The articles in this special issue focus on fundamental approaches to network coding in wireless communications systems. Wireless communication network providers are constantly striving for more efficient and reliable service provision to billions of customers across the globe. As such, there exist great opportunities in the research and development of advanced network coding techniques in emerging wireless communication systems and applications for further improving network capacity and performance. Arguably, bandwidth-hungry applications,such as multimedia, are to benefit the most from the many advantages that wireless network coding can offer, particularly higher throughputs, lower delays, and better scalability. Wireless network coding has a great potential to be applied at the physical layer, harnessing inherent interference in the wireless channel for more spectral efficiency. It can significantly enhance the performance of relay-based, device-to-device, and cooperative communication techniques in current and future wireless systems.

While wireless ad hoc networks have a wide range of applications in environment monitoring, military operations, and disaster recovery, etc, the full potential of such networks is inherently hindered by their diminishing capacity as the network size scales up. Content caching has been previously proposed to improve the availability of contents in a network and thus helps to alleviate the load on content custodians, reduce access latency, and improve the network capacity. This paper studies the scaling laws of the capacity of cache-enabled content distribution wireless ad hoc networks. We consider two basic content access schemes, namely, the Nearest Caching Node scheme where a request is satisfied by the nearest node to the requestor that has the content in its cache, and the Transparent Enroute Caching scheme where a content request is routed towards the content custodian and is satisfied by an intermediate node (or custodian) along the path that has the content in its cache. We first establish the capacity of content distribution wireless ad hoc networks without content caching as a baseline for investigating the benefit of caching. We then obtain the scaling laws of the capacity for the above two content access schemes with content caching. Based on the results we further explore their design and performance implications. Our results show that the capacity exhibits distinct scaling behaviors under different scaling regimes of the network parameters. Under certain conditions increasing the cache size of the nodes can effectively improve the capacity while under other conditions the improvement can be negligible. The characterizations of the capacity allows us to identify the bottleneck of the content access capacity for given network scenarios and choose effective approaches to improve the capacity.

Wireless researchers face the practical challenge of evaluating the performance of new routing protocols for very large-scale heterogeneous mobile systems. While simulation platforms, tools and techniques have matured, doubts remain about the broad validity of simulated results when applied to real world situations. Wireless network emulation provides a middle ground between simulations and real-world testing. However, emulation environments are complex to run and cost-prohibitive for larger-scale tests. In order to maximize fidelity while managing complexity and costs, we developed a common framework that unifies simulation-based analysis and emulation experiment tools, in order to produce calibrated and scalable performance evaluations.

We consider the problem of providing per-customer service guarantees in a high-speed packet switch typically situated at the edge between a set of customers and a service provider network. As basic requirements, the switch should be scalable to high speeds per port, a large number of ports, and a large number of customers (macroflows) with independent guarantees. Existing scalable solutions are based on virtual output queuing, which is computationally complex when required to provide service guarantees for a large number of macroflows. We present a novel architecture for packet switching that provides support for such service guarantees. A cost-effective fabric with small external speedup is combined with a feedback mechanism that enables the fabric to be virtually lossless, thus avoiding packet drops indiscriminate of macroflows' behavior. Through analysis and simulation, we show that this architecture provides accurate support for service guarantees, has low computational complexity, and is scalable to very high port speeds.

We survey advances in theories and models for Internet quality of service (QoS). We start with the theory of network calculus, which lays the foundation for support of deterministic performance guarantees in networks, and illustrate its applications to integrated services, differentiated services, and streaming media playback delays. We also present mechanisms and architecture for scalable support of guaranteed services in the Internet, based on the concept of a stateless core. Methods for scalable control operations are also discussed. We then turn our attention to statistical performance guarantees and describe several new probabilistic results that can be used for a statistical dimensioning of differentiated services. Lastly, we review proposals and results in supporting performance guarantees in a best effort context. These include models for elastic throughput guarantees based on TCP performance modeling, techniques for some QoS differentiation without access control, and methods that allow an application to control the performance it receives, in the absence of network support.

In this work, we investigate mechanisms for Internet congestion control in general, and random early detection (RED) in particular. We first study the current proposals for RED implementation and identify several structural problems such as producing large traffic oscillations and introducing unnecessary overhead in the fast path forwarding. We model RED as a feedback control system and discover fundamental laws governing the traffic dynamics in TCP/IP networks. Based on this understanding, we derive a set of recommendations for the architecture and implementation of congestion control modules in routers, such as RED.

The Differentiated services (diffserv) architecture has been proposed as a scalable solution for providing service differentiation among flows without any per-flow buffer management inside the core of the network. It has been advocated that it is feasible to provide service differentiation among a set of flows by choosing an appropriate “marking profile” for each flow. In this paper, we examine (i) whether it is possible to provide service differentiation among a set of TCP flows by choosing appropriate marking profiles for each flow, (ii) under what circumstances, the marking profiles are able to influence the service that a TCP flow receives, and, (iii) how to choose a correct profile to achieve a given service level. We derive a simple, and yet accurate, analytical model for determining the achieved rate of a TCP flow when edge-routers use “token bucket” packet marking and core-routers use active queue management for preferential packet dropping. From our study, we observe three important results: (i) the achieved rate is not proportional to the assured rate, (ii) it is not always possible to achieve the assured rate and, (iii) there exist ranges of values of the achieved rate for which token bucket parameters have no influence. We find that it is not easy to regulate the service level achieved by a TCP flow by solely setting the profile parameters. In addition, we derive conditions that determine when the bucket size influences the achieved rate, and rates that can be achieved and those that cannot. Our study provides insight for choosing appropriate token bucket parameters for the achievable rates.

