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1.
Haiyun Luo Xiaqiao Meng Ramjee R. Sinha P. Li Li 《Mobile Computing, IEEE Transactions on》2007,6(9):1060-1074
In third-generation (3G) wireless data networks, providing service to low data-rate users is required for maintaining fairness, but at the cost of reducing the cell's aggregate throughput. In this paper, we propose the unified cellular and ad hoc network (UCAN) architecture for enhancing cell throughput while maintaining fairness. In UCAN, a mobile client has both 3G interface and IEEE 802.11 -based peer-to-peer links. The 3G base station forwards packets for destination clients with poor channel quality to proxy clients with better channel quality. The proxy clients then use an ad hoc network composed of other mobile clients and IEEE 802.11 wireless links to forward the packets to the appropriate destinations, thereby improving cell throughput. We refine the 3G base station scheduling algorithm so that the throughput gains are distributed in proportion to users' average channel rates, thereby maintaining fairness. With the UCAN architecture in place, we propose novel greedy and on-demand protocols for proxy discovery and ad hoc routing that explicitly leverage the existence of the 3G infrastructure to reduce complexity and improve reliability. We further propose secure crediting mechanisms to motivate users that are not actively receiving to participate in relaying packets for others. Through both analysis and extensive simulations with HDR and IEEE 802.11b, we show that the UCAN architecture can increase individual user's throughput by more than 100 percent and the aggregate throughput of the HDR downlink by up to 50 percent. 相似文献
2.
This paper develops routing and scheduling algorithms for packet transmission in a low Earth orbit satellite network with a limited number of transmitters and buffer space. We consider a packet switching satellite network, where time is slotted and the transmission time of each packet is fixed and equal to one time slot. Packets arrive at each satellite independently with a some probability during each time slot; their destination satellite is uniformly distributed. With a limited number of transmitters and buffer space on-board each satellite, contention for transmission inevitably occurs as multiple packets arrive at a satellite. First, we establish the stability region of the system in terms of the maximum admissible packet arrival rate that can possibly be supported. We then consider three transmission scheduling schemes for resolving these contentions: random packet win, where the winning packet is chosen at random; oldest packet win, where the packet that has traveled the longest distance wins the contention; and shortest hops win (SHW), where the packet closest to its destination wins the contention. We evaluate the performance of each of the schemes in terms of throughput. For a system without a buffer, the SHW scheme attains the highest throughput. However, when even limited buffer space is available, all three schemes achieve about the same throughput performance. Moreover, even with a buffer size of just a few packets the achieved throughput is close to that of the infinite buffer case. 相似文献
3.
Next generation mobile networks are expected to provide seamless personal mobile communication and quality-of-service (QoS) guaranteed IP-based multimedia services. Providing seamless communication in mobile networks means that the networks have to be able to provide not only fast but also lossless handoff. This paper presents a two-layer downlink queuing model and a scheduling mechanism for providing lossless handoff and QoS in mobile networks, which exploit IP as a transport technology for transferring datagrams between base stations and the high-speed downlink packet access (HSDPA) at the radio layer. In order to reduce handoff packet dropping rate at the radio layer and packet forwarding rate at the IP layer and provide high system performance, e.g., downlink throughput, scheduling algorithms are performed at both IP and radio layers, which exploit handoff priority scheduling principles and take into account buffer occupancy and channel conditions. Performance results obtained by computer simulation show that, by exploiting the downlink queuing model and scheduling algorithms, the system is able to provide low handoff packet dropping rate, low packet forwarding rate, and high downlink throughput. 相似文献
4.
In a multihop network, packets go through a number of hops before they are absorbed at their destinations. In routing to its destination using minimum path, a packet at a node may have a preferential output link (the so-called “care” packet) or may not (the so-called “don't care” packet). Since each node in an optical multihop network may have limited buffer, when such buffer runs out, contention among packets for the same output link can be resolved by deflection. In this paper, we study packet scheduling algorithms and their performance in a buffered regular network with deflection routing. Using shufflenet as an example, we show that high performance (in terms of throughput and delay) can he achieved if “care” packets can be scheduled with higher priority than “don't care” packets. We then analyze the performance of a shufflenet with this priority scheduling given the buffer size per node. Traditionally, the deflection probability of a packet at a node is solved from a transcendental equation by numerical methods which quickly becomes very cumbersome when the buffer size is greater than one packet per node. By exploiting the special topological properties of the shufflenet, we are able to simplify the analysis greatly and obtain a simple closed-form approximation of the deflection probability. The expression allows us to extract analytically the performance trend of the shufflenet with respect to its buffer and network sizes. We show that a shufflenet indeed performs very well with only one buffer, and can achieve performance close to the store-and-forward case using a buffer size as small as four packets per node 相似文献
5.
