TCP-Friendly Rate Control (TFRC) for bursty media flows

TCP-Friendly Rate Control (TFRC) was originally designed for multimedia streaming applications where continuous data was available at the sender. However, TFRC is not well-suited to the variable rate traffic presented by many modern adaptive media codecs. One way to counter this deficiency would be for the sender to continue to transmit at the media rate during periods of silence, known as padding. This use of padding can ensure acceptable application performance. However, it also degrades network performance, and decreases the usefulness of TFRC congestion control. Recent standardisation has resulted in a new revised TFRC specification. This paper describes candidate methods that were evaluated as a part of this revision and presents the first analysis of the new TFRC specification including a comparison this with the proposed Faster Restart method. It evaluates behaviour both in terms of the application performance benefit and the implications on other network traffic that share an Internet bottleneck and shows that the new methods improve the performance of bursty media. Although Faster Restart allowed TFRC to better support bursty applications, the additional gain was determined to be small when combined with the revised TFRC specification. Finally, revised TFRC is shown to remove the former incentive for padding, substantially improving the performance of other network traffic sharing a congested network.     

Packet loss in peer-to-peer video streaming over the Internet

Peer-to-peer streaming has recently gained attention as an effective solution to support large scale media streaming applications over the Internet. One of the main challenges of peer-to-peer video streaming is the cumulative impact of the Internet packet loss due to the decoding dependency of the compressed video frames. In this paper we study the impact of the Internet packet loss on the performance of peer-to-peer video streaming systems, and analyze the efficiency of various packet loss recovery policies in such systems. Our analytical and simulation results show how the Internet packet loss can affect the performance of peer- to-peer video streaming systems and how different packet loss recovery policies can be effective for such systems. Our analysis results give us some insights that can be used in designing efficient peer-to-peer video streaming systems.     

端到端的流媒体传输控制技术研究综述

随着多媒体应用的发展,因特网上的流媒体传输技术已成为研究热点。在因特网上传输音频或视频流需要有带宽、延迟、丢包率等诸多的QoS要求,但当前的因特网并不提供任何QoS保证,这对流媒体在因特网上的传输提出巨大挑战。该文从端到端的传输控制技术角度入手,给出了一个因特网上流媒体传输的总体框架,然后依此框架为线索,对流媒体传输所必需的协议栈、拥塞控制、自适应速率编码、速率整形、差错控制等技术的研究进展进行了概括总结并进行了对比,同时提出了进一步的研究建议。     

Media- and TCP-friendly congestion control for scalable video streams

This paper presents a media- and TCP-friendly rate-based congestion control algorithm (MTFRCC) for scalable video streaming in the Internet. The algorithm integrates two new techniques: i) a utility-based model using the rate-distortion function as the application utility measure for optimizing the overall video quality; and ii) a two-timescale approach of rate averages (long-term and short-term) to satisfy both media and TCP-friendliness. We evaluate our algorithm through simulation and compare the results against the TCP-friendly rate control (TFRC) algorithm. For assessment, we consider five criteria: TCP fairness, responsiveness, aggressiveness, overall video quality, and smoothness of the resulting bit rate. Our simulation results manifest that MTFRCC performs better than TFRC for various congestion levels, including an improvement of the overall video quality.     

A new fast friendly window-based congestion control for real-time streaming media transmission

1 Introduction In the current Internet, not all applications use TCP and they do not follow the same concept of fairly sharing the available bandwidth. The rapid growing of real-time streaming media applications will bring much UDP traffic without integrating TCP compatible congestion control mechanism into Internet. It threats the quality of service (QoS) of real-time applications and the stability of the current Internet. For this reason, it is desirable to define appropriate rate rule…     

无线网络中基于流媒体传输的自适应TFRC机制

在无线网络高误码率的环境下, 经典TFRC机制会将无线误码丢包误认为拥塞丢包, 导致吞吐量过度降低. 针对无线网络实时流媒体业务的传输控制问题, 提出了一种改进型动态自适应TFRC机制(Adaptive-TFRC). 它在接收端利用丢包区分参数来真实反映网络的状态(即拥塞或者误码), 然后反馈至发送端, 同时对经典TFRC机制的吞吐量模型公式进行改进, 最终能够根据实时网络条件动态自适应地调节传输速率. 仿真结果表明, Adaptive-TFRC机制能够有效地提高网络吞吐量, 降低实时业务流的延时抖动, 同时能够进一步改善TCP业务的友好性传输, 从而保证无线网络实时流媒体的服务质量.     

