IMS智能终端的QoS技术

文章通过对全球IP多媒体子系统(IMS)终端服务质量(QoS)研究情况的调查、分析,提出了IMS终端QoS控制的实现方法.文章基于区分服务的QoS策略,设计了一套基于终端设备的QoS控制机制,该机制按服务优先级进行流量调节,对包进行缓存和调度输出实现多媒体通信.     

改善Internet视频流QoS的途径和方法

在当前Internet的尽力而为的服务模式下，网络拥塞和分组丢失不可避免，视频流必须使用有效的拥塞控制和差错控制来改善性能。本文分析了：Internet视频流QoS影响因素，提出了两种QoS解决方案：基于终端和基于网络。本文着重讨论了基于终端的QoS解决方案，在目前Internet的环境下，基于终端的QoS解决方案更具可行性。     

改善Internet视频流QoS的途径和方法

在当前Internet的尽力而为的服务模式下，网络拥塞和分组丢失不可避免，视频流必须使用有效的拥塞控制和差错控制来改善性能，本文分析了Internet视频流QoS影响因素，提出了两种QoS解决方案，基于终端和基于网络，并着重讨论了基于终端的QoS解决方案，在目前的环境下，基于终端的QoS解决方案更具可行性。     

基于流媒体的实时网络传输系统中若干问题研究

流媒体在Internet上的实时传输常因其要求高带宽、低延迟而造成网络拥塞。探讨了基于Internet的实时流媒体中视频流的QoS控制策略,并主要论述了基于速率的网络拥塞控制方法并给出一种视频流编码速率调整算法。这些控制技术应用于终端系统并不需要路由器和网络QoS支持,可以较好地提高视频质量。     

基于Internet的实时视频流的应用层QoS控制策略

本文分析了Internet上实时视频传输的特点，提出了基于Internet的实时视频流的应用层QoS控制策略，主要包括拥塞控制策略和错误控制策略以及相应的控制技术。这些控制技术应用于终端系统并不需要路由器和网络的QoS支持，可以最大限度地提高视频质量。     

基于RSVP的QoS参数控制报文设计与实现

本文提出一种高速网络服务质量(QoS)参数的设计与实现方法.该方法在Internet标准RFC2205的基础上,根据不同的多媒体应用类型和服务质量需求,首先对服务质量进行量化和分类,然后设计和定义基于RSVP(Reservation Protocol)的、可供路由器和端系统识别的各类服务质量控制报文,并设计和实现一种利用这些报文进行QoS控制的方法.这些QoS参数控制报文的提出使得RSVP协议在广域网上的实现成为可能,从而提高Internet传送多媒体应用的能力.     

QoS体系中的主动队列管理算法研究综述

随着Internet的迅速发展，服务质量(QoS)正成为当前研究的热点之一。为了达到QoS的性能指标，拥塞控制作为一个很重要的方面在发挥着作用。主动队列管理是实现拥塞控制的重要手段之一，长期以来一直受到广泛的关注，基于不同理论的各种主动队列管理的算法也随之涌现。这些队列管理算法在一定程度上完成网络拥塞控制的任务，但是也不同程度地在公平性、可扩展性以及算法的复杂度上存在缺陷。本文通过对目前几种主要队列管理算法的实现原理的分析，考察了这些队列管理算法的优点和其可能存在的一些问题，而这些可能存在的问题也是下一步研究的起点。     

流量/拥塞控制技术研究

流量/拥塞控制的基本目的是以分布处理的方式有效地控制结点间的数据流,从而避免网络中出现拥塞。拥塞控制相应的控制策略称为拥塞控制算法(协议)。简述了Internet上基于TCP/IP的拥塞控制机制,分析和比较了TCP/IP上具体实现算法的稳定性,讨论了TCP/IP拥塞控制所面临的问题。     

基于带宽预测的多媒体通信路由拥塞节点调度方法

为提高多媒体通信吞吐量,降低端与端传输延时,开展基于带宽预测的多媒体通信路由拥塞节点调度方法设计研究。将TCP/IP协议作为传输指令,建立多媒体通信路由节点通信模型;根据拥塞情况,预测通信过程中的链路带宽,计算通信路由节点有效传输量;引进QOS协商机制,控制节点流量,并结合不同节点的状态量,进行调度设计。实验证明,提出的方法有效提升多媒体通信路由吞吐量,降低通信过程延时,进一步提高通信数据传输速率。     

一种基于代理的视频点播系统QoS管理模型

随着实时多媒体应用在Internet上的增多,对其进行有力的QoS保证是实现多媒体通信的前提和基础.本文就当前国内外对于QoS的研究现状进行了分析,针对一种特定的多媒体应用--视频点播业务设计了一种基于代理的QoS管理模型,并对其功能和工作过程进行了详细阐述.该模型采用层次化结构,分成应用接入层、QoS管理层和资源管理层,将资源管理功能从QoS管理层中分离出来,单独设置资源管理层,这样可以更加方便地进行功能实现和扩展.     