This article proposes algorithms to determine an optimal choice of the Reed-Solomon forward error correction (FEC) code parameters (n,k) to mitigate the effects of packet loss on multimedia traffic caused by buffer overflow at a wireless base station. A network model is developed that takes into account traffic arrival rates, channel loss characteristics, the capacity of the buffer at the base station, and FEC parameters. For Poisson distributed traffic, the theory of recurrent linear equations is applied to develop a new closed form solution of low complexity of the Markov model for the buffer occupancy. For constant bit rate (CBR) traffic,an iterative procedure is developed to compute the packet loss probabilities after FEC recovery. 相似文献
6.
Jui-Chi Chen 《Wireless Personal Communications》2012,67(3):613-630
Base stations in next-generation broadband mobile networks (NGBMNs) must efficiently schedule different kinds of multimedia packets providing different quality of service (QoS) classes. During the past 10?years, many researchers have experimented with various packet scheduling schemes. In this paper we will propose a batch-arrival queuing model for evaluating NGBMN multimedia packet scheduling systems, and for obtaining three performance measures: packet loss rate (PLR), queuing delay (QD), and bandwidth utilization (BU). The three measures can be used to solve utilization optimization problems with QoS constraints. Specifically, a combination of a traffic statistic plus maximum PLR and QD constraints can be used to maximize BU for a multimedia packet scheduling management architecture. According to results from mathematical tests of the proposed model, it offers an efficient approach to managing scheduling buffers. The model and optimized parameters can be applied to flexible bandwidth deployment and classified buffer size control, thus enhancing profitability. 相似文献
7.
Scheduling Algorithms for a Slotted Packet Switch with either Fixed or Variable Length Packets 总被引:3,自引:2,他引:1
Callegati F. Cerroni W. Corazza G. Develder C. Pickavet M. Demeester P. 《Photonic Network Communications》2004,8(2):163-176
We address the problem of congestion resolution in optical packet switching (OPS). We consider a fairly generic all-optical packet switch architecture with a feedback optical buffer constituted of fiber delay lines (FDL). Two alternatives of switching granularity are addressed for a switch operating in a slotted transfer mode: switching at the slot level (i.e., fixed length packets of a single slot) or at the burst level (variable length packets that are integer multiples of the slot length). For both cases, we show that in spite of the limited queuing resources, acceptable performance in terms of packet loss can be achieved for reasonable hardware resources with an appropriate design of the time/wavelength scheduling algorithms. Depending on the switching units (slots or bursts), an adapted scheduling algorithm needs to be deployed to exploit the bandwidth and buffer resources most efficiently. 相似文献
8.
In current code division multiple access (CDMA) based wireless systems, a base station (BS) schedules packets independently of its neighbours, which may lead to resource wastage and the degradation of the system's performance. In wireless networks, in order to achieve an efficient packet scheduling, there are two conflicting performance metrics that have to be optimized: throughput and fairness. Their maximization is a key goal, particularly in next-generation wireless networks. This paper proposes joint packet scheduling and BS assignment schemes for a cluster of interdependent neighbouring BSs in CDMA-based wireless networks, in order to enhance the system performance through dynamic load balancing. The proposed schemes are based on sector subdivision in terms of average required resource per mobile station and utility function approach. The fairness is achieved by minimizing the variance of the delay for the remaining head-of-queue packets. Inter-cell and intra-cell interferences from scheduled packets are also minimized in order to increase the system capacity and performance. The simulation results show that our proposed schemes perform better than existing schemes available in the open literature. Copyright © 2007 John Wiley & Sons, Ltd. 相似文献
9.
