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Load balancing in publish/subscribe (pub/sub) broker networks is challenging as the workload is multi-dimensional and content-dependent. In this paper we present the framework design of a middleware, called Shuffle, to achieve optimal load balancing in a pub/sub broker network. Shuffle features a suite of active workload management schemes within a single overlay topology on message parsing, matching, delivery, and forwarding, the four types of workload in a publish/subscribe service affected by two inputs-streaming events and stored subscriptions. Shuffle leverages its traffic randomization scheme and Chord, a DHT substrate, to build overlay trees for active workload aggregation and distribution, and we show the optimality property of the load balancing scheme upon any input traffic distribution on individual Shuffle aggregation trees. We also show the NP-hardness of the workload management problem when it has to be done among multiple correlated aggregation trees, and present a heuristic accordingly. Through extensive simulations we validated the design of Shuffle upon dynamic and heavy workload.

In this work, we address the issue of unrealistic simulations of wireless networks using a measurement-based approach. The idea is to use empirical modeling using measurement data as a mechanism to model physical layer behavior. We demonstrate the power of this approach for 802.11-based networks using ns2, a packet-level network simulator. Specifically, we develop two versions of the ns2 simulator that model the wireless physical layer with different levels of fidelity. In both versions, the deferral and reception model are built using measurements. For propagation modeling, one version uses direct measurements and the other uses an empirically derived model. In validation experiments with a 12-node mesh testbed, both these versions were found to be reasonably accurate (85 percentile errors within about 10% of the capacity) relative to regular simulations (85 percentile errors within 50% of capacity).

Real-time services such as VoIP are becoming popular and are major revenue earners for network service providers. These services are no longer confined to the wired domain and are being extended over wireless networks. Although some of the existing wireless technologies can support some low-bandwidth applications, the bandwidth demands of many multimedia applications exceed the capacity of these technologies. The IEEE 802.16-based WiMax promises to be one of the wireless access technologies capable of supporting very high bandwidth applications. In this paper, we exploit the rich set of flexible features offered at the medium access control (MAC) layer of WiMax for the construction and transmission of MAC protocol data units (MPDUs) for supporting multiple VoIP streams. We study the quality of VoIP calls, usually given by R-score, with respect to the delay and loss of packets. We observe that loss is more sensitive than delay; hence, we compromise the delay performance within acceptable limits in order to achieve a lower packet loss rate. Through a combination of techniques like forward error correction, automatic repeat request, MPDU aggregation, and minislot allocation, we strike a balance between the desired delay and loss. Simulation experiments are conducted to test the performance of the proposed mechanisms. We assume a three-state Markovian channel model and study the performance with and without retransmissions. We show that the feedback-based technique coupled with retransmissions, aggregation, and variable length MPDUs are effective and increase the R-score and mean opinion score by about 40 percent.

Multichannel communication in a wireless mesh network with routers having multiple radio interfaces significantly enhances the network capacity. Efficient channel assignment and routing is critical for realization of optimal throughput in such networks. In this paper, we investigate the problem of finding the largest number of links that can be activated simultaneously in a wireless mesh network subject to interference, radio and connectivity constraints. Our goal is to activate all such links and we present an interference aware channel assignment algorithm that realizes this goal. We show that the Link Interference Graph created by utilizing a frequently used interference model gives rise to a special class of graphs, known as overlapping double-disk (ODD) graphs. We prove that the Maximum Independent Set computation problem is NP-complete for this special class of graphs. We provide a Polynomial Time Approximation Scheme (PTAS) for computation of the Maximum Independent Set of an ODD graph. We use this PTAS to develop a channel assignment algorithm for a multiradio multichannel Wireless Mesh Network. We evaluate the performance of our channel assignment algorithm by comparing it with the optimal solution obtained by solving an integer linear program. Experimental results demonstrate that our channel assignment algorithm produces near optimal solution in almost all instances of the problem.

