Abstract

Current packet-switched networks such as the Internet do not provide guaranteed performance measures such as maximum loss rate or delay jitter. One way to support packet audio in these networks is to use control mechanisms which adapt the audio coding and decoding processes based on the state of the network so as to maximize the quality of the audio delivered to the destinations. This quality essentially depends on the characteristics of the loss and the delay jitter processes in the network. In this paper, we describe and analyze a set of efficient control mechanisms which attempt to minimize the impact of these processes. The mechanisms include a a jitter control mechanism and a combined error and rate control mechanism. These mechanisms have been implemented and evaluated over the Internet and the MBone. Experiments indicate that they make it possible to establish and maintain good quality audioconferences even across fairly congested connections.

Citations