A survey of congestion control schemes for multicast video applications

Congestion control for IP multicast on the Internet has been one of the main issues that challenge a rapid deployment of IP multicast. In this article, we survey and discuss the most important congestion control schemes for multicast video applications on the Internet. We start with a discussion of the different elements of a multicast congestion control architecture. A congestion control scheme for multicast video possesses specific requirements for these elements. These requirements are discussed, along with the evaluation criteria for the performance of multicast video. We categorize the schemes we present into end-to-end schemes and router-supported schemes. We start with the end-to-end category and discuss several examples of both single-rate multicast applications and layered multicast applications. For the router-supported category, we first present single-rate schemes that utilize filtering of multicast packets by the routers. Next we discuss receiver-based layered schemes that rely on routers group?flow control of multicast sessions. We evaluate a number of schemes that belong to each of the two categories.     

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.     

基于分层多播的视频传输拥塞控制算法研究

分层多播以多速率方式解决了多播接收者异构性问题,对于提高网络服务质量具有重要意义。本文分析了分层多播传输特性,通过提取MPEG视频流中的I帧、P帧、B帧组成3个帧流,分别放到分层多播的基层、增强层1和增强层2上传输,并在中间节点采用优先级队列机制,提出了一种面向视频流传输的分层多播拥塞控制(VLMCC)算法。仿真实验表明,本文提出的VLMCC算法能够适应视频多播接收者的异构性,大大提高了视频多播传输质量。     

TFMCC versus ASMP: lessons learned from performance evaluation

In this article we present a simulation‐based comparison of one of the best‐known multicast congestion control schemes—TCP‐friendly Multicast Congestion Control (TFMCC)—against our proposed Adaptive Smooth Multicast Protocol (ASMP). ASMP consists of a single‐rate multicast congestion control mechanism which takes advantage of the RTCP Sender (SR) and Receiver Reports (RR) in order to adjust the sender's transmission rate in respect of the network conditions. The innovation in ASMP lays in the ‘smooth’ transmission rate, which is TCP‐friendly and prevents oscillations. We use an integrated simulation environment named Multi‐Evalvid‐RA for the evaluation of the two congestion control schemes. Multi‐Evalvid‐RA provides all the necessary tools to perform simulation studies and assess video quality by using both network‐centric metrics along with video quality measurements. Performance evaluation results show that ASMP is a very efficient solution for rate‐adaptive multimedia applications and a serious competitor to well‐known TFMCC. Copyright © 2012 John Wiley & Sons, Ltd.     

一种新的分层多播拥塞控制协议

提出了一种新的自适应分层多播拥塞控制方案(ALM)。ALM是发送方与接收方共同驱动、由路由器辅助流量控制的拥塞控制方案,通过把发送方的动态分层和接收方的自适应速率调整有机结合,不仅增强了分层多播的适应能力,提高了系统的吞吐量,而且较好地满足了TCP友好性。仿真实验表明,ALM能有效地利用网络带宽,解决网络带宽的异构性问题,并能通过接收端计算TCP友好速率,使接收端达到与TCP流公平竞争网络资源的目的。     

A study of multicast congestion control for UMTS

In this paper, we study the applicability of multicast congestion control over universal mobile telecommunications system (UMTS) networks. We analyze two well‐known multicast congestion control schemes for fixed networks, namely TCP‐friendly multicast congestion control and pragmatic general multicast congestion control. We investigate their behavior when they are employed in UMTS networks and we analyze the problems arose when these mechanisms are applied over the wireless links of the UMTS terrestrial radio‐access network. Additionally, we propose necessary improvements to these legacy schemes and explain the necessity of these modifications. The proposed schemes are implemented in the ns‐2 network simulator and are evaluated under various network conditions and topologies. Finally, we measure the performance of the proposed modified schemes and compare them with the corresponding legacy mechanisms. Copyright © 2009 John Wiley & Sons, Ltd.     