一种报文大小可变的TFRC改进算法

基于Internet的实时多媒体数据传输是一种报文发送速率固定,报文大小变化的应用。该文分析了这类应用对TFRC的影响,通过对TFRC协议的扩展,提出了一种支持报文大小可变应用的改进TFRC拥塞控制算法。这种算法在接收方采用了对报文数量进行加权的方法来计算丢失事件率以支持报文大小变化的应用。同时在网络仿真器ns2中实现了这种改进算法。仿真实验表明：这种改进算法能够支持报文大小变化,报文发送速率固定的应用,并且具有TCP友好性,与TCP相比具有较平缓的流量抖动。     

Multiple TFRC Connections Based Rate Control for Wireless Networks

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 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. This approach, also known as TCP friendly rate control (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 bulk of packet loss is due to error at the physical layer. In this paper, 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 1$times$RTT CDMA wireless data network, and and video streaming simulations using traces from the actual experiments, are carried out to validate, and characterize the performance of our proposed approach.     

基于延迟抖动分析的TCP友好拥塞控制算法

基于因特网的以UDP为传输协议的实时多媒体数据传输需要在保证实时性和可靠性的基础上,能够与因特网其他服务所使用的TCP协议公平共享有限的带宽。本文采用基于实时传输协议（RTP）和实时传输控制协议（RTCP）的反馈拥塞控制算法,提出一种简单的拥塞控制机制,使UDP数据流能与TCP数据流和平共处;研究了基于速率控制的TCP友好拥塞控制策略-TFRC,分析了其基本机制和关键问题;提出利用延迟抖动作为潜在拥塞信号来改进TFRC的速率控制机制,以适应实时业务低抖动的要求,并通过NS仿真验证了改进的TFRC算法对实时业务的良好性能。     

Media streaming via TFRC: An analytical study of the impact of TFRC on user-perceived media quality

TCP-Friendly Rate Control (TFRC) is being adopted in Internet standards for congestion control of streaming media applications. In this paper, we consider the transmission of prerecorded media from a server to a client by using TFRC, and analytically study the impact of TFRC on user-perceived media quality, which is roughly measured by calculating the rebuffering probability. A rebuffering probability is defined to be the probability that the total duration of all rebuffering events experienced by a user is longer than a certain threshold. Several approaches are presented to help an application determine an appropriate initial buffering delay and media playback rate in order to achieve a certain rebuffering probability under a given network condition. First, we derive a closed-form expression to approximate the average TFRC sending rate, which could be used as the maximum allowed playback rate of a media stream. Second, we develop a queueing model for a TFRC client buffer with the traffic described by a Markov-Renewal-Modulated Deterministic Process (MRMDP), which captures the fundamental behavior of TFRC that predicts the immediate future TCP sending rate based on the history of past loss intervals. We present a closed-form solution and a more accurate iterative method to solve the queueing model and calculate the rebuffering probability.     

TFRC拥塞控制策略的研究和改进

分析了目前在实时业务中常用的拥塞控制机制TFRC.针对实时流媒体要求低延迟和低抖动的特点,提出将单程传输延迟的抖动作为反馈信号来调节发送端的发送速率,并针对抖动阈值的选定采用了自适应的调节方法.仿真结果表明改进后的算法减小了延迟抖动,提高了相关的服务质量.     

支持最少速率保证的UDP拥塞控制机制

虽然目前 TCP流量在整个 Internet流量中占有主要地位 ,但随着网络带宽的不断升级改造 ,基于 UDP的音频、视频等实时多媒体流量日益增加 ,而这些实时流量一般都需要一定的带宽保证 ,同时又具有 TCP友好的端端拥塞控制机制 .从端主机和网关队列机制两方面着手 ,提出了一种支持 IETF定义的可控负载服务机制 ,其实现原理是在端主机方配置基于令牌桶的自适应的支持标记的速率调节机制 ,在网关采用加强的 RED(随机早期检测 )队列管理机制对不同的流量进行相应的处理 ,然后在 NS仿真环境下对其公平性、带宽使用效率等方面进行实验 ,证明了该机制的有效性和可行性     

基于模糊逻辑的Internet视频流拥塞控制

为了提高视频传输质量,在Internet上对视频流进行拥塞控制,即利用包发送和接收间隔时间（IPGs）代替丢包率作为拥塞指示,采用模糊逻辑拥塞控制策略（FLC）调整视频发送速率并用遗传算法优化模糊控制规则,提高了拥塞控制性能。仿真结果表明,与TFRC和RAP拥塞控制相比,由于FLC发送速率更平滑、带宽利用率更高,从而减少了丢包,提高了视频传输质量;另外,FLC能够与竞争的TCP流公平地分享带宽并对路由器缓冲区大小保持了很好的鲁棒性。     

Design and implementation of MP3-music on demand system using streaming technology

The success of the Internet and the use of broadband in homes have caused a gradual shift in traffic on the Internet from data to multimedia communication. Multimedia applications typically include a large quantity of video/audio information. Streaming technology is normally adopted to handle the transmission of multimedia traffic and thus reduce the buffer requirement on the client side and the service request/response time. This work focuses on the transmission of MP3 music which has a constant bit rate characteristic. The design of both the server side and the client side of the MP3-music on demand (MoD) system with streaming technology, is considered to meet the quality of service (QoS) requirements of MP3 music. A stream buffering technique is used and an adaptive rate control mechanism is applied in combination with a client feedback packet to prevent stream buffer overflow or underflow on the client side, and thereby accommodate the network delay, jitter, and timing deviation between the server machine and the client host. A server self-timing revision scheme is used to reduce the network overhead of the feedback mechanism. The adaptive rate control mechanism is developed and verified using a computer simulation. Finally, for completeness a MoD system is constructed with a low-cost embedded network system to which an Altera FPGA is applied to provide cut-through data movement and an adaptive rate control mechanism is realized to evaluate QoS.     

