMessageAt = Date.now();
             const message = parser_namespaceObject.Parser.parseMessage(rawMessage, ua.transport.logger);
             ua.onTransportMessage(rawMessage);
             this.eventEmitter.emit(MESSAGE, message);
@@ -5412,6 +5463,16 @@ class WebRTCClient extends Emitter {
     isRegistered() {
         return Boolean(this.registerer && this.registerer.state === registerer_state_namespaceObject.RegistererState.Registered);
     }
+    getLastTransportMessageAt() {
+        return this.lastTransportMessageAt;
+    }
+    isTransportSuspect() {
+        if (!this.isConnected()) return true;
+        if (Object.keys(this.sipSessions).length > 0) return false;
+        const threshold = this.config.transportSilenceThresholdMs;
+        if (!threshold || !this.lastTransportMessageAt) return false;
+        return Date.now() - this.lastTransportMessageAt > threshold;
+    }
     onConnect() {
         web_rtc_client_logger.info('sdk webrtc connected', {
             method: 'delegate.onConnect',
@@ -5435,6 +5496,23 @@ class WebRTCClient extends Emitter {
             this.eventEmitter.emit(UNREGISTERED);
         }
     }
+    async _isRegistrationHealthy() {
+        if (!this.isTransportSuspect()) return true;
+        web_rtc_client_logger.warn('sdk webrtc registered but transport suspect, re-registering', {
+            clientId: this.clientId,
+            connected: this.isConnected(),
+            lastTransportMessageAt: this.lastTransportMessageAt
+        });
+        this._cleanupRegister();
+        if (this.isConnected()) {
+            this.connectionPromise = this._disconnectTransport(true);
+            await this.connectionPromise;
+            this.connectionPromise = null;
+            const transport = this.userAgent?.transport;
+            if (transport && transport.state !== transport_state_namespaceObject.TransportState.Disconnected) transport.transitionState(transport_state_namespaceObject.TransportState.Disconnected);
+        }
+        return false;
+    }
     async register(tries = 0) {
         const logInfo = {
             clientId: this.clientId,
@@ -5447,6 +5525,11 @@ class WebRTCClient extends Emitter {
             skipRegister: this.skipRegister
         };
         this.forceClosed = false;
+        this._closed = false;
+        if (!this._countedAsLive) {
+            this._countedAsLive = true;
+            liveClientCount += 1;
+        }
         if (this.skipRegister) {
             web_rtc_client_logger.info('sdk webrtc skip register...', logInfo);
             return Promise.resolve();
@@ -5456,15 +5539,19 @@ class WebRTCClient extends Emitter {
             web_rtc_client_logger.info('sdk webrtc recreating User Agent');
             this.userAgent = this.createUserAgent(this.uaConfigOverrides);
         }
-        if (!this.userAgent || this.isRegistered()) {
-            web_rtc_client_logger.info('sdk webrtc registering aborted, already registered or no UA can be created');
-            return Promise.resolve();
+        if (!this.userAgent) {
+            web_rtc_client_logger.info('sdk webrtc registering aborted, no UA can be created');
+            return;
+        }
+        if (this.isRegistered() && await this._isRegistrationHealthy()) {
+            web_rtc_client_logger.info('sdk webrtc registering aborted, already registered');
+            return;
         }
         if (this.connectionPromise || this.registerer?.waiting) {
             web_rtc_client_logger.info('sdk webrtc registering aborted due to a registration in progress.', {
                 clientId: this.clientId
             });
-            return Promise.resolve();
+            return;
         }
         const registerOptions = this._isWeb() ? {} : {
             extraContactHeaderParams: [
@@ -5565,7 +5652,6 @@ class WebRTCClient extends Emitter {
                         web_rtc_client_logger.info('sdk webrtc unregistered', {
                             clientId: this.clientId
                         });
-                        if (this.registerer) this.registerer.stateChange.addListener(onRegisterStateChange);
                         this._cleanupRegister();
                         resolve();
                     }
@@ -5759,6 +5845,11 @@ class WebRTCClient extends Emitter {
             force
         });
