试用MEDIASOUP：一款开源流媒体工具

import express from "express";import http from "http";import { Server } from "socket.io";import cors from "cors";import mediasoup from "mediasoup";
const app = express();const port = 4000;const server = http.createServer(app);
// Initialize a Socket.IO server for WebSocket connectionsconst io = new Server(server, {  cors: {    origin: "*",    credentials: true,  },});
// Create a namespace "/mediasoup" for mediasoup-related socket eventsconst peers = io.of("/mediasoup");

let worker: mediasoup.types.Worker<mediasoup.types.AppData>;let router: mediasoup.types.Router<mediasoup.types.AppData>;let producerTransport:  | mediasoup.types.WebRtcTransport<mediasoup.types.AppData>  | undefined;let consumerTransport:  | mediasoup.types.WebRtcTransport<mediasoup.types.AppData>  | undefined;let producer: mediasoup.types.Producer<mediasoup.types.AppData> | undefined;let consumer: mediasoup.types.Consumer<mediasoup.types.AppData> | undefined;
const createWorker = async (): Promise<  mediasoup.types.Worker<mediasoup.types.AppData>  // Create a mediasoup worker with specific configuration    const newWorker = await mediasoup.createWorker({    rtcMinPort: 2000, // Minimum port number for RTC traffic    rtcMaxPort: 2020, // Maximum port number for RTC traffic  });
// Log the worker process ID for reference  console.log(`Worker process ID ${newWorker.pid}`);
// Event handler for the 'died' event on the workernewWorker.on("died", (error) => {    console.error("mediasoup worker has died");    // Gracefully shut down the process to allow for recovery or troubleshooting.    setTimeout(() => {      process.exit();    }, 2000);  });
  return newWorker;};

/** * The media codecs configuration array. * Each object in this array provides configuration for a specific audio or video codec. */const mediaCodecs: mediasoup.types.RtpCodecCapability[] = [  {    /** Indicates this is an audio codec configuration */    kind: "audio",    /**     * Specifies the MIME type for the Opus codec, known for good audio quality at various bit rates.     * Format: <type>/<subtype>, e.g., audio/opus     */    mimeType: "audio/opus",    /**     * Specifies the number of audio samples processed per second (48,000 samples per second for high-quality audio).     * Higher values generally allow better audio quality.     */    clockRate: 48000,    /** Specifies the number of audio channels (2 for stereo audio). */    channels: 2,    /**     * Optional: Specifies a preferred payload type number for the codec.     * Helps ensure consistency in payload type numbering across different sessions or applications.     */    preferredPayloadType: 96, // Example value    /**     * Optional: Specifies a list of RTCP feedback mechanisms supported by the codec.     * Helps optimize codec behavior in response to network conditions.     */    rtcpFeedback: [      // Example values      { type: "nack" },      { type: "nack", parameter: "pli" },    ],  },  {    /** Indicates this is a video codec configuration */    kind: "video",    /** Specifies the MIME type for the VP8 codec, commonly used for video compression. */    mimeType: "video/VP8",    /** Specifies the clock rate, or the number of timing ticks per second (commonly 90,000 for video). */    clockRate: 90000,    /**     * Optional: Specifies codec-specific parameters.     * In this case, sets the starting bitrate for the codec.     */    parameters: {      "x-google-start-bitrate": 1000,    },    preferredPayloadType: 97, // Example value    rtcpFeedback: [      // Example values      { type: "nack" },      { type: "ccm", parameter: "fir" },      { type: "goog-remb" },    ],  },];

// Event handler for new peer connectionspeers.on("connection", async (socket) => { // ... Peer connection event handling ...
 /**  * Create a router.  * Since there are only two peers in this application, we only need one router for each peer.  * In a more complex application, you may need multiple routers to handle multiple peers.  * A router is required to route media to/from this peer.  */ router = await worker.createRouter({   mediaCodecs: mediaCodecs, });
 /**  * Event handler for fetching router RTP capabilities.  * RTP capabilities are required for configuring transports and producers/consumers.  * This function is called when a peer requests the router RTP capabilities.  * The callback function is used to send the router RTP capabilities to the peer.  */ socket.on("getRouterRtpCapabilities", (callback) => {   // ... Handling router RTP capabilities ... });
/**  * Event handler for creating a transport.  * A transport is required for sending or producing media.  * The callback function is used to send the transport parameters to the peer.  * @param {boolean} data.sender - Indicates whether the transport is for sending or receiving media.  * @param {function} callback - A callback function to handle the result of the transport creation.  */ socket.on("createTransport", async ({ sender }, callback) => {   // ... Creating sender/receiver transports ... });

