@wazo/sdk 0.39.15 → 0.40.0

package/esm/web-rtc-client.js

@@ -189,7 +189,6 @@ export default class WebRTCClient extends Emitter {
                     method: 'delegate.onInvite',
                     clientId: this.clientId,
                     id: invitation.id,
-                    // @ts-ignore
                     remoteURI: invitation.remoteURI,
                 });
                 this._setupSession(invitation);
@@ -207,6 +206,7 @@ export default class WebRTCClient extends Emitter {
         ua.transport.onMessage = (rawMessage) => {
             const message = Parser.parseMessage(rawMessage, ua.transport.logger);
             // We have to re-sent the message to the UA ...
+            // @ts-ignore: private
             ua.onTransportMessage(rawMessage);
             // And now do what we want with the message
             this.eventEmitter.emit(MESSAGE, message);
@@ -231,9 +231,9 @@ export default class WebRTCClient extends Emitter {
     onConnect() {
         logger.info('sdk webrtc connected', { method: 'delegate.onConnect', clientId: this.clientId });
         this.eventEmitter.emit(CONNECTED);
-        // @ts-ignore
+        // @ts-ignore: private
         if (!this.isRegistered() && this.registerer?.waiting) {
-            // @ts-ignore
+            // @ts-ignore: private
             this.registerer.waitingToggle(false);
         }
         return this.register();
@@ -245,9 +245,9 @@ export default class WebRTCClient extends Emitter {
         this.eventEmitter.emit(DISCONNECTED, error);
         if (this.isRegistered()) {
             await this.unregister();
-            // @ts-ignore
+            // @ts-ignore: private
             if (this.registerer?.waiting) {
-                // @ts-ignore
+                // @ts-ignore: private
                 this.registerer.waitingToggle(false);
             }
             this.eventEmitter.emit(UNREGISTERED);
@@ -260,7 +260,7 @@ export default class WebRTCClient extends Emitter {
             registered: this.isRegistered(),
             connectionPromise: !!this.connectionPromise,
             registerer: !!this.registerer,
-            // @ts-ignore
+            // @ts-ignore: private
             waiting: this.registerer && this.registerer.waiting,
             tries,
             skipRegister: this.skipRegister,
@@ -279,7 +279,7 @@ export default class WebRTCClient extends Emitter {
             logger.info('sdk webrtc registering aborted, already registered or no UA can be created');
             return Promise.resolve();
         }
-        // @ts-ignore
+        // @ts-ignore: private
         if (this.connectionPromise || this.registerer?.waiting) {
             logger.info('sdk webrtc registering aborted due to a registration in progress.', { clientId: this.clientId });
             return Promise.resolve();
@@ -298,9 +298,9 @@ export default class WebRTCClient extends Emitter {
                 return;
             }
             this.connectionPromise = null;
-            // @ts-ignore
+            // @ts-ignore: private
             if (this.registerer && this.registerer.waiting) {
-                // @ts-ignore
+                // @ts-ignore: private
                 this.registerer.waitingToggle(false);
             }
             if (tries <= MAX_REGISTER_TRIES) {
@@ -352,19 +352,19 @@ export default class WebRTCClient extends Emitter {
         if (!registerer) {
             return;
         }
-        // @ts-ignore
+        // @ts-ignore: private
         const oldWaitingToggle = registerer.waitingToggle.bind(registerer);
-        // @ts-ignore
+        // @ts-ignore: private
         const oldUnregistered = registerer.unregistered.bind(registerer);
-        // @ts-ignore
+        // @ts-ignore: private
         registerer.waitingToggle = (waiting) => {
-            // @ts-ignore
+            // @ts-ignore: private
             if (!registerer || registerer.waiting === waiting) {
                 return;
             }
             oldWaitingToggle(waiting);
         };
-        // @ts-ignore
+        // @ts-ignore: private
         registerer.unregistered = () => {
             if (!registerer || registerer.state === RegistererState.Terminated) {
                 return;
@@ -441,7 +441,7 @@ export default class WebRTCClient extends Emitter {
             inviterOptions.sessionDescriptionHandlerModifiersReInvite = [stripVideo];
         }
         const uri = this._makeURI(number);
