This paper presents the design, implementation and performance of a reliable multicast transport protocol called RMTP. RMTP is based on a hierarchical structure in which receivers are grouped into local regions or domains and in each domain there is a special receiver called a Designated Receiver (DR) which is responsible for sending acknowledgments periodically to the sender, for processing acknowledgements from receivers in its domain and for retransmitting lost packets to the corresponding receivers. Since lost packets are recovered by local retransmissions as opposed to retransmissions from the original sender, end-to-end latency is significantly reduced, and the overall throughput is improved as well. Also, since only the DRs send their acknowledgments to the sender, instead of all receivers sending their acknowledgments to the sender, a single acknowledgement is generated per local region, and this prevents acknowledgement implosion. Receivers in RMTP send their acknowledgments to the DRs periodically, thereby simplifying error recovery. In addition, lost packets are recovered by selective repeat retransmissions, leading to improved throughput at the cost of minimal additional buffering at the receivers. This paper also describes the implementation of RMTP and its performance on the Internet.

Sender-initiated reliable multicast protocols based on the use of positive acknowledgments (ACKs) can suffer performance degradation as the number of receivers increases. This degradation is due to the fact that the sender must bear much of the complexity associated with reliable data transfer (e.g., maintaining state information and timers for each of the receivers and responding to receivers' ACKs). A potential solution to this problem is to shift the burden of providing reliable data transfer to the receivers - thus resulting in receiver-initiated multicast error control protocols based on the use of negative acknowledgments (NAKs). In this paper we determine the maximum throughputs for generic senderinitiated and receiver-initiated protocols for two classes of applications: (i) one-many applications where one participant sends data to a set of receivers, and (ii) many-many applications where all participants simultaneously send and receive data to/from each other. We show that a receiver-initiated error control protocol which requires receivers to transmit NAKs point-to-point to the sender provides higher throughput than a sender-initiated counterpart for both classes of applications. We further demonstrate that, in the case of a one-many application, replacing point-to-point transfer of NAKs with multicasting of NAKs coupled with a random backoff procedure provides a substantial additional increase in the throughput of a receiver-initiated error control protocol over a sender-initiated protocol. We also find, however, that such a modification leads to a throughput degradation in the case of many-many applications.

The recent success of multicast applications such as Internet teleconferencing illustrates the tremendous potential of applications built upon wide-area multicast communication services. A critical issue for such multicast applications and the higher layer protocols required to support them is the manner in which packet losses occur within the multicast network. In this paper we present and analyze packet loss data collected on multicast-capable hosts at 17 geographically distinct locations in Europe and the US and connected via the MBone. We experimentally and quantitatively examine the spatial and temporal correlation in packet loss among participants in a multicast session. Our results show that there is some spatial correlation in loss among the multicast sites. However, the shared loss in the backbone of the MBone is, for the most part, low. We find a fairly significant amount of of burst loss (consecutive losses) at most sites. In every dataset, at least one receiver experienced ...

PROTOCOL AND REAL-TIME SCHEDULING ISSUES FOR MULTIMEDIA APPLICATIONS September 1994 Sridhar Pingali, B.Tech., Indian Institute of Technology M.S., Clemson University Ph.D., University of Massachusetts Amherst Directed by: Professor James F. Kurose The growth of higher bandwidth networks and powerful new workstations has enabled the development of many new multimedia applications. These applications involve the combined use of different media such as voice, video and text. Supporting these applications requires the resolution of a number of issues both within the network and at the end-hosts that originate the applications. In this dissertation we consider real-time scheduling and protocol problems that arise in this arena. In the realm of applications operating under hard time-constraints, we address both network-node and workstation scheduling. In applications involving the use of video or a combination of voice and video, there can be the need to schedule classes of traffic with d...