Showing papers in "IEEE ACM Transactions on Networking in 2002"

Mobility increases the capacity of ad hoc wireless networks

Abstract: The capacity of ad hoc wireless networks is constrained by the mutual interference of concurrent transmissions between nodes. We study a model of an ad hoc network where n nodes communicate in random source-destination pairs. These nodes are assumed to be mobile. We examine the per-session throughput for applications with loose delay constraints, such that the topology changes over the time-scale of packet delivery. Under this assumption, the per-user throughput can increase dramatically when nodes are mobile rather than fixed. This improvement can be achieved by exploiting a form of multiuser diversity via packet relaying.

Bandwidth sharing: objectives and algorithms

Abstract: This paper concerns the design of distributed algorithms for sharing network bandwidth resources among contending flows. The classical fairness notion is the so-called max-min fairness. The alternative proportional fairness criterion has recently been introduced by F. Kelly (see Eur. Trans. Telecommun., vol.8, p.33-7, 1997); we introduce a third criterion, which is naturally interpreted in terms of the delays experienced by ongoing transfers. We prove that fixed-size window control can achieve fair bandwidth sharing according to any of these criteria, provided scheduling at each link is performed in an appropriate manner. We then consider a distributed random scheme where each traffic source varies its sending rate randomly, based on binary feedback information from the network. We show how to select the source behavior so as to achieve an equilibrium distribution concentrated around the considered fair rate allocations. This stochastic analysis is then used to assess the asymptotic behavior of deterministic rate adaption procedures.

The BLUE active queue management algorithms

Abstract: In order to stem the increasing packet loss rates caused by an exponential increase in network traffic, the IETF has been considering the deployment of active queue management techniques such as RED (random early detection) (see Floyd, S. and Jacobson, V., IEEE/ACM Trans. Networking, vol.1, p.397-413, 1993). While active queue management can potentially reduce packet loss rates in the Internet, we show that current techniques are ineffective in preventing high loss rates. The inherent problem with these algorithms is that they use queue lengths as the indicator of the severity of congestion. In light of this observation, a fundamentally different active queue management algorithm, called BLUE, is proposed, implemented and evaluated. BLUE uses packet loss and link idle events to manage congestion. Using both simulation and controlled experiments, BLUE is shown to perform significantly better than RED, both in terms of packet loss rates and buffer size requirements in the network. As an extension to BLUE, a novel technique based on Bloom filters (see Bloom, B., Commun. ACM, vol.13, no.7, p.422-6, 1970) is described for enforcing fairness among a large number of flows. In particular, we propose and evaluate stochastic fair BLUE (SFB), a queue management algorithm which can identify and rate-limit nonresponsive flows using a very small amount of state information.

Compressed Bloom filters

Abstract: A Bloom filter is a simple space-efficient randomized data structure for representing a set in order to support membership queries. Although Bloom filters allow false positives, for many applications the space savings outweigh this drawback when the probability of an error is sufficiently low. We introduce compressed Bloom filters, which improve performance when the Bloom filter is passed as a message, and its transmission size is a limiting factor. For example, Bloom filters have been suggested as a means for sharing Web cache information. In this setting, proxies do not share the exact contents of their caches, but instead periodically broadcast Bloom filters representing their caches. By using compressed Bloom filters, proxies can reduce the number of bits broadcast, the false positive probability, and/or the amount of computation per lookup. The cost is the processing time for compression and decompression, which can use simple arithmetic coding, and more memory use at the proxies, which utilize the larger uncompressed form of the Bloom filter.

