Rate-adaptive video streaming through packet dispersion feedback

The anticipated growth of IPTV makes selection of suitable congestion controllers for video-stream traffic of vital concern. Measurements of packet dispersion at the receiver provide a graded way of estimating congestion, which is particularly suited to video as it does not rely on packet loss. A closed-loop congestion controller, which dynamically adapts the bitstream output of a transcoder or video encoder to a rate less likely to lead to packet loss, is presented. The video congestion controller is based on fuzzy logic with packet dispersion and its rate of change forming the inputs. Compared with TCP emulators such as TCP-friendly rate control (TFRC) and rate adaptation protocol (RAP), which rely on packet loss for real-time congestion control, the fuzzy-logic trained system?s sending rate is significantly smoother when multiple video-bearing sources share a tight link. Using a packet dispersion method similarly results in a fairer allocation of bandwidth than TFRC and RAP. These gains for video traffic are primarily because of better estimation of network congestion through packet dispersion but also result from accurate interpretation by the fuzzy-logic controller.     

TCPW BR: A Wireless Congestion Control Scheme Base on RTT

The wireless network is limited by the transmission medium, and the transmission process is subject to large interference and jitter. This jitter can cause sporadic loss and is mistaken for congestion by the congestion control mechanism. The TCP Westwood protocol (referred to as TCPW) is such that it cannot distinguish between congestion loss and wireless jitter loss, which makes the congestion mechanism too sensitive and reduces bandwidth utilization. Based on this, the TCPW protocol is modified based on the estimate of the Round-Trip Time (referred to as RTT) value-called TCPW BR. The algorithm uses the measured smooth RTT value and divides the congestion level according to the weighted average idea to determine congestion loss and wireless jitter loss. The simulation results show that the TCPW BR algorithm enhances the wireless network’s ability to judge congestion and random errors.     

TCP Yolanda:无线和有线混合网络中的TCP改进算法

在TCP Westwood基础上提出了一种适合无线有线混合网络的TCP改进算法——TCP Yolanda,该算法以瓶颈链路缓冲区长度测量值为参数,采用分两个子阶段的方式改进拥塞避免阶段以更有效进行传输控制,使传输过程能更长时间保持在稳定的高速率状态,并能在误码率较高的无线网络环境下快速恢复拥塞窗口大小,另外在传输发生错误时能判断丢包原因并进行区别处理。基于ns2的仿真实验表明,改进后的TCP算法在网络传输吞吐率、传输稳定性等方面的性能有较大提升,并且保持较好的连接公平性和友好性。     

On window control algorithm for TCP efficiency over wireless networks with high bandwidth-delay product

The existing congestion control algorithms such as TCP Reno are not suitable for efficient utilisation of networks with a high bandwidth-delay product because it takes a very long time to achieve the full link utilisation. Furthermore, it is difficult to guarantee the fairness among the TCP connections with different round-trip times. To overcome these problems, a new window control algorithm using the buffer state and variable gains is proposed. The simulation results show that our algorithm improves the performance of TCP in the wireless networks     

面向多媒体流的主动分层多播拥塞控制方案

针对现有分层多播拥塞控制方案存在较大的拥塞响应延时,吞吐率抖动剧烈,和不满足TCP友好的问题,给出了一个主动分层多播服务模型,通过引入主动标记分层、可用带宽主动测量以及优先级分层过滤机制,提出了一种新的面向多媒体流的自适应主动分层多播拥塞控制方案AALM。仿真结果表明,该方案能够有效地改进分层多播拥塞控制的性能,具有较快的拥塞响应速度、较好的稳定性和TCP友好特性。     

WARD: A deterministic fluid model

Queue management, bandwidth share and congestion control are very important to both the robustness and fairness of the Internet. A new TCP-friendly router-based active queue management scheme, called WARD, approximates the fair queueing policy. WARD is a simple packet dropping algorithm with a random mechanism and discriminates against the flows which submit more packets per second than is allowed by their fair share. By doing this, it not only protects transmission control protocol (TCP) connections from user datagram protocol (UDP) flows, but also solves the problem of competing bandwidth among different TCP versions, such as TCP Vegas and TCP Reno. Furthermore, it is stateless and easy to implement, so WARD controls unresponsive or misbehaving flows with a minimum overhead. In this article, we present a deterministic fluid model of TCP/WARD system, and explain the UDP throughput behaviour with WARD. Also, we prove that, provided the number of TCP flows is large, the UDP bandwidth share peaks at (2e)^-1 = 0.184 when UDP input rate is slightly larger than link capacity and drops to zero as UDP input rate tends to infinity.     