Jacob Chakareski 《Journal of Visual Communication and Image Representation》2012,23(8):1189-1198
The paper studies the benefits of multi-path content delivery from a rate-distortion efficiency perspective. We develop an optimization framework for computing transmission schedules for streaming media packets over multiple network paths that maximize the end-to-end video quality, for the given bandwidth resources. We comprehensively address the two prospective scenarios of content delivery with packet path diversity. In the context of sender-driven systems, our framework enables the sender to compute at every transmission instance the mapping of packets to network paths that meets a rate constraint while minimizing the end-to-end distortion. In receiver-driven multi-path streaming, our framework enables the client to dynamically decide which packets, if any, to request for transmission and from which media servers, such that the end-to-end distortion is minimized for a given transmission rate constraint. Via simulation experiments, we carefully examine the performance of the scheduling framework in both multi-path delivery scenarios. We demonstrate that the optimization framework closely approaches the performance of an ideal streaming system working at channel capacity with an infinite play-out delay. We also show that the optimization leads to substantial gains in rate-distortion performance over a conventional content-agnostic scheduler. Through the concept of error-cost performance for streaming a single packet, we provide another useful insight into the operation of the optimization framework and the conventional scheduling system. 相似文献
10.
In this paper, a cross-layer design is proposed for downlink of orthogonal frequency division multiplexing systems which uses superposition coding (SC) scheme. SC theorem allows two users to share the same subchannel. Firstly, the subchannel will be allocated to degraded user (who is far away from the base station) and then this subchannel will be allowed to be shared by potential user (who is near to the base station). We also employed a packet dependent scheduling at the medium access control layer which decides the transmission order of packets according to the delay, size and quality of service priority level of packets. The weight of each user can be calculated by summing up the weights of all packets in the queues of that user. We have considered that each user is having multiple heterogeneous traffic queues. Simulation results show that the algorithm proposed in this paper is better than the previously reported algorithm in terms of total throughput and packet delay with the same computational complexity. 相似文献
11.
Baqai S. Farrukh Khan M. Miae Woo Shinkai S. Khokhar A.A. Ghafoor A. 《Selected Areas in Communications, IEEE Journal on》1996,14(7):1388-1403
Distributed, networked multimedia information systems will be a critical component of technology-based information infrastructures in the future. We present an infrastructure for supporting multimedia applications. We discuss various characteristics of multimedia data and the effect of the network on the required quality of presentation for multimedia data. We present a suite of synchronization protocols to support the quality of presentation. The crux of these protocols is the scheduling of multimedia information for synchronized delivery, over broadband networks with limited resources, and is identified as an NP-hard problem. We introduce two parameters which can be used to measure the performance of end-to-end synchronization protocols in a network supporting distributed multimedia applications. We propose and implement several heuristic scheduling algorithms, and compare their performance. We deduce the appropriateness of these algorithms in different types of distributed multimedia environments 相似文献
12.
iRGRR(iterative Request-Grant-based Round-Robin)算法是一种输入排队crossbar调度算法,具有硬件易实现、可扩展性强、性能优良等优点。在此基础上,该文提出一种新的高速crossbar分组调度策略iRGRR/PM(iRGRR with Packet Mode),可以更好地支持IP分组的调度,能够被应用于高速、大容量的路由器中。与iRGRR算法相比,iRGRR/PM简化了分组输出重组模块的设计,并且提高了crossbar的带宽资源利用率。文中简单分析了两种算法间的分组时延关系,并进行了详尽的仿真研究。结果表明:在相同条件下,iRGRR/PM算法具有更高的吞吐量,尤其在非均匀业务流下能获得接近100%的吞吐量;调度长分组时,iRGRR/PM算法具有更好的时延性能。 相似文献
13.
1 Introduction Broad-band integrated service digital networks (B-ISDN) based on the asynchronous transfer mode (ATM) are designed to support a wide variety of multimedia with diverse statistical characteristics and quality of services (QoS). Among the var… 相似文献
14.
Next-generation cellular wireless communication networks aim to provide a variety of quality-of-service (QoS)-sensitive packet-based services to downlink users. Included among these are real-time multimedia services, which have stringent delay requirements. Downlink packet scheduling at the base station plays a key role in efficiently allocating system resources to meet the desired level of QoS for various users. In this paper, we employ dynamic programming (DP) to study the design of a downlink packet scheduler capable of supporting real-time multimedia applications. Under well-justified modeling reductions, we extensively characterize structural properties of the optimal control associated with the DP problem. We leverage intuition gained from these properties to propose a heuristic scheduling policy, namely, Channel-Aware Earliest Due Date (CA-EDD), which is based on a "quasi- static" approach to scheduling. The per-time-slot implementation complexity of CA-EDD is only O(K) for a system with K downlink users. Experimental results show that CA-EDD delivers up to 50 percent of performance gains over benchmark schedulers. CA-EDD achieves these performance gains by using channel and deadline information in conjunction with application layer information (relative importance of packets) in a systematic and unified way for scheduling. 相似文献
15.