Transport of TCP/IP traffic over assured forwarding IP-differentiated services

The Internet is facing a twofold challenge: to increase network capacity in order to accommodate a steadily increasing number of users; to guarantee the quality of service for existing applications and for new multimedia applications requiring real-time network response. In order to meet these requirements, IETF is currently defining the differentiated service (DiffServ) architecture, which should offer a simple and scalable platform to guarantee differentiated QoS in the Internet. In the DiffServ domain, the assured forwarding service is designed to provide data applications with acceptable performance, overcoming the limits of the Internet's current best-effort service. Since data applications mostly rely on the TCP transport protocol, it is important to examine the interaction between the congestion avoidance and control mechanisms of TCP and assured forwarding. Our main purpose is to shed light on this interaction, and to show that, in the current DiffServ framework, poor performance of TCP traffic flows can result from the existing mismatch between the assured forwarding traffic conditioning procedures and the TCP congestion management. We propose a new adaptive packet marking policy to deal with congestion situations that may occur. We show that, with this policy, the provisioned rate for TCP flows can be achieved.     

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.     

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。     

Towards a better-than-best-effort forwarding service for multimediaflows

A salient requirement of interactive multimedia applications is that they transmit data continuously at uniform rates with minimum possible end-to-end delay. The majority of these applications do not require hard and fast guarantees of network performance, but the current best-effort forwarding model of the Internet is frequently insufficient for realizing these requirements. Worse still, the requirement of uniform-rate transmission puts many multimedia applications at odds with current and proposed Internet network management practices that assume or require TCP-like reactions to packet loss. We are investigating router-based active queue management, specifically the use of queue occupancy thresholds to isolate TCP flows and to provide a better-than-best-effort forwarding service for flows in need of uniform-rate transmissions. Our current scheme, class-based thresholds (CBT), relies on a packet marking mechanism such as those proposed for realizing differentiated services on the Internet. CBT, when combined with existing active router queue management schemes such as random early detection (RED), provides a performance for TCP that approximates that achievable under a packet scheduling scheme and acceptable performance for multimedia flows. CBT is a simple and efficient mechanism with implementation complexity and run-time overhead comparable to that of RED     

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.     

A novel congestion control protocol with AQM support for IP-based networks

Multimedia services (Real-time and Non real-time) have different demands, including the need for high bandwidth and low delay, jitter and loss. TCP is a dominant protocol on the Internet. In order to have the best performance in TCP, the congestion window size must be set according to some parameters, since the TCP source is not aware of the window size. TCP emphasizes more on reliability than timeliness, so TCP is not suitable for real-time traffic. In this paper an active Queue management support TCP (QTCP) model is presented. Source rate is regulated based on the feedback which is received from intermediate routers. Furthermore, in order to satisfy the requirements of multimedia applications, a new Optimization Based active Queue management (OBQ) mechanism has been developed. OBQ calculates packet loss probabilities based on the queue length, packets priority and delay in routers and the results are sent to source, which can then regulate its sending rate. Simulation results indicate that the QTCP reduces packet loss and buffer size in intermediate nodes, improves network throughput and reduces delay.     

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.     

A time-dominated TCP congestion control over heterogeneous networks

The traditional transmission control protocol (TCP) suffers from performance problems such as throughput bias against flows with longer packet roundtrip time (RTT), which leads to burst traffic flows producing high packet loss, long delays, and high delay jitter. This paper proposes a TCP congestion control mechanism, TD-TCP, that the sender increases the congestion window according to time rather than receipt of acknowledgement. Since this mechanism spaces out data sent into the network, data are not sent in bursts. In addition, the proposed mechanism reduces throughput bias because all flows increase the congestion window at the same rate regardless of their packet RTT. The implementation of the mechanism affects only the protocol stack at the sender; hence, neither the receiver nor the routers need modifications. The mechanism has been implemented in the Linux platform and tested in conjunction with various TCP variants in real environments. The experimental result shows that the proposed mechanism improves transmission performance, especially in networks with congestion and/or high packet loss rates. Experiments in real commercial wireless networks have also been conducted to support the proposed mechanism's practical use. Copyright © 2008 John Wiley & Sons, Ltd.     

Using Quick‐Start to enhance TCP‐friendly rate control performance in bidirectional satellite networks

In the not so distant future, we envisage an Internet where the biggest share of capacity is used by streaming applications. To avoid congestion collapse from unresponsive flows calls for a robust and ubiquitous end‐to‐end multimedia congestion control mechanism, such as TCP‐friendly rate control (TFRC), which provides fair sharing with the other Internet traffic. This paper therefore analyses the implications of using rate‐adaptive congestion control over satellite links that utilize demand allocation multiple access (DAMA) to maximize satellite transponder utilization. The interaction between TFRC and DAMA is explored using simulations supported by fluidic flow models. The analysis shows that DAMA reduces the start‐up phase of TFRC, causing non‐negligible delays. To mitigate this problem, we propose a new cross‐layer method based on the Quick‐Start mechanism. This can accelerate the start‐up of multimedia flows by a judicious allocation of additional capacity derived from cross‐layer signalling. Copyright © 2009 John Wiley & Sons, Ltd.