Search

Reliability has been well recognized as an important design objective in the design of modern high-speed networks. While traditional approaches offer either 100% protection in the presence of single link failure or no protection at all, connections in real networks may have multiple reliability requirements. The concept of differentiated reliability (DiR) has been introduced in the literature to provide multiple reliability requirements in protection schemes that provision spare resources. In this paper, we consider the problem of routing connections with differentiated reliability in wavelength-division multiplexing (WDM) mesh networks when backup sharing is not allowed. Our objective is to route connections with minimum network cost (e.g., network resources) while meeting their required reliability cannot be used for achieving efficiency, the goal is to achieve efficiency by improved path selection. In this paper, we first present an integer linear programming (ILP) formulation for the problem. By solving the ILP formulation, we can obtain an optimal solution with respect to the current network state for each dynamic arrival. To solve the ILP formulation, however, is time consuming for large networks. We thus propose two approximation algorithms for the problem. The first one, called Shortest-Path-Pair-based Auxiliary graph (SPPA), can obtain an r-approximation solution whose cost is at most 1 + ϵ times the optimum in O((n2 (n+ 1) + 2mn)(log log(2n) + 1/ϵ)) time, where n and m are the number of nodes and links in a network, respectively. To reduce the computational complexity of the first algorithm, the second algorithm, called Auxiliary graph-based Two-Step Approach (ATSA), is proposed and can obtain a near optimal solution with cost at most 2 + ϵ times that of the optimal solution in O(mn(log log n+ 1/ϵ)) time. [ABSTRACT FROM AUTHOR]

Traffic classification, a mapping of traffic to network applications, is important for a variety of networking and security issues, such as network measurement, network monitoring, as well as the detection of malware activities. In this paper, we propose Securitas, a network trace-based protocol identification system, which exploits the semantic information in protocol message formats. Securitas requires no prior knowledge of protocol specifications. Deeming a protocol as a language between two processes, our approach is based upon the new insight that the n-grams of protocol traces, just like those of natural languages, exhibit highly skewed frequency-rank distribution that can be leveraged in the context of protocol identification. In Securitas, we first extract the statistical protocol message formats by clustering n-grams with the same semantics, and then use the corresponding statistical formats to classify raw network traces. Our tool involves the following key features: 1) applicable to both connection oriented protocols and connection less protocols; 2) suitable for both text and binary protocols; 3) no need to assemble IP packets into TCP or UDP flows; and 4) effective for both long-live flows and short-live flows. We implement Securitas and conduct extensive evaluations on real-world network traces containing both textual and binary protocols. Our experimental results on BitTorrent, CIFS/SMB, DNS, FTP, PPLIVE, SIP, and SMTP traces show that Securitas has the ability to accurately identify the network traces of the target application protocol with an average recall of about 97.4% and an average precision of about 98.4%. Our experimental results prove Securitas is a robust system, and meanwhile displaying a competitive performance in practice. [ABSTRACT FROM AUTHOR]

It has been recently discovered that heavy-tailed completion times can result from protocol interaction even when file sizes are light-tailed. A key to this phenomenon is the use of a restart policy where if the file is interrupted before it is completed, it needs to restart from the beginning. In this paper, we show that fragmenting a file into pieces whose sizes are either bounded or independently chosen after each interruption guarantees light-tailed completion time as long as the file size is light-tailed; i.e., in this case, heavy-tailed completion time can only originate from heavy-tailed file sizes. If the file size is heavy-tailed, then the completion time is necessarily heavy-tailed. For this case, we show that when the file size distribution is regularly varying, then under independent or bounded fragmentation, the completion time tail distribution function is asymptotically bounded above by that of the original file size stretched by a constant factor. We then prove that if the distribution of times between interruptions has nondecreasing failure rate, the expected completion time is minimized by dividing the file into equal-sized fragments; this optimal fragment size is unique but depends on the file size. We also present a simple blind fragmentation policy where the fragment sizes are constant and independent of the file size and prove that it is asymptotically optimal. Both these policies are also shown to have desirable completion time tail behavior. Finally, we bound the error in expected completion time due to error in modeling of the failure process. [ABSTRACT FROM AUTHOR]

This paper is focused on techniques for maximizing utility across all users within a total network transit cost budget. We present a new method for selecting between replicated servers distributed over the Internet. First, we introduce a novel utility framework that factors in quality of service metrics. Then we design an optimization algorithm, solvable in polynomial time, to allocate user requests to servers based on utility while satisfying network transit cost constraints, mapping service names to service instance locators. We then describe an efficient, low overhead distributed model which only requires knowledge of a fraction of the data required by the global optimization formulation. Next, a load-balancing variant of the algorithm is explored that substantially reduces blocking caused by congested servers. Extensive simulations show that our method is scalable and leads to higher user utility compared with mapping user requests to the closest service replica, while meeting network traffic cost constraints. We discuss several options for real-world deployment that require no changes to end-systems based on either the use of SDN controllers or extensions to the current DNS system. [ABSTRACT FROM AUTHOR]

In this paper, we study the computational complexity and effectiveness of a concept we term "N-hub Shortest-Path Routing" in IP networks. N-hub Shortest-Path Routing allows the ingress node of a routing domain to determine up to N intermediate nodes ("hubs") through which a packet will pass before reaching its final destination. This facilitates better utilization of the network resources, while allowing the network routers to continue to employ the simple and well-known shortest-path routing paradigm. Although this concept has been proposed in the past, this paper is the first to investigate it in depth. We apply N-hub Shortest-Path Routing to the problem of minimizing the maximum load in the network. We show that the resulting routing problem is NP-complete and hard to approximate. However, we propose efficient algorithms for solving it both in the online and the offline contexts. Our results show that N-hub Shortest-Path Routing can increase network utilization significantly even for N = 1. Hence, this routing paradigm should be considered as a powerful mechanism for future datagram routing in the Internet. [ABSTRACT FROM AUTHOR]