         this.forceClosed = force;
+        this._closed = true;
+        if (this._countedAsLive) {
+            this._countedAsLive = false;
+            liveClientCount = Math.max(0, liveClientCount - 1);
+        }
         this._cleanupMedia();
         this.connectionPromise = null;
         Object.values(this.audioElements).forEach((audioElement)=>{
@@ -8647,6 +8738,15 @@ class WebRTCPhone extends Emitter {
     isRegistered() {
         return this.client && this.client.isRegistered();
     }
+    isConnected() {
+        return Boolean(this.client?.isConnected());
+    }
+    isTransportSuspect() {
+        return !this.client || this.client.isTransportSuspect();
+    }
+    getLastTransportMessageAt() {
+        return this.client?.getLastTransportMessageAt() ?? null;
+    }
     enableRinging() {
         WebRTCPhone_logger.info('WebRTC enable ringing');
         this.ringingEnabled = true;
@@ -8966,8 +9066,9 @@ class WebRTCPhone extends Emitter {
         const { state } = sipSession || {};
         const sessionId = this.getSipSessionId(sipSession);
         fromSession = fromSession || this.callSessions[sessionId];
+        const headerCallId = sipSession?.request?.getHeader?.('X-Wazo-Call-ID');
         const callSession = new CallSession({
-            callId: fromSession && fromSession.callId,
+            callId: fromSession?.callId || headerCallId || '',
             sipCallId: sessionId,
             sipStatus: state,
             displayName: identity ? identity.displayName || number : number,
@@ -8988,6 +9089,7 @@ class WebRTCPhone extends Emitter {
             recordingState: fromSession ? fromSession.recordingState : RECORDING_STATE.INACTIVE,
             sipSession,
             diversion: sipSession?.diversion,
+            assertedIdentity: toAssertedIdentity(sipSession?.assertedIdentity) || void 0,
             conference: !!fromSession && fromSession.isConference(),
             ...extra
         });
@@ -9601,6 +9703,7 @@ const protocolDebugMessages = [
 class Phone_Phone extends Emitter {
     client;
     phone;
+    connectInFlight;
     session;
     sipLine;
     SessionState;
@@ -9670,10 +9773,37 @@ class Phone_Phone extends Emitter {
         this.sipLine = sipLine || this.getPrimaryWebRtcLine();
         if (!this.sipLine) throw new Error('Sorry, no sip lines found for this user');
         options.userUuid = session.uuid || '';
-        this.connectWithCredentials(server, this.sipLine, session.displayName(), options);
+        return this.connectWithCredentials(server, this.sipLine, session.displayName(), options);
     }
-    connectWithCredentials(server, sipLine, displayName, rawOptions = {}) {
+    async connectWithCredentials(server, sipLine, displayName, rawOptions = {}) {
         if (this.phone) return;
+        if (this.connectInFlight) {
+            Phone_logger.info('connectWithCredentials: connection already in progress, reusing it and ignoring this call\'s arguments', {
+                server,
+                sipLineId: sipLine?.id
+            });
+            return this.connectInFlight;
+        }
+        this.connectInFlight = this._connectWithCredentials(server, sipLine, displayName, rawOptions).finally(()=>{
+            this.connectInFlight = null;
+        });
+        return this.connectInFlight;
+    }
+    async _connectWithCredentials(server, sipLine, displayName, rawOptions = {}) {
+        if (this.client) {
+            Phone_logger.warn('connectWithCredentials: closing previous client before creating a new one');
+            try {
+                await this.client.close();
+            } catch (error) {
+                Phone_logger.error('connectWithCredentials: error while closing previous client', {
+                    message: error?.message
+                });
+            }
+            if (this.phone) {
+                Phone_logger.info('connectWithCredentials: phone connected while closing previous client, keeping it');
+                return;
+            }
+        }
         const [host, port = 443] = server.split(':');
         const options = rawOptions;
         options.media = options.media || {
@@ -9710,6 +9840,24 @@ class Phone_Phone extends Emitter {
         }
         this.phone = null;
     }
+    async reset() {
+        Phone_logger.info('phone reset', {
+            phone: !!this.phone,
