@whereby.com/media 2.5.2 → 2.5.3

package/dist/index.cjs

@@ -1568,6 +1568,7 @@ class ServerSocket {
         this._wasConnectedUsingWebsocket = false;
         this._reconnectManager = null;
         this._socket = socket_ioClient.io(hostName, Object.assign({ path: DEFAULT_SOCKET_PATH, randomizationFactor: 0.5, reconnectionDelay: 250, reconnectionDelayMax: 5000, timeout: 5000, transports: ["websocket"], withCredentials: true }, optionsOverrides));
+        this.joinRoomFinished = false;
         this._socket.io.on("reconnect", () => {
             this._socket.sendBuffer = [];
         });
@@ -1585,6 +1586,12 @@ class ServerSocket {
         });
         if (glitchFree)
             this._reconnectManager = new ReconnectManager(this._socket);
+        this._socket.on("room_joined", (payload) => {
+            const { error } = payload;
+            if (!error) {
+                this.joinRoomFinished = true;
+            }
+        });
         this._socket.on("connect", () => {
             const transport = this.getTransport();
             if (transport === "websocket") {
@@ -1601,6 +1608,7 @@ class ServerSocket {
             }
         });
         this._socket.on("disconnect", () => {
+            this.joinRoomFinished = false;
             this.disconnectTimestamp = Date.now();
             if (this.noopKeepaliveInterval) {
                 clearInterval(this.noopKeepaliveInterval);
@@ -1622,26 +1630,12 @@ class ServerSocket {
     disconnect() {
         this._socket.disconnect();
     }
-    disconnectOnConnect() {
-        this._socket.once("connect", () => {
-            this._socket.disconnect();
-        });
-    }
     emit(eventName, ...args) {
         this._socket.emit.apply(this._socket, arguments);
     }
-    emitIfConnected(eventName, data) {
-        if (!this.isConnected()) {
-            return;
-        }
-        this.emit(eventName, data);
-    }
     getTransport() {
-        return (this._socket &&
-            this._socket.io &&
-            this._socket.io.engine &&
-            this._socket.io.engine.transport &&
-            this._socket.io.engine.transport.name);
+        var _a, _b, _c, _d;
+        return (_d = (_c = (_b = (_a = this._socket) === null || _a === void 0 ? void 0 : _a.io) === null || _b === void 0 ? void 0 : _b.engine) === null || _c === void 0 ? void 0 : _c.transport) === null || _d === void 0 ? void 0 : _d.name;
     }
     getManager() {
         return this._socket.io;
@@ -2603,6 +2597,8 @@ class P2pRtcManager {
         this._webrtcProvider = webrtcProvider;
         this._features = features || {};
         this._isAudioOnlyMode = false;
+        this._closed = false;
+        this.skipEmittingServerMessageCount = 0;
         this.offerOptions = { offerToReceiveAudio: true, offerToReceiveVideo: true };
         this._pendingActionsForConnectedPeerConnections = [];
         this._audioTrackOnEnded = () => {
@@ -2684,6 +2680,10 @@ class P2pRtcManager {
         }
         return this._replaceTrackToPeerConnections(oldTrack, newTrack);
     }
+    close() {
+        this._closed = true;
+        this.disconnectAll();
+    }
     disconnectAll() {
         Object.keys(this.peerConnections).forEach((peerConnectionId) => {
             this.disconnect(peerConnectionId);
@@ -2869,8 +2869,18 @@ class P2pRtcManager {
             logger$6.error("Error during setting jitter buffer target:", error);
         }
     }
-    _emitServerEvent(eventName, data, callback) {
-        this._serverSocket.emit(eventName, data, callback);
+    _emitServerEvent(eventName, data) {
+        if (this._closed) {
+            logger$6.warn("RtcManager closed. Will not send event", eventName, data);
+            return;
+        }
+        if (this._features.awaitJoinRoomFinished && !this._serverSocket.joinRoomFinished) {
+            rtcStats.sendEvent("skip_emitting_server_message", { eventName });
+            this.skipEmittingServerMessageCount++;
+        }
+        else {
+            this._serverSocket.emit(eventName, data);
+        }
     }
     _emit(eventName, data) {
         this._emitter.emit(eventName, data);
@@ -5506,6 +5516,9 @@ class VegaRtcManager {
             });
         }
     }
+    close() {
+        this.disconnectAll();
+    }
     disconnectAll() {
         var _a, _b, _c, _d;
         this._reconnect = false;

package/dist/index.d.cts

@@ -490,140 +490,6 @@ declare function getUpdatedDevices({ oldDevices, newDevices, currentAudioId, cur
     currentSpeakerId?: string | undefined;
 }): GetUpdatedDevicesResult;
 