-        let session;
+        let session = null;
         if (uri) {
             session = this.userAgent ? new Inviter(this.userAgent, uri, inviterOptions) : null;
         }
@@ -458,23 +458,21 @@ export default class WebRTCClient extends Emitter {
         const inviteOptions = {
             requestDelegate: {
                 onAccept: (response) => {
-                    if (session.sessionDescriptionHandler?.peerConnection) {
-                        session.sessionDescriptionHandler.peerConnection.sfu = conference;
+                    if (session?.sessionDescriptionHandler?.peerConnection) {
+                        (session?.sessionDescriptionHandler).peerConnection.sfu = conference;
                     }
-                    // @ts-ignore
                     this._onAccepted(session, response.session, true);
                 },
-                onProgress: payload => {
+                onProgress: (payload) => {
                     if (payload.message.statusCode === 183) {
-                        // @ts-ignore
                         this._onEarlyProgress(payload.session);
                     }
                 },
                 onReject: (response) => {
                     logger.info('on call rejected', {
-                        id: session.id,
-                        // @ts-ignore
-                        fromTag: session.fromTag,
+                        id: session?.id,
+                        // @ts-ignore: fromTag does not exist
+                        fromTag: session?.fromTag,
                     });
                     this._stopSendingStats(session);
                     this.stopNetworkMonitoring(session);
@@ -484,17 +482,19 @@ export default class WebRTCClient extends Emitter {
             sessionDescriptionHandlerOptions: this.getMediaConfiguration(enableVideo || false, conference),
         };
         if (inviteOptions.sessionDescriptionHandlerOptions) {
-            // @ts-ignore
             inviteOptions.sessionDescriptionHandlerOptions.audioOnly = audioOnly;
         }
         inviteOptions.sessionDescriptionHandlerModifiers = [replaceLocalIpModifier];
         if (audioOnly) {
             inviteOptions.sessionDescriptionHandlerModifiers.push(stripVideo);
         }
-        // Do not await invite here or we'll miss the Establishing state transition
-        session.invitePromise = session.invite(inviteOptions).catch((e) => {
-            logger.warn('sdk webrtc creating call, error', e);
-        });
+        if (session) {
+            // Do not await invite here or we'll miss the Establishing state transition
+            // @ts-ignore: Property 'invitePromise' does not exist
+            session.invitePromise = session.invite(inviteOptions).catch((e) => {
+                logger.warn('sdk webrtc creating call, error', e);
+            });
+        }
         return session;
     }
     answer(session, enableVideo) {
@@ -512,7 +512,7 @@ export default class WebRTCClient extends Emitter {
             sessionDescriptionHandlerOptions: this.getMediaConfiguration(enableVideo || false),
         };
         return this._accept(session, options).then(() => {
-            // @ts-ignore
+            // @ts-ignore: private
             if (session.isCanceled) {
                 const message = 'accepted a canceled session (or was canceled during the accept phase).';
                 logger.error(message, {
@@ -555,8 +555,7 @@ export default class WebRTCClient extends Emitter {
         return Promise.resolve(null);
     }
     async getStats(session) {
-        // @ts-ignore
-        const pc = session.sessionDescriptionHandler.peerConnection;
+        const pc = session.sessionDescriptionHandler?.peerConnection;
         if (!pc) {
             return null;
         }
@@ -625,13 +624,12 @@ export default class WebRTCClient extends Emitter {
         if (this.userAgent) {
             this.userAgent.delegate = undefined;
         }
-        // @ts-ignore: this is odd, may want to validate
+        // @ts-ignore: removeAllListeners does not exist
         this.userAgent.stateChange.removeAllListeners();
         await this._disconnectTransport(force);
         this._cleanupRegister();
         try {
             // Prevent `Connect aborted.` error when disconnecting
-            // @ts-ignore
             this.userAgent.transport.connectReject = () => { };
             // Don't wait here, It can take ~30s to stop ...
