Interference management using basestation coordination in broadband wireless access networks

This paper proposes a transmission‐scheduling algorithm for interference management in broadband wireless access networks. The algorithm aims to minimize the cochannel interference using basestation coordination while still maintaining the other quality of service (QoS) requirements such as packet delay, throughput and packet loss. The interference reduction is achieved by avoiding (or minimizing) concurrent transmission of potential dominant interferers. Dynamic slot allocation based on traffic information in other cells/sectors is employed. In order to implement the algorithm in a distributed manner, basestations (BSs) have to exchange traffic information. Both real‐time and non‐real‐time services are considered in this work. Results show that significant reduction in the packet error rate can be achieved without increasing the packet delay at low to medium loading values and with a higher but acceptable packet delay at high loading values. Since ARQ schemes can also be used for packet error rate reduction, we compare the performance of the proposed scheme with that of ARQ. Results indicate that although ARQ is more effective in reducing packet error rate, the proposed algorithm incurs much less packet delay particularly at medium to high loading. Copyright © 2006 John Wiley & Sons, Ltd.     

LTE系统上行链路调度算法研究

长期演进(Long Term Evolution,LTE)已经成为4G无线技术标准。目前,LTE分组调度的下行链路调度被大多数研究者研究,上行链路的研究相对较少。针对上行链路调度无法保证实时业务分组在延迟期限内传输,存在公平性较差、分组丢弃多的问题。因此,提出了一种新的上行链路调度算法。该算法根据实时业务的延迟约束条件建立目标整数线性规划模型,再根据目标整数线性规划模型进行调度。实验结果表明,该算法能保证实时业务分组在延迟期限内传输,适用于实时业务,能确保公平性,最小化分组丢弃,具有较好的适用性。     

按需轮循的蓝牙微微网调度算法与性能评估

提出了一种低时延、高信道利用率与公平性好的按需轮循的蓝牙微微网调度算法.首先通过速率估计过程计算出上行链路分组到达率的估计值,然后在主调度过程中分别计算各链路对自上次轮循时刻到当前时刻的分组到达数量的估计值与等待时间,根据提出的动态轮循选择规则选出优先度最高的链路对进行轮循,再根据自适应轮循间隙规则计算出下次轮循应该进行的时间.分别在混合、下行与上行三种传输模型的仿真中进行了与其他经典算法的性能比较.结果表明:算法在前两种传输模型中具有最佳的整体性能,尤其是时延分别降低了24.1%与41.4%以上,在第三种传输模型中也具有较佳的整体性能.该算法效率高、实现复杂度低,可应用于蓝牙与其他无线网络.     

Scheduling nonuniform traffic in a packet switching system withlarge propagation delay

Transmission algorithms are introduced for use in a single-hop packet switching system with nonuniform traffic and with propagation delay that Is large relative to the packet transmission time. The traffic model allows arbitrary traffic streams subject only to a constraint on the number of data packets which can arrive at any individual source in the system or for any individual destination in the system over time periods of specified length. The algorithms are based primarily on sending transmission schedules to the receivers immediately before transmitting each data packet multiple times so that the receiver can maximize the number of packets it captures. An algorithm based on matchings in a random graph is shown to provide mean total delay divided by mean propagation delay arbitrarily close to one, as the propagation delay tends to infinity     

Active intra-ONU scheduling with proportional guaranteed bandwidth in long-reach EPONs

The unused slot remainder (USR) problem in Ethernet Passive Optical Networks and long-reach Passive Optical Networks (LR-PONs) results in both a lower bandwidth utilization and a greater packet delay. In a previous study by the current group, an Active intra-ONU Scheduling with predictive queue report scheme was proposed for resolving the USR problem by predicting the granted bandwidth in advance based on the arrival traffic estimates of the optical network units (ONUs). However, it was found that the higher bandwidth prediction error in the proposed scheme prevents the network performance from being improved. Accordingly, the present study proposes a non-predictive-based ONU scheduling method designated as Active intra-ONU Scheduling with proportional guaranteed bandwidth (ASPGB) to improve the performance of LR-PONs. In the proposed method, the maximum guaranteed bandwidth of each ONU is adapted dynamically in accordance with the ratio of the ONU traffic load to the overall system load. Importantly, the proposed dynamic bandwidth allocation approach reduces the dependence of the network performance on the granted bandwidth prediction since the maximum guaranteed bandwidth determined by the Optical Line Terminal more closely approaches the actual bandwidth demand of each ONU. To solve the idle time problem arising in the event of an excess bandwidth reallocation, ASPGB is integrated with an improved early allocation (IEA) algorithm (a kind of Just-In-Time scheduling). The simulation results show that the IEA-ASPGB scheme outperforms previously published methods in terms of bandwidth utilization and average packet delay under both balanced and unbalanced traffic load conditions.     

