无线实时流媒体传输性能的跨层优化设计

提出一种用于无线实时流媒体传输的优化设计策略,以提高接收方的播放质量。该策略采用跨层设计的方法,利用泊松过程分析链路层数据帧的丢失,同时把链路层最大重传次数映射到端到端时延和丢包率的计算中,自适应地调整MPEG视频帧的发送速率,在视频源数据和冗余数据之间动态分配网络带宽。仿真实验结果表明,该策略能使接收方获得最大的可播放帧率,有效提高流媒体传输的可靠性和实时性。     

无线流媒体传输中的速率控制

着眼于无线流媒体服务质量的研究,介绍了目前常用的QoS控制策略,深入研究了视频流传输中基于发送端的速率控制机制。根据无线网络的传输特点,在原有的帧间视频分组调度的基础上,提出了视频分组的帧内调度,有效改善了终端视频的播放质量。     

新型的基于信道状况的自适应HARQ方案

基于信道状况提出自适应的混合自动重传请求（HARQ）方案。充分利用发送端获得的上次成功传输时重传的冗余信息次数和信道估计的信息,判断当前的信道状况,由此确定当前发送的数据帧被成功接收可能需要重传的次数K,将初传信息和K次增量冗余信息一起发送给接收端译码,最大可能地一次性将数据发送成功,减少传输出错再启动重传协议发送增量冗余信息的次数。仿真表明,新的方案在吞吐率基本不变的情况下,传输延时降低,易于实现。     

实时以太网状态分析及其优化策略

为提高实时以太网通信性能和数据帧发送成功率,提出一种基于随机优化理论的实时以太网数据传输优化策略。对实时以太网传输状态进行分析,计算每种状态之间的转移概率,得到状态转移概率矩阵,通过求解平稳状态方程确定以太网处于每种状态的概率,并计算数据帧发生碰撞后每次重传成功概率,以数据帧发送成功率最大为目标函数,对节点发送速率进行优化。实验结果表明,与现有方法相比,该策略在网络传输成功率和吞吐量方面最多提高50．4％和23．4％,在端到端平均延时上也有所改善。     

基于贝叶斯分类器的实时视频优化传输方法

通用UDP协议传输实时视频时存在延时较长的问题。为此,提出一种基于贝叶斯分类器的自适应视频实时传输方法。在发送图像数据前进行帧内压缩,压缩过程根据网络的实时带宽调整压缩质量因子,进行分组发送。在接收端进行数据包重组并解码,利用贝叶斯准则得出自适应调整的阈值,对使用自适应算法与使用固定因子情况下产生的延时进行比较,分析延时产生的原因。实验结果表明,该方法由于优化了传输数据队列的长度,在不增加网络延时的情况下,局部时间内提高了视频图像的传输质量。     

IP网中视频终端的自适应缓冲控制

IP网的数字电视业务应用近年来随着网络技术的飞速进步得到了发展。在这种实时应用中,为进一步减小由IP网传输带来的延时抖动和拥塞对终端视频播放质量造成的影响,该文提出了一种根据视频终端缓冲区容量大小自适应丢帧或重播帧的细粒度缓冲控制机制,使终端缓冲区始终保持在稳定的区域内,同时保证在实施该机制的过程中不影响播放质量。该机制在实践中取得很好的效果,有效地达到了预期的目的。     

一种适合实时视频传输的混合ARQ加条件跳帧算法研究

论文讨论视频数据在无线局域网(WLAN)中的实时传输问题。采用IEEE802.11b协议,WLAN可提供高达11Mbps的传输速率,在没有干扰和没有拥塞的情况下,其环境完全满足实时视频传输的需要。但在干扰和拥塞严重的情况下,视频帧的延时很大,无法满足实时播放的要求。文献犤1犦提出了一种混合ARQ算法,可较好地解决视频帧的传输延迟问题,但在网络负载很重时,丢包率很高,视频序列的重构质量很差。因而在深入研究IEEE802.11b协议的核心MAC层的基础上,参考文献犤1犦,提出了一种新的适合实时视频传输的混合ARQ算法,结合高层的基于反馈信息的条件跳帧技术,可很好地解决视频帧的延时问题,同时传输的可靠性也很高。     

