Adaptive Fair Subcarrier/Rate Allocation in Multirate OFDMA Networks: Radio Link Level Queuing Performance Analysis

This paper presents a semi-analytical methodology for radio link level performance analysis in a multirate "orthogonal frequency-division multiple-access" (OFDMA) network with adaptive fair rate allocation. Multirate transmission is assumed to be achieved through adaptive modulation, and fair rate allocation is based on the principle of generalized processor sharing to allocate the subcarriers adaptively among the users. The fair rate allocation problem is formulated as an optimization problem with the objective of maximizing system throughput while maintaining fairness (in terms of transmission rate) among the users. The "optimal" fair rate allocation is obtained by using the "Hungarian method." A heuristic-based approach, namely the "iterative approach," that is more implementation friendly is also presented. The throughput performance of the iterative fair rate allocation is observed to be as good as that of optimal fair rate allocation and is better than that of the static subcarrier allocation scheme. Also, the iterative fair allocation provides better fairness compared to that for each of the optimal and the static subcarrier allocation schemes. To this end, a queuing model is formulated to analyze radio link level performance measures such as packet dropping probability and packet transmission delay under the above rate allocation schemes. In this formulation, packet arrivals are modeled by the discrete Markov modulated Poisson process, which is flexible to model different types of traffic arrival patterns. The proposed framework for radio link level performance analysis of multirate OFDMA networks is validated by extensive simulations. Also, examples on the application of the proposed model for connection admission control and quality-of-service provisioning are illustrated     

数据流的活动队列管理算法:MBLUE

MBLUE(Modified BLUE)是一种面向数据流的活动队列管理算法.它不是使用平均队列长度指示缓冲区拥塞状态,而是使用数据报丢弃的频率和队列空闲程度来管理网络拥塞.探测瓶颈连接早期的拥塞信息,通过数据报的丢弃和标记避免拥塞.它只维护一个先进先出队列,以较少的数据流状态信息,在不同流之间公平的分配网络带宽.能够适应瞬时的猝发流,能合理控制非TCP数据流,又能够保持较短的平均队列长度,从而控制、减轻网络拥塞.通过TCP/IP网络的模拟,证实算法在公平的分配网络带宽和降低数据报的丢失率上具有较好的鲁棒性.     

A Study on Differentiated Service Queuing Scheme with an Overflow Buffer Allocation Within a UMTS Core Network

According to the QoS features of the four types of UMTS traffic, this study proposes a priority-based queuing scheme to support differentiated services among all UMTS traffic; it bases on packet transmission priorities of four types of UMTS traffic to handle packets forwarding in a gateway within a UMTS core network. In the proposed queuing scheme, a static dedicated logical queuing buffer allocation and a dynamic overflow buffer allocation are used to support packet enqueuing/dequeuing jobs. In this study, the ns2 network simulator is used as a simulation platform and different categories of simulation scenarios are executed. Moreover, the IETF DiffServ scheme is used as a comparison to understand the packet forwarding performance of the proposed scheme. By analyzing simulation data, two important points can be drawn. The proposed queuing scheme supports a differentiated packet forwarding behavior among four types of UMTS traffic. And, the differentiated packet forwarding behaviors with the proposed queuing scheme are similar to the packet forwarding behavior with the IETF DiffServ scheme.     

Core‐stateless fair rate estimation fair queuing

Core‐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.     

Efficient fair queuing using deficit round-robin

Fair queuing is a technique that allows each flow passing through a network device to have a fair share of network resources. Previous schemes for fair queuing that achieved nearly perfect fairness were expensive to implement; specifically, the work required to process a packet in these schemes was O(log(n)), where n is the number of active flows. This is expensive at high speeds. On the other hand, cheaper approximations of fair queuing reported in the literature exhibit unfair behavior. In this paper, we describe a new approximation of fair queuing, that we call deficit round-robin. Our scheme achieves nearly perfect fairness in terms of throughput, requires only O(1) work to process a packet, and is simple enough to implement in hardware. Deficit round-robin is also applicable to other scheduling problems where servicing cannot be broken up into smaller units (such as load balancing) and to distributed queues     

