Heterogeneous multicast (HMC) is a communication paradigm for distributing real-time streams among multiple session participants that have different constraints and reception preferences. In HMC, sources offer streams to the participants, and each participant specifies individualized fidelity level for each stream he/she wants to receive. Participants can change their specification in mid-session. The approach is based on sources encoding streams in layers which can be individually ordered. The network provides transport that allows ordering and selective distribution of individual layers. ATM infrastructure is used for connection-based transport and cell switching. However, ATM signalling is enhanced to enable the establishment and management of multipoint, multi-VC connections for carrying multiple layers. The ATM-based network module also supports admission control, and QOS-based multipoint routes.

Emerging networking technologies are being introduced with their own management protocols for routing, resource reservation and signaling. This diversity restricts the interoperability among QOS-aware multimedia applications written for different networks. We present an approach for managing multiparty, multimedia sessions in a heterogeneous internetwork spanning multiple signaling domains. Participants utilize the native signaling on their respective domains and interact with participants on other domains through signaling gateways that bridge the domains and provide translation of signaling procedures and QOS semantics. Data streams are transmitted using a hierarchical representation, which allows participants to independently adjust the reception quality of each stream according to their resources and interests. We present a particular design and implementation details for connecting ATM and IP signaling domains and conclude with its extension to an arbitrary number of interconnected domains.

Real-time multicast communication is considered, in which each destination makes an individual bid for delivery of a subset of real-time hierarchically encoded streams that are offered to the session by the source. The objective is to customize stream delivery to destinations, based on their requests and network constraints. Admission control procedures, which consider requests for multiple streams from multiple destinations and resolve contention when users' requests exceed available network resources, are presented. Specifications for four admission control procedures are presented and their properties analyzed. A performance measure for admission control is defined to be the total gain for the source that equals the sum of bids by the users taking delivery of the streams they ordered. Simulation results of the performance of the procedures are presented and compared to the performance of an optimal, but highly complex, admission control algorithm. In this paper the assumption is that the routing tree is already in place and the focus is on managing users' requests and deciding on which stream to deliver to each destination. This is done by signaling protocols that transfer messages containing users' requests and resource allocation commands between a source and the destinations. In addition, admission control protocols take users' requests and network resources as their input and issue as outputs the decisions on which request to admit (and allocate resources for) and which to reject.

Multicast is rapidly becoming an important mode of communication as well as a good platform for building group-oriented services. However, to be used for trusted communication, current multicast schemes must be supplemented by mechanisms for protecting traffic, controlling participation, and restricting access of unauthorized users to the data exchanged by the participants. We consider fundamental security issues in building a trusted multicast facility. We discuss techniques for group-based data encryption, authentication of participants, and preventing unauthorized transmissions and receptions.>

Flexibility and efficiency of multipoint, multimedia communications will be greatly enhance if participants get delivery of only the streams, each at a specified level of quality, each user wishes to receive. The authors investigate the problem of optimal distribution of multiple streams from a source to a set of destinations, subject to the destinations' requirements and the limitations on available link capacities. Users' requests are expressed by means of bids, which are non-negative values assigned by each destination to each offered stream. Assuming that the source gains an amount equal to a user's bid whenever a stream is delivered to that user, we seek distributions that maximize the sources' gain. The optimal assignment of streams to the links of a multicast tree is formulated as an integer programming problem, and an algorithm for determining the optimal assignment is presented. The algorithm consists of three procedures: listing candidate assignments on a single link, listing candidate assignments on tree paths, and evaluating the gain for each assignment. The algorithm's correctness is proved, and its computational complexity is ascertained.>

Distributed algorithms for finding two disjoint paths of minimum total length from each node to a destination are presented. The algorithms have both node-disjoint and link-disjoint versions and provide each node with information sufficient to forward data on the disjoint paths. It is shown that this problem can be reduced to the problem of finding a minimal shortest path from each node to the destination in a modified network, and a distributed algorithm on the original network that simulates a shortest-paths algorithm running on the modified network is presented. The algorithm has a smaller space complexity than any previous distributed algorithm for the same problem, and a method for forwarding packets is presented that does not require any additional space complexity. A synchronous implementation of the algorithm is also presented and studied.>