The steady-state performance of a bulk transfer TCP flow (i.e., a flow with a large amount of data to send, such as FTP transfers) may be characterized by the send rate, which is the amount of data sent by the sender in unit time. In this paper we develop a simple analytic characterization of the steady-state send rate as a function of loss rate and round trip time (RTT) for a bulk transfer TCP flow. Unlike the models of Lakshman and Madhow (see IEE/ACM Trans. Networking, vol.5, p.336-50, 1997), Mahdavi and Floyd (1997), Mathis, Semke, Mahdavi and Ott (see Comput. Commun. Rev., vol.27, no.3, 1997) and by by Ott et al., our model captures not only the behavior of the fast retransmit mechanism but also the effect of the time-out mechanism. Our measurements suggest that this latter behavior is important from a modeling perspective, as almost all of our TCP traces contained more time-out events than fast retransmit events. Our measurements demonstrate that our model is able to more accurately predict TCP send rate and is accurate over a wider range of loss rates. We also present a simple extension of our model to compute the throughput of a bulk transfer TCP flow, which is defined as the amount of data received by the receiver in unit time.

In this paper we present methods to evaluate the benefit of using direct ATM connections (shortcuts) between IP nodes in IP over ATM networks, and we identify the combinations of IP and ATM network topologies where ATM shortcut benefits are likely to be high. We model an IP/ATM network with and without ATM shortcuts as two loss networks. We propose a metric for network performance comparison, the network load ratio, that gives the ratio of the number of flows accepted by two networks at the same network blocking probability. We derive an estimator of this metric, the asymptotic load ratio, that has low computational complexity. This estimator forms the basis of a methodology for network performance comparison. We use this method in simulation experiments using random networks. These experiments indicate that in many cases the utilization of an IP/ATM network increases proportionally to the decrease in the average path length when ATM shortcuts are used. We have also found that there is almost no correlation between the increase in network utilization (when using ATM shortcuts) and the IP to ATM node ratio.

In this paper we develop a simple analytic characterization of the steady state throughput, as a function of loss rate and round trip time for a bulk transfer TCP flow, i.e., a flow with an unlimited amount of data to send. Unlike the models in [6, 7, 10], our model captures not only the behavior of TCP's fast retransmit mechanism (which is also considered in [6, 7, 10]) but also the effect of TCP's timeout mechanism on throughput. Our measurements suggest that this latter behavior is important from a modeling perspective, as almost all of our TCP traces contained more time-out events than fast retransmit events. Our measurements demonstrate that our model is able to more accurately predict TCP throughput and is accurate over a wider range of loss rates.

We present algorithms for flow admission control at an earliest deadline first link scheduler when the flows are characterized by piecewise linear traffic envelopes. We show that the algorithms have very low computational complexity and, thus, practical applicability. The complexity can be further decreased by introducing the notion of discretized admission control. Through discretization, the range of positions for the end points of linear segments of the traffic envelopes is restricted to a finite set. Simulation experiments show that discretized admission control can lead to two orders of magnitude decrease in the amount of computation needed to make admission control decisions over that incurred when using exact (nondiscrete) admission control, with the additional benefit that this amount of computation no longer depends on the number of flows. We examine the relative performance degradation (in terms of the number of flows admitted) incurred by the discretization and find that it is small.

We present algorithms for flow admission control at an EDF link scheduler when the flows are characterized by peak rate, average rate and burst size. We show that the algorithms have very low computational complexity and are easily applicable in practice. The complexity can be further decreased by introducing the notion of flex classes. We evaluate the penalty in efficiency that the classes incur to the EDF scheduler. We find that this efficiency degradation can be made arbitrarily small and is acceptable even for a small number of classes.

Many multicast applications, including audio and video, require quality of service (QoS) guarantees from the network. Hence, multicast admission control and resource reservation procedures will be needed. In this paper we present a general framework for admission control and resource reservation for multicast sessions. Within this framework, efficient and practical algorithms that aim to efficiently utilize network resources are developed. The problem of admission control is decomposed into several subproblems that include: the division of end-to-end QoS requirements into local QoS requirements, the mapping of local QoS requirements into resource requirements, and the reclaiming of the resources allocated in excess. These are solved independently of each other yielding a set of mechanisms and policies that can be used to provide admission control and resource reservation for multicast connection establishment. The resource allocation algorithms we consider specifically accommodate receiver heterogeneity (in both end-to-end and per-hop QoS requirements) by reserving necessary and sufficient resources for a multicast session. An application of these algorithms in the context of packetized voice multicast connections over the Mbone is provided to illustrate their applicability.