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mediasoup

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Cutting Edge WebRTC Video Conferencing

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import type { EnhancedEventEmitter } from './enhancedEvents'; import type { Transport, TransportListenInfo, TransportListenIp, TransportTuple, SctpState, BaseTransportDump, BaseTransportStats, TransportEvents, TransportObserverEvents } from './TransportTypes'; import type { SrtpParameters, SrtpCryptoSuite } from './srtpParametersTypes'; import type { SctpParameters, NumSctpStreams } from './sctpParametersTypes'; import type { Either, AppData } from './types'; export type PlainTransportOptions<PlainTransportAppData extends AppData = AppData> = { /** * Use RTCP-mux (RTP and RTCP in the same port). Default true. */ rtcpMux?: boolean; /** * Whether remote IP:port should be auto-detected based on first RTP/RTCP * packet received. If enabled, connect() method must not be called unless * SRTP is enabled. If so, it must be called with just remote SRTP parameters. * Default false. */ comedia?: boolean; /** * Create a SCTP association. Default false. */ enableSctp?: boolean; /** * SCTP streams number. */ numSctpStreams?: NumSctpStreams; /** * Maximum allowed size for SCTP messages sent by DataProducers. * Default 262144. */ maxSctpMessageSize?: number; /** * Maximum SCTP send buffer used by DataConsumers. * Default 262144. */ sctpSendBufferSize?: number; /** * Enable SRTP. For this to work, connect() must be called * with remote SRTP parameters. Default false. */ enableSrtp?: boolean; /** * The SRTP crypto suite to be used if enableSrtp is set. Default * 'AES_CM_128_HMAC_SHA1_80'. */ srtpCryptoSuite?: SrtpCryptoSuite; /** * Custom application data. */ appData?: PlainTransportAppData; } & PlainTransportListen; type PlainTransportListen = Either<PlainTransportListenInfo, PlainTransportListenIp>; type PlainTransportListenInfo = { /** * Listening info. */ listenInfo: TransportListenInfo; /** * Optional listening info for RTCP. */ rtcpListenInfo?: TransportListenInfo; }; type PlainTransportListenIp = { /** * Listening IP address. */ listenIp: TransportListenIp | string; /** * Fixed port to listen on instead of selecting automatically from Worker's * port range. */ port?: number; }; export type PlainTransportDump = BaseTransportDump & { rtcpMux: boolean; comedia: boolean; tuple: TransportTuple; rtcpTuple?: TransportTuple; srtpParameters?: SrtpParameters; }; export type PlainTransportStat = BaseTransportStats & { type: string; rtcpMux: boolean; comedia: boolean; tuple: TransportTuple; rtcpTuple?: TransportTuple; }; export type PlainTransportEvents = TransportEvents & { tuple: [TransportTuple]; rtcptuple: [TransportTuple]; sctpstatechange: [SctpState]; }; export type PlainTransportObserver = EnhancedEventEmitter<PlainTransportObserverEvents>; export type PlainTransportObserverEvents = TransportObserverEvents & { tuple: [TransportTuple]; rtcptuple: [TransportTuple]; sctpstatechange: [SctpState]; }; export interface PlainTransport<PlainTransportAppData extends AppData = AppData> extends Transport<PlainTransportAppData, PlainTransportEvents, PlainTransportObserver> { /** * Transport type. * * @override */ get type(): 'plain'; /** * Observer. * * @override */ get observer(): PlainTransportObserver; /** * PlainTransport tuple. */ get tuple(): TransportTuple; /** * PlainTransport RTCP tuple. */ get rtcpTuple(): TransportTuple | undefined; /** * SCTP parameters. */ get sctpParameters(): SctpParameters | undefined; /** * SCTP state. */ get sctpState(): SctpState | undefined; /** * SRTP parameters. */ get srtpParameters(): SrtpParameters | undefined; /** * Dump PlainTransport. * * @override */ dump(): Promise<PlainTransportDump>; /** * Get PlainTransport stats. * * @override */ getStats(): Promise<PlainTransportStat[]>; /** * Provide the PlainTransport remote parameters. * * @override */ connect({ ip, port, rtcpPort, srtpParameters, }: { ip?: string; port?: number; rtcpPort?: number; srtpParameters?: SrtpParameters; }): Promise<void>; } export {}; //# sourceMappingURL=PlainTransportTypes.d.ts.map