I found this on your blog, there was no comment section.

Aren't Websockets and WebRTC two different things? You call it and name it WebRTC while using WebSockets.

WebRTC - high-performance over UDP
WebSocket - bidirectional over TCP.

https://stackoverflow.com/questions/18799364/webrtc-vs-websockets-if-webrtc-can-do-video-audio-and-data-why-do-i-need-web

Nice stuff,

I got this error: Uncaught SyntaxError: Failed to execute 'addIceCandidate' on 'RTCPeerConnection': The ICE candidate could not be added.

For this line of code: peerConnection.addIceCandidate(new RTCIceCandidate(signal.ice));

And one more question: Do you pipe everything through the websocket. I mean both audio and video data frame at the same time. I use different technique and experienced data clogging or data bottleneck sending video or audio data alone through websocket.

Response will be greatly appreciated. I've a plan to make use of your code with more complicated stuffs. Hope you don't mind about that.

Hi, may I ask how to implement your webrtc example in an ios app?

hi sir,
you done a great job.
i am getting a error is "Upgrade Required".
can you help me to solve this problem?

thank you.

Hi,

I am new to webrtc, and I would like to implement peer-to-server for audio and video. I use existing code as client & signal server, and use "node-webrtc" as "peer-to-server"'s server, it can build up the connection, but the server side cannot receive video stream. Attached file is my test code by node-webrtc.

index.zip

I am trying to run this webrtc with IP address and using self signed certificate which I generated by following the steps from this post: https://medium.com/@antelle/how-to-generate-a-self-signed-ssl-certificate-for-an-ip-address-f0dd8dddf754

I am still getting Not Secure connection after replacing key and cert files in the root folder and running the node server back. Any idea?

google chrome version:
Version 52.0.2743.82 (64-bit)

NET::ERR_CERT_AUTHORITY_INVALID

With the latest Firefox (v56) I get no local video feed and in the console appears the following error:

MediaStreamError { name: "InternalError", message: "Starting video failed", constraint: "", stack: "" }

I have been trying to make a datachannel work without success, Can someone point me in the right direction?

You should remove stun.services.mozilla.com from the list of STUN servers to use, as this currently requires authentication (see Example at MDN RTCConfiguration). I would suggest to replace it with stun.stunprotocol.org.

First, thanks for creating and maintaining this repo. I found it very helpful. 🎉

Issue

On this line instead of using the now deprecated addStream you could use addTrack.

localStream.getTracks().forEach((track) => peerConnection.addTrack(track, localStream))

Install WebRTC Network Limiter extension on chrome (https://chrome.google.com/webstore/detail/webrtc-network-limiter/npeicpdbkakmehahjeeohfdhnlpdklia) and set it to "use only me default public ip address". Then, when you try connecting with two Chrome windows, it fails with the ICEConnectionState going to failed.

Hello, I can not see the remote interlocutor. I did it like in the code, everything works fine and when you press the button, the start of the video connection two times receives the video stream from the remote interlocutor, but there is no video. The signaling server is built on the WebSocket java library, data through this server is perfectly sent to two interlocutors. I also use https on the signaling server, it receives and sends connection data and formats perfectly, and the interlocutors accept this data and the function of receiving the remote video signal is triggered, but the video does not.

Hello there i was unable to show the streams between different clients and server when they were on a different network!.

I tried to use a remote server, but it is not working. The remote video does not appear, only a black screen . The scripts were changed only the WebSocket server path. Any ideas ?

While running on browser, it keeps displaying "waiting for ".

After closing both the the browsers, command window displays an error.
This is the error at the end...

  throw er; // Unhandled 'error' event
  ^

Error: read ECONNRESET
at TLSWrap.onStreamRead (internal/stream_base_commons.js:205:27)
Emitted 'error' event on WebSocket instance at:
at WebSocket.finalize (C:\inetpub\pkVC1\node_modules\ws\lib\WebSocket.js:182:41)
at TLSSocket.emit (events.js:315:20)
at emitErrorNT (internal/streams/destroy.js:92:8)
at emitErrorAndCloseNT (internal/streams/destroy.js:60:3)
at processTicksAndRejections (internal/process/task_queues.js:84:21) {
errno: 'ECONNRESET',
code: 'ECONNRESET',
syscall: 'read'
}
npm ERR! code ELIFECYCLE
npm ERR! errno 1
npm ERR! [email protected] start: node server/server.js
npm ERR! Exit status 1
npm ERR!
npm ERR! Failed at the [email protected] start script.
npm ERR! This is probably not a problem with npm. There is likely additional logging output above.

npm ERR! A complete log of this run can be found in:
npm ERR! C:\Users\HP\AppData\Roaming\npm-cache_logs\2020-08-08T09_45_25_858Z-debug.log

To make it work, I had to update the broadcast from WS:

// Broadcast to all.
wss.broadcast = function broadcast(data) {
wss.clients.forEach(function each(client) {
if (client.readyState === WebSocket.OPEN) {
client.send(data);
}
});
};

I tried your source which is really a good one. But I am facing one problem. On Same netwrok it works perfectly but on different network remote side video & audio is not working. My server is a dedicated Windows 2012 server with static IP. On Same network means if I use it on a PC and laptop with wifi on the same network it works perfectly but when ever I go to different network it doesn't work. What should I do?