+            client: !!this.client
+        });
+        await this.disconnect().catch((error)=>{
+            Phone_logger.error('reset: error while disconnecting phone', {
+                message: error?.message
+            });
+        });
+        if (this.client) await this.client.close().catch((error)=>{
+            Phone_logger.error('reset: error while closing client', {
+                message: error?.message
+            });
+        });
+        this.phone = null;
+        this.client = null;
+    }
     async call(extension, withCamera = false, rawSipLine = null, audioOnly = false, conference = false) {
         if (!this.phone) return;
         const sipLine = rawSipLine || this.getPrimaryWebRtcLine();

package/dist/index.mjs

@@ -1681,6 +1681,7 @@ class CallSession {
     answeredBySystem;
     dialedExtension;
     diversion;
+    assertedIdentity;
     ringing;
     paused;
     sipCallId;
@@ -1698,7 +1699,7 @@ class CallSession {
     recordingState;
     sipSession;
     conference;
-    constructor({ answered, answeredBySystem, isCaller, displayName, callId, muted, videoMuted, number, paused, ringing, startTime, creationTime, endTime, cameraEnabled, dialedExtension, diversion, sipCallId, sipStatus, callerNumber, call, autoAnswer, ignored, screensharing, recording, recordingPaused, recordingState, videoRemotelyDowngraded, sipSession, answerTime, conference }){
+    constructor({ answered, answeredBySystem, isCaller, displayName, callId, muted, videoMuted, number, paused, ringing, startTime, creationTime, endTime, cameraEnabled, dialedExtension, diversion, assertedIdentity, sipCallId, sipStatus, callerNumber, call, autoAnswer, ignored, screensharing, recording, recordingPaused, recordingState, videoRemotelyDowngraded, sipSession, answerTime, conference }){
         this.callId = callId;
         this.sipCallId = sipCallId;
         this.displayName = displayName;
@@ -1717,6 +1718,7 @@ class CallSession {
         this.cameraEnabled = cameraEnabled;
         this.dialedExtension = dialedExtension || '';
         this.diversion = diversion;
+        this.assertedIdentity = assertedIdentity || void 0;
         this.call = call;
         this.sipStatus = sipStatus;
         this.autoAnswer = autoAnswer || false;
@@ -5142,10 +5144,24 @@ const transportEvents = [
 ];
 class CanceledCallError extends Error {
 }
+const toAssertedIdentity = (identity)=>{
+    if (!identity) return null;
+    const number = identity.uri?.user || void 0;
+    const displayName = identity.displayName || number;
+    if (!displayName) return null;
+    return {
+        displayName,
+        number
+    };
+};
 const MAX_REGISTER_TRIES = 5;
 const NO_ANSWER_TIMEOUT = 86400;
+let liveClientCount = 0;
 class WebRTCClient extends Emitter {
     clientId;
+    _countedAsLive;
+    _closed;
+    lastTransportMessageAt;
     config;
     uaConfigOverrides;
     userAgent;
@@ -5216,11 +5232,32 @@ class WebRTCClient extends Emitter {
         web_rtc_client_logger.info('sdk webrtc constructor', {
             clientId: this.clientId
         });
+        liveClientCount += 1;
+        this._countedAsLive = true;
+        this._closed = false;
+        this.lastTransportMessageAt = null;
+        if (liveClientCount > 1) web_rtc_client_logger.info('sdk webrtc multiple clients alive', {
+            liveClientCount,
+            clientId: this.clientId
+        });
         this.uaConfigOverrides = uaConfigOverrides;
         this.config = config;
         this.skipRegister = config.skipRegister;
         this._buildConfig(config, session).then((newConfig)=>{
             this.config = newConfig;
+            if (this._closed) {
+                web_rtc_client_logger.info('sdk webrtc constructor, client closed during config build, skipping UA creation', {
+                    clientId: this.clientId
+                });
+                return;
+            }
+            if (this.userAgent && !session) {
+                web_rtc_client_logger.info('sdk webrtc constructor, userAgent already created, skipping', {
+                    clientId: this.clientId
+                });
+                return;
+            }
+            if (this.registerer) this._cleanupRegister();
             this.userAgent = this.createUserAgent(uaConfigOverrides);
         });