-declare class P2pRtcManager implements RtcManager {
-    _selfId: any;
-    _roomName: any;
-    _roomSessionId: any;
-    peerConnections: any;
-    localStreams: any;
-    enabledLocalStreamIds: any[];
-    _screenshareVideoTrackIds: any[];
-    _socketListenerDeregisterFunctions: any[];
-    _localStreamDeregisterFunction: any;
-    _emitter: any;
-    _serverSocket: any;
-    _webrtcProvider: any;
-    _features: any;
-    _isAudioOnlyMode: boolean;
-    offerOptions: {
-        offerToReceiveAudio: boolean;
-        offerToReceiveVideo: boolean;
-    };
-    _pendingActionsForConnectedPeerConnections: any[];
-    _audioTrackOnEnded: () => void;
-    _videoTrackOnEnded: () => void;
-    totalSessionsCreated: number;
-    _iceServers: any;
-    _turnServers: any;
-    _sfuServer: any;
-    _mediaserverConfigTtlSeconds: any;
-    _fetchMediaServersTimer: any;
-    _wasScreenSharing: any;
-    ipv6HostCandidateTeredoSeen: any;
-    ipv6HostCandidate6to4Seen: any;
-    mdnsHostCandidateSeen: any;
-    _stoppedVideoTrack: any;
-    icePublicIPGatheringTimeoutID: any;
-    _videoTrackBeingMonitored?: CustomMediaStreamTrack;
-    _audioTrackBeingMonitored?: CustomMediaStreamTrack;
-    constructor({ selfId, room, emitter, serverSocket, webrtcProvider, features, }: {
-        selfId: any;
-        room: any;
-        emitter: any;
-        serverSocket: any;
-        webrtcProvider: any;
-        features: any;
-    });
-    numberOfPeerconnections(): number;
-    isInitializedWith({ selfId, roomName, isSfu }: {
-        selfId: any;
-        roomName: any;
-        isSfu: any;
-    }): boolean;
-    supportsScreenShareAudio(): boolean;
-    addNewStream(streamId: string, stream: MediaStream, audioPaused: boolean, videoPaused: boolean, beforeEffectTracks?: CustomMediaStreamTrack[]): void;
-    replaceTrack(oldTrack: CustomMediaStreamTrack | null, newTrack: CustomMediaStreamTrack): Promise<any[]>;
-    disconnectAll(): void;
-    fixChromeAudio(constraints: any): Promise<any[]> | undefined;
-    setupSocketListeners(): void;
-    sendAudioMutedStats(muted: boolean): void;
-    sendVideoMutedStats(muted: boolean): void;
-    sendStatsCustomEvent(eventName: string, data: any): void;
-    rtcStatsDisconnect(): void;
-    rtcStatsReconnect(): void;
-    setAudioOnly(audioOnly: any): void;
-    setRemoteScreenshareVideoTrackIds(remoteScreenshareVideoTrackIds?: never[]): void;
-    setRoomSessionId(roomSessionId: string): void;
-    _setConnectionStatus(session: any, newStatus: any, clientId: string): void;
-    _setJitterBufferTarget(pc: any): void;
-    _emitServerEvent(eventName: string, data?: any, callback?: any): void;
-    _emit(eventName: string, data?: any): void;
-    _addEnabledLocalStreamId(streamId: string): void;
-    _deleteEnabledLocalStreamId(streamId: string): void;
-    _getSession(peerConnectionId: string): any;
-    _getOrCreateSession(peerConnectionId: string, initialBandwidth: any): any;
-    _getLocalCameraStream(): any;
-    _getNonLocalCameraStreamIds(): string[];
-    _isScreensharingLocally(): boolean;
-    _getFirstLocalNonCameraStream(): any;
-    _transformIncomingSdp(original: any, _: any): {
-        type: any;
-        sdp: any;
-    };
-    _transformOutgoingSdp(original: any): {
-        type: any;
-        sdpU: any;
-    };
-    _createSession({ clientId, initialBandwidth, isOfferer, peerConnectionId, shouldAddLocalVideo, }: {
-        clientId: string;
-        initialBandwidth: any;
-        isOfferer: any;
-        peerConnectionId: string;
-        shouldAddLocalVideo: boolean;
-    }): any;
-    _cleanup(peerConnectionId: string): void;
-    _forEachPeerConnection(func: any): void;
-    _addStreamToPeerConnections(stream: any): void;
-    _addTrackToPeerConnections(track: any, stream?: any): void;
-    _replaceTrackToPeerConnections(oldTrack: any, newTrack: any): Promise<any[]>;
-    _removeStreamFromPeerConnections(stream: any): void;
-    _removeTrackFromPeerConnections(track: any): void;
-    _addLocalStream(streamId: string, stream: any): void;
-    _removeLocalStream(streamId: string): void;
-    _updateAndScheduleMediaServersRefresh({ iceServers, turnServers, sfuServer, mediaserverConfigTtlSeconds }: any): void;
-    _clearMediaServersRefresh(): void;
-    _monitorAudioTrack(track: any): void;
-    _monitorVideoTrack(track: CustomMediaStreamTrack): void;
-    _connect(clientId: string): Promise<any>;
-    _maybeRestartIce(clientId: string, session: any): void;
-    _setCodecPreferences(pc: RTCPeerConnection): Promise<void>;
-    _negotiatePeerConnection(clientId: string, session: any, constraints?: any): void;
-    _withForcedRenegotiation(session: any, action: any): void;
-    _changeBandwidthForAllClients(isJoining: boolean): number;
-    _createP2pSession({ clientId, initialBandwidth, shouldAddLocalVideo, isOfferer, }: {
-        clientId: string;
-        initialBandwidth: number;
-        shouldAddLocalVideo: boolean;
-        isOfferer: boolean;
-    }): any;
-    acceptNewStream({ streamId, clientId, shouldAddLocalVideo, }: {
-        streamId: string;
-        clientId: string;
-        shouldAddLocalVideo?: boolean;
-    }): any;
-    disconnect(clientId: string): void;
-    updateStreamResolution(): void;
-    stopOrResumeAudio(): void;
-    _handleStopOrResumeVideo({ enable, track }: {
-        enable: boolean;
-        track: any;
-    }): void;
-    stopOrResumeVideo(localStream: any, enable: boolean): void;
-    _shareScreen(streamId: string, stream: any): void;
-    removeStream(streamId: string, stream: any, requestedByClientId: any): void;
-    hasClient(clientId: string): boolean;
-}
-
 declare const assert: {
     fail: (message?: string | Error) => void;
     ok: (value: any, message?: string | Error) => void;
@@ -774,13 +640,12 @@ declare class ServerSocket {
     noopKeepaliveInterval: any;
     _wasConnectedUsingWebsocket?: boolean;
     disconnectTimestamp: number | undefined;
+    joinRoomFinished: boolean;
     constructor(hostName: string, optionsOverrides?: any, glitchFree?: boolean);
     setRtcManager(rtcManager?: RtcManager): void;
     connect(): void;
     disconnect(): void;
-    disconnectOnConnect(): void;
     emit(eventName: string, ...args: any[]): void;
-    emitIfConnected(eventName: string, data: any): void;
     getTransport(): any;
     getManager(): any;
     isConnecting(): any;
@@ -1206,6 +1071,143 @@ declare function fromLocation({ host, protocol }?: {
     subdomain: string;
 };
 
