@whereby.com/media 1.11.1 → 1.12.0

package/dist/index.cjs

@@ -3548,6 +3548,7 @@ class P2pRtcManager {
         this._isAudioOnlyMode = false;
         this.offerOptions = { offerToReceiveAudio: true, offerToReceiveVideo: true };
         this._pendingActionsForConnectedPeerConnections = [];
+        this._nodeJsClients = [];
         this._audioTrackOnEnded = () => {
             rtcStats.sendEvent("audio_ended", { unloading: unloading$1 });
             this._emit(rtcManagerEvents.MICROPHONE_STOPPED_WORKING, {});
@@ -3629,9 +3630,6 @@ class P2pRtcManager {
         }
         return this._replaceTrackToPeerConnections(oldTrack, newTrack);
     }
-    accept({ clientId, shouldAddLocalVideo }) {
-        return this.acceptNewStream({ streamId: clientId, clientId, shouldAddLocalVideo });
-    }
     disconnectAll() {
         Object.keys(this.peerConnections).forEach((peerConnectionId) => {
             this.disconnect(peerConnectionId);
@@ -3666,6 +3664,11 @@ class P2pRtcManager {
     setupSocketListeners() {
         this._socketListenerDeregisterFunctions = [
             () => this._clearMediaServersRefresh(),
+            this._serverSocket.on(PROTOCOL_RESPONSES.NEW_CLIENT, (data) => {
+                if (data.client.isDialIn && !this._nodeJsClients.includes(data.client.id)) {
+                    this._nodeJsClients.push(data.client.id);
+                }
+            }),
             this._serverSocket.on(PROTOCOL_RESPONSES.MEDIASERVER_CONFIG, (data) => {
                 if (data.error) {
                     logger$4.warn("FETCH_MEDIASERVER_CONFIG failed:", data.error);
@@ -3863,7 +3866,7 @@ class P2pRtcManager {
     _transformOutgoingSdp(original) {
         return { type: original.type, sdpU: original.sdp };
     }
-    _createSession({ clientId, initialBandwidth, isOfferer, peerConnectionId, shouldAddLocalVideo, }) {
+    _createSession({ clientId, initialBandwidth, isOfferer, peerConnectionId, shouldAddLocalVideo, enforceTurnProtocol, }) {
         if (!peerConnectionId) {
             throw new Error("peerConnectionId is missing");
         }
@@ -3899,13 +3902,13 @@ class P2pRtcManager {
                 return entry;
             });
         }
-        if (this._features.useOnlyTURN) {
+        if (enforceTurnProtocol || this._features.useOnlyTURN) {
             peerConnectionConfig.iceTransportPolicy = "relay";
             const filter = {
                 onlyudp: /^turn:.*transport=udp$/,
                 onlytcp: /^turn:.*transport=tcp$/,
                 onlytls: /^turns:.*transport=tcp$/,
-            }[this._features.useOnlyTURN];
+            }[enforceTurnProtocol || this._features.useOnlyTURN];
             if (filter) {
                 peerConnectionConfig.iceServers = peerConnectionConfig.iceServers.filter((entry) => entry.url && entry.url.match(filter));
             }
@@ -4157,17 +4160,26 @@ class P2pRtcManager {
     }
     _connect(clientId) {
         this.rtcStatsReconnect();
-        const shouldAddLocalVideo = true;
         let session = this._getSession(clientId);
-        let bandwidth = (session && session.bandwidth) || 0;
+        let initialBandwidth = (session && session.bandwidth) || 0;
         if (session) {
             logger$4.warn("Replacing peer session", clientId);
             this._cleanup(clientId);
         }
         else {
-            bandwidth = this._changeBandwidthForAllClients(true);
+            initialBandwidth = this._changeBandwidthForAllClients(true);
         }
-        session = this._createP2pSession(clientId, bandwidth, shouldAddLocalVideo, true);
+        let enforceTurnProtocol;
+        if (this._nodeJsClients.includes(clientId)) {
+            enforceTurnProtocol = this._features.isNodeSdk ? "onlyudp" : "onlytls";
+        }
+        session = this._createP2pSession({
+            clientId,
+            initialBandwidth,
+            shouldAddLocalVideo: true,
+            isOfferer: true,
+            enforceTurnProtocol,
+        });
         this._negotiatePeerConnection(clientId, session);
         return Promise.resolve(session);
     }
@@ -4338,13 +4350,14 @@ class P2pRtcManager {
         });
         return bandwidth;
     }
-    _createP2pSession(clientId, initialBandwidth, shouldAddLocalVideo, isOfferer) {
+    _createP2pSession({ clientId, initialBandwidth, shouldAddLocalVideo = false, isOfferer = false, enforceTurnProtocol, }) {
         const session = this._createSession({
             peerConnectionId: clientId,
             clientId,
             initialBandwidth,
             shouldAddLocalVideo,
             isOfferer,
+            enforceTurnProtocol,
         });
         const pc = session.pc;
         if (this._features.increaseIncomingMediaBufferOn) {
@@ -4458,20 +4471,26 @@ class P2pRtcManager {
         };
         return session;
     }
-    acceptNewStream({ streamId, clientId, shouldAddLocalVideo, }) {
+    acceptNewStream({ streamId, clientId, shouldAddLocalVideo, enforceTurnProtocol, }) {
         let session = this._getSession(clientId);
         if (session && streamId !== clientId) {
             return session;
         }
-        let bandwidth = (session && session.bandwidth) || 0;
+        let initialBandwidth = (session && session.bandwidth) || 0;
         if (session) {
             logger$4.warn("Replacing peer session", clientId);
             this._cleanup(clientId);
         }
         else {
-            bandwidth = this._changeBandwidthForAllClients(true);
+            initialBandwidth = this._changeBandwidthForAllClients(true);
         }
-        session = this._createP2pSession(clientId, bandwidth, shouldAddLocalVideo);
+        session = this._createP2pSession({
+            clientId,
+            initialBandwidth,
+            shouldAddLocalVideo: !!shouldAddLocalVideo,
+            enforceTurnProtocol,
+            isOfferer: false,
+        });
         this._emitServerEvent(RELAY_MESSAGES.READY_TO_RECEIVE_OFFER, {
             receiverId: clientId,
         });