We study the problem of supporting VoIP calls in a wireless mesh network. Specifically, we propose solutions for call admission control (CAC) and route selection for VoIP calls. Call admission decisions must evaluate how the capacity of the mesh network is utilized by the existing calls. We address this issue via a measurement-based modeling effort to model mutual interference between wireless links. The modeling approach evaluates whether capacity constraints (or, required QoS metrics) will be satisfied if a new call is admitted with a given route. Evaluations with a 6-node 802.11a testbed demonstrate excellent accuracy of the model and thus also the CAC performance. We address the issue of route selection by also using a modeling approach that considers models of transmission and interference ranges to develop a polynomial-time algorithm to search for feasible routes. This problem takes exponential time for wireless networks without such modeling. In addition to studying feasibility, we study several routing metrics such as shortest feasible path and maximum residual feasible path. Finally, we develop a new method for routing using call statistics that uses prior calling patterns to avoid potentially critical links. We evaluate the performance of these route selection techniques via extensive simulations and demonstrate the superiority of using max residual feasible path over simply shortest feasible path, and routing using call statistics over max residual feasible path.

Background transfers, or transfers that humans do not actively wait on, dominate the Internet today. In today's best-effort Internet, background transfers can interfere with foreground transfers causing long wait times, thereby hurting human productivity. In this paper, we present the design and implementation of a background network, Harp, that addresses this problem. Harp has three significant advantages over recent end-host based background transport protocols; Harp (i) uses multiple paths to exploit path diversity and load imbalance in the Internet to tailor network resource allocation to human needs, (ii) provides better fairness and utilization compared to unipath end-host protocols, and (ill) can be deployed at either end-hosts or enterprise gateways, thereby aligning the incentive for deployment with the goals of network customers. Our evaluation using simulations and a prototype on Planetlab suggests that Harp improves foreground TCP transfer time by a factor of five and background transfer time by a factor of two using just two extra paths per connection.

802.11a, b, and g standards were designed for deployment in cooperative environments, and hence do not include mechanisms to protect from jamming attacks. In this paper, we explore how to protect 802.11 networks from jamming attacks by having the legitimate transmission hop among channels to hide the transmission from the jammer. Using a combination of mathematical analysis and prototype experimentation in an 802.11a environment, we explore how much throughput can be maintained in comparison to the maintainable throughput in a cooperative, jam-free environment. Our experimental and analytical results show that in today's conventional 802.11a networks, we can achieve up to 60% of the original throughput. Our mathematical analysis allows us to extrapolate the throughput that can be maintained when the constraint on the number of orthogonal channels used for both legitimate communication and for jamming is relaxed.

We present a practical, measurement-based model that captures the effect of interference in 802.11-based wireless LAN or mesh networks. The goal is to model capacity of any given link in the presence of any given number of interferers in a deployed network, carrying any specified amount of offered load. Central to our modeling approach is a MAC-layer model for 802.11 that is fed by PHY-layer models for deferral and packet capture behaviors, which in turn are profiled based on measurements. The target network to be evaluated needs only O(N) measurement steps to gather metrics for individual links that seed the models. We provide two solution approaches - one based on direct simulation (slow, but accurate) and the other based on analytical methods (faster, but approximate). We present elaborate validation results for a 12 node 802.11b mesh network using upto 5 interfering transmissions. We demonstrate, using as comparison points three simpler modeling approaches, that the accuracy of our approach is much better, predicting link capacities with errors within 10% of the base channel datarate for about 90% of the cases.

In this paper, we exploit the flexible features in the medium access control (MAC) layer of WiMax for construction and transmission of MAC protocol data units (MPDU) for supporting multiple VoIP streams over a WiMax link. Quality of VoIP calls, usually given by R-score, is studied with respect to delay and loss of packets. We observe that loss is more sensitive than delay, hence we trade delay for loss. We propose a combination of techniques that exploit the flexibility of the WiMax MAC layer to strike a balance between loss and delay. These techniques are forward error correction, automatic repeat request, MPDU aggregation, and minislot allocation. Simulation experiments are conducted to test the performance of the proposed mechanisms. We assume a three-state Markovian channel model and study the performance with and without retransmissions. We show that the feedback-based technique coupled with retransmissions, aggregation, and variable length MPDUs are effective and increases the R-score by about 40%.