+declare class P2pRtcManager implements RtcManager {
+    _selfId: any;
+    _roomName: any;
+    _roomSessionId: any;
+    peerConnections: any;
+    localStreams: any;
+    enabledLocalStreamIds: any[];
+    _screenshareVideoTrackIds: any[];
+    _socketListenerDeregisterFunctions: any[];
+    _localStreamDeregisterFunction: any;
+    _emitter: any;
+    _serverSocket: ServerSocket;
+    _webrtcProvider: any;
+    _features: any;
+    _isAudioOnlyMode: boolean;
+    offerOptions: {
+        offerToReceiveAudio: boolean;
+        offerToReceiveVideo: boolean;
+    };
+    _pendingActionsForConnectedPeerConnections: any[];
+    _audioTrackOnEnded: () => void;
+    _videoTrackOnEnded: () => void;
+    totalSessionsCreated: number;
+    _iceServers: any;
+    _turnServers: any;
+    _sfuServer: any;
+    _mediaserverConfigTtlSeconds: any;
+    _fetchMediaServersTimer: any;
+    _wasScreenSharing: any;
+    ipv6HostCandidateTeredoSeen: any;
+    ipv6HostCandidate6to4Seen: any;
+    mdnsHostCandidateSeen: any;
+    _stoppedVideoTrack: any;
+    icePublicIPGatheringTimeoutID: any;
+    _videoTrackBeingMonitored?: CustomMediaStreamTrack;
+    _audioTrackBeingMonitored?: CustomMediaStreamTrack;
+    _closed: boolean;
+    skipEmittingServerMessageCount: number;
+    constructor({ selfId, room, emitter, serverSocket, webrtcProvider, features, }: {
+        selfId: any;
+        room: any;
+        emitter: any;
+        serverSocket: ServerSocket;
+        webrtcProvider: any;
+        features: any;
+    });
+    numberOfPeerconnections(): number;
+    isInitializedWith({ selfId, roomName, isSfu }: {
+        selfId: any;
+        roomName: any;
+        isSfu: any;
+    }): boolean;
+    supportsScreenShareAudio(): boolean;
+    addNewStream(streamId: string, stream: MediaStream, audioPaused: boolean, videoPaused: boolean, beforeEffectTracks?: CustomMediaStreamTrack[]): void;
+    replaceTrack(oldTrack: CustomMediaStreamTrack | null, newTrack: CustomMediaStreamTrack): Promise<any[]>;
+    close(): void;
+    disconnectAll(): void;
+    fixChromeAudio(constraints: any): Promise<any[]> | undefined;
+    setupSocketListeners(): void;
+    sendAudioMutedStats(muted: boolean): void;
+    sendVideoMutedStats(muted: boolean): void;
+    sendStatsCustomEvent(eventName: string, data: any): void;
+    rtcStatsDisconnect(): void;
+    rtcStatsReconnect(): void;
+    setAudioOnly(audioOnly: any): void;
+    setRemoteScreenshareVideoTrackIds(remoteScreenshareVideoTrackIds?: never[]): void;
+    setRoomSessionId(roomSessionId: string): void;
+    _setConnectionStatus(session: any, newStatus: any, clientId: string): void;
+    _setJitterBufferTarget(pc: any): void;
+    _emitServerEvent(eventName: string, data?: any): void;
+    _emit(eventName: string, data?: any): void;
+    _addEnabledLocalStreamId(streamId: string): void;
+    _deleteEnabledLocalStreamId(streamId: string): void;
+    _getSession(peerConnectionId: string): any;
+    _getOrCreateSession(peerConnectionId: string, initialBandwidth: any): any;
+    _getLocalCameraStream(): any;
+    _getNonLocalCameraStreamIds(): string[];
+    _isScreensharingLocally(): boolean;
+    _getFirstLocalNonCameraStream(): any;
+    _transformIncomingSdp(original: any, _: any): {
+        type: any;
+        sdp: any;
+    };
+    _transformOutgoingSdp(original: any): {
+        type: any;
+        sdpU: any;
+    };
+    _createSession({ clientId, initialBandwidth, isOfferer, peerConnectionId, shouldAddLocalVideo, }: {
+        clientId: string;
+        initialBandwidth: any;
+        isOfferer: any;
+        peerConnectionId: string;
+        shouldAddLocalVideo: boolean;
+    }): any;
+    _cleanup(peerConnectionId: string): void;
+    _forEachPeerConnection(func: any): void;
+    _addStreamToPeerConnections(stream: any): void;
+    _addTrackToPeerConnections(track: any, stream?: any): void;
+    _replaceTrackToPeerConnections(oldTrack: any, newTrack: any): Promise<any[]>;
+    _removeStreamFromPeerConnections(stream: any): void;
+    _removeTrackFromPeerConnections(track: any): void;
+    _addLocalStream(streamId: string, stream: any): void;
+    _removeLocalStream(streamId: string): void;
+    _updateAndScheduleMediaServersRefresh({ iceServers, turnServers, sfuServer, mediaserverConfigTtlSeconds }: any): void;
+    _clearMediaServersRefresh(): void;
+    _monitorAudioTrack(track: any): void;
+    _monitorVideoTrack(track: CustomMediaStreamTrack): void;
+    _connect(clientId: string): Promise<any>;
+    _maybeRestartIce(clientId: string, session: any): void;
+    _setCodecPreferences(pc: RTCPeerConnection): Promise<void>;
+    _negotiatePeerConnection(clientId: string, session: any, constraints?: any): void;
+    _withForcedRenegotiation(session: any, action: any): void;
+    _changeBandwidthForAllClients(isJoining: boolean): number;
+    _createP2pSession({ clientId, initialBandwidth, shouldAddLocalVideo, isOfferer, }: {
+        clientId: string;
+        initialBandwidth: number;
+        shouldAddLocalVideo: boolean;
+        isOfferer: boolean;
+    }): any;
+    acceptNewStream({ streamId, clientId, shouldAddLocalVideo, }: {
+        streamId: string;
+        clientId: string;
+        shouldAddLocalVideo?: boolean;
+    }): any;
+    disconnect(clientId: string): void;
+    updateStreamResolution(): void;
+    stopOrResumeAudio(): void;
+    _handleStopOrResumeVideo({ enable, track }: {
+        enable: boolean;
+        track: any;
+    }): void;
+    stopOrResumeVideo(localStream: any, enable: boolean): void;
+    _shareScreen(streamId: string, stream: any): void;
+    removeStream(streamId: string, stream: any, requestedByClientId: any): void;
+    hasClient(clientId: string): boolean;
+}
+
 declare class RtcManagerDispatcher {
     emitter: {
         emit: <K extends keyof RtcEvents>(eventName: K, args?: RtcEvents[K]) => void;
@@ -1626,6 +1628,7 @@ declare class VegaRtcManager implements RtcManager {
         width: number;
         height: number;
     }): void;
+    close(): void;
     disconnectAll(): void;
     sendAudioMutedStats(muted: boolean): void;
     sendVideoMutedStats(muted: boolean): void;

package/dist/index.d.mts

@@ -490,140 +490,6 @@ declare function getUpdatedDevices({ oldDevices, newDevices, currentAudioId, cur
     currentSpeakerId?: string | undefined;
 }): GetUpdatedDevicesResult;
 