Energy-efficient packet transmission over a wireless link

Abstract: The paper considers the problem of minimizing the energy used to transmit packets over a wireless link via lazy schedules that judiciously vary packet transmission times. The problem is motivated by the following observation. With many channel coding schemes, the energy required to transmit a packet can be significantly reduced by lowering transmission power and code rate, and therefore transmitting the packet over a longer period of time. However, information is often time-critical or delay-sensitive and transmission times cannot be made arbitrarily long. We therefore consider packet transmission schedules that minimize energy subject to a deadline or a delay constraint. Specifically, we obtain an optimal offline schedule for a node operating under a deadline constraint. An inspection of the form of this schedule naturally leads us to an online schedule which is shown, through simulations, to perform closely to the optimal offline schedule. Taking the deadline to infinity, we provide an exact probabilistic analysis of our offline scheduling algorithm. The results of this analysis enable us to devise a lazy online algorithm that varies transmission times according to backlog. We show that this lazy schedule is significantly more energy-efficient compared to a deterministic (fixed transmission time) schedule that guarantees queue stability for the same range of arrival rates.

The stable paths problem and interdomain routing

Abstract: Dynamic routing protocols such as RIP and OSPF essentially implement distributed algorithms for solving the shortest paths problem. The border gateway protocol (BGP) is currently the only interdomain routing protocol deployed in the Internet. BGP does not solve a shortest paths problem since any interdomain protocol is required to allow policy-based metrics to override distance-based metrics and enable autonomous systems to independently define their routing policies with little or no global coordination. It is then natural to ask if BGP can be viewed as a distributed algorithm for solving some fundamental problem. We introduce the stable paths problem and show that BGP can be viewed as a distributed algorithm for solving this problem. Unlike a shortest path tree, such a solution does not represent a global optimum, but rather an equilibrium point in which each node is assigned its local optimum. We study the stable paths problem using a derived structure called a dispute wheel, representing conflicting routing policies at various nodes. We show that if no dispute wheel can be constructed, then there exists a unique solution for the stable paths problem. We define the simple path vector protocol (SPVP), a distributed algorithm for solving the stable paths problem. SPVP is intended to capture the dynamic behavior of BGP at an abstract level. If SPVP converges, then the resulting state corresponds to a stable paths solution. If there is no solution, then SPVP always diverges. In fact, SPVP can even diverge when a solution exists. We show that SPVP will converge to the unique solution of an instance of the stable paths problem if no dispute wheel exists.

Single-packet IP traceback

Abstract: The design of the IP protocol makes it difficult to reliably identify the originator of an IP packet Even in the absence of any deliberate attempt to disguise a packet's origin, widespread packet forwarding techniques such as NAT and encapsulation may obscure the packet's true source Techniques have been developed to determine the source of large packet flows, but, to date, no system has been presented to track individual packets in an efficient, scalable fashion We present a hash-based technique for IP traceback that generates audit trails for traffic within the network, and can trace the origin of a single IP packet delivered by the network in the recent past We demonstrate that the system is effective, space efficient (requiring approximately 05% of the link capacity per unit time in storage), and implementable in current or next-generation routing hardware We present both analytic and simulation results showing the system's effectiveness

Proportional differentiated services: delay differentiation and packet scheduling

Abstract: The proportional differentiation model provides the network operator with the 'tuning knobs' for adjusting the per-hop quality-of-service (QoS) ratios between classes, independent of the class loads. This paper applies the proportional model in the differentiation of queueing delays, and investigates appropriate packet scheduling mechanisms. Starting from the proportional delay differentiation (PDD) model, we derive the average queueing delay in each class, show the dynamics of the class delays under the PDD constraints, and state the conditions in which the PDD model is feasible. The feasibility model of the model can be determined from the average delays that result with the strict priorities scheduler. We then focus on scheduling mechanisms that can implement the PDD model, when it is feasible to do so. The proportional average delay (PAD) scheduler meets the PDD constraints, when they are feasible, but it exhibits a pathological behavior in short timescales. The waiting time priority (WTP) scheduler, on the other hand, approximates the PDD model closely, even in the short timescales of a few packet departures, but only in heavy load conditions. PAD and WTP serve as motivation for the third scheduler, called hybrid proportional delay (HPD). HPD approximates the PDD model closely, when the model is feasible, independent of the class load distribution. Also, HPD provides predictable delay differentiation even in short timescales.

