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Showing papers in "IEEE ACM Transactions on Networking in 2003"


Journal ArticleDOI
TL;DR: Results from theoretical analysis and simulations show that Chord is scalable: Communication cost and the state maintained by each node scale logarithmically with the number of Chord nodes.
Abstract: A fundamental problem that confronts peer-to-peer applications is the efficient location of the node that stores a desired data item. This paper presents Chord, a distributed lookup protocol that addresses this problem. Chord provides support for just one operation: given a key, it maps the key onto a node. Data location can be easily implemented on top of Chord by associating a key with each data item, and storing the key/data pair at the node to which the key maps. Chord adapts efficiently as nodes join and leave the system, and can answer queries even if the system is continuously changing. Results from theoretical analysis and simulations show that Chord is scalable: Communication cost and the state maintained by each node scale logarithmically with the number of Chord nodes.

3,518 citations


Journal ArticleDOI
TL;DR: For the multicast setup it is proved that there exist coding strategies that provide maximally robust networks and that do not require adaptation of the network interior to the failure pattern in question.
Abstract: We take a new look at the issue of network capacity. It is shown that network coding is an essential ingredient in achieving the capacity of a network. Building on recent work by Li et al.(see Proc. 2001 IEEE Int. Symp. Information Theory, p.102), who examined the network capacity of multicast networks, we extend the network coding framework to arbitrary networks and robust networking. For networks which are restricted to using linear network codes, we find necessary and sufficient conditions for the feasibility of any given set of connections over a given network. We also consider the problem of network recovery for nonergodic link failures. For the multicast setup we prove that there exist coding strategies that provide maximally robust networks and that do not require adaptation of the network interior to the failure pattern in question. The results are derived for both delay-free networks and networks with delays.

2,628 citations


Journal ArticleDOI
TL;DR: In this article, the authors explore and evaluate the use of directed diffusion for a simple remote-surveillance sensor network analytically and experimentally and demonstrate that directed diffusion can achieve significant energy savings and can outperform idealized traditional schemes under the investigated scenarios.
Abstract: Advances in processor, memory, and radio technology will enable small and cheap nodes capable of sensing, communication, and computation. Networks of such nodes can coordinate to perform distributed sensing of environmental phenomena. In this paper, we explore the directed-diffusion paradigm for such coordination. Directed diffusion is data-centric in that all communication is for named data. All nodes in a directed-diffusion-based network are application aware. This enables diffusion to achieve energy savings by selecting empirically good paths and by caching and processing data in-network (e.g., data aggregation). We explore and evaluate the use of directed diffusion for a simple remote-surveillance sensor network analytically and experimentally. Our evaluation indicates that directed diffusion can achieve significant energy savings and can outperform idealized traditional schemes (e.g., omniscient multicast) under the investigated scenarios.

2,550 citations


Journal ArticleDOI
TL;DR: This work describes an end-to-end methodology, called self-loading periodic streams (SLoPS), for measuring avail-bw, and uses pathload, a nonintrusive tool, to evaluate the variability ("dynamics") of the avail- bw in Internet paths.
Abstract: The available bandwidth (avail-bw) in a network path is of major importance in congestion control, streaming applications, quality-of-service verification, server selection, and overlay networks. We describe an end-to-end methodology, called self-loading periodic streams (SLoPS), for measuring avail-bw. The basic idea in SLoPS is that the one-way delays of a periodic packet stream show an increasing trend when the stream's rate is higher than the avail-bw. We have implemented SLoPS in a tool called pathload. The accuracy of the tool has been evaluated with both simulations and experiments over real-world Internet paths. Pathload is nonintrusive, meaning that it does not cause significant increases in the network utilization, delays, or losses. We used pathload to evaluate the variability ("dynamics") of the avail-bw in Internet paths. The avail-bw becomes significantly more variable in heavily utilized paths, as well as in paths with limited capacity (probably due to a lower degree of statistical multiplexing). We finally examine the relation between avail-bw and TCP throughput. A persistent TCP connection can be used to measure roughly the avail-bw in a path, but TCP saturates the path and increases significantly the path delays and jitter.

765 citations


Journal ArticleDOI
TL;DR: A duality model of end-to-end congestion control is proposed and applied to understand the equilibrium properties of TCP and active queue management schemes to maximize aggregate utility subject to capacity constraints.
Abstract: We propose a duality model of end-to-end congestion control and apply it to understanding the equilibrium properties of TCP and active queue management schemes. The basic idea is to regard source rates as primal variables and congestion measures as dual variables, and congestion control as a distributed primal-dual algorithm over the Internet to maximize aggregate utility subject to capacity constraints. The primal iteration is carried out by TCP algorithms such as Reno or Vegas, and the dual iteration is carried out by queue management algorithms such as DropTail, RED or REM. We present these algorithms and their generalizations, derive their utility functions, and study their interaction.