-declare class P2pRtcManager implements RtcManager {
-    _selfId: any;
-    _roomName: any;
-    _roomSessionId: any;
-    peerConnections: any;
-    localStreams: any;
-    enabledLocalStreamIds: any[];
-    _screenshareVideoTrackIds: any[];
-    _socketListenerDeregisterFunctions: any[];
-    _localStreamDeregisterFunction: any;
-    _emitter: any;
-    _serverSocket: any;
-    _webrtcProvider: any;
-    _features: any;
-    _isAudioOnlyMode: boolean;
-    offerOptions: {
-        offerToReceiveAudio: boolean;
-        offerToReceiveVideo: boolean;
-    };
-    _pendingActionsForConnectedPeerConnections: any[];
-    _audioTrackOnEnded: () => void;
-    _videoTrackOnEnded: () => void;
-    totalSessionsCreated: number;
-    _iceServers: any;
-    _turnServers: any;
-    _sfuServer: any;
-    _mediaserverConfigTtlSeconds: any;
-    _fetchMediaServersTimer: any;
-    _wasScreenSharing: any;
-    ipv6HostCandidateTeredoSeen: any;
-    ipv6HostCandidate6to4Seen: any;
-    mdnsHostCandidateSeen: any;
-    _stoppedVideoTrack: any;
-    icePublicIPGatheringTimeoutID: any;
-    _videoTrackBeingMonitored?: CustomMediaStreamTrack;
-    _audioTrackBeingMonitored?: CustomMediaStreamTrack;
-    constructor({ selfId, room, emitter, serverSocket, webrtcProvider, features, }: {
-        selfId: any;
-        room: any;
-        emitter: any;
-        serverSocket: any;
-        webrtcProvider: any;
-        features: any;
-    });
-    numberOfPeerconnections(): number;
-    isInitializedWith({ selfId, roomName, isSfu }: {
-        selfId: any;
-        roomName: any;
-        isSfu: any;
-    }): boolean;
-    supportsScreenShareAudio(): boolean;
-    addNewStream(streamId: string, stream: MediaStream, audioPaused: boolean, videoPaused: boolean, beforeEffectTracks?: CustomMediaStreamTrack[]): void;
-    replaceTrack(oldTrack: CustomMediaStreamTrack | null, newTrack: CustomMediaStreamTrack): Promise<any[]>;
-    disconnectAll(): void;
-    fixChromeAudio(constraints: any): Promise<any[]> | undefined;
-    setupSocketListeners(): void;
-    sendAudioMutedStats(muted: boolean): void;
-    sendVideoMutedStats(muted: boolean): void;
-    sendStatsCustomEvent(eventName: string, data: any): void;
-    rtcStatsDisconnect(): void;
-    rtcStatsReconnect(): void;
-    setAudioOnly(audioOnly: any): void;
-    setRemoteScreenshareVideoTrackIds(remoteScreenshareVideoTrackIds?: never[]): void;
-    setRoomSessionId(roomSessionId: string): void;
-    _setConnectionStatus(session: any, newStatus: any, clientId: string): void;
-    _setJitterBufferTarget(pc: any): void;
-    _emitServerEvent(eventName: string, data?: any, callback?: any): void;
-    _emit(eventName: string, data?: any): void;
-    _addEnabledLocalStreamId(streamId: string): void;
-    _deleteEnabledLocalStreamId(streamId: string): void;
-    _getSession(peerConnectionId: string): any;
-    _getOrCreateSession(peerConnectionId: string, initialBandwidth: any): any;
-    _getLocalCameraStream(): any;
-    _getNonLocalCameraStreamIds(): string[];
-    _isScreensharingLocally(): boolean;
-    _getFirstLocalNonCameraStream(): any;
-    _transformIncomingSdp(original: any, _: any): {
-        type: any;
-        sdp: any;
-    };
-    _transformOutgoingSdp(original: any): {
-        type: any;
-        sdpU: any;
-    };
-    _createSession({ clientId, initialBandwidth, isOfferer, peerConnectionId, shouldAddLocalVideo, }: {
-        clientId: string;
-        initialBandwidth: any;
-        isOfferer: any;
-        peerConnectionId: string;
-        shouldAddLocalVideo: boolean;
-    }): any;
-    _cleanup(peerConnectionId: string): void;
-    _forEachPeerConnection(func: any): void;
-    _addStreamToPeerConnections(stream: any): void;
-    _addTrackToPeerConnections(track: any, stream?: any): void;
-    _replaceTrackToPeerConnections(oldTrack: any, newTrack: any): Promise<any[]>;
-    _removeStreamFromPeerConnections(stream: any): void;
-    _removeTrackFromPeerConnections(track: any): void;
-    _addLocalStream(streamId: string, stream: any): void;
-    _removeLocalStream(streamId: string): void;
-    _updateAndScheduleMediaServersRefresh({ iceServers, turnServers, sfuServer, mediaserverConfigTtlSeconds }: any): void;
-    _clearMediaServersRefresh(): void;
-    _monitorAudioTrack(track: any): void;
-    _monitorVideoTrack(track: CustomMediaStreamTrack): void;
-    _connect(clientId: string): Promise<any>;
-    _maybeRestartIce(clientId: string, session: any): void;
-    _setCodecPreferences(pc: RTCPeerConnection): Promise<void>;
-    _negotiatePeerConnection(clientId: string, session: any, constraints?: any): void;
-    _withForcedRenegotiation(session: any, action: any): void;
-    _changeBandwidthForAllClients(isJoining: boolean): number;
-    _createP2pSession({ clientId, initialBandwidth, shouldAddLocalVideo, isOfferer, }: {
-        clientId: string;
-        initialBandwidth: number;
-        shouldAddLocalVideo: boolean;
-        isOfferer: boolean;
-    }): any;
-    acceptNewStream({ streamId, clientId, shouldAddLocalVideo, }: {
-        streamId: string;
-        clientId: string;
-        shouldAddLocalVideo?: boolean;
-    }): any;
-    disconnect(clientId: string): void;
-    updateStreamResolution(): void;
-    stopOrResumeAudio(): void;
-    _handleStopOrResumeVideo({ enable, track }: {
-        enable: boolean;
-        track: any;
-    }): void;
-    stopOrResumeVideo(localStream: any, enable: boolean): void;
-    _shareScreen(streamId: string, stream: any): void;
-    removeStream(streamId: string, stream: any, requestedByClientId: any): void;
-    hasClient(clientId: string): boolean;
-}
-
 declare const assert: {
     fail: (message?: string | Error) => void;
     ok: (value: any, message?: string | Error) => void;
@@ -774,13 +640,12 @@ declare class ServerSocket {
     noopKeepaliveInterval: any;
     _wasConnectedUsingWebsocket?: boolean;
     disconnectTimestamp: number | undefined;
+    joinRoomFinished: boolean;
     constructor(hostName: string, optionsOverrides?: any, glitchFree?: boolean);
     setRtcManager(rtcManager?: RtcManager): void;
     connect(): void;
     disconnect(): void;
-    disconnectOnConnect(): void;
     emit(eventName: string, ...args: any[]): void;
-    emitIfConnected(eventName: string, data: any): void;
     getTransport(): any;
     getManager(): any;
     isConnecting(): any;
@@ -1206,6 +1071,143 @@ declare function fromLocation({ host, protocol }?: {
     subdomain: string;
 };
 
