Markov-based channel characterization for tractable performance analysis in wireless packet networks

Finite-state Markov chain (FSMC) models have often been used to characterize the wireless channel. The fitting is typically performed by partitioning the range of the received signal-to-noise ratio (SNR) into a set of intervals (states). Different partitioning criteria have been proposed in the literature, but none of them was targeted to facilitating the analysis of the packet delay and loss performance over the wireless link. In this paper, we propose a new partitioning approach that results in an FSMC model with tractable queueing performance. Our approach utilizes Jake's level-crossing analysis, the distribution of the received SNR, and the elegant analytical structure of Mitra's producer-consumer fluid queueing model. An algorithm is provided for computing the various parameters of the model, which are then used in deriving closed-form expressions for the effective bandwidth (EB) subject to packet loss and delay constraints. Resource allocation based on the EB is key to improving the perceived capacity of the wireless medium. Numerical investigations are carried out to study the interactions among various key parameters, verify the adequacy of the analysis, and study the impact of error control parameters on the allocated bandwidth for guaranteed packet loss and delay performance.     

On the performance of early packet discard

In a previous paper, one of the authors gave a worst case analysis for the early packet discard (EPD) technique for maintaining packet integrity during overload in ATM switches. This analysis showed that to ensure 100% goodput during overload under worst case conditions requires a buffer with enough storage for one maximum length packet from every active virtual circuit. This paper refines that analysis, using assumptions that are closer to what we expect to see in practice, and examines how EPD performs when the buffer is not large enough to achieve 100% goodput. We show that 100% goodput can be achieved with substantially smaller buffers than predicted by the worst case analysis, although the required buffer space can be significant when the link speed is substantially higher than the rate of the individual virtual circuits. We also show that high goodputs can be achieved with more modest buffer sizes, but that EPD exhibits anomalies with respect to buffer capacity, in that there are situations in which increasing the amount of buffering can cause the goodput to decrease. These results are validated by comparison with simulation     

一种基于捕获的扩频分组无线网MAC层协议及其性能分析

在分组接入过程中,当每个用户终端以随机竞争的方式获得所需的频谱资源时,会引起用户间的冲突和资源的浪费,造成系统性能的降低。本文引入了一种基于捕获的接入方法,当一个时隙中存在多个干扰分组时,该方法仍有捕获到某个分组的可能,该方法在系统高负载的情况下能有效地利用资源。性能分析和计算结果表明,基于捕获的方法改善了系统的性能。     

Split-channel pipelined packet scheduling for wireless networks

To reduce medium access control (MAC) overhead and improve channel utilization, there has been extensive research on dynamically adjusting the channel access behavior of a contending station based on channel feedback information. This paper explores an alternative approach, named pipelined packet scheduling, to reduce the MAC overhead. MAC overheads can be divided into bandwidth-dependent and bandwidth-independent components and these overheads can both be reduced by using split-channel pipelining mechanisms, as demonstrated in this paper. In the past, pipelining mechanisms have not been well studied. This paper introduces two total pipelining schemes that attempt to fully pipeline contention resolution with data transmission. Further, the paper identifies shortcomings of total pipelining in the wireless environment and proposes a partial pipelining approach to overcome these shortcomings. Simulation results show that substantial performance improvement in channel utilization, average packet access delay, and access energy cost can be achieved with a properly designed scheme.     

QoS-guaranteed packet scheduling in wireless networks

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.     

