Socket
Book a DemoInstallSign in
Socket

msc-node

Package Overview
Dependencies
Maintainers
1
Versions
26
Alerts
File Explorer

Advanced tools

Socket logo

Install Socket

Detect and block malicious and high-risk dependencies

Install

msc-node

mediasoup client side Node.js library

Source
npmnpm
Version
0.0.6
Version published
Weekly downloads
233
-36.34%
Maintainers
1
Weekly downloads
 
Created
Source

mediasoup-client-node [WIP]

Node.js client side (not browser,just node.js) library for building mediasoup based applications.

Using werift-webrtc for webrtc protocol stack

Installation

npm install msc-node

requires at least Node.js 14

Usage Example

import {
  Device,
  RTCRtpCodecParameters,
  useAbsSendTime,
  useFIR,
  useNACK,
  usePLI,
  useREMB,
  useSdesMid,
  MediaStreamTrack,
} from "msc-node";
import { exec } from "child_process";
import { createSocket } from "dgram";
import mySignaling from "./my-signaling"; // Our own signaling stuff.

// Create a device with RtpCapabilities
const device = new Device({
  headerExtensions: {
    video: [useSdesMid(), useAbsSendTime()],
  },
  codecs: {
    video: [
      new RTCRtpCodecParameters({
        mimeType: "video/VP8",
        clockRate: 90000,
        payloadType: 98,
        rtcpFeedback: [useFIR(), useNACK(), usePLI(), useREMB()],
      }),
    ],
  },
});

// Communicate with our server app to retrieve router RTP capabilities.
const routerRtpCapabilities = await mySignaling.request(
  "getRouterCapabilities"
);

// Load the device with the router RTP capabilities.
await device.load({ routerRtpCapabilities });

// Check whether we can produce video to the router.
if (!device.canProduce("video")) {
  console.warn("cannot produce video");

  // Abort next steps.
}

// Create a transport in the server for sending our media through it.
const { id, iceParameters, iceCandidates, dtlsParameters, sctpParameters } =
  await mySignaling.request("createTransport", {
    sctpCapabilities: device.sctpCapabilities,
  });

// Create the local representation of our server-side transport.
const sendTransport = device.createSendTransport({
  id,
  iceParameters,
  iceCandidates,
  dtlsParameters,
  sctpParameters,
});

// Set transport "connect" event handler.
sendTransport.on("connect", async ({ dtlsParameters }, callback, errback) => {
  // Here we must communicate our local parameters to our remote transport.
  try {
    await mySignaling.request("transport-connect", {
      transportId: sendTransport.id,
      dtlsParameters,
    });

    // Done in the server, tell our transport.
    callback();
  } catch (error) {
    // Something was wrong in server side.
    errback(error);
  }
});

// Set transport "produce" event handler.
sendTransport.on(
  "produce",
  async ({ kind, rtpParameters, appData }, callback, errback) => {
    // Here we must communicate our local parameters to our remote transport.
    try {
      const { id } = await mySignaling.request("produce", {
        transportId: sendTransport.id,
        kind,
        rtpParameters,
        appData,
      });

      // Done in the server, pass the response to our transport.
      callback({ id });
    } catch (error) {
      // Something was wrong in server side.
      errback(error);
    }
  }
);

// Set transport "producedata" event handler.
sendTransport.on(
  "producedata",
  async (
    { sctpStreamParameters, label, protocol, appData },
    callback,
    errback
  ) => {
    // Here we must communicate our local parameters to our remote transport.
    try {
      const { id } = await mySignaling.request("produceData", {
        transportId: sendTransport.id,
        sctpStreamParameters,
        label,
        protocol,
        appData,
      });

      // Done in the server, pass the response to our transport.
      callback({ id });
    } catch (error) {
      // Something was wrong in server side.
      errback(error);
    }
  }
);

// Produce our rtp video.
exec(
  "ffmpeg -re -f lavfi -i testsrc=size=640x480:rate=30 -vcodec libvpx -cpu-used 5 -deadline 1 -g 10 -error-resilient 1 -auto-alt-ref 1 -f rtp rtp://127.0.0.1:5030"
);
const udp = createSocket("udp4");
udp.bind(5030);
const rtpTrack = new MediaStreamTrack({ kind: "video" });
udp.addListener("message", (data) => {
  rtpTrack.writeRtp(data);
});
const rtpProducer = await sendTransport.produce({ track: rtpTrack });

// Produce data (DataChannel).
const dataProducer = await sendTransport.produceData({
  ordered: true,
  label: "foo",
});

Authors

  • shinyoshiaki [github]

Original Authors

License

ISC

Original License

ISC License

Copyright © 2015, Iñaki Baz Castillo <ibc@aliax.net>

FAQs

Package last updated on 21 May 2021

Did you know?

Socket

Socket for GitHub automatically highlights issues in each pull request and monitors the health of all your open source dependencies. Discover the contents of your packages and block harmful activity before you install or update your dependencies.

Install

Related posts