This service delivers a real-time video communication platform built using WebRTC, enabling secure, low-latency peer-to-peer audio and video calls directly in the browser without external plugins.
Core Capabilities
Real-Time Video & Audio Communication
High-quality peer-to-peer video and voice calls with minimal latency.
Web-Based Access
Fully browser-based solution compatible with modern desktop and mobile browsers.
Signaling Server Architecture
Custom signaling implemented using Node.js, Express, and Socket.IO to establish and manage peer connections.
Secure Media Transmission
Encrypted audio/video streams using WebRTC’s built-in DTLS and SRTP protocols.
Room-Based or One-to-One Calls
Support for private one-to-one sessions and scalable room-based communication.
Frontend & Backend Separation
Frontend deployed on platforms like Netlify, with backend signaling server deployed independently (e.g., Render).
Technology Stack
Frontend: HTML, CSS, JavaScript (WebRTC APIs)
Backend: Node.js, Express.js
Real-Time Communication: Socket.IO
Protocols: WebRTC, STUN (TURN-ready architecture)
Deployment: Netlify (Frontend), Render / Cloud VM (Backend)