DSMAC:一种适于无线多媒体传感器网络的信道接入协议

多媒体传感器网络作为一种多媒体信息获取和处理方式,已在军事、民用及商业领域中显示出广阔的应用前景.信道接入协议能否高效地使用无线信道是保证无线多媒体传感器网络通信的最关键的因素之一.分析支持多媒体业务传输的无线传感器网络信道接入协议的要求,提出适于多媒体传感器网络提供区分服务的信道接入协议--DSMAC(different service medium access control),对实时业务与非实时业务实现了区分服务,在信道接入帧内的随机竞争期实现突发业务及时接入,支持突发多媒体业务实时传输,并提出了多信道簇间传输方式,避免了隐终端冲突.最后,对协议的服务区分、实时性、吞吐量以及能量有效性等性能进行了仿真实验,验证了其优良性能.     

报警网络中随机信息最佳传输模型研究

在城市联网报警与监控系统的研究中,为保障报警业务数据的有效传输,在不同业务负载下如何设置合适的系统参数成为大型报警联网系统体系结构设计中的一个关键和难点问题.为解决上述问题,对大型报警联网系统中随机业务的最佳信息传输问题进行研究,利用随机过程理论分析了随机业务情况下,前端数据采集节点的个数分布参数、采集数据量分布参数、信道传输容量、数据缓存区大小对传输过程中平均丢包率的影响模型.通过对无缓存模式、有缓存模式、单数据包、复数据包情况下随机业务传输进行实验,提出的模型与仿真结果高度一致.改进算法可有效描述不同参数下的随机信息传输丢包情况,为大型报警联网系统中的节点参数设计提供指导.     

无线通信系统中信道拥塞动态调整的研究

无线通信系统的信道拥塞率能准确地反映信道的使用情况。分析了无线通信网覆盖中产生的信道拥塞,提出一种基于信道拥塞率的动态调整算法。该算法通过对无线通信系统网络侧统计数据的分析,动态调整系统中的信令信道和业务信道的比例,以解决无线通信系统中信道拥塞的问题。在相同的话务模型下,对算法应用前后进行仿真比较。仿真结果表明,采用所提算法可以提高网络的质量,特别是当短时间多用户同时申请信道时,可以明显提高系统的接通率。     

密集D2D网络中提升接入率的资源分配研究

在蜂窝网络中引入D2D通信可以增大系统吞吐量,提升资源利用率,减小终端功耗。传统资源分配算法只解决稀疏D2D网络中资源分配的问题,如何以有限的频谱资源将所有设备用接入网络是目前急需解决的问题。提出一种高接入率的资源分配算法,在基站的控制下为每条蜂窝信道选取满足接入条件的最优D2D链路接入。仿真验证表明,该算法有效提高了混合网络中D2D链路的接入率,同时有效增加了每条蜂窝信道的复用增益。     

M/P/C/K排队系统呼损性能研究

大量的研究表明网络中的业务量呈现自相似特征,但精确分析自相似业务流下的含多个服务装置、多缓存系统的排队性能非常困难。通过仿真得到服务时间是Pareto分布的M/P/C/K模型的平均呼损率（CBP）,将它与服务时间是指数分布的传统网络排队模型M/M/C/K的CBP公式比较,结果表明,在有缓存情况下,自相似业务量导致网络的排队呼损性能劣化。     

一种新的基于HCCA的流量调度算法

IEEE802.11e中MAC协议传送方式采用了混合协调功能(HCF),其中混合式协调控制信道访问(HCCA)是基于集中轮询方式接入业务,它提供了参数化的QoS接入,着重讨论变比特率数据流的接入控制,提出了一种新的根据网络的实际流量、动态调整服务间隔和传输机会,来灵活地分配网络资源的算法.仿真结果表明,该算法能有效提高网络的吞吐量和降低丢包率,大大提高系统性能.     