+declare class P2pRtcManager implements RtcManager {
+    _selfId: any;
+    _roomName: any;
+    _roomSessionId: any;
+    peerConnections: any;
+    localStreams: any;
+    enabledLocalStreamIds: any[];
+    _screenshareVideoTrackIds: any[];
+    _socketListenerDeregisterFunctions: any[];
+    _localStreamDeregisterFunction: any;
+    _emitter: any;
+    _serverSocket: ServerSocket;
+    _webrtcProvider: any;
+    _features: any;
+    _isAudioOnlyMode: boolean;
+    offerOptions: {
+        offerToReceiveAudio: boolean;
+        offerToReceiveVideo: boolean;
+    };
+    _pendingActionsForConnectedPeerConnections: any[];
+    _audioTrackOnEnded: () => void;
+    _videoTrackOnEnded: () => void;
+    totalSessionsCreated: number;
+    _iceServers: any;
+    _turnServers: any;
+    _sfuServer: any;
+    _mediaserverConfigTtlSeconds: any;
+    _fetchMediaServersTimer: any;
+    _wasScreenSharing: any;
+    ipv6HostCandidateTeredoSeen: any;
+    ipv6HostCandidate6to4Seen: any;
+    mdnsHostCandidateSeen: any;
+    _stoppedVideoTrack: any;
+    icePublicIPGatheringTimeoutID: any;
+    _videoTrackBeingMonitored?: CustomMediaStreamTrack;
+    _audioTrackBeingMonitored?: CustomMediaStreamTrack;
+    _closed: boolean;
+    skipEmittingServerMessageCount: number;
+    constructor({ selfId, room, emitter, serverSocket, webrtcProvider, features, }: {
+        selfId: any;
+        room: any;
+        emitter: any;
+        serverSocket: ServerSocket;
+        webrtcProvider: any;
+        features: any;
+    });
+    numberOfPeerconnections(): number;
+    isInitializedWith({ selfId, roomName, isSfu }: {
+        selfId: any;
+        roomName: any;
+        isSfu: any;
+    }): boolean;
+    supportsScreenShareAudio(): boolean;
+    addNewStream(streamId: string, stream: MediaStream, audioPaused: boolean, videoPaused: boolean, beforeEffectTracks?: CustomMediaStreamTrack[]): void;
+    replaceTrack(oldTrack: CustomMediaStreamTrack | null, newTrack: CustomMediaStreamTrack): Promise<any[]>;
+    close(): void;
+    disconnectAll(): void;
+    fixChromeAudio(constraints: any): Promise<any[]> | undefined;
+    setupSocketListeners(): void;
+    sendAudioMutedStats(muted: boolean): void;
+    sendVideoMutedStats(muted: boolean): void;
+    sendStatsCustomEvent(eventName: string, data: any): void;
+    rtcStatsDisconnect(): void;
+    rtcStatsReconnect(): void;
+    setAudioOnly(audioOnly: any): void;
+    setRemoteScreenshareVideoTrackIds(remoteScreenshareVideoTrackIds?: never[]): void;
+    setRoomSessionId(roomSessionId: string): void;
+    _setConnectionStatus(session: any, newStatus: any, clientId: string): void;
+    _setJitterBufferTarget(pc: any): void;
+    _emitServerEvent(eventName: string, data?: any): void;
+    _emit(eventName: string, data?: any): void;
+    _addEnabledLocalStreamId(streamId: string): void;
+    _deleteEnabledLocalStreamId(streamId: string): void;
+    _getSession(peerConnectionId: string): any;
+    _getOrCreateSession(peerConnectionId: string, initialBandwidth: any): any;
+    _getLocalCameraStream(): any;
+    _getNonLocalCameraStreamIds(): string[];
+    _isScreensharingLocally(): boolean;
+    _getFirstLocalNonCameraStream(): any;
+    _transformIncomingSdp(original: any, _: any): {
+        type: any;
+        sdp: any;
+    };
+    _transformOutgoingSdp(original: any): {
+        type: any;
+        sdpU: any;
+    };
+    _createSession({ clientId, initialBandwidth, isOfferer, peerConnectionId, shouldAddLocalVideo, }: {
+        clientId: string;
+        initialBandwidth: any;
+        isOfferer: any;
+        peerConnectionId: string;
+        shouldAddLocalVideo: boolean;
+    }): any;
+    _cleanup(peerConnectionId: string): void;
+    _forEachPeerConnection(func: any): void;
+    _addStreamToPeerConnections(stream: any): void;
+    _addTrackToPeerConnections(track: any, stream?: any): void;
+    _replaceTrackToPeerConnections(oldTrack: any, newTrack: any): Promise<any[]>;
+    _removeStreamFromPeerConnections(stream: any): void;
+    _removeTrackFromPeerConnections(track: any): void;
+    _addLocalStream(streamId: string, stream: any): void;
+    _removeLocalStream(streamId: string): void;
+    _updateAndScheduleMediaServersRefresh({ iceServers, turnServers, sfuServer, mediaserverConfigTtlSeconds }: any): void;
+    _clearMediaServersRefresh(): void;
+    _monitorAudioTrack(track: any): void;
+    _monitorVideoTrack(track: CustomMediaStreamTrack): void;
+    _connect(clientId: string): Promise<any>;
+    _maybeRestartIce(clientId: string, session: any): void;
+    _setCodecPreferences(pc: RTCPeerConnection): Promise<void>;
+    _negotiatePeerConnection(clientId: string, session: any, constraints?: any): void;
+    _withForcedRenegotiation(session: any, action: any): void;
+    _changeBandwidthForAllClients(isJoining: boolean): number;
+    _createP2pSession({ clientId, initialBandwidth, shouldAddLocalVideo, isOfferer, }: {
+        clientId: string;
+        initialBandwidth: number;
+        shouldAddLocalVideo: boolean;
+        isOfferer: boolean;
+    }): any;
+    acceptNewStream({ streamId, clientId, shouldAddLocalVideo, }: {
+        streamId: string;
+        clientId: string;
+        shouldAddLocalVideo?: boolean;
+    }): any;
+    disconnect(clientId: string): void;
+    updateStreamResolution(): void;
+    stopOrResumeAudio(): void;
+    _handleStopOrResumeVideo({ enable, track }: {
+        enable: boolean;
+        track: any;
+    }): void;
+    stopOrResumeVideo(localStream: any, enable: boolean): void;
+    _shareScreen(streamId: string, stream: any): void;
+    removeStream(streamId: string, stream: any, requestedByClientId: any): void;
+    hasClient(clientId: string): boolean;
+}
+
 declare class RtcManagerDispatcher {
     emitter: {
         emit: <K extends keyof RtcEvents>(eventName: K, args?: RtcEvents[K]) => void;
@@ -1626,6 +1628,7 @@ declare class VegaRtcManager implements RtcManager {
         width: number;
         height: number;
     }): void;
+    close(): void;
     disconnectAll(): void;
     sendAudioMutedStats(muted: boolean): void;
     sendVideoMutedStats(muted: boolean): void;

package/dist/index.d.ts

@@ -490,140 +490,6 @@ declare function getUpdatedDevices({ oldDevices, newDevices, currentAudioId, cur
     currentSpeakerId?: string | undefined;
 }): GetUpdatedDevicesResult;
 