This paper is concerned with flow control and resource allocation problems in computer networks in which real-time applications may have hard quality of service (QoS) requirements. Recent optimal flow control approaches are unable to deal with these problems since QoS utility functions generally do not satisfy the strict concavity condition in real-time applications. For elastic traffic, we show that bandwidth allocations using the existing optimal flow control strategy can be quite unfair. If we consider different QoS requirements among network users, it may be undesirable to allocate bandwidth simply according to the traditional max-min fairness or proportional fairness. Instead, a network should have the ability to allocate bandwidth resources to various users, addressing their real utility requirements. For these reasons, this paper proposes a new distributed flow control algorithm for multiservice networks, where the application's utility is only assumed to be continuously increasing over the available bandwidth. In this, we show that the algorithm converges, and that at convergence, the utility achieved by each application is well balanced in a proportionally (or max-mm) fair manner. [ABSTRACT FROM AUTHOR]

WDM optical networks provide unprecedented high speed and reliability for message transfer among the nodes. All-to-all routing is a fundamental routing problem in such networks and has been well studied on single hop WDM networks. However, the number of wavelengths to realize all-to-all routing on the single hop model typically is very large. One way to reduce the number of wavelengths is to use/c-hop routing, in which each routing path consists of k segments and each segment is assigned a different wavelength, where k usually is a small constant. Because of the complexity of design and analysis for such a routing problem, only few papers discussed and proposed all-to-all routing by k ≥ 2 hops. However, the proposed algorithms are usually exceeding complicated even for ring topologies. Often, an ad hoc approach is employed to deal with each individual topology. In this paper we propose a generic method for all-to-all routing in multi-hop WDM networks, which aims to minimize the number of wavelengths. We illustrate the approach for several optical networks of commonly used topology, including lines, rings, tori, meshes, and complete binary trees. For each case an upper bound on the number of wavelengths is obtained. The results show that this approach produces clear routing paths, requires less wavelengths, and can easily incorporate load balancing. For simple topologies such as fines and rings, this approach easily produces the same bounds on the number of wavelengths that were hard-obtained previously. Moreover, this general approach provides a unified routing algorithm for any d-dimensional torus, which seems impossible to obtain by the previous approach. [ABSTRACT FROM AUTHOR]

This paper proposes a new dynamic bandwidth allocation system for Ethernet Passive Optical Networks (EPONs), subject to requirements of fairness, efficiency, and cost. An Optical Line Terminal (OLT)-centric bandwidth allocation model is proposed which employs a credit pooling technique combined with a weighted-share policy to partition the upstream bandwidth among different classes of service, and to prevent Optical Network Units (ONUs) from monopolizing the bandwidth. The OLT-centric model allows global optimization of network resources, a characteristic which is not found in many earlier proposals. Supported by the new bandwidth allocation, the paper proposes a joint-ONU interval-based packet scheduling algorithm, referred to herein as COPS (Class-of-service Oriented Packet Scheduling), that meets the requirements set out above. We compare COPS with another well-known scheduling algorithm which employed a standard priority-based bandwidth sharing. We show that COPS is superior in terms of network utilization and maximum packet delay, with the consequence of an increase in average packet delay for the premium traffic. This drawback is overcome by combining COPS with a rate-based optimization scheme. [ABSTRACT FROM AUTHOR]

This paper presents a distanceless networking approach for wireless sensor networks sparsely deployed in large areas. By leveraging rateless codes, we provide distanceless transmission to expand the communication range of sensor motes and fully exploit network diversity. We address a variety of practical challenges to accommodate rateless coding on resource-constrained sensor motes and devise a communication protocol to efficiently coordinate the distanceless link transmissions. We propose a new metric (expected distanceless transmission time) for routing selection and further adapt the distanceless transmissions to low duty-cycled sensor networks. We implement the proposed scheme in TinyOS on the TinyNode platform and deploy the sensor network in a real-world project, in which 12 wind measurement sensors are installed around a large urban reservoir of 2.5\,\times\,3.0 km^2 to monitor the field wind distribution. Extensive experiments show that our proposed scheme significantly outperforms the state-of-the-art approaches for data collection in sparse sensor networks. [ABSTRACT FROM PUBLISHER]

Recent literature including our past work provides analysis and solutions for using: 1) erasure coding; 2) parallelism; or 3) variable slicing/chunking (i.e., dividing an object of a specific size into a variable number of smaller chunks) in speeding up the I/O performance of storage clouds. However, a comprehensive approach that considers all three dimensions together to achieve the best throughput-delay tradeoff curve had been lacking. This paper presents the first set of solutions that can pick the best combination of coding redundancy ratio and object chunking/slicing options as the load dynamically changes. Our specific contributions are as follows: 1) We establish via measurements that combining variable redundancy ratio and chunking is mostly feasible over a popular public cloud. 2) We relate the delay-optimal values for chunking level and code redundancy ratio to the queue backlogs via an approximate queuing analysis. 3) Based on this analysis, we propose TOFEC that adapts the chunking level and redundancy ratio against the queue backlogs. Our trace-driven simulation results show that TOFEC's adaptation mechanism converges to an appropriate code that provides the optimal throughput-delay tradeoff without reducing system capacity. Compared to a nonadaptive strategy optimized for throughput, TOFEC delivers 2.5\times lower latency under light workloads; compared to a nonadaptive strategy optimized for latency, TOFEC can scale to support over 3\times as many requests. 4)We propose a simpler greedy solution that performs on a par with TOFEC in average delay performance, but exhibits significantly more performance variations. [ABSTRACT FROM PUBLISHER]

Video streaming generates a substantial fraction of the traffic on the Internet. The demands of video streaming also increase the workload on the video server, which in turn leads to substantial slowdowns. In order to resolve the slowdown problem, and to provide a scalable and robust infrastructure to support on-demand streaming, helper-assisted video-on-demand (VoD) systems have been introduced. In this architecture, helper nodes, which are micro-servers with limited storage and bandwidth resources, download and store the user-requested videos from a central server to decrease the load on the central server. Multi-layer videos, in which a video is divided into different layers, can also be used to improve the scalability of the system. In this paper, we study the problem of utilizing the helper nodes to minimize the pressure on the central servers. We formulate the problem as a linear programming using joint inter- and intra-layer network coding. Our solution can also be implemented in a distributed manner. We show how our method can be extended to the case of wireless live streaming, in which a set of videos is broadcasted. Moreover, we extend the proposed method to the case of unreliable connections. We carefully study the convergence and the gain of our distributed approach. [ABSTRACT FROM PUBLISHER]