基于5G通信技术的变电站SF6气瓶智能管理系统设计

为确保变电站六氟化硫(SF₆)气瓶智能管理的时效性和可靠性,设计基于5G通信技术的变电站SF₆气瓶智能管理系统。通过SF₆气体采集器等装置,感知并采集变电站SF₆气瓶的状态数据。将该数据通过由5G通信技术构建的5G传输模块,传送至管理模块,并采用5G切片技术划分网络,以实现按需通信。管理模块接收经由5G传输模块传送的数据后,基于非分光红外差分检测方法,在线智能检测SF₆气瓶的泄漏浓度,并校核SF₆气体在线监测结果,以实现SF₆气瓶的智能管理。测试结果表明：网络传输平均速率在8～10 Gbit/s之间,极大程度接近传输速率峰值,通信的实时性良好;能够实时监测SF₆气瓶运行状态,在线监测精度较高,平均偏移程度低于0.035。该系统可全面呈现气体的泄漏情况,以及发生泄漏的时间和泄漏气体的浓度情况。     

无线通信系统中信道拥塞动态调整的研究

无线通信系统的信道拥塞率能准确地反映信道的使用情况。分析了无线通信网覆盖中产生的信道拥塞，提出一种基于信道拥塞率的动态调整算法。该算法通过对无线通信系统网络侧统计数据的分析，动态调整系统中的信令信道和业务信道的比例，以解决无线通信系统中信道拥塞的问题。在相同的话务模型下，对算法应用前后进行仿真比较。仿真结果表明，采用所提算法可以提高网络的质量，特别是当短时间多用户同时申请信道时，可以明显提高系统的接通率。     

无线接入网络中TCP流的上下行信道公平算法

在无线接入网络中,上行TCP流会极大地压制下行TCP流,导致严重的上下行信道TCP流不公平问题.本文指出TCP流的ACK包在接入节点下行缓存中的侵占性是上下行TCP不公平问题的直接原因,从限制缓存大小的新角度提出了MBA(Maximum Buffer for ACKs)算法.MBA算法基于上下行TCP流的不公平比例和缓存大小的关系,自适应地调节ACK包的最大缓存空间.理论分析和仿真实验结果表明MBA算法不但能通过限制ACK包的缓存空间实现上下行TCP流公平,还能通过减少无线信道ACK包传输概率提高网络总有效吞吐率.     

Adhoc网络Internet接入的动态QoS研究

Adhoc与Internet混合网络的QoS策略不同于传统的QoS保障,针对Adhoc与Internet混合的泛在网络有效感知,提出一种新型的Adhoc网络Internet接入的动态QoS模型RATD(ResourceAssignmentandTrafficDetection).该模型通过链路层信号测量激发网络上层的切换决策模块,基于网络接入点QoS状态、信道资源的实时监测与分配、应用业务类别以及呼叫接入方式,对已有的QoS成果进行拓展,提高了接入切换控制的效率,理论分析和仿真实验均验证了RATD模型的有效性.     

Investigation of the M2/G2/1/∞,N queue with restricted admission of priority customers and its application to HSDPA mobile systems

This paper investigates a queuing system for QoS optimization of multimedia traffic consisting of aggregated streams with diverse QoS requirements transmitted to a mobile terminal over a common downlink shared channel. The queuing system, proposed for buffer management of aggregated single-user traffic in the base station of High-Speed Downlink Packet Access (HSDPA), allows for optimum loss/delay/jitter performance for end-user multimedia traffic with delay-tolerant non-real-time streams and partially loss tolerant real-time streams. In the queuing system, the real-time stream has non-preemptive priority in service but the number of the packets in the system is restricted by a constant. The non-real-time stream has no service priority but is allowed unlimited access to the system. Both types of packets arrive in the stationary Poisson flow. Service times follow general distribution depending on the packet type. Stability condition for the model is derived. Queue length distribution for both types of customers is calculated at arbitrary epochs and service completion epochs. Loss probability for priority packets is computed. Waiting time distribution in terms of Laplace–Stieltjes transform is obtained for both types of packets. Mean waiting time and jitter are computed. Numerical examples presented demonstrate the effectiveness of the queuing system for QoS optimization of buffered end-user multimedia traffic with aggregated real-time and non-real-time streams.     

