Performance analysis of relative service using TCP‐aware marking and dynamic WRED

The implementation of successful assured forwarding (AF) services according to the DiffServ framework remains a challenging problem today, despite the numerous proposals for assured forwarding per‐hop‐behaviour (AF PHB) mechanisms and AF‐based service implementations. The interaction of the TCP and UDP traffic under an AF‐based service and a number of relative issues such as fairness among flows, achievable bandwidth guarantees and qualitative performance have been taken into consideration in this work in order to address the existing limitations. We propose two modules, the TCP‐window aware marker (TWAM) and the dynamic WRED (WRED) mechanism for implementing the differentiated services (DiffServ) AF PHB. We provide analytical models and an experimental evaluation in order to demonstrate how they succeed in enhancing the quality, improving the performance and easing the deployment of a production level AF‐based service. Copyright © 2008 John Wiley & Sons, Ltd.     

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     

区分服务结构及其TCP性能分析

本文首先对区分服务(DiffServ)结构中的边缘路由器和核心路由器机制进行了系统的分类研究,分析并比较了各种机制.接着,对区分服务结构中TCP的性能问题进行了研究,总结了国内外对这一问题的仿真、解析模型分析和实验研究的成果.找出了关键的问题所在,并根据我们的研究成果提出了一些区分服务结构本身的改进建议.     

Enabling novel premium service classes in DiffServ over MPLS‐enabled network

Network resources dimensioning and traffic engineering influence the quality in provisioned services required by the Expedited Forwarding (EF) traffic in production networks established through DiffServ over MPLS‐enabled network. By modeling EF traffic flows and the excess of network resources reserved for it, we derive the range of delay values which are required to support these flows at DiffServ nodes. This enables us to develop an end‐to‐end (e2e) delay budget‐partitioning mechanism and traffic‐engineering techniques within a framework for supporting new premium QoS levels, which are differentiated based on e2e delay, jitter and loss. This framework enables ingress routers to control EF traffic flow admission and select appropriate routing paths, with the goal of EF traffic balancing, avoiding traffic congestion and getting the most use out of the available network resources through traffic engineering. As a result, this framework should enable Internet service providers to provide three performance levels of EF service class to their customers provided that their network is DiffServ MPLS TE aware. Copyright © 2008 John Wiley & Sons, Ltd.     

一种支持多媒体通信QoS的拥塞控制机制

本文针对Internet传输协议TCP的和式增加积式减少(AIMD)拥塞控制机制不适应多媒体通信,而目前拥塞控制的研究又大多集中在尽量做好(Best-effort)服务上的问题,结合Internet上多媒体通信的特点及其对QoS的要求,提出了一种将多媒体通信服务质量(QoS)控制和基于速率拥塞控制结合起来的拥塞控制的新机制.本文详细地研究了这一机制,并提出了源端多媒体数据流的带宽控制策略、基于动态部分缓存共享(DPBS)的数据包丢失控制方案和接收端计算包丢失率p的方法.最后给出了整个拥塞控制机制的系统结构.     

TCP-LP: low-priority service via end-point congestion control

Service prioritization among different traffic classes is an important goal for the Internet. Conventional approaches to solving this problem consider the existing best-effort class as the low-priority class, and attempt to develop mechanisms that provide "better-than-best-effort" service. In this paper, we explore the opposite approach, and devise a new distributed algorithm to realize a low-priority service (as compared to the existing best effort) from the network endpoints. To this end, we develop TCP Low Priority (TCP-LP), a distributed algorithm whose goal is to utilize only the excess network bandwidth as compared to the "fair share" of bandwidth as targeted by TCP. The key mechanisms unique to TCP-LP congestion control are the use of one-way packet delays for early congestion indications and a TCP-transparent congestion avoidance policy. The results of our simulation and Internet experiments show that: 1) TCP-LP is largely non-intrusive to TCP traffic; 2) both single and aggregate TCP-LP flows are able to successfully utilize excess network bandwidth; moreover, multiple TCP-LP flows share excess bandwidth fairly; 3) substantial amounts of excess bandwidth are available to the low-priority class, even in the presence of "greedy" TCP flows; 4) the response times of web connections in the best-effort class decrease by up to 90% when long-lived bulk data transfers use TCP-LP rather than TCP; 5) despite their low-priority nature, TCP-LP flows are able to utilize significant amounts of available bandwidth in a wide-area network environment.     

基于光GMPLS接入网中区分服务的QoS研究

通过配置区分服务体系实现基于光GMPLS环状网络的QoS测量,涉及到不同的实时多媒体服务.讨论并分析区分服务结构的三种服务级别的特性.仿真结果表明,在网络过载的情况下,快速转发类和保证转发类都有很好的保护作用.加速转发类适合于实时服务,而最大努力类对任何类型的服务}生能都非常低,同时保证转发非常适合于视频流.     

