An empirical study of packet voice distribution over a campus-wide network

Abstract

Support for digital audio and video streams (continuous media) in desktop computers is increasingly widespread and hence the demand for high-quality network distribution of continuous media streams is growing. While current local area and campus-wide networks have high-bandwidth links and powerful routers, the delay sensitivities of continuous media streams pose a challenge. Our paper provides an empirical investigation of the feasibility of transmitting real-time packet streams across a large extant campus network, the University of Virginia campus network. This network is representative of contemporary large enterprise networks consisting of Ethernet segments connected by high-performance routers to high-speed back-bones, e.g., FDDI rings. We focus on interactive packet voice. Delay and loss measurements are obtained for packet voice streams transmitted over three paths through the network. These measurements are used to examine the voice protocol parameters for this environment, including a novel retransmission-based error control proposed by Dempsey (1994).<>