IEEE 802.16 Mesh模式下区分服务时隙调度算法

为了满足用户需求,IEEE802.16支持多种QoS。在IEEE802.16-2004定义的mesh模式的调度机制下,针对实时业务和非实时业务对时延具有不同敏感性的特点,提出了一种区分服务的时隙调度算法。该算法优先处理实时业务,并通过拥塞情况,自适应地调整实时业务的预留带宽。最后对算法进行了仿真,结果表明该算法下业务的延迟、服务请求拒绝率、时隙利用率均有所改善。     

A hybrid channel allocation algorithm with priority to handoff calls in mobile cellular networks

This paper proposes and investigates a novel analytical model of a hybrid channel allocation algorithm within wireless cellular networks. Each cell of the network consists of a predesigned fixed number of channels and the network may approve the request for extra channels for both new and handoff calls if all predesigned channels are occupied. This approval depends on the types of new and handoff calls, as well as the number of approved additional channels in the cell. If a request is denied for the arriving new call, this call will be blocked and cleared from the system. However, if a request is denied for an arriving handoff call, this call will not be blocked immediately but rather put on hold in a buffer with finite space. The implication behind this is to give priority to handoff calls. For this proposed hybrid channel allocation scheme, we first obtain the stationary distribution of each cell when there are i calls connecting to the system and j calls holding on in the buffer. We then derive new and handoff call blocking probabilities, the average number of borrowed channels, and the average delay period of handoff calls. The numerical results show that the proposed hybrid algorithm is more efficient than other approaches, specifically, in comparison with methods without a borrowing capability for new calls and those without a reserved buffer priority for handoff calls. The idea and results presented in this paper are expected to provide guidelines for field data processing within current wireless and mobile network design and performance evaluation.     

Admission control scheme based on priority access for wireless LANs

In order to support the quality of service (QoS) requirements at the medium access control (MAC) layer, the enhanced distributed channel access (EDCA) has been developed in IEEE 802.11e standard. However, it cannot guarantee the stringent real-time constraints of multimedia applications such as voice and video without an efficient method of controlling network loads. In this paper, we propose a measurement-based admission control scheme, which is made up of two parts: priority access and admission control. First, in order to measure the channel status per traffic type, we propose a priority access mechanism in which each priority traffic is distinguished by a busy tone, and separately performs its own packet transmission operation. Then, admission control mechanism protects existing flows from new ones, and maintains the QoS of the admitted flows based on the measured channel status information. Performance of the proposed scheme is evaluated by simulation. Our results show that the proposed scheme is very effective in guaranteeing the QoS of multimedia applications as well as in avoiding the performance starvation of low priority traffics.     

Constant bit rate network transmission of variable bit rate continuous media in Video-On-Demand servers

Multimedia computing is rapidly emerging as the next generation standard for human-computer interaction. One class of multimedia applications that has been gaining much attention is the real-time display of continuous media data such as video and audio, commonly known as Video-On-Demand (VOD) service. Although advances in computer and network technologies have made VOD service feasible, providing guaranteed quality, real-time video delivery still poses many technical challenges. One such challenge involves the transmission of continuous media traffic over high-speed networks.In this paper, we present an algorithm for determining the minimum buffer requirement for avoiding overflow or underflow at the client video display process, allowing the network scheduler at the VOD server to enforce a constant bit rate delivery of variable bit rate encoded continuous media. This strategy results in reduced congestion and cell loss at the network switch, and in simplified admission control parameters. Initial results indicate that buffer requirements for typical video streams range from 3.7 to 14.6 Megabytes, which is acceptable by today's multimedia PC standards. Further, we show that this approach increases the number of streams that can be multiplexed by a factor of 4.6 to 9.9 times when compared to peak and 90%-of-peak bandwidth allocation strategies.ECE Dept., Syracuse University, Syracuse, USACIS Dept., Syracuse University, Syracuse, USA     

