End-to-end TCP-friendly streaming protocol and bit allocation for scalable video over wireless Internet

With the convergence of wired-line Internet and mobile wireless networks, as well as the tremendous demand on video applications in mobile wireless Internet, it is essential to an design effective video streaming protocol and resource allocation scheme for video delivery over wireless Internet. Taking both network conditions in the Internet and wireless networks into account, in this paper, we first propose an end-to-end transmission control protocol (TCP)-friendly multimedia streaming protocol for wireless Internet, namely WMSTFP, where only the last hop is wireless. WMSTFP can effectively differentiate erroneous packet losses from congestive losses and filter out the abnormal round-trip time values caused by the highly varying wireless environment. As a result, WMSTFP can achieve higher throughput in wireless Internet and can perform rate adjustment in a smooth and TCP-friendly manner. Based upon WMSTFP, we then propose a novel loss pattern differentiated bit allocation scheme, while applying unequal loss protection for scalable video streaming over wireless Internet. Specifically, a rate-distortion-based bit allocation scheme which considers both the wired and the wireless network status is proposed to minimize the expected end-to-end distortion. The global optimal solution for the bit allocation scheme is obtained by a local search algorithm taking the characteristics of the progressive fine granularity scalable video into account. Analytical and simulation results demonstrate the effectiveness of our proposed schemes.     

Enhancement of TCP over wired/wireless networks with packet loss classifiers inferred by supervised learning

TCP is suboptimal in heterogeneous wired/wireless networks because it reacts in the same way to losses due to congestion and losses due to link errors. In this paper, we propose to improve TCP performance in wired/wireless networks by endowing it with a classifier that can distinguish packet loss causes. In contrast to other proposals we do not change TCP’s congestion control nor TCP’s error recovery. A packet loss whose cause is classified as link error will simply be ignored by TCP’s congestion control and recovered as usual, while a packet loss classified as congestion loss will trigger both mechanisms as usual. To build our classification algorithm, a database of pre-classified losses is gathered by simulating a large set of random network conditions, and classification models are automatically built from this database by using supervised learning methods. Several learning algorithms are compared for this task. Our simulations of different scenarios show that adding such a classifier to TCP can improve the throughput of TCP substantially in wired/wireless networks without compromizing TCP-friendliness in both wired and wireless environments.     

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.     

TCP-Jersey for wireless IP communications

Improving the performance of the transmission control protocol (TCP) in wireless Internet protocol (IP) communications has been an active research area. The performance degradation of TCP in wireless and wired-wireless hybrid networks is mainly due to its lack of the ability to differentiate the packet losses caused by network congestions from the losses caused by wireless link errors. In this paper, we propose a new TCP scheme, called TCP-Jersey, which is capable of distinguishing the wireless packet losses from the congestion packet losses, and reacting accordingly. TCP-Jersey consists of two key components, the available bandwidth estimation (ABE) algorithm and the congestion warning (CW) router configuration. ABE is a TCP sender side addition that continuously estimates the bandwidth available to the connection and guides the sender to adjust its transmission rate when the network becomes congested. CW is a configuration of network routers such that routers alert end stations by marking all packets when there is a sign of an incipient congestion. The marking of packets by the CW configured routers helps the sender of the TCP connection to effectively differentiate packet losses caused by network congestion from those caused by wireless link errors. This paper describes the design of TCP-Jersey, and presents results from experiments using the NS-2 network simulator. Results from simulations show that in a congestion free network with 1% of random wireless packet loss rate, TCP-Jersey achieves 17% and 85% improvements in goodput over TCP-Westwood and TCP-Reno, respectively; in a congested network where TCP flow competes with VoIP flows, with 1% of random wireless packet loss rate, TCP-Jersey achieves 9% and 76% improvements in goodput over TCP-Westwood and TCP-Reno, respectively. Our experiments of multiple TCP flows show that TCP-Jersey maintains the fair and friendly behavior with respect to other TCP flows.     

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。     

JTCP: jitter-based TCP for heterogeneous wireless networks

Transmission control protocol (TCP), a widely used transport protocol performs well over the traditional network which is constructed by purely wired links. As wireless access networks are growing rapidly, the wired/wireless mixed internetwork, a heterogeneous environment will get wide deployment in the next-generation ALL-IP wireless networks. TCP which detects the losses as congestion events could not suit the heterogeneous network in which the losses will be introduced by higher bit-error rates or handoffs. There exist some unsolved challenges for applying TCP over wireless links. End-to-end congestion control and fairness issues are two significant factors. To satisfy these two criteria, we propose a jitter-based scheme to adapt sending rates to the packet losses and jitter ratios. The experiment results show that our jitter-based TCP (JTCP) conducts good performance over the heterogeneous network.     