701 citations


Journal ArticleDOI
TL;DR: A framework for designing end-to-end congestion control schemes in a network where each user may have a different utility function and may experience noncongestion-related losses is presented and ECN marking levels can be designed to nearly eliminate losses in the network.
Abstract: We present a framework for designing end-to-end congestion control schemes in a network where each user may have a different utility function and may experience noncongestion-related losses. We first show that there exists an additive-increase-multiplicative-decrease scheme using only end-to-end measurable losses such that a socially optimal solution can be reached. We incorporate round-trip delay in this model, and show that one can generalize observations regarding TCP-type congestion avoidance to more general window flow control schemes. We then consider explicit congestion notification (ECN) as an alternate mechanism (instead of losses) for signaling congestion and show that ECN marking levels can be designed to nearly eliminate losses in the network by choosing the marking level independently for each node in the network. While the ECN marking level at each node may depend on the number of flows through the node, the appropriate marking level can be estimated using only aggregate flow measurements, i.e., per-flow measurements are not required.

529 citations


Journal ArticleDOI
TL;DR: This paper presents a power-control framework called utility-based power control (UBPC) by reformulating the problem using a softened SIR requirement (utility) and adding a penalty on power consumption (cost).
Abstract: Distributed power-control algorithms for systems with hard signal-to-interference ratio (SIR) constraints may diverge when infeasibility arises. In this paper, we present a power-control framework called utility-based power control (UBPC) by reformulating the problem using a softened SIR requirement (utility) and adding a penalty on power consumption (cost). Under this framework, the goal is to maximize the net utility, defined as utility minus cost. Although UBPC is still noncooperative and distributed in nature, some degree of cooperation emerges: a user will automatically decrease its target SIR (and may even turn off transmission) when it senses that traffic congestion is building up. This framework enables us to improve system convergence and to satisfy heterogeneous service requirements (such as delay and bit error rate) for integrated networks with both voice users and data users. Fairness, adaptiveness, and a high degree of flexibility can be achieved by properly tuning parameters in UBPC.

505 citations


Journal ArticleDOI
TL;DR: This paper proposes an architecture that significantly reduces this implementation complexity yet still achieves approximately fair bandwidth allocations, called Core-Stateless Fair Queueing, and presents simulations and analysis on the performance.
Abstract: Router mechanisms designed to achieve fair bandwidth allocations, such as Fair Queueing, have many desirable properties for congestion control in the Internet. However, such mechanisms usually need to maintain state, manage buffers, and/or perform packet scheduling on a per-flow basis, and this complexity may prevent them from being cost-effectively implemented and widely deployed. In this paper, we propose an architecture that significantly reduces this implementation complexity yet still achieves approximately fair bandwidth allocations. We apply this approach to an island of routers--that is, a contiguous region of the network--and we distinguish between edge routers and core routers. Edge routers maintain per-flow state; they estimate the incoming rate of each flow and insert a label into each packet based on this estimate. Core routers maintain no per-flow state; they use first-in-first-out packet scheduling augmented by a probabilistic dropping algorithm that uses the packet labels and an estimate of the aggregate traffic at the router. We call the scheme Core-Stateless Fair Queueing. We present simulations and analysis on the performance of this approach.

428 citations


Journal ArticleDOI
TL;DR: It is shown that the topology can be described efficiently with power laws and that the power laws hold even in the most recent and more complete topology with correlation coefficient above 99% for the degree-based power law.
Abstract: In this paper, we study and characterize the topology of the Internet at the autonomous system (AS) level. First, we show that the topology can be described efficiently with power laws. The elegance and simplicity of the power laws provide a novel perspective into the seemingly uncontrolled Internet structure. Second, we show that power laws have appeared consistently over the last five years. We also observe that the power laws hold even in the most recent and more complete topology with correlation coefficient above 99% for the degree-based power law. In addition, we study the evolution of the power-law exponents over the five-year interval and observe a variation for the degree-based power law of less than 10%. Third, we provide relationships between the exponents and other topological metrics.

