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Free WebRTC signaling server: peer to peer WebRTC live streaming, handles multiple channels (streams) and viewers per channel, support for STUN/TURN (tested with Coturn), accounts and resource limitation plans.

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VideoWhisper WebRTC : Signaling Server with STUN/TURN Support

This is a WebRTC signaling server designed for VideoWhisper HTML5 Videochat, that can also be used with new apps. It's built in NodeJS, supports SSL, TURN/STUN configuration, authentication with static key or accounts (MySQL), streaming limitations & plans, API. Can be used to publish a stream from a broadcaster to 1 or more subscribed viewers and possible applications include 1 to 1 (1 way or 2 way private), 1 to multiple, multiple to multiple (conferencing) live video streaming.

Live Demos without Registration

HTML5 Videochat / P2P Video Call

Features

Free Server for Testing, Development

If you want to test with an existing server, register for a Free Developers account with VideoWhisper, to get a server address and token. Includes STUN/TURN server. Limited availability.

Installation

  • Install NodeJS if not already available on your server (ie. yum install nodejs)
  • Deploy these files in a server folder
  • Run npm update in folder to get dependencies
  • Get a certificate with CRT & KEY files for SSL
  • For relaying in production, install Coturn server or other TURN server (not required for intranet testing)
  • Configure your server in .env file (path to certificate files, TURN server configuration, static token and/or MySQL)
  • Install PM2 npm install pm2 -g(process manager for NodeJS), configure startup options pm2 startup
  • Development: pm2 start ecosystem.config.js --env development --watch --attach --ignore-watch="node_modules/*"
  • Production: pm2 start ecosystem.config.js --env production
  • Manage: pm2 status / pm2 stop 0 / pm2 start 0 (for id 0)
  • Troubleshooting: Run from folder with npm start to see live server output in terminal and end it with Ctrl+C

Configure HTML5 Videochat

You will need the signaling server address, based on certificate and port you configured, like wss://yourserverdomain.com:3000 and the SecretToken you configured or account token from accounts table in MySQL database.

  • For Paid Videochat / PPV Live Webcams from WordPress plugin settings, WebRTC tab, select WebRTC Streaming Server : VideoWhisper or Auto and fill Address, Token for VideoWhisper WebRTC. Auto requires both VideoWhisper WebRTC for private chats and Wowza SE as relay for 1 to many group chats (recommended for scaling to many viewers).

  • For 2Way VideoCalls and Random Chat from WordPress plugin settings, Server tab, select WebRTC Streaming Server : VideoWhisper and fill Address, Token for VideoWhisper WebRTC.

  • For PHP HTML5 Videochat editions P2P 2 Way Videocall , Live Video Streaming , configure settings.php :

 $options = array(
	'serverType' => 'videowhisper', //videowhisper/wowza 
	'vwsSocket' => 'wss://yourserver.com:3000/',
	'vwsToken' => 'YourSecretToken',
    ...

If you implement your own videochat apps, you need to use same functions and logic.

API

Server currently implements these GET requests:

  • https://yourServer.com:3000/ - Shows server version & features
  • https://yourServer.com:3000/connections/?apikey=API_KEY - Shows current list of connections,
  • https://yourServer.com:3000/channels/?apikey=API_KEY - Shows current list of published channels (with streaming parameters like resolution, bitrate)
  • https://yourServer.com:3000/stats/?apikey=API_KEY - Shows usage stats by account (number of connections, bitrate)
  • https://yourDomain:PORT/update-accounts?apikey=API_KEY - Reloads accounts from MySQL without restarting server (call after adding a new account)

Configure API_KEY in .env. In development mode the apikey parameter is not required.

Commercial Services

Consult VideoWhisper for:

  • Turnkey Setups
  • Technical Support
  • Hosting (WebRTC Signaling, WowzaSE, TURN, Webhosting cPanel & WordPress)
  • Custom Development
  • Commercial Solutions

MySQL Structure

For managing connections from different accounts (websites, setups) and/or setting limitations, use a MySQL with unique token keys (per account).

DROP TABLE IF EXISTS accounts;
  CREATE TABLE accounts (
    id INT NOT NULL AUTO_INCREMENT,
    name VARCHAR(255) NOT NULL,
    token VARCHAR(255) NOT NULL,
    properties TEXT NOT NULL,
    planId INT,
    meta TEXT,
    contactId INT,
    created INT,
    PRIMARY KEY (id),
    UNIQUE KEY (token), 
    KEY (planId),
    KEY (contactId),
    KEY (created)
  );

  DROP TABLE IF EXISTS plans;
  CREATE TABLE plans (
    id INT NOT NULL AUTO_INCREMENT,
    name VARCHAR(255) NOT NULL,
    properties TEXT NOT NULL,
    PRIMARY KEY (id)
  );

Properties are JSON encoded, for accounts and plans. Some fields like contactId, created, meta are used with external integration and may not be required for your project.

Sample test data showcasing supported properties:

INSERT INTO accounts (name, token, properties, meta, planId) VALUES ('Test Account', 'testToken', '{}', '', 1);
INSERT INTO accounts (name, token, properties, meta, planId) VALUES ('Suspended Account', 'suspendedToken', '{"suspended":true}', '', 1);
INSERT INTO plans (name, properties) VALUES ('Developers Plan', '{"connections":5,"totalBitrate":2700, "bitrate":500,"framerate":15,"audioBitrate":32,"width":640,"height":360}');

Consider both bitrate and audioBitrate when setting an account limit for totalBitrate. In example 1064 for 2 users with 500 bitrate and 32 audioBitrate.

How does it work?

To use with own WebRTC streaming application(s) see a brief description below:

Server

  1. For each stream server manages a channel with broadcaster & player(s)
  2. Players call subscribe to register in a channel
  3. Broadcaster calls publish to register and get players in channel
  4. Peers exchange offers, answers, ice candidates with messagePeer

Broadcaster

  1. For each player, makes a peerConnection and creates, sets, sends offer when peerConnection.onnegotiationneeded
  2. Sends the ice candidate on peerConnection.onicecandidate
  3. Sets ice candidates received from server with peerConnection.addIceCandidate

Player

  1. On offer message from server creates peerConnection, sets remote description, creates sets, sends answer
  2. Sends the ice candidate on peerConnection.onicecandidate
  3. Sets ice candidates received from server with peerConnection.addIceCandidate

About

Free WebRTC signaling server: peer to peer WebRTC live streaming, handles multiple channels (streams) and viewers per channel, support for STUN/TURN (tested with Coturn), accounts and resource limitation plans.

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