DNS performance and the effectiveness of caching

Abstract: This paper presents a detailed analysis of traces of domain name system (DNS) and associated TCP traffic collected on the Internet links of the MIT Laboratory for Computer Science and the Korea Advanced Institute of Science and Technology (KAIST). The first part of the analysis details how clients at these institutions interact with the wide-area domain name system, focusing on client-perceived performance and the prevalence of failures and errors. The second part evaluates the effectiveness of DNS caching. In the most recent MIT trace, 23% of lookups receive no answer; these lookups account for more than half of all traced DNS packets since query packets are retransmitted overly persistently. About 13% of all lookups result in an answer that indicates an error condition. Many of these errors appear to be caused by missing inverse (IP-to-name) mappings or NS records that point to nonexistent or inappropriate hosts. 27% of the queries sent to the root name servers result in such errors. The paper also presents the results of trace-driven simulations that explore the effect of varying time-to-live (TTL) and varying degrees of cache sharing on DNS cache hit rates. Due to the heavy-tailed nature of name accesses, reducing the TTL of address (A) records to as low as a few hundred seconds has little adverse effect on hit rates, and little benefit is obtained from sharing a forwarding DNS cache among more than 10 or 20 clients. These results suggest that client latency is not as dependent on aggressive caching as is commonly believed, and that the widespread use of dynamic low-TTL A-record bindings should not greatly increase DNS related wide-area network traffic.

Utility-based rate control in the Internet for elastic traffic

Abstract: In a communication network, a good rate allocation algorithm should reflect the utilities of the users while being fair. We investigate this fundamental problem of achieving the system optimal rates in the sense of maximizing aggregate utility, in a distributed manner, using only the information available at the end hosts of the network. This is done by decomposing the overall system problem into subproblems for the network and for the individual users by introducing a pricing scheme. The users are to solve the problem of maximizing individual net utility, which is the utility less the amount they pay. We provide algorithms for the network to adjust its prices and the users to adjust their window sizes such that at an equilibrium the system optimum is achieved. Further, the equilibrium prices are such that the system optimum achieves weighted proportional fairness. It is notable that the update algorithms of the users do not require any explicit feedback from the network, rendering them easily deployable over the Internet. Our scheme is incentive compatible in that there is no benefit to the users to lie about their utilities.

Algebra and algorithms for QoS path computation and hop-by-hop routing in the Internet

Abstract: Prompted by the advent of quality-of-service routing in the Internet, we investigate the properties that path weight functions must have so that hop-by-hop routing is possible and optimal paths can be computed with a generalization of E.W. Dijkstra's algorithm (see Numer. Math., vol.1, p.269-71, 1959). We define an algebra of weights which contains a binary operation, for the composition of link weights into path weights, and an order relation. Isotonicity is the key property of the algebra. It states that the order relation between the weights of any two paths is preserved if both of them are either prefixed or appended by a common, third, path. We show that isotonicity is both necessary and sufficient for a generalized Dijkstra's algorithm to yield optimal paths. Likewise, isotonicity is also both necessary and sufficient for hop-by-hop routing. However, without strict isotonicity, hop-by-hop routing based on optimal paths may produce routing loops. They are prevented if every node computes what we call lexicographic-optimal paths. These paths can be computed with an enhanced Dijkstra's algorithm that has the same complexity as the standard one. Our findings are extended to multipath routing as well. As special cases of the general approach, we conclude that shortest-widest paths can neither be computed with a generalized Dijkstra's algorithm nor can packets be routed hop-by-hop over those paths. In addition, loop-free hop-by-hop routing over widest and widest-shortest paths requires each node to compute lexicographic-optimal paths, in general.

