Share Of Bandwidth Research Articles

The LOTTERYBUS on-chip communication architecture

On-chip communication architectures play an important role in determining the overall performance of System-on-Chip (SoC) designs. Communication architectures should be flexible so as to offer high performance over a wide range of traffic characteristics. In particular, the resource sharing mechanism of the communication architecture, which determines how the often-conflicting requirements of different components are served, is of utmost importance. Conventional SoC architectures typically employ priority or time-division multiple-access (TDMA)-based communication architectures. However, these techniques are often inadequate. In the former, low-priority components may suffer from starvation, while in the latter, depending on the request profile, high-priority traffic may be subject to large latencies. This paper presents LOTTERYBUS, a high-performance SoC communication architecture based on new randomized on-chip communication protocols that addresses the shortcomings mentioned above. LOTTERYBUS provides each SoC component with a flexible, proportional, and probabilistically guaranteed share of the on-chip communication bandwidth. We present two variants of LOTTERYBUS. In the first variant, its architectural parameters are statically configured, leading to relatively low hardware overhead and design complexity. In the second variant, these parameters are allowed to vary dynamically, enabling more sophisticated use of LOTTERYBUS, at additional hardware cost. We have performed experiments to investigate the performance of LOTTERYBUS across a range of communication traffic characteristics. We have used LOTTERYBUS in designing a 4times4 ATM switch subsystem, and have compared its performance with conventional architectures. The results show that LOTTERYBUS provides fine-grained control over bandwidth allocation, and also provides significant reduction in average transaction latencies (up to 85%) compared to conventional architectures. Hardware implementations using a commercial 0.15-mum cell-based library indicate that the advantages provided by LOTTERYBUS are accompanied by modest hardware overheads compared to conventional architectures

Fair service for mice in the presence of elephants

We show how randomized caches can be used in resource-poor partial-state routers to provide a fair share of bandwidth to short-lived flows that are known as mice when long-lived flows known as elephants are present.

Rate-distortion optimized distributed packet scheduling of multiple video streams over shared communication resources

We consider the problem of distributed packet selection and scheduling for multiple video streams sharing a communication channel. An optimization framework is proposed, which enables the multiple senders to coordinate their packet transmission schedules, such that the average quality over all video clients is maximized. The framework relies on rate-distortion information that is used to characterize a video packet. This information consists of two quantities: the size of the packet in bits, and its importance for the reconstruction quality of the corresponding stream. A distributed streaming strategy then allows for trading off rate and distortion, not only within a single video stream, but also across different streams. Each of the senders allocates to its own video packets a share of the available bandwidth on the channel in proportion to their importance. We evaluate the performance of the distributed packet scheduling algorithm for two canonical problems in streaming media, namely adaptation to available bandwidth and adaptation to packet loss through prioritized packet retransmissions. Simulation results demonstrate that, for the difficult case of scheduling nonscalably encoded video streams, our framework is very efficient in terms of video quality, both over all streams jointly and also over the individual videos. Compared to a conventional streaming system that does not consider the relative importance of the video packets, the gains in performance range up to 6 dB for the scenario of bandwidth adaptation, and even up to 10 dB for the scenario of random packet loss adaptation.

Dynamic bandwidth allocation algorithm for full-band utilization

Abstract To improve and optimize the bandwidth utilization for multi-service packet transporting system, a kind of Dynamic Full Bandwidth Utilized (DFBU) allocation algorithm allowing a single link to use far beyond its fair share bandwidth is presented. Three important parameters as the bound on max and minimum bandwidth, the maximum packet delay and the minimum bandwidth utilization are discussed and analyzed. Results of experiments show that the DFBU-algorithm is capable of making a single link in the system use all the spare bandwidth (up to full-bandwidth) while the performance of fairness and QoS requirement is still guaranteed.