Robust end-to-end loss differentiation scheme for transport control protocol over wired/wireless networks

The authors propose a robust end-to-end loss differentiation scheme to identify the packet losses because of congestion for transport control protocol (TCP) connections over wired/wireless networks. The authors use the measured round trip time (RTT) values to determine whether the cause of packet loss is because of the congestion over wired path or regular bit errors over wireless paths. The classification should be as accurate as possible to achieve high throughput and maximum fairness for the TCP connections sharing the wired/wireless paths. The accuracies of previous schemes in the literature depends on varying network parameters such as RTT, buffer size, amount of cross traffic, wireless loss rate and congestion loss rate. The proposed scheme is robust in that the accuracy remains rather stable under varying network parameters. The basic idea behind the scheme is to set the threshold for the classification to be a function of the minimum RTT and the current sample RTT, so that it may automatically adapt itself to the current congestion level. When the congestion level of the path is estimated to be low, the threshold for a packet loss to be classified as a congestion loss is increased. This avoids unnecessary halving of the congestion window on packet loss because of the regular bit errors over the wireless path and hence improves the TCP throughput. When the congestion level of the path is estimated to be high, the threshold for a packet loss to be classified as the congestion loss not to miss any congestion loss is decreased and hence improves the TCP fairness. In ns 2 simulations, the proposed scheme correctly classifies the congestion losses under varying network parameters whereas the previous schemes show some dependency on subsets of parameters.     

一种能处理无线链路拥塞的TCP/ARQ改进算法

采用链路层自动请求重传（ARQ）方法来对分组差错进行恢复，并对传统的TCP／ARQ算法加以改进，加入了在无线链路拥塞情况下的自适应拥塞避免算法。仿真结果表明，这种能处理无线链路拥塞的TCP／ARQ算法对于提高无线网络的吞吐量非常有效。     

An Adjust Duty Cycle Method for Optimized Congestion Avoidance and Reducing Delay for WSNs

With the expansion of the application range and network scale of wireless sensor networks in recent years, WSNs often generate data surges and delay queues during the transmission process, causing network paralysis, even resulting in local or global congestion. In this paper, a dynamically Adjusted Duty Cycle for Optimized Congestion based on a real-time Queue Length (ADCOC) scheme is proposed. In order to improve the resource utilization rate of network nodes, we carried out optimization analysis based on the theory and applied it to the adjustment of the node’s duty cycle strategy. Using this strategy to ensure that the network lifetime remains the same, can minimize system delay and maximize energy efficiency. Firstly, the problems of the existing RED algorithm are analyzed. We introduce the improved SIG-RED algorithm into the ADCOC mechanism. As the data traffic changes, the RED protocol cannot automatically adjust the duty cycle. A scheduler is added to the buffer area manager, referring to a weighted index of network congestion, which can quickly determine the status of network congestion. The value of the weighting coefficient W is adjusted by the Bayesian method. The scheduler preferably transmits severely urgent data, alleviating the memory load. Then we combined improved data fusion technology and information gain methods to adjust the duty cycle dynamically. By simulating the algorithm, it shows that it has faster convergence speed and smaller queue jitter. Finally, we combine the adjusted congestion weight and the duty cycle growth value to adjust the data processing rate capability in the real-time network by dynamically adjusting it to adapt to bursts of data streams. Thus, the frequency of congestion is reduced to ensure that the system has higher processing efficiency and good adaptability.     

Enhanced token bucket policer using reduced fair queuing for Ethernet access networks

An enhanced traffic policer capable of adaptive rate control to provide fair distribution of available network bandwidth is proposed. The policer also performs maximum rate control, limiting flow's use of network resources within a pre-defined bound. Since the enhanced policer is based on a token bucket policer, a flow is guaranteed to receive service at its reserved rate. The proposed policer partially employs a reduced fair queuing algorithm, which is designed for adaptive rate control based on the perceived traffic congestion level. The adaptive rate control requires measurement of the congestion level, which is approximated through examination of local buffer usage. As a result, the allocated service rate increase provided to the flows by the policer is inversely proportional to the network congestion level. In addition, far-travelling flows in multi-hop networks receive the same service rate enhancements as short-travelling flows. The proposed scheme is useful in Ethernet networks, especially access networks, where quality of service is not well organised.     

一种融合MAC层拥塞通告的混合网络TCP协议

在研究无线网络媒体接入控制(MAC)层拥塞测度的基础上,提出了一种跨层的显式拥塞通告(ECN)机制,即:当数据包中记录的请求发送(RTS)次数超过给定阈值时,通过ECN向传输控制协议(TCP)源端发送拥塞通告,从而启动TCP拥塞控制.这种跨层设计是对有线网络中基于主动队列管理(AQM)的拥塞控制的有效补充,由此可以得到一种与已有的协议无缝连接的混合网络TCP模型.通过在网络模拟器NS2中构造多流无线局域网和多跳无线/有线混合网络,对所提出的方法进行了仿真,实验结果说明该方法能够提高混合网络的性能,并且具备良好的扩展性.     