414 citations


Journal ArticleDOI
TL;DR: ZigZag and the hybrid algorithm are the fairest among all LDAs, and all of the LDAs are reasonably fair when competing with TCP, and their fairness among flows using the same LDA depends on the network topology.
Abstract: In this paper, we explore end-to-end loss differentiation algorithms (LDAs) for use with congestion-sensitive video transport protocols for networks with either backbone or last-hop wireless links. As our basic video transport protocol, we use UDP in conjunction with a congestion control mechanism extended with an LDA. For congestion control, we use the TCP-Friendly Rate Control (TFRC) algorithm. We extend TFRC to use an LDA when a connection uses at least one wireless link in the path between the sender and receiver. We then evaluate various LDAs under different wireless network topologies, competing traffic, and fairness scenarios to determine their effectiveness. In addition to evaluating LDAs derived from previous work, we also propose and evaluate a new LDA, ZigZag, and a hybrid LDA, ZBS, that selects among base LDAs depending upon observed network conditions. We evaluate these LDAs via simulation, and find that no single base algorithm performs well across all topologies and competition. However, the hybrid algorithm performs well across topologies and competition, and in some cases exceeds the performance of the best base LDA for a given scenario. All of the LDAs are reasonably fair when competing with TCP, and their fairness among flows using the same LDA depends on the network topology. In general, ZigZag and the hybrid algorithm are the fairest among all LDAs.

371 citations


Journal ArticleDOI
TL;DR: A new generic graph model for traffic grooming in heterogeneous WDM mesh networks, based on the auxiliary graph, is proposed which can achieve various objectives using different grooming policies, while taking into account various constraints such as transceivers, wavelengths, wavelength-conversion capabilities, and grooming capabilities.
Abstract: As the operation of our fiber-optic backbone networks migrates from interconnected SONET rings to arbitrary mesh topology, traffic grooming on wavelength-division multiplexing (WDM) mesh networks becomes an extremely important research problem. To address this problem, we propose a new generic graph model for traffic grooming in heterogeneous WDM mesh networks. The novelty of our model is that, by only manipulating the edges of the auxiliary graph created by our model and the weights of these edges, our model can achieve various objectives using different grooming policies, while taking into account various constraints such as transceivers, wavelengths, wavelength-conversion capabilities, and grooming capabilities. Based on the auxiliary graph, we develop an integrated traffic-grooming algorithm (IGABAG) and an integrated grooming procedure (INGPROC) which jointly solve several traffic-grooming subproblems by simply applying the shortest-path computation method. Different grooming policies can be represented by different weight-assignment functions, and the performance of these grooming policies are compared under both nonblocking scenario and blocking scenario. The IGABAG can be applied to both static and dynamic traffic grooming. In static grooming, the traffic-selection scheme is key to achieving good network performance. We propose several traffic-selection schemes based on this model and we evaluate their performance for different network topologies.

Journal ArticleDOI
TL;DR: A mobile tracking scheme that exploits the predictability of user mobility patterns in wireless PCS networks, and proposes a dynamic Gauss-Markov parameter estimator that provides the mobility parameters to the prediction algorithm.
Abstract: This paper presents a mobile tracking scheme that exploits the predictability of user mobility patterns in wireless PCS networks. In this scheme, a mobile's future location is predicted by the network, based on the information gathered from the mobile's recent report of location and velocity. When a call is made, the network pages the destination mobile around the predicted location. A mobile makes the same location prediction as the network does; it inspects its own location periodically and reports the new location when the distance between the predicted and the actual locations exceeds a threshold. To more realistically represent the various degrees of velocity correlation in time, a Gauss-Markov mobility model is used. For practical systems where the mobility pattern varies over time, we propose a dynamic Gauss-Markov parameter estimator that provides the mobility parameters to the prediction algorithm.Based on the Gauss-Markov model, we describe an analytical framework to evaluate the cost of mobility management for the proposed scheme. We also present an approximation method that reduces the computational complexity of the cost evaluation for multidimensional systems. We then compare the cost of predictive mobility management against that of the regular, nonpredictive distance-based scheme, for both the case with ideal Gauss-Markov mobility pattern and the case with time-varying mobility pattern.The performance advantage of the proposed scheme is demonstrated under various mobility patterns, call patterns, location inspection cost, location updating cost, mobile paging cost, and frequencies of mobile location inspections. As a point of reference, prediction can reduce the mobility management cost by more than 50% for all systems, where a the mobile users have moderate mean velocity and where performing a single location update is as least as expensive as paging a mobile in one cell.