We consider the problem of supporting VoIP calls over a wireless mesh network. We particularly consider the call admission control (CAC) decision problem for VoIP calls on a mesh network. The goal of the call admission decision is to maintain the quality of VoIP calls as given by the R-score measure. In this work, we define a notion of interference capacity model that can be efficiently used to design a CAC algorithm. We show through extensive simulation of 802.11 mesh using ns-2, that our CAC performs well in multi-hop scenarios. Specifically, the proposed CAC provides less than 20% incorrect decisions for different sizes of a multihop linear topology.

Abstract — We introduce MobiStream — a video streaming system that exploits the perceptual value in the video content and characteristics of the link layer and physical layer channels to enable efficient error-resilient video streaming over wireless widearea networks (WWANs). The key building block in MobiStream is the use of link-layer based, but application-controlled, virtual channels (ViCs) abstraction. Each virtual channel in MobiStream offers a level of reliability and statistical loss gaurantee using ‘awareness ’ of the characteristics of link-layer and physical layer channels. Video applications can dynamically instantiate new virtual channels, control their loss behavior, and/or flexibly switch video transmission across channels. MobiStream achieves fine-grained errorresilience by partitioning the video frames into number of small, independently decodable, blocks of data (called ‘slices’) and assigns priority to each individual slice based on its perceptual (visual) usefulness. MobiStream augments a number of other enhancements for error-resilience: multiple description video coding, perceptual slice-structured coding, low-delay inter-frame and intra-frame slice interleaving, dynamic unequal error protection, and priority-based video-data scheduling to enable efficient and error-resilient video streaming over wireless wide-area links. MobiStream has been implemented and evaluated using loss distributions from tests conducted over a commercial wide-area wireless (CDMA2000 3G) network. Results show that, even in stationary conditions, MobiStream, on average, can improve video picture quality by at least 4 dB. We conclude that significant benefits to end-user experience can be obtained by deploying such a system. I.

Abstract — We present Sieve – an infrastructure for fast content-based data distribution to interested users. The ability of Sieve to filter and forward high-bandwidth data streams stems from its distributed pipelined architecture. The complex message filtering task is broken-up into a sequence of light-weight filtering components resulting in high end-to-end throughput. Furthermore, since each component is assigned to a node based on its resource constraints, the queue buildup inside the nodes is minimal resulting in low end-to-end latency. Our experimental results based on real system implementation show that Sieve can sustain a throughput of more than 5000 messages per second for 100000 subscriptions with predicates of 10 attributes. Index Terms — Content-based Information Dissemination, Publish-Subscribe System, Event Stream Filtering

Abstract — Performance in multihop wireless networks is known to degrade with the number of hops for both TCP and UDP traffic. For VoIP, the wireless network presents additional challenges as the perceived quality is dependent on both loss and delay. We investigate several methods to improve voice quality and present experimental results from an 802.11b testbed optimized for voice delivery. Use of multiple interfaces, path diversity and aggregation are shown to provide a combined improvement of 13 times in number of calls supported in our 15 node 802.11 mesh system. VoIP, multihop, mesh I.

