Optimization of TFRC Loss History Initialization

This letter deals with the initialization of the loss history structure in the TFRC (TCP-friendly rate control) mechanism. This initialization occurs after the detection of the first loss event after every slowstart phase. The loss history is crucial for the algorithm since it returns the packet loss rate estimation. This estimation is used in the TFRC equation to compute the sending rate. In this letter, we propose a new method to compute the packet loss rate which is more computationally efficient and remains as accurate as the classical commonly used method. The motivation of this work is to reduce the computation time and formulate a unified computation scheme. This method is based on the Newton's algorithm issued from numerical analysis of the TCP throughput equation. This proposal is evaluated analytically and the results show a significant improvement in terms of the computation time     

TFRC协议友好性与平稳性改进算法研究

 本文针对TFRC(TCP-Friendly Rate Control)流与TCP流竞争带宽时的友好性问题,分析了影响TFRC协议TCP友好性的因素,通过对TFRC速率计算公式中丢包率的不同幂级项引入权重系数,增加网络拥塞严重时的发送速率,减少网络拥塞较轻时的发送速率,从而降低了网络拥塞程度对TFRC流传输速率的影响.仿真实验表明该方法对TFRC协议具有较明显改进作用,提高了TFRC流的传输平稳度和TCP友好性,从而能更有效地适应多媒体流的传输要求.     

一种基于多路复用的多媒体流TCP友好拥塞控制机制

本文重点研究在多路复用的链路环境中,TCP友好与多媒体流最低速率阈值限定之间的权衡关系,提出了一种基于多路复用的TCP友好速率控制算法-MTCRC(Multiplexing and threshold-constrained rate control).MTCRC引入基于概率的随机试验技术,以保证多媒体流在多路复用时,当友好速率低于限定的最低速率时,通过在适当的时间对部分流的挂起操作,使多媒体流的平均吞吐量仍保持TCP友好.MTCRC是对TFRC(TCP-friendly rate control)的改进,它在保持TFRC良好的速率平滑性的同时,增加了对多媒体流最低速率阈值限定特性及多路复用链路环境的考虑,使其既能尽量保持多媒体流应用的有效性,又能与竞争的TCP流公平地分享带宽.模拟结果显示:MTCRC的性能明显优于TFRC.     

无线网络中TCP友好流媒体传输改进机制

为保持无线网络中多媒体业务对TCP的友好性，提出了一种适用于无线网络的动态自适应的流媒体传输速率调节机制。该机制通过在接收端区分网络拥塞丢包和链路错误随机丢包，准确判断网络的拥塞状况结合接收端缓存区占用程度，自适应实施多级速率调节，实现了TCP流友好性和流媒体服务质量（QoS）的折中。由于准确区分出无线链路误码丢包和动态调整流媒体QoS要求，该机制能维持较高的网络利用率。仿真实验结果显示在连接数为2和32，链路误码率从0到0．1变化时TCP,TFRC和吞吐量幅度下降幅度较大，WTFCC幅度下降相对较慢，最大相差达2M；在网络负载重时，尽管链路误码率较低，WTFCC区分链路错误与拥塞丢包，因此，端到端丢包率高于TCP和TFRC，但整体传输吞吐量也高于两者。归一化吞吐量显示WTFCC对TCP流友好。     

VIRAL: coupling congestion control with fair video quality metric

Video streaming is often carried out by congestion controlled transport protocols to preserve network sustainability. However, the success of the growth of such non-live video flows is linked to the user quality of experience. Thus, one possible solution is to deploy complex quality of service systems inside the core network. Another possibility would be to keep the end-to-end principle while making aware transport protocols of video quality rather than throughput. The objective of this article is to investigate the latter by proposing a novel transport mechanism which targets video quality fairness among video flows. Our proposal, called VIRAL for virtual rate-quality curve, allows congestion controlled transport protocols to provide fairness in terms of both throughput and video quality. VIRAL is compliant with any rate-based congestion control mechanisms that enable a smooth sending rate for multimedia applications. Implemented inside TFRC a TCP-friendly protocol, we show that VIRAL enables both intra-fairness between video flows in terms of video quality and inter-fairness in terms of throughput between TCP and video flows.     