无线上行视频流的跨层容错传输方法

如何通过资源受限的移动通信终端提升无线上行视频流的抗误性能是亟待解决的重要问题。通过不同通信层次的联合调度,提出了一种跨层容错传输方案。移动通信终端的网络层代理首先利用容错包调度为视频流的延时约束帧集合提供重要性分类,随后该终端的链路层代理利用无线链路单元的优先级调度实现选择性重传。在调度延时与传输带宽限制下,跨层容错传输能够将突发错误转移到延时约束帧集合的低优先级视频数据中,从而在突发易错传输环境中实现了无线链路单元粒度的渐进式传输和平稳退化。     

TCP重传计时器的实现和改进

传输控制协议(TCP)是Internet上数据传输采用的核心协议.为提高性能和保证数据的可靠传输,TCP使用了重传计时器的概念.文章对Xinu操作系统中的TCP重传计时器管理和往返时间(RTT：Round Trip Time)测量两方面的问题进行了分析,并提出了改进的方案.     

基于混合流拥塞控制的Ad Hoc网络视频传输

基于TCP反馈的主动队列管理算法会无差别地丢弃UDP分组,导致视频质量下降.根据TCP/UDP发送特性和混合流排队机制,推导拥塞窗口、丢弃概率及混合流队列长度的微分方程,通过线性化获得Ad Hoc网络混合流AQM(active queue management)控制模型,提出了一种减少UDP丢弃率的PI-V主动队列管理算法.当视频画质下降时,算法依据视频帧权重与分组长度,动态地调整UDP视频流的丢弃概率.NS(network simulator)仿真表明,PI-V算法优于PI与去尾算法,提高了视频传输质量0.82 d B~2.22 d B的峰值信噪比(PSNR).     

Why TCP timers (still) don’t work well

We argue that the design principles of the TCP timeout algorithm are based solely on RTT estimations and may lead to flow synchronization, unnecessary retransmission effort and unfair resource allocation. We present a new Window-Based Retransmission Timeout algorithm (WB-RTO) for TCP, which exhibits two major properties: (i) it cancels retransmission synchronization, which dominates when resource demand exceeds resource supply and (ii) it reschedules flows on the basis of their contribution to congestion. WB-RTO achieves better fairness and slightly better goodput with significant less retransmission effort.     

Real-time video streaming over multipath in multi-hop wireless networks

Given the limited wireless link throughput, high loss rate, and varying end-to-end delay, supporting video applications in multi-hop wireless networks becomes a challenging task. Path diversity exploits multiple routes for each session simultaneously, which achieves higher aggregated bandwidth and potentially decreases delay and packet loss. Unfortunately, for TCP-based video streaming, naive load splitting often results in inaccurate estimation of round trip time (RTT) and packet reordering. As a result, it can suffer from significant instability or even throughput reduction, which is also validated by our analysis and simulation in multi-hop wireless networks. To make real-time TCP-based streaming viable over multi-hop wireless networks, we propose a novel cross-layer design with a smart traffic split scheme, namely, multiple path retransmission (MPR). MPR differentiates the original data packets and the retransmitted packets and works with a novel QoS-aware multi-path routing protocol, QAOMDV, to distribute them separately. MPR does not suffer from the RTT underestimation and extra packet reordering, which ensures stable throughput improvement over single-path routing. Through extensive simulations, we further demonstrate that, as compared with state-of-the-art multi-path protocols, our MPR with QAOMDV noticeably enhances the TCP streaming throughput and reduces bandwidth fluctuation, with no obvious impact to fairness.     

Design of feedback controls supporting TCP based on the state-space approach

This paper investigates how to design feedback controls supporting transmission control protocol (TCP) based on the state-space approach for the linearized system of the well-known additive increase multiplicative decrease (AIMD) dynamic model. We formulate the feedback control design problem as state-space models without assuming its structure in advance. Thereby, we get three results that have not been observed by previous studies on the congestion control problem. 1) In order to fully support TCP, we need a proportional-derivative (PD)-type state-feedback control structure in terms of queue length (or RTT: round trip time). This backs up the conjecture in the networking literature that the AQM RED is not enough to control TCP dynamic behavior, where RED can be classified as a P-type AQM (or as an output feedback control for the linearized AIMD model). 2) In order to fully support TCP in the presence of delays, we derive delay-dependent feedback control structures to compensate for delays explicitly under the assumption that RTT, capacity and number of sources are known, where all existing AQMs including RED, REM/PI and AVQ are delay-independent controls. 3) In an attempt to interpret different AQM structures in a unified manner rather than to compare them via simulations, we propose a PID-type mathematical framework using integral control action. As a performance index to measure the deviation of the closed-loop system from an equilibrium point, we use a linear quadratic (LQ) cost of the transients of state and control variables such as queue length, aggregate rate, jitter in the aggregate rate, and congestion measure. Stabilizing gains of the feedback control structures are obtained minimizing the LQ cost. Then, we discuss the impact of the control structure on performance using the PID-type mathematical framework. All results are extended to the case of multiple links and heterogeneous delays.     