Multicast-based inference of network-internal delay distributions

Abstract: Packet delay greatly influences the overall performance of network applications. It is therefore important to identify causes and locations of delay performance degradation within a network. Existing techniques, largely based on end-to-end delay measurements of unicast traffic, are well suited to monitor and characterize the behavior of particular end-to-end paths. Within these approaches, however, it is not clear how to apportion the variable component of end-to-end delay as queueing delay at each link along a path. Moreover, there are issues of scalability for large networks. In this paper, we show how end-to-end measurements of multicast traffic can be used to infer the packet delay distribution and utilization on each link of a logical multicast tree. The idea, recently introduced in Caceres et al. (1999), is to exploit the inherent correlation between multicast observations to infer performance of paths between branch points in a tree spanning a multicast source and its receivers. The method does not depend on cooperation from intervening network elements; because of the bandwidth efficiency of multicast traffic, it is suitable for large-scale measurements of both end-to-end and internal network dynamics. We establish desirable statistical properties of the estimator, namely consistency and asymptotic normality. We evaluate the estimator through simulation and observe that it is robust with respect to moderate violations of the underlying model.

Fixed-alternate routing and wavelength conversion in wavelength-routed optical networks

Abstract: Consider an optical network which employs wavelength-routing crossconnects that enable the establishment of wavelength-division-multiplexed (WDM) connections between node pairs. In such a network, when there is no wavelength conversion, a connection is constrained to be on the same wavelength channel along its route. Alternate routing can improve the blocking performance of such a network by providing multiple possible paths between node pairs. Wavelength conversion can also improve the blocking performance of such a network by allowing a connection to use different wavelengths along its route. This work proposes an approximate analytical model that incorporates alternate routing and sparse wavelength conversion. We perform simulation studies of the relationships between alternate routing and wavelength conversion on three representative network topologies. We demonstrate that alternate routing generally provides significant benefits, and that it is important to design alternate routes between node pairs in an optimized fashion to exploit the connectivity of the network topology. The empirical results also indicate that fixed-alternate routing with a small number of alternate routes asymptotically approaches adaptive routing in blocking performance.

Dynamic parallel access to replicated content in the Internet

Abstract: Popular content is frequently replicated in multiple servers or caches in the Internet to offload origin servers and improve end-user experience. However, choosing the best server is a nontrivial task and a bad choice may provide poor end user experience. In contrast to retrieving a file from a single server, we propose a parallel-access scheme where end users access multiple servers at the same time, fetching different portions of that file from different servers and reassembling them locally. The amount of data retrieved from a particular server depends on the resources available at that server or along the path from the user to the server. Faster servers will deliver bigger portions of a file while slower servers will deliver smaller portions. If the available resources at a server or along the path change during the download of a file, a dynamic parallel access will automatically shift the load from congested locations to less loaded parts (server and links) of the Internet. The end result is that users experience significant speedups and very consistent response times. Moreover, there is no need for complicated server selection algorithms and load is dynamically shared among all servers. The dynamic parallel-access scheme presented in this paper does not require any modifications to servers or content and can be easily included in browsers, peer-to-peer applications or content distribution networks to speed up delivery of popular content.

Resource management with hoses: point-to-cloud services for virtual private networks

Abstract: As IP technologies providing both tremendous capacity and the ability to establish dynamic security associations between endpoints emerge, virtual private networks (VPNs) are going through dramatic growth. The number of endpoints per VPN is growing and the communication pattern between endpoints is becoming increasingly hard to predict. Consequently, users are demanding dependable, dynamic connectivity between endpoints, with the network expected to accommodate any traffic matrix, as long as the traffic to the endpoints does not overwhelm the capacity of the respective ingress and egress links. We propose a new service interface, termed a hose, to provide the appropriate performance abstraction. A hose is characterized by the aggregate traffic to and from one endpoint in the VPN to a set of other endpoints in the VPN, and by an associated performance guarantee. Hoses provide important advantages to a VPN customer: (1) flexibility to send traffic to a set of endpoints without having to specify the detailed traffic matrix, and (2) reduction in the size of access links through multiplexing gains obtained from the natural aggregation of the flows between endpoints. As compared with the conventional point-to-point (or customer pipe) model for managing quality of service (QoS), hoses provide reduction in the state information a customer must maintain. On the other hand, hoses would appear to increase the complexity of the already difficult problem of resource management to support QoS. To manage network resources in the face of this increased uncertainty, we consider both conventional statistical multiplexing techniques, and a new resizing technique based on online measurements. To study these performance issues, we run trace-driven simulations, using traffic derived from AT&T's voice network and from a large corporate data network. From the customer's perspective, we find that aggregation of traffic at the hose level provides significant multiplexing gains. From the provider's perspective, we find that the statistical multiplexing and resizing techniques deal effectively with uncertainties about the traffic, providing significant gains over the conventional alternative of a mesh of statically sized customer pipes between endpoints.