package/dist/index.d.cts

@@ -309,140 +309,6 @@ 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;
-    _sfuServer: any;
-    _mediaserverConfigTtlSeconds: any;
-    _fetchMediaServersTimer: any;
-    _wasScreenSharing: any;
-    ipv6HostCandidateTeredoSeen: any;
-    ipv6HostCandidate6to4Seen: any;
-    mdnsHostCandidateSeen: any;
-    _lastReverseDirectionAttemptByClientId: 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;
-    maybeRestrictRelayBandwidth(session: any): void;
-    addNewStream(streamId: string, stream: MediaStream, audioPaused: boolean, videoPaused: boolean, beforeEffectTracks?: CustomMediaStreamTrack[]): void;
-    replaceTrack(oldTrack: CustomMediaStreamTrack | null, newTrack: CustomMediaStreamTrack): Promise<any[]>;
-    accept({ clientId, shouldAddLocalVideo }: {
-        clientId: string;
-        shouldAddLocalVideo?: boolean;
-    }): 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, 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: any, vp9On: any, av1On: any, redOn: any): void;
-    _negotiatePeerConnection(clientId: string, session: any, constraints?: any): void;
-    _withForcedRenegotiation(session: any, action: any): void;
-    _changeBandwidthForAllClients(isJoining: boolean): number;
-    _createP2pSession(clientId: string, initialBandwidth: any, shouldAddLocalVideo: any, isOfferer?: any): 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;
@@ -823,6 +689,7 @@ interface SignalRequests {
     };
     stop_recording: void;
 }
+type TurnTransportProtocol = "onlyudp" | "onlytcp" | "onlytls";
 
 declare function fromLocation({ host, protocol }?: {
     host?: string | undefined;
@@ -836,6 +703,145 @@ 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: any;
+    _webrtcProvider: any;
+    _features: any;
+    _isAudioOnlyMode: boolean;
+    offerOptions: {
+        offerToReceiveAudio: boolean;
+        offerToReceiveVideo: boolean;
+    };
+    _pendingActionsForConnectedPeerConnections: any[];
+    _audioTrackOnEnded: () => void;
+    _videoTrackOnEnded: () => void;
+    totalSessionsCreated: number;
+    _iceServers: any;
+    _sfuServer: any;
+    _mediaserverConfigTtlSeconds: any;
+    _fetchMediaServersTimer: any;
+    _wasScreenSharing: any;
+    ipv6HostCandidateTeredoSeen: any;
+    ipv6HostCandidate6to4Seen: any;
+    mdnsHostCandidateSeen: any;
+    _lastReverseDirectionAttemptByClientId: any;
+    _stoppedVideoTrack: any;
+    icePublicIPGatheringTimeoutID: any;
+    _videoTrackBeingMonitored?: CustomMediaStreamTrack;
+    _audioTrackBeingMonitored?: CustomMediaStreamTrack;
+    _nodeJsClients: string[];
+    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;
+    maybeRestrictRelayBandwidth(session: any): void;
+    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, enforceTurnProtocol, }: {
+        clientId: string;
+        initialBandwidth: any;
+        isOfferer: any;
+        peerConnectionId: string;
+        shouldAddLocalVideo: boolean;
+        enforceTurnProtocol?: TurnTransportProtocol;
+    }): 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, 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: any, vp9On: any, av1On: any, redOn: any): void;
+    _negotiatePeerConnection(clientId: string, session: any, constraints?: any): void;
+    _withForcedRenegotiation(session: any, action: any): void;
+    _changeBandwidthForAllClients(isJoining: boolean): number;
+    _createP2pSession({ clientId, initialBandwidth, shouldAddLocalVideo, isOfferer, enforceTurnProtocol, }: {
+        clientId: string;
+        initialBandwidth: number;
+        shouldAddLocalVideo: boolean;
+        isOfferer: boolean;
+        enforceTurnProtocol?: TurnTransportProtocol;
+    }): any;
+    acceptNewStream({ streamId, clientId, shouldAddLocalVideo, enforceTurnProtocol, }: {
+        streamId: string;
+        clientId: string;
+        shouldAddLocalVideo?: boolean;
+        enforceTurnProtocol?: TurnTransportProtocol;
+    }): 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;
@@ -1410,4 +1416,4 @@ declare class RtcStream {
     static getTypeFromId(id: string): string;
 }
 