面向ABR业务的公平排队策略

在基于ＡＴＭ－ＴＣＰ技术实现ＡＢＲ中,一个信元的丢失会导致该信元所属分组破坏,致命该信元所属分组的网络中传输都变得无效,重传机制来确保传输正确性,分组多次重传浪费大量带宽,并进一步加剧网络拥塞状况。ＡＴＭ－ＴＣＰ只保证每个信元对网络访问的公平性,而忽略了ＴＣ究组的公平性。本文提出了一种与分组长度无关的公平排队策略,提出以抛弃信元的代价为权值,来决定要抛弃的信元。它不仅能保证分组的公平性,而且能使网     

Earliest Deadline First Scheduling with Active Buffer Management for Real-Time Traffic in the Internet

We studied the problem of QoS guarantee for differentiated services. A two-level hierarchical scheduling framework was deployed to separate QoS metrics. Due to its desirable property of minimizing the maximum packet lateness, the Earliest Deadline First (EDF) scheduling was adopted to provide the in-class scheduling for the time-sensitive traffic. We employed an EDF scheduler combined with an active buffer management scheme (CHOKe) to improve the fairness of resource allocation and to maintain a good delay performance for real-time applications. Simulation results showed that the proposed scheme can achieve a better delay performance and make a more fair bandwidth allocation between the real-time TCP and UDP connections than the First Come First Served (FCFS) scheduling with the drop-tail buffer management which is commonly deployed in traditional IP routers.     

泛在无线环境中多无线接入网协同的业务分配策略

This paper introduces an adaptive traffic allocation scheme with cooperation of multiple Radio Access Networks (RANs) in universal wireless environments. The different cooperation scenarios are studied, and based on the scenario of cooperation in both network layer and terminal layer, an open queuing system model, which is aiming to depict the characteristics of packet loss rate of wireless communication networks, is proposed to optimize the traffic allocation results. The analysis and numerical simulations indicate that the proposed scheme achieves internetworking load balance to minimize the whole transmission delay and expands the communication ability of single-mode terminals to support high data rate traffics.     

Congestion-free communication in high-speed packet networks

The process of packet clustering in a network with well-regulated input traffic is studied and a strategy for congestion-free communication in packet networks is proposed. The strategy provides guaranteed services per connection with no packet loss and an end-to-end delay which is a constant plus a small bounded jitter term. It is composed of an admission policy imposed per connection at the source node, and a particular queuing scheme practiced at the switching nodes, which is called stop-and-go queuing. The admission policy requires the packet stream of each connection to possess a certain smoothness property upon arrival at the network. This is equivalent to a peak bandwidth allocation per connection. The queuing scheme eliminates the process of packet clustering and thereby preserves the smoothness property as packets travel inside the network. Implementation is simple     

TCP-Jersey for wireless IP communications

Improving the performance of the transmission control protocol (TCP) in wireless Internet protocol (IP) communications has been an active research area. The performance degradation of TCP in wireless and wired-wireless hybrid networks is mainly due to its lack of the ability to differentiate the packet losses caused by network congestions from the losses caused by wireless link errors. In this paper, we propose a new TCP scheme, called TCP-Jersey, which is capable of distinguishing the wireless packet losses from the congestion packet losses, and reacting accordingly. TCP-Jersey consists of two key components, the available bandwidth estimation (ABE) algorithm and the congestion warning (CW) router configuration. ABE is a TCP sender side addition that continuously estimates the bandwidth available to the connection and guides the sender to adjust its transmission rate when the network becomes congested. CW is a configuration of network routers such that routers alert end stations by marking all packets when there is a sign of an incipient congestion. The marking of packets by the CW configured routers helps the sender of the TCP connection to effectively differentiate packet losses caused by network congestion from those caused by wireless link errors. This paper describes the design of TCP-Jersey, and presents results from experiments using the NS-2 network simulator. Results from simulations show that in a congestion free network with 1% of random wireless packet loss rate, TCP-Jersey achieves 17% and 85% improvements in goodput over TCP-Westwood and TCP-Reno, respectively; in a congested network where TCP flow competes with VoIP flows, with 1% of random wireless packet loss rate, TCP-Jersey achieves 9% and 76% improvements in goodput over TCP-Westwood and TCP-Reno, respectively. Our experiments of multiple TCP flows show that TCP-Jersey maintains the fair and friendly behavior with respect to other TCP flows.     