This paper studies network resource allocation between users that manage multiple connections, possibly through different routes, where each connection is subject to congestion control. We formulate a user-centric Network Utility Maximization problem that takes into account the aggregate rate a user obtains from all connections, and we propose decentralized means to achieve this fairness objective. In a first proposal, cooperative users control their number of active connections based on congestion prices from the transport layer to emulate suitable primal-dual dynamics in the aggregate rate; we show this control achieves asymptotic convergence to the optimal user-centric allocation. For the case of noncooperative users, we show that network stability and user-centric fairness can be enforced by a utility-based admission control implemented at the network edge. We also study stability and fairness issues when routing of incoming connections is enabled at the edge router. We obtain in this case a characterization of the stability region of loads that can be served with routing alone and a generalization of our admission control policy to ensure user-centric fairness when the stability condition is not met. The proposed algorithms are implemented at the packet level in ns2 and demonstrated through simulation. [ABSTRACT FROM AUTHOR]

It is well known that without admission control, network congestion is bound to occur. However, to implement admission control is difficult in IP-based networks, which are constructed out of the end-to-end principle, and semantics of most major signaling protocols can only be interpreted at the edge of the network. Even if routers can perform admission control internally, the path computation and the state updating activities required for setting up and tearing down each flow will overwhelm the network. A new QoS architecture, called a nonblocking network, has been proposed recently, and it requires no internal admission control and can still offer hard QoS guarantees. In this architecture, as long as each edge node admits not more than a specified amount of traffic, the network will never experience link congestion. For multicast networks, the main problem with this approach is low throughput. Conventional tree-based multicast routing algorithms lead to a throughput so low that the nonblocking concept is rendered impractical. In this paper, we show how network coding can solve this problem. We demonstrate that a nonblocking unicast network and a multicast network share the same optimal paths, and that a nonblocking multicast network with network coding can admit the same amount of traffic as a nonblocking unicast network. The above conclusions apply to explicit-routing (MPLS-like) and shortest-path routing (IP-like) networks. [ABSTRACT FROM PUBLISHER]

In this paper, we propose to tackle the problem of up-dating the access network in order to connect new subscribers and to satisfy the new class of service requirements for the existing subscribers to offer, for instance, new services such as high-definition television (HDTV) over the Internet protocol (IPTV). Four important access network architectures/technologies are considered: the digital subscriber line (xDSL) technologies deployed directly from the central office (CO), the fiber-to-the-node (FTTN), the fiber-to-the-micro-node (FTTn) and the fiber-to-the-premises (FTTP). An integer mathematical programming model is proposed for this network planning problem. Next, a heuristic algorithm based on the tabu search principle is proposed to find "good" feasible solutions within a reasonable amount of computational time. Finally, numerical results are presented and analyzed. To assess the quality of the solutions found with the proposed algorithm, they are compared to the optimal solutions found using a commercial implementation of the branch-and-bound algorithm. [ABSTRACT FROM AUTHOR]

The concept of dynamic spectrum access will allow the radio spectrum to be traded in a market like scenario allowing wireless service providers (WSPs) to tease chunks of spectrum on a short-term basis. Such market mechanisms will lead to competition among WSPs where they not only compete to acquire spectrum but also attract and retain users. Currently, there is little understanding on how such a dynamic trading system will operate so as to make the system feasible under economic terms. In this paper, we propose an economic framework that can be used to guide i) the dynamic spectrum allocation process and ii) the service pricing mechanisms that the providers can use. We propose a knapsack based auction model that dynamically allocates spectrum to the WSPs such that revenue and spectrum usage are maximized. We borrow techniques from game theory to capture the conflict of interest between WSPs and end users. A dynamic pricing strategy for the providers is also proposed. We show that even in a greedy and non-cooperative behavioral game model, it is in the best interest of the WSPs to adhere to a price and channel threshold which is a direct consequence of price equilibrium. Through simulation results, we show that the proposed auction model entices WSPs to participate in the auction, makes optimal use of the spectrum, and avoids collusion among WSPs. We demonstrate how pricing can be used as an effective tool for providing incentives to the WSPs to upgrade their network resources and offer better services. [ABSTRACT FROM AUTHOR]

Hose-model virtual private networks (VPNs) provide customers with more flexibility in specifying bandwidth requirements than pipe-model VPNs. Many hose-model VPN provisioning algorithms have been proposed, and they focus on the bandwidth efficiency in the construction of a single hose-model VPN. In practice, however, VPNs come and go and the dynamics will affect the performance of these VPN provisioning algorithms. If the frequency of adding and deleting VPNs is high, these algorithms will have a scalability problem. We propose in this paper a new network architecture for dynamic VPN construction. In the proposed architecture, adding a new VPN is much simpler and faster, and all that is required is to check if the edge routers have enough bandwidth. There is no need to check the bandwidth left on each internal link because the architecture guarantees that as long as the edge routers have enough capacities to accept the VPN, the internal links will never experience overflow caused by adding the new VPN. We present a linear programming formulation for finding the optimal routing that maximizes the amount of admissible VPN traffic in the network. We then exploit the underlying network flow structure and convert the linear programming problem into a subgradient iterative search problem. The resulting solution is significantly faster than the linear programming approach. [ABSTRACT FROM AUTHOR]

We consider the problem of congestion-aware multi-path routing in the Internet. Currently, Internet routing protocols select only a single path between a source and a destination. However, due to many policy routing decisions, single-path routing may limit the achievable throughput. In this paper, we envision a scenario where multi-path routing is enabled in the Internet to take advantage of path diversity. Using minimal congestion feedback signals from the routers, we present a class of algorithms that can be implemented at the sources to stably and optimally split the flow between each source-destination pair. We then show that the connection-level throughput region of such multi-path routing/congestion control algorithms can be larger than that of a single-path congestion control scheme. [ABSTRACT FROM AUTHOR]