-export { type AddSpotlightRequest, type AudioEnableRequest, type AudioEnableRequestedEvent, type AudioEnabledEvent, BandwidthTester, type ChatMessage, type ClientKickedEvent, type ClientLeftEvent, type ClientMetadataPayload, type ClientMetadataReceivedEvent, type ClientRole, type ClientUnableToJoinEvent, type CloudRecordingStartedEvent, type Credentials, type CustomMediaStreamTrack, EVENTS, type GetConstraintsOptions, type GetDeviceDataResult, type GetMediaConstraintsOptions, type GetStreamOptions, type GetStreamResult, type GetUpdatedDevicesResult, type IdentifyDeviceRequest, type JoinRoomRequest, KNOCK_MESSAGES, KalmanFilter, type KnockAcceptedEvent, type KnockRejectedEvent, type KnockRoomRequest, type KnockerLeftEvent, type LiveTranscriptionStartedEvent, type LiveTranscriptionStoppedEvent, Logger, MAXIMUM_TURN_BANDWIDTH, MAXIMUM_TURN_BANDWIDTH_UNLIMITED, MEDIA_JITTER_BUFFER_TARGET, type NewClientEvent, NoDevicesError, P2pRtcManager, PROTOCOL_ERRORS, PROTOCOL_EVENTS, PROTOCOL_REQUESTS, PROTOCOL_RESPONSES, RELAY_MESSAGES, ReconnectManager, type RemoveSpotlightRequest, type RoleName, type RoomJoinedEvent, type RoomKnockedEvent, type RoomLockedEvent, type RoomSessionEndedEvent, type RtcClientConnectionStatusChangedPayload, RtcEventNames, type RtcEvents, type RtcLocalStreamTrackAddedPayload, type RtcLocalStreamTrackRemovedPayload, type RtcManager, type RtcManagerCreatedPayload, RtcManagerDispatcher, RtcStream, type RtcStreamAddedPayload, STREAM_TYPES, type ScreenshareStartedEvent, type ScreenshareStoppedEvent, type SendClientMetadataRequest, ServerSocket, Session, SfuV2Parser, type SignalClient, type SignalEvents, type SignalKnocker, type SignalRequests, type SocketConf, type SocketManager, type Spotlight, type SpotlightAddedEvent, type SpotlightRemovedEvent, type StatsMonitorOptions, type StatsMonitorState, TYPES, VegaConnection, VegaMediaQualityMonitor, VegaRtcManager, type VideoEnabledEvent, addAbsCaptureTimeExtMap, addExtMap, assert, buildDeviceList, calculateStd, captureAudioSsrcMetrics, captureCandidatePairInfoMetrics, captureCommonSsrcMetrics, captureSsrcInfo, captureVideoSsrcMetrics, changeMediaDirection, compareLocalDevices, createACFCalculator, createMicAnalyser, createWorker, deprioritizeH264, detectMicrophoneNotWorking, enumerate, filterMidExtension, filterMsidSemantic, fromLocation, generateByteString, getConstraints, getCurrentPeerConnections, getDeviceData, getDisplayMedia, getHandler, getIssuesAndMetrics, getMediaConstraints, getMediaSettings, getOptimalBitrate, getPeerConnectionIndex, getStats, getStream, getStream2, getUpdatedDevices, getUpdatedStats, getUserMedia, hasGetDisplayMedia, ipRegex, isMobile, isRelayed, maybeRejectNoH264, maybeTurnOnly, modifyMediaCapabilities, removePeerConnection, replaceSSRCs, replaceTracksInStream, _default as rtcManagerEvents, rtcStats, setClientProvider, setCodecPreferenceSDP, setPeerConnectionsForTests, setVideoBandwidthUsingSetParameters, standardDeviation, startPerformanceMonitor, stopStreamTracks, subscribeIssues, subscribeStats, variance };
+export { type AddSpotlightRequest, type AudioEnableRequest, type AudioEnableRequestedEvent, type AudioEnabledEvent, BandwidthTester, type ChatMessage, type ClientKickedEvent, type ClientLeftEvent, type ClientMetadataPayload, type ClientMetadataReceivedEvent, type ClientRole, type ClientUnableToJoinEvent, type CloudRecordingStartedEvent, type Credentials, type CustomMediaStreamTrack, EVENTS, type GetConstraintsOptions, type GetDeviceDataResult, type GetMediaConstraintsOptions, type GetStreamOptions, type GetStreamResult, type GetUpdatedDevicesResult, type IdentifyDeviceRequest, type JoinRoomRequest, KNOCK_MESSAGES, KalmanFilter, type KnockAcceptedEvent, type KnockRejectedEvent, type KnockRoomRequest, type KnockerLeftEvent, type LiveTranscriptionStartedEvent, type LiveTranscriptionStoppedEvent, Logger, MAXIMUM_TURN_BANDWIDTH, MAXIMUM_TURN_BANDWIDTH_UNLIMITED, MEDIA_JITTER_BUFFER_TARGET, type NewClientEvent, NoDevicesError, P2pRtcManager, PROTOCOL_ERRORS, PROTOCOL_EVENTS, PROTOCOL_REQUESTS, PROTOCOL_RESPONSES, RELAY_MESSAGES, ReconnectManager, type RemoveSpotlightRequest, type RoleName, type RoomJoinedEvent, type RoomKnockedEvent, type RoomLockedEvent, type RoomSessionEndedEvent, type RtcClientConnectionStatusChangedPayload, RtcEventNames, type RtcEvents, type RtcLocalStreamTrackAddedPayload, type RtcLocalStreamTrackRemovedPayload, type RtcManager, type RtcManagerCreatedPayload, RtcManagerDispatcher, RtcStream, type RtcStreamAddedPayload, STREAM_TYPES, type ScreenshareStartedEvent, type ScreenshareStoppedEvent, type SendClientMetadataRequest, ServerSocket, Session, SfuV2Parser, type SignalClient, type SignalEvents, type SignalKnocker, type SignalRequests, type SocketConf, type SocketManager, type Spotlight, type SpotlightAddedEvent, type SpotlightRemovedEvent, type StatsMonitorOptions, type StatsMonitorState, TYPES, type TurnTransportProtocol, VegaConnection, VegaMediaQualityMonitor, VegaRtcManager, type VideoEnabledEvent, addAbsCaptureTimeExtMap, addExtMap, assert, buildDeviceList, calculateStd, captureAudioSsrcMetrics, captureCandidatePairInfoMetrics, captureCommonSsrcMetrics, captureSsrcInfo, captureVideoSsrcMetrics, changeMediaDirection, compareLocalDevices, createACFCalculator, createMicAnalyser, createWorker, deprioritizeH264, detectMicrophoneNotWorking, enumerate, filterMidExtension, filterMsidSemantic, fromLocation, generateByteString, getConstraints, getCurrentPeerConnections, getDeviceData, getDisplayMedia, getHandler, getIssuesAndMetrics, getMediaConstraints, getMediaSettings, getOptimalBitrate, getPeerConnectionIndex, getStats, getStream, getStream2, getUpdatedDevices, getUpdatedStats, getUserMedia, hasGetDisplayMedia, ipRegex, isMobile, isRelayed, maybeRejectNoH264, maybeTurnOnly, modifyMediaCapabilities, removePeerConnection, replaceSSRCs, replaceTracksInStream, _default as rtcManagerEvents, rtcStats, setClientProvider, setCodecPreferenceSDP, setPeerConnectionsForTests, setVideoBandwidthUsingSetParameters, standardDeviation, startPerformanceMonitor, stopStreamTracks, subscribeIssues, subscribeStats, variance };

package/dist/index.d.mts

@@ -309,140 +309,6 @@ 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;
-    _sfuServer: any;
-    _mediaserverConfigTtlSeconds: any;
-    _fetchMediaServersTimer: any;
-    _wasScreenSharing: any;
-    ipv6HostCandidateTeredoSeen: any;
-    ipv6HostCandidate6to4Seen: any;
-    mdnsHostCandidateSeen: any;
-    _lastReverseDirectionAttemptByClientId: 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;
-    maybeRestrictRelayBandwidth(session: any): void;
-    addNewStream(streamId: string, stream: MediaStream, audioPaused: boolean, videoPaused: boolean, beforeEffectTracks?: CustomMediaStreamTrack[]): void;
-    replaceTrack(oldTrack: CustomMediaStreamTrack | null, newTrack: CustomMediaStreamTrack): Promise<any[]>;
-    accept({ clientId, shouldAddLocalVideo }: {
-        clientId: string;
-        shouldAddLocalVideo?: boolean;
-    }): 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, 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: any, vp9On: any, av1On: any, redOn: any): void;
-    _negotiatePeerConnection(clientId: string, session: any, constraints?: any): void;
-    _withForcedRenegotiation(session: any, action: any): void;
-    _changeBandwidthForAllClients(isJoining: boolean): number;
-    _createP2pSession(clientId: string, initialBandwidth: any, shouldAddLocalVideo: any, isOfferer?: any): 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;
@@ -823,6 +689,7 @@ interface SignalRequests {
     };
     stop_recording: void;
 }
+type TurnTransportProtocol = "onlyudp" | "onlytcp" | "onlytls";
 