Low-complexity video coding for receiver-driven layered multicast

The “Internet Multicast Backbone,” or MBone, has risen from a small, research curiosity to a large-scale and widely used communications infrastructure. A driving force behind this growth was the development of multipoint audio, video, and shared whiteboard conferencing applications. Because these real-time media are transmitted at a uniform rate to all of the receivers in the network, a source must either run at the bottleneck rate or overload portions of its multicast distribution tree. We overcome this limitation by moving the burden of rate adaptation from the source to the receivers with a scheme we call receiver-driven layered multicast, or RLM. In RLM, a source distributes a hierarchical signal by striping the different layers across multiple multicast groups, and receivers adjust their reception rate by simply joining and leaving multicast groups. We describe a layered video compression algorithm which, when combined with RLM, provides a comprehensive solution for scalable multicast video transmission in heterogeneous networks. In addition to a layered representation, our coder has low complexity (admitting an efficient software implementation) and high loss resilience (admitting robust operation in loosely controlled environments like the Internet). Even with these constraints, our hybrid DCT/wavelet-based coder exhibits good compression performance. It outperforms all publicly available Internet video codecs while maintaining comparable run-time performance. We have implemented our coder in a “real” application-the UCB/LBL videoconferencing tool vic. Unlike previous work on layered video compression and transmission, we have built a fully operational system that is currently being deployed on a very large scale over the MBone     

Transporting real-time video over the Internet: challenges andapproaches

Delivering real-time video over the Internet is an important component of many Internet multimedia applications. Transmission of real-time video has bandwidth, delay, and loss requirements. However the current Internet does not offer any quality of service (QoS) guarantees to video transmission over the Internet. In addition, the heterogeneity of the networks and end systems makes it difficult to multicast Internet video in an efficient and flexible way. Thus, designing protocols and mechanisms for Internet video transmission poses many challenges. In this paper, we take a holistic approach to these challenges and present solutions from both transport and compression perspectives. With the holistic approach, we design a framework for transporting real-time Internet video, which includes two components, namely, congestion control and error control. Specifically congestion control consists of rate control, rate-adaptive encoding, and rate shaping; error control consists of forward error correction (FEC), retransmission error resilience, and error concealment. For the design of each component in the framework, we classify approaches and summarize representative research work. We point out there exists a design space which can be explored by video application designers and suggest that the synergy of both transport and compression could provide good solutions     

因特网上视频多点传输算法研究

本文研究因特网上进行视频多点传输的问题.在分析了资源预约协议和智体反馈控制机制的基础上,基于分层编码技术提出一种新的视频传输算法.文中利用ns-2网络模拟器进行了性能评价,结果表明该算法在保证视频基本服务质量的条件下,具有较好的公平性和可扩放性.     

Feedback control mechanisms for real-time multipoint video services

While existing research shows that reactive congestion control mechanisms are capable of providing high video quality and channel utilization for point-to-point real-time video, there has been relatively little study of the reactive congestion control of point-to-multipoint video, especially in ATM networks. Problems complicating the provision of multipoint, feedback-based real-time video service include: (1) implosion of feedback returning to the source as the number of multicast destinations increases and (2) variance in the amount of available bandwidth on different branches in the multipoint connection. A new service architecture is proposed for real-time multicast video, and two multipoint feedback mechanisms to support this service are introduced and studied. The mechanisms support a minimum bandwidth guarantee and the best effort support of video traffic exceeding the minimum rate. They both rely on adaptive, multilayered coding at the video source and closed-loop feedback from the network in order to control both the high and low priority video generation rates of the video encoder. Simulation results show that the studied feedback mechanisms provide, at the minimum, a quality of video comparable to a constant bit rate (CBR) connection reserving the same amount of bandwidth. When unutilized network bandwidth becomes available, the mechanisms are capable of exploiting it to dynamically improve video quality beyond the minimum guaranteed level     

FLID-DL: congestion control for layered multicast

We describe fair layered increase/decrease with dynamic layering (FLID-DL): a new multirate congestion control algorithm for layered multicast sessions. FLID-DL generalizes the receiver-driven layered congestion control protocol (RLC) introduced by Vicisano et al. (Proc. IEEE INFOCOM, San Francisco, CA, , p.996-1003, Mar. 1998)ameliorating the problems associated with large Internet group management protocol (IGMP) leave latencies and abrupt rate increases. Like RLC, FLID-DL, is a scalable, receiver-driven congestion control mechanism in which receivers add layers at sender-initiated synchronization points and leave layers when they experience congestion. FLID-DL congestion control coexists with transmission control protocol (TCP) flows as well as other FLID-DL sessions and supports general rates on the different multicast layers. We demonstrate via simulations that our congestion control scheme exhibits better fairness properties and provides better throughput than previous methods. A key contribution that enables FLID-DL and may be useful elsewhere is dynamic layering (DL), which mitigates the negative impact of long IGMP leave latencies and eliminates the need for probe intervals present in RLC. We use DL to respond to congestion much faster than IGMP leave operations, which have proven to be a bottleneck in practice for prior work.     