In this paper, we propose two schemes for the load balanced Birkhoff-von Neumann switches to provide guaranteed rate services. The first scheme is based on an earliest eligible time first (EETF) policy. In such a scheme, we assign every packet of a guaranteed rate flow a targeted departure time that is the departure time from the corresponding work conserving link with capacity equal to the guaranteed rate. By implementing the EETF policy with jitter control mechanisms and first come first serve (FCFS) queues, we show that the end-to-end delay for every packet of a guaranteed rate flow is bounded by the sum of its targeted departure time and a constant that only depends on the number of flows and the size of the switch. Our second scheme is a frame based scheme as in Keslassy and McKeown, 2002. There, time slots are grouped into fixed size frames. Packets are placed in appropriate bins (buffers) according to their arrival times and their flows. We show that if the incoming traffic satisfies certain rate assumptions, then the end-to-end delay for every packet and the size of the central buffers are both bounded by constants that only depend on the size of the switches and the frame size. The second scheme is much simpler than the first one in many aspects: 1) the on-line complexity is O(1) as there is no need for complicated scheduling; 2) central buffers are finite and thus can be built into a single chip; 3) connection patterns of the two switch fabrics are changed less frequently; 4) there is no need for resequencing-and-output buffer after the second stage; and 5) variable length packets may be handled without segmentation and reassembly. [ABSTRACT FROM AUTHOR]

The addition of the two-bit Explicit Congestion Notification (ECN) field to the IP header provides routers with a mechanism for conveying link price information necessary for the successful operation of a number of congestion control schemes. Two recent proposals for probabilistic packet marking at the routers allow receivers to estimate path price from the fraction of marked packets. In this paper we introduce an alternative deterministic marking scheme for encoding path price. Each router quantizes the price of its outgoing link to a fixed number of bits. Every data packet sent along the path encodes a partial sum of the quantized link prices in its ECN field, allowing the receiver to estimate the path price. We evaluate the performance of our algorithm in terms of its error in representing prices, and compare it to probabilistic marking. We show that based on empirical Internet traffic characteristics, our algorithm performs better when estimating time-varying prices and static path price using small blocks of packets. [ABSTRACT FROM AUTHOR]

Rate control methodologies that are currently available in IEEE 802.11 network cards seriously underutilize network resources and, in addition, per-second throughputs suffer from high variability. In this paper, we introduce an algorithm, H-RCA, that overcomes these shortcomings, giving substantially higher, and less variable, throughput. The approach solely uses information already available at the driver-level to function and can be implemented on 802.11e commodity hardware. H-RCA's design objective is to minimize the average time each packet spends on the medium (including retries) in order to maximize total network throughput. It uses a development of a recently proposed estimation scheme to distinguish transmission failures due to collisions from those caused by channel noise. It employs an estimate of the packet loss ratio due to noise in assessing whether it is appropriate to change rate. We demonstrate experimentally that packet loss ratio is not necessarily a monotonic increasing function of rate; this is accounted for in H-RCA's design. As H-RCA statistically separates noise losses from those caused by collisions, ns-2 simulations show that it is robust to changing environments. H-RCA does not require specific hardware support nor any change to the IEEE 802.11 protocol. This point is substantiated with results from an experimental implementation. [ABSTRACT FROM AUTHOR]

Recent botnets such as Conficker, Kraken, and Torpig have used DNS-based “domain fluxing” for command-and-control, where each Bot queries for existence of a series of domain names and the owner has to register only one such domain name. In this paper, we develop a methodology to detect such “domain fluxes” in DNS traffic by looking for patterns inherent to domain names that are generated algorithmically, in contrast to those generated by humans. In particular, we look at distribution of alphanumeric characters as well as bigrams in all domains that are mapped to the same set of IP addresses. We present and compare the performance of several distance metrics, including K–L distance, Edit distance, and Jaccard measure. We train by using a good dataset of domains obtained via a crawl of domains mapped to all IPv4 address space and modeling bad datasets based on behaviors seen so far and expected. We also apply our methodology to packet traces collected at a Tier-1 ISP and show we can automatically detect domain fluxing as used by Conficker botnet with minimal false positives, in addition to discovering a new botnet within the ISP trace. We also analyze a campus DNS trace to detect another unknown botnet exhibiting advanced domain-name generation technique. [ABSTRACT FROM AUTHOR]

The abuse of chat services by automated programs, known as chat bots, poses a serious threat to Internet users. Chat bots target popular chat networks to distribute spam and malware. In this paper, we first conduct a series of measurements on a large commercial chat network. Our measurements capture a total of 16 different types of chat bots ranging from simple to advanced. Moreover, we observe that human behavior is more complex than bot behavior. Based on the measurement study, we propose a classification system to accurately distinguish chat bots from human users. The proposed classification system consists of two components: 1) an entropy-based classifier; and 2) a Bayesian-based classifier. The two classifiers complement each other in chat bot detection. The entropy-based classifier is more accurate to detect unknown chat bots, whereas the Bayesian-based classifier is faster to detect known chat bots. Our experimental evaluation shows that the proposed classification system is highly effective in differentiating bots from humans. [ABSTRACT FROM PUBLISHER]

Alias resolution, the task of identifying IP addresses belonging to the same router, is an important step in building traceroute-based Internet topology maps. Inaccuracies in alias resolution affect the representativeness of constructed topology maps. This in turn affects the conclusions derived from studies that use these maps. This paper presents two complementary studies on alias resolution. First, we present an experimental study to demonstrate the impact of alias resolution on topology measurement studies. Then, we introduce an alias resolution approach called analytic and probe-based alias resolver (APAR). APAR consists of an analytical component and a probe-based component. Given a set of path traces, the analytical component utilizes the common IP address assignment scheme to infer IP aliases. The probe-based component introduces a minimal probing overhead to improve the accuracy of APAR. Compared to the existing state-of-the-art tool ally, APAR uses an orthogonal approach to resolve a large number of IP aliases that ally fails to identify. Our extensive verification study on sample data sets shows that our approach is effective in resolving many aliases with good accuracy. Our evaluations also indicate that the two approaches (ally and APAR) should be used together to maximize the success of the alias resolution process. [ABSTRACT FROM AUTHOR]

Network wide broadcasting is a fundamental operation in ad hoc networks. In broadcasting, a source node sends a message to all the other nodes in the network. In this paper, we consider the problem of collision-free broadcasting in ad hoc networks. Our objective is to minimize the latency and the number of transmissions in the broadcast. We show that minimum latency broad-casting is NP-complete for ad hoc networks. We also present a simple distributed collision-free broadcasting algorithm for broadcasting a message. For networks with bounded node transmission ranges, our algorithm simultaneously guarantees that the latency and the number of transmissions are within O(1) times their respective optimal values. Our algorithm and analysis extend to the case when multiple messages are broadcast from multiple sources. Experimental studies indicate that our algorithms perform much better in practice than the analytical guarantees provided for the worst case. [ABSTRACT FROM AUTHOR]