F. Callegati D. Careglio W. Cerroni G. Muretto C. Raffaelli J. Solé Pareta P. Zaffoni 《Optical Switching and Networking》2005,2(3):137-147
This paper deals with optical packet switches with limited buffer capabilities, subject to asynchronous, variable-length packets and connection-oriented operation. The focus is put on buffer scheduling policies and queuing performance evaluation. In particular a combined use of the wavelength and time domain is exploited in order to obtain contention resolution algorithms that guarantee the sequence preservation of packets belonging to the same connection. Four simple algorithms for strict and loose packet sequence preservation are proposed. Their performance is studied and compared with previously proposed algorithms. Simulation results suggest that by accepting some additional processing effort it is possible to guarantee very low packet loss probabilities while avoiding the out-of-sequence delivery. 相似文献
16.
To guarantee the quality of service (QoS) of a wireless network, a new packet scheduling algorithm using cross-layer design technique is proposed in this article. First, the demand of packet scheduling for multimedia transmission in wireless networks and the deficiency of the existing packet scheduling algorithms are analyzed. Then the model of the QoS-guaranteed packet scheduling (QPS) algorithm of high speed downlink packet access (HSDPA) and the cost function of packet transmission are designed. The calculation method of packet delay time for wireless channels is expounded in detail, and complete steps to realize the QPS algorithm are also given. The simulation results show that the QPS algorithm that provides the scheduling sequence of packets with calculated values can effectively improve the performance of delay and throughput. 相似文献
17.
A principal challenge in supporting real-time video services over ATM is the need to provide synchronous play-out in the face of stochastic end-to-end network delays. In this paper, an intelligent traffic smooth mechanism ( ITSM ) is proposed to meet the continuity requirement which is composed of a back-propagation neural network ( BPNN ) traffic predictor, a play-out buffer, and a fuzzy neural network ( FNN ) based play-out rate determinator. The BPNN traffic predictor online predicts the mean packet rate of the traffic in the future interval ( FI ) and the FNN is designed to adaptively determinate the play-out time according to the number of packets in the buffer and the traffic character predicted. Simulation results show that compared to the window mechanism, ITSM achieves high continuity with accepted delay. Furthermore, ITSM can be adaptively modified to meet the QoS of different kinds of services by FNN parameter training. 相似文献
18.
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. 相似文献
19.
现有无线分组调度算法一般以最大化系统吞吐量或者保证用户之间公平性为目标。然而,不同视频分组之间存在着重要性差异,使得传统的内容无关调度算法不适用于无线视频传输。充分发掘视频分组之间存在的重要性差异并准确衡量视频分组的传输失真,能够在资源受限的无线环境下更合理地分配资源,提高视频传输质量。该文首先提出一种分组级别视频传输失真衡量模型,用于预测视频分组丢失对视频质量的影响,然后考虑分组时限的影响,提出分组级别的时限扩展模型。在此基础上,利用正交频分复用(OFDM)技术在无线资源(时域、频域及功率)分配上的灵活性,提出一种基于梯度的失真和时限感知调度算法。仿真结果表明该文所提出的调度算法相对于传统内容无关算法最大有4.3 dB的平均峰值信噪比(PSNR)增益。 相似文献
20.
Efficient utilization of network resources is a key goal for emerging BWAS. This is a complex goal to achieve due to the heterogeneous service nature and diverse QoS requirements of various applications that BWAS support. Packet scheduling is an important activity that affects BWAS QoS outcomes. This paper proposes a new packet scheduling mechanism that improves QoS in mobile wireless networks which exploit IP as a transport technology for data transfer between BWAS base stations and mobile users at the radio transmission layer. In order to improve BWAS QoS the new packet algorithm makes changes at both the IP and the radio layers. The new algorithm exploits handoff priority scheduling principles and takes into account buffer occupancy and channel conditions. The packet scheduling mechanism also incorporates the concept of fairness. The algorithm offers an opportunity to maximize the carriers’ revenue at various traffic situations. Simulation results were compared to well-known algorithms which demonstrated the new packet scheduling algorithm is able to provide a low handoff packet drop rate, low packet forwarding rate, low packet delay, ensure fairness amongst the users of different services and generates higher revenue. Furthermore this research proposes a new and novel measure named “Satisfaction Factor” to measure the efficacy of various scheduling schemes and finally proposes four performance metrics for NodeB’s of in Next Generation Wireless Networks. 相似文献