Journal ArticleDOI
TL;DR: A power-allocation policy is developed which stabilizes the system whenever the rate vector lies within the capacity region and provides a performance bound for the Choose-the-K-Largest-Connected-Queues policy.
Abstract: We consider power and server allocation in a multibeam satellite downlink which transmits data to N different ground locations over N time-varying channels. Packets destined for each ground location are stored in separate queues and the server rate for each queue, i, depends on the power, p/sub i/(t), allocated to that server and the channel state, c/sub i/(t), according to a concave rate-power curve /spl mu//sub i/(p/sub i/,c/sub i/). We establish the capacity region of all arrival rate vectors (/spl lambda//sub 1/,...,/spl lambda//sub N/) which admit a stabilizable system. We then develop a power-allocation policy which stabilizes the system whenever the rate vector lies within the capacity region. Such stability is guaranteed even if the channel model and the specific arrival rates are unknown. Furthermore, the algorithm is shown to be robust to arbitrary variations in the input rates and a bound on average delay is established. As a special case, this analysis verifies stability and provides a performance bound for the choose-the-K-largest-connected-queues policy when channels can be in one of two states (ON or OFF ) and K servers are allocated at every timestep (K

Journal ArticleDOI
TL;DR: This paper forms the bandwidth provisioning problem for an SON which buys bandwidth from the underlying network domains to provide end-to-end value-added QoS sensitive services such as VoIP and Video-on-Demand mathematically and develops approximate solutions.
Abstract: We advocate the notion of service overlay network (SON) as an effective means to address some of the issues, in particular, end-to-end quality of service (QoS), plaguing the current Internet, and to facilitate the creation and deployment of value-added Internet services such as VoIP, Video-on-Demand, and other emerging QoS-sensitive services. The SON purchases bandwidth with certain QoS guarantees from the individual network domains via bilateral service level agreement (SLA) to build a logical end-to-end service delivery infrastructure on top of the existing data transport networks. Via a service contract, users directly pay the SON for using the value-added services provided by the SON. In this paper, we study the bandwidth provisioning problem for a SON which buys bandwidth from the underlying network domains to provide end-to-end value-added QoS sensitive services such as VoIP and Video-on-Demand. A key problem in the SON deployment is the problem of bandwidth provisioning, which is critical to cost recovery in deploying and operating the value-added services over the SON. The paper is devoted to the study of this problem. We formulate the bandwidth provisioning problem mathematically, taking various factors such as SLA, service QoS, traffic demand distributions, and bandwidth costs. Analytical models and approximate solutions are developed for both static and dynamic bandwidth provisioning. Numerical studies are also performed to illustrate the properties of the proposed solutions and demonstrate the effect of traffic demand distributions and bandwidth costs on SON bandwidth provisioning.

Journal ArticleDOI
TL;DR: A new approach to the virtual-topology reconfiguration problem for a wavelength-division-multiplexing-based optical wide-area mesh network under dynamic traffic demand is presented, and it is found that this method adapts very well to the changes in the offered traffic.
Abstract: We present a new approach to the virtual-topology reconfiguration problem for a wavelength-division-multiplexing- based optical wide-area mesh network under dynamic traffic demand. By utilizing the measured Internet backbone traffic characteristics, we propose an adaptation mechanism to follow the changes in traffic without a priori knowledge of the future traffic pattern. Our work differs from most previous studies on this subject which redesign the virtual topology according to an expected (or known) traffic pattern, and then modify the connectivity to reach the target topology. The key idea of our approach is to adapt the underlying optical connectivity by measuring the actual traffic load on lightpaths continuously (periodically based on a measurement period) and reacting promptly to the load imbalances caused by fluctuations on the traffic, by either adding or deleting one or more lightpath at a time. When a load imbalance is encountered, it is corrected either by tearing down a lightpath that is lightly loaded or by setting up a new lightpath when congestion occurs. We introduce high and low watermark parameters on lightpath loads to detect any over- or underutilized lightpath, and to trigger an adaptation step. We formulate an optimization problem which determines whether or not to add or delete lightpaths at the end of a measurement period, one lightpath at a time, as well as which lightpath to add or delete. This optimization problem turns out to be a mixed-integer linear program. Simulation experiments employing the adaptation algorithm on realistic network scenarios reveal interesting effects of the various system parameters (high and low watermarks, length of the measurement period, etc.). Specifically, we find that this method adapts very well to the changes in the offered traffic.

Journal ArticleDOI
TL;DR: This work adopts a (quasi-)static view of the RWA problem and proposes new integer-linear programming formulations, which can be addressed with highly efficient linear programming methods and yield optimal or near-optimal RWA policies.
Abstract: The problem of routing and wavelength assignment (RWA) is critically important for increasing the efficiency of wavelength-routed all-optical networks. Given the physical network structure and the required connections, the RWA problem is to select a suitable path and wavelength among the many possible choices for each connection so that no two paths sharing a link are assigned the same wavelength. In work to date, this problem has been formulated as a difficult integer programming problem that does not lend itself to efficient solution or insightful analysis. In this work, we propose several novel optimization problem formulations that offer the promise of radical improvements over the existing methods. We adopt a (quasi-)static view of the problem and propose new integer-linear programming formulations, which can be addressed with highly efficient linear (not integer) programming methods and yield optimal or near-optimal RWA policies. The fact that this is possible is surprising, and is the starting point for new and greatly improved methods for RWA. Aside from its intrinsic value, the quasi-static solution method can form the basis for suboptimal solution methods for the stochastic/dynamic settings.