Fair scheduling in wireless packet networks

Fair scheduling of delay and rate-sensitive packet flows over a wireless channel is not addressed effectively by most contemporary wireline fair-scheduling algorithms because of two unique characteristics of wireless media: (1) bursty channel errors and (2) location-dependent channel capacity and errors. Besides, in packet cellular networks, the base station typically performs the task of packet scheduling for both downlink and uplink flows in a cell; however, a base station has only a limited knowledge of the arrival processes of uplink flows. We propose a new model for wireless fair-scheduling based on an adaptation of fluid fair queueing (FFQ) to handle location-dependent error bursts. We describe an ideal wireless fair-scheduling algorithm which provides a packetized implementation of the fluid mode, while assuming full knowledge of the current channel conditions. For this algorithm, we derive the worst-case throughput and delay bounds. Finally, we describe a practical wireless scheduling algorithm which approximates the ideal algorithm. Through simulations, we show that the algorithm achieves the desirable properties identified in the wireless FFQ model     

Throughput and delay analysis of multi-channel wireless infrastructure networks

Wireless infrastructure networks (WINs) provide ubiquitous connectivity to mobile nodes in metro areas. The nodes in such backbone networks are often equipped with multiple transceivers to allow for concurrent transmissions in multiple orthogonal channels. In this study, we develop an analytical model for the estimation of the delay and throughput performance of wireless infrastructure networks employing slotted ALOHA channel access and slotted Carrier Sense Multiple Access with Collision Avoidance (CSMA/CA) over multiple channels. The analytical model, which takes into account the correlation due to multi-hop transmissions, approximates the performance observed through simulations accurately.     

Concerning mean packet delay as a performance measure in single channel networks

It is shown that, under certain conditions, it is possible to have mean packet delay greater than mean message delay in single channel networks. The analysis includes overheads and transmission interference. It is shown further that for exponential message lengths, under the same assumptions, mean packet delay is always greater than mean message delay.     

Providing deterministic packet delays and packet losses in multimedia wireless networks

‘Anytime, anywhere’ communication, information access and processing are much cherished in modern societies because of their ability to bring flexibility, freedom and increased efficiency to individuals and organizations. Wireless communications, by providing ubiquitous and tetherless network connectivity to mobile users, are therefore bound to play a major role in the advancement of our society. Although initial proposals and implementations of wireless communications are generally focused on near‐term voice and electronic messaging applications, it is recognized that future wireless communications will have to evolve towards supporting a wider range of applications, including voice, video, data, images and connections to wired networks. This implies that future wireless networks must provide quality‐of‐service (QoS) guarantees to various multimedia applications in a wireless environment. Typical traffic in multimedia applications can be classified as either Constant‐Bit‐Rate (CBR) traffic or Variable‐Bit‐Rate (VBR) traffic. In particular, scheduling the transmission of VBR multimedia traffic streams in a wireless environment is very challenging and is still an open problem. In general, there are two ways to guarantee the QoS of VBR multimedia streams, either deterministically or statistically. In particular, most connection admission control (CAC) algorithms and medium access control (MAC) protocols that have been proposed for multimedia wireless networks only provide statistical, or soft, QoS guarantees. In this paper, we consider deterministic QoS guarantees in multimedia wireless networks. We propose a method for constructing a packet‐dropping mechanism that is based on a mathematical framework that determines how many packets can be dropped while the required QoS can still be preserved. This is achieved by employing: (1) An accurate traffic characterization of the VBR multimedia traffic streams; (2) A traffic regulator that can provide bounded packet loss and (3) A traffic scheduler that can provide bounded packet delay. The combination of traffic characterization, regulation and scheduling can provide bounded loss and delay deterministically. This is a distinction from traditional deterministic QoS schemes in which a 0% packet loss are always assumed with deterministically bounding the delay. We performed a set of performance evaluation experiments. The results will demonstrate that our proposed QoS guarantee schemes can significantly support more connections than a system, which does not allow any loss, at the same required QoS. Moreover, from our evaluation experiments, we found that the proposed algorithms are able to out‐perform scheduling algorithms adopted in state‐of‐the‐art wireless MAC protocols, for example Mobile Access Scheme Based on Contention and Reservation for ATM (MASCARA) when the worst‐case traffic is being considered. Copyright © 2002 John Wiley & Sons, Ltd.     