             this.userAgent.stop().catch(console.error);
@@ -646,7 +644,7 @@ export default class WebRTCClient extends Emitter {
         if (!session) {
             return null;
         }
-        // @ts-ignore
+        // @ts-ignore: private
         return session.remoteIdentity.uri._normal.user;
     }
     mute(session) {
@@ -666,7 +664,6 @@ export default class WebRTCClient extends Emitter {
             return false;
         }
         let muted = true;
-        // @ts-ignore
         const pc = session.sessionDescriptionHandler.peerConnection;
         if (!pc) {
             return false;
@@ -711,7 +708,7 @@ export default class WebRTCClient extends Emitter {
         logger.info('sdk webrtc hold', {
             sessionId,
             keys: Object.keys(this.heldSessions),
-            // @ts-ignore
+            // @ts-ignore: private
             pendingReinvite: !!session.pendingReinvite,
             isConference,
             hasVideo,
@@ -719,7 +716,7 @@ export default class WebRTCClient extends Emitter {
         if (sessionId in this.heldSessions) {
             return Promise.resolve();
         }
-        // @ts-ignore
+        // @ts-ignore: private
         if (session.pendingReinvite) {
             return Promise.resolve();
         }
@@ -731,7 +728,6 @@ export default class WebRTCClient extends Emitter {
         // `sendonly` direction
         this.mute(session);
         session.sessionDescriptionHandlerOptionsReInvite = {
-            // @ts-ignore
             hold: true,
             conference: isConference,
         };
@@ -746,7 +742,7 @@ export default class WebRTCClient extends Emitter {
         };
         // Avoid sdh to create a new stream
         if (session.sessionDescriptionHandler) {
-            // @ts-ignore
+            // @ts-ignore: private
             session.sessionDescriptionHandler.localMediaStreamConstraints = options.constraints;
         }
         // Send re-INVITE
@@ -760,19 +756,18 @@ export default class WebRTCClient extends Emitter {
         logger.info('sdk webrtc unhold', {
             sessionId,
             keys: Object.keys(this.heldSessions),
-            // @ts-ignore
+            // @ts-ignore: private
             pendingReinvite: !!session.pendingReinvite,
             isConference,
             hasVideo,
         });
-        // @ts-ignore
+        // @ts-ignore: private
         if (session.pendingReinvite) {
             return Promise.resolve();
         }
         this.unmute(session);
         delete this.heldSessions[this.getSipSessionId(session)];
         session.sessionDescriptionHandlerOptionsReInvite = {
-            // @ts-ignore
             hold: false,
             conference: isConference,
         };
@@ -794,13 +789,12 @@ export default class WebRTCClient extends Emitter {
     }
     // Returns true if a re-INVITE is required
     async upgradeToVideo(session, constraints, isConference) {
-        // @ts-ignore
-        const pc = session.sessionDescriptionHandler.peerConnection;
+        const pc = session.sessionDescriptionHandler?.peerConnection;
         // Check if a video sender already exists
         let videoSender;
         if (isConference) {
             // We search for the last transceiver without `video-` in the mid (video- means remote transceiver)
-            const transceivers = pc.getTransceivers();
+            const transceivers = pc?.getTransceivers() || [];
             const transceiverIdx = lastIndexOf(transceivers, transceiver => transceiver.sender.track === null && transceiver.mid && transceiver.mid.indexOf('video') === -1);
             videoSender = transceiverIdx !== -1 ? transceivers[transceiverIdx].sender : null;
         }
@@ -819,8 +813,7 @@ export default class WebRTCClient extends Emitter {
         }
         // Add previous local audio track
         if (constraints && !constraints.audio) {
-            // @ts-ignore
-            const localVideoStream = session.sessionDescriptionHandler.localMediaStream;
+            const localVideoStream = session.sessionDescriptionHandler?.localMediaStream;
             const localAudioTrack = localVideoStream.getTracks().find(track => track.kind === 'audio');
             if (localAudioTrack) {
                 newStream.addTrack(localAudioTrack);
@@ -835,13 +828,12 @@ export default class WebRTCClient extends Emitter {
     }
     downgradeToAudio(session) {
         // Release local video stream when downgrading to audio
-        // @ts-ignore
-        const localStream = session.sessionDescriptionHandler.localMediaStream;
-        // @ts-ignore
-        const pc = session.sessionDescriptionHandler.peerConnection;