 /**  * Event handler for producing media.  * This function sets up a producer for sending media to the peer.  * A producer represents the source of a single media track (audio or video).  * @param {object} data.dtlsParameters - Datagram Transport Layer Security (DTLS) parameters.  * These parameters are necessary for securing the transport with encryption.  */ socket.on("connectProducerTransport", async ({ dtlsParameters }) => {   // ... Connecting the sending transport ... });
 // Event handler for producing media socket.on("transport-produce", async ({ kind, rtpParameters }, callback) => {   // ... Producing media ... });
 // Event handler for connecting the receiving transport socket.on("connectConsumerTransport", async ({ dtlsParameters }) => {   // ... Connecting the receiving transport ... });
 /**  * Event handler for consuming media.  * This function sets up a consumer for receiving media from the peer.  * A consumer represents the endpoint for receiving media of a single kind  * (audio or video) from a remote peer. Creating a consumer involves multiple  * steps to ensure that the media can be received and decoded correctly.  */ socket.on("consumeMedia", async ({ rtpCapabilities }, callback) => {   // ... Consuming media ... });
 // Event handler for resuming media consumption socket.on("resumePausedConsumer", async (data) => {   // ... Resuming media consumption ... });});

const createWebRtcTransport = async (  callback: (arg0: {    params: mediasoup.types.WebRtcTransportOptions | { error: unknown };  }) => void) => { // ... WebRTC transport creation, configuration, and event handling ...};

server.listen(port, () => {  console.log(`Server running at http://localhost:${port}`);});

import { useEffect, useRef, useState } from "react";import { io } from "socket.io-client";import { Device } from "mediasoup-client";
export default function Home() { /**  * References to the local and remote video HTML elements.  * These refs are used to attach media streams to the video elements for playback.  */ const videoRef = useRef<HTMLVideoElement | null>(null); const remoteVideoRef = useRef<HTMLVideoElement | null>(null);
 /**  * State to hold encoding parameters for the media stream.  * Encoding parameters control the quality and bandwidth usage of the transmitted video.  * Each object in the encoding array represents a different layer of encoding,  * allowing for scalable video coding (SVC). The parameters defined here are:  * - rid: The encoding layer identifier.  * - maxBitrate: The maximum bitrate for this layer.  * - scalabilityMode: The scalability mode which specifies the temporal and spatial scalability.  *  * Additionally, codecOptions are provided to control the initial bitrate.  */ const [params, setParams] = useState({   encoding: [     { rid: "r0", maxBitrate: 100000, scalabilityMode: "S1T3" }, // Lowest quality layer     { rid: "r1", maxBitrate: 300000, scalabilityMode: "S1T3" }, // Middle quality layer     { rid: "r2", maxBitrate: 900000, scalabilityMode: "S1T3" }, // Highest quality layer   ],   codecOptions: { videoGoogleStartBitrate: 1000 }, // Initial bitrate });
 /**  * State to hold references to various mediasoup client-side entities.  * These entities are crucial for managing the media transmission and reception.  */ const [device, setDevice] = useState<any>(null); // mediasoup Device const [socket, setSocket] = useState<any>(null); // Socket for signaling const [rtpCapabilities, setRtpCapabilities] = useState<any>(null); // RTP Capabilities for the device const [producerTransport, setProducerTransport] = useState<any>(null); // Transport for sending media const [consumerTransport, setConsumerTransport] = useState<any>(null); // Transport for receiving media
 /**  * Effect to initialize the socket connection on component mount.  * The socket is used for signaling to coordinate media transmission.  * On successful connection, the camera is started to obtain a media stream.  */ useEffect(() => {   const socket = io("http://localhost:4000/mediasoup");
   setSocket(socket);   socket.on("connection-success", (data) => {     startCamera();   });   return () => {     socket.disconnect();   }; }, []);
 /**  * Function to start the camera and obtain a media stream.  * This stream is then attached to the local video element for preview.  */ const startCamera = async () => {   try {     const stream = await navigator.mediaDevices.getUserMedia({ video: true });     if (videoRef.current) {       const track = stream.getVideoTracks()[0];       videoRef.current.srcObject = stream;       setParams((current) => ({ ...current, track }));     }   } catch (error) {     console.error("Error accessing camera:", error);   } };
}

//...      <video ref={videoRef} id="localvideo" autoPlay playsInline />      <video ref={remoteVideoRef} id="remotevideo" autoPlay playsInline />//...

<div style={{ display: "flex", flexDirection: "column", gap: "20px" }}>        <button onClick={getRouterRtpCapabilities}>          Get Router RTP Capabilities        </button>        <button onClick={createDevice}>Create Device</button>        <button onClick={createSendTransport}>Create send transport</button>        <button onClick={connectSendTransport}>          Connect send transport and produce        </button>        <button onClick={createRecvTransport}>Create recv transport</button>        <button onClick={connectRecvTransport}>          Connect recv transport and consume        </button>      </div>

* Step 1: Retrieve the Router's RTP Capabilities.   * This function requests the router's RTP capabilities from the server,   * which are essential to configure the mediasoup Device.   * The router's RTP capabilities describe the codecs and RTP parameters supported by the router.   * This information is crucial for ensuring that the Device is compatible with the router.   */  const getRouterRtpCapabilities = async () => {    socket.emit("getRouterRtpCapabilities", (data: any) => {      setRtpCapabilities(data.routerRtpCapabilities);      console.log(`getRouterRtpCapabilities: ${data.routerRtpCapabilities}`);    });  };

