Performance Analysis of WebRTC and SIP-based Audio and Video Communication Systems
- PDF / 3,504,991 Bytes
- 22 Pages / 595.276 x 790.866 pts Page_size
- 63 Downloads / 276 Views
ORIGINAL RESEARCH
Performance Analysis of WebRTC and SIP‑based Audio and Video Communication Systems T. P. Fowdur1 · N. Ramkorun1 · P. K. Chiniah1 Received: 15 October 2020 / Accepted: 19 October 2020 / Published online: 2 November 2020 © Springer Nature Singapore Pte Ltd 2020
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. Keywords WebRTC · SIP · SDP · QoS · SSNR · PSNR
Introduction The internet is widely used for different applications, such as e-shopping, browsing, and communication. The internet has assisted the evolution of communication by providing easy access to real-time communication through thirdparty applications such as Skype, WhatsApp, and other platforms. Two commonly used frameworks for developing real-time communication applications are the web realtime communication (WebRTC), which is an open source framework and the sessions initiation protocol (SIP). Using WebRTC, current web browsers can exchange video and audio (in a peer-to-peer manner) with other capable * T. P. Fowdur [email protected] N. Ramkorun [email protected] P. K. Chiniah [email protected] 1
Department of Electrical and Electronics Engineering, University of Mauritius, Reduit, Mauritius
browsers. Browser-to-browser communication has proven to be a groundbreaking evolution as it permits web developers to build real-time multimedia applications which operate plugin-free, that is, those applications can work without proprietary plugins like Adobe Flash [1]. WebRTC has been standardized by International communities such as The World Wide Web Consortium (W3C) and The Internet Engineering Task Force (IETF) which are working hand in hand into elaborating JavaScript APIs, HTML tags, and commun
Data Loading...
