Performance Evaluation of WebRTC-Based Video Conferencing: A Comprehensive Analysis

Abstract

In an ever-evolving technological landscape, addressing the performance challenges of real-time communication protocols is crucial. Real-time communication, facilitated by streaming media protocols, utilizes peer-to-peer or client-server models to enhance Quality of Service (QoS). WebRTC (Web Real-Time Communication) stands as a widely adopted, browser-based, open-source, peer-to-peer protocol, offering real-time media transmission through JavaScript APIs without third-party plugins. This paper presents an in-depth performance evaluation of a WebRTC-based video conferencing system using Socket.io services on a Node.js server. Our research expands on recent studies by introducing a comprehensive set of performance parameters, including Processing delay, CPU Utilization, Latency, Jitter, and Packet Loss, and packet delay. Our findings indicate that WebRTC performs exceptionally well within specific latency thresholds. However, scalability concerns emerge when a large number of clients are introduced, especially in bandwidth-constrained environments.