Adaptive Head-to-Tail: Active Queue Management based on implicit congestion signals

Active Queue Management is a convenient way to administer the network load without increasing the complexity of end-user protocols. Current AQM techniques work in two ways; the router either drops some of its packets with a given probability or creates different queues with corresponding priorities. Head-to-Tail introduces a novel AQM approach: the packet rearrange scheme. Instead of dropping, HtT rearranges packets, moving them from the head of the queue to its tail. The additional queuing delay triggers a sending rate decrease and congestion events can be avoided. The HtT scheme avoids explicit packet drops and extensive retransmission delays. In this work, we detail the HtT algorithm and demonstrate when and how it outperforms current AQM implementations. We also approach analytically its impact on packet delay and conduct extensive simulations. Our experiments show that HtT achieves better results than Droptail and RED methods in terms of retransmitted packets and Goodput.     

A collision avoidance scheme for the synchronized broadcast packets in a multi-AP Wi-Fi broadcasting system

A Wi-Fi broadcasting system is a kind of Mobile-TV system that transmits multimedia content over Wi-Fi networks. The specialty of the system is that it takes advantage of broadcast packets for streaming to be scalable to the number of users. However, the loss rate of broadcast packets is much higher than that of unicast ones because MAC layer retransmission is not applied on broadcast packets. To recover lost packets, a packet level Forward Error Correction (FEC) scheme is usually used in Wi-Fi broadcasting systems. But it introduces additional transmission overhead, which is usually proportional to the packet loss rate. So it is important to reduce the packet loss rate to build an efficient and reliable Wi-Fi broadcasting system. While past studies have considered only single-AP systems, our study focuses on a multi-AP system which is designed to cover a much larger area. We found a specific packet collision problem that increases packet loss rate significantly in a multi-AP system. It is caused by the simultaneous arrival and transmission of a broadcast packet at and by APs. We identify two scenarios of the collision that depend on the channel state at the time of packet arrival. We propose two collision avoidance methods to handle these scenarios: Broadcast Packet Scheduling Method (BPSM) and Adaptive Contention Window-Sizing Method(ACWSM). We implement both methods in our multi-AP Wi-Fi broadcasting system and verify their effectiveness through experiments.     

Proportional loss rate differentiation in a FIFO queue

The relative differentiated service model is one of several models proposed for service differentiation in networks [IEEE Network Sept/Oct (1999) 26]. In this model, an assurance is given that ‘higher classes will be better, or at least no worse than lower classes.’ This paper describes a relative loss rate differentiation scheme based on RED. The scheme is used for differentially dropping packets in a FIFO queue during times of congestion. The main idea is, if packet losses are unavoidable in the FIFO queuing system, then they should be distributed among the different service classes in the queue in inverse proportion to the service price or weight assigned to each class. The simulation studies using TCP traffic show that the scheme is very effective in ensuring relative loss rate differentiation between service classes.     

Lateral error recovery for media streaming in application-level multicast

We consider media streaming using application-level multicast (ALM) where packet loss has to be recovered via retransmission in a timely manner. Since packets may be lost due to congestion, node failures, and join and leave dynamics, traditional "vertical" recovery approach where upstream nodes retransmit the lost packets is no longer effective. We therefore propose lateral error recovery (LER). In LER, hosts are divided into a number of planes, each of which forms an independent ALM tree. Since error correlation across planes is low, a node effectively recovers its error by "laterally" requesting retransmission from nearby nodes in other planes. We present analysis on the complexity and recovery delay on LER. Using Internet-like topologies, we show via simulations that LER is an effective error recovery mechanism. It achieves low overhead in terms of delivery delay (i.e., relative delay penalty) and physical link stress. As compared with traditional recovery schemes, LER attains much lower residual loss rate (i.e., loss rate after retransmission) under a certain deadline constraint. The performance can be substantially improved in the presence of some reliable proxies.     