+        const sessionDescriptionHandler = session.sessionDescriptionHandler;
+        const localStream = sessionDescriptionHandler?.localMediaStream;
+        const pc = sessionDescriptionHandler?.peerConnection;
         const videoTracks = localStream.getVideoTracks();
         // Remove video senders
-        if (pc.getSenders) {
+        if (pc?.getSenders) {
             pc.getSenders().filter((sender) => sender.track && sender.track.kind === 'video').forEach((videoSender) => {
                 videoSender.replaceTrack(null);
             });
@@ -855,7 +847,7 @@ export default class WebRTCClient extends Emitter {
     async getStreamFromConstraints(constraints, conference = false) {
         const video = constraints && constraints.video;
         const { constraints: newConstraints, } = this.getMediaConfiguration(video, conference, constraints);
-        let newStream;
+        let newStream = null;
         try {
             newStream = await wazoMediaStreamFactory(newConstraints);
         }
@@ -864,7 +856,6 @@ export default class WebRTCClient extends Emitter {
         if (!newStream) {
             return null;
         }
-        // @ts-ignore
         newStream.local = true;
         return newStream;
     }
@@ -995,7 +986,7 @@ export default class WebRTCClient extends Emitter {
         logger.warn('pingServer: null fromURI');
     }
     getState() {
-        // @ts-ignore
+        // @ts-ignore: something fishy here: `states` content doesn't match UserAgentState at all
         return this.userAgent ? states[this.userAgent.state] : UserAgentState.Stopped;
     }
     getContactIdentifier() {
@@ -1026,9 +1017,7 @@ export default class WebRTCClient extends Emitter {
             // audioElement is an array of HTMLAudioElements, and HTMLAudioElement inherits the method from HTMLMediaElement
             // https://developer.mozilla.org/en-US/docs/Web/API/HTMLAudioElement
             // https://developer.mozilla.org/en-US/docs/Web/API/HTMLMediaElement/setSinkId
-            // @ts-ignore
             if (audioElement.setSinkId) {
-                // @ts-ignore
                 audioElement.setSinkId(id);
             }
         });
@@ -1070,8 +1059,7 @@ export default class WebRTCClient extends Emitter {
         }
         if (session && navigator.mediaDevices) {
             const sdh = session.sessionDescriptionHandler;
-            // @ts-ignore
-            const pc = sdh.peerConnection;
+            const pc = sdh?.peerConnection;
             const constraints = {
                 audio: {
                     deviceId: {
@@ -1120,7 +1108,6 @@ export default class WebRTCClient extends Emitter {
             return Promise.resolve();
         }
         const sdh = session.sessionDescriptionHandler;
-        // @ts-ignore
         const pc = sdh.peerConnection;
         const sessionId = this.getSipSessionId(session);
         const localStream = this.getLocalStream(sessionId);
@@ -1164,18 +1151,16 @@ export default class WebRTCClient extends Emitter {
         });
     }
     getAudioDeviceId() {
-        // @ts-ignore
-        return this.audio && typeof this.audio === 'object' && 'deviceId' in this.audio ? this.audio.deviceId.exact : undefined;
+        return this.audio && typeof this.audio === 'object' && 'deviceId' in this.audio ? this.audio.deviceId?.exact : undefined;
     }
     getVideoDeviceId() {
-        // @ts-ignore
-        return this.video && typeof this.video === 'object' && 'deviceId' in this.video ? this.video.deviceId.exact : undefined;
+        return this.video && typeof this.video === 'object' && 'deviceId' in this.video ? this.video.deviceId?.exact : undefined;
     }
     reinvite(sipSession, newConstraints = null, conference = false, audioOnly = false, iceRestart = false) {
         if (!sipSession) {
             return Promise.resolve();
         }
-        // @ts-ignore
+        // @ts-ignore: private
         if (sipSession.pendingReinvite) {
             return Promise.resolve();
         }
@@ -1200,7 +1185,6 @@ export default class WebRTCClient extends Emitter {
             sipSession.sessionDescriptionHandlerModifiersReInvite = modifiers.filter(modifier => modifier !== stripVideo);
         }
         sipSession.sessionDescriptionHandlerOptionsReInvite = { ...sipSession.sessionDescriptionHandlerOptionsReInvite,
-            // @ts-ignore
             conference,
             audioOnly,
         };
@@ -1210,13 +1194,13 @@ export default class WebRTCClient extends Emitter {
                 onAccept: (response) => {
                     // Update the SDP body to be able to call sessionWantsToDoVideo correctly in `_setup[Local|Remote]Media`.
