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。     

融合IP和HFC的多通道协同传输系统的设计和优化

分析了IP双向交互通道和HFC单向广播通道的特性,提出一种在有线电视网上实现数据可靠传输的优化方法。该方法融合IP网络双向交互通道和有线电视网单向通道构建了多通道协同传输系统,使用有线电视网单向通道和IP双向通道协同下发数据,使用IP双向通道交互信令,充分利用有线电视网高带宽的优势实现非对称的宽带数据通信;基于有线电视网单向保序的特性,优化了数据包的重传策略,使得丢失的数据包尽快地重传,提高了数据传输的速率。实验结果表明,所提出的优化方法能够有效地缓解接收缓存阻塞,提升了传输系统的吞吐量。     

并行多路传输中数据调度算法的研究

针对传统并行多路传输中数据调度算法存在的问题,基于MPTCP协议,提出了带宽预测和前向时延的数据调度算法(data-scheduling algorithm using bandwidth estimation and forward trip-time,DA-BEFT)。该算法充分考虑子流间传输时延差较大的影响,结合性能好的重传选路策略,减轻接收端因数据乱序导致的缓存阻塞,提高整个连接吞吐量。通过仿真实验验证了DA-BEFT在子流时延差变化时能够提高带宽利用率,提高网络的吞吐量。     

多路并行传输中数据调度算法的优化

针对异构无线网络环境中,基于流控制传输协议（SCTP）的多路并行传输协议（CMT-SCTP）存在接收缓存阻塞和路径负载失衡等问题,提出一种改进的轮询数据调度算法。该算法根据每条路径上的发送队列信息和拥塞状况对网络状况进行估计,并按照各路径上的网络状况分配相应的传输任务量,缩短数据包在接收端缓冲区的平均排队时延,减少接收端乱序数据包的数量。仿真结果表明,改进的轮询数据调度算法能有效提升CMT-SCTP在异构无线网络环境中的传输效率,有效缓解接收缓存的阻塞,且对不同的网络场景具有很好的适应性。     

基于丢包率预测的视频传输纠错算法

为解决视频图像在互联网中进行传输时,其质量易受网络丢包率、时延等因素的影响而显著降低的问题,提出了一种基于丢包率预测的视频传输纠错算法。该算法采用隐马尔可夫模型预测网络丢包率,根据丢包率的大小自适应地选择FEC或ARQ对视频图像进行纠错操作。当预测出的丢包率较高时,为避免FEC算法在丢包率较高时降低带宽利用率,采用选择性ARQ算法恢复丢失的视频数据包,并通过限制其重传次数使视频传输的实时性得到了保证;当预测出的丢包率较低时,则采用优化了RS冗余值的FEC算法进行纠错操作。在OPNET modeler中进行的仿真实验表明,与HARQ算法相比,使用该纠错算法,视频图像的PSNR的平均值提高了1.6 dB,平均时延减少了0.24 s左右。该算法不但降低了视频传输的平均时延和丢包率,而且提高了接收端视频图像的重建质量,具有复杂度低、实现简单的特点。     

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。     

WSN中基于网络编码的最小广播重传算法

为了减少在无线传感网络中数据的重传次数,提高无线传感网的数据传输效率,提高服务质量,文中提出了一种网络编码的广播算法（WMBR）。在该算法中,依据网络接收节点的丢包情况,创建生成丢包的哈希表,选择并生成高效的重传数据包,然后对数据包再进行二次编码,通过这种方法有效地提高了重传效率。经过软件的仿真,结果表明：相比于普通重传方法及现有的算法而言,这种方法能够有效地降低数据包的重传次数,提高了无线传感网中数据通信的效率。     

结合视频运动特性的数据包调度策略研究

为了提高传输效率和视频播出质量,提出一种结合视频运动特性数据包调度策略.在率失真模型和人眼视觉感知模型的基础上,将人眼对不同视频内容运动变化的敏感度作为数据包重要性的分析依据之一,在带宽和时延约束条件下,使接收视频质量最优.NS2仿真结果表明,该调度策略能够提高视频播出质量,对于运动变化剧烈的复杂视频同样效果明显.     