无线网络中基于网络编码的重传机制

鉴于目前网络编码在单跳无线网络重传中的应用研究都是基于单发-多收场景的局限性,提出一种应用于多发-多收无线网络中的重传机制NCWRM。该重传机制中,网络节点既可以是发送方也可以是接收方,节点发送的数据包在直传和第一次重传中都失败后,第二次重传时广播发送由多个丢失数据包编码组合得到的编码包。多个接收方通过解码编码包得到丢失的数据包,从而实现一次重传获得多个丢失的数据包,可以有效提高重传效率。理论分析和仿真结果表明,NCWRM算法能够明显提高系统的饱和吞吐量,同时减小开销及丢包率。     

Performance of ARQ-protocols for HDX-transmission in hierarchical polling systems

Two different ARQ-protocols are analyzed for half-duplex transmission between a primary station and an arbitrary number of polled secondary stations. The protocols differ in the retransmission scheme after error detection of transmitted frames (packets): immediate retransmission or retransmission within the next polling cycle. Both cases are modeled by a multi-queue, single server system with feedback, batch arrival processes, general service (transmission and overhead) times, and cyclic interqueue discipline. Both queueing models are analyzed through an imbedded Markov chain on the basis of an independence assumption for the cycle times. The analysis yields explicit results for the Laplace-Stieltjes transforms and averages of the packet delay and interdeparture times as functions of the most important design parameters as number of stations, load, arrival clusters, packet length, overhead (control, change of transmission direction), error rate, and retransmission schemes. Some numerical results are given which show the influences of these parameters on the system performance.     

基于终端差异化与D2D网络的IDNC协作重传方案

为了满足终端用户的个性化需求并且降低D2D网络的传输时延,提出了一种基于终端差异化的立即可解网络编码（IDNC）协作重传方案。首先,针对PC-D2D网络存在的解码冲突以及传输冲突问题提出一种新的IDNC算法框架并且在此框架的基础上搜索极大独立集（MIS）,综合考虑数据包的接收情况、终端用户需求以及链路丢包率情况设计权重,衡量权重选取一次重传时延增量最小的并发协作重传终端以及数据包组合生成编码包;同时,考虑不需要数据包提供的未来解码机会,优化终端不需要的数据包,进一步降低传输时延。仿真结果表明,所提方案在满足终端个性化需求的同时能够有效地降低解码时延和完成时间。     

Multi-indicator Active Queue Management Method

A considerable number of applications are running over IP networks. This increased the contention on the network resource, which ultimately results in congestion. Active queue management (AQM) aims to reduce the serious consequences of network congestion in the router buffer and its negative effects on network performance. AQM methods implement different techniques in accordance with congestion indicators, such as queue length and average queue length. The performance of the network is evaluated using delay, loss, and throughput. The gap between congestion indicators and network performance measurements leads to the decline in network performance. In this study, delay and loss predictions are used as congestion indicators in a novel stochastic approach for AQM. The proposed method estimates the congestion in the router buffer and then uses the indicators to calculate the dropping probability, which is responsible for managing the router buffer. The experimental results, based on two sets of experiments, have shown that the proposed method outperformed the existing benchmark algorithms including RED, ERED and BLUE algorithms. For instance, in the first experiment, the proposed method resides in the third-place in terms of delay when compared to the benchmark algorithms. In addition, the proposed method outperformed the benchmark algorithms in terms of packet loss, packet dropping, and packet retransmission. Overall, the proposed method outperformed the benchmark algorithms because it preserves packet loss while maintaining reasonable queuing delay.     

一种改进的网络编码单播重传方案

针对现有无线单播重传存在有效性低、频谱效率低等问题,提出了一种改进的网络编码单播重传方案。该方案的基本原理:在重传阶段,发送端将首次传输有误的数据包与新的数据包分别以不同的速率编码后进行合并,然后再将对应的混合数据包发送到接收端;接收端将收到的重传数据包与首次传输的数据包进行软合并及译码处理,以恢复出所传输的原始数据包信息,同时被合并的2个数据包在重传过程中共享能量、空间和时间。仿真结果表明,相比于传统的数据包独立自动请求重传方案,该网络编码单播重传方案的误比特率更低,有效提高了无线单播链路的传输效率。     

Adaptive hybrid error correction model for video streaming over wireless networks