                     // Can't set directly sipSession.body because it's a getter.
-                    // @ts-ignore
+                    // @ts-ignore: private
                     if (sipSession instanceof Inviter && sipSession.outgoingRequestMessage.body) {
-                        // @ts-ignore
+                        // @ts-ignore: private
                         sipSession.outgoingRequestMessage.body.body = response.message.body;
                     }
                     else if (sipSession instanceof Invitation) {
-                        // @ts-ignore
+                        // @ts-ignore: private
                         sipSession.incomingInviteRequest.message.body = response.message.body;
                     }
                     logger.info('on re-INVITE accepted', {
@@ -1228,7 +1212,6 @@ export default class WebRTCClient extends Emitter {
                     if (wasMuted) {
                         this.mute(sipSession);
                     }
-                    // @ts-ignore
                     this._onAccepted(sipSession, response.session, false, false);
                     if (shouldDoScreenSharing) {
                         this.eventEmitter.emit(ON_SCREEN_SHARING_REINVITE, sipSession, response, desktop);
@@ -1242,7 +1225,6 @@ export default class WebRTCClient extends Emitter {
             },
             sessionDescriptionHandlerOptions: {
                 constraints,
-                // @ts-ignore
                 conference,
                 audioOnly,
                 offerOptions: {
@@ -1263,13 +1245,11 @@ export default class WebRTCClient extends Emitter {
         if (!sipSession) {
             return null;
         }
-        // @ts-ignore
         return sipSession.sessionDescriptionHandler ? sipSession.sessionDescriptionHandler.peerConnection : null;
     }
     // Local streams
     getLocalStream(sessionId) {
         const sipSession = this.sipSessions[sessionId];
-        // @ts-ignore
         return sipSession?.sessionDescriptionHandler?.localMediaStream || null;
     }
     getLocalTracks(sessionId) {
@@ -1292,7 +1272,6 @@ export default class WebRTCClient extends Emitter {
     // Remote streams
     getRemoteStream(sessionId) {
         const sipSession = this.sipSessions[sessionId];
-        // @ts-ignore
         return sipSession?.sessionDescriptionHandler?.remoteMediaStream || null;
     }
     getRemoteTracks(sessionId) {
@@ -1333,15 +1312,14 @@ export default class WebRTCClient extends Emitter {
         if (!sipSession) {
             return '';
         }
-        // @ts-ignore
-        if (sipSession.message?.callId) {
-            // @ts-ignore
-            return sipSession.message.callId.substring(0, SIP_ID_LENGTH);
+        const message = sipSession.message;
+        if (message?.callId) {
+            return message.callId.substring(0, SIP_ID_LENGTH);
         }
         // For Inviter
-        // @ts-ignore
+        // @ts-ignore: private
         if (sipSession instanceof Inviter && sipSession.outgoingRequestMessage) {
-            // @ts-ignore
+            // @ts-ignore: private
             return sipSession.outgoingRequestMessage.callId.substring(0, SIP_ID_LENGTH);
         }
         // For Invitation
@@ -1394,7 +1372,7 @@ export default class WebRTCClient extends Emitter {
         if (!this.userAgent) {
             return;
         }
-        // @ts-ignore
+        // @ts-ignore: private
         this.userAgent.attemptReconnection();
     }
     storeSipSession(session) {
@@ -1419,7 +1397,7 @@ export default class WebRTCClient extends Emitter {
         });
         if (sipSession.sessionDescriptionHandler) {
             // eslint-disable-next-line no-underscore-dangle
-            // @ts-ignore
+            // @ts-ignore: protected
             sipSession.sessionDescriptionHandler._localMediaStream = newStream;
         }
     }
@@ -1533,14 +1511,13 @@ export default class WebRTCClient extends Emitter {
             return;
         }
         if (this.userAgent?.transport) {
-            // @ts-ignore
             this.userAgent.transport.configuration.traceSip = true;
         }
-        // @ts-ignore
+        // @ts-ignore: private
         this.userAgent.loggerFactory.builtinEnabled = true;
-        // @ts-ignore
+        // @ts-ignore: private
         this.userAgent.loggerFactory.level = 3; // debug
-        // @ts-ignore
+        // @ts-ignore: private
         this.userAgent.loggerFactory.connector = logConnector;
     }
     _onHeartbeat(message) {
@@ -1567,8 +1544,9 @@ export default class WebRTCClient extends Emitter {
         }
         if (this.userAgent && this.userAgent.transport) {
             // Disconnect from WS and triggers events, but do not trigger disconnect if already disconnecting...
