Real-Time Communication / Media

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.

Part of my work withPharaoh Soft Ltd for vivdbe.

Real-Time
WebRTC
Performance

vivdbe overview with audio and video calling, session controls, and WebRTC and WebSocket interface concepts. — Project visualization · Interface conceptOpen image at full size

Context

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

Approach

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

Outcome

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.

Built with

WebRTCWebSocketsReactTypeScriptNode.js