一种CDMA2000链路层自适应重传算法

TCP协议应用在3G无线网络中时，其传输性能受到了无线网络物理层帧差错率的严重影响。链路层重传技术可以大大改善TCP数据传输性能因无线信道误码率大而下降的情况。主要研究了CDMA2000中链路层重传技术对无线TCP数据传输的影响，并结合TCP层重传超时机制，提出了一种新的自适应链路层重传算法，提高了TCP在无线链路的数据传输性能。最后对该算法进行仿真，验证了此算法可以显著提高TCP吞吐率。     

TCP Vegas拥塞控制机制的改进

针对Vegas拥塞控制技术的不足，提出了一种改进方法。首先对发送分组根据长度进行分类，分别计算各长度类别的“往返时间”（RTT）和RTT最小值（base_rtt）；在更新发送窗口以及设置重传定时器时，根据分组长度选择属于同一长度范围的RTT和base_rtt值作为输入参数。这在一定程度上消除分组长度差异对发送窗口更新和重传定时器设定值所产生的影响，进一步提高协议吞吐率。仿真结果表明，改进的Vegas能够将原协议的吞吐率提高多达24％。     

CIR、RTT和RTO感知的三色标记算法

往返时间(RTT)、目标速度（CIR）以及重传超时(RTO)等因素导致了带宽分配的不均匀。通过研究带宽与这些因素的关系,在时间滑动窗口三色标记器（TSW3CM）的基础上,提出一种CIR、RTT、RTO感知的标记器(CRR3CM)。该标记器完成了在各汇聚流之间对剩余带宽的公平分配,减少了目标速度、往返时间,以及重传超时因素对带宽分配的影响。模拟实验表明,与TSW3CM相比,该算法有效地提高了TCP流之间带宽分配的公平性。     

Layer-encoded video in scalable adaptive streaming

Combining the concepts of caching and transmission control protocol (TCP)-friendly streaming of layer-encoded video bears the problem that those videos might not be cached in full quality. Therefore, we focus in this work on the scheduling of retransmissions of missing segments of a cached video in a manner that allows clients to receive the content in an improved quality. In a first step, we conducted subjective assessments of variations in layer-encoded video with the goal to validate existing quality metrics, including our own, which are based on certain assumptions. A statistical analysis of the subjective assessment validates these assumptions. We also show that the frequently used peak signal-to-noise ratio (PSNR) is not an appropriate metric for variations in layer-encoded video. With the insight from the subjective assessment we develop heuristics for retransmission scheduling and prove their applicability by conducting a series of simulations.     

基于仿真的TCP拥塞控制研究

研究了几种不同的TCP拥塞控制算法原理。通过对于TCP Reno和TCP Vegas协议的实验仿真,研究在不同的数据流和不同的网络条件下算法的性能差异。提出了一种改进的RTT估计方法,在拥塞避免阶段,采用时延的指数滑动平均值取代瞬时的RTT。实验表明,这种改进增强了TCP Vcgas对于时延扰动的鲁棒性。     

DCCP下的一种流媒体重传机制

由于差错重传机制存在各种缺点,并不适合流媒体传输,若它的处理过程占用最小网络带宽和时间,将大大提高流媒体的质量。提出了一种基于DCCP的源端控制的选择重传机制（TBSR）,其重传由决策算法决定,而发送速率和拥塞控制由TFRC控制。仿真结果表明,机制能大大提高流媒体视频质量。     

基于运动估计与回溯自适应正交匹配追踪的视频压缩传感

为了解决传统视频压缩传感方法中对视频逐帧单独重构所产生的图像模糊,将压缩传感理论与MPEG标准视频编码的相关技术相结合,提出了一种基于运动估计与运动补偿的视频压缩传感方法,以消除视频信号在空域和时域上的冗余。该方法在充分考虑视频序列时域相关性的同时,首先对视频图像进行前、后向和双向预测和补偿,然后采用回溯自适应正交匹配追踪(BAOMP)算法,对运动预测残差进行重构,最后实现当前帧的重构。实验结果表明,该方法较逐帧重构的视频图像质量有较大改善,且可获得更高的峰值信噪比。