 declare function fromLocation({ host, protocol }?: {
     host?: string | undefined;
@@ -836,6 +703,145 @@ 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: any;
+    _webrtcProvider: any;
+    _features: any;
+    _isAudioOnlyMode: boolean;
+    offerOptions: {
+        offerToReceiveAudio: boolean;
+        offerToReceiveVideo: boolean;
+    };
+    _pendingActionsForConnectedPeerConnections: any[];
+    _audioTrackOnEnded: () => void;
+    _videoTrackOnEnded: () => void;
+    totalSessionsCreated: number;
+    _iceServers: any;
+    _sfuServer: any;
+    _mediaserverConfigTtlSeconds: any;
+    _fetchMediaServersTimer: any;
+    _wasScreenSharing: any;
+    ipv6HostCandidateTeredoSeen: any;
+    ipv6HostCandidate6to4Seen: any;
+    mdnsHostCandidateSeen: any;
+    _lastReverseDirectionAttemptByClientId: any;
+    _stoppedVideoTrack: any;
+    icePublicIPGatheringTimeoutID: any;
+    _videoTrackBeingMonitored?: CustomMediaStreamTrack;
+    _audioTrackBeingMonitored?: CustomMediaStreamTrack;
+    _nodeJsClients: string[];
+    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;
+    maybeRestrictRelayBandwidth(session: any): void;
+    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, enforceTurnProtocol, }: {
+        clientId: string;
+        initialBandwidth: any;
+        isOfferer: any;
+        peerConnectionId: string;
+        shouldAddLocalVideo: boolean;
+        enforceTurnProtocol?: TurnTransportProtocol;
+    }): 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, 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: any, vp9On: any, av1On: any, redOn: any): void;
+    _negotiatePeerConnection(clientId: string, session: any, constraints?: any): void;
+    _withForcedRenegotiation(session: any, action: any): void;
+    _changeBandwidthForAllClients(isJoining: boolean): number;
+    _createP2pSession({ clientId, initialBandwidth, shouldAddLocalVideo, isOfferer, enforceTurnProtocol, }: {
+        clientId: string;
+        initialBandwidth: number;
+        shouldAddLocalVideo: boolean;
+        isOfferer: boolean;
+        enforceTurnProtocol?: TurnTransportProtocol;
+    }): any;
+    acceptNewStream({ streamId, clientId, shouldAddLocalVideo, enforceTurnProtocol, }: {
+        streamId: string;
+        clientId: string;
+        shouldAddLocalVideo?: boolean;
+        enforceTurnProtocol?: TurnTransportProtocol;
+    }): 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;
@@ -1410,4 +1416,4 @@ declare class RtcStream {
     static getTypeFromId(id: string): string;
 }
 
-export { type AddSpotlightRequest, type AudioEnableRequest, type AudioEnableRequestedEvent, type AudioEnabledEvent, BandwidthTester, type ChatMessage, type ClientKickedEvent, type ClientLeftEvent, type ClientMetadataPayload, type ClientMetadataReceivedEvent, type ClientRole, type ClientUnableToJoinEvent, type CloudRecordingStartedEvent, type Credentials, type CustomMediaStreamTrack, EVENTS, type GetConstraintsOptions, type GetDeviceDataResult, type GetMediaConstraintsOptions, type GetStreamOptions, type GetStreamResult, type GetUpdatedDevicesResult, type IdentifyDeviceRequest, type JoinRoomRequest, KNOCK_MESSAGES, KalmanFilter, type KnockAcceptedEvent, type KnockRejectedEvent, type KnockRoomRequest, type KnockerLeftEvent, type LiveTranscriptionStartedEvent, type LiveTranscriptionStoppedEvent, Logger, MAXIMUM_TURN_BANDWIDTH, MAXIMUM_TURN_BANDWIDTH_UNLIMITED, MEDIA_JITTER_BUFFER_TARGET, type NewClientEvent, NoDevicesError, P2pRtcManager, PROTOCOL_ERRORS, PROTOCOL_EVENTS, PROTOCOL_REQUESTS, PROTOCOL_RESPONSES, RELAY_MESSAGES, ReconnectManager, type RemoveSpotlightRequest, type RoleName, type RoomJoinedEvent, type RoomKnockedEvent, type RoomLockedEvent, type RoomSessionEndedEvent, type RtcClientConnectionStatusChangedPayload, RtcEventNames, type RtcEvents, type RtcLocalStreamTrackAddedPayload, type RtcLocalStreamTrackRemovedPayload, type RtcManager, type RtcManagerCreatedPayload, RtcManagerDispatcher, RtcStream, type RtcStreamAddedPayload, STREAM_TYPES, type ScreenshareStartedEvent, type ScreenshareStoppedEvent, type SendClientMetadataRequest, ServerSocket, Session, SfuV2Parser, type SignalClient, type SignalEvents, type SignalKnocker, type SignalRequests, type SocketConf, type SocketManager, type Spotlight, type SpotlightAddedEvent, type SpotlightRemovedEvent, type StatsMonitorOptions, type StatsMonitorState, TYPES, VegaConnection, VegaMediaQualityMonitor, VegaRtcManager, type VideoEnabledEvent, addAbsCaptureTimeExtMap, addExtMap, assert, buildDeviceList, calculateStd, captureAudioSsrcMetrics, captureCandidatePairInfoMetrics, captureCommonSsrcMetrics, captureSsrcInfo, captureVideoSsrcMetrics, changeMediaDirection, compareLocalDevices, createACFCalculator, createMicAnalyser, createWorker, deprioritizeH264, detectMicrophoneNotWorking, enumerate, filterMidExtension, filterMsidSemantic, fromLocation, generateByteString, getConstraints, getCurrentPeerConnections, getDeviceData, getDisplayMedia, getHandler, getIssuesAndMetrics, getMediaConstraints, getMediaSettings, getOptimalBitrate, getPeerConnectionIndex, getStats, getStream, getStream2, getUpdatedDevices, getUpdatedStats, getUserMedia, hasGetDisplayMedia, ipRegex, isMobile, isRelayed, maybeRejectNoH264, maybeTurnOnly, modifyMediaCapabilities, removePeerConnection, replaceSSRCs, replaceTracksInStream, _default as rtcManagerEvents, rtcStats, setClientProvider, setCodecPreferenceSDP, setPeerConnectionsForTests, setVideoBandwidthUsingSetParameters, standardDeviation, startPerformanceMonitor, stopStreamTracks, subscribeIssues, subscribeStats, variance };
+export { type AddSpotlightRequest, type AudioEnableRequest, type AudioEnableRequestedEvent, type AudioEnabledEvent, BandwidthTester, type ChatMessage, type ClientKickedEvent, type ClientLeftEvent, type ClientMetadataPayload, type ClientMetadataReceivedEvent, type ClientRole, type ClientUnableToJoinEvent, type CloudRecordingStartedEvent, type Credentials, type CustomMediaStreamTrack, EVENTS, type GetConstraintsOptions, type GetDeviceDataResult, type GetMediaConstraintsOptions, type GetStreamOptions, type GetStreamResult, type GetUpdatedDevicesResult, type IdentifyDeviceRequest, type JoinRoomRequest, KNOCK_MESSAGES, KalmanFilter, type KnockAcceptedEvent, type KnockRejectedEvent, type KnockRoomRequest, type KnockerLeftEvent, type LiveTranscriptionStartedEvent, type LiveTranscriptionStoppedEvent, Logger, MAXIMUM_TURN_BANDWIDTH, MAXIMUM_TURN_BANDWIDTH_UNLIMITED, MEDIA_JITTER_BUFFER_TARGET, type NewClientEvent, NoDevicesError, P2pRtcManager, PROTOCOL_ERRORS, PROTOCOL_EVENTS, PROTOCOL_REQUESTS, PROTOCOL_RESPONSES, RELAY_MESSAGES, ReconnectManager, type RemoveSpotlightRequest, type RoleName, type RoomJoinedEvent, type RoomKnockedEvent, type RoomLockedEvent, type RoomSessionEndedEvent, type RtcClientConnectionStatusChangedPayload, RtcEventNames, type RtcEvents, type RtcLocalStreamTrackAddedPayload, type RtcLocalStreamTrackRemovedPayload, type RtcManager, type RtcManagerCreatedPayload, RtcManagerDispatcher, RtcStream, type RtcStreamAddedPayload, STREAM_TYPES, type ScreenshareStartedEvent, type ScreenshareStoppedEvent, type SendClientMetadataRequest, ServerSocket, Session, SfuV2Parser, type SignalClient, type SignalEvents, type SignalKnocker, type SignalRequests, type SocketConf, type SocketManager, type Spotlight, type SpotlightAddedEvent, type SpotlightRemovedEvent, type StatsMonitorOptions, type StatsMonitorState, TYPES, type TurnTransportProtocol, VegaConnection, VegaMediaQualityMonitor, VegaRtcManager, type VideoEnabledEvent, addAbsCaptureTimeExtMap, addExtMap, assert, buildDeviceList, calculateStd, captureAudioSsrcMetrics, captureCandidatePairInfoMetrics, captureCommonSsrcMetrics, captureSsrcInfo, captureVideoSsrcMetrics, changeMediaDirection, compareLocalDevices, createACFCalculator, createMicAnalyser, createWorker, deprioritizeH264, detectMicrophoneNotWorking, enumerate, filterMidExtension, filterMsidSemantic, fromLocation, generateByteString, getConstraints, getCurrentPeerConnections, getDeviceData, getDisplayMedia, getHandler, getIssuesAndMetrics, getMediaConstraints, getMediaSettings, getOptimalBitrate, getPeerConnectionIndex, getStats, getStream, getStream2, getUpdatedDevices, getUpdatedStats, getUserMedia, hasGetDisplayMedia, ipRegex, isMobile, isRelayed, maybeRejectNoH264, maybeTurnOnly, modifyMediaCapabilities, removePeerConnection, replaceSSRCs, replaceTracksInStream, _default as rtcManagerEvents, rtcStats, setClientProvider, setCodecPreferenceSDP, setPeerConnectionsForTests, setVideoBandwidthUsingSetParameters, standardDeviation, startPerformanceMonitor, stopStreamTracks, subscribeIssues, subscribeStats, variance };