Wireless 802.11 hotspots have grown in an uncoordinated fashion with highly variable deployment densities. Such uncoordinated deployments, coupled with the difficulty of implementing coordination protocols, has often led to conflicting configurations (e.g., in choice of transmission power and channel of operation) among the corresponding Access Points (APs). Overall, such conflicts cause both unpredictable network performance and unfairness among clients of neighboring hotspots. In this paper, we focus on the fairness problem for uncoordinated deployments. We study this problem from the channel assignment perspective. Our solution is based on the notion of channel-hopping, and meets all the important design considerations for control methods in uncoordinated deployments - distributed in nature, minimal to zero coordination among APs belonging to different hotspots, simple to implement, and interoperable with existing standards. In particular, we propose a specific algorithm called MAXchop, which works efficiently when using only non-overlapping wireless channels, but is particularly effective in exploiting partially-overlapped channels that have been proposed in recent literature. We also evaluate how our channel assignment approach complements previously proposed carrier sensing techniques in providing further performance improvements. Through extensive simulations on real hotspot topologies and evaluation of a full implementation of this technique, we demonstrate the efficacy of these techniques for not only fairness, but also the aggregate throughput, metrics.We believe that this is the first work that brings into focus the fairness properties of channel hopping techniques and we hope that the insights from this research will be applied to other domains where a fair division of a system's resources is an important consideration.

In this paper we design and evaluate Ganges, a wireless mesh network architecture that can efficiently transport real-time video streams from multiple sources to a central monitoring station. Video quality suffers from deterioration in the presence of bursty network losses and due to packets missing their playback deadline. Ganges spatially separates the paths to reduce inter-flow contention. It finds out a fair rate allocation for the different video sources. The wireless routers in the mesh network implement several optimizations in order to reduce the end-to-end delay variation. Ganges improves the network capacity by over 10-15% over a shortest path tree, and improves the video picture quality by 7-10 dB

In this paper, we present the design of the WiMAX based radio access network to provide data coverage to 802.16 enabled devices. The design provides a multihop solution to provide wide area coverage to users and targets maximizing the radio resource utilization. We propose efficient route construction, data scheduling for increasing the utilization of the backhaul network and base station selection algorithm for end users for efficient load balancing. Through extensive solution, we should attain the benefits by each of the proposed design methods

Performance in multihop wireless networks is known to degrade with the number of hops for both TCP and UDP traffic. For VoIP, the wireless network presents additional challenges as the perceived quality is dependent on loss, delay and jitter. Furthermore, small size of VoIP packets aggravates network utilization of the 802.11 based wireless mesh network. In this work, we evaluate the packet aggregation algorithms to reduce VoIP protocol overhead, compare the aggregation algorithms and propose a distributed packet aggregation mechanism. The experimental results demonstrates that the proposed distributed packet aggregation algorithm improves the VoIP quality and results in significant increase in the number high quality VoIP calls supported over multihop mesh.

Current web search engines are retrospective in that they limit users to searches against already existing pages. Prospective search engines, on the other hand, allow users to upload queries that will be applied to newly discovered pages in the future. We study and compare algorithms for efficiently matching large numbers of simple keyword queries against a stream of newly discovered pages.

In the recent past, there has been a tremendous increase in the popularity of VoIP services as a result of huge growth in broadband access. The same voice-over-Internet protocol (VoIP) service poses new challenges when deployed over a wireless mesh network, while enabling users to make voice calls using WiFi phones. Packet losses and delay due to interference in a multiple-hop mesh network with limited capacity can significantly degrade the end-to-end VoIP call quality. In this work, we discuss the basic requirements for efficient deployment of VoIP services over a mesh network. We present and evaluate practical optimizing techniques that can enhance the network capacity, maintain the VoIP quality and handle user mobility efficiently. Extensive experiments conducted on a real testbed and ns-2 provide insights into the performance issues and demonstrate the level of improvement that can be obtained by the proposed techniques. Specifically, we find that packet aggregation along with header compression can increase the number of supported VoIP calls in a multihop network by 2-3 times. The proposed fast path switching is highly effective in maintaining the VoIP quality. Our fast handoff scheme achieves almost negligible disruption during calls to roaming clients