Equation-based end-to-end single-rate multicast congestion control

Among the recently proposed single-rate multicast congestion control protocols is transmission control protocol-friendly multicast congestion control (TFMCC; Widmer and Handley 2001; Floyd et al. 2000; Widmer et al. IEEE Netw 15:28–37, 2001), which is an equation-based single-rate protocol that extends the mechanisms of the unicast TCP-friendly rate control (TFRC) protocol into the multicast domain. In TFMCC, each receiver estimates its throughput using an equation that estimates the steady-state throughput of a TCP source. The source then adjusts its sending rate according to the slowest receiver within the session (a.k.a., current-limiting receiver, CLR). TFMCC is a relatively simple, scalable, and TCP-friendly multicast congestion control protocol. However, TFMCC is hindering its throughput performance by adopting an equation derived from the unicast TFRC protocol. Further, TFMCC is slow to react to congestion conditions that usually result in a change of the CLR. This paper is motivated by these two observations and proposes an improved version of TFMCC, which we refer to as hybrid-TFMCC (or H-TFMCC for short). First, each receiver estimates its throughput using an equation that models the steady-state throughput of a multicast source controlled according to the additive increase multiplicative decrease (AIMD) approach. The second modification consists of adopting a hybrid sender/receiver-based rate control strategy, where the sending rate can be adjusted by the source or initiated by the current or a new CLR. The source monitors RTT variations on the CLR path, in order to rapidly adjust the sending rate to network conditions. Simulation results show that these modifications result in remarkable performance improvement with respect to throughput, time to react, and magnitude of oscillations. We also show that H-TFMCC remains TCP-friendly and achieves a higher fairness index than that achieved by TFMCC.     

Taxonomy and Evaluation of TCP-Friendly Congestion-Control Schemes on Fairness, Aggressiveness, and Responsiveness

Many TCP-friendly congestion control schemes have been proposed to pursue the TCP-equivalence criterion, which states that a TCP-equivalent flow should have the same throughput with TCP if it experiences identical network conditions as TCP. Additionally, the throughput should converge as fast as TCP when the packet-loss conditions change. This study classifies eight typical TCP-friendly schemes according to their underlying policies on fairness, aggressiveness, and responsiveness. The schemes are evaluated to verify whether they meet TCP-equivalence and TCP-equal share. TCP-equal share is a more realistic but more challenging criterion than TCP-equivalence and states that a flow should have the same throughput with TCP if competing with TCP for the same bottleneck. Simulation results indicate that one of the selected schemes, TCP-friendly rate control (TFRC), meets both criteria under more testing scenarios than the others. Additionally, the results under non-periodic losses, low-multiplexing, two-state losses, and bursty losses reveal the causes that bring fault cases to the schemes. Finally, appropriate policies are recommended for an ideal scheme.     

On the long-run behavior of equation-based rate control

We consider unicast equation-based rate control, where, at some points in time, a source adjusts its rate to f(p,r). Here p is an on-line estimate of the loss-event rate, r, of the mean round-trip time, both as observed by this source, and f is a TCP throughput formula. It was generally believed that such a source would be TCP-friendly, that is, under the same operating conditions, its long-run time-average send rate (throughput) would not be larger than that of a TCP source. Our goal is to identify whether, and how far, this is true. First, we identify factors that play a role in TCP friendliness and find that it is important to study them separately. Then we analyze the importance of individual factors. A first factor is conservativeness (= throughput not larger than f(p,r)). We show that conservativeness is influenced by some convexity properties of f(p,r) with respect to p, and the covariance of the loss process. We show that in many real life cases these conditions result in conservativeness and, sometimes, excessive conservativeness. This explains the previously observed phenomena of throughput-drop when losses are high and f is the so-called PFTK formula. The second factor is that the source may experience considerably different loss-event rate than a TCP source. We identify and analyze two limit cases where this may lead to either TCP-friendliness or, in contrast, non-TCP-friendliness. Other factors such as round trip time and obedience of TCP to its own formula are found to be less significant. Our claims are obtained by analysis, and verified by numerical examples, simulations, laboratory and Internet experiments. Our results suggest that TCP-friendliness is difficult to verify in practice, whereas conservativeness is easier.     