A new traffic allocation algorithm in Ad hoc networks

~~A new traffic allocation algorithm in Ad hoc networks1. Su Kai-rong, ZHENG Wu,XIAO Shu-cheng, et al. Research on technology of mobile Ad hoc network access. Joural of Chongqing University of Posts and Telecommunications: Natural Science. 2005, 17(5): 61…     

B—DMB快速分组交换机的性能分析

由于不能预先对进入B-DMB交换机的各分组进行调整,可能出现同一时隙有2个或2个以上的分组都要到达某一输出端的情形,因此在B-DMB交换机中不可避免地要加进一些缓冲器。本文对B-DMB交换机的输入、输出缓冲器长度和分组时延进行了分析和计算,得出了一些有用的结论。     

Performance evaluation of wireless multimedia CDMA networks usingadaptive transmission control

When multimedia information including speech, image, data etc. are transmitted, the difference in information rates, required quality, and traffic performance should be taken into account. A wireless spread spectrum system can achieve a flexible balance of these differences because of inherent asynchronous code-division multiple-access (CDMA) capability. In order to accomplish wireless multimedia communication, three parameters are investigated: the transmission power, packet length, and time slot. We propose and investigate two methods to accomplish wireless multimedia communication by using these parameters: (1) a method based on control of both transmission power and transmission packet length and (2) a method based on control of both transmission power and adaptive time slot. We derive the optimal transmission packet length and transmission power in (1) and derive optimal adaptive time slot and transmission power in (2) by using traffic of measurement and nonlinear programming. By the comparison, the two proposed methods are compared to the minislotted alternating priority (MSAP). Numerical results demonstrate that the proposed methods, when the priority of each medium is graded, show better performance than other systems in terms of average throughput and delay time of all media     

Frame‐based adaptive uplink scheduling algorithm in orthogonal frequency division multiple access‐based Worldwide Interoperability for Microwave Access networks

Because the orthogonal frequency division multiple access physical resource available for scheduling in Worldwide Interoperability for Microwave Access networks is frame by frame, an uplink scheduler located at the base station must efficiently allocate available resources to the subscriber stations in response to constant or bursty data traffic on a per‐frame basis. Available resources for real‐time and nonreal‐time traffics, called frame‐based adaptive bandwidth allocation and minimum guarantee and weight‐based bandwidth allocation, respectively, are proposed in this paper. Moreover, both short‐term and long‐term bandwidth predictions for traffic are incorporated so that the long‐term bandwidth prediction can have sustainable throughput requirement, and the short‐term bandwidth prediction can meet the objectives of low delay and jitter. For the scenarios studied, it shows that system performance of the proposed algorithm is better than the hybrid (earliest deadline first + weighted fair queuing + FIFO) algorithm in terms of packet delay, jitter, throughput, and fairness. Copyright © 2011 John Wiley & Sons, Ltd.     

具有多重码的预约Slotted ALOHA协议性能分析

本文对具有多重码的预约Slotted ALOHA协议进行了性能分析。它是基于帧的协议且在一个时隙内有多重码可用来传输分组信息。文中利用离散时间,离散状态和Markov链来分析一个小区上行链路的话音分组吞吐量和数据分组延时性能,并提出了两种改进碰撞解决的方法,得到了较好的系统特性。     

A novel adaptive traffic prediction AQM algorithm

In the Internet, network congestion is becoming an intractable problem. Congestion results in longer delay, drastic jitter and excessive packet losses. As a result, quality of service (QoS) of networks deteriorates, and then the quality of experience (QoE) perceived by end users will not be satisfied. As a powerful supplement of transport layer (i.e. TCP) congestion control, active queue management (AQM) compensates the deficiency of TCP in congestion control. In this paper, a novel adaptive traffic prediction AQM (ATPAQM) algorithm is proposed. ATPAQM operates in two granularities. In coarse granularity, on one hand, it adopts an improved Kalman filtering model to predict traffic; on the other hand, it calculates average packet loss ratio (PLR) every prediction interval. In fine granularity, upon receiving a packet, it regulates packet dropping probability according to the calculated average PLR. Simulation results show that ATPAQM algorithm outperforms other algorithms in queue stability, packet loss ratio and link utilization.     