Dynamic bandwidth allocation algorithm utilizing full band

A kind of Dynamic Full Bandwidth Utilized (DFBU) allocation algorithm is introduced. This algorithm allows a single link to use bandwidth far beyond its fair share bandwidth in a multi-service packet transporting system. Three important parameters as the bound on maximum and minimum bandwidth, the maximum packet delay and the minimum band width utilization are discussed and analyzed. Results of experiments show that the DFBU-algorithm is capable of making a single link in the system to use all the spare bandwidth (up to full-bandwidth) while the performance of fairness and QoS requirement is still guaranteed.

Core‐stateless fair rate estimation fair queuing

AbstractCore‐stateless mechanisms, such as core‐stateless fair queuing (CSFQ), reduce the complexity of fair queuing, which usually need to maintain states, manage buffers, and perform flow scheduling on a per‐flow basis. However, they require executing label rewriting and dropping decision on a per‐packet basis, thus preventing them from being widely deployed. In this paper, we propose a novel architecture based on CSFQ without per‐packet labelling. Similarly, we distinguish between edge routers and core routers. Edge routers maintain the per‐flow state by employing a fair queuing mechanism to allocate each flow a fair bandwidth share locally and a token bucket mechanism to regulate those flows with feedback packets sent from egress edge routers. Core routers do not maintain per‐flow state; they use FIFO packet scheduling extended by a fare rate alarm mechanism by estimating the arrival rate and the number of flows using a matching–mismatching algorithm. The novel scheme is called core‐stateless fair rate estimation fair queuing (CSFREFQ). CSFREFQ is proven to be capable of achieving max–min fairness. Furthermore, we present and discuss simulations and experiments on the performance under different traffic scenarios. Copyright © 2005 John Wiley & Sons, Ltd.

End-to-end bandwidth guarantees through fair local spectrum share in wireless ad-hoc networks

Sharing the common spectrum among the links in a vicinity is a fundamental problem in wireless ad-hoc networks. Lately, some scheduling approaches have been proposed that guarantee fair share of bandwidth among the links. The quality of service perceived by the applications however depends on the end-to-end bandwidth allocated to the multihop sessions. We propose an algorithm that provides provably maxmin fair end-to-end bandwidth to sessions. The algorithm combines a link scheduling that avoids collisions, a fair session service discipline per link, and a hop-by-hop window flow control. All the stages of the algorithm are implementable based on local information, except the link scheduling part that needs some network-wide coordination.

Dynamic Bandwidth Management in Single-Hop Ad Hoc Wireless Networks

Distributed weighted fair scheduling schemes for Quality of Service (QoS) support in wireless local area networks have not yet become standard. Therefore, we propose an Admission Control and Dynamic Bandwidth Management scheme that provides fairness and a soft rate guarantee in the absence of distributed MAC-layer weighted fair scheduling. This scheme is especially suitable for smart- rooms where peer-to-peer multimedia transmissions need to adapt their transmission rates co-operatively. We present a mapping scheme to translate the bandwidth requirements of an application into its channel time requirements. The center piece of our scheme is a Bandwidth Manager, which allots each flow a share of the channel, depending on the flow's requirements relative to the requirements of other flows in the network. Admitted flows control their transmission rates so they only occupy the channel for the fraction of time allotted to them. Thus co-operation between flows is achieved and the channel time is fair shared. As the available channel capacity changes and the traffic characteristics of various flows change, the Bandwidth Manager dynamically re-allocates the channel access time to the individual flows. Our simulation experiments show that, at a very low cost and with high probability, every admitted flow in the network will receive at least its minimum requested share of the network bandwidth. We also present extensive testbed experiments with our scheme using a real-time audio streaming application running between Linux laptops equipped with standard IEEE 802.11 network cards.