-declare class P2pRtcManager implements RtcManager {
-    _selfId: any;
-    _roomName: any;
-    _roomSessionId: any;
-    peerConnections: any;
-    localStreams: any;
-    enabledLocalStreamIds: any[];
-    _screenshareVideoTrackIds: any[];
-    _socketListenerDeregisterFunctions: any[];
-    _localStreamDeregisterFunction: any;
-    _emitter: any;
-    _serverSocket: any;
-    _webrtcProvider: any;
-    _features: any;
-    _isAudioOnlyMode: boolean;
-    offerOptions: {
-        offerToReceiveAudio: boolean;
-        offerToReceiveVideo: boolean;
-    };
-    _pendingActionsForConnectedPeerConnections: any[];
-    _audioTrackOnEnded: () => void;
-    _videoTrackOnEnded: () => void;
-    totalSessionsCreated: number;
-    _iceServers: any;
-    _turnServers: any;
-    _sfuServer: any;
-    _mediaserverConfigTtlSeconds: any;
-    _fetchMediaServersTimer: any;
-    _wasScreenSharing: any;
-    ipv6HostCandidateTeredoSeen: any;
-    ipv6HostCandidate6to4Seen: any;
-    mdnsHostCandidateSeen: any;
-    _stoppedVideoTrack: any;
-    icePublicIPGatheringTimeoutID: any;
-    _videoTrackBeingMonitored?: CustomMediaStreamTrack;
-    _audioTrackBeingMonitored?: CustomMediaStreamTrack;
-    constructor({ selfId, room, emitter, serverSocket, webrtcProvider, features, }: {
-        selfId: any;
-        room: any;
-        emitter: any;
-        serverSocket: any;
-        webrtcProvider: any;
-        features: any;
-    });
-    numberOfPeerconnections(): number;
-    isInitializedWith({ selfId, roomName, isSfu }: {
-        selfId: any;
-        roomName: any;
-        isSfu: any;
-    }): boolean;
-    supportsScreenShareAudio(): boolean;
-    addNewStream(streamId: string, stream: MediaStream, audioPaused: boolean, videoPaused: boolean, beforeEffectTracks?: CustomMediaStreamTrack[]): void;
-    replaceTrack(oldTrack: CustomMediaStreamTrack | null, newTrack: CustomMediaStreamTrack): Promise<any[]>;
-    disconnectAll(): void;
-    fixChromeAudio(constraints: any): Promise<any[]> | undefined;
-    setupSocketListeners(): void;
-    sendAudioMutedStats(muted: boolean): void;
-    sendVideoMutedStats(muted: boolean): void;
-    sendStatsCustomEvent(eventName: string, data: any): void;
-    rtcStatsDisconnect(): void;
-    rtcStatsReconnect(): void;
-    setAudioOnly(audioOnly: any): void;
-    setRemoteScreenshareVideoTrackIds(remoteScreenshareVideoTrackIds?: never[]): void;
-    setRoomSessionId(roomSessionId: string): void;
-    _setConnectionStatus(session: any, newStatus: any, clientId: string): void;
-    _setJitterBufferTarget(pc: any): void;
-    _emitServerEvent(eventName: string, data?: any, callback?: any): void;
-    _emit(eventName: string, data?: any): void;
-    _addEnabledLocalStreamId(streamId: string): void;
-    _deleteEnabledLocalStreamId(streamId: string): void;
-    _getSession(peerConnectionId: string): any;
-    _getOrCreateSession(peerConnectionId: string, initialBandwidth: any): any;
-    _getLocalCameraStream(): any;
-    _getNonLocalCameraStreamIds(): string[];
-    _isScreensharingLocally(): boolean;
-    _getFirstLocalNonCameraStream(): any;
-    _transformIncomingSdp(original: any, _: any): {
-        type: any;
-        sdp: any;
-    };
-    _transformOutgoingSdp(original: any): {
-        type: any;
-        sdpU: any;
-    };
-    _createSession({ clientId, initialBandwidth, isOfferer, peerConnectionId, shouldAddLocalVideo, }: {
-        clientId: string;
-        initialBandwidth: any;
-        isOfferer: any;
-        peerConnectionId: string;
-        shouldAddLocalVideo: boolean;
-    }): any;
-    _cleanup(peerConnectionId: string): void;
-    _forEachPeerConnection(func: any): void;
-    _addStreamToPeerConnections(stream: any): void;
-    _addTrackToPeerConnections(track: any, stream?: any): void;
-    _replaceTrackToPeerConnections(oldTrack: any, newTrack: any): Promise<any[]>;
-    _removeStreamFromPeerConnections(stream: any): void;
-    _removeTrackFromPeerConnections(track: any): void;
-    _addLocalStream(streamId: string, stream: any): void;
-    _removeLocalStream(streamId: string): void;
-    _updateAndScheduleMediaServersRefresh({ iceServers, turnServers, sfuServer, mediaserverConfigTtlSeconds }: any): void;
-    _clearMediaServersRefresh(): void;
-    _monitorAudioTrack(track: any): void;
-    _monitorVideoTrack(track: CustomMediaStreamTrack): void;
-    _connect(clientId: string): Promise<any>;
-    _maybeRestartIce(clientId: string, session: any): void;
-    _setCodecPreferences(pc: RTCPeerConnection): Promise<void>;
-    _negotiatePeerConnection(clientId: string, session: any, constraints?: any): void;
-    _withForcedRenegotiation(session: any, action: any): void;
-    _changeBandwidthForAllClients(isJoining: boolean): number;
-    _createP2pSession({ clientId, initialBandwidth, shouldAddLocalVideo, isOfferer, }: {
-        clientId: string;
-        initialBandwidth: number;
-        shouldAddLocalVideo: boolean;
-        isOfferer: boolean;
-    }): any;
-    acceptNewStream({ streamId, clientId, shouldAddLocalVideo, }: {
-        streamId: string;
-        clientId: string;
-        shouldAddLocalVideo?: boolean;
-    }): any;
-    disconnect(clientId: string): void;
-    updateStreamResolution(): void;
-    stopOrResumeAudio(): void;
-    _handleStopOrResumeVideo({ enable, track }: {
-        enable: boolean;
-        track: any;
-    }): void;
-    stopOrResumeVideo(localStream: any, enable: boolean): void;
-    _shareScreen(streamId: string, stream: any): void;
-    removeStream(streamId: string, stream: any, requestedByClientId: any): void;
-    hasClient(clientId: string): boolean;
-}
-
 declare const assert: {
     fail: (message?: string | Error) => void;
     ok: (value: any, message?: string | Error) => void;
@@ -774,13 +640,12 @@ declare class ServerSocket {
     noopKeepaliveInterval: any;
     _wasConnectedUsingWebsocket?: boolean;
     disconnectTimestamp: number | undefined;
+    joinRoomFinished: boolean;
     constructor(hostName: string, optionsOverrides?: any, glitchFree?: boolean);
     setRtcManager(rtcManager?: RtcManager): void;
     connect(): void;
     disconnect(): void;
-    disconnectOnConnect(): void;
     emit(eventName: string, ...args: any[]): void;
-    emitIfConnected(eventName: string, data: any): void;
     getTransport(): any;
     getManager(): any;
     isConnecting(): any;
@@ -1206,6 +1071,143 @@ declare function fromLocation({ host, protocol }?: {
     subdomain: string;
 };
 