package/dist/index.d.ts

@@ -309,140 +309,6 @@ 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;
-    _sfuServer: any;
-    _mediaserverConfigTtlSeconds: any;
-    _fetchMediaServersTimer: any;
-    _wasScreenSharing: any;
-    ipv6HostCandidateTeredoSeen: any;
-    ipv6HostCandidate6to4Seen: any;
-    mdnsHostCandidateSeen: any;
-    _lastReverseDirectionAttemptByClientId: 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;
-    maybeRestrictRelayBandwidth(session: any): void;
-    addNewStream(streamId: string, stream: MediaStream, audioPaused: boolean, videoPaused: boolean, beforeEffectTracks?: CustomMediaStreamTrack[]): void;
-    replaceTrack(oldTrack: CustomMediaStreamTrack | null, newTrack: CustomMediaStreamTrack): Promise<any[]>;
-    accept({ clientId, shouldAddLocalVideo }: {
-        clientId: string;
-        shouldAddLocalVideo?: boolean;
-    }): 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, 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: any, vp9On: any, av1On: any, redOn: any): void;
-    _negotiatePeerConnection(clientId: string, session: any, constraints?: any): void;
-    _withForcedRenegotiation(session: any, action: any): void;
-    _changeBandwidthForAllClients(isJoining: boolean): number;
-    _createP2pSession(clientId: string, initialBandwidth: any, shouldAddLocalVideo: any, isOfferer?: any): 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;
@@ -823,6 +689,7 @@ interface SignalRequests {
     };
     stop_recording: void;
 }
+type TurnTransportProtocol = "onlyudp" | "onlytcp" | "onlytls";
 
 declare function fromLocation({ host, protocol }?: {
     host?: string | undefined;
@@ -836,6 +703,145 @@ 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: any;
+    _webrtcProvider: any;
+    _features: any;
+    _isAudioOnlyMode: boolean;
+    offerOptions: {
+        offerToReceiveAudio: boolean;
+        offerToReceiveVideo: boolean;
+    };
+    _pendingActionsForConnectedPeerConnections: any[];
+    _audioTrackOnEnded: () => void;
+    _videoTrackOnEnded: () => void;
+    totalSessionsCreated: number;
+    _iceServers: any;
+    _sfuServer: any;
+    _mediaserverConfigTtlSeconds: any;
+    _fetchMediaServersTimer: any;
+    _wasScreenSharing: any;
+    ipv6HostCandidateTeredoSeen: any;
+    ipv6HostCandidate6to4Seen: any;
+    mdnsHostCandidateSeen: any;
+    _lastReverseDirectionAttemptByClientId: any;
+    _stoppedVideoTrack: any;
+    icePublicIPGatheringTimeoutID: any;
+    _videoTrackBeingMonitored?: CustomMediaStreamTrack;
+    _audioTrackBeingMonitored?: CustomMediaStreamTrack;
+    _nodeJsClients: string[];
+    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;
+    maybeRestrictRelayBandwidth(session: any): void;
+    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, enforceTurnProtocol, }: {
+        clientId: string;
+        initialBandwidth: any;
+        isOfferer: any;
+        peerConnectionId: string;
+        shouldAddLocalVideo: boolean;
+        enforceTurnProtocol?: TurnTransportProtocol;
+    }): 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, 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: any, vp9On: any, av1On: any, redOn: any): void;
+    _negotiatePeerConnection(clientId: string, session: any, constraints?: any): void;
+    _withForcedRenegotiation(session: any, action: any): void;
+    _changeBandwidthForAllClients(isJoining: boolean): number;
+    _createP2pSession({ clientId, initialBandwidth, shouldAddLocalVideo, isOfferer, enforceTurnProtocol, }: {
+        clientId: string;
+        initialBandwidth: number;
+        shouldAddLocalVideo: boolean;
+        isOfferer: boolean;
+        enforceTurnProtocol?: TurnTransportProtocol;
+    }): any;
+    acceptNewStream({ streamId, clientId, shouldAddLocalVideo, enforceTurnProtocol, }: {
+        streamId: string;
+        clientId: string;
+        shouldAddLocalVideo?: boolean;
+        enforceTurnProtocol?: TurnTransportProtocol;
+    }): 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;
@@ -1410,4 +1416,4 @@ declare class RtcStream {
     static getTypeFromId(id: string): string;
 }
 