Incentive-compatible adaptation of Internet real-time multimedia

The rapid deployment of new applications and the interconnection of networks with increasing diversity of technologies and capacity make it more challenging to provide end-to-end quality assurance to the value-added services, such as the transmission of real-time multimedia and mission critical data. In a network with enhancements for QoS support, pricing of network services based on the level of service, usage, and congestion provides a natural and equitable incentive for multimedia applications to adapt their sending rates according to network conditions. We have developed an intelligent service architecture that integrates resource reservation, negotiation, pricing and adaptation in a flexible and scalable way. In this paper, we present a generic pricing structure that characterizes the pricing schemes widely used in the current Internet, and introduce a dynamic, congestion-sensitive pricing algorithm that can be used with the proposed service framework. We also develop the demand behavior of adaptive users based on a physically reasonable user utility function. We introduce our multimedia testbed and describe how the proposed intelligent framework can be implemented to manage a video conference system. We develop a simulation framework to compare the performance of a network supporting congestion-sensitive pricing and adaptive reservation to that of a network with a static pricing policy. We study the stability of the dynamic pricing and reservation mechanisms, and the impact of various network control parameters. The results show that the congestion-sensitive pricing system takes advantage of application adaptivity to achieve significant gains in network availability, revenue, and user-perceived benefit relative to the fixed-price policy. Congestion-based pricing is stable and effective in limiting utilization to a targeted level. Users with different demand elasticity are seen to share bandwidth fairly, with each user having a bandwidth share proportional to its relative willingness to pay for bandwidth. The results also show that even a small proportion of adaptive users may result in a significant performance benefit and better service for the entire user population-both adaptive and nonadaptive users. The performance improvement given by the congestion-based adaptive policy further improves as the network scales and more connections share the resources. Finally, we complement the simulation with experimental results demonstrating important features of the adaptation process.

Translating scalable video streams from wide‐area to access networks

Transmitting video over user datagram protocol (UDP) has been considered advantageous because it allows for discarding of packets in favor of retransmissions, and sender‐controlled timing. Using UDP has been criticized because it allows video streams to consume more than their fair share of bandwidth, which is typically associated with the back‐off behavior of transmission control protocol (TCP). TCP‐friendly algorithms are meant as a middle path. However, UDP delivery to end systems may still be prevented by firewalls or for other reasons, and TCP must be used. This in turn suffers from bandwidth fluctuations. Investigates an architecture that separates the transfer of a video stream over long distances. Considers a proxy server to translate the traffic and two straightforward approaches for the translation of a layered video stream transmission from the TCP‐friendly transport protocol to TCP. Does not expect that one of these two approaches is by itself suited for the task, but investigating them will provide insights into their basic functions and help in discovering appropriate modifications. For the investigation, an experimental approach was used where network behavior and content are emulated.

Fair Scheduling With QoS Support in Wireless Ad Hoc Networks

We study fair scheduling with quality of service (QoS) support for wireless ad hoc networks. Two types of flows are considered: best effort and guaranteed flows. The goal is to satisfy the minimum bandwidth requirements of guaranteed flows and to provide a fair share of residual bandwidth to all flows. We compare timestamp-based and credit-based mechanisms, and evaluate the feasibility of all existing fair scheduling schemes of these two types of mechanisms for multimedia wireless multihop networks. We suggest a flow weight calculation scheme for existing timestamp-based mechanisms to support both best effort and guaranteed flows, and propose a credit-based mechanism called credit-based slot allocation protocol (CSAP) for fair scheduling with QoS support. For comparison purposes, several metrics are defined to evaluate the performances of these two kinds of mechanisms. The simulation results show that CSAP outperforms the other approaches in terms of meeting the minimum requirements of guaranteed flows, fairly sharing the residual bandwidth among all flows, and improving overall system throughput.

Reallocation of paths of ABR services in a multistage interconnection network (MIN)-type ATM switch network

In this paper, we develop algorithms for reallocating paths of available bit rate (ABR) services in an asynchronous transfer mode (ATM) switch network. ATM traffic control for fair share bandwidth allocation is usually performed under the assumption that paths of all services in a switch network are fixed. However, each connection may have multiple paths from an ingress queue to an egress queue since most ATM switch networks have the structure of the multistage interconnection network of switch elements. Therefore, paths already established for ABR connections may have to be changed to enhance throughput of the switch, if the ATM switch has the capability of adjusting paths of ABR connections while they are being served. We present three algorithms, in which throughput for ABR connections is estimated to decide whether or not paths of the connections should be changed. These algorithms are compared with an existing traffic management algorithm through simulation experiments. Results of the experiments show that the suggested algorithms give higher throughput in terms of the number of transmitted ABR cells without increasing the delay time of ABR services as well as quality of service (QoS) guaranteed services or decreasing the number of transmitted cells of QoS guaranteed services.

