A cross-layer framework for optimal delay-margin,network lifetime and utility tradeoff in wireless visual sensor networks

For wireless multimedia sensor networks a distributed cross-layer framework is proposed, which not only achieves an optimal tradeoff between network lifetime and its utility but also provides end-to-end delay-margin. The delay-margin, defined as the gap between maximum end-to-end delay threshold and the actual end-to-end delay incurred by the network, is exploited by the application layer to achieve any desired level of delay quality-of-service. For optimal performance tradeoff an appropriate objective function for delay-margin is required, which is obtained by employing sensitivity analysis. Sensitivity analysis is performed by incorporating delay-margin in the end-to-end delay constraints while penalizing its price in the objective function. For distributed realization of proposed cross-layer framework, the optimal tradeoff problem is decomposed into network lifetime, utility and delay-margin subproblems coupled through dual variables. The numerical results for performance evaluation show that compromising network utility does not guarantee both lifetime and delay-margin improvement, simultaneously, for the set of operating points. Performance evaluation results also reveal that the fairness among different delay-margins, corresponding to different source–destination node pairs, can be improved by relaxing the end-to-end delay threshold.     

End-to-End Performance Improvement of the Wireless Channel for Video Transmission

Real time video transmission in wireless environment considers various parameters of wireless channel like information rate, error resiliency, security, end-to-end latency, quality of service etc. The available internet protocols are transmission control protocol and user datagram protocol (UDP). But most of the real-time applications uses UDP as their transport protocol. UDP is a fast protocol suitable for delay sensitive applications like video and audio transmission as it does not provide flow control or error recovery and does not require connection management. Due to the tremendous growth in wired and wireless real-time applications, some improvements should be made in the existing systems or protocols. Various techniques to improve end-to-end performance of system for real time video transmission over wireless channel are available in literature. Authors claim that the solution suggested in the paper provide more reliability in wireless video transmission. In the proposed solution, adaptive redundant packets are added in every block (or datagram) transmitted in order to achieve a desired recovery rate at the receiver. The suggested method dose not use any retransmission mechanism. The network simulator NS-2 is used to evaluate the method and the simulation results indicate that the proposed method can guarantee satisfied end-to-end performance by increased packet delivery ratio, reduced end-to-end delay and hence increased network throughput for video transmission in wireless network.     

Ad Hoc网络中一种高效的接纳控制算法

提出了一种保证Ad hoc网络中实时多媒体业务服务质量（QoS）的高效接纳控制方案。该机制采用跨层设计思想,以网络中每个节点MAC层感知的信息为基本依据,在新业务申请进入网络时在路由层发起接纳判决过程,在保证已有业务QoS不受损害的同时,新业务的QoS要求能够得到满足时才允许接入。该接纳过程具有以下特点:首先,它由目的节点发起反向逐跳进行,相对于以往由源节点发起的接纳控制过程可以节省一半的控制负载;其次,在每跳节点进行接纳判决的同时,对带宽进行暂时预留以防止过度接纳和过度预留情况的发生。分析和仿真证明了该机制能很好的保证实时业务的吞吐量,端到端延迟和延迟抖动等QoS参数,并且跟现有机制相比具有控制负载小,带宽利用率高的特点。      

Buffered fixed routing: a routing protocol for real-time transportin grid networks

We propose a new routing protocol called buffered fixed routing (BFR) for real-time applications on grid networks. While previous routing protocols for grid networks have been designed to improve network throughput, the BFR scheme is proposed to guarantee the end-to-end packet delay and sequencing without loss by using finite buffers at each node. Thus the proposed scheme can satisfy quality-of-service (QoS) requirements of real-time applications. The BFR scheme uses the token on the row ring to provide QoS guarantees. The performance of the BFR scheme is analyzed by using the Geom/Geom/1 queueing system under uniform traffic. In the simulation, the BFR scheme shows the zero-loss, high-throughput performance with the minimum delay variation compared to other routing protocols such as store and forward routing, deflection routing and vertical routing. In addition, it has shown the smallest average delay at intermediate and heavy loads     