-export { type AddSpotlightRequest, type AudioEnableRequest, type AudioEnableRequestedEvent, type AudioEnabledEvent, BandwidthTester, type ChatMessage, type ClientKickedEvent, type ClientLeftEvent, type ClientMetadataPayload, type ClientMetadataReceivedEvent, type ClientRole, type ClientUnableToJoinEvent, type CloudRecordingStartedEvent, type Credentials, type CustomMediaStreamTrack, EVENTS, type GetConstraintsOptions, type GetDeviceDataResult, type GetMediaConstraintsOptions, type GetStreamOptions, type GetStreamResult, type GetUpdatedDevicesResult, type IdentifyDeviceRequest, type JoinRoomRequest, KNOCK_MESSAGES, KalmanFilter, type KnockAcceptedEvent, type KnockRejectedEvent, type KnockRoomRequest, type KnockerLeftEvent, type LiveTranscriptionStartedEvent, type LiveTranscriptionStoppedEvent, Logger, MAXIMUM_TURN_BANDWIDTH, MAXIMUM_TURN_BANDWIDTH_UNLIMITED, MEDIA_JITTER_BUFFER_TARGET, type NewClientEvent, NoDevicesError, P2pRtcManager, PROTOCOL_ERRORS, PROTOCOL_EVENTS, PROTOCOL_REQUESTS, PROTOCOL_RESPONSES, RELAY_MESSAGES, ReconnectManager, type RemoveSpotlightRequest, type RoleName, type RoomJoinedEvent, type RoomKnockedEvent, type RoomLockedEvent, type RoomSessionEndedEvent, type RtcClientConnectionStatusChangedPayload, RtcEventNames, type RtcEvents, type RtcLocalStreamTrackAddedPayload, type RtcLocalStreamTrackRemovedPayload, type RtcManager, type RtcManagerCreatedPayload, RtcManagerDispatcher, RtcStream, type RtcStreamAddedPayload, STREAM_TYPES, type ScreenshareStartedEvent, type ScreenshareStoppedEvent, type SendClientMetadataRequest, ServerSocket, Session, SfuV2Parser, type SignalClient, type SignalEvents, type SignalKnocker, type SignalRequests, type SocketConf, type SocketManager, type Spotlight, type SpotlightAddedEvent, type SpotlightRemovedEvent, type StatsMonitorOptions, type StatsMonitorState, TYPES, VegaConnection, VegaMediaQualityMonitor, VegaRtcManager, type VideoEnabledEvent, addAbsCaptureTimeExtMap, addExtMap, assert, buildDeviceList, calculateStd, captureAudioSsrcMetrics, captureCandidatePairInfoMetrics, captureCommonSsrcMetrics, captureSsrcInfo, captureVideoSsrcMetrics, changeMediaDirection, compareLocalDevices, createACFCalculator, createMicAnalyser, createWorker, deprioritizeH264, detectMicrophoneNotWorking, enumerate, filterMidExtension, filterMsidSemantic, fromLocation, generateByteString, getConstraints, getCurrentPeerConnections, getDeviceData, getDisplayMedia, getHandler, getIssuesAndMetrics, getMediaConstraints, getMediaSettings, getOptimalBitrate, getPeerConnectionIndex, getStats, getStream, getStream2, getUpdatedDevices, getUpdatedStats, getUserMedia, hasGetDisplayMedia, ipRegex, isMobile, isRelayed, maybeRejectNoH264, maybeTurnOnly, modifyMediaCapabilities, removePeerConnection, replaceSSRCs, replaceTracksInStream, _default as rtcManagerEvents, rtcStats, setClientProvider, setCodecPreferenceSDP, setPeerConnectionsForTests, setVideoBandwidthUsingSetParameters, standardDeviation, startPerformanceMonitor, stopStreamTracks, subscribeIssues, subscribeStats, variance };
+export { type AddSpotlightRequest, type AudioEnableRequest, type AudioEnableRequestedEvent, type AudioEnabledEvent, BandwidthTester, type ChatMessage, type ClientKickedEvent, type ClientLeftEvent, type ClientMetadataPayload, type ClientMetadataReceivedEvent, type ClientRole, type ClientUnableToJoinEvent, type CloudRecordingStartedEvent, type Credentials, type CustomMediaStreamTrack, EVENTS, type GetConstraintsOptions, type GetDeviceDataResult, type GetMediaConstraintsOptions, type GetStreamOptions, type GetStreamResult, type GetUpdatedDevicesResult, type IdentifyDeviceRequest, type JoinRoomRequest, KNOCK_MESSAGES, KalmanFilter, type KnockAcceptedEvent, type KnockRejectedEvent, type KnockRoomRequest, type KnockerLeftEvent, type LiveTranscriptionStartedEvent, type LiveTranscriptionStoppedEvent, Logger, MAXIMUM_TURN_BANDWIDTH, MAXIMUM_TURN_BANDWIDTH_UNLIMITED, MEDIA_JITTER_BUFFER_TARGET, type NewClientEvent, NoDevicesError, P2pRtcManager, PROTOCOL_ERRORS, PROTOCOL_EVENTS, PROTOCOL_REQUESTS, PROTOCOL_RESPONSES, RELAY_MESSAGES, ReconnectManager, type RemoveSpotlightRequest, type RoleName, type RoomJoinedEvent, type RoomKnockedEvent, type RoomLockedEvent, type RoomSessionEndedEvent, type RtcClientConnectionStatusChangedPayload, RtcEventNames, type RtcEvents, type RtcLocalStreamTrackAddedPayload, type RtcLocalStreamTrackRemovedPayload, type RtcManager, type RtcManagerCreatedPayload, RtcManagerDispatcher, RtcStream, type RtcStreamAddedPayload, STREAM_TYPES, type ScreenshareStartedEvent, type ScreenshareStoppedEvent, type SendClientMetadataRequest, ServerSocket, Session, SfuV2Parser, type SignalClient, type SignalEvents, type SignalKnocker, type SignalRequests, type SocketConf, type SocketManager, type Spotlight, type SpotlightAddedEvent, type SpotlightRemovedEvent, type StatsMonitorOptions, type StatsMonitorState, TYPES, type TurnTransportProtocol, VegaConnection, VegaMediaQualityMonitor, VegaRtcManager, type VideoEnabledEvent, addAbsCaptureTimeExtMap, addExtMap, assert, buildDeviceList, calculateStd, captureAudioSsrcMetrics, captureCandidatePairInfoMetrics, captureCommonSsrcMetrics, captureSsrcInfo, captureVideoSsrcMetrics, changeMediaDirection, compareLocalDevices, createACFCalculator, createMicAnalyser, createWorker, deprioritizeH264, detectMicrophoneNotWorking, enumerate, filterMidExtension, filterMsidSemantic, fromLocation, generateByteString, getConstraints, getCurrentPeerConnections, getDeviceData, getDisplayMedia, getHandler, getIssuesAndMetrics, getMediaConstraints, getMediaSettings, getOptimalBitrate, getPeerConnectionIndex, getStats, getStream, getStream2, getUpdatedDevices, getUpdatedStats, getUserMedia, hasGetDisplayMedia, ipRegex, isMobile, isRelayed, maybeRejectNoH264, maybeTurnOnly, modifyMediaCapabilities, removePeerConnection, replaceSSRCs, replaceTracksInStream, _default as rtcManagerEvents, rtcStats, setClientProvider, setCodecPreferenceSDP, setPeerConnectionsForTests, setVideoBandwidthUsingSetParameters, standardDeviation, startPerformanceMonitor, stopStreamTracks, subscribeIssues, subscribeStats, variance };

