waashk/dblp huggingface dataset:
a reliable and parallel double-ring fddi metropolitan area network. the authors present a reliable double-ring metropolitan area network architecture, called dorfman, based on an enhanced fiber distributed data interface (fddi) protocol. dorfman is composed of two local ring networks interconnected via two monitors. the two monitors are used to perform simple routing and error recovery logic as well as mode transitions of dorfman. there are three modes defined in dorfman: local, global, and transition modes. during the local mode, dorfman allows double rings to operate in parallel. during the global mode, dorfman behaves as an fddi network. modes are dynamically switched between local and global modes on a demand basis. the bandwidth waste resulting from the transition of modes has been reduced to a minimum. performance analysis and simulation results show that with a reasonable degree of locality assumed, twice the throughput of an fddi network can be achieved in dorfman. an error recovery scheme for dorfman is addressed
an end-to-end qos framework with on-demand bandwidth reconfiguration. this paper proposes a new qos framework, called the on-demand qos path framework (odp). it provides end-to-end qos guarantees to individual flows with minimal overhead, while keeping the scalability characteristic of diffserv. odp exercises per-flow admission control and end-to-end bandwidth reservation at the edge of the network and only differentiates service types in the core of the network. in addition, to adapt to dynamically changing traffic load, odp monitors the bandwidth utilization of the network and performs dynamic bandwidth reconfiguration in the network core based on the monitored bandwidth utilization. through extensive simulations, the performance of odp is investigated and compared with that of intserv and diffserv frameworks. the simulation results clearly showed that odp provides end-to-end qos guarantees to individual flows, which diffserv can not provide, with much less overhead than intserv.
a tdm-based multibus packet switch. a novel packet switch architecture using two sets of time division multiplexed (tdm) buses is proposed. the horizontal buses collect packets from the input ports while the vertical buses distribute the packets to the output ports. the two sets of buses are connected by a set of switching elements which coordinate the connections between the horizontal buses and the vertical buses so that each vertical bus is connected to only one horizontal bus at a time. the switch has the advantages of: (1) it adds input and output ports without increasing the bus and i/o adaptor speed; (2) it is internally unbuffered; (3) it has a very simple control circuit; and (4) it has 100% potential throughput under uniform traffic. a combined analytical-simulation method is used to obtain the packet delay and packet loss probability. numerical results show that for satisfactory performance the buses need to run about 30% faster than the input line rate. with this speedup, even at a utilization factor of 0.9, the input queue can give a packet loss of 10 -6 with only 31 buffers per input adaptor. the output queue behaves essentially as an m/d/1 queue
transient behaviors of tcp-friendly congestion control protocols. we investigate the fairness, smoothness, responsiveness, and aggressiveness of tcp and three representative tcp-friendly congestion control protocols: gaimd, tfrc, and tear. the properties are evaluated both analytically and experimentally by studying protocol responses to three network environment changes. the first environment change is the inherent fluctuations in a stationary network environment. we consider three types of sending rate variations: smoothness, short-term fairness, and long-term fairness. for a stationary environment, we observe that smoothness and fairness are positively correlated. we derive an analytical expression for the sending rate coefficient of variation for each of the four protocols. these analytical results match well with experimental results. the other two environment changes we study are a step increase of network congestion and a step increase of available bandwidth. protocol responses to these changes reflect their responsiveness and aggressiveness, respectively.
alternate path routing for multicast. current network-layer multicast routing protocols build multicast trees based only on hop count and policy. if a tree cannot meet application requirements, the receivers have no alternative. in this paper, we propose a general and modular architecture that integrates alternate path routing with the network's multicast services. this enables individual multicast receivers to reroute a multicast tree according to their needs, subject to policy restrictions. our design focuses on the two primary components of this architecture--a loop-free path installation protocol and a scalable, distributed path computation algorithm. based on a simulation study, we demonstrate that using alternate path routing enables receivers to find acceptable paths nearly as well as a link-state protocol, with much lower overhead. we also show that our approach scales to large networks and that performance improves as a multicast group grows in size.
statistical synchronization among participants in real-time multimedia conference. in this paper, an algorithm to determine the set of packets generated continuously and periodically from different participants that are arriving at a node either for mixing at the master of a conference, or for simply playing back at a regular participant of a conference, is proposed. the essence of the algorithm is to estimate the average packet arrival time (or reference time) for each participant. with the reference time at hand, the maximum jitter and the optimum waiting time for a mixer to wait packets from all participants can be determined. the error of the proposed algorithm is enumerated by the chernoff bound and is shown to be acceptable in practical application.