Detecting shared congestion of flows via end-to-end measurement

Abstract: Current Internet congestion control protocols operate independently on a per-flow basis. Recent work has demonstrated that cooperative congestion control strategies between flows can improve performance for a variety of applications, ranging from aggregated TCP transmissions to multiple-sender multicast applications. However, in order for this cooperation to be effective, one must first identify the flows that are congested at the same set of resources. In this paper, we present techniques based on loss or delay observations at end hosts to infer whether or not two flows experiencing congestion are congested at the same network resources. Our novel result is that such detection can be achieved for unicast flows, but the techniques can also be applied to multicast flows. We validate these techniques via queueing analysis, simulation, and experimentation within the Internet. In addition, we demonstrate preliminary simulation results that show that the delay-based technique can determine whether two TCP flows are congested at the same set of resources. We also propose metrics that can be used as a measure of the amount of congestion sharing between two flows.

Heuristic algorithms for multiconstrained quality-of-service routing

Abstract: Multiconstrained quality-of-service (QoS) routing deals with finding routes that satisfy multiple independent QoS constraints. This problem is NP-hard. In this paper, two heuristics, the limited granularity heuristic and the limited path heuristic, are investigated. Both heuristics extend the Bellman-Ford shortest path algorithm and solve general k-constrained QoS routing problems. Analytical and simulation studies are conducted to compare the time/space requirements of the heuristics and the effectiveness of the heuristics in finding paths that satisfy the QoS constraints. The major results of this paper are the following. For an N-nodes and E-edges network with k (a small constant) independent QoS constraints, the limited granularity heuristic must maintain a table of size O(|N|k- 1) in each node to be effective, which results in a time complexity of O (|N|K|E|); while the limited path heuristic can achieve very high performance by maintaining O (|N|2lg(|N|)) entries in each node. These results indicate that the limited path heuristic is relatively insensitive to the number of constraints and is superior to the limited granularity heuristic in solving k-constrained QoS routing problems when k > 3.

MLSR: a novel routing algorithm for multilayered satellite IP networks

Abstract: Several IP-based routing algorithms have been developed for low-Earth orbit (LEO) satellite networks in recent years. The performance of the satellite IP networks can be improved drastically if multiple satellite constellations are used in the architecture. In this work, a multilayered satellite IP network is introduced that consists of LEO, medium-Earth orbit (MEO), and geostationary Earth orbit (GEO) satellites. A new multilayered satellite routing algorithm (MLSR) is developed that calculates routing tables efficiently using the collected delay measurements. The performance of the multilayered satellite network and MLSR is evaluated through simulations and analysis.

Algorithms for provisioning virtual private networks in the hose model

Abstract: Virtual Private Networks (VPNs) provide customers with predictable and secure network connections over a shared network. The recently proposed hose model for VPNs allows for greater flexibility since it permits traffic to and from a hose endpoint to be arbitrarily distributed to other endpoints. In this paper, we develop novel algorithms for provisioning VPNs in the hose model. We connect VPN endpoints using a tree structure and our algorithms attempt to optimize the total bandwidth reserved on edges of the VPN tree. We show that even for the simple scenario in which network links are assumed to have infinite capacity, the general problem of computing the optimal VPN tree is NP-hard. Fortunately, for the special case when the ingress and egress bandwidths for each VPN endpoint are equal, we can devise an algorithm for computing the optimal tree whose time complexity is O(mn), where m and n are the number of links and nodes in the network, respectively. We present a novel integer programming formulation for the general VPN tree computation problem (that is, when ingress and egress bandwidths of VPN endpoints are arbitrary) and develop an algorithm that is based on the primal-dual method. Our experimental results with synthetic network graphs indicate that the VPN trees constructed by our proposed algorithms dramatically reduce bandwidth requirements (in many instances, by more than a factor of 2) compared to scenarios in which Steiner trees are employed to connect VPN endpoints.