/**   * Step 2: Create and Initialize the mediasoup Device.   * This function creates a new mediasoup Device instance and loads the router's RTP capabilities into it.   * The Device is a client-side entity that provides an API for managing sending/receiving media with a mediasoup server.   * Loading the router's RTP capabilities ensures that the Device is aware of the codecs and RTP parameters it needs to use   * to successfully send and receive media with the server.   *   * If the Device is unable to load the router's RTP capabilities (e.g., due to an unsupported browser),   * an error is logged to the console.   */  const createDevice = async () => {    try {      const newDevice = new Device();
      await newDevice.load({ routerRtpCapabilities: rtpCapabilities });
      setDevice(newDevice);    } catch (error: any) {      console.log(error);      if (error.name === "UnsupportedError") {        console.error("Browser not supported");      }    }  };

/*** Step 3: Create a Transport for Sending Media.* This function initiates the creation of a transport on the server-side for sending media,* and then replicates the transport on the client-side using the parameters returned by the server.*/const createSendTransport = async () => { // Request the server to create a send transport socket.emit(   "createSendTransport",   { sender: true },   ({ params }: { params: any }) => {     if (params.error) {       console.log(params.error);       return;     }
     /**      * Replicate the send transport on the client-side.      * The `device.createSendTransport` method creates a send transport instance on the client-side      * using the parameters provided by the server.      */     let transport = device.createSendTransport(params);
     // Update the state to hold the reference to the created transport     setProducerTransport(transport);
     /**        * Event handler for the "connect" event on the transport.        * This event is triggered when the transport is ready to be connected.        * The `dtlsParameters` are provided by the transport and are required to establish        * the DTLS connection between the client and the server.        * This event it emitted as a result of calling the `producerTransport?.produce(params)`        * method in the next step. The event will only be emitted if this is the first time        */     transport.on(       "connect",       async ({ dtlsParameters }: any, callback: any, errback: any) => {         try {           console.log("----------> producer transport has connected");           // Notify the server that the transport is ready to connect with the provided DTLS parameters           await socket.emit("transport-connect", { dtlsParameters });           // Callback to indicate success           callback();         } catch (error) {           // Errback to indicate failure           errback(error);         }       }     );
     /**        * Event handler for the "produce" event on the transport.        * This event is triggered when the transport is ready to start producing media.        * The `parameters` object contains the necessary information for producing media,        * including the kind of media (audio or video) and the RTP parameters.        * The event is emitted as a result of calling the `producerTransport?.produce(params)`        * method in the next step.        */     transport.on(       "produce",       async (parameters: any, callback: any, errback: any) => {         const { kind, rtpParameters } = parameters;
         console.log("----------> transport-produce");
         try {           // Notify the server to start producing media with the provided parameters           socket.emit(             "transport-produce",             { kind, rtpParameters },             ({ id }: any) => {               // Callback to provide the server-generated producer ID back to the transport               callback({ id });             }           );         } catch (error) {           // Errback to indicate failure           errback(error);         }       }     );   } );};

const connectSendTransport = async () => { // Producing media using the send transport let localProducer = await producerTransport.produce(params);
 // Event handlers for track ending and transport closing events localProducer.on("trackended", () => {   console.log("trackended"); }); localProducer.on("transportclose", () => {   console.log("transportclose"); });}

const createRecvTransport = async () => { // Requesting the server to create a receive transport socket.emit(   "createTransport",   { sender: false },   ({ params }: { params: any }) => {     if (params.error) {       return;     }
     // Creating a receive transport on the client-side using the server-provided parameters     let transport = device.createRecvTransport(params);     setConsumerTransport(transport);
     // Event handler for the connect event on the transport     transport.on(       "connect",       async ({ dtlsParameters }: any, callback: any, errback: any) => {         try {           // Notifying the server to connect the receive transport with the provided DTLS parameters           await socket.emit("transport-recv-connect", { dtlsParameters });           callback();         } catch (error) {           errback(error);         }       }     );   } );};

const connectRecvTransport = async () => { // Requesting the server to start consuming media await socket.emit(   "consume",   { rtpCapabilities: device.rtpCapabilities },   async ({ params }: any) => {     if (params.error) {       return;     }
     // Consuming media using the receive transport     let consumer = await consumerTransport.consume({       id: params.id,       producerId: params.producerId,       kind: params.kind,       rtpParameters: params.rtpParameters,     });
     // Accessing the media track from the consumer     const { track } = consumer;
     // Attaching the media track to the remote video element for playback     if (remoteVideoRef.current) {       remoteVideoRef.current.srcObject = new MediaStream([track]);     }
     // Notifying the server to resume media consumption     socket.emit("consume-resume");   } );};