1974420
Performance evaluation of multiple time scale TCP under self-similar traffic conditions
Measurements of network traffic have shown that self-similarity is a ubiquitous phenomenon spanning across diverse network environments. In previous work, we have explored the feasibility of exploiting long-range correlation structure in self-similar traffic for congestion control. We have advanced the framework of multiple time scale congestion control and shown its effectiveness at enhancing performance for rate-based feedback control. In this article, we extend the multiple time scale control framework to window-based congestion control, in particular, TCP. This is performed by interfacing TCP with a large time scale module that adjusts the aggressiveness of bandwidth consumpton behavior exhibited by TCP as a function of large time scale network state, that is, information that exceeds the time horizon of the feedback loop as determined by RTT. How to effectively utilize such information—due to its probabilistic nature, dispersion over multiple time scales, and realization on top of existing window-based congestion controls—is a nontrivial problem. First, we define a modular extension of TCP (a function call with a simple interface that applies to various flavors of TCP, e.g., Tahoe, Reno, and Vegas) and show that it significantly improves performance. Second, we show that multiple time scale TCP endows the underlying feedback control with proacativity by bridging the uncertainty gap associated with reactive controls which is exacerbated by the high delay-bandwidth product in broadband wide area networks. Third, we investigate the influence of three traffic control dimensions—tracking ability, connection duration, and fairness—on performance. Performance evaluation of multiple time scale TCP is facilitated by a simulation benchmark environment based on physical modeling of self-similar traffic. We explicate our methodology for disc
15690470
Multiple Time Scale Congestion Control for Self-Similar Network Traffic
Abstract Analytical and empirical studies have shown that self-similar traffic can have a detrimental impact on network performance including amplified queuing delay and packet loss rate. Given the ubiquity of scale-invariant burstiness observed across diverse networking contexts, finding effective traffic control algorithms capable of detecting and managing self-similar traffic has become an important problem. In this paper, we study congestion control algorithms for improving network performance — in particular, throughput — under self-similar traffic conditions. Although scale-invariant burstiness implies the existence of concentrated periods of contention and idleness, the long-range dependence associated with self-similar traffic leaves open the possibility that correlation structure at larger time scales may be exploited for performance enhancement purposes. We construct a 2-level multiple time scale congestion control protocol that exercises congestion control concurrently across two time scales an order of magnitude apart. The first component — acting at the smaller time scale — is a generic linear increase/exponential decrease feedback congestion control that uses implicit prediction afforded by feedback to affect rate control sensitive to changes in network state at 20–200 ms time scales. The second component — acting at 2–5 s time scales — uses explicit prediction to detect persistent shifts in overall network contention and uses this information to modulate the aggressiveness exhibited by the first component. We show that cooperative interaction between the two congestion control modules acting on information at different time scales leads to improved performance vis-a-vis the case when the large time scale component is absent. We show that the improvement factor increases with long-range dependence and we show that as the number of flows engaging in multiple time scale congestion control (MTSC) increases, both fairness and efficiency are preserved.
34711054
Towards communication-sensitive load balancing
Recent studies have shown that if the overhead associated with process migration is not high when compared to the estimated lifetime of processes, migrating isolated processes to underutilized workstations can improve system performance. However, if tightly-coupled processes from groups of communicating processes or other existing dependencies are separated, system performance can decrease due to increased communication cost. Current load balancers do not make effective use of communication dependencies in their algorithms. We have implemented a communication-sensitive load balancer for DUNES (Distributed UNix ExtenSion)-a user-level distributed operating system-that uses the run-time communication pattern between processes when balancing load. We examine communication-sensitive load balancing under different workload conditions and compare its performance against a load balancer that is not communication-aware. Our results show that a communication-sensitive load balancer performs substantially better than a load balancer that is not communication-aware under workloads that have a "good mix" of CPU-bound and I/O-bound processes.
