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• I received connection refused error while trying to stream live video through RTMP with FFMPEG

  25 septembre 2020, par Femzy

  I am working on a nodeJs app that can send camera stream to third party plartform i.e Facebook and Youtube using the RTMP protoco ;.. It works well on my localhost but once i deploy to the server, it only give me errors. The error I get is below on this content..
Here is my codes

  


  server.js

  


  


  


  const child_process = require('child_process'); // To be used later for running FFmpeg
const express = require('express');
const http = require('http');
const WebSocketServer = require('ws').Server;

const app = express();
const server = http.createServer(app).listen(4000, () => {
  console.log('Listening...');
});

// Serve static files out of the www directory, where we will put our HTML page
app.use(express.static(__dirname + '/www'));


const wss = new WebSocketServer({
  server: server
});
wss.on('connection', (ws, req) => {
  
  
  
  const rtmpUrl = 'rtmp://a.rtmp.youtube.com/live2/MyStreamId';
  console.log('Target RTMP URL:', rtmpUrl);
  
  // Launch FFmpeg to handle all appropriate transcoding, muxing, and RTMP.
  // If 'ffmpeg' isn't in your path, specify the full path to the ffmpeg binary.
  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.  Please also read the security considerations
    // later on in this tutorial.
    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');
  });
  
});

  


  


  



  On the server.js file i create a websocket to receive stream data from the client side and then use FFMPEG to send the stream data over to youtube via the RTMP url

  


  Here is my client.js code

  


  


  


  const ws = new WebSocket(
             'wss://my-websocket-server.com'

        );
         ws.addEventListener('open', (e) => {
             console.log('WebSocket Open', e);
             drawVideosToCanvas();
             mediaStream = getMixedVideoStream(); // 30 FPS
             mediaRecorder = new MediaRecorder(mediaStream, {
               mimeType: 'video/webm;codecs=h264',
               //videoBitsPerSecond : 3000000000
               bitsPerSecond: 6000000
             });

             mediaRecorder.addEventListener('dataavailable', (e) => {
               ws.send(e.data);
             });
             mediaRecorder.onstop = function() {
              ws.close.bind(ws);
              isRecording = false;
              actionBtn.textContent = 'Start Streaming';
              actionBtn.onclick = startRecording;
             }
             mediaRecorder.onstart = function() {
              isRecording = true;
              actionBtn.textContent = 'Stop Streaming';
              actionBtn.onclick = stopRecording;
              screenShareBtn.onclick = startSharing;
              screenShareBtn.disabled = false;
             }
             //mediaRecorder.addEventListener('stop', ws.close.bind(ws));

             mediaRecorder.start(1000); // Start recording, and dump data every second

           });

  


  


  



  On my client.js file, i captured users camera and then open the websocket server to send the data to the server.. Every thing works fine on local host expect for when i deploy it to live server..
i am wondering if there is a bad configuration on the server.. The server is Centos 7.8 and the app was runing on Apache software
Here is how i configured the virtual host for the websocket domain

  


  


  


  ServerName my-websocket.com

  RewriteEngine on
  RewriteCond %{HTTP:Upgrade} websocket [NC]
  RewriteCond %{HTTP:Connection} upgrade [NC]
  RewriteRule .* "ws://127.0.0.1:3000/$1" [P,L]

  ProxyPass "/" "http://127.0.0.1:3000/$1"
  ProxyPassReverse "/" "http://127.0.0.1:3000/$1"
  ProxyRequests off

  


  


  



  I don't know much about server configuration but i just thought may be the configuration has to do with why FFMPEg can not open connection to RTMP protocol on the server.

  


  here is the error am getting

  


  


  