一种面向实时业务的Ad Hoc网络MAC协议

为降低AdHoe网络中实时业务的端到端时延,提出了基于802．11DCF的改进协议。协议采用3种机制降低实时业务的时延：面向路径的连续转发机制将RTS中的转发信息携带在ACK中发送,给实时业务提供较高的接入优先级;标签交换机制使得中间节点可以在MAC层获取转发信息,加快了实时业务数据包的转发速度;重传控制机制减少了无效传输的超时数据包。仿真结果表明,在重负载条件下,改进协议中实时业务的时延比802．11DCF有大幅度的下降,网络吞吐量也有所提高。     

A framework for deterministic delay guarantee in OBS networks

In OBS networks, the delay of control packets in the switch control unit (SCU) of core nodes influences burst loss performance in the optical switching and should be constrained. Furthermore, the end-to-end (E2E) delay requirements of premium services need queueing delay guarantee in network nodes throughout the transmission path. For this purpose, a framework for deterministic delay guarantee is proposed in this article. It incorporates the deterministic delay model in the ingress edge node as well as in the SCUs of core nodes. On this basis, the configuration of the assembler and the offset time is addressed by means of an optimization problem under the delay constraints. Scenario studies are carried out with reference to realistic transport network topologies. Compared to statistical delay models in the literature, the deterministic model has advantages in rendering robust absolute delay guarantee for individual FEC flows, which is especially appreciated in the provisioning of premium services. By performance evaluation in comparison with the statistical models, it is shown that the adopted deterministic delay models lead to practical delay bounds in a magnitude that is close to the delay estimations by stochastic analysis.     

Measurement-Based Admission Control at Edge Routers

It is very important to allocate and manage resources for multimedia traffic flows with real-time performance requirements in order to guarantee quality of service (QoS). In this paper, we develop a scalable architecture and an algorithm for admission control of real-time flows. Since individual management of each traffic flow on each transit router can cause a fundamental scalability problem in both data and control planes, we consider that each flow is classified at the ingress router and data traffic is aggregated according to the class inside the core network as in a DiffServ framework. In our approach, admission decision is made for each flow at the edge (ingress) routers, but it is scalable because per-flow states are not maintained and the admission algorithm is simple. In the proposed admission control scheme, an admissible bandwidth, which is defined as the maximum rate of a flow that can be accommodated additionally while satisfying the delay performance requirements for both existing and new flows, is calculated based on the available bandwidth measured by edge routers. The admissible bandwidth is a threshold for admission control, and thus, it is very important to accurately estimate the admissible bandwidth. The performance of the proposed algorithm is evaluated by taking a set of simulation experiments using bursty traffic flows.     

Earliest Deadline First Scheduling with Active Buffer Management for Real-Time Traffic in the Internet

We studied the problem of QoS guarantee for differentiated services. A two-level hierarchical scheduling framework was deployed to separate QoS metrics. Due to its desirable property of minimizing the maximum packet lateness, the Earliest Deadline First (EDF) scheduling was adopted to provide the in-class scheduling for the time-sensitive traffic. We employed an EDF scheduler combined with an active buffer management scheme (CHOKe) to improve the fairness of resource allocation and to maintain a good delay performance for real-time applications. Simulation results showed that the proposed scheme can achieve a better delay performance and make a more fair bandwidth allocation between the real-time TCP and UDP connections than the First Come First Served (FCFS) scheduling with the drop-tail buffer management which is commonly deployed in traditional IP routers.     

准静止状态下的高空平台实时业务信道分配

受平台准静止状态的影响,高空平台（ HAPS)通信网络内存在大量的切换呼叫,且业务量动态变化。 HAPS网络可传输多种业务,其中实时业务在切换过程中具有较高的时延要求。通过为切换呼叫预留信道可降低平台不稳定对服务质量（ QoS)造成的影响。在基于服务优先级的多业务信道分配算法基础上,重点对实时业务的信道分配算法进行改进,提出了一种基于概率的预留信道借用策略。该算法可根据网络内业务量的实时统计数据控制新呼叫业务的准入。仿真结果表明：与固定预留信道算法和门限预留信道算法相比,该算法能够适应网络内业务量的动态变化,在保证切换呼叫掉线率满足期望值的条件下提升系统的整体性能,降低平台不稳定造成的性能损失。     

