Performance Analysis of WebRTC and SIP-based Audio and Video Communication Systems

Abstract

Video and audio communications have become an integral part of all spheres of life. Two commonly used real-time communication protocols for IP-based video and audio communications are the session initiation protocol (SIP) and web real-time communications (WebRTC). These two protocols have been widely used in softphone and video conferencing applications. The main aim of this paper is to make a comparative analysis of the performance of client server applications for video and audio communications developed by both SIP and WebRTC. The SIP system comprises of a desktop client developed in C#, a mobile client developed in Android studio and a FreeSwitch server. For the WebRTC application, the client was developed in JavaScript and the server in Node.js. The WebRTC application also included a speech translation API for real-time speech translation and employed two different codecs namely VP8 and VP9 via a modification of the Session Description Protocol (SDP) header. Results showed that WebRTC peer-to-peer audio communication provides a much lower quality in terms of SSNR than the SIP audio calls. However, for video communications, WebRTC provides better quality in terms of PSNR than SIP. The performance of the WebRTC application was also evaluated with the different codecs in terms of PSNR as well as the impact of the real-time speech translation delay.