基于STK实现中继卫星业务运行可视化

跟踪与数据中继卫星系统搭建了中低轨航天器及其地面站通信的桥梁,实现了信息的实时传输,并提高了突发事件的应对能力。为对跟踪与数据中继卫星系统(TDRSS)调度业务的质量以及效率进行有效评估,在STK功能和接口的基础上,提出了基于STK通信组件(STKX)的跟踪与中继卫星实时场景仿真及其原型系统实现的技术方法,设计并实现了中继卫星针对中继业务的运行可视化系统。仿真结果表明,基于STKX的仿真平台能够满足中继卫星地面终端对中继任务的实时任务监控以及业务调度评估需求。     

基于PF调度的多源多中继协作网络节点选择

针对多源多中继协作网络模型,提出一种公平的节点选择与用户调度方案。出于优化平均信道质量的思想,每个传输时隙,选择信道条件差异最大的两个用户同时进行传输信息。中继节点选择基于PF比例公平调度算法,能保证中继节点处的负载均衡。然后在中继端采用一种网络编码传输方案。最后,与其他2种调度方案相比较仿真系统的误码率性能。     

基于协作干扰的全双工中继系统混合安全传输协议

针对全双工中继系统的安全问题,提出一种基于协作干扰的物理层安全混合传输协议.系统根据自身信道状态信息,自适应地选择基于中继节点与信源节点轮流发送干扰(RSCJ)协议或基于中继节点与目的节点轮流发送干扰(RDCJ)协议中安全容量相对较高的协议进行信息的传输.在全双工节点自干扰因素被抑制的条件下,给出了系统的传输协议选择策略,并理论推导了采用RSCJ或RDCJ传输协议下系统的安全中断概率表达式.仿真结果表明,利用所提的混合安全传输协议可以显著提高全双工中继系统的安全中断性能,与传统未利用协作干扰的传输机制相比,具有明显的安全性能优势.     

无线双向中继网络编码系统用户调度方法性能分析

提出了一种用于无线双向中继网络编码系统的用户调度方法,基站或中继在调度用户时,综合考虑基站与中继和用户与中继之间信道的影响,使得基站采用原有预编码算法时,具有了波束赋形增益。仿真表明,系统吞吐量和首次传输的正确率明显提升,且算法复杂度低、用户公平性好,便于在实际系统使用。     

基于USRP的中继系统的实现

基于通用软件无线电外设(Universal Software Radio Peripheral,USRP)和通用计算机组成的软件无线电平台,在基于DVB-T标准的无线多媒体传输系统中设计实现数字基带放大转发(Amplify-and-Forward,AF)系统和译码转发(Decode-and-Forward,DF)系统,并实现了动态参数配置和中继信号可视化。实现的中继系统具有很大的灵活性和可扩展性。实际测试结果表明,所设计的AF和DF中继系统可实现信号中继转发,有效扩大信号的覆盖范围,提高数据传输可靠性。     

一种针对HSDPA中继系统的比例公平调度算法

由于系统在资源分配上的差异,现有的3GPP长期演进项目的后续演进(Long Term Evolution Advanced,LTE-A)中继系统中的调度算法并不能直接应用于高速下行链路分组接入(High Speed Downlink Packet Access,HSDPA)中继系统。提出了一种适合HSDPA中继系统的比例公平调度算法,并给出了算法流程。通过系统级仿真,比较了该算法与没有中继时的PF调度算法和两层比例公平(THPF)调度算法的性能。仿真结果表明,所提出的调度算法能够在提高用户吞吐率的同时更好地保证用户公平性要求。     

宽带无线中继传输在应急通信中的应用分析

以固定通信系统为载体,积极构建骨干传输链路,这就是TD-LTE宽带集群通信系统,与新时期突发公共事件现场的可视化指挥调度需求予以充分结合,将宽带无线中继传输系统加入其中,进而形成无线骨干传输备份链路.基于此,本文将宽带无线中继传输作为研究重点,并阐述其在应急通信当中的具体应用,希望有所帮助.     

中继卫星资源调度问题研究现状与展望

随着跟踪与数据中继卫星系统(TDRSS)的发展以及中继应用需求的增加,其资源调度成为决定系统性能的关键因素之一.分析了包括描述方法、活动与资源、约束条件等在内的中继卫星资源调度问题基本内涵,研究了目前中继卫星资源调度的数学模型和实现算法,比较了不同数学模型和实现算法的优缺点.针对未来卫星激光和微波链路联合组网的发展趋势,提出了基于多终端-多节点及基于资源故障的激光/微波混合链路中继卫星资源调度的基本方法.     