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.     

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。     

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.     

Adaptive Rate Optimal Design for Wireless Heterogeneous Networks: Jointly Congestion Discrimination and Queue Updating

In this paper, we propose a novel rate adaptive optimization scheme for streaming media transmission over wireless heterogeneous IP networks. In the proposed adaptive scheme, through the analysis of the packet loss characteristics in wireless channel, we develop the relationship between the packet loss rates and the packet sizes. Furthermore, the scheme detects the nature of packet losses by sending large and small packets alternately, and then adopts an adaptive rate optimization strategy to decrease the network congestion and increase the network throughput. Using congestion discrimination and updating factor, the scheme can adapt to the changes of network states quickly and improve delivery quality of wireless multimedia streaming. Simulation results show that, in comparisons to the existing rate optimization algorithms, our proposed scheme offers significantly improved performance in terms of throughput and network congestion, especially when the channel quality is poor in different network topology environments.     

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.     

The impact of multihop wireless channel on TCP performance

This paper studies TCP performance in a stationary multihop wireless network using IEEE 802.11 for channel access control. We first show that, given a specific network topology and flow patterns, there exists an optimal window size W* at which TCP achieves the highest throughput via maximum spatial reuse of the shared wireless channel. However, TCP grows its window size much larger than W* leading to throughput reduction. We then explain the TCP throughput decrease using our observations and analysis of the packet loss in an overloaded multihop wireless network. We find out that the network overload is typically first signified by packet drops due to wireless link-layer contention, rather than buffer overflow-induced losses observed in the wired Internet. As the offered load increases, the probability of packet drops due to link contention also increases, and eventually saturates. Unfortunately the link-layer drop probability is insufficient to keep the TCP window size around W'*. We model and analyze the link contention behavior, based on which we propose link RED that fine-tunes the link-layer packet dropping probability to stabilize the TCP window size around W*. We further devise adaptive pacing to better coordinate channel access along the packet forwarding path. Our simulations demonstrate 5 to 30 percent improvement of TCP throughput using the proposed two techniques.     

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.     

Enhancing Fairness and Throughput of TCP in Heterogeneous Wireless Access Networks

Most of the recent research on TCP over heterogeneous wireless networks has concentrated on differentiating between packet drops caused by congestion and link errors, to avoid significant throughput degradations due to the TCP sending window being frequently shut down, in response to packet losses caused not by congestion but by transmission errors over wireless links. However, TCP also exhibits inherent unfairness toward connections with long round-trip times or traversing multiple congested routers. This problem is aggravated by the difference of bit-error rates between wired and wireless links in heterogeneous wireless networks. In this paper, we apply the TCP Bandwidth Allocation (TBA) algorithm, which we have proposed previously, to improve TCP fairness over heterogeneous wireless networks with combined wireless and wireline links. To inform the sender when congestion occurs, we propose to apply Wireless Explicit Congestion Notification (WECN). By controlling the TCP window behavior with TBA and WECN, congestion control and error-loss recovery are effectively separated. Further enhancement is also incorporated to smooth traffic bursts. Simulation results show that not only can the combined TBA and WECN mechanism improve TCP fairness, but it can maintain good throughput performance in the presence of wireless losses as well. A salient feature of TBA is that its main functions are implemented in the access node, thus simplifying the sender-side implementation.     

无线Mesh网中基于网络编码的多路径TCP研究

在无线多跳网络中,本地重传和网络编码已经被成功地应用到多路径技术上以增加吞吐量并减少丢包。然而,在提高UDP传输性能的同时,也产生了数据包重排序和延迟等副作用,严重影响了TCP性能。针对此问题,主要提出一种基于网络编码的多路径传输方案NC-MPTCP,即在无线mesh网络的多条路径中引入网络编码、执行拥塞控制以及使用一个基于信用的方法控制节点的传输速率,提高网络的吞吐量以及增加网络传输的可靠性。该方案使用一个简单的算法,评估丢包率以及发送线性组合数据包的速率,用来降低目的节点的数据包解码延迟和防止TCP的超时重传。仿真结果表明设计的NC-MPTCP有效。     