基于业务安全策略需求的路由模型研究

首先对业务进行分类,不同的业务对网络不同的要求使其具有不同的QoS参数约束.然后研究并提出了基于智能业务识别的QoS路由模型和路由结构,根据动态配置的安全/QoS策略,在业务识别的基础上,标志数据包,根据DiffServ代码点DSCP值选择合适的路由算法.并针对带宽-时延-时延抖动-丢包率限制路由提出了一种改进的启发式路由算法,将丢包率转化为可加性条件,并把带宽限制作为剪枝条件,最后通过实验证明了其可行性.     

基于大数据调度的OMS配网一体化调控算法研究

针对当前OMS配网一体化调控方法存在的带宽利用率低、数据传输时延高和数据丢包率高的问题,提出基于大数据调度的OMS配网一体化调控算法.根据任务的稀缺度和紧急度计算任务在OMS配网中的优先请求级别,采用历史信息统计法结合节点能力影响因素计算节点在OMS配网中的上传能力.在任务优先请求级别和节点上传能力的基础上,计算路径在OMS配网中的可用带宽和前向传输时延,并将最大优先算法应用到发生数据丢包的现象中,重新选择传输路径,避免配网中接收端出现乱序的现象,根据计算结果结合重选路策略实现OMS配网的一体化调控.实验结果表明,所提算法的带宽利用率高、数据传输时延低、数据丢包率低.     

A path switching scheme for SCTP based on round trip delays

Due to the rapid development of network applications, today the Internet plays an important role in our everyday life. Users hope that the network is always speedy enough to help them access the Internet without any delay. But the real situation is far from the ideal case. In the future, network researchers will continuously improve the network speed, and try to develop networks that are robust, without any crashes or packet loss. In this paper, we propose an aggressive path switching scheme for SCTP. Before data transmission, the scheme selects the fastest path as the primary path to transmit packets. When the path fails or transmission quality is poor, this scheme evaluates alternate paths, and selects the one with the best quality as the new primary path to substitute for the original one. After that, packets are delivered through the new path. Several factors are considered in the evaluation, including bandwidth, encryption/decryption, size of the congestion window, retransmission policy, routing policy, etc.     

Back pressure based multicast scheduling for fair bandwidth allocation

We study the fair allocation of bandwidth in multicast networks with multirate capabilities. In multirate transmission, each source encodes its signal in layers. The lowest layer contains the most important information and all receivers of a session should receive it. If a receiver's data path has additional bandwidth, it receives higher layers which leads to a better quality of reception. The bandwidth allocation objective is to distribute the layers fairly. We present a computationally simple, decentralized scheduling policy that attains the maxmin fair rates without using any knowledge of traffic statistics and layer bandwidths. This policy learns the congestion level from the queue lengths at the nodes, and adapts the packet transmissions accordingly. When the network is congested, packets are dropped from the higher layers; therefore, the more important lower layers suffer negligible packet loss. We present analytical and simulation results that guarantee the maxmin fairness of the resulting rate allocation, and upper bound the packet loss rates for different layers.     

Online FPS games: effect of router buffer and multiplexing techniques on subjective quality estimators

First Person Shooters are a genre of online games in which users demand a high interactivity, because the actions and the movements are very fast. They usually generate high rates of small packets which have to be delivered to the server within a deadline. When the traffic of a number of players shares the same link, these flows can be aggregated in order to save bandwidth. Certain multiplexing techniques are able to merge a number of packets, in a similar way to voice trunking, creating a bundle which is transmitted using a tunnel. In addition, the headers of the original packets can be compressed by means of standard algorithms. The characteristics of the buffers of the routers which deliver these bundled packets may have a strong influence on the network impairments (mainly delay, jitter and packet loss) which determine the quality of the game. A subjective quality estimator has been used in order to study the mutual influence of the buffer and multiplexing techniques. Taking into account that there exist buffers which size is measured in terms of bytes, and others measured in packets, both kinds of buffers have been tested, using different sizes. Traces from real game parties have been merged in order to obtain the traffic of 20 simultaneous players sharing the same Internet access. The delay and jitter produced by the buffer of the access router have been obtained using simulations. In general, the quality is expected to be reduced as the background traffic grows, but the results show an anomalous region in which the quality rises with the background traffic amount. Small buffers present better subjective quality results than bigger ones. When the total traffic amount gets above the available bandwidth, the buffers measured in bytes add to the packets a fixed delay, which grows with buffer size. They present a jitter peak when the offered traffic is roughly the link capacity. On the other hand, buffers which size is measured in packets add a smaller delay, but they increase packet loss for gaming traffic. The obtained results illustrate the need of knowing the characteristics of the buffer in order to make the correct decision about traffic multiplexing. As a conclusion, it would be interesting for game developers to identify the behaviour of the router buffer so as to adapt the traffic to it.     