finite buffer discrete-time queues with multiple markovian arrivals and services in atm networks. an efficient computational procedure to calculate the queue size distribution in a finite buffer queuing system with a number of independent sources is presented. each source is characterized by a finite state discrete-time markov chain. an explicit expression for the queue size distribution is obtained directly. the method is based on computing the modified spectral expansion of the state distribution of the system. this representation requires evaluating the roots of the entire system characteristic function and solving a set of linear system equations. exact results show that the cell loss probability depends only on the ratio of the buffer capacity to burst length. an approximation expression for the cell loss probability based on asymptotic analysis is also presented. numerical results show that the approximation is accurate, especially when the system load is heavy. the effect of burst length on the cell loss probability is investigated
logarithmically scalable routing algorithms in large optical networks. in this paper, we present three logarithmically scalable routing algorithms for very large optical networks. the algorithms are based on a hierarchical approach. each algorithm requires a different amount of information to be stored at each node, and results in a different efficiency. the network capacity provided by each algorithm is compared with that provided by the global shortest path algorithm (which is not scalable). it is found that the network capacities of the three algorithms are almost the same as that of the global shortest path algorithm. for fixed routing algorithms, we use an approximate analysis from which the blocking probability at a fixed network load can iterately be obtained (we use call blocking probability as our quality objective and compare network capacity by observing the difference in the offered load which produces the same blocking probability). the approximate analysis is validated through simulation and is found to be very accurate. scalable, quasi-dynamic routing algorithms are also studied. numerical results show that, when the node-capacity is equal to the capacity of a single wavelength and the bandwidth required per call is large ( greater than 24% of the link capacity), the capacity provided by the quasi-dynamic routing algorithm is very close to that of an infinite capacity centralized switch (lowest possible call blocking caused exclusively by congestion on the finite capacity user input/output links, never by the switch fabric itself). in all the examples considered, the improvement of the quasi-dynamic routing algorithm over the fixed routing algorithm is significant.
using the small-world model to improve freenet performance. efficient data retrieval in an unstructured peer-to-peer system like freenet is a challenging problem. in this paper, we study the impact of workload on the performance of freenet. we find that there is a steep reduction in the hit ratio of document requests with increasing load in freenet. we show that a slight modification of freenet's routing table cache replacement scheme (from lru to a replacement scheme that enforces clustering in the key space) can significantly improve performance. our modification is based on intuition from the small-world models and theoretical results by kleinberg; our replacement scheme forces the routing tables to resemble neighbor relationships in a small-world acquaintance graph--clustering with light randomness. our simulations show that this new scheme improves the request hit ratio dramatically while keeping the small average hops per successful request comparable to lru. a simple, highly idealized model of freenet under clustering with light randomness proves that the expected message delivery time in freenet is o(log n) if the routing tables satisfy the small-world model and have the size θ(log2n).
maximal average loss rates for a single gps server system with finite buffers. considers a single server system with finite buffers that services.
a convergence proof for an iterative method for atm networks. a flow model for evaluating the performance of a network of asynchronous transfer mode (atm) switches is presented. the performance measures used include the link (nodal) and end-to-end cell loss probabilities as well as the link (nodal) and end-to-end cell delays. in the model, the routing assignments are assumed to be given. the assumed form of routing assignments may be used to represent either virtual circuit or datagram service. due to the nonlinear relationship between cell losses and offered flows, the flow model is a system of nonlinear equations. the authors develop a sufficient condition for the existence of a unique solution to the nonlinear system of equations. they present an iterative model and prove that it converges to a unique fixed point provided the sufficient condition is satisfied. the unique fixed point corresponds to the unique solution to the flow model
modeling of motion classified vbr video codecs. the authors use a motion adaptive variable-bit-rate (vbr) video codec and propose a motion classified model to represent the characteristics of various classes of motion activities. the codec switches between interframe, motion compensated, and intraframe coding corresponding to low, medium, and high motions and scene changes, respectively. the model captures the motion of various video scenes and the codec structure by providing the statistics of vbr-coded video' traffic through a first-order composite autoregressive process with three motion classes. the parameters of this model are derived from a vbr-coded sample video sequence such that the bit rate distribution and the autocorrelation in bit rates of two successive frames are matched. the validity and accuracy of the model are verified. using this model, the characteristics of aggregated traffic sources are discussed
topological design of loss-free switch-based lans. the paper presents a new design methodology and tools to construct a switch-based lan with (i) scalable throughput, (ii) no loss due to congestion, and (iii) two routing modes: fifo or non-fifo. more specifically, given a bounded degree (number of switch ports) at each node, the design is based on the construction of multiple virtual rings under the following constraints: (i) the virtual rings are pairwise edge-disjoint, and (ii) there is at least one virtual ring between any pair of nodes. the target topology is obtained from the edge union of the multiple virtual rings. the objectives of the above two constraints are (i) to ensure no loss due to congestion inside the network of bursty traffic sources, and (ii) to ensure convergence of packets/cells to their destinations. the virtual rings are constructed by a new methodology that employs combinatorial block designs together with a new algorithm for realizing any size networks. it is shown that the bound on the maximum route length, under the two constraints, is o(/spl radic/n) for an n-node network.
