server.js

const child_process = require('child_process');
const express = require('express');
const WebSocketServer = require('ws').Server;
const http = require('http');

const app = express();
const server = http.createServer(app).listen(3000, () => {
  console.log('Listening...');
});

const wss = new WebSocketServer({
  server: server
});

app.use((req, res, next) => {
  console.log('HTTP Request: ' + req.method + ' ' + req.originalUrl);
  return next();
});

app.use(express.static(__dirname + '/www'));

wss.on('connection', (ws, req) => {
  
  const streamKey = "sk_us-west-**************************";
  const rtmpUrl = 'rtmps://******.global-*******-video.net:443/app/'+streamKey;
  
  console.log('Target RTMP URL:', rtmpUrl);
  
  // Launch FFmpeg to handle all appropriate transcoding, muxing, and RTMP
  const ffmpeg = child_process.spawn('ffmpeg', [
    // Facebook requires an audio track, so we create a silent one here.
    // Remove this line, as well as `-shortest`, if you send audio from the browser.
    '-f', 'lavfi', '-i', 'anullsrc',
    
    // FFmpeg will read input video from STDIN
    '-i', '-',
    
    // Because we're using a generated audio source which never ends,
    // specify that we'll stop at end of other input.  Remove this line if you
    // send audio from the browser.
    '-shortest',
    
    // If we're encoding H.264 in-browser, we can set the video codec to 'copy'
    // so that we don't waste any CPU and quality with unnecessary transcoding.
    // If the browser doesn't support H.264, set the video codec to 'libx264'
    // or similar to transcode it to H.264 here on the server.
    '-vcodec', 'copy',
    
    // AAC audio is required for Facebook Live.  No browser currently supports
    // encoding AAC, so we must transcode the audio to AAC here on the server.
    '-acodec', 'aac',
    
    // FLV is the container format used in conjunction with RTMP
    '-f', 'flv',
    
    // The output RTMP URL.
    // For debugging, you could set this to a filename like 'test.flv', and play
    // the resulting file with VLC.
    rtmpUrl 
  ]);
  
  // If FFmpeg stops for any reason, close the WebSocket connection.
  ffmpeg.on('close', (code, signal) => {
    console.log('FFmpeg child process closed, code ' + code + ', signal ' + signal);
    ws.terminate();
  });
  
  // Handle STDIN pipe errors by logging to the console.
  // These errors most commonly occur when FFmpeg closes and there is still
  // data to write.  If left unhandled, the server will crash.
  ffmpeg.stdin.on('error', (e) => {
    console.log('FFmpeg STDIN Error', e);
  });
  
  // FFmpeg outputs all of its messages to STDERR.  Let's log them to the console.
  ffmpeg.stderr.on('data', (data) => {
    console.log('FFmpeg STDERR:', data.toString());
  });

  // When data comes in from the WebSocket, write it to FFmpeg's STDIN.
  ws.on('message', (msg) => {
    console.log('DATA', msg);
    ffmpeg.stdin.write(msg);
  });
  
  // If the client disconnects, stop FFmpeg.
  ws.on('close', (e) => {
    ffmpeg.kill('SIGINT');
  });
  
});