         this.audioOutputDeviceId = config.audioDeviceOutput;
@@ -5279,6 +5316,19 @@ class WebRTCClient extends Emitter {
         this.audio = media.audio;
     }
     createUserAgent(uaConfigOverrides) {
+        this.lastTransportMessageAt = null;
+        if (this.userAgent) {
+            web_rtc_client_logger.warn('sdk webrtc, stopping existing UA before creating a new one', {
+                clientId: this.clientId
+            });
+            this.userAgent.stop().catch((error)=>{
+                web_rtc_client_logger.error('sdk webrtc, error stopping previous UA', {
+                    error: error?.message,
+                    clientId: this.clientId
+                });
+            });
+            this.userAgent = null;
+        }
         const uaOptions = this._createUaOptions(uaConfigOverrides);
         web_rtc_client_logger.info('sdk webrtc, creating UA', {
             uaOptions: {
@@ -5308,6 +5358,7 @@ class WebRTCClient extends Emitter {
             web_rtc_client_logger.warn('Transport connect error', e);
         });
         ua.transport.onMessage = (rawMessage)=>{
+            this.lastTransportMessageAt = Date.now();
             const message = __WEBPACK_EXTERNAL_MODULE_sip_js_lib_core_messages_parser_9974076a__.Parser.parseMessage(rawMessage, ua.transport.logger);
             ua.onTransportMessage(rawMessage);
             this.eventEmitter.emit(MESSAGE, message);
@@ -5326,6 +5377,16 @@ class WebRTCClient extends Emitter {
     isRegistered() {
         return Boolean(this.registerer && this.registerer.state === __WEBPACK_EXTERNAL_MODULE_sip_js_lib_api_registerer_state_6e58ddd6__.RegistererState.Registered);
     }
+    getLastTransportMessageAt() {
+        return this.lastTransportMessageAt;
+    }
+    isTransportSuspect() {
+        if (!this.isConnected()) return true;
+        if (Object.keys(this.sipSessions).length > 0) return false;
+        const threshold = this.config.transportSilenceThresholdMs;
+        if (!threshold || !this.lastTransportMessageAt) return false;
+        return Date.now() - this.lastTransportMessageAt > threshold;
+    }
     onConnect() {
         web_rtc_client_logger.info('sdk webrtc connected', {
             method: 'delegate.onConnect',
@@ -5349,6 +5410,23 @@ class WebRTCClient extends Emitter {
             this.eventEmitter.emit(UNREGISTERED);
         }
     }
+    async _isRegistrationHealthy() {
+        if (!this.isTransportSuspect()) return true;
+        web_rtc_client_logger.warn('sdk webrtc registered but transport suspect, re-registering', {
+            clientId: this.clientId,
+            connected: this.isConnected(),
+            lastTransportMessageAt: this.lastTransportMessageAt
+        });
+        this._cleanupRegister();
+        if (this.isConnected()) {
+            this.connectionPromise = this._disconnectTransport(true);
+            await this.connectionPromise;
+            this.connectionPromise = null;
+            const transport = this.userAgent?.transport;
+            if (transport && transport.state !== __WEBPACK_EXTERNAL_MODULE_sip_js_lib_api_transport_state_2606bc5d__.TransportState.Disconnected) transport.transitionState(__WEBPACK_EXTERNAL_MODULE_sip_js_lib_api_transport_state_2606bc5d__.TransportState.Disconnected);
+        }
+        return false;
+    }
     async register(tries = 0) {
         const logInfo = {
             clientId: this.clientId,
@@ -5361,6 +5439,11 @@ class WebRTCClient extends Emitter {
             skipRegister: this.skipRegister
         };
         this.forceClosed = false;
+        this._closed = false;
+        if (!this._countedAsLive) {
+            this._countedAsLive = true;
+            liveClientCount += 1;
+        }
         if (this.skipRegister) {
             web_rtc_client_logger.info('sdk webrtc skip register...', logInfo);
             return Promise.resolve();
@@ -5370,15 +5453,19 @@ class WebRTCClient extends Emitter {
             web_rtc_client_logger.info('sdk webrtc recreating User Agent');
             this.userAgent = this.createUserAgent(this.uaConfigOverrides);
         }
-        if (!this.userAgent || this.isRegistered()) {
-            web_rtc_client_logger.info('sdk webrtc registering aborted, already registered or no UA can be created');
-            return Promise.resolve();
+        if (!this.userAgent) {