A substantial portion of the emerging wireless data service consists of non-real-time applications such as content download. The existing mechanisms based on per-packet performance guarantees used mainly for voice and streaming media do not suffice for the elastic nature of non-real-time traffic. For a non-real-time user data services, the key performance measure of interest is the total download time. In this paper, we propose a novel scheduling framework for wireless content service. Specifically, we present a two-layer scheduling architecture that combines content-aware scheduling with opportunistic scheduling. In terms of content-awareness, the proposed scheduling policy provides guarantees on the download time of content. In the second stage, the instantaneous channel conditions of different users are exploited in an opportunistic manner so as to maximize the throughput of the system. We define service differentiation in two modes - differential and guaranteed - and provide polynomial time algorithms for both that manipulate the stretch ratio but within allowable limits. Extensive simulations are conducted that verify the efficiency of the proposed schemes and provide insights into the behavior of the scheduling algorithms for non-real-time data.

To bring TCP-based services to the mobile devices in a cellular network, it is necessary that TCP be extended over the wireless link. However, the performance of TCP severely degrades in a wireless medium. Hence, radio link protocols (RLPs) are used as an interface between TCP and the physical medium. RLPs fragment TCP segments into frames and use robust error correcting codes and fast retransmission schemes to shield the channel related losses from TCP, thus preventing TCP throughput degradation. In this paper, we show the limitations of the existing RLPs, which do not differentiate the frames generated from the same TCP segment. We claim that if selective frames are made more robust to transmission failures, then the performance of RLP and, hence, TCP can be improved. We identify such decisive frames and categorize them as crucial and noncrucial. Our claim is based on the fact that initial frames can afford a few trials of retransmissions, whereas the later ones cannot. We treat the frames differentially with respect to forward error correcting (FEC) coding and automatic repeat request (ARQ) schemes. We consider specific cases of FEC and ARQ strategies and show the qualitative difference in the performance of the RLP through analysis and simulations. The gain in the performance is more prominent when both FEC and ARQ (hybrid-ARQ) are used. The increase in TCP throughput with the proposed RLP is also demonstrated

We present SPIDER-a system for fast replication or distribution of large content from a single source to multiple sites interconnected over Internet or via a private network. In order to exploit spatial diversity of the underlying network, SPIDER uses an overlay structure composed of dedicated transit nodes (TNs). The data transport mechanism in SPIDER leverages this overlay structure to provide a coordinated approach that minimizes the maximum time to replicate to all destination sites (the make span of content replication). In order to achieve this objective, SPIDER employs two orthogonal components: a) creation of multiple dynamic distribution trees using the transit nodes b) end-to-end reliable data transport with flow control on these trees by chaining point-to-point TCPs. We further present simulations based results to quantify benefits of tree construction algorithms in random topologies. We evaluate the real implementation of the SPIDER in Planet Lab and observe a 2-6 times speed up compared to different existing schemes.

Non-uniform coverage and low system throughput due to location dependent fading in cellular networks is a major bottleneck in providing quality of service. Moreover, the growing number of stand-alone Wi-Fi hotspots is creating competition for 3G cellular networks. We propose an architecture which augments the existing 3G downlink technology with Wi-Fi like relay nodes leveraging benefits of both forms to improve the 3G system performance and provide uniform coverage throughout the cell. The relay nodes in the proposed WRN (Wi-Fi like relay network) architecture are equipped with a dual radio interface to communicate with the base station and other relay nodes. We compare the performance of the existing CDMA/HDR-based 3G cellular network with the proposed WRN architecture in terms of throughput and blocking probability. We demonstrate how pre-engineered deployment of relay nodes can yield better performance than random deployment of relay nodes. Simulation experiments are conducted to corroborate our analytical findings. The system throughput is not only enhanced, but the fairness among users is also maintained.