hop-by-hop congestion control over a wireless multi-hop network. this paper focuses on congestion control over multi-hop, wireless networks. in a wireless network, an important constraint that arises is that due to the mac (media access control) layer. many wireless macs use a time-division strategy for channel access, where, at any point in space, the physical channel can be accessed by a single user at each instant of time. in this paper, we develop a fair hop-by-hop congestion control algorithm with the mac constraint being imposed in the form of a channel access time constraint, using an optimization-based framework. in the absence of delay, we show that this algorithm are globally stable using a lyapunov-function-based approach. next, in the presence of delay, we show that the hop-by-hop control algorithm has the property of spatial spreading. in other words, focused loads at a particular spatial location in the network get "smoothed" over space. we derive bounds on the "peak load" at a node, both with hop-by-hop control, as well as with end-to-end control, show that significant gains are to be had with the hop-by-hop scheme, and validate the analytical results with simulation.
efficient selective frame discard algorithms for stored video delivery across resource constrained networks. video delivery from a server to a client across a network is an important component of many multimedia applications. while delivering a video stream across a resource constrained network, loss of frames may be unavoidable. under such circumstances, it is desirable to find a server transmission schedule that can efficiently utilize the network resources while maximizing the perceived quality-of-service (qos) at the client. to address this issue, in this paper we introduce the notion of selective frame discard at the server and formulate the optimal selective frame discard problem using a qos-based cost function. given network bandwidth and client buffer constraints, we develop an o (n log n) algorithm to find the minimum number of frames that must be discarded in order to meet these constraints. the correctness of the algorithm is also formally established. we present a dynamic programming based algorithm for solving the problem of optimal selective frame discard. since the computational complexity of the optimal algorithm is prohibitively high in general, we also develop several efficient heuristic algorithms for selective frame discard. these algorithms are evaluated using jpeg and mpeg video traces.
supporting cooperative caching in ad hoc networks. most researches in ad hoc networks focus on routing and not much work has been done on data access. a common technique used to improve the performance of data access is caching. cooperative caching, which allows the sharing and coordination of cached data among multiple nodes, can further explore the potential of the caching techniques. due to mobility and resource constraints of ad hoc networks, cooperative caching techniques designed for wired networks may not be applicable to ad hoc networks. in this paper, we design and evaluate cooperative caching techniques to efficiently support data access in ad hoc networks. we first propose two schemes: cachedata, which caches the data, and cachepath, which caches the data path. after analyzing the performance of those two schemes, we propose a hybrid approach (hybridcache), which can further improve the performance by taking advantage of cachedata and cachepath while avoiding their weaknesses. cache replacement policies are also studied to further improve the performance. simulation results show that the proposed schemes can significantly reduce the query delay and message complexity when compared to other caching schemes.
dirsmin: a fault-tolerant switch for b-isdn applications using dilated reduced-stage min. develops and analyzes a dilated high performance fault tolerant fast packet multistage interconnection network (min). in this new design, the links at the input and the output stages of a dilated banyan-based min are rearranged to create multiple routes for each source-destination pair in the network after removing one stage in the network. these multiple paths are link- and node-disjoint. fault tolerance at low latency is achieved by sending multiple copies of each input packet simultaneously using different routes and different priorities. this guarantees that high throughput is maintained even in the presence of fault. throughput is analyzed using simulation and analysis and the authors show that the new design has considerably higher performance in the presence of a faulty switching element (se) or link in comparison to dilated networks.