Modeling and performance analysis of multihop packet radio networks

The design of packet radio networks involves a large number of issues which interact in a very complex fashion. Many of these pertain to the RF channel and its use, others pertain to the operational protocols. Clearly, no single model can be formulated which incorporates all the necessary parameters and leads to the optimum solution. The one essential element which complicates matters is that, contrary to point-to-point networks in which each channel is utilized by a single pair of nodes, the radio channel in packet radio networks is a multiaccess broadcast resource: i) in a given locality determined by radio connectivity, the channel is shared by many contending users, hence the need for channel access protocols; ii) radio is a broadcast medium and thus the action taken by a node has an effect on the actions taken by neighboring nodes and their outcome. Despite the complexity of the problem, there has been significant progress worth reporting on. The work accomplished so far has been either the analysis of specific examples of networks or an attempt to create models that would be useful in the design of general networks. The purpose of this paper is to survey the various modeling techniques that have been used for the performance analysis of packet radio networks, and to discuss the assumptions underlying these models, their scope of applicability, and some of the results obtained.     

Adaptive packet prioritisation for large wireless sensor networks

Large wireless sensor networks consisting of hundreds of devices are often used to conduct distinct monitoring tasks simultaneously. Furthermore, the significant events that can be detected by the network may be detected in many different parts of the network. As a consequence, these multiple tasks and diverse events may also cause resource contention and degradation of network quality of service. For instance, congestion created by regularly occurring events which do not need urgent forwarding to the monitoring centres may create delays for other events which need to be reported very rapidly. The Random Re-Routing (RRR) protocol for wireless sensor networks has therefore been proposed to adaptively prioritise the packets of certain urgent events above those of other less urgent events which may be of a more routine nature. In this paper we extend the RRR protocol to improve performance, and we use simulation experiments to illustrate heuristic algorithms for matching routing priorities to the requirements of the sensor network??s monitoring tasks. The approaches are based on selecting the protocol??s parameters prior to network deployment, and on using an acknowledgement mechanism for adapting the parameters during network operation.     

Distributed adaptive rate system for wireless packet networks

A distributed adaptive rate system for wireless packet networks is proposed. Compared to the centralized adaptive rate system that needs to know the packet retransmission probability to maximize the throughput, this system need not know the packet retransmission probability in advance and can achieve the optimal adaptive rate system that maximizes the throughput for wireless packet networks.     

无线传感器网络中丢包现象的研究

无线传感器网络(Wireless Sensor Network,WSN)中通信链路质量的好坏影响通信的效率与可靠性,因而研究影响WSN中包发送成功与否的原因具有现实意义。在一定范围内,节点间的距离和包的接收没有必然的关系。同时,由于应用程序设计的原因,只要节点在某一时刻没有收到包,就会在下一时刻出现丢包的情况。最后,来自无线局域网(Wireless Local AreaNetwork,WLAN)的干扰对WSN通信具有较大影响。     

On the delay performance in multi-antenna wireless networks using contention-based feedback

When a spread spectrum contention-based feedback channel is employed, it has been shown that most of the multiuser diversity gain can be maintained without any delay constraint. To address the delay in sending packets, we investigate how a spread spectrum contention-based feedback channel affects the delay and throughput performance. Using large deviations techniques, we show that most of the multiuser diversity gain can be maintained while satisfying a delay constraint in which the probability that the maximum delay of any bits in any users? queues is less than a specified value. In addition, we show that the spectral efficiency improves as the number of users increases while maintaining a fixed normalized maximum delay (which is defined as the ratio of the maximum delay and the number of users) as the number of users increases.     

TCP throughput and fairness performance in presence of delay spikes in wireless networks

Wireless networks are likely to experience delay spikes exceeding several times the typical round‐trip‐time figures, which can cause spurious timeouts that lead to unnecessary retransmissions and reduction of the TCP sender's transmission rate, and thus, the throughput of the TCP is degraded. This paper presents some research results on the effect of delay spikes caused by handover on TCP performance by using three different mobility models. It is shown that the throughput of TCP connection over a single bottleneck link is decreased in the presence of delay spikes significantly. Furthermore, it is shown that the fairness feature of TCP is also severely affected in the presence of delay spikes. Copyright © 2005 John Wiley & Sons, Ltd.     