Internet pricing with a game theoretical approach: concepts and examples

Abstract: The basic concepts of three branches of game theory, leader-follower, cooperative, and two-person nonzero sum games, are reviewed and applied to the study of the Internet pricing issue. In particular, we emphasize that the cooperative game (also called the bargaining problem) provides an overall picture for the issue. With a simple model for Internet quality of service (QoS), we demonstrate that the leader-follower game may lead to a solution that is not Pareto optimal and in some cases may be "unfair," and that the cooperative game may provide a better solution for both the Internet service provider (ISP) and the user. The practical implication of the results is that government regulation or arbitration may be helpful. The QoS model is also applied to study the competition between two ISPs, and we find a Nash equilibrium point from which the two ISPs would not move out without cooperation. The proposed approaches can be applied to other Internet pricing problems such as the Paris Metro pricing scheme.

A dynamic call admission policy with precision QoS guarantee using stochastic control for mobile wireless networks

Abstract: Call admission control is one of the key elements in ensuring the quality of service in mobile wireless networks. The traditional trunk reservation policy and its numerous variants give preferential treatment to the handoff calls over new arrivals by reserving a number of radio channels exclusively for handoffs. Such schemes, however, cannot adapt to changes in traffic pattern due to the static nature. This paper introduces a novel stable dynamic call admission control mechanism (SDCA), which can maximize the radio channel utilization subject to a predetermined bound on the call dropping probability. The novelties of the proposed mechanism are: (1) it is adaptive to wide range of system parameters and traffic conditions due to its dynamic nature; (2) the control is stable under overloading traffic conditions, thus can effectively deal with sudden traffic surges; (3) the admission policy is stochastic, thus spreading new arrivals evenly over a control period, and resulting in more effective and accurate control; and (4) the model takes into account the effects of limited channel capacity and time dependence on the call dropping probability, and the influences from nearest and next-nearest neighboring cells, which greatly improve the control precision. In addition, we introduce local control algorithms based on strictly local estimations of the needed traffic parameters, without requiring the status information exchange among different cells, which makes it very appealing in actual implementation. Most of the computational complexities lie in off-line precalculations, except for the nonlinear equation of the acceptance ratio, in which a coarse-grain numerical integration is shown to be sufficient for stochastic control. Extensive simulation results show that our scheme steadily satisfies the hard constraint on call dropping probability while maintaining a high channel throughput.

Concurrent round-robin-based dispatching schemes for Clos-network switches

Abstract: A Clos-network switch architecture is attractive because of its scalability. Previously proposed implementable dispatching schemes from the first stage to the second stage, such as random dispatching (RD), are not able to achieve high throughput unless the internal bandwidth is expanded. This paper presents two round-robin-based dispatching schemes to overcome the throughput limitation of the RD scheme. First, we introduce a concurrent round-robin dispatching (CRRD) scheme for the Clos-network switch. The CRRD scheme provides high switch throughput without expanding internal bandwidth. CRRD implementation is very simple because only simple round-robin arbiters are adopted. We show via simulation that CRRD achieves 100% throughput under uniform traffic. When the offered load reaches 1.0, the pointers of round-robin arbiters at the first- and second-stage modules are completely desynchronized and contention is avoided. Second, we introduce a concurrent master-slave round-robin dispatching (CMSD) scheme as an improved version of CRRD to make it more scalable. CMSD uses hierarchical round-robin arbitration. We show that CMSD preserves the advantages of CRRD, reduces the scheduling time by 30% or more when arbitration time is significant and has a dramatically reduced number of crosspoints of the interconnection wires between round-robin arbiters in the dispatching scheduler with a ratio of 1//spl radic/N, where N is the switch size. This makes CMSD easier to implement than CRRD when the switch size becomes large.