ATM交换机EPD性能分析

利用ＡＴＭ网络传送ＴＣＰ分组时的一个重要问题是分组中任何一个信元丢失都半导致分组的重传,为解决这一问题,一个方法是发生拥塞时交换机有选择丢弃信元,在有限的存储器容量下,交换机智能化的信元集合源状态与缓存器占用的联合分布,基于这一结果,本文推导出的ＥＰＤ系统的分组丢失率的上界和下界,通过分析比较发现,使用ＥＰＤ的系统,当门限设为缓存器大小时其性能将优于无控制系统。     

Cross-layer resource allocation for QoS guarantee with finite queue constraint in multirate OFDMA wireless networks

Efficient radio resource allocation is essential to provide quality of service （QoS） for wireless networks. In this article, a cross-layer resource allocation scheme is presented with the objective of maximizing system throughput, while providing guaranteed QoS for users. With the assumption of a finite queue for arrival packets, the proposed scheme dynamically a/locates radio resources based on user＇s channel characteristic and QoS metrics derived from a queuing model, which considers a packet arrival process modeled by discrete Markov modulated Poisson process （dMMPP）, and a multirate transmission scheme achieved through adaptive modulation. The cross-layer resource allocation scheme operates over two steps. Specifically, the amount of bandwidth allocated to each user is first derived from a queuing analytical model, and then the algorithm finds the best subcarrier assignment for users. Simulation results show that the proposed scheme maximizes the system throughput while guaranteeing QoS for users.     

An explicit and fair window adjustment method to enhance TCP efficiency and fairness over multihops Satellite networks

Transmission control protocol (TCP) is the most widely used transport protocol in today's Internet. Despite the fact that several mechanisms have been presented in recent literature to improve TCP, there remain some vexing attributes that impair TCPs performance. This paper addresses the issue of the efficiency and fairness of TCP in multihops satellite constellations. It mainly focuses on the effect of the change in flows count on TCP behavior. In case of a handover occurrence, a TCP sender may be forced to be sharing a new set of satellites with other users resulting in a change of flows count. This paper argues that the TCP rate of each flow should be dynamically adjusted to the available bandwidth when the number of flows that are competing for a single link, changes over time. An explicit and fair scheme is developed. The scheme matches the aggregate window size of all active TCP flows to the network pipe. At the same time, it provides all the active connections with feedbacks proportional to their round-trip time values so that the system converges to optimal efficiency and fairness. Feedbacks are signaled to TCP sources through the receiver's advertised window field in the TCP header of acknowledgments. Senders should accordingly regulate their sending rates. The proposed scheme is referred to as explicit and fair window adjustment (XFWA). Extensive simulation results show that the XFWA scheme substantially improves the system fairness, reduces the number of packet drops, and makes better utilization of the bottleneck link.     

W-CDMA/UMTS下一种解决TCP有效性和公平性的跨层方案

在UMTS系统中,由于MAC层的包调度引入了用户间的竞争,带来了传输层TCP有效性和公平性问题,为了解决这个问题,该文提出了一种主动ACK包转发控制策略(AAFC),它的基本思想是在基站处嵌入AAFC模块,通过MAC层和传输层之间的消息传递来控制基站处ACK包的转发,以此来保证用户间TCP流的公平性以及链路层无线资源的最充分利用。仿真结果证明了AAFC方案的有效性。     

OSLG: A new granting scheme in WDM Ethernet passive optical networks

Several granting schemes have been proposed to grant transmission window and dynamic bandwidth allocation (DBA) in passive optical networks (PON). Generally, granting schemes suffer from bandwidth wastage of granted windows. Here, we propose a new granting scheme for WDM Ethernet PONs, called optical network unit (ONU) Side Limited Granting (OSLG) that conserves upstream bandwidth, thus resulting in decreasing queuing delay and packet drop ratio. In OSLG instead of optical line terminal (OLT), each ONU determines its transmission window. Two OSLG algorithms are proposed in this paper: the OSLG_GA algorithm that determines the size of its transmission window in such a way that the bandwidth wastage problem is relieved, and the OSLG_SC algorithm that saves unused bandwidth for more bandwidth utilization later on. The OSLG can be used as granting scheme of any DBA to provide better performance in the terms of packet drop ratio and queuing delay. Our performance evaluations show the effectiveness of OSLG in reducing packet drop ratio and queuing delay under different DBA techniques.     