Congestion control for multimedia applications in the wireless internet

This paper provides a parallel review of two important issues for the next‐generation multimedia networking. Firstly, the emerging multimedia applications require a fresh approach to congestion control in the Internet. Currently, congestion control is performed by TCP; it is optimised for data traffic flows, which are inherently elastic. Audio and video traffic do not find the sudden rate fluctuations imposed by the TCP multiplicative‐decrease control algorithm optimal. The second important issue is the mobility support for multimedia applications. Wireless networks are characterized by a substantial packet loss due to the imperfection of the radio medium. This increased packet loss disturbs the foundation of TCP's loss‐based congestion control. This paper contributes to the ongoing discussion about the Internet congestion control by providing a parallel analysis of these two issues. The paper describes the main challenges, design guidelines, and existing proposals for the Internet congestion control, optimised for the multimedia traffic in the wireless network environment. Copyright © 2004 John Wiley & Sons, Ltd.     

A new ATM adaptation layer for TCP/IP over wireless ATM networks

This paper describes the design and performance of a new ATM adaptation layer protocol (AAL‐T) for improving TCP performance over wireless ATM networks. The wireless links are characterized by higher error rates and burstier error patterns in comparison with the fiber links for which ATM was introduced in the beginning. Since the low performance of TCP over wireless ATM networks is mainly due to the fact that TCP always responds to all packet losses by congestion control, the key idea in the design is to push the error control portion of TCP to the AAL layer so that TCP is only responsible for congestion control. The AAL‐T is based on a novel and reliable ARQ mechanism to support quality‐critical TCP traffic over wireless ATM networks. The proposed AAL protocol has been validated using the OPNET tool with the simulated wireless ATM network. The simulation results show that the AAL‐T provides higher throughput for TCP over wireless ATM networks compared to the existing approach of TCP with AAL 5. This revised version was published online in July 2006 with corrections to the Cover Date.     

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).     

ARC: the analytical rate control scheme for real-time traffic in wireless networks

Next-generation wireless Internet (NGWI) is expected to provide a wide range of services including real-time multimedia to mobile users. However, the real-time multimedia traffic transport requires rate control deployment to protect shared Internet from unfairness and further congestion collapse. The transmission rate control method must also achieve high throughput and satisfy multimedia requirements such as delay or jitter bound. However, the existing solutions are mostly for the wired Internet, and hence, they do not address the challenges in the wireless environments which are characterized by high bit error rates. In this paper, a new analytical rate control (ARC) protocol for real-time multimedia traffic over wireless networks is presented. It is intended to achieve high throughput and multimedia support for real-time traffic flows while preserving fairness to the TCP sources sharing the same wired link resources. Based on the end-to-end path model, the desired behavior of a TCP source over lossy links is captured via renewal theory. The resulting asymptotic throughput equation is designated as the driving equation for the proposed rate control method. Performance evaluation via simulation experiments reveals that ARC achieves high throughput and meets multimedia traffic expectations without violating good citizenship rules for the shared Internet.     

TCP over optical burst-switched networks with controlled burst retransmission

For optical burst-switched (OBS) networks in which TCP is implemented at a higher layer, the loss of bursts can lead to serious degradation of TCP performance. Due to the bufferless nature of OBS, random burst losses may occur, even at low traffic loads. Consequently, these random burst losses may be mistakenly interpreted by the TCP layer as congestion in the network. The TCP sender will then trigger congestion control mechanisms, thereby reducing TCP throughput unnecessarily. In this paper, we introduce a controlled retransmission scheme in which the bursts lost due to contention in the OBS network are retransmitted at the OBS layer. The OBS retransmission scheme can reduce the burst loss probability in the OBS core network. Also, the OBS retransmission scheme can reduce the probability that the TCP layer falsely detects congestion, thereby improving the TCP throughput. We develop an analytical model for evaluating the burst loss probability in an OBS network that uses a retransmission scheme, and we also analyze TCP throughput when the OBS layer implements burst retransmission. We develop a simulation model to validate the analytical results. Simulation and analytical results show that an OBS layer with controlled burst retransmission provides up to two to three orders of magnitude improvement in TCP throughput over an OBS layer without burst retransmission. This significant improvement is primarily because the TCP layer triggers fewer time-outs when the OBS retransmission scheme is used.