package/dist/index.mjs

@@ -3527,6 +3527,7 @@ class P2pRtcManager {
         this._isAudioOnlyMode = false;
         this.offerOptions = { offerToReceiveAudio: true, offerToReceiveVideo: true };
         this._pendingActionsForConnectedPeerConnections = [];
+        this._nodeJsClients = [];
         this._audioTrackOnEnded = () => {
             rtcStats.sendEvent("audio_ended", { unloading: unloading$1 });
             this._emit(rtcManagerEvents.MICROPHONE_STOPPED_WORKING, {});
@@ -3608,9 +3609,6 @@ class P2pRtcManager {
         }
         return this._replaceTrackToPeerConnections(oldTrack, newTrack);
     }
-    accept({ clientId, shouldAddLocalVideo }) {
-        return this.acceptNewStream({ streamId: clientId, clientId, shouldAddLocalVideo });
-    }
     disconnectAll() {
         Object.keys(this.peerConnections).forEach((peerConnectionId) => {
             this.disconnect(peerConnectionId);
@@ -3645,6 +3643,11 @@ class P2pRtcManager {
     setupSocketListeners() {
         this._socketListenerDeregisterFunctions = [
             () => this._clearMediaServersRefresh(),
+            this._serverSocket.on(PROTOCOL_RESPONSES.NEW_CLIENT, (data) => {
+                if (data.client.isDialIn && !this._nodeJsClients.includes(data.client.id)) {
+                    this._nodeJsClients.push(data.client.id);
+                }
+            }),
             this._serverSocket.on(PROTOCOL_RESPONSES.MEDIASERVER_CONFIG, (data) => {
                 if (data.error) {
                     logger$4.warn("FETCH_MEDIASERVER_CONFIG failed:", data.error);
@@ -3842,7 +3845,7 @@ class P2pRtcManager {
     _transformOutgoingSdp(original) {
         return { type: original.type, sdpU: original.sdp };
     }
-    _createSession({ clientId, initialBandwidth, isOfferer, peerConnectionId, shouldAddLocalVideo, }) {
+    _createSession({ clientId, initialBandwidth, isOfferer, peerConnectionId, shouldAddLocalVideo, enforceTurnProtocol, }) {
         if (!peerConnectionId) {
             throw new Error("peerConnectionId is missing");
         }
@@ -3878,13 +3881,13 @@ class P2pRtcManager {
                 return entry;
             });
         }
-        if (this._features.useOnlyTURN) {
+        if (enforceTurnProtocol || this._features.useOnlyTURN) {
             peerConnectionConfig.iceTransportPolicy = "relay";
             const filter = {
                 onlyudp: /^turn:.*transport=udp$/,
                 onlytcp: /^turn:.*transport=tcp$/,
                 onlytls: /^turns:.*transport=tcp$/,
-            }[this._features.useOnlyTURN];
+            }[enforceTurnProtocol || this._features.useOnlyTURN];
             if (filter) {
                 peerConnectionConfig.iceServers = peerConnectionConfig.iceServers.filter((entry) => entry.url && entry.url.match(filter));
             }
@@ -4136,17 +4139,26 @@ class P2pRtcManager {
     }
     _connect(clientId) {
         this.rtcStatsReconnect();
-        const shouldAddLocalVideo = true;
         let session = this._getSession(clientId);
-        let bandwidth = (session && session.bandwidth) || 0;
+        let initialBandwidth = (session && session.bandwidth) || 0;
         if (session) {
             logger$4.warn("Replacing peer session", clientId);
             this._cleanup(clientId);
         }
         else {
-            bandwidth = this._changeBandwidthForAllClients(true);
+            initialBandwidth = this._changeBandwidthForAllClients(true);
         }
-        session = this._createP2pSession(clientId, bandwidth, shouldAddLocalVideo, true);
+        let enforceTurnProtocol;
+        if (this._nodeJsClients.includes(clientId)) {
+            enforceTurnProtocol = this._features.isNodeSdk ? "onlyudp" : "onlytls";
+        }
+        session = this._createP2pSession({
+            clientId,
+            initialBandwidth,
+            shouldAddLocalVideo: true,
+            isOfferer: true,
+            enforceTurnProtocol,
+        });
         this._negotiatePeerConnection(clientId, session);
         return Promise.resolve(session);
     }
@@ -4317,13 +4329,14 @@ class P2pRtcManager {
         });
         return bandwidth;
     }
-    _createP2pSession(clientId, initialBandwidth, shouldAddLocalVideo, isOfferer) {
+    _createP2pSession({ clientId, initialBandwidth, shouldAddLocalVideo = false, isOfferer = false, enforceTurnProtocol, }) {
         const session = this._createSession({
             peerConnectionId: clientId,
             clientId,
             initialBandwidth,
             shouldAddLocalVideo,
             isOfferer,
+            enforceTurnProtocol,
         });
         const pc = session.pc;
         if (this._features.increaseIncomingMediaBufferOn) {
@@ -4437,20 +4450,26 @@ class P2pRtcManager {
         };
         return session;
     }
-    acceptNewStream({ streamId, clientId, shouldAddLocalVideo, }) {
+    acceptNewStream({ streamId, clientId, shouldAddLocalVideo, enforceTurnProtocol, }) {
         let session = this._getSession(clientId);
         if (session && streamId !== clientId) {
             return session;
         }
-        let bandwidth = (session && session.bandwidth) || 0;
+        let initialBandwidth = (session && session.bandwidth) || 0;
         if (session) {
             logger$4.warn("Replacing peer session", clientId);
             this._cleanup(clientId);
         }
         else {
-            bandwidth = this._changeBandwidthForAllClients(true);
+            initialBandwidth = this._changeBandwidthForAllClients(true);
         }
-        session = this._createP2pSession(clientId, bandwidth, shouldAddLocalVideo);
+        session = this._createP2pSession({
+            clientId,
+            initialBandwidth,
+            shouldAddLocalVideo: !!shouldAddLocalVideo,
+            enforceTurnProtocol,
+            isOfferer: false,
+        });
         this._emitServerEvent(RELAY_MESSAGES.READY_TO_RECEIVE_OFFER, {
             receiverId: clientId,
         });