Combined Connection Admission Control and Packet Transmission Scheduling for Mobile Internet Services

Mobile Internet access is expected to be the most popular communication service in the near future. In this paper, we investigate radio resource management for mobile Internet multimedia systems that use the orthogonal frequency division multiple access and adopt the adaptive modulation and coding technique. It is assumed that real-time (RT) service such as streaming video and best-effort (BE) services such as file transfer protocol and hypertext transfer protocol coexist in the systems. We suggest two levels of radio resource management schemes: the connection admission control (CAC) scheme at the first level and the packet transmission scheduler at the second level. The proposed scheduler does not assign higher priority to RT packets over BE packets unconditionally. Instead, only the RT packets that are close to the deadline are given higher priority. Therefore, the performance of BE services is improved at the cost of RT services. To control the performance degradation in RT services within an acceptable level, the CAC algorithm functions as a congestion controller. The combined effects of the proposed CAC and packet scheduling by using the cross-layer simulation that covers from the physical layer to the Internet application layer are evaluated. The numerical results show that the proposed schemes greatly improve the performance of BE services while maintaining the quality of video service at an acceptable level.     

Fine-grained Layered multicast with STAIR

Traditional approaches to receiver-driven layered multicast have advocated the benefits of cumulative layering, which can enable coarse-grained congestion control that complies with TCP-friendliness equations over large time scales. In this paper, we quantify the costs and benefits of using noncumulative layering and present a new, scalable multicast congestion control scheme called STAIR that embodies this approach. Our first main contribution is a set of performance criteria on which we base a comparative evaluation of layered multicast schemes. In contrast to the conventional wisdom, we demonstrate that fine-grained rate adjustment can be achieved with only modest increases in the number of layers, aggregate bandwidth consumption and control traffic. The STAIR protocol that we subsequently define and evaluate is a multiple rate congestion control scheme that provides a fine-grained approximation to the behavior of TCP additive increase/multiplicative decrease (AIMD) on a per-receiver basis.     

基于网络控制的分层多点播送速率控制机制研究

本文在视频分层编码及分层传输协议的基础上，将基于网络控制与接收端控制机制相结合，提出一种新的分层多点播送速率控制机制。文中给出了该机制的拥塞检测、流行度权衡和数据层增加和丢弃算法。实验结果表明，该速率控制机制通史对拥塞做也快速响应，并在较好利用宽带的基础上保证高流行度的会话流具有较好的服务质量。     

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     

A cross-layer framework for multi-layer-video multicast with QoS requirements in multirate wireless networks

We propose a cross-layer framework for efficient multi-layer-video multicast with rate adaptation and quality-of-service (QoS) requirements in multirate wireless networks. We employ time division multiple access at the physical layer to transmit different video layers' data. The multicast sender then dynamically regulates the transmission rate and time-slot allocation based on the channel state information (CSI) and loss QoS requirements imposed by upper protocol layers. Under our proposed cross-layer framework, we first design a rate adaptation algorithm to fulfill the diverse loss QoS requirements for all video layers while achieving high multicast throughput. We then develop a time-slot allocation scheme which synchronizes data transmission across different video layers. Also conducted are simulation results to validate and evaluate our designed adaptive multicasting schemes under the proposed cross-layer framework.     

基于可用带宽测量的分层组播拥塞控制机制

针对现有的组播拥塞控制机制对接收端可用带宽估计精度较低的问题,提出了一种基于可用带宽测量的分层组播拥塞控制机制ABM-LMCC.在分析了现有可用带宽估计方法不足的基础上,提出一种适用于组播的可用带宽测量算法,并设计了分层组播拥塞控制机制的具体操作规程.通过调节组播数据包的发送间隔,使其呈现降速率的指数分布,从而实现各接收端对可用带宽的准确测量,并根据其测量值迅速调节期望速率,从而达到组播拥塞控制的目的.仿真表明,ABM-LMCC能够有效避免拥塞,提高链路利用率,显著降低丢包率,具有良好的响应性、稳定性.     

一种缓冲区驱动的自适应视频多播系统

利用基于控制理论的自适应传输机制,提出一种接收方缓冲区驱动的自适应视频多播系统,实现了在不可预约带宽网络环境中的视频自适应传输.仿真结果验证了方法的有效性,可适用于不同的拥塞控制算法和编码算法,扩展性良好.