A Black Box Analysis of Webrtc Mouth-To-Ear Delays

Abstract

Due to the recent shift toward cloud based computing, some of the world‘s leading standardization bodies have combined forces to provide guidelines and standards for native implementation of RealTime Communication (RTC) in the browsers. The World Wide Web Consortium (W3C) works on WebRTC APIs [1], while the Internet Engineering Task Force (IETF) is concerned with underlying standards [2]. Their efforts led to the development of WebRTC project by Google, Mozilla and Opera, which is an open framework that allows browsers to be RTC ready [3]. In this paper, we examine WebRTC from a Quality of Service (QoS) perspective, focusing on Mouth-to-Ear (M2E) delays under various application configurations.