基于端系统的视频流组播速率控制方法

针对目前Internet未对多媒体应用提供QoS保障的问题,分析了视频流组播的难点,提出了一种基于缓冲区管理的网络自适应组播发送速率控制方法.该机制可以合理控制服务器的发送速率,既能自适应网络状况的变化,又满足流媒体实时播放的需求.实验结果表明,该机制通过控制发送缓冲区占有率,降低了分组丢包率,提高了终端的接收质量,具有良好的实用价值.     

IP网络实时视频流的传输控制算法AVTC的研究

随着实时音视频流和多播应用的发展，越来越多的非TCP流将出现在IP网上，它们可能和TCP流不公平地竞争网络带宽．针对TFRC算法存在的一些缺陷、Rtt的计算以及具体的实现等问题进行了讨论，并提出了IP网的自适应实时视频传输控制算法AVTC(adaptive video transmission control)．在AVTC算法中，发送端通过从接收端得到当前网络的状态信息，从而估计得到比较合适的发送速率，并动态地调整发送端的发送速率以适合当前网络的状况．AVTC算法满足实时视频流传输的实时性要求以及与TCP流公平地分享带宽的TCP友好性要求．     

Empirical evaluation of the congestion responsiveness of RealPlayer video streams

Increasingly popular commercial streaming media applications over the Internet often use UDP as the underlying transmission protocol for performance reasons. Hand-in-hand with the increase in streaming media comes the impending threat of unresponsive UDP traffic, often cited as the major threat to the stability of the Internet. Unfortunately, there are few empirical studies that analyze the responsiveness, or lack of it, of commercial streaming media applications. In this work, we evaluate the responsiveness of RealNetworks’ RealVideo over UDP by measuring the performance of numerous streaming video clips selected from a variety of RealServers on the Internet, analyze the TCP-Friendliness of the UDP streams and correlate the results with network and application layer statistics. We find that most RealVideo UDP streams respond to Internet congestion by reducing the application layer encoding rate, and streams with a minimum encoding rate less than the fair share of the capacity often achieve a TCP-Friendly rate. In addition, our results suggest that a reason streaming applications choose not to use TCP is that the TCP API hides network information, such as loss rate and round-trip time, making it difficult to estimate the available capacity for effective media scaling.     

基于MPC的音视频同步码率自适应测试

随着信息化的发展,音视频流媒体技术应用面越来越广,为了使得音视频流媒体技术尤其是在直播方面拥有更好的性能,得到更多用户的好评,采用在原本HTTP的动态自适应流标准的视频流媒体架构下引入MPC控制算法并将MPC模型预测控制与码率自适应算法相结合的方法,进行对AAC优化、确定预测模型、测试音视频同步的影响因素以及PSNR-Y分量、测试切片时长与跳帧时延,计算最终的QoE用户评价指标来进一步检测音视频流媒体技术的优劣。经实验仿真测试可知,相比前人的相关算法,在不同直播场景下以及不同网络环境下均有更加良好的QoE值,平均QoE用户评价指标明显更高,为1237.2826。综上分析可知,MPC的音视频同步码率自适应算法各项性能最好。     

分组交换网络中拥塞计费策略及拥塞率的计算

随着网络视频和流媒体等应用的逐渐普及,用户对带宽的需求日益增加.基于多TCP流的应用大量挤占网络带宽,影响网页浏览和电子邮件等低数据量应用的体验.如何在不过度限制高带宽需求应用的前提下,有效改善低数据量应用的体验,成为目前分组交换网络亟待解决的问题.拥塞计费是该问题的一种可行的解决方法.其基本思想是在网络发生拥塞时按照用户所造成拥塞量的多少来对用户进行收费.本文介绍了基于拥塞量收费的网络拥塞计费策略,建立了多拥塞点网络的数学模型,给出了计算拥塞率的方法,并对其算法复杂度进行了分析.最后,本文通过网络仿真对所提出的模型进行了验证.     

Adaptive efficient video transmission over the Internet based on congestion control and RS coding

An approach based on adaptive congestion control and adaptive error recovery with RS (Reed-Solomon) coding method is presented for efficient video transmission over the Internet. Featured by weighted moving average rate control and TCP-friendliness, AVSP, a novel adaptive video streaming protocol, is designed with adjustable rate control parameters so as to respond quickly to the QoS status fluctuation during video transmission over the Internet. Combined with congestion control policy, an adaptive RS coding error recovery scheme with variable parameters is presented to enhance the robustness of MPEG video transmission over the Internet with restriction to the total system bandwidth .