As the Internet evolves into a ubiquitous communication infrastructure and supports increasingly important services, its dependability in the presence of various failures becomes critical. In this paper, we analyze IS-IS routing updates from the Sprint IP backbone network to characterize failures that affect IP connectivity. Failures are first classified based on patterns observed at the IP-layer; in some cases, it is possible to further infer their probable causes, such as maintenance activities, router-related and optical layer problems. Key temporal and spatial characteristics of each class are analyzed and, when appropriate, parameterized using well-known distributions. Our results indicate that 20% of all failures happen during a period of scheduled maintenance activities. Of the unplanned failures, almost 30% are shared by multiple links and are most likely due to router-related and optical equipment-related problems, respectively, while 70% affect a single link at a time. Our classification of failures reveals the nature and extent of failures in the Sprint IP backbone. Furthermore, our characterization of the different classes provides a probabilistic failure model, which can be used to generate realistic failure scenarios, as input to various network design and traffic engineering problems. [ABSTRACT FROM AUTHOR]

Topology-transparent scheduling for mobile wireless ad hoc networks has been treated as a theoretical curiosity. This paper makes two contributions towards its practical deployment: 1) We generalize the combinatorial requirement on the schedules and show that the solution is a cover-free family. As a result, a much wider number and variety of constructions for schedules exist to match network conditions. 2) In simulation, we closely match the theoretical bound on expected throughput. The bound was derived assuming acknowledgments are available immediately. We use rateless forward error correction (RFEC) as an acknowledgment scheme with minimal computational overhead. Since the wireless medium is inherently unreliable, RFEC also offers some measure of automatic adaptation to channel load. These contributions renew interest in topology-transparent scheduling when delay is a principal objective. [ABSTRACT FROM AUTHOR]

This paper presents an approach to IP traceback based on the probabilistic packet marking paradigm. Our approach, which we call randomize-and-link, uses large checksum cords to "link" message fragments in a way that is highly scalable, for the checksums serve both as associative addresses and data integrity verifiers. The main advantage of these checksum cords is that they spread the addresses of possible router messages across a spectrum that is too large for the attacker to easily create messages that collide with legitimate messages. [ABSTRACT FROM AUTHOR]

We study limitations of an equation-based congestion control protocol, called TCP-Friendly Rate Control (TFRC). It examines how the three main factors that determine TFRC throughput, namely, the TCP-friendly equation, loss event rate estimation, and delay estimation, can influence the long-term throughput imbalance between TFRC and TCP. Especially, we show that different sending rates of competing flows cause these flows to experience different loss event rates. There are several fundamental reasons why TFRC and TCP flows have different average sending rates, from the first place. Earlier work shows that the convexity of the TCP-friendly equation used in TFRC causes the sending rate difference. We report two additional reasons in this paper: 1) the convexity of 1/χ where χ is a loss event period and 2) different retransmission timeout period (RTO) estimations of TCP and TFRC. These factors can be the reasons for TCP and TFRC to experience initially different sending rates. But we find that the loss event rate difference due to the differing sending rates greatly amplifies the initial throughput difference; in some extreme cases, TFRC uses around 20 times more, or sometimes 10 times less, bandwidth than TCP. Despite these factors influencing the throughput difference, we also find that simple heuristics can greatly mitigate the problem. [ABSTRACT FROM AUTHOR]

In today's Internet, users can choose their local Internet service providers (ISPs), but once their packets have entered the network, they have little control over the overall routes their packets take. Giving a user the ability to choose between provider-level routes has the potential of fostering ISP competition to offer enhanced service and improving end-to-end performance and reliability. This paper presents the design and evaluation of a new Internet routing architecture (NIRA) that gives a user the ability to choose the sequence of providers his packets take. NIRA addresses a broad range of issues, including practical provider compensation, scalable route discovery, efficient route representation, fast route fall-over, and security. NIRA supports user choice without running a global link-state routing protocol. It breaks an end-to-end route into a sender part and a receiver part and uses address assignment to represent each part. A user can specify a route with only a source and a destination address, and switch routes by switching addresses. We evaluate NIRA using a combination of network measurement, simulation, and analysis. Our evaluation shows that NIRA supports user choice with low overhead. [ABSTRACT FROM AUTHOR]

Most existing reliable multicast congestion control (RMCC) mechanisms try to emulate TCP congestion control behaviors for achieving TCP-compatibility. However, different loss recovery mechanisms employed in reliable multicast protocols, especially NAK-based retransmission and local loss recovery mechanisms, may lead to different behaviors and performance of congestion control. As a result, reliable multicast flows might be identified and treated as non-TCP-friendly by routers in the network. It is essential to understand those influences and take them into account in the development and deployment of reliable multicast services. In this paper, we study the influences comprehensively through analysis, modelling and simulations. We demonstrate that NAK-based retransmission and/or local loss recovery mechanisms are much more robust and efficient in recovering from single or multiple packet losses within a single round-trip time (RTT). For a better understanding on the impact of loss recovery on RIVICC, we derive expressions for steady-state throughput of NAK-based RMCC schemes, which clearly brings out the throughput advantages of NAK-based RMCC over TCP Reno. We also show that timeout effects have little impact on shaping the performance of NAK-based RMCC schemes except for extremely high loss rates (>0.2). Finally, we use simulations to validate our findings and show that local loss recovery may further increase the throughput and deteriorate the fairness properties of NAK-based RMCC schemes. These findings and insights could provide useful recommendations for the design, testing and deployment of reliable multicast protocols and services. [ABSTRACT FROM AUTHOR]