Dynamic connectivity and path formation time in Poisson networks

The connectivity of wireless networks is commonly analyzed using static geometric graphs. However, with half-duplex radios and due to interference, static or instantaneous connectivity cannot be achieved. It is not necessary, either, since packets take multiple time slots to propagate through the network. For example, if a packet traverses a link in one time slot, it is irrelevant if the next link is available in that time slot also, but it is relevant if the next hop exists in the next time slot. To account for half-duplex constraints and the dynamic changes in the transmitting set of nodes due to MAC scheduling and traffic loads, we introduce a random multi-digraph that captures the evolution of the network connectivity in a dynamic fashion. To obtain concrete results, we focus on Poisson networks, where transmitters form a Poisson point process on the plane at all time instants. We first provide analytical results for the degree distribution of the graph and derive the distributional properties of the end-to-end connection delay using techniques from first-passage percolation and epidemic processes. Next, we prove that under some assumptions, the delay scales linearly with the source–destination distance even in the presence of interference. We also provide simulation results in support of the theoretical results.     

Low‐jitter slot assignment algorithm for deadline‐aware packet transmission in wireless video surveillance sensor networks

This work presents a distributed time‐slot assignment algorithm which adopts TDMA as Medium Access Control, specially suited to support applications with strict delay, jitter, and throughput requirements characterized by convergecast traffic pattern in sensor networks. (e.g. wireless video surveillance sensor networks). The proposed algorithm has three characteristics: (1) every node is guaranteed a path to the base station for its data delivery. In the path, sufficient resource is reserved and weighted fairness can be achieved. (2) It uses cascading time‐slot assignment and jitter minimization algorithm in each node to minimize jitter and end‐to‐end delay. (3) Nodes are only active during their scheduled slots and sleep otherwise. This offers energy saving by reducing idle listening and avoiding overhearing. The performance of the proposed algorithm is evaluated over simulations and analyzed theoretically in comparison with existing slot assignment algorithm. The results show that our algorithm provides lower end‐to‐end delay, jitter, and higher throughput. Copyright © 2010 John Wiley & Sons, Ltd.     

Performance measures and scheduling policies in ring networks

A unidirectional ring network is considered. A node may transmit at most one packet per slot to its downstream neighbor. Potentially all nodes may transmit at the same slot. The achievable performance is studied and policies are proposed for both the evacuation mode and continual operation. In the evacuation mode each node has initially an amount of packets destined for every other node of the ring, and no more packets are generated later. It is shown that the furthest destination first (FDF) policy, that gives priority to the packet with the longest way to go at each node, minimizes the time until every packet reaches its destination. Furthermore it is shown that the closest destination first (CDF) policy, that gives priority to the packet with the shortest way to go at each node, minimizes the average packet delivery time. A formula for the optimal evacuation time is obtained. The continual operation of the ring is considered then where packets are generated according to some arrival process. For any arrival sample path, the PDF maximizes the fraction of the time at which the ring is empty. The performance analysis of individual origin-destination traffic streams under FDF is facilitated based on the following. For each traffic stream, a single server priority queue is identified such that the average sojourn time of the traffic stream in the ring is equal to the aggregate transmission time plus the queueing delay of the low priority stream in the queue. Formulas for the sojourn time are obtained for iid arrivals. The performance of CDF and FIFO in continual operation is studied by simulation. It turns out that the CDF, minimum delay policy for the evacuation, has the worst performance in continual operation     

Network queue scheduling algorithm based on self-similar traffic level grading prediction

Self-similarity characteristic of network traffic will lead to the continuous burstness of data in the network.In order to effectively reduce the queue delay and packet loss rate caused by network traffic burst,improve the transmission capacity of different priority services,and guarantee the service quality requirements,a queue scheduling algorithm P-DWRR based on the self-similarity of network traffic was proposed.A dynamic weight allocation method and a service quantum update method based on the self-similar traffic level grading prediction results were designed,and the service order of the queue according was determined to the service priority and queue waiting time,so as to reduce the queuing delay and packet loss rate.The simulation results show that the P-DWRR algorithm can reduce the queueing delay,delay jitter and packet loss rate on the basis of satisfying the different service priority requirements of the network,and its performance is better than that of DWRR and VDWRR.     

Dynamic cache allocation routing strategy of Internet of things satellite node based on traffic prediction

Aiming at the routing problem of low earth orbit (LEO) Internet of things (IoT) satellite systems,a dynamic cache allocation routing strategy based on traffic prediction for IoT satellite nodes was proposed.Firstly,the space-time characteristics of traffic distribution in the LEO coverage area were analyzed,and an end-to-end traffic prediction model was proposed.Then,according to the traffic prediction result,a dynamic cache allocation routing strategy was proposed.The satellite node periodically monitored the traffic load of the inter-satellite link,dynamically allocated the cache resources of each inter-satellite link between the neighboring nodes.The cache allocation process was divided into two phases,initialization and system operation.At the same time,the traffic offload and packet forwarding strategy when the node was congested was proposed.By comparing the queuing delay and the forwarding delay,it was determined whether the data packet needs to be rerouted.The simulation results show that the proposed routing strategy effectively reduces the packet loss rate and average end-to-end delay,and improves the traffic distribution in the whole network.     