The Hybrid ARQ (HARQ) mechanism is the well-known error packet recovery solution composed of the Automation Repeat reQuest (ARQ) mechanism and the Forward Error Correction (FEC) mechanism. However, the HARQ mechanism neither retransmits the packet to the receiver in time when the packet cannot be recovered by the FEC scheme nor dynamically adjusts the number of FEC redundant packets according to network conditions. In this paper, the Adaptive Hybrid Error Correction Model (AHECM) is proposed to improve the HARQ mechanism. The AHECM can limit the packet retransmission delay to the most tolerable end-to-end delay. Besides, the AHECM can find the appropriate FEC parameter to avoid network congestion and reduce the number of FEC redundant packets by predicting the effective packet loss rate. Meanwhile, when the end-to-end delay requirement can be met, the AHECM will only retransmit the necessary number of redundant FEC packets to receiver in comparison with legacy HARQ mechanisms. Furthermore, the AHECM can use an Unequal Error Protection to protect important multimedia frames against channel errors of wireless networks. Besides, the AHECM uses the Markov model to estimate the burst bit error condition over wireless networks. The AHECM is evaluated by several metrics such as the effective packet loss rate, the error recovery efficiency, the decodable frame rate, and the peak signal to noise ratio to verify the efficiency in delivering video streaming over wireless networks.     

一种新颖的移动自组网灰洞攻击检测方案

移动自组网(mobile ad hoc networks,MANETs)是典型的分布式网络,没有集中式的管理节点,网络拓扑动态变化,而且网络带宽有限.移动自组网无网络基础设施的特点,使其易于受到各种拒绝服务攻击(denial of service,DoS).灰洞攻击是一种类型的拒绝服务攻击,攻击者在网络状态良好的情况下,首先以诚实的方式参与路由发现过程,然后以不被察觉的方式丢弃部分或全部转发数据包.首先介绍了相关工作、DSR算法、聚合签名算法和网络模型.然后基于聚合签名算法,给出了用于检测丢包节点的3个相关算法:证据产生算法、审查算法和诊断算法.证据产生算法用于节点产生转发证据;审查算法用于审查源路由节点;诊断算法用于确定丢包节点.最后分析了算法的效率.ns-2仿真结果表明,在移动速度中等的网络中,提出的算法可以检测出多数丢包节点,且路由包开销较低.舍弃含丢包节点的路由后,数据发送率有相应的改善.     

MANET中基于信号强度的TCP拥塞控制算法研究

针对MANET中较高比特误码率、路由故障等因素引起的大量数据丢包和重传提出跨层设计的TCP拥塞控制算法.算法根据接收到的信号强度预测路由断点并提前建立备份路由,拯救传输中的数据包,解决传统TCP的拥塞控制算法在MANET中存在的问题.实验结果表明,优化算法极大地提高了TCP流的平均吞吐量.     

抑制CMT中乱序程度的路径优化方案

为减轻并行多路径传输(CMT)中接收端的乱序程度,文中提出了一种新的路径优化方案。该方案以MPTCP协议为基础,在三维笛卡尔坐标系下建立基于路径往返时延、丢包率和带宽的路径评价模型,将二分法与改进的基于密度的聚类分析方法相融合,根据所需路径数在坐标系下选择出一组带宽大、丢包率小且时延差也小的路径作为活跃路径。仿真结果显示,本方案与基于权重的路径选择方案相比降低了接收端的乱序长度与重传数据包个数,从而可以提高吞吐量与数据包传输速率。     

MTCP: scalable TCP-like congestion control for reliable multicast

We present MTCP, a congestion control scheme for large-scale reliable multicast. Congestion control for reliable multicast is important, because of its wide applications in multimedia and collaborative computing, yet non-trivial, because of the potentially large number of receivers involved. Many schemes have been proposed to handle the recovery of lost packets in a scalable manner, but there is little work on the design and implementation of congestion control schemes for reliable multicast. We propose new techniques that can effectively handle instances of congestion occurring simultaneously at various parts of a multicast tree.Our protocol incorporates several novel features: (1) hierarchical congestion status reports that distribute the load of processing feedback from all receivers across the multicast group, (2) the relative time delay concept which overcomes the difficulty of estimating round-trip times in tree-based multicast environments, (3) window-based control that prevents the sender from transmitting faster than packets leave the bottleneck link on the multicast path through which the sender's traffic flows, (4) a retransmission window that regulates the flow of repair packets to prevent local recovery from causing congestion, and (5) a selective acknowledgment scheme that prevents independent (i.e., non-congestion-related) packet loss from reducing the sender's transmission rate. We have implemented MTCP both on UDP in SunOS 5.6 and on the simulator ns, and we have conducted extensive Internet experiments and simulation to test the scalability and inter-fairness properties of the protocol. The encouraging results we have obtained support our confidence that TCP-like congestion control for large-scale reliable multicast is within our grasp.     