TCP-friendly flow control of wireless multimedia using ECN marking

In a wireless network packet losses can be caused not only by network congestion but also by unreliable error-prone wireless links. Therefore, flow control schemes which use packet loss as a congestion measure cannot be directly applicable to a wireless network because there is no way to distinguish congestion losses from wireless losses. In this paper, we extend the so-called TCP-friendly flow control scheme, which was originally developed for the flow control of multimedia flows in a wired IP network environment, to a wireless environment. The main idea behind our scheme is that by using explicit congestion notification (ECN) marking in conjunction with random early detection (RED) queue management scheme intelligently, it is possible that not only the degree of network congestion is notified to multimedia sources explicitly in the form of ECN-marked packet probability but also wireless losses are hidden from multimedia sources. We calculate TCP-friendly rate based on ECN-marked packet probability instead of packet loss probability, thereby effectively eliminating the effect of wireless losses in flow control and thus preventing throughput degradation of multimedia flows travelling through wireless links. In addition, we refine the well-known TCP throughput model which establishes TCP-friendliness of multimedia flows in a way that the refined model provides more accurate throughput estimate of a TCP flow particularly when the number of TCP flows sharing a bottleneck link increases. Through extensive simulations, we show that the proposed scheme indeed improves the quality of the delivered video significantly while maintaining TCP-friendliness in a wireless environment for the case of wireless MPEG-4 video.     

gTFRC, a TCP friendly QoS-aware rate control for DiffServ Assured Service

This study addresses the end-to-end congestion control support over the DiffServ Assured Forwarding (AF) class. The resulting Assured Service (AS) provides a minimum level of throughput guarantee. In this context, this article describes a new end-to-end mechanism for continuous transfer based on TCP-Friendly Rate Control (TFRC). The proposed approach modifies TFRC to take into account the QoS negotiated. This mechanism, named gTFRC, is able to reach the minimum throughput guarantee whatever the flow's RTT and target rate. Simulation measurements and implementation over a real QoS testbed demonstrate the efficiency of this mechanism either in over-provisioned or exactly-provisioned network. In addition, we show that the >frc mechanism can be used in the same DiffServ/AF class with TCP or TFRC flows.     

一种拥塞感知的TFRC协议慢启动算法

 本文分析了TFRC(TCP-Friendly Rate Control)协议在慢启动阶段采用类似TCP协议的倍增发送速率机制存在的问题,提出了一种利用回路响应时间(Round Trip Time,RTT)来自适应调节慢启动阶段速率的算法.通过分析实际RTT值和EWMA(Exponentially Weighted Moving Average)处理后的平均RTT值来感知网络当前的拥塞状况,以调节发送速率的激进程度.仿真实验表明,该方法对TFRC协议具有明显的改进作用,减少了慢启动阶段结束时的报文丢失率,提高了协议的传输平稳度和吞吐量,从而能更有效地适应多媒体流的传输要求.     