Journal ArticleDOI
TL;DR: The findings indicate that although voice services can be adequately provided by some ISPs, a significant number of Internet backbone paths lead to poor performance.
Abstract: As the Internet evolves into a ubiquitous communication infrastructure and provides various services including telephony, it will be expected to meet the quality standards achieved in the public switched telephone network. Our objective in this paper is to assess to what extent today's Internet meets this expectation. Our assessment is based on delay and loss measurements taken over wide-area backbone networks and uses subjective voice quality measures capturing the various impairments incurred. First, we compile the results of various studies into a single model for assessing the voice-over-IP (VoIP) quality. Then, we identify different types of typical Internet paths and study their VoIP performance. For each type of path, we identify those characteristics that affect the VoIP perceived quality. Such characteristics include the network loss and the delay variability that should be appropriately handled by the playout scheduling at the receiver. Our findings indicate that although voice services can be adequately provided by some ISPs, a significant number of Internet backbone paths lead to poor performance.

Journal ArticleDOI
TL;DR: This study addresses the routing and wavelength-assignment problem in a network with path protection under duct-layer constraints in a wavelength-division multiplexing (WDM) network in which failures occur due to fiber cuts.
Abstract: This study investigates the problem of fault management in a wavelength-division multiplexing (WDM)-based optical mesh network in which failures occur due to fiber cuts. In reality, bundles of fibers often get cut at the same time due to construction or destructive natural events, such as earthquakes. Fibers laid down in the same duct have a significant probability to fail at the same time. When path protection is employed, we require the primary path and the backup path to be duct-disjoint, so that the network is survivable under single-duct failures. Moreover, if two primary paths go through any common duct, their backup paths cannot share wavelengths on common links. This study addresses the routing and wavelength-assignment problem in a network with path protection under duct-layer constraints. Off-line algorithms for static traffic is developed to combat single-duct failures. The objective is to minimize total number of wavelengths used on all the links in the network. Both integer linear programs and a heuristic algorithm are presented and their performance is compared through numerical examples.

Journal ArticleDOI
TL;DR: This paper describes a simple, lossless method of preventing deadlocks and livelocks in backpressured packet networks that represents a new networking paradigm in which internal network losses are avoided (thereby simplifying the design of other network protocols) and internal network delays are bounded.
Abstract: No packets will be dropped inside a packet network, even when congestion builds up, if congested nodes send backpressure feedback to neighboring nodes, informing them of unavailability of buffering capacity-stopping them from forwarding more packets until enough buffer becomes available. While there are potential advantages in backpressured networks that do not allow packet dropping, such networks are susceptible to a condition known as deadlock in which throughput of the network or part of the network goes to zero (i.e., no packets are transmitted). In this paper, we describe a simple, lossless method of preventing deadlocks and livelocks in backpressured packet networks. In contrast with prior approaches, our proposed technique does not introduce any packet losses, does not corrupt packet sequence, and does not require any changes to packet headers. It represents a new networking paradigm in which internal network losses are avoided (thereby simplifying the design of other network protocols) and internal network delays are bounded.

Journal ArticleDOI
TL;DR: It is shown that an aggregate information scenario which uses only aggregated and not per-path information provides sufficient information for a suitably developed algorithm to be able to perform almost as well as the complete information scenario.
Abstract: This paper presents new algorithms for dynamic routing of restorable bandwidth-guaranteed paths. We assume that connection requests one-by-one and have to be routed with no a priori knowledge of future arrivals. In order to guarantee restorability, in addition to determining an active path to route each request, an alternate link (node) disjoint backup (restoration) path has to be determined for the request at the time of connection initiation. This joint on-line routing problem is becoming particularly important in optical networks and in multiprotocol label switching (MPLS)-based networks due to the trend in backbone networks toward dynamic provisioning of bandwidth-guaranteed or wavelength paths. A straightforward solution for the restoration problem is to find two disjoint paths. However, this results in excessive resource usage. Given a restoration objective, such as protection against single-link failures, backup path bandwidth usage can be reduced by judicious sharing of backup paths amongst certain active paths while still maintaining restorability. The best sharing performance is achieved if the routing of every path in progress in the network is known to the routing algorithm at the time of a new path setup. We give an integer programming formulation for this problem which is new. Complete path routing knowledge is a reasonable assumption for a centralized routing algorithm. However, it is not often desirable, particularly when distributed routing is preferred. We show that an aggregate information scenario which uses only aggregated and not per-path information provides sufficient information for a suitably developed algorithm to be able to perform almost as well as the complete information scenario. Disseminating this aggregate information is feasible using proposed traffic engineering extensions to routing protocols. We formulate the dynamic restorable bandwidth routing problem in this aggregate information scenario and develop efficient routing algorithms. We show that the performance of our aggregate information-based algorithm is close to the complete information bound.