A simple, scalable, and stable explicit rate allocation algorithmfor max-min flow control with minimum rate guarantee

We present a novel control-theoretic explicit rate (ER) allocation algorithm for the max-min flow control of elastic traffic services with minimum rate guarantee in the setting of the ATM available bit rate (ABR) service. The proposed ER algorithm is simple in that the number of operations required to compute it at a switch is minimized, scalable in that per-virtual-circuit (VC) operations including per-VC queueing, per-VC accounting, and per-VC state management are virtually removed, and stable in that by employing it, the user transmission rates and the network queues are asymptotically stabilized at a unique equilibrium point at which max-min fairness with minimum rate guarantee and target queue lengths are achieved, respectively. To improve the speed of convergence, we normalize the controller gains of the algorithm by the estimate of the number of locally bottlenecked VCs. The estimation scheme is also computationally simple and scalable since it does not require per-VC accounting either. We analyze the theoretical performance of the proposed algorithm and verify its agreement with the practical performance through simulations in the case of multiple bottleneck nodes. We believe that the proposed algorithm will serve as an encouraging solution to the max-min flow control of elastic traffic services, the deployment of which has been debated long due to their lack of theoretical foundation and implementation complexity     

Rate Control Algorithm Considering Mobility in Mobile Ad Hoc Networks

For reducing the loss of data packet due to network congestion and mobility, rate control algorithm is effective means for ensuring the fair allocation of network resource among communication flows and alleviating network congestion. The Transport Control Protocol (TCP) was originally designed for wire-line networks, where the links are assumed to be reliable and with fixed capacities. However, the performance of TCP becomes very poor when it is directly used in ad hoc networks. Some optimization-based congestion control schemes have been proposed to improve TCP performance in ad hoc networks, but the mobility of hosts and route change frequently are not considered for designing rate control algorithm. In this paper, we propose rate control algorithms in a more practical way by taking into account link congestion and node movement. Numerical results show that the rate control algorithms can approach the globally optimal value and show the effect of the node mobility on the algorithms.     

A Study on Adaptive Time Token Priority-Based Queuing Scheme

Several famous priority-based queuing schemes operated in a gateway to support differentiated services among internet traffic. Examining packet forwarding operations in these queueing schemes, they only support a priority-based service either in a packet enqueuing process or in a packet dequeuing process. If a queuing scheme can support priority-based services in both packet enqueuing/dequeuing processes; it would enhance differentiated service performance for internet traffic. This study proposes a priority-based queuing scheme with an adaptive time token allotment measure to support a differentiated packet forwarding process for different types of IP traffic both in packet enqueuing/dequeuing processes. Depending on packet sizes and packet forwarding priorities of IP traffic, the proposed queuing scheme assigns fix and adaptive time token thresholds dynamically to logical queuing buffers separately. With assigned time tokens, logical queuing buffers allow arrival IP packets to be enqueued in a differentiated way. Moreover, the proposed queuing scheme uses a transferred WRR dequeuing measure to enhance a differentiated packet forwarding process. The simulation results show that the proposed queuing scheme supports a differentiated packet forwarding process for different types of IP traffic. The differentiated packet forwarding performance supported by the proposed scheme is close to the IETF DiffServ scheme; this result shows that the proposed scheme can support differentiated packet forwarding performance for different types of IP traffic with a lower operation cost.     

A FIFO-based buffer management approach for the ATM GFR services

We investigate the TCP packet transmission over the ATM Guaranteed Frame Rate (GFR) service using a selective packet-discard with buffer vacancy tracking strategy and a packet-based push-out buffering scheme. Our approach is based on FIFO queueing discipline, it features fair sharing of available resources as well as feasibility in hardware implementation     

QoS based fair resource allocation in multi-cell TD/CDMA communication systems

Abstract-In a wireless multimedia code division multiple access (CDMA) system, the resources in terms of transmission rate and power should be efficiently distributed to each user to guarantee its quality-of-service (QoS) requirements. In, this paper, a resource allocation algorithm which combines packet scheduling and power assignment is proposed to achieve efficient resource utilization under QoS constraints. The packet scheduling is based on the fair packet loss sharing (FPLS) principle, and the power assignment is determined by the received power limited (RPL) scheme. The basic idea of FPLS is to schedule the transmission of multimedia packets in such a way that, all the users have a fair share of packet loss according to their QoS requirements, which maximizes the number of the served users with QoS satisfaction. The RPL scheme minimizes the received power for each packet. Given the propagation path loss, it in turn minimizes the transmitted power as well. The intercell interference from the scheduled packets is also limited in order to increase the system capacity.