QUORUM—Quality of Service in Wireless Mesh Networks

Wireless mesh networks (WMNs) can provide seamless broadband connectivity to network users with low setup and maintenance costs. To support next-generation applications with real-time requirements, however, these networks must provide improved quality of service guarantees. Current mesh protocols use techniques that fail to accurately predict the performance of end-to-end paths, and do not optimize performance based on knowledge of mesh network structures. In this paper, we propose QUORUM, a routing protocol optimized for WMNs that provides accurate QoS properties by correctly predicting delay and loss characteristics of data traffic. QUORUM integrates a novel end-to-end packet delay estimation mechanism with stability-aware routing policies, allowing it to more accurately follow QoS requirements while minimizing misbehavior of selfish nodes.     

光突发交换网络中Jitter-EDF调度算法的研究

提出了一种应用于OBS网络中的Jitter-EDF调度算法来减小时延抖动对实时性突发QoS的影响,设计了基于该算法的OBS网络中核心路由器的主要结构,讨论了此算法实现的可能性,并推导出端到端时延的上下限和时延抖动的上限,最后结合具体网络拓扑进行了仿真和分析。结果表明,Jitter-EDF调度算法能够将时延抖动的幅度平均降低到30%～40％,有效保证了实时性应用对时延抖动的要求。     

Group priority scheduling

We present an end-to-end delay guarantee theorem for a class of guaranteed deadline (GD) servers. The theorem can be instantiated to obtain end-to-end delay bounds for a variety of source control mechanisms and GD servers. We then propose the idea of group priority, and specialize the theorem to a subclass of GD servers that use group priority in packet scheduling. With the use of group priority, the work of packet schedulers can be substantially reduced. We work out a detailed example, for the class of burst scheduling networks, to illustrate how group sizes can be designed such that the worst case end-to-end delay of application data units in a real-time flow is unaffected by the use of group priority. Group priority also can be used in packet schedulers that provide integrated services (best effort as well as real-time services) to achieve statistical performance gains, which we illustrate with empirical results from simulation experiments     

Efficient topology construction and routing for IEEE 802.15.4m-based smart grid networks

IEEE 802.15.4m TVWS Multi-Channel Tree PAN (TMCTP) standard that uses the vacant TV frequency of a region is the key to provide a flexible, scalable and cost-effective AMI smart grid networks. However, the performance of the IEEE 802.15.4m based AMI network can suffer from network interruption, varying transmission reliability and energy consumption problems due to the excessive number of channels and periodic channel scanning. To resolve these issues, we presented an enhanced IEEE 802.15.4m TMCTP called TVWS Orphan channel scanning with Multi-Channel Tree PAN Routing (TOMTPR). The proposed TOMTPR framework includes pilot-channel based Multi-Channel beaconing and interleaving-based TVWS orphan channel scanning. Furthermore, a capacity-aware routing tree is constructed during the neighbor discovery procedure. The proposed protocol suite is designed to provide compatibility with the IEEE 802.15.4 family standards with lower architecture complexity. The simulation results in presence of realistic AMI traffic and AMI network model show that TOMTPR can not only satisfy delay requirements of the AMI traffic, but also outperforms IEEE 802.15.4m TMCTP with IEEE 802.15.5 layer 2 mesh routing in terms of topology construction delay, end-to-end transmission reliability, and energy efficiency.     

适于低轨卫星IP网络的特定源组播算法

为了解决低轨卫星IP网络中现有特定源组播算法的信道资源浪费问题,本文提出了一套新的特定源组播算法,即基于核心群的特定源组播算法(CSSM)和加权的CSSM算法(w-CSSM).CSSM算法以源节点作为初始核心群,通过核心群和剩余组成员的最短路径方法逐步扩展直至整棵组播树构建完成,所得的树代价最小,从而大大提高了网络的带宽利用率和传输效率.在w-CSSM算法中,加权因子可以自适应调整以适度增大树代价、降低端到端传播时延,以支持某些有严格端到端时延要求的实时组播业务.通过与低轨卫星IP网络中典型特定源组播算法MRA的仿真比较,可以看出CSSM和w-CSSM算法的树代价性能比MRA有较大改善,不过端到端传播时延略高.     