In this paper, we develop failure-resilient techniques for monitoring link delays and faults in a Service Provider or Enterprise IP network. Our two-phased approach attempts to minimize both the monitoring infrastructure costs as well as the additional traffic due to probe messages. In the first phase, we compute the locations of a minimal set of monitoring stations such that all network links are covered, even in the presence of several link failures. Subsequently, in the second phase, we compute a minimal set of probe messages that are transmitted by the stations to measure link delays and isolate network faults. We show that both the station selection problem as well as the probe assignment problem are NP-hard. We then propose greedy approximation algorithms that achieve a logarithmic approximation factor for the station selection problem and a constant factor for the probe assignment problem. These approximation ratios are provably very close to the best possible bounds for any algorithm. [ABSTRACT FROM AUTHOR]

The demand assigned capacity management (DACM) problem in IP over optical (IPO) network aims at devising efficient bandwidth replenishment schedules from the optical domain conditioned upon traffic evolution processes in the IP domain. A replenishment schedule specifies the location, sizing, and sequencing of link capacity expansions to support the growth of Internet traffic demand in the IP network subject to economic considerations. A major distinction in the approach presented in this paper is the focus of attention on the economics of ‘excess bandwidth’ in the IP domain, which can be viewed as an inventory system that is endowed with fixed and variable costs and depletes with increase in IP traffic demand requiring replenishment from the optical domain. We develop mathematical models to address the DACM problem in IPO networks based on a class of inventory management replenishment methods. We apply the technique to IPO networks that implement capacity adaptive routing in the IP domain and networks without capacity adaptive routing. We analyze the performance characteristics under both scenarios, in terms of minimizing cumulative replenishment cost over an interval of time. For the non-capacity adaptive routing scenario, we consider a shortest path approach in the IP domain, specifically OSPF. For the capacity adaptive scenario, we use an online constraint-based routing scheme. This study represents an application of integrated traffic engineering which concerns collaborative decision making targeted towards network performance improvement that takes into consideration traffic demands, control capabilities, and network assets at different levels in the network hierarchy. [ABSTRACT FROM AUTHOR]

Using ternary content addressable memory (TCAM) for high-speed IP address lookup has been gaining popularity due to its deterministic high performance. However, restricted by the slow improvement of memory accessing speed, the,, route lookup engines for next-generation terabit routers demand exploiting parallelism among multiple TCAM chips. Traditional parallel methods always incur excessive redundancy and high power consumption. We propose in this paper an original TCAM-based IP lookup scheme that achieves both ultra-high lookup throughput and optimal utilization of the memory while being power-efficient. In our multi-chip scheme, we devise a load-balanced TCAM table construction algorithm together with an adaptive load balancing mechanism. The power efficiency is well controlled by decreasing the number of TCAM entries triggered in each lookup operation. Using four 133 MHz TCAM chips and given 25% more TCAM entries than the original route table, the proposed scheme achieves a lookup throughput of up to 533 MPPS while remains simple for ASIC implementation. [ABSTRACT FROM AUTHOR]

There has been much interest in admission control schemes that place the burden of admission control decisions on the end users. In these schemes, referred to as Endpoint Admission Control, the decision to join the network is taken by the user, based on the probing of the network using probe packets. Depending on the level of congestion, routers mark the probe packets and thus inform the user of the state of the network. In this paper, we analyze three mechanisms for providing Endpoint Admission Control: virtual-queue marking, random-early marking and tail drop. For each scheme, we analyze the probing duration necessary to guarantee the required QoS and achieve high link utilization. Our main conclusion is that very few probe packets have to be sent when early marking is used, whereas tail drop requires a large number of probe packets. [ABSTRACT FROM AUTHOR]

To disseminate data to a heterogeneous body of receivers efficiently, congestion control protocols for IP multicast compose a session from several multicast groups and prescribe guidelines that enable each receiver to subscribe to an appropriate subset of the groups. However, a misbehaving receiver can ignore the group subscription rules and inflate its subscription to acquire unfairly high throughput. In this paper, we present the first solution for the problem of inflated subscription. Our design guards access to multicast groups with dynamic keys and consists of two independent components: DELTA (Distribution of ELigibility To Access)—a novel method for in-band distribution of group keys to receivers that are eligible to access the groups according to the congestion control protocol, and SIGMA (Secure Internet Group Management Architecture)—a generic architecture for key-based group access at edge routers. We apply DELTA and SIGMA to derive robust versions of prominent RLM and FLID-DL protocols. [ABSTRACT FROM AUTHOR]

Routers have the ability to output statistics about packets and flows of packets that traverse them. Since, however, the generation of detailed traffic statistics does not scale well with link speed, increasingly routers and measurement boxes implement sampling strategies at the packet level. In this paper, we study both theoretically and practically what information about the original traffic can be inferred when sampling, or ‘thinning’, is performed at the packet level. While basic packet level characteristics such as first order statistics can be fairly directly recovered, other aspects require more attention. We focus mainly on the spectral density, a second-order statistic, and the distribution of the number of packets per flow, showing how both can be exactly recovered, in theory. We then show in detail why in practice this cannot be done using the traditional packet based sampling, even for high sampling rate. We introduce an alternative flow-based thinning, where practical inversion is possible even at arbitrarily low sampling rate. We also investigate the theory and practice of fitting the parameters of a Poisson cluster process, modeling the full packet traffic, from sampled data. [ABSTRACT FROM AUTHOR]

Providing real-time communication services to multimedia applications and subscription-based Internet access often requires that sufficient network resources be reserved for real-time traffic. However, the reserved network resource is susceptible to resource theft and abuse. Without a resource access control mechanism that can efficiently differentiate legitimate real-time traffic from attacking packets, the traffic conditioning and policing enforced at Internet Service Provider (ISP) edge routers cannot protect the reserved network resource from embezzlement. On the contrary to the usual expectation, the traffic policing at edge routers aggravates their vulnerability to flooding attacks by blindly dropping packets, in this paper, we propose a fast and lightweight IP network-edge resource access control mechanism, called IP Easy-pass, to prevent unauthorized access to reserved network resources at edge devices. We attach a unique pass to each legitimate real-time packet so that an ISP edge router can validate the legitimacy of the incoming IP packet very quickly and simply by checking its pass. We present the generation of Easy-pass, its embedding, and verification procedures. We implement the IP Easy-pass mechanism in the Linux kernel, and measure its overhead on our testbed. Finally, we demonstrate its effectiveness against packet forgery and resource embezzlement attempts by conducting a series of experiments. [ABSTRACT FROM AUTHOR]

