Browsing by Author "Hussain, Afaque"

Web Real Time Communication (WebRTC) is a project that aims to enable plug-in free, real time communications across multiple web-browsers. With WebRTC enabled web-browsers, users can make direct browser to browser audio calls, video calls or transfer arbitrary data. In a simple network with no Network Address Translation (NAT), firewalls or HTTP-proxies, WebRTC applications work well without any problems. But in the real world, the networks are complex and clients are usually behind NAT, firewalls and (or) HTTP proxies. Introduction of such network middle boxes creates problems for WebRTC media flow, leading to a failure in WebRTC call establishment. WebRTC uses Interactive Connectivity Establishment (ICE) framework to work around the problems posed by these middle boxes. ICE uses a combination of Session Traversal Utilities for NAT (STUN) and Traversal Using Relays around NAT (TURN) depending on the network configuration and tries to give the WebRTC media the best possible chance to traverse these middle boxes. A TURN server is required to relay the WebRTC media between peers when STUN methods fail. In this work, we evaluate the different network configurations in which WebRTC peers can be present and how the WebRTC connectivity problem can be solved using the ICE framework, when peers are present in such network configurations. We also evaluate the TURN server for its computational, memory and bandwidth requirements for relaying different types of WebRTC calls.