Next-generation wireless networks such as LTE and WiMax can achieve throughputs of several Mbps with TCP. These higher throughputs, however, can easily be destroyed by frequent handoffs, which occur in urban environments due to shadowing. A primary reason for the throughput drop during handoffs is the out of order arrival of packets at the receiver. As a result, in this paper, we model the precise effect of packet-reordering on the goodput of TCP NewReno. Specifically, we develop a TCP NewReno model that captures the goodput of TCP as a function of round-trip time, average time duration between packet-reorder events, average number of packets reordered during every reorder event, and the congestion window threshold of TCP NewReno. We also developed an emulator that runs on a router to implement packet reordering events from time to time. We validate our NewReno model by comparing the goodput results obtained by transferring data between two hosts connected via the emulator to the goodput results that our model predicts.

A transmission and multiplexing strategy appropriate for voice over asynchronous transfer mode (ATM), called delayed frame queueing (DFQ), is proposed. This frame-based strategy has features in common with the synchronous transfer mode and is thus well suited to service synchronous applications such as voice, while retaining the statistical multiplexing capabilities of ATM. In particular, the DFQ service discipline can provide explicit and nontrivial bounds for queue delay and jitter, for both bursty as well as continuous traffic streams. Furthermore, the DFQ discipline can combine a wide range of delay and jitter bounds while also managing the distribution of quality of service violations among the traffic streams when congestion occurs. Jitter control is performed at the network periphery and thus does not negatively influence multiplexing gain at intermediate nodes. This efficient strategy has major implications in terms of the preferred alternatives chosen by clients when implementing source clock recovery for voice. DFQ allows the entire range of implementation alternatives for voice over ATM to be appropriately serviced, such as ATM adaptation layer types 1 and 2 (AAL1/2), adaptive playout, and immediate playout.

We propose a simple first-in first-out (FIFO)-based service protocol which is appropriate for a multimedia ATM satellite system. The main area of interest is to provide real-time traffic with upper bounds on the end-to-end delay, jitter, and loss experienced at various service queues within a satellite network. Various service protocols, each based on a common underlying strategy, are developed in light of the requirements and limitations imposed at each of the satellite's subsystems. These subsystems include the uplink (UL) earth station (ES) service queue, on-board processing (OBP) queues, and the downlink (DL) ES service queue feeding into a wireline ATM network or directly to an end-user application. Numerous network simulation results demonstrate the tractability, efficiency, and versatility of the underlying service discipline. Key features of our strategy are its algorithmic and architectural simplicity, its non-ad-hoc scheduling approach, and its unified treatment of all real-time streams at all service queues. In addition, the delay and jitter bounds are uncoupled. In this way, end-to-end jitter can be tightly controlled even if medium access requires long indeterminate waiting durations.

A generic satellite interworking unit (IWU) is proposed for delivering real-time traffic across an arbitrary satellite bridge employing asynchronous transfer mode (ATM). End-to-end cell transfer delay (CTD) and cell delay variation (CDV) bounds are provided by logically associating cells to short duration link-level frames. In addition, a generic scheduler is proposed for the uplink earth station (ES). This scheduler only defines the service priorities and deadlines of queued cells but does not define the particular fashion in which an ES gains access to the shared transmission medium. Simulation results demonstrate that the quality of service (QoS) can be efficiently managed in order to maximize satellite throughput of real-time traffic.

In traditional precedence-constrained scheduling a task is ready to execute when all its predecessors are complete. We call such a task an AND task. In this paper we allow certain tasks to be ready when just one of their predecessors is complete. These tasks are known as OR tasks. We analyze the complexity of two types of real-time AND/OR task scheduling problems. In the first type of problem, all the predecessors of every OR task must eventually be completed, but in the second type of problem, some OR predecessors may be left unscheduled. We show that most problems involving tasks with individual deadlines are NP-complete, and then present two priority-driven heuristic algorithms to minimize completion time on a multiprocessor. These algorithms provide the same level of worst-case performance as some previous priority-driven algorithms for scheduling AND-only task systems.

An examination is made of the worst-case performance of a class of scheduling algorithms commonly known as priority-driven or list-scheduling algorithms. These algorithms have anomalous, unpredictable performance when used to suboptimally schedule nonpreemptive tasks with precedence constraints. A general method is presented for deriving the worst-case performance of these algorithms. This method is easy to use, yet powerful enough to yield tight performance bounds for many classes of scheduling problems. It is demonstrated that the method has wide applicability. Several task systems are presented for which list-scheduling algorithms have worst-case performance, and the general characteristics of these task systems are discussed. It is believed that these task systems are sometimes ignored in simulation studies; consequently, the results of these studies may be overly optimistic.>