package/dist/legacy-esm.js

@@ -3527,6 +3527,7 @@ class P2pRtcManager {
         this._isAudioOnlyMode = false;
         this.offerOptions = { offerToReceiveAudio: true, offerToReceiveVideo: true };
         this._pendingActionsForConnectedPeerConnections = [];
+        this._nodeJsClients = [];
         this._audioTrackOnEnded = () => {
             rtcStats.sendEvent("audio_ended", { unloading: unloading$1 });
             this._emit(rtcManagerEvents.MICROPHONE_STOPPED_WORKING, {});
@@ -3608,9 +3609,6 @@ class P2pRtcManager {
         }
         return this._replaceTrackToPeerConnections(oldTrack, newTrack);
     }
-    accept({ clientId, shouldAddLocalVideo }) {
-        return this.acceptNewStream({ streamId: clientId, clientId, shouldAddLocalVideo });
-    }
     disconnectAll() {
         Object.keys(this.peerConnections).forEach((peerConnectionId) => {
             this.disconnect(peerConnectionId);
@@ -3645,6 +3643,11 @@ class P2pRtcManager {
     setupSocketListeners() {
         this._socketListenerDeregisterFunctions = [
             () => this._clearMediaServersRefresh(),
+            this._serverSocket.on(PROTOCOL_RESPONSES.NEW_CLIENT, (data) => {
+                if (data.client.isDialIn && !this._nodeJsClients.includes(data.client.id)) {
+                    this._nodeJsClients.push(data.client.id);
+                }
+            }),
             this._serverSocket.on(PROTOCOL_RESPONSES.MEDIASERVER_CONFIG, (data) => {
                 if (data.error) {
                     logger$4.warn("FETCH_MEDIASERVER_CONFIG failed:", data.error);
@@ -3842,7 +3845,7 @@ class P2pRtcManager {
     _transformOutgoingSdp(original) {
         return { type: original.type, sdpU: original.sdp };
     }
-    _createSession({ clientId, initialBandwidth, isOfferer, peerConnectionId, shouldAddLocalVideo, }) {
+    _createSession({ clientId, initialBandwidth, isOfferer, peerConnectionId, shouldAddLocalVideo, enforceTurnProtocol, }) {
         if (!peerConnectionId) {
             throw new Error("peerConnectionId is missing");
         }
@@ -3878,13 +3881,13 @@ class P2pRtcManager {
                 return entry;
             });
         }
-        if (this._features.useOnlyTURN) {
+        if (enforceTurnProtocol || this._features.useOnlyTURN) {
             peerConnectionConfig.iceTransportPolicy = "relay";
             const filter = {
                 onlyudp: /^turn:.*transport=udp$/,
                 onlytcp: /^turn:.*transport=tcp$/,
                 onlytls: /^turns:.*transport=tcp$/,
-            }[this._features.useOnlyTURN];
+            }[enforceTurnProtocol || this._features.useOnlyTURN];
             if (filter) {
                 peerConnectionConfig.iceServers = peerConnectionConfig.iceServers.filter((entry) => entry.url && entry.url.match(filter));
             }
@@ -4136,17 +4139,26 @@ class P2pRtcManager {
     }
     _connect(clientId) {
         this.rtcStatsReconnect();
-        const shouldAddLocalVideo = true;
         let session = this._getSession(clientId);
-        let bandwidth = (session && session.bandwidth) || 0;
+        let initialBandwidth = (session && session.bandwidth) || 0;
         if (session) {
             logger$4.warn("Replacing peer session", clientId);
             this._cleanup(clientId);
         }
         else {
-            bandwidth = this._changeBandwidthForAllClients(true);
+            initialBandwidth = this._changeBandwidthForAllClients(true);
         }
-        session = this._createP2pSession(clientId, bandwidth, shouldAddLocalVideo, true);
+        let enforceTurnProtocol;
+        if (this._nodeJsClients.includes(clientId)) {
+            enforceTurnProtocol = this._features.isNodeSdk ? "onlyudp" : "onlytls";
+        }
+        session = this._createP2pSession({
+            clientId,
+            initialBandwidth,
+            shouldAddLocalVideo: true,
+            isOfferer: true,
+            enforceTurnProtocol,
+        });
         this._negotiatePeerConnection(clientId, session);
         return Promise.resolve(session);
     }
@@ -4317,13 +4329,14 @@ class P2pRtcManager {
         });
         return bandwidth;
     }
-    _createP2pSession(clientId, initialBandwidth, shouldAddLocalVideo, isOfferer) {
+    _createP2pSession({ clientId, initialBandwidth, shouldAddLocalVideo = false, isOfferer = false, enforceTurnProtocol, }) {
         const session = this._createSession({
             peerConnectionId: clientId,
             clientId,
             initialBandwidth,
             shouldAddLocalVideo,
             isOfferer,
+            enforceTurnProtocol,
         });
         const pc = session.pc;
         if (this._features.increaseIncomingMediaBufferOn) {
@@ -4437,20 +4450,26 @@ class P2pRtcManager {
         };
         return session;
     }
-    acceptNewStream({ streamId, clientId, shouldAddLocalVideo, }) {
+    acceptNewStream({ streamId, clientId, shouldAddLocalVideo, enforceTurnProtocol, }) {
         let session = this._getSession(clientId);
         if (session && streamId !== clientId) {
             return session;
         }
-        let bandwidth = (session && session.bandwidth) || 0;
+        let initialBandwidth = (session && session.bandwidth) || 0;
         if (session) {
             logger$4.warn("Replacing peer session", clientId);
             this._cleanup(clientId);
         }
         else {
-            bandwidth = this._changeBandwidthForAllClients(true);
+            initialBandwidth = this._changeBandwidthForAllClients(true);
         }
-        session = this._createP2pSession(clientId, bandwidth, shouldAddLocalVideo);
+        session = this._createP2pSession({
+            clientId,
+            initialBandwidth,
+            shouldAddLocalVideo: !!shouldAddLocalVideo,
+            enforceTurnProtocol,
+            isOfferer: false,
+        });
         this._emitServerEvent(RELAY_MESSAGES.READY_TO_RECEIVE_OFFER, {
             receiverId: clientId,
         });

package/package.json

@@ -1,7 +1,7 @@
 {
     "name": "@whereby.com/media",
     "description": "Media library for Whereby",
-    "version": "1.11.1",
+    "version": "1.12.0",
     "license": "MIT",
     "homepage": "https://github.com/whereby/sdk",
     "repository": {
@@ -12,6 +12,7 @@
         "url": "https://github.com/whereby/sdk/issues"
     },
     "scripts": {
+        "clean": "rimraf dist node_modules .turbo",
         "build": "rimraf dist && rollup -c rollup.config.js",
         "test": "npm run test:lint && npm run test:unit",
         "test:unit": "jest",