-            // @ts-ignore
-            if (!this.userAgent.transport.transitioningState && !this.userAgent.transport.disconnectPromise) {
+            const transport = this.userAgent.transport;
+            // @HEADS UP
+            if (!transport.transitioningState && !transport.disconnectPromise) {
                 try {
                     await this.userAgent.transport.disconnect();
                 }
@@ -1590,14 +1568,14 @@ export default class WebRTCClient extends Emitter {
         return this.hasAudio;
     }
     _getAudioConstraints() {
-        // @ts-ignore
+        // @ts-ignore: media constraints
         return this.audio?.deviceId?.exact ? this.audio : true;
     }
     _getVideoConstraints(video = false) {
         if (!video) {
             return false;
         }
-        // @ts-ignore
+        // @ts-ignore: media constraints
         return this.video?.deviceId?.exact ? this.video : true;
     }
     _connectIfNeeded() {
@@ -1612,7 +1590,6 @@ export default class WebRTCClient extends Emitter {
         }
         if (this.isConnecting()) {
             logger.info('webrtc sdk, already connecting...');
-            // @ts-ignore
             this.connectionPromise = this.userAgent.transport.connectPromise;
             return Promise.resolve(this.connectionPromise);
         }
@@ -1623,11 +1600,10 @@ export default class WebRTCClient extends Emitter {
         logger.info('WebRTC UA needs to connect');
         // Force UA to reconnect
         if (this.userAgent && this.userAgent.state !== UserAgentState.Stopped) {
-            // @ts-ignore
+            // @ts-ignore: private
             this.userAgent.transitionState(UserAgentState.Stopped);
         }
         if (this.userAgent.transport && this.userAgent.transport.state !== TransportState.Disconnected) {
-            // @ts-ignore
             this.userAgent.transport.transitionState(TransportState.Disconnected);
         }
         this.connectionPromise = this.userAgent.start().catch(console.error);
@@ -1641,11 +1617,8 @@ export default class WebRTCClient extends Emitter {
         const client = new ApiClient({
             server: `${config.host}:${String(config.port || 443)}`,
         });
-        // @ts-ignore
         client.setToken(session.token);
-        // @ts-ignore
         client.setRefreshToken(session.refreshToken);
-        // @ts-ignore
         return client.confd.getUserLineSipFromToken(session.uuid).then(sipLine => ({
             authorizationUser: sipLine.username,
             password: sipLine.secret,
@@ -1681,7 +1654,6 @@ export default class WebRTCClient extends Emitter {
             sessionDescriptionHandlerFactory: (session, options = {}) => {
                 const uaLogger = session.userAgent.getLogger('sip.WazoSessionDescriptionHandler');
                 const isWeb = this._isWeb();
-                // @ts-ignore
                 const iceGatheringTimeout = 'peerConnectionOptions' in options ? options.peerConnectionOptions.iceGatheringTimeout || DEFAULT_ICE_TIMEOUT : DEFAULT_ICE_TIMEOUT;
                 const sdhOptions = { ...options,
                     iceGatheringTimeout,
@@ -1743,8 +1715,7 @@ export default class WebRTCClient extends Emitter {
             session.delegate = {};
         }
         session.delegate.onSessionDescriptionHandler = (sdh) => {
-            // @ts-ignore
-            sdh.on('error', e => {
+            sdh.on('error', (e) => {
                 this.eventEmitter.emit(ON_ERROR, e);
             });
             sdh.peerConnectionDelegate = {
@@ -1759,12 +1730,12 @@ export default class WebRTCClient extends Emitter {
                 },
             };
         };
-        // @ts-ignore
+        // @ts-ignore: protected
         const oldInviteRequest = session.onInviteRequest.bind(session);
         let hadRemoteVideo = false;
         // Monkey patch `onInviteRequest` to be able to know if there was a remote video stream before `onInvite` is called