TCP Vegas-A: Improving the Performance of TCP Vegas

While it has been shown that TCP Vegas provides better performance compared to TCP Reno, studies have identified various issues associated with the protocol. We propose modifications to the congestion avoidance mechanism of the TCP Vegas to overcome these limitations. Unlike the solutions proposed in the past, our solution, named TCP Vegas-A, is neither dependent on optimising any critical parameter values nor on the buffer management scheme implemented at the routers and hence can be implemented solely at the end host. Our simulation experiments over wired as well as over geosynchronous and lower earth orbit satellite links show that TCP Vegas-A is able to overcome several of the identified problems-it can obtain a fairer share of the network bandwidth in wired and satellite scenarios, tackle rerouting issues, rectify Vegas's bias against higher bandwidth flows and prevail over fluctuating RTT conditions of a lower earth orbit satellite link. At the same time, Vegas-A is able to preserve the unique properties of Vegas that had made it a noteworthy protocol.

Understanding CHOKe: Throughput and Spatial Characteristics

A recently proposed active queue management, CHOKe, is stateless, simple to implement, yet surprisingly effective in protecting TCP from UDP flows. We present an equilibrium model of TCP/CHOKe. We prove that, provided the number of TCP flows is large, the UDP bandwidth share peaks at (e+1)/sup -1/=0.269 when UDP input rate is slightly larger than link capacity, and drops to zero as UDP input rate tends to infinity. We clarify the spatial characteristics of the leaky buffer under CHOKe that produce this throughput behavior. Specifically, we prove that, as UDP input rate increases, even though the total number of UDP packets in the queue increases, their spatial distribution becomes more and more concentrated near the tail of the queue, and drops rapidly to zero toward the head of the queue. In stark contrast to a nonleaky FIFO buffer where UDP bandwidth shares would approach 1 as its input rate increases without bound, under CHOKe, UDP simultaneously maintains a large number of packets in the queue and receives a vanishingly small bandwidth share, the mechanism through which CHOKe protects TCP flows.

Application of a hash function to discourage MAC-layer misbehaviour in wireless LANs

Contention-based MAC protocols for wireless ad hoc LANs rely on random deferment of packet transmissions to avoid collisions. By selfishly modifying the probabilities of deferments greedy stations can grab more bandwidth than regular stations that apply standard-prescribed probabilities. To discourage such misbehaviour we propose a protocol called RT-hash whereby the winner of a contention is determined using a public hash function of the channel feedback. RT-hash is effective in a full hearability topology, assuming that improper timing of control frames is detectable and that greedy stations do not resort to malicious actions. Simulation experiments show that RT-hash protects regular stations` bandwidth share against various sophisticated greedy strategies of deferment selection; as such it may contribute to MAC-layer network security.

Performance evaluation and comparison of Westwood+, New Reno, and Vegas TCP congestion control

TCP congestion control has been designed to ensure Internet stability along with fair and efficient allocation of the network bandwidth. During the last decade, many congestion control algorithms have been proposed to improve the classic Tahoe/Reno TCP congestion control. This paper aims at evaluating and comparing three control algorithms, which are Westwood+, New Reno and Vegas TCP, using both Ns-2 simulations and live Internet measurements. Simulation scenarios are carefully designed in order to investigate goodput, fairness and friendliness provided by each of the algorithms. Results show that Westwood+ TCP is friendly towards New Reno TCP and improves fairness in bandwidth allocation whereas Vegas TCP is fair but it is not able to grab its bandwidth share when coexisting with Reno or in the presence of reverse traffic because of its RTT-based congestion detection mechanism. Finally results show that Westwood+ remarkably improves utilization of wireless links that are affected by losses not due to congestion.