14470122
An architecture for noncooperative QoS provision in many-switch systems
With the proliferation of high-speed networks and networked services, provisioning differentiated services to a diverse user base with heterogeneous QoS requirements has become an important problem. The traditional approach of resource reservation and admission control provides both guarantees and graded services, however, at the cost of potentially underutilized resources and limited scalability. We describe a WAN QoS provision architecture that adaptively organizes best-effort bandwidth into stratified services with graded QoS properties such that the QoS needs of a diverse user base can be effectively met. This architecture-SBS (stratified best-effort service)-promotes a simple user/simple network realization where neither the user nor the network is burdened with complex computational responsibilities. SBS is scalable, efficient, and adaptive, and it complements the guaranteed service architecture, sharing a common network substrate comprised of GPS routers. It is also a functional complement, provisioning QoS efficiently commensurate with user needs, albeit at the cost of weaker protection. SBS is suited to noncooperative network environments where users behave selfishly and resource contention resolution is mediated by the principle of competitive interaction. A principal feature of SBS is the transformation of user-centric QoS provision mechanisms-a defining characteristic of competitive interaction entailing intimate user control of internal network resources-into network-centric mechanisms while preserving the former's resource allocation paradigm. End-to-end QoS control is facilitated by decentralized control based on Lagrangian optimization-achieve a target end-to-end QoS at minimum cost or resource usage-which, in turn, is amenable to distributed implementation. SBS achieves per-flow QoS control with zero per-flow state at routers and a packet header whose size is independent of hop count. SBS, in spite of foregoing both resource reservation and admission control, is able to provision stable, graded QoS.
14277915
QoS-sensitive transport of real-time MPEG video using adaptive redundancy control
This paper presents an adaptive end-to-end protocol for quality of service (QoS)-sensitive transport of real-time MPEG video using packet-level forward error correction in dynamic networks. The objective is to facilitate a user-specified QoS end-to-end - i.e. without special network support - for real-time MPEG video whose timing constraints rule out the use of retransmission-based congestion control and QoS provisioning schemes. The degree of redundancy - overcode in forward error correction - injected into the network is adjusted as a function of network state, decreasing when the network is well-behaved to minimize unnecessary network resource consumption, and increasing when it is not to compensate for adverse network effects so as to maintain an invariant level of end-to-end QoS. We describe an adaptive packet-level forward error correction protocol called AFEC and analyze its properties with respect to optimality and stability. The optimal control problem is nontrivial due to the fact that increased redundancy, beyond a certain point, can backfire resulting in self-induced congestion which impedes the timely recovery of information - MPEG video frames - at the receiver. We experimentally evaluate the efficacy of our end-to-end QoS control by implementing and customizing AFEC to the transport of real-time MPEG video. We realize the system as a transport/application layer protocol running on UNIX workstations, and measure its performance over controlled network environments. We show that AFEC is able to effectively hide potentially adverse network effects such as packet drops and delays stemming from traffic burstiness and nonstationary structural changes using adaptive redundancy control, exporting a constant QoS service commensurate with user-specified QoS.
35766217
Efficient Shaping of User-Specified QoS Using Aggregate-Flow Control
Architecting networks capable of providing scalable, efficient, and fair services to users with diverse QoS requirements is a pressing problem. The two principal issues are: design of "good" per-hop behavior and edge control. In previous work [2,3], we studied aggregate-flow QoS control from a noncooperative resource provisioning context. In [20], the framework was generalized by, one, solving an optimal aggregate-flow per-hop behavior problem, and two, showing how it can be used coupled with end-to-end label control to facilitate scalable and fair QoS when driven by selfish users and service providers. In this paper, we focus on optimal aggregate-flow per-hop control and complement analysis by experimental performance evaluation. We show that user-specified, diverse QoS is efficiently facilitated over the optimal per-hop behavior network substrate using adaptive label control end-to-end.
11662197
pKASSO: Towards Seamless Authentication Providing Non-Repudiation on Resource-Constrained Devices
PKI is generally considered as the most appropriate solution for e-commerce and mutual authentication, owing to its digital signature and non-repudiation features. Asymmetric key operations of PKI require by far more CPU cycles than a symmetric cryptographic algorithm. It hampers the usability of PKI on resource-constrained devices. To overcome these limitations, we propose a new PKI- based authentication protocol and security infrastructure enhanced with single sign-on and delegation technology for a device with a restricted computing power. Although a conventional delegation mechanism cannot support non-repudiation mechanism against malicious user's behavior, our proposed protocol and security infrastructure can provide the mechanism by devising a referee server that generates binding information between a device and authentication messages, and retains the information in its local storage for future accusation.