         // Because when `onInvite` is called we already got the video track
-        // @ts-ignore
+        // @ts-ignore: protected
         session.onInviteRequest = request => {
             hadRemoteVideo = this.hasARemoteVideoTrack(sipSessionId);
             oldInviteRequest(request);
@@ -1773,7 +1744,7 @@ export default class WebRTCClient extends Emitter {
             let updatedCalleeName = null;
             let updatedNumber = null;
             if (session.assertedIdentity) {
-                // @ts-ignore
+                // @ts-ignore: private
                 updatedNumber = session.assertedIdentity.uri.normal.user;
                 updatedCalleeName = session.assertedIdentity.displayName || updatedNumber;
             }
@@ -1786,12 +1757,12 @@ export default class WebRTCClient extends Emitter {
             // Update SDP
             // Remote video is handled by the `track` event. Here we're dealing with video stream removal.
             if (session instanceof Invitation) {
-                // @ts-ignore
+                // @ts-ignore: private
                 session.incomingInviteRequest.message.body = request.body;
-                // @ts-ignore
+                // @ts-ignore: private
             }
             else if (session instanceof Inviter && session.outgoingInviteRequest.message.body) {
-                // @ts-ignore
+                // @ts-ignore: private
                 session.outgoingInviteRequest.message.body.body = request.body;
             }
             this.updateRemoteStream(sipSessionId, false);
@@ -1812,13 +1783,11 @@ export default class WebRTCClient extends Emitter {
         logger.info('on call accepted', {
             id: session.id,
             clientId: this.clientId,
-            // @ts-ignore
             remoteTag: session.remoteTag,
         });
         this.storeSipSession(session);
         this._setupMedias(session);
         this.updateRemoteStream(this.getSipSessionId(session), initAllTracks);
-        // @ts-ignore
         const pc = session.sessionDescriptionHandler?.peerConnection;
         const onTrack = (event) => {
             const isAudioOnly = this._isAudioOnly(session);
@@ -1837,13 +1806,11 @@ export default class WebRTCClient extends Emitter {
             }
             this.eventEmitter.emit(ON_TRACK, session, event);
         };
-        // @ts-ignore
-        if (session.sessionDescriptionHandler.peerConnection) {
-            // @ts-ignore
-            session.sessionDescriptionHandler.peerConnection.addEventListener('track', onTrack);
+        const sessionDescriptionHandler = session.sessionDescriptionHandler;
+        if (sessionDescriptionHandler?.peerConnection) {
+            sessionDescriptionHandler.peerConnection.addEventListener('track', onTrack);
         }
-        // @ts-ignore
-        session.sessionDescriptionHandler.remoteMediaStream.onaddtrack = onTrack;
+        sessionDescriptionHandler.remoteMediaStream.onaddtrack = onTrack;
         if (pc) {
             pc.oniceconnectionstatechange = () => {
                 logger.info('on ice connection state changed', {
@@ -1880,11 +1847,10 @@ export default class WebRTCClient extends Emitter {
             this.createAudioElementFor(sessionId);
         }
         const audioElement = this.audioElements[sessionId];
-        // @ts-ignore
         const sipSession = this.sipSessions[session.callId];
         const removeStream = this.getRemoteStream(sessionId);
-        // @ts-ignore
-        const earlyStream = sipSession && sipSession.sessionDescriptionHandler ? sipSession.sessionDescriptionHandler.remoteMediaStream : null;
+        const sdh = sipSession.sessionDescriptionHandler;
+        const earlyStream = sdh ? sdh.remoteMediaStream : null;
         const stream = newStream || removeStream || earlyStream;
         if (!stream) {
             logger.info('Setup media no stream to attach, bailing');