A generalized hierarchical fair service curve algorithm for high network utilization and link-sharing

The number of real-time applications, such as video-on-demand and video conferencing, is rapidly increasing. Real-time data now occupies a significant portion of network traffic. These applications require real-time service; as such, they need to bound end-to-end delays. Generally, real-time service is provided by reserving bandwidth in advance. Thus, to admit a large number of real-time applications, bandwidth needs to be reserved efficiently. In addition, to share limited bandwidth effectively among multiple agencies, a link-sharing service is required. A link-sharing service provides a guaranteed share of link bandwidth to each agency. To provide both real-time and link-sharing services simultaneously, several hierarchical link-sharing schemes have been proposed. However, existing hierarchical schemes still have problems in achieving a high level of network utilization, especially when variable-bit-rate (VBR) video sessions participate in a hierarchy. The problem is serious since a large portion of real-time applications transmit MPEG-coded VBR data. We propose a new scheduling algorithm that can achieve a high level of network utilization even with VBR video sessions. The proposed scheduling algorithm is a hierarchical link-sharing scheme; it is an extension of the H-FSC generalized with a new bandwidth reservation scheme. The new scheme provides tight bandwidth reservations for VBR video sessions. Even with the advantage in network utilization, the generalized H-FSC does not require high overhead for scheduling; the scheduling complexity remains the same as that of the H-FSC.

RED-DTB: A dual token bucket based queue management algorithm

Improving the Quality of Service (QoS) of Internet traffic is widely recognized as a critical issue for the next-generation networks. In this paper, we present a new algorithm for the active queue management, namely RED-DTB. This buffer control technique is used to enforce approximate fairness among a large number of concurrent Internet flows. Like RED (Random Early Detection) algorithm, the RED-DTB mechanism can be deployed to actively respond to the gateway congestion, keep the gateway in a healthy state, and protect the fragile flows from being stolen bandwidth by greedy ones. The algorithm is based on the so-called Dual Token Bucket (DTB) pattern. That is, on the one hand, every flow is rate-limited by its own token bucket, to ensure that it can not consume more than its fair share of bandwidth; On the other hand, to make some compensations to less aggressive flows, such as connections with larger round trip time or smaller sending window, and to gain a relatively higher system utilization coefficient, all flows, depending on their individual behavior, may have a chance to fetch tokens from the public token bucket when they run out of their own share of tokens. The algorithm is analyzed and evaluated by simulations, and is proved to be effective in protecting the gateway buffer and controlling the fair allocation of bandwidth among flows.

Rate-based congestion control in ATM switching networks using a recursive digital filter

The present paper proposes a control-theoretic approach to design rate-based controllers in order to flow-regulate the best-effort service in asynchronous transfer mode (ATM) switching networks. The proposed control uses a recursive digital filtering controller, where the control parameters can be designed to ensure the stability of the control loop in a control-theoretic sense. The stability of closed-loop congestion-controlled systems is analyzed by using Schur–Cohn stability criterion, which leads to certain necessary and sufficient stability condition under which the controlled ATM switching network is asymptotically stable in terms of buffer occupancy. Such proposed stability condition is then shown to be a key tool in designing a wide scope of adaptive controllers. Further, we demonstrate that a fair share of the available bandwidth at the bottleneck node can be achieved according to the proposed control policy. Simulations are performed that show good performance of both local area networks and wide area networks if implemented by the proposed control schemes.

An efficient scheme to achieve weighted max-min fairness on bandwidth allocation

In this paper, a scheme is proposed to achieve weighted max-min fairness on bandwidth allocation. The scheme supports various bandwidth requirements of users according to their weights. With the reservation strategy, the minimum bandwidth of each flow is guaranteed. The scheme achieves fair allocation rapidly when the network condition changes. The basic idea is to find the bandwidth share of each active flow at its bottleneck link and then assign this bottleneck share as its fair allocation. The correctness of the scheme is also proven. The scheme is efficient since it takes O(log N ) time to allocate bandwidth to N active flows. Simulations performed validate that the scheme reacts quickly to the changes of network condition and re-establishes the state of fair allocation fast.