Journal ArticleDOI
TL;DR: This research focuses on delay-based congestion avoidance algorithms (DCA), like TCP/Vegas, which attempt to utilize the congestion information contained in packet round-trip time (RTT) samples, and shows evidence suggesting that a single deployment of DCA is not a viable enhancement to TCP over high-speed paths.
Abstract: The set of TCP congestion control algorithms associated with TCP/Reno (e.g., slow-start and congestion avoidance) have been crucial to ensuring the stability of the Internet. Algorithms such as TCP/NewReno (which has been deployed) and TCP/Vegas (which has not been deployed) represent incrementally deployable enhancements to TCP as they have been shown to improve a TCP connection's throughput without degrading performance to competing flows. Our research focuses on delay-based congestion avoidance algorithms (DCA), like TCP/Vegas, which attempt to utilize the congestion information contained in packet round-trip time (RTT) samples. Through measurement and simulation, we show evidence suggesting that a single deployment of DCA (i.e., a TCP connection enhanced with a DCA algorithm) is not a viable enhancement to TCP over high-speed paths. We define several performance metrics that quantify the level of correlation between packet loss and RTT. Based on our measurement analysis we find that although there is useful congestion information contained within RTT samples, the level of correlation between an increase in RTT and packet loss is not strong enough to allow a TCP/Sender to reliably improve throughput. While DCA is able to reduce the packet loss rate experienced by a connection, in its attempts to avoid packet loss, the algorithm will react unnecessarily to RTT variation that is not associated with packet loss. The result is degraded throughput as compared to a similar flow that does not support DCA.

Journal ArticleDOI
TL;DR: It is found that caching alone is seldom effective at insulating services from failures but that the combination of mobile extension code and prefetching can improve average unavailability by as much as an order of magnitude for classes of service whose semantics support disconnected operation.
Abstract: This paper seeks to understand how network failures affect the availability of service delivery across wide-area networks (WANs) and to evaluate classes of techniques for improving end-to-end service availability. Using several large-scale connectivity traces, we develop a model of network unavailability that includes key parameters such as failure location and failure duration. We then use trace-based simulation to evaluate several classes of techniques for coping with network unavailability. We find that caching alone is seldom effective at insulating services from failures but that the combination of mobile extension code and prefetching can improve average unavailability by as much as an order of magnitude for classes of service whose semantics support disconnected operation. We find that routing-based techniques may provide significant improvements but that the improvements of many individual techniques are limited because they do not address all significant categories of network failures. By combining the techniques we examine, some systems may be able to reduce average unavailability by as much as one or two orders of magnitude.

Journal ArticleDOI
TL;DR: In this article, the authors present analytic models to estimate the latency and steady-state throughput of TCP Tahoe, Reno, and SACK and validate their models using both simulations and TCP traces collected from the Internet.
Abstract: Continuing the process of improvements made to TCP through the addition of new algorithms in Tahoe and Reno, TCP SACK aims to provide robustness to TCP in the presence of multiple losses from the same window. In this paper we present analytic models to estimate the latency and steady-state throughput of TCP Tahoe, Reno, and SACK and validate our models using both simulations and TCP traces collected from the Internet. In addition to being the first models for the latency of finite Tahoe and SACK flows, our model for the latency of TCP Reno gives a more accurate estimation of the transfer times than existing models. The improved accuracy is partly due to a more accurate modeling of the timeouts, evolution of cwnd during slow start and the delayed ACK timer. Our models also show that, under the losses introduced by the droptail queues which dominate most routers in the Internet, current implementations of SACK can fail to provide adequate protection against timeouts and a loss of roughly more than half the packets in a round will lead to timeouts. We also show that with independent losses SACK performs better than Tahoe and Reno and, as losses become correlated, Tahoe can outperform both Reno and SACK.