+            web_rtc_client_logger.info('sdk webrtc registering aborted, no UA can be created');
+            return;
+        }
+        if (this.isRegistered() && await this._isRegistrationHealthy()) {
+            web_rtc_client_logger.info('sdk webrtc registering aborted, already registered');
+            return;
         }
         if (this.connectionPromise || this.registerer?.waiting) {
             web_rtc_client_logger.info('sdk webrtc registering aborted due to a registration in progress.', {
                 clientId: this.clientId
             });
-            return Promise.resolve();
+            return;
         }
         const registerOptions = this._isWeb() ? {} : {
             extraContactHeaderParams: [
@@ -5479,7 +5566,6 @@ class WebRTCClient extends Emitter {
                         web_rtc_client_logger.info('sdk webrtc unregistered', {
                             clientId: this.clientId
                         });
-                        if (this.registerer) this.registerer.stateChange.addListener(onRegisterStateChange);
                         this._cleanupRegister();
                         resolve();
                     }
@@ -5673,6 +5759,11 @@ class WebRTCClient extends Emitter {
             force
         });
         this.forceClosed = force;
+        this._closed = true;
+        if (this._countedAsLive) {
+            this._countedAsLive = false;
+            liveClientCount = Math.max(0, liveClientCount - 1);
+        }
         this._cleanupMedia();
         this.connectionPromise = null;
         Object.values(this.audioElements).forEach((audioElement)=>{
@@ -8561,6 +8652,15 @@ class WebRTCPhone extends Emitter {
     isRegistered() {
         return this.client && this.client.isRegistered();
     }
+    isConnected() {
+        return Boolean(this.client?.isConnected());
+    }
+    isTransportSuspect() {
+        return !this.client || this.client.isTransportSuspect();
+    }
+    getLastTransportMessageAt() {
+        return this.client?.getLastTransportMessageAt() ?? null;
+    }
     enableRinging() {
         WebRTCPhone_logger.info('WebRTC enable ringing');
         this.ringingEnabled = true;
@@ -8880,8 +8980,9 @@ class WebRTCPhone extends Emitter {
         const { state } = sipSession || {};
         const sessionId = this.getSipSessionId(sipSession);
         fromSession = fromSession || this.callSessions[sessionId];
+        const headerCallId = sipSession?.request?.getHeader?.('X-Wazo-Call-ID');
         const callSession = new CallSession({
-            callId: fromSession && fromSession.callId,
+            callId: fromSession?.callId || headerCallId || '',
             sipCallId: sessionId,
             sipStatus: state,
             displayName: identity ? identity.displayName || number : number,
@@ -8902,6 +9003,7 @@ class WebRTCPhone extends Emitter {
             recordingState: fromSession ? fromSession.recordingState : RECORDING_STATE.INACTIVE,
             sipSession,
             diversion: sipSession?.diversion,
+            assertedIdentity: toAssertedIdentity(sipSession?.assertedIdentity) || void 0,
             conference: !!fromSession && fromSession.isConference(),
             ...extra
         });
@@ -9515,6 +9617,7 @@ const protocolDebugMessages = [
 class Phone_Phone extends Emitter {
     client;
     phone;
+    connectInFlight;
     session;
     sipLine;
     SessionState;
@@ -9584,10 +9687,37 @@ class Phone_Phone extends Emitter {
         this.sipLine = sipLine || this.getPrimaryWebRtcLine();
         if (!this.sipLine) throw new Error('Sorry, no sip lines found for this user');
         options.userUuid = session.uuid || '';
-        this.connectWithCredentials(server, this.sipLine, session.displayName(), options);
+        return this.connectWithCredentials(server, this.sipLine, session.displayName(), options);
     }
-    connectWithCredentials(server, sipLine, displayName, rawOptions = {}) {
+    async connectWithCredentials(server, sipLine, displayName, rawOptions = {}) {
         if (this.phone) return;
+        if (this.connectInFlight) {
+            Phone_logger.info('connectWithCredentials: connection already in progress, reusing it and ignoring this call\'s arguments', {
+                server,
+                sipLineId: sipLine?.id
+            });
+            return this.connectInFlight;