基于ECN带预测验证的拥塞控制算法

提出了一种新的拥塞控制源算法FAV(Forecast and Verify),这是一种在接收方测量分组到达的延迟来预测网络中是否将出现拥塞的算法,并结合ECN来验证预测的准确性。通过和TCP Reno、TCP Vegas算法进行比较实验,表明FAV算法能够有效地减小丢失率和链路延迟,同时保持较高的链路利用率,从而使网络中的拥塞得到较好控制。     

多时滞无线通信网络的功率和速率最优跟踪

研究了无线通信网络的功率和速率控制这一热点问题。针对传统的无线通信网络模型并没有同时考虑状态时滞和输入时滞,已有的功率和速率控制算法也不能保证系统具有最优输出跟踪性能的情况,重新建立了无线通信网络的模型,在建模过程中考虑了多时滞,得到了含有不同的状态时滞和输入时滞的功率和速率控制系统模型,在此基础上,研究了一般形式的多时滞模型的时滞补偿算法,并将该算法应用在提出的多时滞模型中,给出了无线通信网络功率和速率的最优跟踪控制律。经过计算机仿真验证,提出的最优控制律具有良好的控制性能。     

An interference and latency aware uplink scheduling mechanism in IEEE 802.16j networks

IEEE 802.16j spreads out the coverage of WiMAX networks and strengthens wireless signal transmission using relay technology. To take advantage of relaying in IEEE 802.16j networks, an efficient scheduling schedule with quality of service (QoS) provision for multiple link transmissions is necessary, especially when link interference exists. In this paper, we propose an uplink scheduling mechanism in the transparent mode of IEEE 802.16j, which enables multiple devices to transmit without interference. The maximum latency of each connection has been considered in order to optimize the violation and transmission rate. An interference detection task is first carried out, and then a resource allocation algorithm and a dynamic frame adjustment method are developed. Two simulation experiments were conducted with different interference levels. The results demonstrate that under a fixed QoS type of connection, when the total number of connections goes up to 360 and 420 and the maximum latency violation rate approaches 20%, the average uplink transmission rate of the proposed mechanism can achieve 6.67 and 7.92 Mbps, which apparently outperform regular relay scheduling schemes with rate of 4 and 3.91 Mbps, respectively.     

一种基于缓存交互的命名数据网络拥塞控制算法

研究了命名数据网络(NDN)的拥塞控制。为了解决突发流量问题和提高吞吐量及网络资源利用率,考虑了路由器缓冲区大小与拥塞控制机制的相互影响以及NDN内部署缓存这一重要特性,提出了一种基于缓存交互的NDN拥塞控制算法。该算法通过利用NDN中的路由器缓存,在逻辑上动态扩充缓冲区大小并控制Data包的发送速率,同时与现有的NDN拥塞控制算法相结合,动态调整Interest包发送速率阈值,以平滑突发流量,缓解网络拥塞。基于ndn SIM的仿真实验结果表明,该算法能有效提高NDN的传输效率、吞吐量和网络资源利用率。     

基于部分可观察马尔可夫决策过程的受控无线网络系统动态资源分配

研究了受控无线网络的动态资源分配。针对传统无线通信传输模型的局限性随着无线通信系统架构的发展日益凸显的问题,提出了一种引入反馈控制策略的受控无线网络模型。该模型结合部分可观察马尔可夫决策过程(POMDP),将用户接收功率与数据传输误码率作为反馈观测对象,对通信小区内基站天线开启数与用户接入数进行动态资源最优匹配。仿真结果表明,这种方法能够有效提升系统传输能效性与可靠性,降低传输误码率,改善系统资源动态匹配控制性能。     

基于广播中继的自组网组播路由协议改进(ODMRP-MPR)

提出一种对ODMRP协议的改进,即ODMRP-MPR路由协议。该协议继承了ODMRP协议对拓扑频繁变化的良好适应性,并采用广播中继机制MPR(Multipoint Relay),优化了组播转发网格并减少了协议开销,提高了协议对网络规模的可扩展性和吞吐量,并且提供拥塞控制,有效地解决单向链路问题,增强了协议的健壮性。通过对ODMRP-MPR和ODMRP协议的模拟仿真实验比较,验证了ODMRP-MPR协议的改进效果。     

Optimal resource allocation for multicast sessions in multi-hop wireless networks

In this paper, we extend recent results on fair and stable resource allocation in wireless networks to include multicast sessions, in particular multi-rate multicast. The solution for multi-rate multicast is based on scheduling virtual (shadow) 'traffic' that 'moves' in reverse direction from destinations to sources. This shadow scheduling algorithm can also be used to control delays in wireless networks.     

宽带CDMA无线多媒体接入系统的研究

提出了可变传输速率的宽带CDMA扩频调制方法、非平衡功率控制算法以及自适应接入速率的无线接入方法。基于上述技术,给出了由基站和多媒体终端组成的实验系统参数。该系统实现了舆速率分别为8kbp和144skbps的语音和数据通信。     

Congestion-aware proactive vertical handoff algorithm in heterogeneous wireless networks

In heterogeneous wireless networks, when a mobile host/handset (MH) with multiple wireless interfaces changes its location or requires a certain network service, the MH will require a switch between different wireless networks (namely vertical handoff). A congestion-aware proactive vertical handoff algorithm is proposed, which uses a data pre-deployment technology to realise soft handoff between cellular interface and ad hoc interface. Here, the vertical handoff algorithm is implemented in an experimental heterogeneous network structure called converged ad hoc and cellular network, which is an ad hoc overlay system considering the balancing of the traffic between adjacent cellular cells. By evaluations, it is shown that the proposed algorithm can realise low handoff delay and low packet losses, and help to ease congestion issue existing in the heterogeneous networks.