Fuzzy-based rate control for real-time MPEG video

We propose a fuzzy logic-based control scheme for real-time motion picture expert group (MPEG) video to avoid long delay or excessive loss at the user-network interface (UNI) in an asynchronous transfer mode (ATM) network. The system consists of a shaper whose role is to smooth the MPEG output traffic to reduce the burstiness of the video stream. The input and output rates of the shaper buffer are controlled by two fuzzy logic-based controllers. To avoid a long delay at the shaper, the first controller aims to tune the output rate of the shaper in the video frame time scale based on the number of available transmission credits at the UNI and the occupancy of the shaper's buffer. Based on the average occupancy of the shaper's buffer and its variance, the second controller tunes the input rate to the shaper over a much larger time scale by applying a closed-loop MPEG encoding scheme. With this approach, the traffic enters the network at an almost constant bit rate (with a very small variation) allowing simple network management functions such as admission control and bandwidth allocation, while guaranteeing a relatively constant video quality since the encoding rate is changed only in critical periods when the shaper buffer “threatens” to overflow. The performance of the proposed scheme is evaluated through numerical tests on real video sequences     

Real-Time Virtual Channel Flow Control

Real-time communication system support for large scale parallel multicomputers becomes an important issue as the number of real-time applications developed on these systems increases. Flow control is a key component that affects the performance of the communication subsystem. We develop a range of new real-time virtual channel flow control schemes for wormhole networks. The flow control schemes differ in their priority mapping strategies, priority adjustment methods, and arbitration functions. The priority mapping strategy and priority adjustment method of a flow control scheme determine the priority of a message. The priority of a message is used for the virtual channel assignment and the physical channel arbitration. We discuss the trade-off between the performance and the hardware cost of each flow control scheme. A simulator is implemented for studying the performance of the schemes, and simulation experiments are designed to compare the importance of priority mapping, priority adjustment and arbitration toward the system performance. As wormhole networks scale to larger sizes, the average distance between source and destination nodes increases. The flits of messages in wormhole networks, which are buffered in nodes along the path from the source to the destination, consume network resources in these nodes. Therefore, increased scaling may lead to increased resource consumption, congestion, and late messages. In real-time systems, messages lose their value when they miss their deadlines. In order to reduce congestion, we provide a scheme for dropping messages that miss their deadlines.     

智能配电网通信多信道调度策略

为了有效提高基于无线传感网的配电网服务质量,进一步增强配电网中数据实时性,减少延迟,提出了一种基于优先级的多信道调度策略。首先,根据实时的信道链路状态信息,提出了基于最小跳数生成树的链路路由算法（LB-MHST）,克服了无线射频干扰,保证智能电网的服务质量;然后,针对配电网中不同数据包对于延迟要求的不同,考虑了数据优先级传输,有效地提高了感知节点的数据传输效率,进一步满足了配电网中的QoS性能要求。实验结果表明,与最小跳数生成树（MHST）算法相比,尤其是在多信道延迟感知传输中,提出算法在单信道、8信道和16信道时分别将高负载流量延迟性能提高了12%,15.2%和18%。     

IEEE 802.16系统中带有收益因子的动态呼叫接纳控制算法*

针对IEEE 80216 OFDMA系统,提出了一种考虑运营收益并且动态调整预留信道的呼叫接纳控制算法。该算法将一个连接申请潜在的运营收益值作为接入优先级函数的因子,从而提高系统的运营收益;其次,算法根据系统内资源的使用情况动态调整预留信道,在降低系统掉话率的同时,也尽可能地将呼叫阻塞率保持在较低水平。将所提算法与传统接纳控制方法进行了仿真比较,结果表明该算法能够有效控制系统掉话率和阻塞率,达到用户使用满意度要求,同时增大系统运营收益。     

抗时延敏感性跨层自适应资源分配方案*

为了对抗多用户OFDM系统中用户实时业务对时延的敏感性,提出一种利用Hopfield神经网络(HNN)算法的跨层自适应资源分配方案。该方案设置用户调度优先级时同时考虑物理层的信道状态信息,及媒体接入层的用户队列状态信息和等待时间等;采用HNN算法,最大化系统容量的同时降低了平均时延和丢包率。仿真结果表明,相比于传统资源分配方案,该方案可以有效保障用户的服务质量,并提高了系统的整体性能。