Journal ArticleDOI
Qing Zhao1, Lang Tong1
TL;DR: An adaptive medium-access control (MAC) protocol for heterogeneous networks with finite population is proposed and has superior throughput and delay performance as compared to the slotted ALOHA with the optimal retransmission probability.
Abstract: An adaptive medium-access control (MAC) protocol for heterogeneous networks with finite population is proposed. Referred to as the multiqueue service room (MQSR) protocol, this scheme is capable of handling users with different quality-of-service (QoS) constraints. By exploiting the multipacket reception (MPR) capability, the MQSR protocol adaptively grants access to the MPR channel to a number of users such that the expected number of successfully received packets is maximized in each slot. The optimal access protocol avoids unnecessary empty slots for light traffic and excessive collisions for heavy traffic. It has superior throughput and delay performance as compared to, for example, the slotted ALOHA with the optimal retransmission probability. This protocol can be applied to random-access networks with multimedia traffic.

Journal ArticleDOI
TL;DR: Contrary to conventional belief, it is shown that appropriate modifications of the weighted round-robin (WRR) service discipline can, in fact, provide tight fairness properties and efficient delay guarantees to multiple sessions.
Abstract: Weighted fair queueing (WFQ)-based packet scheduling schemes require processing at line speeds for tag computation and tag sorting. This requirement presents a bottleneck for their implementation at high transmission speeds. In this paper, we propose an alternative and lower complexity approach to packet scheduling, based on modifications of the classical round-robin scheduler. Contrary to conventional belief, we show that appropriate modifications of the weighted round-robin (WRR) service discipline can, in fact, provide tight fairness properties and efficient delay guarantees to multiple sessions. Two such modifications are described: 1) list-based round robin, in which the server visits different sessions according to a precomputed list which is designed to obtain the desirable scheduling properties and 2) multiclass round robin, a version of hierarchical round robin with controls designed for good scheduling properties. The schemes considered are compared with well-known WFQ schemes and with deficit round robin (a credit-based WRR), on the basis of desirable properties such as bandwidth guarantees, fairness in excess bandwidth sharing, worst-case fairness, and efficiency of latency (delay guarantee) tuning. The scheduling schemes proposed and analyzed here operate with fixed packet sizes, and hence can be used in applications such as cell scheduling in ATM networks, time-slot scheduling on wireless links as in GPRS air interface, etc. A credit-based extension of the proposed schemes to handle variable packet sizes is also possible.

Journal ArticleDOI
TL;DR: An algorithm to estimate the parameters and size of a discrete MMPP (D-MMPP) from a data trace, which requires only two passes through the data and is used as the arrival process in a matrix-analytic queueing model.
Abstract: We start with the premise, and provide evidence that it is valid, that a Markov-modulated Poisson process (MMPP) is a good model for Internet traffic at the packet/byte level. We present an algorithm to estimate the parameters and size of a discrete MMPP (D-MMPP) from a data trace. This algorithm requires only two passes through the data. In tandem-network queueing models, the input to a downstream queue is the output from an upstream queue, so the arrival rate is limited by the rate of the upstream queue. We show how to modify the MMPP describing the arrivals to the upstream queue to approximate this effect. To extend this idea to networks that are not tandem, we show how to approximate the superposition of MMPPs without encountering the state-space explosion that occurs in exact computations. Numerical examples that demonstrate the accuracy of these methods are given. We also present a method to convert our estimated D-MMPP to a continuous-time MMPP, which is used as the arrival process in a matrix-analytic queueing model.

Journal ArticleDOI
TL;DR: It is shown that it is theoretically possible for a PPS to emulate a first-come first-served (FCFS) output-queued (OQ) packet switch if each lower speed packet switch operates at a rate of approximately 2R/k, and that if the lower speed packets are switched, the resulting PPS can emulate an FCFS-OQ switch within a delay bound.
Abstract: Our work is motivated by the desire to design packet switches with large aggregate capacity and fast line rates. In this paper, we consider building a packet switch from multiple lower speed packet switches operating independently and in parallel. In particular, we consider a (perhaps obvious) parallel packet switch (PPS) architecture in which arriving traffic is demultiplexed overk identical lower speed packet switches, switched to the correct output port, then recombined (multiplexed) before departing from the system. Essentially, the packet switch performs packet-by-packet load balancing, or inverse multiplexing, over multiple independent packet switches. Each lower speed packet switch operates at a fraction of the line rate R. For example, each packet switch can operate at rateR/k. It is a goal of our work that all memory buffers in the PPS run slower than the line rate. Ideally,a PPS would share the benefits of an output-queued switch, i.e., the delay of individual packets could be precisely controlled, allowing the provision of guaranteed qualities of service.In this paper, we ask the question: Is it possible for a PPS to precisely emulate the behavior of an output-queued packet switch with the same capacity and with the same number of ports? We show that it is theoretically possible for a PPS to emulate a first-come first-served (FCFS) output-queued (OQ) packet switch if each lower speed packet switch operates at a rate of approximately 2R/k. We further show that it is theoretically possible for a PPS to emulate a wide variety of quality-of-service queueing disciplines if each lower speed packet switch operates at a rate of approximately 3R/k. It turns out that these results are impractical because of high communication complexity, but a practical high-performance PPS can be designed if we slightly relax our original goal and allow a small fixed-size coordination buffer running at the line rate in both the demultiplexer and the multiplexer. We determine the size of this buffer and show that it can eliminate the need for a centralized scheduling algorithm, allowing a full distributed implementation with low computational and communication complexity. Furthermore, we show that if the lower speed packet switch operates at a rate ofR/k (i.e., without speedup), the resulting PPS can emulate an FCFS-OQ switch within a delay bound.