+        }
+        this.connectInFlight = this._connectWithCredentials(server, sipLine, displayName, rawOptions).finally(()=>{
+            this.connectInFlight = null;
+        });
+        return this.connectInFlight;
+    }
+    async _connectWithCredentials(server, sipLine, displayName, rawOptions = {}) {
+        if (this.client) {
+            Phone_logger.warn('connectWithCredentials: closing previous client before creating a new one');
+            try {
+                await this.client.close();
+            } catch (error) {
+                Phone_logger.error('connectWithCredentials: error while closing previous client', {
+                    message: error?.message
+                });
+            }
+            if (this.phone) {
+                Phone_logger.info('connectWithCredentials: phone connected while closing previous client, keeping it');
+                return;
+            }
+        }
         const [host, port = 443] = server.split(':');
         const options = rawOptions;
         options.media = options.media || {
@@ -9624,6 +9754,24 @@ class Phone_Phone extends Emitter {
         }
         this.phone = null;
     }
+    async reset() {
+        Phone_logger.info('phone reset', {
+            phone: !!this.phone,
+            client: !!this.client
+        });
+        await this.disconnect().catch((error)=>{
+            Phone_logger.error('reset: error while disconnecting phone', {
+                message: error?.message
+            });
+        });
+        if (this.client) await this.client.close().catch((error)=>{
+            Phone_logger.error('reset: error while closing client', {
+                message: error?.message
+            });
+        });
+        this.phone = null;
+        this.client = null;
+    }
     async call(extension, withCamera = false, rawSipLine = null, audioOnly = false, conference = false) {
         if (!this.phone) return;
         const sipLine = rawSipLine || this.getPrimaryWebRtcLine();

package/dist/types/__tests__/web-rtc-client-multi-client.test.d.ts

@@ -0,0 +1,2 @@
+export {};
+//# sourceMappingURL=web-rtc-client-multi-client.test.d.ts.map

package/dist/types/__tests__/web-rtc-client-multi-client.test.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"web-rtc-client-multi-client.test.d.ts","sourceRoot":"","sources":["../../../src/__tests__/web-rtc-client-multi-client.test.ts"],"names":[],"mappings":""}

package/dist/types/domain/CallSession.d.ts

@@ -1,9 +1,10 @@
 import Call, { type RecordingStateType } from './Call';
-import { WazoSession } from './types';
+import { WazoSession, AssertedIdentity } from './types';
 type CallSessionArguments = {
     answered: boolean;
     answerTime?: Date | null | undefined;
     answeredBySystem: boolean;
+    assertedIdentity?: AssertedIdentity;
     call: Call | null | undefined;
     callId: string;
     callerNumber: string;
@@ -49,6 +50,7 @@ export default class CallSession {
     answeredBySystem: boolean;
     dialedExtension: string;
     diversion?: string[];
+    assertedIdentity?: AssertedIdentity;
     ringing: boolean;
     paused: boolean;
     sipCallId: string;
@@ -66,7 +68,7 @@ export default class CallSession {
     recordingState: RecordingStateType;
     sipSession: WazoSession | undefined;
     conference: boolean;
-    constructor({ answered, answeredBySystem, isCaller, displayName, callId, muted, videoMuted, number, paused, ringing, startTime, creationTime, endTime, cameraEnabled, dialedExtension, diversion, sipCallId, sipStatus, callerNumber, call, autoAnswer, ignored, screensharing, recording, recordingPaused, recordingState, videoRemotelyDowngraded, sipSession, answerTime, conference, }: CallSessionArguments);
+    constructor({ answered, answeredBySystem, isCaller, displayName, callId, muted, videoMuted, number, paused, ringing, startTime, creationTime, endTime, cameraEnabled, dialedExtension, diversion, assertedIdentity, sipCallId, sipStatus, callerNumber, call, autoAnswer, ignored, screensharing, recording, recordingPaused, recordingState, videoRemotelyDowngraded, sipSession, answerTime, conference, }: CallSessionArguments);
     resume(): void;
     hold(): void;
     mute(): void;

package/dist/types/domain/CallSession.d.ts.map

@@ -1 +1 @@
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