Hey.
Today my browser said me "RTCPeerConnection.onaddstream is deprecated. Use RTCPeerConnection.ontrack".
:)

Hi.

I just tried your code, curious about this tech, and there is a little problem:

In webrtc.js, the following variable is not correct (taken from the source repo):

var peerConnectionConfig = {
  'iceServers': [
    {'urls': 'stun:stun.stunprotocol.org:3478'},
    {'urls': 'stun:stun.l.google.com:19302'},
  ]
};

The error is the following:

SyntaxError: RTCPeerConnection constructor passed invalid RTCConfiguration - improper scheme: stun1.l.google.com webrtc.js:42:20

However, using the variable your provide at https://shanetully.com/2014/09/a-dead-simple-webrtc-example/ fixes the error.

var peerConnectionConfig = {'iceServers': [{'url': 'stun:stun.services.mozilla.com'}, {'url': 'stun:stun.l.google.com:19302'}]};

So I have the same exact code as you with the following exception:

peerConnection.createOffer(offerOptions).then(createdDescription).catch(errorHandler);
where

const offerOptions = {
  offerToReceiveAudio: 1,
  offerToReceiveVideo: 1
};

Furthermore, I am also using a TURN server: {'urls': 'turn:numb.viagenie.ca','credential': '**mypass','username': '**myusername'}

But when I check the peerConnection.connectionState in the console, it says "failed".

What I do is I open 2 incognito windows in Chrome and go to

https://localhost:8443

in each window.

Then I click the "Start Video" on one of the pages and then the remoteVideo shows up with an image (i.e. a snapshot) of the frame (on both the windows). Then the localVideo changes as I move the camera, but the remoteVideo stays as a snapshot of when the connection happened (on both the windows). Then I check to see the connectionState of the peerConnection and it says it failed.

I have tried looking for solutions, but I cannot find any. Is there something I am doing wrong please let me know.

can not establish wss on firefox
https://picasaweb.google.com/106185541018774360364/IMx6#6285590650062588018

With latest Chrome (v62) I am getting this warning.
The warning is self-explanatory of what needs to be done, i.e., replace, localVideo.src = window.URL.createObjectURL(stream); with localVideo.srcObject = stream; and remoteVideo.src = window.URL.createObjectURL(event.stream); with remoteVideo.srcObject = event.stream;.

got description
VM36 webrtc.js:92 DOMException: Failed to set remote offer sdp: Called in wrong state: kHaveLocalOffer
6VM36 webrtc.js:92 DOMException: Error processing ICE candidate

First of all - thank you for your solution on multiple user videochat. I was studying over other examples, but ended up here, because it is open source and simple (but not much for me, as I am a begginer) to use. I would be very grateful if you could guide me quick through some points.

The thing is: I deployed your code on a Plesk server with the following changes:

server.js

const HTTPS_PORT = 8443; //default port for https is 443
const HTTP_PORT = 8443; //default port for http is 80

and changed paths to files

 if (request.url === '/webrtc.js') {
    response.writeHead(200, { 'Content-Type': 'application/javascript' });
    response.end(fs.readFileSync('public/webrtc.js'));
  } else if (request.url === '/style.css') {
    response.writeHead(200, { 'Content-Type': 'text/css' });
    response.end(fs.readFileSync('public/style.css'));
  } else {
    response.writeHead(200, { 'Content-Type': 'text/html' });
    response.end(fs.readFileSync('public/index.html'));
  }
};

webrtc.js

      .then(() => {
        serverConnection = new WebSocket('wss://' + 'localhost' + ':' + WS_PORT + '/');

Node.js:
Document root: /httpdocs/public
Application URL: http://someurl.dev
Application root: /httpdocs

So the structure on the server is:

**httpdocs/**
    |__ server.js
    |__ package.json
    |__ package-lock.json
    |__ key.perm
    |__ cert.perm
    |__ generate_cert.sh
    |__ install_service.js
    |__ **public/**
             |__ webrtc.js
             |__ index.html
             |__ style.css

Locally, I can open multiple tabs in Chrome and I see myself many times duplicated. Which is ok.
I can also duplicate tabs on remote server and I can see all my duplications in one tab. Which is also ok.

But if I test it on the remote server with my colleague: he sees only himself and I see only myself. Which is absolutelly horrible. And no errors or whatsoever in console.
I am struggling around this problem for some days already, obviously, a lack of experience doesn't let me solve it by myself.
Could you let me know, what I am doing wrong?

Error on remote server:
webrtc.js:44 WebSocket connection to 'wss://localhost:8443/' failed: Error in connection establishment: net::ERR_CONNECTION_REFUSED

P.S. I couldn't connect from my mobile also. I saw a topic here already, but I don't find any mobile stun servers.

use firefox 41.0.2
http://webrtc.github.io/adapter/adapter-latest.js
fail line
var makeMapStats = (stats, legacyStats) => {
//////////////
npm install webrtc-adapter
after change to adapter.js then works fine...
/////////////
firefox 41.0.2
webrtc.js error in this line
serverConnection = new WebSocket('wss://' + window.location.hostname + ':8443');
change to chrome then works fine
//////////////

can improve for firefox?

First of all I'm sorry for my English, I'm Brazilian. Your code works very well, but in a test with 6 computers, when you click the "Start" button a second time, it will no longer work. What I wanted was to switch to another pair. Can you help?