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Real-TimeWebRTCPerformance

vivdbe Real-Time Calling and Communication System

Low-latency audio and video calling system built using native WebRTC with WebSocket signaling to support real-time communication and reliable session handling.

Senior Frontend Engineer

The Problem

Real-time communication requires extremely low latency, reliable signaling, and stable session handling, especially under unstable network conditions.

The Solution

Implemented a WebRTC-based media layer combined with WebSocket signaling, supported by robust lifecycle management and error-handling patterns to ensure smooth real-time communication.

What I Built

  • Implemented real-time audio and video communication using native WebRTC
  • Built WebSocket-based signaling for call setup, negotiation, and reconnection
  • Delivered low-latency call experiences with stable session lifecycle management
  • Handled edge cases such as reconnections, network drops, and state recovery
  • Designed maintainable UI states for call flows and real-time user feedback

Impact

Enabled reliable, low-latency audio and video communication suitable for real-time collaboration use cases, improving call stability, reducing reconnection issues, and delivering a smoother user experience under varying network conditions.

Tech Stack

WebRTCWebSocketsReactTypeScriptNode.js