@@ -1944,8 +1910,8 @@ export default class WebRTCClient extends Emitter {
         stream.getTracks().filter(track => track.enabled).forEach(track => track.stop());
     }
     _toggleAudio(session, muteAudio) {
-        // @ts-ignore
-        const pc = session.sessionDescriptionHandler ? session.sessionDescriptionHandler.peerConnection : null;
+        const sdh = session.sessionDescriptionHandler;
+        const pc = (sdh?.peerConnection || null);
         if (!pc) {
             return;
         }
@@ -1967,9 +1933,9 @@ export default class WebRTCClient extends Emitter {
         }
     }
     _toggleVideo(session, muteCamera) {
-        // @ts-ignore
-        const pc = session.sessionDescriptionHandler?.peerConnection;
-        if (pc.getSenders) {
+        const sdh = session.sessionDescriptionHandler;
+        const pc = sdh?.peerConnection;
+        if (pc?.getSenders) {
             pc.getSenders().forEach((sender) => {
                 if (sender && sender.track && sender.track.kind === 'video') {
                     // eslint-disable-next-line
@@ -2007,14 +1973,14 @@ export default class WebRTCClient extends Emitter {
     }
     _cleanupRegister() {
         if (this.registerer) {
-            // @ts-ignore
+            // @ts-ignore: removeAllListeners does not exist
             this.registerer.stateChange.removeAllListeners();
             this.registerer = null;
         }
     }
     _startSendingStats(session) {
-        // @ts-ignore
-        const pc = session.sessionDescriptionHandler.peerConnection;
+        const sdh = session.sessionDescriptionHandler;
+        const pc = sdh.peerConnection;
         if (!pc) {
             return;
         }
@@ -2046,22 +2012,19 @@ export default class WebRTCClient extends Emitter {
         return UserAgent.makeURI(`sip:${target}@${this.config.host}`);
     }
     async _disconnectTransport(force = false) {
-        if (force && this.userAgent && this.userAgent.transport) {
+        const transport = this.userAgent?.transport;
+        if (force && transport) {
             // Bypass sip.js state machine that prevent to close WS with the state `Connecting`
-            // @ts-ignore
-            this.userAgent.transport.disconnectResolve = () => { };
-            // @ts-ignore
-            if (this.userAgent.transport._ws) {
-                // @ts-ignore
-                this.userAgent.transport._ws.close(1000);
+            transport.disconnectResolve = () => { };
+            if (transport._ws) {
+                transport._ws.close(1000);
             }
             return;
         }
         // Check if `disconnectPromise` is not present to avoid `Disconnect promise must not be defined` errors.
-        // @ts-ignore
-        if (this.userAgent && this.userAgent.transport && !this.userAgent.transport.disconnectPromise) {
+        if (transport && !transport.disconnectPromise) {
             try {
-                await this.userAgent.transport.disconnect();
+                await transport.disconnect();
             }
             catch (e) {
                 logger.error('WebRTC transport disconnect, error', e);
@@ -2175,7 +2138,7 @@ export default class WebRTCClient extends Emitter {
             logger.warn(error, { state: session.state, sessionId: session.id });
             throw new Error(error);
         }
-        // @ts-ignore
+        // @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 });
@@ -2193,7 +2156,7 @@ export default class WebRTCClient extends Emitter {
             logger.warn('Trying to reject a session in a wrong state', { state: session.state, sessionId: session.id });
             return;
         }
-        // @ts-ignore
+        // @ts-ignore: private
         if (!session.incomingInviteRequest.rejectable) {
             logger.warn('Trying to reject a non `rejectable` session', { state: session.state, sessionId: session.id });
             return;