Transport and Display Mechanisms for Multimedia Conferencing Across Packet-Switched Networks

A transport protocol that supports real-time communication of audio/video frames across campus-area packet switched networks is presented. It is a “best effort” protocol that attempts to ameliorate the effect of jitter, load variation, and packet loss, to provide low latency, synchronized audio and video communications. This goal is realized through four transport and display mechanisms, and a real-time implementation of these mechanisms that integrates operating system services (e.g., scheduling and resource allocation, and device management) with network communication services (e.g., transport protocols), and with application code (e.g., display routines). The four mechanisms are: a facility for varying the synchronization between audio and video to achieve continuous audio in the face of jitter, a network congestion monitoring mechanism that is used to control audio/video latency, a queueing mechanism at the sender that is used to maximize frame throughput without unnecessarily increasing latency, and a forward error correction mechanism for transmitting audio frames multiple times to ameliorate the effects of packet loss in the network. The effectiveness of these techniques is demonstrated by measuring the performance of the protocol when transmitting audio and video across congested networks. Published in: Computer Networks and ISDN Systems, Vol. 26, No. 10, (July 1994) pp. 1281-1304. * This work supported in parts by grants from the National Science Foundation (numbers CCR-9110938 and ICI9015443), and by the Digital Equipment Corporation and the IBM Corporation.