A novel multipoint communication paradigm, in which each destination receives a subset of the source's signal that corresponds to that destination's terminal and access bandwidth constraints, is presented. The approach to realizing this paradigm is based on integration of layered coding of the source's signal, routing based on bandwidth demand, optimization of signal parameters, and layered error control. The author gives an overview of several hierarchical signal coding techniques; presents methods for finding maximum bandwidth available to destination, establishing maximum-bandwidth routes; and optimally assigns bandwidth to the signal layers to maximize overall reception quality. Error control procedures whereby the network, source, and destinations cooperate to maintain layered-based data integrity, using erasure recovery coding and prioritized packet detection are also presented.>

A communications system in which multiple parallel channels are available to carry traffic from a transmitter to a receiver is considered, and an extension of the selective-repeat automatic repeat request (SR-ARQ) protocol that dynamically assigns packets to channels for each (re)transmission is presented. Because of selective retransmission, packets arrive at the receiver out of order and must be stored in a resequencing buffer. A queuing model for the resequencing buffer is constructed. The generating function of the buffer occupancy and the packet-delay distribution are derived, and procedures for simplifying the computation are presented. The dynamic assignment scheme is compared with, and shown to have performance superior to, a static assignment scheme.>

The authors consider a selective repeat (SR) automatic-repeat-request (ARQ) protocol with one source and multiple receivers. Each receiver acknowledges all packets and handles its resequencing buffer based only on the packets that it receives error-free. An analysis of the resequencing delay and buffer occupancy at a receiver is presented. The authors construct a model that makes it possible to derive steady-state results, taking into consideration such system parameters as number of receivers, propagation delay, packet error probabilities, and acknowledgments. The authors focus on two measures of occupancy. The first measure corresponds to the number of packets waiting to be resequenced whereas the second measure includes, in addition, the buffer space reserved for packets that cause resequencing delays. The main results are the distribution of the resequencing delay and the distribution of the number of packets occupying the receiver's buffer.>

A technique for fiber-optic networks based on forward-error correction (FEC) that allows the destination to reconstruct missing data packets by using redundant parity packets that the source adds to each block of data packets is presented. Methods for generating several types of parity packets are described, along with decoding techniques and their implementations. Algorithms are presented for packet interleaving and selective rejection of packets from node buffers, both of which disperse missing packets among many blocks, thereby reducing the required coding complexity. Performance evaluation, by both analytic and simulation models, shows that this technique can result in a reduction of up to three orders of magnitude in the packet loss rate.>

A distributed asynchronous algorithm is presented for finding two disjoint paths of minimum total length from each possible source node to a destination. The algorithm has both node-disjoint and link-disjoint versions, and provides each node with information sufficient to make incremental routing decisions for forwarding packets over the disjoint paths. For a network in which all links have unit length, it is shown that a synchronous implementation of the algorithm has communication and time complexities O( mod E mod +D mod N mod ) and O(D/sub 2/), respectively, where D is the networks diameter and D/sub 2/ is the maximum, over all nodes i, of the total number of links in the shortest pair of disjoint paths from i to the destination.>

Consider a communication network that regulates retransmissions of erroneous packets by a selective-repeat (SR) automatic repeat request (ARQ) protocol. Packets are assigned consecutive integers, and the transmitter continuously transmits them in order until a negative acknowledgement or a time-out is observed. The receiver, upon receipt of a packet, checks for errors and returns positive/negative acknowledgement (ACK/NACK) accordingly. Only packets for which either NACK or time-out have been observed are retransmitted. Under SR ARQ, the receiver accepts packets that are out of order and must store them temporarily if it has to deliver them in sequence. The resequencing buffer requirements and the resulting packet delay constitute major factors in overall system considerations. The authors derive the distributions of the buffer occupancy and the resequencing delay at the receiver under a heavy traffic situation. This enables the network designer to determine how much buffer capacity at the receiver guarantees certain specified performance measures.>

The authors consider an SR ARQ (selective-repeat automatic-repeat-request) protocol with the one source and multiple receivers, where each receiver acknowledges all packets and handles its resequencing buffer based only on the packets that it receives error-free. An analysis of the resequencing delay and buffer occupancy at a receiver is presented. They construct a model that allows them to derive steady-state results and that makes into consideration system parameters such as number of receivers, propagation delay, and error probabilities of packets and acknowledgements. The main results of the analysis are the distribution of the resequencing delay and the distribution of the number of packets occupying the receiver's buffer. They also derive simple expressions for the mean buffer occupancy at the limit as the packet error probability tends to one. Numerical results are provided to illustrate the effects of system parameters on the behavior of the expected resequencing delay and the expected buffer occupancy.>