Revisiting the fair queuing paradigm for end-to-end congestion control

Today, the dominant paradigm for congestion control in the Internet is based on the notion of TCP friendliness. To be TCP-friendly, a source must behave in such a way as to achieve a bandwidth that is similar to the bandwidth obtained by a TCP flow that would observe the same round-trip time (RTT) and the same loss rate. However, with the success of the Internet comes the deployment of an increasing number of applications that do not use TCP as a transport protocol. These applications can often improve their own performance by not being TCP-friendly, which severely penalizes TCP flows. To design new applications to be TCP-friendly is often a difficult task. The idea of the fair queuing (FQ) paradigm as a means to improve congestion control was first introduced by Keshav (1991). While Keshav made a fundamental step toward a new paradigm for the design of congestion control protocols, he did not formalize his results so that his findings could be extended for the design of new congestion control protocols. We make this step and formally define the FQ paradigm as a paradigm for the design of new end-to-end congestion control protocols. This paradigm relies on FQ scheduling with per-flow scheduling and longest queue drop buffer management in each router. We assume only selfish and noncollaborative end users. Our main contribution is the formal statement of the congestion control problem as a whole, which enables us to demonstrate the validity of the FQ paradigm. We also demonstrate that the FQ paradigm does not adversely impact the throughput of TCP flows and explain how to apply the FQ paradigm for the design of new congestion control protocols. As a pragmatic validation of the FQ paradigm, we discuss a new multicast congestion control protocol called packet pair receiver-driven layered multicast (PLM).     

End-to-end optimized TCP-friendly rate control for real-time video streaming over wireless multi-hop networks

Rate control is an important issue in video streaming applications. The most popular rate control scheme over wired networks is TCP-Friendly Rate Control (TFRC), which is designed to provide optimal transport service for unicast multimedia delivery based on the TCP Reno’s throughput equation. It assumes perfect link quality, treating network congestion as the only reason for packet losses. Therefore, when used in wireless environment, it suffers significant performance degradation because of packet losses arising from time-varying link quality. Most current research focuses on enhancing the TFRC protocol itself, ignoring the tightly coupled relation between the transport layer and other network layers. In this paper, we propose a new approach to address this problem, integrating TFRC with the application layer and the physical layer to form a holistic design for real-time video streaming over wireless multi-hop networks. The proposed approach can achieve the best user-perceived video quality by jointly optimizing system parameters residing in different network layers, including real-time video coding parameters at the application layer, packet sending rate at the transport layer, and modulation and coding scheme at the physical layer. The problem is formulated and solved as to find the optimal combination of parameters to minimize the end-to-end expected video distortion constrained by a given video playback delay, or to minimize the video playback delay constrained by a given end-to-end video distortion. Experimental results have validated 2–4 dB PSNR performance gain of the proposed approach in wireless multi-hop networks by using H.264/AVC and NS-2.     

Cross-Layer Interaction of TCP and Ad Hoc Routing Protocols in Multihop IEEE 802.11 Networks

In this research, we first investigate the cross-layer interaction between TCP and routing protocols in the IEEE 802.11 ad hoc network. On-demand ad hoc routing protocols respond to network events such as channel noise, mobility, and congestion in the same manner, which, in association with TCP, deteriorates the quality of an existing end-to-end connection. The poor end-to-end connectivity deteriorates TCP's performance in turn. Based on the well-known TCP-friendly equation, we conduct a quantitative study on the TCP operation range using static routing and long-lived TCP flows and show that the additive-increase, multiplicative-decrease (AIMD) behavior of the TCP window mechanism is aggressive for a typical multihop IEEE 802.11 network with a low-bandwidth-delay product. Then, to address these problems, we propose two complementary mechanisms, that is, the TCP fractional window increment (FeW) scheme and the Route-failure notification using BUIk-losS Trigger (ROBUST) policy. The TCP FeW scheme is a preventive solution used to reduce the congestion-driven wireless link loss. The ROBUST policy is a corrective solution that enables on-demand routing protocols to suppress overreactions induced by the aggressive TCP behavior. It is shown by computer simulation that these two mechanisms result in a significant improvement of TCP throughput without modifying the basic TCP window or the wireless MAC mechanisms.     