Journal ArticleDOI
TL;DR: The new protocols, Reliable Periodic Broadcast and Reliable Bandwidth Skimming, are simple to implement and achieve nearly the best possible scalability and efficiency for a given set of client characteristics and desirable/feasible media quality.
Abstract: Previous scalable on-demand streaming protocols do not allow clients to recover from packet loss. This paper develops new protocols that: (1) have a tunably short latency for the client to begin playing the media; (2) allow heterogeneous clients to recover lost packets without jitter as long as each client's cumulative loss rate is within a tunable threshold; and (3) assume a tunable upper bound on the transmission rate to each client that can be as small as a fraction (e.g., 25%) greater than the media play rate. Models are developed to compute the minimum required server bandwidth for a given loss rate and playback latency. The results of the models are used to develop the new protocols and assess their performance. The new protocols, Reliable Periodic Broadcast and Reliable Bandwidth Skimming, are simple to implement and achieve nearly the best possible scalability and efficiency for a given set of client characteristics and desirable/feasible media quality. Furthermore, the results show that the new reliable protocols that transmit to each client at only twice the media play rate have similar performance to previous protocols that require clients to receive at many times the play rate.

Journal ArticleDOI
TL;DR: This paper introduces a novel graph-theoretic algorithm, called turn-prohibition (TP), that breaks all the cycles in a network and, thus, prevents any interdependence between flows, and proves that the TP-algorithm prohibits the use of at most 1/3 of the total number turns in anetwork, for any network topology.
Abstract: Network calculus is known to apply in general only to feedforward routing networks, i.e., networks where routes do not create cycles of interdependent packet flows. In this paper, we address the problem of using network calculus in networks of arbitrary topology. For this purpose, we introduce a novel graph-theoretic algorithm, called turn-prohibition (TP), that breaks all the cycles in a network and, thus, prevents any interdependence between flows. We prove that the TP-algorithm prohibits the use of at most 1/3 of the total number turns in a network, for any network topology. Using analysis and simulation, we show that the TP-algorithm significantly outperforms other approaches for breaking cycles, such as the spanning tree and up/down routing algorithms, in terms of network utilization and delay bounds. Our simulation results also show that the network utilization achieved with the TP-algorithm is within a factor of two of the maximum theoretical network utilization, for networks of up to 50 nodes of degree four. Thus, in many practical cases, the restriction of network calculus to feedforward routing networks may not represent a too significant limitation.

Journal ArticleDOI
TL;DR: It is shown that, by exploiting the typical hierarchical structure of large-scale networks, one can achieve a substantial improvement in terms of computational complexity.
Abstract: Precomputation-based methods have recently been proposed as an instrument to facilitate scalability, improve response time, and reduce computation load on network elements. The key idea is to effectively reduce the time needed to handle an event by performing a certain amount of computations in advance, i.e., prior to the event's arrival. Such computations are performed as background processes, thus enabling to promptly provide a solution upon a request, through a simple, fast procedure.In this paper, we investigate precomputation methods in the context of Quality-of-Service (QoS) routing. Precomputation is highly desirable for QoS routing schemes due to the high computation complexity of selecting QoS paths on the one hand, and the need to promptly provide a satisfactory path upon a request on the other hand. We consider two major settings of QoS routing. The first is the case where the QoS constraint is of the "bottleneck" type, e.g., a bandwidth requirement, and network optimization is sought through hop minimization. The second is the more general setting of "additive" QoS constraints (e.g., delay) and general link costs.This paper mainly focuses on the first setting. First, we show that, by exploiting the typical hierarchical structure of large-scale networks, one can achieve a substantial improvement in terms of computational complexity. Next, we consider networks with topology aggregation. We indicate that precomputation is a necessary element for any QoS routing scheme and establish a precomputation scheme appropriate for such settings. Finally, we consider the case of additive QoS constraints (e.g., delay) and general link costs. As the routing problem becomes NP-hard, we focus on e-optimal approximations and derive a precomputation scheme that offers a major improvement over the standard approach.