integrated rate and credit feedback control for abr service in atm networks. we propose a flow-control scheme that combines the merits of credit- and rate-based flow-control schemes by applying direct control over both bandwidth and buffer resources. the goal of the proposed scheme is to design an optimal rate-control policy for a given finite buffer capacity that maximizes average throughput and bounds end-to-end delay. by applying higher-order rate control, the proposed scheme not only makes the rate process converge to the neighborhood of link bandwidth, but also confines the queue-length fluctuation to a regime bounded by buffer capacity (thus guaranteeing lossless transmission). using the fluid approximation method, we model the proposed flow-control scheme and study the system dynamic behavior for abr (available bit rate) service under the most stressful traffic condition. we derive the expressions for queue build-ups and average throughput in both transient and equilibrium states. the analytical results have shown the proposed scheme to be stable and efficient in that the source rate and bottleneck queue length rapidly converge to the designated operating region. also, presented are examples showing that the proposed scheme outperforms the other existing schemes.
on-line optimal wavelength assignment in wdm networks with shared wavelength converter pool. in this paper, we study on-line wavelength assignment in wavelength-routed wdm networks under both unicast and multicast traffic where nodes in the networks have wavelength conversion ability. since wavelength converters are still expensive and difficult to implement, we consider the case where nodes in networks have only a limited number of converters that are shared by all input channels. we study the problem of setting up connections in such networks using minimum number of wavelength converters. for unicast traffic, we first study the problem of setting up a lightpath on a given link-path with minimum number of conversions. we give a simple algorithm that solves it in o(tk) time where t is the number of links on the path and k is the number of wavelengths per fiber, as compared to the best known existing method that needs to construct an auxiliary graph and apply the dijkstra's algorithm. we also consider the problem of setting up a lightpath while using wavelength converters at nodes with fewer available converters only when necessary, and give an o(tk) time algorithm. we then generalize this technique to wdm networks with arbitrary topologies and give an algorithm that sets up an optimal lightpath network-wide in o(nk+lk) time, where n and l are the number of nodes and links in the network, respectively. we also consider multicast traffic in this paper. finding an optimal multicast light-tree is known to be np-hard and is usually solved by first finding a link-tree then finding a light-tree on the link-tree. finding an optimal link-tree is also np-hard and has been extensively studied. thus, we focus on the second problem which is to set up a light-tree on a given link-tree with minimum number of conversions. we propose a new multicast conversion model with which the output of the wavelength converter is split-table to save the usage of converters. we show that under this model the problem of setting up an optimal light-tree is np-hard and then give efficient heuristics to solve it approximately.
design of robust congestion controllers for atm networks. we propose an approach to design a rate-based proportional traffic controller in order to flow-regulate the best-effort service (e.g., abr traffic) and guaranteed service traffic through an atm switch. the controller is distributed and it has a very simple structure. its local controller at each source node is open-loop stable and only requires the knowledge of the buffer occupancy at the bottleneck switch. we show that this controller is fair and is not sensitive to the change of vcs over time. it does not have oscillation and can achieve a high utilization.
network bandwidth requirements for scalable on-demand streaming. previously proposed streaming protocols using broadcast or multicast are able to deliver multimedia files on-demand with required server bandwidth that grows much slower than linearly with request rate, or with the inverse of client start-up delay. the same efficiencies can be achieved for network bandwidth if delivery is over a true broadcast channel. this paper considers the required network bandwidth for on-demand streaming over multicast delivery trees. we consider both simple canonical delivery trees, and more complex cases in which delivery trees are constructed using both existing and new algorithms for randomly generated network topologies and client site locations. results in this paper quantify the potential savings from use of multicast trees that are configured to minimize network bandwidth rather than the latency to the content server. further, we determine the network bandwidth usage of particular immediate service and periodic broadcast on-demand streaming protocols. the periodic broadcast protocol is able to simultaneously achieve close to the minimum possible network and server bandwidth usage.
performance analysis of finite output-buffered multistage atm switching fabrics. a new and accurate markov chain model for evaluating the performance of multistage atm switching fabrics with 2/spl times/2 finite output-buffered switching elements is proposed. by comparing the results obtained from the proposed model, existing models, and simulations, it has been shown that the proposed model is much more accurate than existing models in the presence of a non-uniform traffic in the switch. the results from existing models are unsatisfactory in the presence of an increased blocking in the switch arising due to a non-uniform traffic in the switch. on the contrary, the proposed model is very robust even under severe blocking inside the switch.