A distributed algorithm to reconfigure a double-loop token-ring local area network following topological changes is presented. Each node, upon detection of a broken link, attempts to use another link, and when this is not possible it connects its input and output lines to form a loopback. The nodes communicate only with their neighbors and base their actions on the messages they receive and on their local observations. The algorithm guarantees that at all times the network is organized to provide maximum possible connectivity among the nodes-either one loop that encompasses all nodes in the network or several subloops, each of which operates as a separate token-ring network. Tokens are generated or eliminated as necessary to result in one and only one token in each (sub)loop. The algorithm is formally specified and its properties are verified.>

Packet radio networks that employ several parallel multiple-access channels are considered. An architecture for such a network dictates the selection of channels for packet transmission. We propose and analyze two multichannel architectures. In the first, each node employs a single radio and is assigned a channel on which it listens when it does not transmit. To transmit a packet, the node tunes its radio to the channel of the intended receiver for that transmission only. The second architecture requires each radio to use a single channel for both transmission and reception, but provides some of the nodes with more than one radio each, allowing them to serve as bridges between channels. Within these architectures, one can further select the amount of routing information held by each node and the channel-access protocol, both of which greatly affect the network performance. To ascertain the effects of the various parameters, we calculate the throughput in both architectures. The channel-access protocols we consider are slotted ALOHA and CSMA with and without capture. We also evaluate the effect of increasing the amount of routing information held by the nodes.

The technology of packet switching over multihop, multiple-access channels has evolved to the point at which its protocols can now support internetwork operation of medium-size networks whose nodes possess some degree of mobility. As regards the needs and challenges of the future operating environment, it is clear that these can be met only by enhancing the packet radio architecture and its protocols. We discuss several enhancements that allow the organization of large, dynamic networks that can operate over multiple channels, adapt to varying conditions, and possess self-monitoring and self-control capabilities. As these areas are examined, the attendant issues and tradeoffs are discussed; in addition, some protocols and information regarding their performance are presented.

No abstract available.

The operation of a sequential decoder in a packet-switching environment is considered. Packets arrive randomly at the decoder, and a packet is stored in a buffer if the decoder is busy upon its arrival. The decoder devotes no more than a time-out period of predetermined length to the decoding of any single packet. If packet decoding is completed within that period, the packet leaves the system. Otherwise, it is retransmitted and its decoding starts anew. While a packet is retransmitted, the decoder decodes another packet that resides in its buffer. An upper bound on the maximum rate of packets that can be supported by the channel-decoder combination is derived, and the optimum time-out that maximizes that rate is determined. A discrete-time model of the decoder's queue is presented, and the average queue length and throughput are evaluated.

A research effort to provide speech-carrying capabilities to a data-oriented packet-switching radio network is described. The features of the network that limit its ability to carry packetized speech are discussed, and their effects on the network performance are analyzed. A new protocol, called duct routing, that enhances the network capabilities in a mobile environment is presented. That protocol makes use of repeater redundancy to compensate for loss of communication connectivity due to node mobility. A series of experiments to evaluate the network performance in carrying speech traffic, both with data and voice protocols, is described, and the results are presented and discussed.

A protocol that provides multiple access with two levels of priority to a common radio channel is described. A probabilistic model of the protocol is presented and its analysis yields the throughput-delay characteristics of the two classes of packets supported by that protocol, along with their interaction. The model is analyzed by a method similar to the one used in priority queues; this method allows consideration of each class separately, thus reducing the computational effort considerably.

The stop-and-wait ARQ protocol with convolutional code and sequential decoding is considered. The channel throughput and optimum time-out before retransmission are obtained and the effects of the link parameters discussed. The results are extended to a twohop radio link, where a repeater receives the source's data and forwards them on the same radio channel to the destination. The ARQ protocols for the source and the repeater are discussed and a method presented for choosing the optimum time-out intervals for both.

Orthogonal quantization is a partition of signal space that is achieved by independent quantization of each of itsMorthogonal axes. A closed form expression is derived for the quantized channel cutoff rate and for the optimum orthogonal quantization similar to the one that has been derived for binary signaling. While orthogonal quantization is natural for communication systems in which the transmitted signals are themselves orthogonal, it can also be profitably applied to other signals, e.g., a simplex set in a lower dimensional space. Though orthogonal quantization is inferior to optimal quantization, it is essentially simpler and does not incur great loss in performance. A numerical example illustrates the relative merits of optimal and orthogonal quantization for the simplex set in the plane.