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mediasoup

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Cutting Edge WebRTC Video Conferencing

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import { EnhancedEventEmitter } from './enhancedEvents'; import type { Channel } from './Channel'; import type { Router, PipeToRouterOptions, PipeToRouterResult, PipeTransportPair, RouterDump, RouterEvents, RouterObserver } from './RouterTypes'; import type { Transport } from './TransportTypes'; import type { WebRtcTransport, WebRtcTransportOptions } from './WebRtcTransportTypes'; import type { PlainTransport, PlainTransportOptions } from './PlainTransportTypes'; import type { PipeTransport, PipeTransportOptions } from './PipeTransportTypes'; import type { DirectTransport, DirectTransportOptions } from './DirectTransportTypes'; import type { ActiveSpeakerObserver, ActiveSpeakerObserverOptions } from './ActiveSpeakerObserverTypes'; import type { AudioLevelObserver, AudioLevelObserverOptions } from './AudioLevelObserverTypes'; import type { RtpCapabilities, RouterRtpCodecCapability } from './rtpParametersTypes'; import type { AppData } from './types'; export type RouterInternal = { routerId: string; }; type RouterData = { rtpCapabilities: RtpCapabilities; }; export declare class RouterImpl<RouterAppData extends AppData = AppData> extends EnhancedEventEmitter<RouterEvents> implements Router { #private; constructor({ internal, data, channel, appData, }: { internal: RouterInternal; data: RouterData; channel: Channel; appData?: RouterAppData; }); get id(): string; get closed(): boolean; get rtpCapabilities(): RtpCapabilities; get appData(): RouterAppData; set appData(appData: RouterAppData); get observer(): RouterObserver; /** * Just for testing purposes. * * @private */ get transportsForTesting(): Map<string, Transport>; close(): void; workerClosed(): void; dump(): Promise<RouterDump>; createWebRtcTransport<WebRtcTransportAppData extends AppData = AppData>({ webRtcServer, listenInfos, listenIps, port, enableUdp, enableTcp, preferUdp, preferTcp, initialAvailableOutgoingBitrate, enableSctp, numSctpStreams, maxSctpMessageSize, sctpSendBufferSize, iceConsentTimeout, appData, }: WebRtcTransportOptions<WebRtcTransportAppData>): Promise<WebRtcTransport<WebRtcTransportAppData>>; createPlainTransport<PlainTransportAppData extends AppData = AppData>({ listenInfo, rtcpListenInfo, listenIp, port, rtcpMux, comedia, enableSctp, numSctpStreams, maxSctpMessageSize, sctpSendBufferSize, enableSrtp, srtpCryptoSuite, appData, }: PlainTransportOptions<PlainTransportAppData>): Promise<PlainTransport<PlainTransportAppData>>; createPipeTransport<PipeTransportAppData extends AppData = AppData>({ listenInfo, listenIp, port, enableSctp, numSctpStreams, maxSctpMessageSize, sctpSendBufferSize, enableRtx, enableSrtp, appData, }: PipeTransportOptions<PipeTransportAppData>): Promise<PipeTransport<PipeTransportAppData>>; createDirectTransport<DirectTransportAppData extends AppData = AppData>({ maxMessageSize, appData, }?: DirectTransportOptions<DirectTransportAppData>): Promise<DirectTransport<DirectTransportAppData>>; pipeToRouter({ producerId, dataProducerId, router, keepId, listenInfo, listenIp, enableSctp, numSctpStreams, enableRtx, enableSrtp, }: PipeToRouterOptions): Promise<PipeToRouterResult>; addPipeTransportPair(pipeTransportPairKey: string, pipeTransportPairPromise: Promise<PipeTransportPair>): void; createActiveSpeakerObserver<ActiveSpeakerObserverAppData extends AppData = AppData>({ interval, appData, }?: ActiveSpeakerObserverOptions<ActiveSpeakerObserverAppData>): Promise<ActiveSpeakerObserver<ActiveSpeakerObserverAppData>>; createAudioLevelObserver<AudioLevelObserverAppData extends AppData = AppData>({ maxEntries, threshold, interval, appData, }?: AudioLevelObserverOptions<AudioLevelObserverAppData>): Promise<AudioLevelObserver<AudioLevelObserverAppData>>; canConsume({ producerId, rtpCapabilities, }: { producerId: string; rtpCapabilities: RtpCapabilities; }): boolean; updateMediaCodecs(mediaCodecs: RouterRtpCodecCapability[]): void; private handleListenerError; } export {}; //# sourceMappingURL=Router.d.ts.map