基于跨层机制的最佳协作中继选择及其系统实现

为了获得最佳的协作中继用户,通过推导协作中继传输系统的误符号率,提出了基于放大重传和解码重传下最佳协作中继的选择准则,并结合跨层协作的思想给出了基于MAC层来实现最佳协作中继选择的分布式协议.与此同时,提出了协作分集系统的物理层传输方案,分析了瑞利衰落信道环境下所提方案的系统性能,得到了系统误比特率的闭式解.此外,针对协作中继选择准则和协作分集传输方案的误码性能进行了系统仿真和性能比较,结果表明,所提出的协作中继选择准则明显优于传统中继选择准则,且协作传输方案可以获得明显的分集增益,系统平均误比特率显著下降.     

基于卫星中继的无人机图像传输系统

基于卫星中继的无人机图像传输系统在遥感图像的传输中有着十分重要的作用。首先讨论了这种系统的应用意义 ,然后提出了一种具体的实现方案 ,对系统中的几个关键部分的设计进行了详细分析。目前的实验成果表明 ,这种基于卫星中继的无人机图像传输系统方案是可行的。     

基于VoIP队列的主动丢包的调度算法

讨论了网络拥塞对VoIP应用的影响,基于VoIP应用的特征,提出了一种新的适应于VoIP队列的主动丢包的调度算法——VASA算法。结合VASA和RED算法可实现新的VoIP队列管理和调度模式,从而可以有效改善网络拥塞状态下语音包转发的性能。     

一种新的语音流队列管理调度机制

本文提出一种新的语音流队列管理调度机制,结合随机早期探测(RED)和主动丢包调度算法实现因特网语音流的队列管理和调度.采用仿真方法分析了新机制的性能特征,并与RED做了性能对比.当网络拥塞时,该算法可有效改善包转发的性能.语音质量测试表明新机制是可行的和有效的.     

Performance evaluation of the MAC protocol in IEEE 802.16 systems with data and VoiP traffic scheduling

In the last few years, the metropolitan area networks (MAN) have increased their popularity and attracted the interest of the most important research groups all over the world. Among several standards, IEEE 802.16 has taken a relevant role providing high data rate in a big covering range with low implementation costs and multi‐traffic communications. The IEEE 802.16 networks can have a pre‐defined structure, with a central base station (BS) covering a cell in which a variable number of subscriber stations (SSs) can work. This paper deals with the proposal of a quality of service (QoS) driven scheduling algorithm to be used in an IEEE 802.16 network where different traffic types coexist. In particular, the paper mainly focuses on best effort data and VoIP communications, by proposing a scheduling technique that allows an efficient resource management of both traffic types by considering their specific QoS flavor. The performance evaluation has been carried out by considering both the phases of contention and packet scheduling, by means of a theoretical approach and computer simulations. Numerical results show the performance of the proposed algorithm by focusing on a scenario where the BS schedules the best effort and VoIP traffics of several SSs. Copyright © 2008 John Wiley & Sons, Ltd.     

Scheduling flows over LEO constellations on LMS channels

Satellite systems typically use physical and link layer reliability schemes to compensate the significant channel impairments, especially for the link between a satellite and a mobile end‐user. These schemes have been introduced at the price of an increase in the end‐to‐end delay, high jitter, or out‐of‐order packets. This is shown to have a negative impact both on multimedia and best‐effort traffic, decreasing the Quality of Experience (QoE) of users. In this paper, we propose to solve this issue by scheduling data transmission as a function of the channel condition. We first investigate existing scheduling mechanisms and analyze their performance for two kinds of traffic: VoIP and best‐effort. In the case of VoIP traffic, the objective is to lower both latency and jitter, which are the most important metrics to achieve a consistent VoIP service. We select the best candidate among several schedulers and propose a novel algorithm specifically designed to carry VoIP over LEO constellations. We then investigate the performance of the scheduling policies on internet‐browsing traffic carried by TCP, where the goal is now the maximize the users' goodput and select the best candidate in this case.     