  FFmpeg STDERR: Input #0, lavfi, from &#x27;anullsrc&#x27;:&#xA;  Duration:&#xA;FFmpeg STDERR: N/A, start: 0.000000, bitrate: 705 kb/s&#xA;    Stream #0:0: Audio: pcm_u8, 44100 Hz, stereo, u8, 705 kb/s&#xA;&#xA;DATA <buffer 1a="1a">&#xA;DATA <buffer 45="45" df="df" a3="a3" 42="42" 86="86" 81="81" 01="01" f7="f7" f2="f2" 04="04" f3="f3" 08="08" 82="82" 88="88" 6d="6d" 61="61" 74="74" 72="72" 6f="6f" 73="73" 6b="6b" 87="87" 0442="0442" 85="85" 02="02" 18="18" 53="53" 80="80" 67="67" ff="ff" 53991="53991" more="more" bytes="bytes">&#xA;DATA <buffer 40="40" c1="c1" 81="81" 00="00" f0="f0" 80="80" 7b="7b" 83="83" 3e="3e" 3b="3b" 07="07" d6="d6" 4e="4e" 1c="1c" 11="11" b4="b4" 7f="7f" cb="cb" 5e="5e" 68="68" 9b="9b" d5="d5" 2a="2a" e3="e3" 06="06" c6="c6" f3="f3" 94="94" ff="ff" 29="29" 16="16" b2="b2" 60="60" 04ac="04ac" 37="37" fb="fb" 1a="1a" 15="15" ea="ea" 39="39" a0="a0" cd="cd" 02="02" b8="b8" 56206="56206" more="more" bytes="bytes">&#xA;FFmpeg STDERR: Input #1, matroska,webm, from &#x27;pipe:&#x27;:&#xA;  Metadata:&#xA;    encoder         :&#xA;FFmpeg STDERR: Chrome&#xA;  Duration: N/A, start: 0.000000, bitrate: N/A&#xA;    Stream #1:0(eng): Audio: opus, 48000 Hz, mono, fltp (default)&#xA;    Stream #1:1(eng): Video: h264 (Constrained Baseline), yuv420p(progressive), 1366x768, SAR 1:1 DAR 683:384, 30.30 fps, 30 tbr, 1k tbn, 60 tbc (default)&#xA;&#xA;FFmpeg STDERR: [tcp @ 0xe5fac0] Connection to tcp://a.rtmp.youtube.com:1935 failed (Connection refused), trying next address&#xA;[rtmp @ 0xe0fb80] Cannot open connection tcp://a.rtmp.youtube.com:1935&#xA;&#xA;FFmpeg STDERR: rtmp://a.rtmp.youtube.com/live2/mystreamid: Network is unreachable&#xA;&#xA;FFmpeg child process closed, code 1, signal null</buffer></buffer></buffer>

  


  


  



  I will really appreciate if I could get some insight on what may be causing this issue or what i can do to solve it..Thanks in advance..

  


• Use Google Analytics and risk fines, after CJEU ruling on Privacy Shield

  27 août 2020, par Joselyn Khor — Privacy
  EU websites using Google Analytics and Facebook are being targeted by European privacy group noyb after the invalidation of the Privacy Shield. They filed a complaint against 101 websites for continuing to send data to the US. 

  “A quick analysis of the HTML source code of major EU webpages shows that many companies still use Google Analytics or Facebook Connect one month after a major judgment by the Court of Justice of the European Union (CJEU) - despite both companies clearly falling under US surveillance laws, such as FISA 702. Neither Facebook nor Google seem to have a legal basis for the data transfers.”

  

  The Privacy Shield previously allowed for EU data to be transferred to the US. However, this was invalidated by the Court of Justice of the European Union (CJEU) on July 16, 2020. The CJEU deemed it illegal for any websites to transfer the personal data of European citizens to the US. 

  They also made it clear in a press release that “data subjects can claim compensation for inadmissible data exports (marginal no. 143 of the judgment). This should in particular include non-material damage (“compensation for pain and suffering”) and must be of a deterrent amount under European law.” Which puts extra financial pressure on websites to take the new ruling seriously.

  Immediate action is required after Google Privacy Shield invalidation

  The Berlin Commissioner for Data Protection and Freedom of Information therefore calls on all those responsible under its supervision to observe the decision of the ECJ [CJEU]. Those responsible who transfer personal data to the USA - especially when using cloud services - are now required to immediately switch to service providers in the European Union or in a country with an adequate level of data protection.

  As the ruling is effective immediately, there’s a pressing need for websites using Google Analytics to act, or face getting fined.

  What does this mean for you ?

  If you’re using Google Analytics the safest bet is to stop using it immediately. 

  "Neither Google Analytics nor Facebook Connect are necessary for the operation of these websites and could therefore have been replaced or at least deactivated in the meantime."

  If you still need to use it, then you’ll need to inform your visitors via a clear consent screen. This banner needs to make clear their personal data will be sent to the US, and to educate them about any potential risk related to this. They will then need to explicitly agree to this. 