The IEEE 802.16 standard (commonly known as WiMAX), which has emerged as a broadband wireless access technology, is capable of delivering very high data rates. However, providing performance guarantees to delay sensitive applications like streaming media is still a challenge. We study the media access control (MAC) layer of WiMAX and exploit its flexible features to construct MAC packet data units (MPDU) dynamically. The sizes of the MPDUs are constantly modified based on channel state information. The desired payload is obtained either by aggregation or fragmentation of the upper layer data units. The robustness of MPDUs is also made tunable by means of cyclic redundancy code bits. We consider both the scenarios - with and without feedback. We adhere to the 802.16 specifications and propose adapting the MPDU length for streaming media for better performance. Three metrics are defined: restore probability; goodput; dropping probability. Simulation experiments are conducted which show the performance enhancements of the proposed ARQ-enabled adaptive algorithm in terms of these three metrics.

We propose enhancements to radio link protocols by identifying decisive frames and categorizing them as crucial and non-crucial. The fact that initial frames from the same upper layer segment can afford a few trials of retransmissions and the later frames cannot, motivates our work. We treat the frames differentially with respect to FEC coding and ARQ schemes. We consider specific cases of FEC and ARQ strategies and qualitatively show how the differential treatment of frames can improve the performance of the RLP.

Efficient transfer of bulk data requires routing that minimizes the net transfer time instead of providing flow level bandwidth guarantees. In this work, we consider a realistic scenario where available bandwidth for each link in a given network is time varying. In such a network and for a given file size, we provide an optimal algorithm that minimizes the total time required to transfer the file from source to destination. We further consider the problem where path shifting is allowed to increase the net throughput for bulk data transfer. For this problem, we provide a dynamic programming based optimal algorithm that minimizes the number of path shifts required to transfer the file in the specified time. Solution to the above problems addresses both the performance and scalability issues that arise in large bulk data transfer for long duration of time.

Hybrid hierarchical optical cross-connects leverages the benefits of both all-optical (OOO) and optical-electrical (OEO) switching for a cost-effective solution in future optical networks. The design goal using hybrid cross-connects is to minimize the sum of two cost metrics: OEO switching cost and OOO switching cost. This paper proposes two different waveband merging architectures and minimum cost routing and merging algorithms for offline traffic scenario. The simulation based results show that the proposed algorithms provide higher cost benefit as compared to existing algorithms that do not rely on waveband merging architecture.

Non-real time content delivery constitutes a major portion wireless data services in 3G networks. The per-packet performance guarantees required for real-time applications, such as voice and streaming media, is achieved through conventional flow-based scheduling mechanisms. However, for data services, such scheduling mechanisms cannot be used since the key performance measure for non-real time applications is the total download time. In this paper, we present a two-layer scheduling architecture that combines content-aware scheduling with opportunistic scheduling. In terms of content-awareness, the proposed scheduling policy provides guarantees to the download time of a content. Furthermore, through parallelizing the schedule and using opportunistic channel usage, the scheme achieves higher throughput. Extensive simulations verify the efficiency of our proposed schemes and provide insights into the behavior of the scheduling algorithms.

Non-uniform coverage is a major concern in cellular data networks based on HSDPA/HDR access technologies. Poor coverage lowers the overall utilization of the cell and results in location-dependent downlink throughput for mobile users. We focus on the planning of ad hoc relay network (ARN) in providing an improved cellular coverage. Specifically, we present and discuss issues and approaches for relay node placement in cellular space. Through extensive simulation modeling, we provide the evaluation of the improvement in the location dependent cellular data rate by employing the ARN.

We present a QoS-enhanced routing framework designed for an administrative system that runs a QoS-aware link-state routing protocol supporting MPLS. This routing framework is based on using only edge nodes and does not require any modifications at the core. The framework also does not employ any bandwidth reservation; instead, congestion is relieved by adaptive rerouting of aggregated traffic flows. Route oscillation is prevented by using randomization-based back-off techniques for triggering route changes. The proposed mechanism is distributed and non-cooperative, which makes it suitable for practical implementation. The results of simulations demonstrate that a significant reduction of packet drop ratio can be achieved with a moderate number of path shifts.