+declare class P2pRtcManager implements RtcManager {
+    _selfId: any;
+    _roomName: any;
+    _roomSessionId: any;
+    peerConnections: any;
+    localStreams: any;
+    enabledLocalStreamIds: any[];
+    _screenshareVideoTrackIds: any[];
+    _socketListenerDeregisterFunctions: any[];
+    _localStreamDeregisterFunction: any;
+    _emitter: any;
+    _serverSocket: ServerSocket;
+    _webrtcProvider: any;
+    _features: any;
+    _isAudioOnlyMode: boolean;
+    offerOptions: {
+        offerToReceiveAudio: boolean;
+        offerToReceiveVideo: boolean;
+    };
+    _pendingActionsForConnectedPeerConnections: any[];
+    _audioTrackOnEnded: () => void;
+    _videoTrackOnEnded: () => void;
+    totalSessionsCreated: number;
+    _iceServers: any;
+    _turnServers: any;
+    _sfuServer: any;
+    _mediaserverConfigTtlSeconds: any;
+    _fetchMediaServersTimer: any;
+    _wasScreenSharing: any;
+    ipv6HostCandidateTeredoSeen: any;
+    ipv6HostCandidate6to4Seen: any;
+    mdnsHostCandidateSeen: any;
+    _stoppedVideoTrack: any;
+    icePublicIPGatheringTimeoutID: any;
+    _videoTrackBeingMonitored?: CustomMediaStreamTrack;
+    _audioTrackBeingMonitored?: CustomMediaStreamTrack;
+    _closed: boolean;
+    skipEmittingServerMessageCount: number;
+    constructor({ selfId, room, emitter, serverSocket, webrtcProvider, features, }: {
+        selfId: any;
+        room: any;
+        emitter: any;
+        serverSocket: ServerSocket;
+        webrtcProvider: any;
+        features: any;
+    });
+    numberOfPeerconnections(): number;
+    isInitializedWith({ selfId, roomName, isSfu }: {
+        selfId: any;
+        roomName: any;
+        isSfu: any;
+    }): boolean;
+    supportsScreenShareAudio(): boolean;
+    addNewStream(streamId: string, stream: MediaStream, audioPaused: boolean, videoPaused: boolean, beforeEffectTracks?: CustomMediaStreamTrack[]): void;
+    replaceTrack(oldTrack: CustomMediaStreamTrack | null, newTrack: CustomMediaStreamTrack): Promise<any[]>;
+    close(): void;
+    disconnectAll(): void;
+    fixChromeAudio(constraints: any): Promise<any[]> | undefined;
+    setupSocketListeners(): void;
+    sendAudioMutedStats(muted: boolean): void;
+    sendVideoMutedStats(muted: boolean): void;
+    sendStatsCustomEvent(eventName: string, data: any): void;
+    rtcStatsDisconnect(): void;
+    rtcStatsReconnect(): void;
+    setAudioOnly(audioOnly: any): void;
+    setRemoteScreenshareVideoTrackIds(remoteScreenshareVideoTrackIds?: never[]): void;
+    setRoomSessionId(roomSessionId: string): void;
+    _setConnectionStatus(session: any, newStatus: any, clientId: string): void;
+    _setJitterBufferTarget(pc: any): void;
+    _emitServerEvent(eventName: string, data?: any): void;
+    _emit(eventName: string, data?: any): void;
+    _addEnabledLocalStreamId(streamId: string): void;
+    _deleteEnabledLocalStreamId(streamId: string): void;
+    _getSession(peerConnectionId: string): any;
+    _getOrCreateSession(peerConnectionId: string, initialBandwidth: any): any;
+    _getLocalCameraStream(): any;
+    _getNonLocalCameraStreamIds(): string[];
+    _isScreensharingLocally(): boolean;
+    _getFirstLocalNonCameraStream(): any;
+    _transformIncomingSdp(original: any, _: any): {
+        type: any;
+        sdp: any;
+    };
+    _transformOutgoingSdp(original: any): {
+        type: any;
+        sdpU: any;
+    };
+    _createSession({ clientId, initialBandwidth, isOfferer, peerConnectionId, shouldAddLocalVideo, }: {
+        clientId: string;
+        initialBandwidth: any;
+        isOfferer: any;
+        peerConnectionId: string;
+        shouldAddLocalVideo: boolean;
+    }): any;
+    _cleanup(peerConnectionId: string): void;
+    _forEachPeerConnection(func: any): void;
+    _addStreamToPeerConnections(stream: any): void;
+    _addTrackToPeerConnections(track: any, stream?: any): void;
+    _replaceTrackToPeerConnections(oldTrack: any, newTrack: any): Promise<any[]>;
+    _removeStreamFromPeerConnections(stream: any): void;
+    _removeTrackFromPeerConnections(track: any): void;
+    _addLocalStream(streamId: string, stream: any): void;
+    _removeLocalStream(streamId: string): void;
+    _updateAndScheduleMediaServersRefresh({ iceServers, turnServers, sfuServer, mediaserverConfigTtlSeconds }: any): void;
+    _clearMediaServersRefresh(): void;
+    _monitorAudioTrack(track: any): void;
+    _monitorVideoTrack(track: CustomMediaStreamTrack): void;
+    _connect(clientId: string): Promise<any>;
+    _maybeRestartIce(clientId: string, session: any): void;
+    _setCodecPreferences(pc: RTCPeerConnection): Promise<void>;
+    _negotiatePeerConnection(clientId: string, session: any, constraints?: any): void;
+    _withForcedRenegotiation(session: any, action: any): void;
+    _changeBandwidthForAllClients(isJoining: boolean): number;
+    _createP2pSession({ clientId, initialBandwidth, shouldAddLocalVideo, isOfferer, }: {
+        clientId: string;
+        initialBandwidth: number;
+        shouldAddLocalVideo: boolean;
+        isOfferer: boolean;
+    }): any;
+    acceptNewStream({ streamId, clientId, shouldAddLocalVideo, }: {
+        streamId: string;
+        clientId: string;
+        shouldAddLocalVideo?: boolean;
+    }): any;
+    disconnect(clientId: string): void;
+    updateStreamResolution(): void;
+    stopOrResumeAudio(): void;
+    _handleStopOrResumeVideo({ enable, track }: {
+        enable: boolean;
+        track: any;
+    }): void;
+    stopOrResumeVideo(localStream: any, enable: boolean): void;
+    _shareScreen(streamId: string, stream: any): void;
+    removeStream(streamId: string, stream: any, requestedByClientId: any): void;
+    hasClient(clientId: string): boolean;
+}
+
 declare class RtcManagerDispatcher {
     emitter: {
         emit: <K extends keyof RtcEvents>(eventName: K, args?: RtcEvents[K]) => void;
@@ -1626,6 +1628,7 @@ declare class VegaRtcManager implements RtcManager {
         width: number;
         height: number;
     }): void;
+    close(): void;
     disconnectAll(): void;
     sendAudioMutedStats(muted: boolean): void;
     sendVideoMutedStats(muted: boolean): void;