  Another downside of cookie consent screens is that you may also suffer a damaging loss of visitors. After implementing cookie consent best practices, the UK’s data regulator the Information Commissioner’s Office (ICO) found a 90% drop in traffic, “implying a ninety percent drop in opt-in rates.”

  With an acceptance rate for such consent screens being lower than 10% your analytics becomes guesswork rather than science. 

  Looking for a privacy-respecting alternative to Google Analytics ?

  Privacy compliant Matomo Analytics is one of the best Google Analytics alternatives availalble. 

  With Matomo you’re able to continue using analytics without facing the wrath of both the GDPR and the CJEU. Matomo On-Premise lets you choose where your data is stored, so you can ensure no data is processed in the US. 

  Matomo is privacy-friendly and can be tweaked to comply with all privacy laws. Including the GDPR, HIPAA, CCPA and PECR. The benefits of this include : not needing to use tracking or cookie consent screens (like with GA) ; and avoiding fines because no personal data is collected. You also get 100% accurate data and the ability to protect your user’s privacy.

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• MediaRecorder - FFMPEG - RTMP issue

  5 octobre 2017, par Bear10

  Currently I’m working on streaming Audio and Video from a video captured in Chrome 61.0.3163.100 with the MediaRecorder API.

  The backend is running on node and it spawns a child process which runs ffmpeg :

  ffmpeg -threads 4 -i udp://127.0.0.1:41234 -t 5400 -ar 44100
  
  -b:a 128k -c:v libx264 -c:a libfdk_aac -pix_fmt yuv420p -r 30 -g 60 -vb 2048k -preset slow -minrate 2000k -maxrate 4000k test.mp4

  On the client side I send the raw Blob from Chrome to the Node server via web socket :

   var mediaStream = new MediaStream([videoOutput.getVideoTracks()[0], audioOutput.getAudioTracks()[0]]),
  
              recorder = new MediaRecorder(mediaStream, {
  
                  mimeType: 'video/webm;codecs=vp8'
  
              });
  
  
  
          socket = io('http://localhost:' + port);
  
  
  
          recorder.addEventListener('dataavailable', function (evt) {
  
              socket.emit('video:data', evt.data);
  
          });

  And in Node we simply resend via udp to the ffmpeg process :

  let client = dgram.createSocket('udp4'),
  
  
  
  client.send(chunk, 0, chunk.length, 41234, '127.0.0.1', function (err, bytes) {
  
              if (err) {
  
                  throw err;
  
              }
  
          });

  The resulting mp4 is viewable on VLC without any issues, despite many of the following warnings showing up in the FFMPEG console :

  stderr: frame= 1751 fps= 31 q=17.0 size=   15997kB time=00:00:58.39 bitrate=2244.3kbits/s dup=449 drop=32 speed=1.03x
  
  stderr: [mp4 @ 0x7fefe6019000] Non-monotonous DTS in output stream 0:1; previous: 2573919, current: 2573792; changing to 2573920. This may result in incorrect timestamps in the output file.
  
  
  
  stderr: frame= 1767 fps= 31 q=17.0 size=   16166kB time=00:00:58.90 bitrate=2248.4kbits/s dup=452 drop=32 speed=1.03x
  
  stderr: [libmp3lame @ 0x7fefe7004c00] Queue input is backward in time
  
  
  
  stderr: [mp4 @ 0x7fefe6019000] Non-monotonous DTS in output stream 0:1; previous: 2614768, current: 2614686; changing to 2614769. This may result in incorrect timestamps in the output file.

  However when I try to re stream this specific video on Youtube or any other RTMP platform (ex : Facebook) the audio is choppy, the command I use :

  ffmpeg -i ./test.mp4 -f flv rtmp://a.rtmp.youtube.com/live2/MYAPIKEY

  On the other hand any other "good" video not made from the aforementioned process re streams just fine and I suspect it might be from the warnings I’m getting.

  As an added bit of information streaming directly from my PC with avfoundation to youtube also works fine, and writing from avfoundation to a file and then streaming to youtube also works ok.

  The goal is to stream from my browser to the node server and directly to the Youtube RTMP without the choppy audio issue.

  If someone knows how to get rid of the warning to ensure ffmpeg isn’t the issue or can point me in the direction to achieve the desired result that’d be great.