Efficient bandwidth allocation strategy with simultaneous fulfillment of QoS requirement of a user in a mobile cellular network is still a critical and an important practical issue. We explore the problem of finding the reservation schedule that would minimize the amount of time for which bandwidth has to be allocated in a cell while meeting the QoS constraint. With the knowledge about the arrival and residence time distribution of a user in a cell, the above problem can be optimally solved using a dynamic programming based approach in polynomial time. To be able to use the solution, we provide a mechanism for constructing the arrival/residence time distribution based on the measurement of hand-off events in a cell. The above solution allows us to propose an optimal time based bandwidth reservation and call admission scheme. By being scalable and distributed, the proposed scheme justifies for practical implementation. Simulations results are also presented to show the effectiveness of the scheme to achieve the target QoS level and optimal bandwidth utilization.

Aggregation of individual wavelengths into wavebands for their subsequent switching and routing as a single group is an attractive way for scalable and cost-efficient optical networks. We analyze the implications of this waveband hierarchy for a single optical node by analyzing two issues: the proper selection of waveband sizes and the assignment of wavebands for a limited set of input-output patterns of traffic. We formulate a general model and propose optimal algorithmic solutions for both problems. The performance of resulting sets of nonuniform wavebands is studied for several representative cases (a single node, an optical ring network, an optical mesh network). The results demonstrate improved optical throughput and reduced cost of switching and routing when using nonuniform waveband hierarchy.

It is an inherently challenging task to provide quality of service (QoS) to real time applications in cellular networks owing to their mobility-induced uncertainties. Here we consider an adaptive framework where services may undergo bandwidth degradation. In contrast to the existing schemes, we here provide a reservation states based approach to provision the QoS to the individual adaptive applications. By using information related to users' mobility, the proposed approach can assure probabilistic QoS to multiple level of degradation as maybe required by such adaptive services. Simulation results show the effectiveness of the scheme in realizing the target QoS.

Resource reservation - both just in time and advance in time - has been advocated as a means of guaranteeing quality of service (QoS) to mobile users. Often, advance bandwidth reservation results in low utilization due to bandwidth fragmentation wherein holes of unused bandwidth in time may be too small to allocate to any user yet the sum of several holes may be sufficient to satisfy the user. In this paper, we introduce and investigate the problem of bandwidth fragmentation and explore different methods of minimizing bandwidth fragmentation. One is a selective call rejection scheme and the other a multi-class bandwidth sharing scheme. Through an extensive set of simulation results, we show that our proposed schemes reduces bandwidth fragmentation.

We study the path provisioning as a mechanism to deliver service level agreements in IP Differentiated Services networks. There is no known polynomial time solution for this problem, so we propose and analyze (by simulations) several heuristic algorithms for solving the problem. As our simulations demonstrate, a centralized server consistently delivers a better performance than a distributed solution. We also show that the performance of one of the proposed algorithms, the greedy algorithm with backtracking, can be very close to the optimal one, while being computationally feasible.

We consider hierarchical optical networks with different granularities of paths (wavelengths and wavebands). We introduce classification layers for routing models in these networks (separated and integrated routing, homogeneous and heterogeneous networks, online and offline routing). We also propose and analyze two heuristic routing and aggregation algorithms (online and offline) to be used for homogeneous networks in the separate routing framework. The analysis is based on ring topology; its results are supported by simulations of large scale networks (Europe and USA). These simulations demonstrate a significant cost reduction achieved by employing hierarchical routing (from 33% in the online algorithm to almost 60% in the offline one).

It is an inherently challenging task to provide quality of service (QoS) to real time multimedia users in a cellular network owing to their mobility induced uncertainty. We propose an implicit resource allocation strategy and justify the need of such a strategy toward an effective solution to QoS provisioning. The reservation scheme presented, based on this proposed strategy, aims at meeting the users' QoS requirement while maintaining a better utilization as compared to the conventional resource reservation schemes. Simulation results are presented which show the effectiveness of the scheme under a varying level of user mobility.