In this paper, we develop, analyze and implement a congestion control scheme in a noncooperative game framework, where each user's cost function is composed of a pricing function proportional to the queueing delay experienced by the user, and a fairly general utility function which captures the user demand for bandwidth. Using a network model based on fluid approximations and through a realistic modeling of queues, we establish the existence of a unique equilibrium as well as its global asymptotic stability for a general network topology, where boundary effects are also taken into account. We also provide sufficient conditions for system stability when there is a bottleneck link shared by multiple users experiencing nonnegligible communication delays. In addition, we study an adaptive pricing scheme using hybrid systems concepts. Based on these theoretical foundations, we implement a window-based, end-to-end congestion control scheme, and simulate it in ns-2 network simulator on various network topologies with sizable propagation delays. [ABSTRACT FROM AUTHOR]

One of the central issues in router performance is IP address lookup based on longest prefix matching. IP address lookup algorithms can be evaluated on a number of metrics—lookup time, update time, memory usage, and to a less important extent, the time to construct the data structure used to support lookups and updates. Many of the existing methods are geared toward optimizing a specific metric, and do not scale well with the ever expanding routing tables and the forthcoming IPv6 where the IP addresses are 128 bits long. In contrast, our effort is directed at simultaneously optimizing multiple metrics and provide solutions that scale to IPv6, with its longer addresses and much larger routing tables. In this paper, we present two IP address lookup schemes—Elevator-Stairs algorithm and logW-Elevators algorithm. For a routing table with N prefixes, The Elevator-Stairs algorithm uses optimal O(N) memory, and achieves better lookup and update times than other methods with similar memory requirements. The logW-Elevators algorithm gives O(log W) lookup time, where W is the length of an IP address, while improving upon update time and memory usage. Experimental results using the MAE-West router with 29487 prefixes show that the Elevator. Stairs algorithm gives an average throughput of 15.7 Million lookups per second (Mlps) using 459 KB of memory, and the logW-Elevators algorithm gives an average throughput of 21.41 Mips with a memory usage of 1259 KB. [ABSTRACT FROM AUTHOR]

The development and adoption of new protocols (or of extensions to existing protocols) is arguably central to the Internet’s evolution. However, and in spite of over 40 years of experience with this process, we have limited understanding of what factors may contribute to a protocol’s success. A sound technical design and a well-grounded purpose are obviously important, but we have many examples of failures that met those two criteria. What other factors affect a protocol’s likelihood of success, and under what circumstances? We investigate this question through a statistical approach, based on analyzing a set of about 250 Internet standard documents, Internet engineering task force request for comments (RFCs). We characterize these RFCs using a number of key features, which we then seek to associate with positive or negative odds when it comes to success. Our high-level results are intuitive, e.g., protocols that call for Internet-wide adoption face greater challenges. Focusing on more targeted subsets of protocols reveals more subtle and possibly more interesting differences between areas of the Internet landscape. We also apply our prediction framework to IPv6, and use different “what-if” scenarios to explore what might have affected its deployment. [ABSTRACT FROM PUBLISHER]

Orthogonal frequency-division multiple access (OFDMA) is one of the most important modulation and access methods for the future mobile networks. Before transmitting a frame on the downlink, an OFDMA base station has to invoke an algorithm that determines which of the pending packets will be transmitted, what modulation should be used for each of them, and how to construct the complex OFDMA frame matrix as a collection of rectangles that fit into a single matrix with fixed dimensions. We propose efficient algorithms, with performance guarantee, that solve this intricate OFDMA scheduling problem by breaking it down into two subproblems, referred to as macro and micro scheduling. We analyze the computational complexity of these subproblems and develop efficient algorithms for solving them. [ABSTRACT FROM AUTHOR]

Existing Internet protocols rely on cooperative behavior of end users. We present a control-theoretic algorithm to counteract uncooperative users which change their congestion control schemes to gain larger bandwidth. This algorithm rectifies uncooperative users; that is, forces them to comply with their fair share, by adjusting the prices fed back to them. It is to be implemented at the edge of the network (e.g., by ISPs), and can be used with any congestion notification policy deployed by the network. Our design achieves a separation of time-scales between the network congestion feedback loop and the price-adjustment loop, thus recovering the fair allocation of bandwidth upon a fast transient phase. [ABSTRACT FROM AUTHOR]

The Differentiated Services framework (DiffServ) has been proposed to provide multiple Quality of Service (QoS) classes over IP networks. A network supporting multiple classes of service also requires a differentiated pricing structure. In this work, we propose a pricing algorithm in a DiffServ environment based on the cost of providing different levels of services, and on long-term average user resource demand of a service class. We integrate the proposed service-dependent pricing scheme with a dynamic pricing and service negotiation environment by considering a dynamic and congestion-sensitive pricing component. Pricing network services dynamically based on the level of service, usage, and congestion allows a more competitive price to be offered, allows the network to be used more efficiently, and provides a natural and equitable incentive for applications to adapt their service requests according to network conditions. We also develop the demand behavior of adaptive users based on a physically reasonable user utility function. Simulation results show that a congestion-sensitive pricing policy coupled with user rate adaptation is able to control congestion and allows a service class to meet its performance assurances under large or bursty offered loads, even without explicit admission control. Users are able to maintain a stable expenditure, and allowing users to migrate between service classes in response to price increases further stabilizes the individual service prices. When admission control is enforced, congestion-sensitive pricing still provides an advantage in terms of a much lower connection blocking rate at high loads. [ABSTRACT FROM AUTHOR]

We present a thorough characterization of what we believe to be the first significant live Internet streaming media workload in the scientific literature. Our characterization of over 3.5 million requests spanning a 28-day period is done at three increasingly granular levels, corresponding to clients, sessions, and transfers. Our findings support two important conclusions. First, we show that the nature of interactions between users and objects is fundamentally different for live versus stored objects. Access to stored objects is user driven, whereas access to live objects is object driven. This reversal of active/passive roles of users and objects leads to interesting dualities. For instance, our analysis underscores a Zipf-like profile for user interest in a given object, which is in contrast to the classic Zipf-like popularity of objects for a given user. Also, our analysis reveals that transfer lengths are highly variable and that this variability is due to client stickiness to a particular live object, as opposed to structural (size) properties of objects. Second, by contrasting two live streaming workloads from two radically different applications, we conjecture that some characteristics of live media access workloads are likely to be highly dependent on the nature of the live content being accessed. This dependence is clear from the strong temporal correlation observed in the traces, which we attribute to the impact of synchronous access to live content. Based on our analysis, we present a model for live media workload generation that incorporates many of our findings, and which we implement in GISMO. [ABSTRACT FROM AUTHOR]