多媒体组播中的RTP包丢失恢复方案

论文提出了一种在实时多媒体组播应用中的RTP包丢失恢复方案,该方案是基于重传机制的,利用“分层缓冲路由器”作为辅助,配合接收端的缓冲播放机制,使得RTP包的本地丢失恢复成为可能。同时,该方案还考虑在网络拥塞时对重传请求进行缓冲等待处理,这样就不会因为向上级缓冲路由器发送该请求而加重拥塞的程度,甚至造成该重传请求的最终丢失。     

多通道并行传输中接收缓存阻塞的缓解方法

针对多通道并行传输中的接收缓存阻塞问题,分析了引起接收缓存阻塞的原因,提出一种改进的缓解接收缓存阻塞的数据包调度方法,综合考虑通道的带宽、时延和丢包率,引入通道质量的评价函数,优化多通道之间的数据包调度,选择质量最好的通道进行传输,减少由于通道特性不同造成的接收端数据包乱序;提出一种改进的数据包重传策略,基于时延和丢包率选择能使数据包最快到达接收端的通道进行重传;提出一种根据通道的带宽-延迟积估算所需接收缓存大小的方法。仿真实验表明,所提出的调度方法和重传策略能够有效地减轻接收缓存阻塞,与CMT-SCTP相比具有更优的性能,所提出的缓存大小的估算方法也能够准确估算所需接收缓存的大小。     

一种改进的主动拥塞控制算法

在已有的主动网络拥塞控制机制(ACC)中,通过丢包的方式缓减拥塞。TCP拥塞控制机制起主导作用,但不能满足现有网络环境要求。该文提出一种改进的ACC算法,该算法将拥塞控制参数嵌入到每个主动包中,通过驻留在主动路由器中的拥塞控制代码,对参数进行相应修改,从而达到拥塞预防和控制的目的。仿真结果表明,该方法具有较好的网络带宽利用率和公平性,丢包概率接近0。     

NNRA-CAC: NARX Neural Network-based Rate Adjustment for Congestion Avoidance and Control in Wireless Sensor Networks

A wireless sensor network (WSN) is an application area that is valuable in various fields, such as healthcare monitoring, environmental monitoring, and so on. Application areas require WSNs with high throughput and low degree of packet loss. Due to congestion in the network, the throughput of the network is affected, which imposes the need for congestion control in the network. This article proposes a method, titled NARX Neural network-based Rate Adjustment (NNRA) for avoiding and controlling congestion in the network. Initially, congestion in the network is avoided by dropping packets and the NNRA is used to control congestion in the network when congestion is present. Performance analysis is carried out in terms of throughput, delay, size of the queue, packet loss, and the level of the congestion using two setups. The results of the proposed method are compared with the existing methods to prove the effectiveness of the proposed method. The proposed method attained a maximum throughput at a rate of 0.9585 and minimum values for delay, queue size, packet loss, and the congestion level.     

基于TCP友好速率控制和前向纠错的MPEG-2视频传输

针对Internet视频传输面临拥塞控制和数据包丢失的问题，结合TCP友好的速率控制算法和前向纠错机制建立视频传输的分层体系构架和控制策略。传输体系同时采用以GOP为基本分析单元的视频帧速率预测模型，实现根据网络丢包率的变化动态地优化配置前向纠错的冗余信息。实验证明，传输体系采用动态优化的前向纠错能实时地适应带宽的变化，有效地降低数据包丢失带来的影响，从而改善视频回放质量。