基于RTT的TCP流带宽公平性保障机制

TCP端到端的拥塞控制机制使得TCP连接获得的瓶颈带宽反比于RTT(数据包往返时间)。为了缓解TCP对于RTT较小流的偏向,区分服务的流量调节机制在RTT较小的流取得目标速率且获得多余资源的情况下可以确保RTT较大流不至于饥饿。现有的基于RTT的流量调节机制在网络拥塞程度较轻时非常有效,但是当网络拥塞程度较重时,由于对RTT较大流的过分保护而导致RTT较小流饥饿。因此,通过引进自适应的思想提出了改进方法,其主要思想就是根据网络的拥塞程度自适应地调整对RTT较大流的保护程度。大量的仿真试验表明所提的机制能有效保障TCP流的带宽公平性并且比现有的方法具有更好的强壮性。     

Performance evaluation of selected Transmission Control Protocol variants over a digital video broadcasting‐second generation broadband satellite multimedia system with QoS

This paper presents an analysis of several Transmission Control Protocol (TCP) variants working over a digital video broadcasting‐second generation (DVB‐S2) satellite link with the support of the Differentiated Services (DiffServ) architecture to provide quality of service (QoS). This analysis is carried out using the NS‐2 simulator tool. Three TCP variants are considered: SACK TCP, Hybla TCP, and CUBIC TCP. These TCP variants are taken as a starting point because they have proven to be the most suitable variants to deal with long delays present in satellite links. The DVB‐S2 link also introduces the challenge of dealing with variable bandwidth, whereas the DiffServ architecture introduces the challenge of dealing with different priorities. In this paper, we propose a DiffServ model that includes a modified queuing mechanism to enhance the goodput of the assured forwarding traffic class. This modified DiffServ model is simulated and tested, considering the interaction of the selected TCP variants. In addition, we present evaluation metrics, significant simulations results, and conclusions about the performance of these TCP variants evaluated over the proposed scenario. As a general conclusion, we show that CUBIC TCP is the TCP variant that shows the best performance in terms of goodput, latency, and friendliness. Copyright © 2012 John Wiley & Sons, Ltd.     

Understanding and improving TCP performance over networks withminimum rate guarantees

A large number of Internet applications are sensitive to overload conditions in the network. While these applications have been designed to adapt somewhat to the varying conditions in the Internet, they can benefit greatly from an increased level of predictability in network services. We propose minor extensions to the packet queueing and discard mechanisms used in routers, coupled with simple control mechanisms at the source that enable the network to guarantee minimal levels of throughput to different sessions while sharing the residual network capacity in a cooperative manner. The service realized by the proposed mechanisms is an interpretation of the controlled-load service being standardized by the Internet Engineering Task Force. Although controlled-load service can be used in conjunction with any transport protocol, our focus in this paper is on understanding its interaction with Transmission Control Protocol (TCP). Specifically, we study the dynamics of TCP traffic in an integrated services network that simultaneously supports both best-effort and controlled-load sessions. In light of this study, we propose and experiment with modifications to TCP's congestion control mechanisms in order to improve its performance in networks where a minimum transmission rate is guaranteed. We then investigate the effect of network transients, such as changes in traffic load and in service levels, on the performance of controlled-load as well as best-effort connections. To capture the evolution of integrated services in the Internet, we also consider situations where only a selective set of routers are capable of providing service differentiation between best-effort and controlled-load traffic. Finally, we show how the service mechanisms proposed here can be embedded within other packet and link scheduling frameworks in a fully evolved integrated services Internet     

Implicit admission control

Internet protocols currently use packet-level mechanisms to detect and react to congestion. Although these controls are essential to ensure fair sharing of the available resource between multiple flows, in some cases they are insufficient to ensure overall network stability. We believe that it is also necessary to take account of higher level concepts, such as connections, flows, and sessions when controlling network congestion. This becomes of increasing importance as more real-time traffic is carried on the Internet, since this traffic is less elastic in nature than traditional Web traffic. We argue that, in order to achieve better utility of the network as a whole, higher level congestion controls are required. By way of example, we present a simple connection admission control (CAC) scheme which can significantly improve the overall performance. This paper discusses our motivation for the use of admission control in the Internet, focusing specifically on control for TCP flows. The technique is not TCP specific, and can be applied to any type of flow in a modern IP infrastructure. Simulation results are used to show that it can drastically improve the performance of TCP over bottleneck links. We go on to describe an implementation of our algorithm for a router running the Linux 2.2.9 operating system. We show that by giving routers at bottlenecks the ability to intelligently deny admission to TCP connections, the goodput of existing connections can be significantly increased. Furthermore, the fairness of the resource allocation achieved by TCP is improved     

DiffServ resource allocation for fast handoff in wireless mobileInternet

The next-generation wireless networks are evolving toward a versatile IP-based network that can provide various real-time multimedia services to mobile users. Two major challenges in establishing such a wireless mobile Internet are support of fast handoff and provision of quality of service (QoS) over IP-based wireless access networks. In this article, a DiffServ resource allocation architecture is proposed for the evolving wireless mobile Internet. The registration-domain-based scheme supports fast handoff by significantly reducing mobility management signaling. The registration domain is integrated with the DiffServ mechanism and provisions QoS guarantee for each service class by domain-based admission control. Furthermore, an adaptive assured service is presented for the stream class of traffic, where resource allocation is adjusted according to the network condition in order to minimize handoff call dropping and new call blocking probabilities     

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     

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.     