基于IEEE 802.16协议的VoIP服务质量调度算法

本文分析了VoIP的特点和IEEE802．16协议对服务质量（QoS）的定义。基于已有的无线城域网对VoIP的QOS控制策略，提出一种新型的调度算法。通过数学建模和仿真试验，证明了新算法对减小延迟和增加信道利用率方面效果显著。     

VoIP Scheduling Algorithm for AMR Speech Codec in IEEE 802.16e/m System

This letter proposes an efficient uplink scheduling algorithm for voice over Internet protocol (VoIP) services with adaptive multi-rate (AMR) speech codec in IEEE 802.16e/m systems. The proposed scheduling algorithm adopts the random access scheme during silent-period to reduce the waste of uplink bandwidth considering the characteristics of AMR speech codec. The numerical results show that the proposed algorithm can increase the maximum supportable number of voice users by 26% compared to the conventional extended real-time polling service (ertPS).     

Enhanced VoIP Support in OFDMA-based Packet Radio Networks

As the first OFDMA-based cellular packet radio network, Evolved UTRAN has already been well studied in 3GPP and semi-persistent scheduling was selected as the most relevant packet scheduling scheme for VoIP traffic. With further evolution of Evolved UTRAN to meet the ambitious goal set by IMT-A, the performance of VoIP needs to be enhanced. In this article, we establish a semi-persistent scheduling centered framework for efficient VoIP support in OFDMA-based packet radio network and evaluate different enhancement methods, such as adaptive transmission bandwidth in persistent resource allocation, dynamic packet bundling, and frequency diversity transmission that can be added into this framework. In addition, we investigate practical constraints, e.g. wideband CQI reporting on real network performance as well as possible corrective measures. The whole system design is verified through large-scale network level simulations in which all the details of various VoIP-specific radio resource management algorithms and enhancement mechanisms as well as interactions between them are explicitly modeled. Simulation results indicate that VoIP in Evolved UTRAN can be well supported using this framework and further improved through different enhancement methods. Compared with fixed transmission bandwidth approach, adaptive transmission bandwidth approach can provide additional capacity gain of 11 and 8% respectively in 3GPP Macro Case 1 and Case 3 deployment scenario. In a highly-loaded network, switching from adaptive transmission bandwidth to dynamic packet bundling can accommodate 18 and 15% more users in Case 1 and 3 respectively. The performance degradation due to wideband CQI reporting can be partly compensated with frequency diversity transmission. It is finally concluded that the enhanced performance of VoIP is sufficient to meet the ambitious target set by IMT-A in all four ITU deployment scenarios.     

Comparative Performance Study of LTE Downlink Schedulers

Long time evolution (LTE) represents an emerging and promising technology for providing broadband, mobile Internet access. Because of the limitation of available spectrum resource, high spectrum efficiency technologies such as channel-aware scheduling need to be explored. In this work, we evaluate the performance of three scheduling algorithms proposed for LTE downlink transmission. The evaluation takes place in mixed traffic scenarios and aims at exploring strengths and weaknesses of the proposed algorithms. Simulation results illustrate the importance of real-time traffic awareness by schedulers when a specified level of quality of service is required. The research shows that lack of prioritisation of multimedia traffic will lead to severe degradation of video and VoIP services even with a relatively low network load.     

Performance Analysis of Uplink Scheduling Algorithms for VoIP Services in the IEEEE 802.16e OFDMA System

While the voice over Internet protocol (VoIP) services is expected to be widely supported in wireless mobile networks, the performance of VoIP services has not previously been evaluated in the IEEE 802.16e orthogonal frequency division multiple access (OFDMA) system taking the adaptive modulation and coding scheme into consideration. To support real-time uplink service flows, three different types of scheduling have been designed in the IEEE 802.16e standard: the unsolicited grant service (UGS), the real-time polling service (rtPS), and the extended rtPS (ertPS). In this paper, we compare the three real-time scheduling algorithms in terms of the performance of VoIP services by using the analytical and simulation models that we developed.

Assessing the quality of voice communications over Internet backbones

As the Internet evolves into a ubiquitous communication infrastructure and provides various services including telephony, it will be expected to meet the quality standards achieved in the public switched telephone network. Our objective in this paper is to assess to what extent today's Internet meets this expectation. Our assessment is based on delay and loss measurements taken over wide-area backbone networks and uses subjective voice quality measures capturing the various impairments incurred. First, we compile the results of various studies into a single model for assessing the voice-over-IP (VoIP) quality. Then, we identify different types of typical Internet paths and study their VoIP performance. For each type of path, we identify those characteristics that affect the VoIP perceived quality. Such characteristics include the network loss and the delay variability that should be appropriately handled by the playout scheduling at the receiver. Our findings indicate that although voice services can be adequately provided by some ISPs, a significant number of Internet backbone paths lead to poor performance.