Fair resource allocation with guaranteed statistical QoS for multimedia traffic in wideband CDMA cellular network

A dynamic fair resource allocation scheme is proposed to efficiently support real-time and non-real-time multimedia traffic with guaranteed statistical quality of service (QoS) in the uplink of a wideband code-division multiple access (CDMA) cellular network. The scheme uses the generalized processor sharing (GPS) fair service discipline to allocate uplink channel-resources, taking into account the characteristics of channel fading and intercell interference. In specific, the resource allocated to each traffic flow is proportional to an assigned weighting factor. For real-time traffic, the assigned weighting factor is a constant in order to guarantee the traffic statistical delay bound requirement; for non-real-time traffic, the assigned weighting factor can be adjusted dynamically according to fading, channel states and the traffic statistical fairness bound requirement. Compared with the conventional static-weight scheme, the proposed dynamic-weight scheme achieves capacity gain. A flexible trade-off between the GPS fairness and efficient resource utilization can also be achieved. Analysis and simulation results demonstrate that the proposed scheme enhances radio resource utilization and guarantees statistical QoS under different fairness bound requirements.     

一种基于IEEE802.11支持QoS自适应调度器模型和算法

这篇文章提出了能够有效支持QoS的IEEE802．11自适应调度器模型。和以往的支持QoS的调度器模型相比，这里根据在主控接入点(MAP：Master Access Point)和用户终端(UT：User Termination)的各自延时需求对满足不同的传输机制的分组流进行自适应调度。这种机制不仅完全兼容当前支持QoS的IEEE802．11MAC协议标准，而且能够有效减少由于实时传输带来的分组延时，增加满足不同信道负载和带宽要求的数据流的吞吐率。实时测试得到的数据基本验证了这个要求。     

保障流媒体时间结构的网络时延分析

流媒体同步对端到端时延和时延抖动提出了确定的要求,而终端抖动缓存一方面能消除时延抖动的影响,一方面却增加了端到端时延,流媒体同步保障对网络时延的要求不明确。论文从概率保障流媒体同步的角度,确定了保障流媒体同步的抖动缓存容量范围,提出了流媒体同步网络保障的充分条件,针对基于Internet VoIP(Voice over IP)业务的实际网络测试结果,给出了应用流媒体同步网络保障充分条件进行同步保障评价的应用实例并验证了其正确性。     

基于遗传算法的时延受限代价最小组播路由选择方法

目前多媒体业务的涌现对网络提出了更高的要求。希望既能满足实时性，又能够高效地利用网络资源。本文提出了一种基于遗传算法的组播路由选择方法，该方法在满足时延限制条件的基础上寻找代价最小的组播树。实验表明，该算法收敛速度快，可靠性高，能够满足多媒体网络对实时性的要求。     

复杂对抗下军事通信系统的随机延迟分析

随着信息化技术的发展,现代军事指挥越来越依靠综合化、集群化的军事通信系统。在战争局势复杂多变,尤其是敌方干扰严重、战场环境复杂等对抗性强的环境下,如何确保有效军事情报及时地准确地传递给指挥者显得尤为重要。针对战场有效数据产生的随机性和传输过程的不稳定性,提出了一种基于随机网络演算的端到端延迟分析方法。由此获得的端到端延迟上界,不仅为有效信息的传输提供了实时性能保证,而且能充分的利用系统资源。这对于军事通信系统的设计以及传输系统的性能改善有着非常重要的指导意义。     

基于无线Mesh网络的分布式协调功能的研究

随着无线mesh网络的迅速发展和实时多媒体业务的日益增长,要求无线网络不仅能提供简单的连通性,还需要提供非常有效的QoS保证机制,针对这一问题,以时分多址/时分双工(TDMA/TDD)技术为基础,提出一种基于无线mesh网络的分布式协调功能,信道存取与数据传输发生在不同的时隙中,能够为端到端数据流公平的分配带宽,并且保证了实时服务传输的QoS.仿真及分析表明该协调功能可提高系统吞吐量并降低端到端延迟,有着更好的QoS性能.