12402823
A BEM Formulation for Transient Axisymmetric Thermoelastic Analysis Using Particular Integrals
A particular integral formulation is presented for the first time in a purely axisymmetric transient uncoupled thermoelastic analysis. The governing axisymmetric elastostatic equation is used as the complementary solution and the particular integrals for displacements and tractions are derived by integrating the three-dimensional formulation along the circumferential direction leading to elliptic integrals. The numerical results for three example problems are given and compared with their analytical solutions. Generally, the agreement among all of those results is satisfactory, if a few interior points are used to include the known values of temperature.
13632261
On the relationship between file sizes, transport protocols, and self-similar network traffic
Measurements of LAN and WAN traffic show that network traffic exhibits variability on different scales. We examine a mechanism that gives rise to self-similar network traffic and discuss performance. The mechanism we study is the transfer of files or messages whose size is drawn from a heavy-tailed distribution. In a realistic client/server network the degree to which file sizes are heavy-tailed can directly determine the degree of traffic self-similarity at the link level. This causal relationship is robust relative to changes in network resources, network topology, the influence of cross-traffic, and the distribution of interarrival times. Properties of the transport layer play an important role in preserving and modulating this relationship. The reliable transmission and flow control mechanisms of TCP serve to maintain the long-range dependency structure induced by heavy-tailed file size distributions. In contrast, if a non-flow-controlled and unreliable (UDP-based) transport protocol is used, the resulting traffic shows little self-similarity: although still bursty at short time scales, it has little long-range dependence. Performance implications of self-similarity are discussed as represented by various performance measures. Increased self-similarity as expected, results in degradation of performance. Queueing delay, in particular is discussed. Throughput-related measures such as packet loss and retransmission rate, however increase only gradually with increasing traffic self-similarity as long as reliable, flow-controlled transport protocol is used.
17340710
Stochastic analysis of packet-pair probing for network bandwidth estimation
In this paper, we perform a stochastic analysis of the packet-pair technique, which is a widely used method for estimating the network bandwidth in an end-to-end manner. There has been no explicit delay model of the packet-pair technique primarily because the stochastic behavior of a packet pair has not been fully understood. Our analysis is based on a novel insight that the transient analysis of the G/D/1 system can accurately describe the behavior of a packet pair, providing an explicit stochastic model. We first investigate a single-hop case and derive an analytical relationship between the input and the output probing gaps of a packet pair. Using this single-hop model, we provide a multi-hop model under an assumption of a single tight link. Our model shows the following two important features of the packet-pair technique: (i) The difference between the proposed model and the previous fluid model becomes significant when the input probing gap is around the characteristic value. (ii) The available bandwidth of any link after the tight link is not observable. We verify our model via ns-2 simulations and empirical results. We give a discussion on recent packet-pair models in relation to the proposed model and show that most of them can be regarded as special cases of the proposed model.
9690053
Robust Delay Estimation for Internet Multimedia Applications
Traffic estimation and modeling has been a crucial issue in many research areas of communication networks. For example, intermediate routers in networks estimate the rate of packet flows via queue size information in order to maximize throughput and provide fairness between flows. Also, many senders in the end system such as multi-media applications estimate end-to-end delay between a sender and a receiver to reduce dropping probability of packet. In this position paper, we propose a robust adaptive estimation scheme for Internet multimedia applications. The proposed scheme adopts an autoregressive (AR) model for the process and identifies the parameters of the AR process via a robust identification algorithm. This robust identification algorithm usually leads to better performance when the noise is correlated and/or non-stationary, and it is also more robust to modeling uncertainties. Here, we consider the problem of end-to-end delay estimation for audio playout mechanisms. We rigorously formulate the proposed scheme in the realm of audio playout mechanisms and give some preliminary simulation result which shows effectiveness of the proposed scheme. Even though we apply the proposed scheme to the estimation of end-to-end delay in audio applications, the scheme itself is very general and we expect that it can be applied to many other estimation problems in Internet multimedia applications.