Adaptive proportional routing: a localized QoS routing approach

Abstract: Most of the QoS routing schemes proposed so far require periodic exchange of QoS state information among routers, imposing both communication overhead on the network and processing overhead on core routers. Furthermore, stale QoS state information causes the performance of these QoS routing schemes to degrade drastically. In order to circumvent these problems, we focus on localized QoS routing schemes where the edge routers make routing decisions using only local information and thus reducing the overhead at core routers. We first describe virtual capacity based routing (vcr), a theoretical scheme based on the notion of virtual capacity of a route. We then propose proportional sticky routing, an easily realizable approximation of vcr and analyze its performance. We demonstrate through extensive simulations that adaptive proportional routing is indeed a viable alternative to the global QoS routing approach.

Computing shortest paths for any number of hops

Abstract: In this paper, we introduce and investigate a "new" path optimization problem that we denote the all hops optimal path (AHOP) problem. The problem involves identifying, for all hop counts, the optimal, i.e., minimum weight, path(s) between a given source and destination(s). The AHOP problem arises naturally in the context of quality-of-service (QoS) routing in networks, where routes (paths) need to be computed that provide services guarantees, e.g., delay or bandwidth, at the minimum possible "cost" (amount of resources required) to the network. Because service guarantees are typically provided through some form of resource allocation on the path (links) computed for a new request, the hop count, which captures the number of links over which resources are allocated, is a commonly used cost measure. As a result, a standard approach for determining the cheapest path available that meets a desired level of service guarantees is to compute a minimum hop shortest (optimal) path. Furthermore, for efficiency purposes, it is desirable to precompute such optimal minimum hop paths for all possible service requests. Providing this information gives rise to solving the AHOP problem. The paper's contributions are to investigate the computational complexity of solving the AHOP problem for two of the most prevalent cost functions (path weights) in networks, namely, additive and bottleneck weights. In particular, we establish that a solution based on the Bellman-Ford algorithm is optimal for additive weights, but show that this does not hold for bottleneck weights for which a lower complexity solution exists.

Delay jitter bounds and packet scale rate guarantee for expedited forwarding

Abstract: We consider the definition of the expedited forwarding per-hop behavior (EF PHB) as given in RFC 2598 and its impact on worst case end-to-end delay jitter. On the one hand, the definition in RFC 2598 can be used to predict extremely low end-to-end delay jitter, independent of the network scale. On the other hand, we find that the worst case delay jitter can be made arbitrarily large, while each flow traverses at most a specified number of hops, if we allow networks to become arbitrarily large; this is in contradiction with the previous statement. We analyze where the contradiction originates and find the explanation. It resides in the fact that the definition in RFC 2598 is not easily implementable in schedulers we know of, mainly because it is not formal enough, and also because it does not contain an error term. We propose a new definition for the EF PHB, called "packet scale rate guarantee" (PSRG) that preserves the spirit of RFC 2598 while allowing a number of reasonable implementations and has very useful properties for per-node and end-to-end network engineering. We show that this definition implies a rate-latency service curve property. We also show that it is equivalent, in some sense, to the stronger concept of "adaptive service guarantee." Then we propose some proven bounds on delay jitter for networks implementing this new definition, both in cases without loss and with loss.