It is generally admitted that interdomain peering links represent nowadays the main bottleneck of the Internet, particularly because of lack of coordination between providers, which use independent and “selfish” routing policies. We are interested in identifying possible “light” coordination strategies that would allow carriers to better control their peering links while preserving their independence and respective interests. We propose a robust multipath routing coordination framework for peering carriers, which relies on the multiple-exit discriminator (MED) attribute of Border Gateway Protocol (BGP) as signaling medium. Our scheme relies on a game theory modeling, with a non-cooperative potential game considering both routing and congestions costs. Peering equilibrium multipath (PEMP) coordination policies can be implemented by selecting Pareto-superior Nash equilibria at each carrier. We compare different PEMP policies to BGP Multipath schemes by emulating a realistic peering scenario. Our results show that the routing cost can be decreased by roughly 10% with PEMP. We also show that the stability of routes can be significantly improved and that congestion can be practically avoided on the peering links. Finally, we discuss practical implementation aspects and extend the model to multiple players highlighting the possible incentives for the resulting extended peering framework. [ABSTRACT FROM PUBLISHER]

Availability in connection-oriented service in networks has traditionally been "all-or-nothing," i.e., when a failure occurs, a connection either is unprotected or fully protected. The differences in availability and cost between these two extremes can be quite high. A general framework for service availability will be presented that fills the gap. It is shown how network resources and cost are related to service parameters of the framework for networks that are bandwidth-efficient. In addition, a simple revenue model is presented and characterized, revealing when nontraditional service agreements may be attractive. [ABSTRACT FROM AUTHOR]

A novel storage design for IP routing table construction is introduced on the basis of a single set-associative hash table to support fast longest prefix matching (LPM). The proposed design involves two key techniques to lower table storage required drastically: 1) storing transformed prefix representations; and 2) accommodating multiple prefixes per table entry via prefix aggregation, achieving superior storage-efficiency (SUSE). With each prefix (p(x)) maneuvered as a polynomial, p(x) = q(x) × g(x) + r(x) based on a divisor g(x), SUSE keeps only q(x) rather than full and long p(x) in an r(x)-indexed table with 2degree(g(x)) entries, because q(x) and r(x) uniquely identify p(x). Additionally, using r(x) as the hash index exhibits better distribution than do original prefixes, reducing hash collisions, which can be tolerated further by the set-associative design. Given a set of chosen prefix lengths (called "treads"), all prefixes are rounded down to nearest treads under SUSE before hashed to the table using their transformed representations so that prefix aggregation opportunities abound in hash entries. SUSE yields significant table storage reduction and enjoys fast lookups and speedy incremental updates not possible for a typical trie-based design, with the worst-case lookup time shown upper-bounded theoretically by the number of treads (ς) but found experimentally to be 4 memory accesses when ς equals 8. SUSE makes it possible to fit a large routing table with 256 K (or even 1 M) prefixes in on-chip SRAM by today's ASIC technology. It solves both the memory- and the bandwidth-intensive problems faced by IP routing. [ABSTRACT FROM AUTHOR]

Flooding is a commonly used technique for network resource and topology information dissemination in the data communication networks. However, due to the well-known N-squared problem it causes network delay in response or even congestion. We propose a new flooding technique, called Lazy Flooding; it floods only when links reach a certain status. It significantly cuts down the number of floods and thus improves the data communication network response time. On the other hand, it has negligible effect on the network performance due to the selected flooding. [ABSTRACT FROM AUTHOR]

We consider a problem motivated by the desire to provide flexible, rate-based, quality of service guarantees for packets sent over input queued switches and switch networks. Our focus is solving a type of online traffic scheduling problem, whose input at each time step is a set of desired traffic rates through the switch network. These traffic rates in general cannot be exactly achieved since they assume arbitrarily small fractions of packets can be transmitted at each time step. The goal of the traffic scheduling problem is to closely approximate the given sequence of traffic rates by a sequence of transmissions in which only whole packets are sent. We prove worst-case bounds on the additional buffer use, which we call backlog, that results from using such an approximation. We first consider the N × N, input queued, crossbar switch. Our main result is an online packet-scheduling algorithm using no speedup that guarantees backlog at most (N + 1)²/4 packets at each input port and each output port. Upper bounds on worst-case backlog have been proved for the case of constant fluid schedules, such as the N² - 2N + 2 bound of Chang, Chen, and Huang (INFOCOM, 2000). Our main result for the crossbar switch is the first, to our knowledge, to bound backlog in terms of switch size N for arbitrary, time-varying fluid schedules, without using speedup. Our main result for Banyan networks is an exact characterization of the speedup required to maintain bounded backlog, in terms of polytopes derived from the network topology. [ABSTRACT FROM AUTHOR]

Geolocation of Internet hosts enables a new class of location-aware applications. Previous measurement-based approaches use reference hosts, called landmarks, with a well-known geographic location to provide the location estimation of a target host. This leads to a discrete space of answers, limiting the number of possible location estimates to the number of adopted landmarks. In contrast, we propose Constraint-Based Geolocation (CBG), which infers the geographic location of Internet hosts using multilateration with distance constraints to establish a continuous space of answers instead of a discrete one. However, to use multilateration in the Internet, the geographic distances from the landmarks to the target host have to be estimated based on delay measurements between these hosts. This is a challenging problem because the relationship between network delay and geographic distance in the Internet is perturbed by many factors, including queueing delays and the absence of great-circle paths between hosts. CBG accurately transforms delay measurements to geographic distance constraints, and then uses multilateration to infer the geolocation of the target host. Our experimental results show that CBG outperforms previous geolocation techniques. Moreover, in contrast to previous approaches, our method is able to assign a confidence region to each given location estimate. This allows a location-aware application to assess whether the location estimate is sufficiently accurate for its needs. [ABSTRACT FROM AUTHOR]