Limitations of Equation-Based Congestion Control

We study limitations of an equation-based congestion control protocol, called TCP-friendly rate control (TFRC). It examines how the three main factors that determine TFRC throughput, namely, the TCP-friendly equation, loss event rate estimation, and delay estimation, can influence the long-term throughput imbalance between TFRC and TCP. Especially, we show that different sending rates of competing flows cause these flows to experience different loss event rates. There are several fundamental reasons why TFRC and TCP flows have different average sending rates, from the first place. Earlier work shows that the convexity of the TCP-friendly equation used in TFRC causes the sending rate difference. We report two additional reasons in this paper: 1) the convexity of where is a loss event period and 2) different retransmission timeout period (RTO) estimations of TCP and TFRC. These factors can be the reasons for TCP and TFRC to experience initially different sending rates. But we find that the loss event rate difference due to the differing sending rates greatly amplifies the initial throughput difference; in some extreme cases, TFRC uses around 20 times more, or sometimes 10 times less, bandwidth than TCP. Despite these factors influencing the throughput difference, we also find that simple heuristics can greatly mitigate the problem.     

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.     

一种在接收端实现的TCP-Friendly拥塞控制机制

本文提出了一种基于速率的单播TCP-Friendly拥塞控制算法——RAAR(Rate Adaptation at Receivers)控制机制.RAAR是一种接收端的速率自适应算法,它抛弃了每包反馈机制,采用GAIMD(General Additive Increase Multiplicative Decrease)策略进行拥塞控制,其主要控制操作由接收方完成.本文建立了简化的数学模型对其进行吞吐量的分析,得到在RAAR中用于TCP-Friendly 的GAIMD拥塞控制中α与 β的关系.通过与TFRC及TEAR这两种重要的TCP-Friendly协议进行对比研究发现,RAAR协议在对TCP协议的友好性,协议内的公平性以及速率的平滑性等方面具有更好的综合性能.由于RAAR不需进行每包反馈,且主要功能在接收方实现,因此可方便地将该机制引入多媒体组播传输系统中.     

Leveraging single rate schemes in multiple rate multicast congestion control design

A significant impediment to deployment of multicast services is the daunting technical complexity of developing, testing and validating congestion control protocols fit for wide-area deployment. Protocols such as pragmatic general multicast congestion control (pgmcc) and TCP-friendly multicast congestion control (TFMCC) have recently made considerable progress on the single rate case, i.e., where one dynamic reception rate is maintained for all receivers in the session. However, these protocols have limited applicability, since scaling to session sizes beyond tens of participants with heterogeneous available bandwidth necessitates the use of multiple rate protocols. Unfortunately, while existing multiple rate protocols exhibit better scalability, they are both less mature than single rate protocols and suffer from high complexity. We propose a new approach to multiple rate congestion control that leverages proven single rate congestion control methods by orchestrating an ensemble of independently controlled single rate sessions. We describe a new multiple rate congestion control algorithm for layered multicast sessions that employs a single rate multicast congestion control as the primary underlying control mechanism for each layer. Our new scheme combines the benefits of single rate congestion control with the scalability and flexibility of multiple rates to provide a sound multiple rate multicast congestion control policy.     

一种新的逐跳TCP友好的主动分层多播拥塞控制方案

基于Internet的多媒体多播应用的迅猛发展对多播拥塞控制提出了要求.分层多播是适应网络异构性较有效的方案.针对现有分层多播存在的问题,将主动网技术思想引入到分层多播拥塞控制中,提出了一种逐跳TCP友好的主动分层多播拥塞控制方案(HTLMA),采用主动标记分层、逐跳探测TCP友好可用带宽,以及主动速率控制机制.仿真实验表明,HTLMA方案大大改进了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.     

On max-min fair congestion control for multicast ABR service in ATM

In the ATM Forum activities, considerable efforts have focused on the congestion control of point-to-point available bit rate (ABR) service. We present a novel approach that extends existing point-to-point (unicast) congestion control protocols to a point-to-multipoint (multicast) environment. In particular, we establish a unified framework to derive a multicast congestion control protocol for an ABR service from a given rate-based unicast protocol. We generalize a known necessary and sufficient condition on the max-min fairness of unicast rate allocation for a multicast service. Using this condition, we show that the resulting multicast protocol derived using our framework preserves the fairness characteristics of the underlying unicast protocol. The practical significance of our approach is illustrated by extending a standard congestion control mechanism for an ABR service to a multicast environment. The performance of the resulting multicast protocol is examined using benchmark network configurations suggested by the traffic management subworking group at the ATM Forum, and simulation results are presented to substantiate our claims     

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).     

基于显式速率的TCP友好的UDP拥塞控制策略

本文提出了一种基于显式速率的UDP拥塞控制策略：通过源端和网络中的路由器相互配合，使得实时UDP应用能够根据网络的反馈以瓶颈链路的公平带宽为速率发送数据。此种控制策略对TCP应用是友好的，并且提高了网络的吞吐量和利用率。仿真结果表明：基于显式速率的UDP拥塞控制策略与采用TFRC(TCP—Friendly Rate Control)的UDP拥塞控制策略相比，在吞吐量、TCP友好性等方面性能有较大提高。     