Pricing in multiservice loss networks: static pricing, asymptotic optimality and demand substitution effects

Abstract: We consider a communication network with fixed routing that can accommodate multiple service classes, differing in bandwidth requirements, demand pattern, call duration and routing. The network charges a fee per call which can depend on the current congestion level and which affects user's demand. Building on the single-node results of I.Ch. Paschalidis and J.N. Tsitsiklis (see IEEE/ACM Trans. Networking, vol.8, p.171-84, 2000), we consider both problems of revenue and of welfare maximization, and show that static pricing is asymptotically optimal in a regime of many, relatively small, users. In particular, the performance of an optimal (dynamic) pricing strategy is closely matched by a suitably chosen class-dependent static price, which does not depend on instantaneous congestion. This result holds even when we incorporate demand substitution effects into the demand model. More specifically, we model the situation where price increases for a class of service might lead users to use another class as an imperfect substitute. For both revenue and welfare maximization objectives we characterize the structure of the asymptotically optimal static prices, expressing them as a function of a parsimonious number of parameters. We employ a simulation-based approach to tune those parameters and to compute efficiently an effective policy away from the limiting regime. Our approach can handle large, realistic, instances of the problem.

Generalized loop-back recovery in optical mesh networks

Abstract: Current means of providing loop-back recovery, which is widely used in SONET, rely on ring topologies, or on overlaying logical ring topologies upon physical meshes. Loop-back is desirable to provide rapid preplanned recovery of link or node failures in a bandwidth-efficient distributed manner. We introduce generalized loop-back, a novel scheme for performing loop-back in optical mesh networks. We present an algorithm to perform recovery for link failure and one to perform generalized loop-back recovery for node failure. We illustrate the operation of both algorithms, prove their validity, and present a network management protocol algorithm, which enables distributed operation for link or node failure. We present three different applications of generalized loop-back. First, we present heuristic algorithms for selecting recovery graphs, which maintain short maximum and average lengths of recovery paths. Second, we present WDM-based loop-back recovery for optical networks where wavelengths are used to back up other wavelengths. We compare, for WDM-based loop-back, the operation of generalized loop-back operation with known ring-based ways of providing loop-back recovery over mesh networks. Finally, we introduce the use of generalized loop-back to provide recovery in a way that allows dynamic choice of routes over preplanned directions.

Explicit window adaptation: a method to enhance TCP performance

Abstract: We study the performance of TCP in an internetwork consisting of both rate-controlled and nonrate-controlled segments. A common example of such an environment occurs when the end systems are part of IP datagram networks interconnected by a rate-controlled segment, such as an ATM network using the available bit rate (ABR) service. In the absence of congestive losses in either segment, TCP keeps increasing its window to its maximum size. Mismatch between the TCP window and the bandwidth-delay product of the network will result in accumulation of large queues and possibly buffer overflows in the devices at the edges of the rate-controlled segment, causing degraded throughput and unfairness. We develop an explicit feedback scheme, called Explicit Window Adaptation, based on modifying the receiver's advertised window in TCP acknowledgments returning to the source. The window size indicated to TCP is a function of the free buffer in the edge device. Results from simulations with a wide range of traffic scenarios show that this explicit window adaptation scheme can control the buffer occupancy efficiently at the edge device, and results in significant improvements in packet loss rate, fairness, and throughput over a packet discard policy such as Random Early Detection (RED).

A multicast routing algorithm for LEO satellite IP networks

Abstract: Satellite networks provide global coverage and support a wide range of services. Since Low Earth Orbit (LEO) satellites provide short round-trip delays, they are becoming increasingly important for real-time applications such as voice and video traffic. Many applications require a mechanism to deliver information to multiple recipients. In this paper, a multicast routing algorithm for datagram traffic is introduced for LEO satellite IP networks. The new scheme creates multicast trees by using the Datagram Routing Algorithm. The bandwidth utilization and delay characteristics are assessed through simulations.

Packet-mode scheduling in input-queued cell-based switches

Abstract: We consider input-queued switch architectures dealing at their interfaces with variable-size packets, but internally operating on fixed-size cells. Packets are segmented into cells at input ports, transferred through the switching fabric, and reassembled at output ports. Cell transfers are controlled by a scheduling algorithm, which operates in packet-mode: all cells belonging to the same packet are transferred from inputs to outputs without interruption. We prove that input-queued switches using packet-mode scheduling can achieve 100% throughput, and we show by simulation that, depending on the packet size distribution, packet-mode scheduling may provide advantages over cell-mode scheduling.