Method of data cleaning for network traffic classification

Network traffic classification aims at identifying the application types of network packets. It is important for Internet service providers （ISPs） to manage bandwidth resources and ensure the quality of service for different network applications However, most classification techniques using machine learning only focus on high flow accuracy and ignore byte accuracy. The classifier would obtain low classification performance for elephant flows as the imbalance between elephant flows and mice flows on Internet. The elephant flows, however, consume much more bandwidth than mice flows. When the classifier is deployed for traffic policing, the network management system cannot penalize elephant flows and avoid network congestion effectively. This article explores the factors related to low byte accuracy, and secondly, it presents a new traffic classification method to improve byte accuracy at the aid of data cleaning. Experiments are carried out on three groups of real-world traffic datasets, and the method is compared with existing work on the performance of improving byte accuracy. Experiment shows that byte accuracy increased by about 22.31% on average. The method outperforms the existing one in most cases.     

A random early demotion and promotion marker for assured services

The differentiated services (DiffServ) model, proposed to evolve the current best-effort Internet to a quality-of-service-aware Internet, provides packet level service differentiation on a per-hop basis. The end-to-end service differentiation may be provided by extending the per-hop behavior over multiple network domains through service level agreements between domains. The edge routers of each of the domains monitor the aggregate flow of the incoming packets and demote packets when the aggregate incoming traffic exceeds the negotiated interdomain service agreement. A demoted packet may encounter other edge routers on its path that have sufficient resources to route the packet with its original marking. In this paper, we propose a random early demotion and promotion (REDP) technique that works at the aggregate traffic level and allows (1) fair demotion of packets belonging to different flows, and (2) easy and fair detection and promotion of the demoted packets. Using early and random decisions on packets REDP ensures fairness in promotion and demotion. It uses a three color marking mechanism, reserving one color fur differentiating between a demoted packet and a packet with the original out-of-profile marking. We experiment with the proposed REDP scheme using the ns2 simulator for both TCP and UDP streams. The results demonstrate the fairness of REDP scheme in demoting and promoting packets. Furthermore, we show a variety of results that demonstrates that REDP provides better assured services compared to the previously proposed RIO scheme with or without the provision of promotion     

A Study on Differentiated Service Queuing Scheme with an Overflow Buffer Allocation Within a UMTS Core Network

According to the QoS features of the four types of UMTS traffic, this study proposes a priority-based queuing scheme to support differentiated services among all UMTS traffic; it bases on packet transmission priorities of four types of UMTS traffic to handle packets forwarding in a gateway within a UMTS core network. In the proposed queuing scheme, a static dedicated logical queuing buffer allocation and a dynamic overflow buffer allocation are used to support packet enqueuing/dequeuing jobs. In this study, the ns2 network simulator is used as a simulation platform and different categories of simulation scenarios are executed. Moreover, the IETF DiffServ scheme is used as a comparison to understand the packet forwarding performance of the proposed scheme. By analyzing simulation data, two important points can be drawn. The proposed queuing scheme supports a differentiated packet forwarding behavior among four types of UMTS traffic. And, the differentiated packet forwarding behaviors with the proposed queuing scheme are similar to the packet forwarding behavior with the IETF DiffServ scheme.     

Size-based scheduling to improve the performance of short TCP flows

The Internet today carries different types of traffic that have different service requirements. A large fraction of the traffic is either Web traffic requiring low response time or peer-to-peer traffic requiring high throughput. Meeting both performance requirements in a network where routers use droptail or RED for buffer management and FIFO as the service policy is an elusive goal. It is therefore worthwhile to investigate alternative scheduling and buffer management policies for bottleneck links. We propose to use the least attained service (LAS) policy to improve the response time of Web traffic. Under LAS, the next packet to be served is the one belonging to the flow that has received the least amount of service. When the buffer is full, the packet dropped belongs to the flow that has received the most service. We show that under LAS, as compared to FIFO with droptail, the transmission time and loss rate for short TCP flows are significantly reduced, with only a negligible increase in transmission time for the largest flows. The improvement seen by short TCP flows under LAS is mainly due to the way LAS interacts with the TCP protocol in the slow start phase, which results in shorter round-trip times and zero loss rates for short flows.