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。     

High-throughput multicast routing metrics in wireless mesh networks

The stationary nature of nodes in a mesh network has shifted the main design goal of routing protocols from maintaining connectivity between source and destination nodes to finding high-throughput paths between them. Numerous link-quality-based routing metrics have been proposed for choosing high-throughput routing paths in recent years. In this paper, we study routing metrics for high-throughput tree or mesh construction in multicast protocols. We show that there is a fundamental difference between unicast and multicast routing in how data packets are transmitted at the link layer, and accordingly how the routing metrics for unicast routing should be adapted for high-throughput multicast routing. We propose a low-overhead adaptive online algorithm to incorporate link-quality metrics to a representative multicast routing protocol. We then study the performance improvement achieved by using different link-quality-based routing metrics via extensive simulation and experiments on a mesh-network testbed, using ODMRP as a representative multicast protocol.Our extensive simulation studies show that: (1) ODMRP equipped with any of the link-quality-based routing metrics can achieve higher throughput than the original ODMRP. In particular, under a tree topology, on average, ODMRP enhanced with link-quality routing metrics achieve up to 34% higher throughput than the original ODMRP under low multicast sending rate; (2) the improvement reduces to 21% under high multicast sending rate due to higher interference experienced by the data packets from the probe packets; (3) heavily penalizing lossy links is an effective way in the link-quality metric design to avoid low-throughput paths; and (4) the path redundancy from a mesh data dissemination topology in mesh-based multicast protocols provides another degree of robustness to link characteristics and reduces the additional throughput gain achieved by using link-quality-based routing metrics. Finally, our experiments on an eight-node testbed show that on average, ODMRP using SPP and PP achieves 14% and 17% higher throughput over ODMRP, respectively, validating the simulation results.     

A survey on TCP-friendly congestion control

New trends in communication, in particular the deployment of multicast and real-time audio/video streaming applications, are likely to increase the percentage of non-TCP traffic in the Internet. These applications rarely perform congestion control in a TCP-friendly manner; they do not share the available bandwidth fairly with applications built on TCP, such as Web browsers, FTP, or e-mail clients. The Internet community strongly fears that the current evolution could lead to congestion collapse and starvation of TCP traffic. For this reason, TCP-friendly protocols are being developed that behave fairly with respect to coexistent TCP flows. We present a survey of current approaches to TCP friendliness and discuss their characteristics. Both unicast and multicast congestion control protocols are examined, and an evaluation of the different approaches is presented     

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.     

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.     

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.     

The impact of multicast layering on network fairness

Many definitions of fairness for multicast networks assume that sessions are single rate, requiring that each multicast session transmits data to all of its receivers at the same rate. These definitions do not account for multirate approaches, such as layering, that permit receiving rates within a session to be chosen independently. We identify four desirable fairness properties for multicast networks, derived from properties that hold within the max-min fair allocations of unicast networks. We extend the definition of multicast max-min fairness to networks that contain multirate sessions, and show that all four fairness properties hold in a multirate max-min fair allocation, but need not hold in a single-rate max-min fair allocation. We then show that multirate max-min fair rate allocations can be achieved via intra-session coordinated joins and leaves of multicast groups. However, in the absence of coordination, the resulting max-min fair rate allocation uses link bandwidth inefficiently, and does not exhibit some of the desirable fairness properties. We evaluate this inefficiency for several layered multirate congestion control schemes, and find that, in a protocol where the sender coordinates joins, this inefficiency has minimal impact on desirable fairness properties. Our results indicate that sender-coordinated layered protocols show promise for achieving desirable fairness properties for allocations in large-scale multicast networks     

A study of a simple preventive transport protocol

The main qualities of a protocol for multimedia flows transportation are related to the way congestions are handled. This paper addresses the problem of end-to-end congestion control performed in the Internet transport layer. We present a simple protocol called Primo, which determines the appropriate sending rate in order to maximize network resources usage and minimize packets loss. Comparison with existing transport protocols (Tcp Reno, Sack, Vegas andTfrc) are considered, regarding various efficiency criteria such as sending and reception rates stability, loss rate, resources occupancy rate and fairness.     

Multicast Capacity of Packet-Switched Ring WDM Networks

Packet-switched unidirectional and bidirectional ring wavelength division multiplexing (WDM) networks with destination stripping provide an increased capacity due to spatial wavelength reuse. Besides unicast traffic, future destination stripping ring WDM networks also need to support multicast traffic efficiently. This article examines the largest achievable transmitter throughput, receiver throughput, and multicast throughput of both unidirectional and bidirectional ring WDM networks with destination stripping. A probabilistic analysis evaluates both the nominal capacity, which is based on the mean hop distances traveled by the multicast packet copies, and the effective capacity, which is based on the ring segment with the highest utilization probability, for each of the three throughput metrics. The developed analytical methodology accommodates not only multicast traffic with arbitrary multicast fanout but also unicast and broadcast traffic. Numerical investigations compare the nominal transmission, receiver, and multicast capacities with the effective transmission, receiver, and multicast capacities and examine the impact of number of ring nodes and multicast fanout on the effective transmission, reception, and multicast capacity of both types of ring networks for different unicast, multicast, and broadcast traffic scenarios and different mixes of unicast and multicast traffic. The presented analytical methodology enables the evaluation and comparison of future multicast-capable medium access control (MAC) protocols for unidirectional and bidirectional ring WDM networks in terms of transmitter, receiver, and multicast throughput efficiency.     

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.     

Cryptanalysis and Improvement of Robust Deniable Authentication Protocol

A deniable authentication protocol enables a receiver to identify the source of a given message, but the receiver cannot prove the source of the message to a third party. Recently, Yoon et al. (Wirel Pers Commun 55:81–90, 2010) proposed a robust deniable authentication protocol based on ElGamal cryptosystem. Although they proved that their protocol satisfies the deniable authentication, mutual authentication and confidentiality, we show that their protocol does not satisfy the deniable authentication property. The receiver can prove the source of a given message to a third party. In addition, we propose an improved protocol that removes this weakness.