基于HSDPA的增强型分组调度算法研究

从系统吞吐量、用户公平性等方面分析研究了HSDPA系统中支持非实时业务的三种经典分组调度算法RR、Max C/I和PF。针对PF算法重传时延过长问题,提出了一种结合混合自动请求重传HARQ的增强分组调度算法。该算法通过提高重传分组的优先级降低重传时延,有效地避免系统资源的浪费。MATLAB仿真结果表明,该算法在降低单用户重传时延的同时,仍能保证用户间的公平性和系统的吞吐量。     

一种按需分配的多路径传输分组调度算法

利用多路径传输协议,多宿主主机可以通过多条路径并行传输数据,从而有效提高系统的吞吐率和鲁棒性.但是由于不同路径在带宽、延迟和丢包率等方面存在差异,接收端必须缓存大量乱序到达的分组.数学分析表明,减少接收端的缓存开销有两条途径:一是最小化每条路径的发送队列中积压分组的数量,二是降低分组发送速率.由前者,提出依据每条路径的空闲发送窗口大小进行分组调度的算法SOD(Scheduling On Demand);由后者,提出利用窗口通告机制限制分组发送速率的流控方法.模拟实验结果表明:与现有算法相比,SOD的缓存开销最小;在接收端进行流控限制的情况下,SOD的吞吐率最大,并且在不同实验场景中性能表现稳定.     

Evaluating the influence of multiplexing schemes and buffer implementation on perceived VoIP conversation quality

This work presents a study of RTP multiplexing schemes, which are compared with the normal use of RTP, in terms of experienced quality. Bandwidth saving, latency and packet loss for different options are studied, and some tests of Voice over IP (VoIP) traffic are carried out in order to compare the quality obtained using different implementations of the router buffer. Voice quality is calculated using ITU R-factor, which is a widely accepted quality estimator. The tests show the bandwidth savings of multiplexing, and also the importance of packet size for certain buffers, as latency and packet loss may be affected. The customer’s experience improvement is measured, showing that the use of multiplexing can be interesting in some scenarios, like an enterprise with different offices connected via the Internet. The system is also tested using different numbers of samples per packet, and the distribution of the flows into different tunnels is found to be an important factor in order to achieve an optimal perceived quality for each kind of buffer. Grouping all the flows into a single tunnel will not always be the best solution, as the increase of the number of flows does not improve bandwidth efficiency indefinitely. If the buffer penalizes big packets, it will be better to group the flows into a number of tunnels. The router processing capacity has to be taken into account too, as the limit of packets per second it can manage must not be exceeded. The obtained results show that multiplexing is a good way to improve customer’s experience of VoIP in scenarios where many RTP flows share the same path.     

Path-Diversity P2P Overlay Retransmission for Reliable IP-Multicast

IP-multicast is a bandwidth efficient transmission mechanism for group communications. Reliability in IP-multicast, however, poses a set of significant challenges. To address the reliability and scalability issues in IP-multicast, this paper proposes a novel, highly distributed, and lightweight overlay peer-to-peer retransmission architecture that exploits path-diversity by taking advantages of both IP-multicast and an overlay network. An approach that leverages both disjoint-path-finding and periodic selective probing to take into account peer's recent packet loss probability, retransmission delay and recent retransmission success rate is proposed to effectively construct an efficient and dynamic overlay retransmission network. We show that the proposed path diversity overlay retransmission architecture has the potential to significantly reduce the retransmission delay, improve the reliability, playback quality, and scalability of IP-multicast based multimedia applications. Given a deployed IP-multicast network, the proposed overlay retransmission architecture is practical, scalable, and easy to deploy, requiring no change to the existing network infrastructure.