package/dist/index.mjs

@@ -1547,6 +1547,7 @@ class ServerSocket {
         this._wasConnectedUsingWebsocket = false;
         this._reconnectManager = null;
         this._socket = io(hostName, Object.assign({ path: DEFAULT_SOCKET_PATH, randomizationFactor: 0.5, reconnectionDelay: 250, reconnectionDelayMax: 5000, timeout: 5000, transports: ["websocket"], withCredentials: true }, optionsOverrides));
+        this.joinRoomFinished = false;
         this._socket.io.on("reconnect", () => {
             this._socket.sendBuffer = [];
         });
@@ -1564,6 +1565,12 @@ class ServerSocket {
         });
         if (glitchFree)
             this._reconnectManager = new ReconnectManager(this._socket);
+        this._socket.on("room_joined", (payload) => {
+            const { error } = payload;
+            if (!error) {
+                this.joinRoomFinished = true;
+            }
+        });
         this._socket.on("connect", () => {
             const transport = this.getTransport();
             if (transport === "websocket") {
@@ -1580,6 +1587,7 @@ class ServerSocket {
             }
         });
         this._socket.on("disconnect", () => {
+            this.joinRoomFinished = false;
             this.disconnectTimestamp = Date.now();
             if (this.noopKeepaliveInterval) {
                 clearInterval(this.noopKeepaliveInterval);
@@ -1601,26 +1609,12 @@ class ServerSocket {
     disconnect() {
         this._socket.disconnect();
     }
-    disconnectOnConnect() {
-        this._socket.once("connect", () => {
-            this._socket.disconnect();
-        });
-    }
     emit(eventName, ...args) {
         this._socket.emit.apply(this._socket, arguments);
     }
-    emitIfConnected(eventName, data) {
-        if (!this.isConnected()) {
-            return;
-        }
-        this.emit(eventName, data);
-    }
     getTransport() {
-        return (this._socket &&
-            this._socket.io &&
-            this._socket.io.engine &&
-            this._socket.io.engine.transport &&
-            this._socket.io.engine.transport.name);
+        var _a, _b, _c, _d;
+        return (_d = (_c = (_b = (_a = this._socket) === null || _a === void 0 ? void 0 : _a.io) === null || _b === void 0 ? void 0 : _b.engine) === null || _c === void 0 ? void 0 : _c.transport) === null || _d === void 0 ? void 0 : _d.name;
     }
     getManager() {
         return this._socket.io;
@@ -2582,6 +2576,8 @@ class P2pRtcManager {
         this._webrtcProvider = webrtcProvider;
         this._features = features || {};
         this._isAudioOnlyMode = false;
+        this._closed = false;
+        this.skipEmittingServerMessageCount = 0;
         this.offerOptions = { offerToReceiveAudio: true, offerToReceiveVideo: true };
         this._pendingActionsForConnectedPeerConnections = [];
         this._audioTrackOnEnded = () => {
@@ -2663,6 +2659,10 @@ class P2pRtcManager {
         }
         return this._replaceTrackToPeerConnections(oldTrack, newTrack);
     }
+    close() {
+        this._closed = true;
+        this.disconnectAll();
+    }
     disconnectAll() {
         Object.keys(this.peerConnections).forEach((peerConnectionId) => {
             this.disconnect(peerConnectionId);
@@ -2848,8 +2848,18 @@ class P2pRtcManager {
             logger$6.error("Error during setting jitter buffer target:", error);
         }
     }
-    _emitServerEvent(eventName, data, callback) {
-        this._serverSocket.emit(eventName, data, callback);
+    _emitServerEvent(eventName, data) {
+        if (this._closed) {
+            logger$6.warn("RtcManager closed. Will not send event", eventName, data);
+            return;
+        }
+        if (this._features.awaitJoinRoomFinished && !this._serverSocket.joinRoomFinished) {
+            rtcStats.sendEvent("skip_emitting_server_message", { eventName });
+            this.skipEmittingServerMessageCount++;
+        }
+        else {
+            this._serverSocket.emit(eventName, data);
+        }
     }
     _emit(eventName, data) {
         this._emitter.emit(eventName, data);
@@ -5485,6 +5495,9 @@ class VegaRtcManager {
             });
         }
     }
+    close() {
+        this.disconnectAll();
+    }
     disconnectAll() {
         var _a, _b, _c, _d;
         this._reconnect = false;

package/dist/legacy-esm.js

@@ -1547,6 +1547,7 @@ class ServerSocket {
         this._wasConnectedUsingWebsocket = false;
         this._reconnectManager = null;
         this._socket = io(hostName, Object.assign({ path: DEFAULT_SOCKET_PATH, randomizationFactor: 0.5, reconnectionDelay: 250, reconnectionDelayMax: 5000, timeout: 5000, transports: ["websocket"], withCredentials: true }, optionsOverrides));
+        this.joinRoomFinished = false;
         this._socket.io.on("reconnect", () => {
             this._socket.sendBuffer = [];
         });
@@ -1564,6 +1565,12 @@ class ServerSocket {
         });
         if (glitchFree)
             this._reconnectManager = new ReconnectManager(this._socket);
+        this._socket.on("room_joined", (payload) => {
+            const { error } = payload;
+            if (!error) {
+                this.joinRoomFinished = true;
+            }
+        });
         this._socket.on("connect", () => {
             const transport = this.getTransport();
             if (transport === "websocket") {
@@ -1580,6 +1587,7 @@ class ServerSocket {
             }
         });
         this._socket.on("disconnect", () => {
+            this.joinRoomFinished = false;
             this.disconnectTimestamp = Date.now();
             if (this.noopKeepaliveInterval) {
                 clearInterval(this.noopKeepaliveInterval);
@@ -1601,26 +1609,12 @@ class ServerSocket {
     disconnect() {
         this._socket.disconnect();
     }
-    disconnectOnConnect() {
-        this._socket.once("connect", () => {
-            this._socket.disconnect();
-        });
-    }
     emit(eventName, ...args) {
         this._socket.emit.apply(this._socket, arguments);
     }
-    emitIfConnected(eventName, data) {
-        if (!this.isConnected()) {
-            return;
-        }
-        this.emit(eventName, data);
-    }
     getTransport() {
-        return (this._socket &&
-            this._socket.io &&
-            this._socket.io.engine &&
-            this._socket.io.engine.transport &&
-            this._socket.io.engine.transport.name);
+        var _a, _b, _c, _d;
+        return (_d = (_c = (_b = (_a = this._socket) === null || _a === void 0 ? void 0 : _a.io) === null || _b === void 0 ? void 0 : _b.engine) === null || _c === void 0 ? void 0 : _c.transport) === null || _d === void 0 ? void 0 : _d.name;
     }
     getManager() {
         return this._socket.io;
@@ -2582,6 +2576,8 @@ class P2pRtcManager {
         this._webrtcProvider = webrtcProvider;
         this._features = features || {};
         this._isAudioOnlyMode = false;
+        this._closed = false;
+        this.skipEmittingServerMessageCount = 0;
         this.offerOptions = { offerToReceiveAudio: true, offerToReceiveVideo: true };
         this._pendingActionsForConnectedPeerConnections = [];
         this._audioTrackOnEnded = () => {
@@ -2663,6 +2659,10 @@ class P2pRtcManager {
         }
         return this._replaceTrackToPeerConnections(oldTrack, newTrack);
     }
+    close() {
+        this._closed = true;
+        this.disconnectAll();
+    }
     disconnectAll() {
         Object.keys(this.peerConnections).forEach((peerConnectionId) => {
             this.disconnect(peerConnectionId);
@@ -2848,8 +2848,18 @@ class P2pRtcManager {
             logger$6.error("Error during setting jitter buffer target:", error);
         }
     }
-    _emitServerEvent(eventName, data, callback) {
-        this._serverSocket.emit(eventName, data, callback);
+    _emitServerEvent(eventName, data) {
+        if (this._closed) {
+            logger$6.warn("RtcManager closed. Will not send event", eventName, data);
+            return;
+        }
+        if (this._features.awaitJoinRoomFinished && !this._serverSocket.joinRoomFinished) {
+            rtcStats.sendEvent("skip_emitting_server_message", { eventName });
+            this.skipEmittingServerMessageCount++;
+        }
+        else {
+            this._serverSocket.emit(eventName, data);
+        }
     }
     _emit(eventName, data) {
         this._emitter.emit(eventName, data);
@@ -5485,6 +5495,9 @@ class VegaRtcManager {
             });
         }
     }
+    close() {
+        this.disconnectAll();
+    }
     disconnectAll() {
         var _a, _b, _c, _d;
         this._reconnect = false;

package/package.json

@@ -1,7 +1,7 @@
 {
   "name": "@whereby.com/media",
   "description": "Media library for Whereby",
-  "version": "2.5.2",
+  "version": "2.5.3",
   "license": "MIT",
   "homepage": "https://github.com/whereby/sdk",
   "repository": {