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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