Efficient Traffic Scheduling for Real Time VBR MPEG Video Transmission Over DOCSIS-Based HFC Networks

Data-over-cable service interface specifications (DOCSIS), the de facto standard in the cable industry, defines a scheduling service called real-time polling service (rtPS) to provision quality of service (QoS) transmission of real-time variable bit rate (VBR) videos. However, the rtPS service intrinsically has high latency, which makes it not applicable to real-time traffic transport. In this paper, we present a novel traffic scheduling algorithm for hybrid fiber coax (HFC) networks based on DOCSIS that aims to provide QoS for real-time VBR video transmissions. The novel characteristics of this algorithm, as compared to those described in published literatures, include 1) it predicts the bandwidth requirements for future traffic using a novel traffic predictor designed to provide simple yet accurate online prediction; and 2) it takes the attributes of physical (PHY) layer, media access control (MAC) layer and application layer into consideration. In addition, the proposed traffic scheduling algorithm is completely compatible with the DOCSIS specification and does not require any protocol changes. We analyze the performance of the proposed traffic predictor and traffic scheduling algorithm using real-life MPEG video traces. Simulation results indicate that 1) the proposed traffic predictor significantly outperforms previously published techniques with respect to the prediction error and 2) Compared with several existing scheduling algorithms, the proposed traffic scheduling algorithm surpasses other mechanisms in terms of channel utilization, buffer usage, packet delay, and packet loss rate.     

基于多终端协同的高清视频多流并发自适应传输控制方法

摘要：针对高清视频在异构无线网络中以多流并发的方式进行传输,以提高传输速率,从而增强用户体验的问题,以最小化系统传输时延以及各路径间时延差为优化目标,联合考虑了视频发送端和接收端,自适应调整视频发送速率和接收端缓存大小以提高用户体验,建立了异构无线网络中视频多流并发传输的控制模型,并基于Pareto分布和P/P/1排队理论对具有自相似性和长相关性的视频流进行了研究,推导了并发传输系统的时延统计特性,并在此基础上提出了一种异构无线网络视频流自适应分流决策方法。仿真结果表明,与一般的负载均衡分流决策方法相比,提出的异构网络多流并发自适应传输控制方法在时延和分组丢失率方面都有一定的优越性。     

A tree-based slot allocation algorithm for loss-free slotted OBS networks

Optical burst switching (OBS) is regarded as one of the most promising switching technologies for next generation optical networks. However, the data burst contention problem is still unresolved thoroughly even though slotted OBS (SOBS) is studied as a new paradigm reducing the blocking rate. In this article, we propose a tree-based slot allocation (TSA) algorithm for loss-free SOBS networks, where the TSA algorithm originally avoids contention of the time-slots by reserving the time-slots with different time-slot positions for the source nodes, respectively. In order to manage the time-slots efficiently, we also propose an OBS superframe, which is a cyclic period and consists of multiple time-slots transmitted by the source nodes toward the same incoming port of a destination node. In addition, we attempt to optimize multiplexing of the OBS superframes to reduce wavelength consumption. On the other hand, when incoming traffic is beyond expectation, a source node may need more time-slots to prevent packet loss because of buffer overflow. For reallocation of the time-slots, we propose a flow control scheme managing some number of shared time-slots, where a control node adaptively allocates (or redeems) the time-slots to (or from) source nodes by utilizing the shared time-slots based on fluctuating traffic condition. Simulation results show that the blocking rate of the proposed TSA–OBS scheme is zero with acceptable queueing delay at moderate traffic offered loads. In addition, multiplexing optimization simulated in the 14-node NSFNET achieves a 63% reduction of wavelength consumption. Moreover, the proposed flow control scheme assisting the TSA algorithm maintains a target upper-bound of queueing delay at the source node, so that packet loss caused by buffer overflow is prevented.     

Idle-signal casting multiple access with data slot reservation(ICMA-DR) for packet radio communications

A population of terminals communicating with a central station over a packet-switched multiple access radio channel is investigated with regard to multiple access control schemes. The authors describe the ICMA-DR, which is an advanced idle-signal casting multiple access (ICMA) scheme characterized by data slot reservation. This improved central controlled multiple-access scheme for packet transmission in terrestrial radio communications is evaluated in terms of throughput traffic, throughput delay characteristics, and handling capacity. It is shown that the throughput characteristics of ICMA-DR are superior than those of ICMA or slotted ALOHA when a packet for data slot reservation is relatively short in comparison to that for upward data. Thus, it is shown that ICMA-DR is suitable for the packet radio multiple-access scheme, especially in the case where fading packet error occurs frequently and ordered traffic is heavy. The ICMA-DR scheme has been utilized for the access control channel of NTT's new 800-MHz-band high-capacity land mobile communication system since the Spring of 1988