Rate control for streaming video over wireless

Rate control is an important issue in video streaming applications for both wired and wireless networks. A widely accepted rate control method in wired networks is TCP-friendly rate control (TFRC) (Floyd, 2000). It is equation-based rate control in which the TCP-friendly rate is determined as a function of packet loss rate, round-trip time, and packet size. TFRC assumes that packet loss in wired networks is primarily due to congestion, and as such is not applicable to wireless networks in which the main cause of packet loss is at the physical layer. In this article we review existing approaches to solve this problem. Then we propose multiple TFRC connections as an end-to-end rate control solution for wireless video streaming. We show that this approach not only avoids modifications to the network infrastructure or network protocol, but also results in full utilization of the wireless channel. NS-2 simulations, actual experiments over a 1/spl times/RTT CDMA wireless data network, and video streaming simulations using traces from the actual experiments are carried out to characterize the performance and show the efficiency of our proposed approach.     

无线网络下一种改进的TCP拥塞控制机制TCP_RD

本文探讨了无线网络下误码丢包对TCP性能的影响,认为短期内误码丢包严重时,可以通过数据发送速率的降低来有效避免不必要的误码丢包,从而提高数据发送的可靠性,减少移动主机不必要的能源消耗和系统的额外开销。基于此,本文提出了一种改进的TCP拥塞控制机制TCP_RD,此机制实现简单,系统额外开销小,既能有效提高数据发送的可靠性,又不会过多降低系统吞吐量和加大系统时延。     

新的多速率多播拥塞控制方案

提出了一种新的基于数据包束探测(packet-bunch probe)和TCP吞吐量公式的多速率多播拥塞控制方案PTMCC(packet-bunch probe and TCP-formula based multicast congestion control)。这种接收端驱动的拥塞控制,采用数据包束来探测网络的可用带宽,利用TCP吞吐量公式得到TCP友好速率,并采用了新的速率调节算法。仿真实验表明,PTMCC在收敛性、灵敏性以及TCP友好性上具有较好的性能。     

TCP-Aware Channel Allocation in CDMA Networks

This paper explores the use of rate adaptation in cellular networks to maximize throughput of long-lived TCP sessions. We focus on the problem of maximizing the throughput of TCP connections and propose a joint optimization of MAC and physical layer parameters with respect to TCP sending rate. In particular, we propose a simple TCP-aware channel scheduler that adapts the wireless channel rate to changes in the TCP sending rate and explore its performance for both single and multiple concurrent sessions. In the case of a single TCP session, we develop a fluid model of its steady-state behavior in such a system that adapts between two channel rates. Our results indicate that a two-rate scheme improves TCP throughput by 15% to 20% over a system that does not exploit rate adaptation and that little additional benefit accrues from the addition of a third channel rate. Finally, we extend the framework to scenarios where bandwidth is shared by multiple TCP sessions. We propose two channel allocation algorithms and explore their performance through simulation. Our results indicate that TCP throughput is relatively insensitive to either channel allocation algorithm and adaptive rate variation is the dominant factor in performance.     

解决FAST TCP缓存溢出相关问题的改进pacing technique算法和α参数调整算法

分析FAST TCP在缓存溢出发生时的性能,发现在缓存溢出场景中,收敛中的FAST TCP流经历严重的报文段丢失。相反,已经收敛了的FAST TCP流维持着高吞吐量和低报文段丢失概率。这种不公平是由FAST TCP缩减其窗口时的零传输率导致的。通过修改FAST TCP pacing算法,可以解决此问题。文中提出的α-adjusting算法,通过动态调整FAST TCP协议中的α参数来避免频繁的缓存溢出。通过分析ns2仿真结果,证明该算法在公平性和稳定性方面可获得令人满意的性能。     

基于模糊逻辑的无线网络流量自适应控制

文章在分析现有提高无线TCP性能方案的基础上，提出一种新的流量控制方法，即基于显式窗口反馈的无线网络流量控制方案。在Snoop中引入有线网络的显示窗口自适应（EWA）算法，通过对BS共享缓存的实时监测，应用模糊控制算法预测当前拥塞窗口（cwnd）的大小，并显式反馈给发送端，使TCP的发送窗口能快速响应网络负荷状况的变化．避免分组的丢失。仿真结果表明．该方法增强了网络对拥塞的自适应性以及对无线信道差错的实时处理能力．提高了网络的吞吐量。