A packet delay analysis for cellular digital packet data

We formulate and analyze a model of voice and data burst traffic for cellular digital packet data (CDPD). To develop a tractible model, we make some reasonable simplifying assumptions to model the voice traffic and one of the logical CDPD channels as an M/M/1 queue in a random environment. Rather than explicitly solve the resulting matrix quadratic equation, we develop an asymptotic analysis that gives us simple approximate formulas (with error bounds) for many performance measures of interest. In particular, one metric that we highlight comes from using our approximate mean data burst delay formula to characterize the performance of a set of interacting logical CDPD channels. We do so by describing it in terms of an equivalent number of independent, dedicated CDPD channels, We can use this metric to suggest what the optimal number of logical CDPD channels for a given system should be     

Towards an end-to-end delay analysis of wireless multihop networks

In wireless multihop networks, end-to-end (e2e) delay is a critical parameter for quality of service (QoS) guarantees. We employ discrete-time queueing theory to analyze the end-to-end (e2e) delay of wireless multihop networks for two MAC schemes, m-phase TDMA and slotted ALOHA. In one-dimensional (1-D) networks, due to the lack of sufficient multiplexing and splitting, a space–time correlation structure exists, the nodes are spatially correlated with each other, and the e2e performance cannot be analyzed as in general two-dimensional networks by assuming all nodes independent of each other. This paper studies an 1-D network fed with a single flow, an extreme scenario in which there is no multiplexing and splitting. A decomposition approach is used to decouple the whole network into isolated nodes. Each node is modeled as a GI/Geo/1 queueing system. First, we derive the complete per-node delay distribution and departure characterization, accounting for both the queueing delay and access delay. Second, based on the departure process approximation, we define a parameter to measure the spatial correlation and its influence on the e2e delay variance. Our study shows that traffic burstiness of the source flow and MAC together determines the sign of the correlation.     

Numerical analysis of hardware architecture for header compression and packet aggregation on wireless networks

This paper proposes a numerical analysis model to predict the processing delay of a hardware architecture for robust header compression and packet aggregation on wireless mesh networks. The analysis model is composed of a series of queue systems such as G/M/1, M ^[K]/M/1, M/M/1, and M/M/∞ that are one-to-one mapped into the constructed hardware components to characterize the concurrent operations and interactional relationship between encoding and decoding paths. Based on the co-simulation method which integrates NS-2 and SystemC, we show the analysis model properly approximates the processing delay of the hardware architecture. Additionally, the variation of processing delay occurring when a part of hardware components are differently configured is suitably characterized by the proposed model, and the overall mesh network behaviors is predicted by applying the numerical results into NS-2 simulations.     

无线传感器网络中基于链路质量的路径延时分析

基于链路质量给出了路径满足实时性概率的上界,并证明了计算其上界的时间复杂度为指数级。另外在考虑链路质量的基础上,提出了一种在给定的延迟阈值下最大化端到端数据分组发送成功概率的贪心算法(RROP)。根据给定的延迟阈值和链路质量,RROP算法通过设置每跳链路的最大重传次数来优化端到端数据分组发送成功的概率。证明该算法能够在多项式时间内找到最优解并且通过该最优解获得路径满足实时性概率的一个近似最优的下界。实验结果表明给出的路径延迟分析上界和下界是准确的,并且提出的RROP算法在节省能量和满足实时性上比传统的方法能够获得高出10%以上的性能。     

End-to-end delay in wireless random networks

The focus of this letter is to derive a scaling law for the ene-to-end delay of wireless random networks under node mobility, where n nodes randomly move with the speed of v. To that end, we apply the cover time analysis and relate it to the delay scaling law. As a result, we derive that the mean delay per S-D pair as θ(n) or θ(√n÷v), and the worst case delay is θ(n log n) or θ(√log n÷v), corresponding to one slot time length that is either constant or 1÷v√n .