Scalable services via egress admission control

Allocating resources for multimedia traffic flows with real-time performance requirements is an important challenge for future packet networks. However, in large-scale networks, individually managing each traffic flow on each of its traversed routers has fundamental scalability limitations, in both the control plane's requirements for signaling, state management, and admission control, and the data plane's requirements for per-flow scheduling mechanisms. In this paper, we develop a scalable architecture and algorithm for quality-of-service management termed egress admission control. In our approach, resource management and admission control are performed only at egress routers, without any coordination among backbone nodes or per-flow management. Our key technique is to develop a framework for admission control under a general “black box” model, which allows for cross traffic that cannot be directly measured, and scheduling policies that may be ill-described across many network nodes. By monitoring and controlling egress routers' class-based arrival and service envelopes, we show how network services can be provisioned via scalable control at the network edge. We illustrate the performance of our approach with a set of simulation experiments using highly bursty traffic flows and find that despite our use of distributed admission control, our approach is able to accurately control the system's admissible region under a wide range of conditions     

An Admission Control Scheme Based on Online Measurement for VBR Video Streams Over Wireless Home Networks

This paper presents an online measurement-based admission control scheme on the basis that the aggregate VBR video traffic is lognormally distributed. The proposed scheme consists of two components: measurement process and admission decision. The measurement process applies a linear Kalman filter to estimate statistical parameters of aggregate VBR video traffic. The estimated statistical parameters are used to calculate the effective bandwidth for admission decision. Variable bit rate (VBR) video traffic with high data rate is expected to occupy a dominant proportion of bandwidth for future wireless broadband home networks. To guarantee quality-of-service (QoS) of such VBR video streams, while achieving a high level of channel utilization, an efficient admission control scheme is urgently required, especially for emerging wireless multimedia indoor services, such as HDTV, online video game, etc. The proposed scheme is computationally efficient and accurate without much prior traffic information. Simulation results verify its effectiveness and show that it performs well for both a small number of connections and a large number of connections.     

基于模糊控制的ATM网络VBR视频传输平滑策略

VBR视频传输的突发性是影响ATM网络服务质量的关键因素,文中通过模糊控制方法对视频传输流量的阈值进行监控,实现了接入的平滑并可动态调整传输速度。文中以传输流量及这发级作为评价指标进行分析,结果表明VBR视频传输自适应平滑策略的实效性。     

Admission control scheme based on priority access for wireless LANs

In order to support the quality of service (QoS) requirements at the medium access control (MAC) layer, the enhanced distributed channel access (EDCA) has been developed in IEEE 802.11e standard. However, it cannot guarantee the stringent real-time constraints of multimedia applications such as voice and video without an efficient method of controlling network loads. In this paper, we propose a measurement-based admission control scheme, which is made up of two parts: priority access and admission control. First, in order to measure the channel status per traffic type, we propose a priority access mechanism in which each priority traffic is distinguished by a busy tone, and separately performs its own packet transmission operation. Then, admission control mechanism protects existing flows from new ones, and maintains the QoS of the admitted flows based on the measured channel status information. Performance of the proposed scheme is evaluated by simulation. Our results show that the proposed scheme is very effective in guaranteeing the QoS of multimedia applications as well as in avoiding the performance starvation of low priority traffics.     

Threshold configuration and routing optimization for PCN-based resilient admission control

Pre-Congestion Notification (PCN) in IP networks uses packet metering and marking within a PCN domain to notify its egress nodes whether link-specific admissible or supportable rate thresholds have been exceeded by high-priority traffic. Based on this information simple admission control and flow termination is implemented. The latter is a new flow control function and useful in case of overload through high priority traffic which can occur in spite of admission control, e.g., when traffic is rerouted in failure cases. Resilient admission control admits only so much traffic that admitted traffic can be rerouted without causing congestion on backup paths in case of a likely failures, e.g., single link failures.We propose algorithms to configure the link-specific PCN rate thresholds such that resources are utilized efficiently and fairly by competing traffic aggregates while meeting resilience constraints. This is done for the single and dual marking PCN architecture whereby the single marking case is more demanding since it requires that the supportable rate is a fixed multiple of the admissible rate on all links within a single PCN domain. Furthermore, we derive objective functions to optimize the underlying routing system for both cases. Our performance results for various network types show that the dual marking PCN architecture leads to significantly better resource efficiency than the single marking PCN architecture.     

Performance analysis of multimedia based web traffic with QoS constraints

During the recent years, there has been a tremendous growth in the development and deployment of multimedia based networked applications such as video streaming, IP telephony, interactive games, among others. These applications, in contrast to elastic applications such as email and data sharing, are delay and delay jitter sensitive but can tolerate certain level of packet loss. A vital element of end-to-end delay and delay jitter is the random queueing delays in network switches and routers. Analysis of robust mechanisms for buffer management at network routers needs to be carried out in order to reduce end-to-end delay for traffic generated by multimedia applications. In this context, a threshold based buffer management scheme for accommodating multiple class multimedia traffic in network routers has been analysed. This technique effectively controls the allocation of buffer to various traffic classes according to their delay constraints. The forms of the joint state probabilities, as well as basic performance measures such as blocking probabilities are analytically established at equilibrium. Typical numerical experiments are included to illustrate the credibility of the proposed mechanism in the context of different quality of service (QoS) grades for various network traffic classes. This model, therefore, can be used as a powerful tool to provide a required grade of service to a particular class of multimedia based web traffic in any heterogeneous network.     

Above packet level admission control and bandwidth allocation for IEEE 802.16 wireless MAN

IEEE 802.16 wireless Metropolitan Area Network (MAN) is expected to revolutionize the broadband wireless access technology. Efficient resource management is essential in providing scalability in such large IEEE 802.16-based wireless network and Quality of Service (QoS) for multimedia applications (VoIP, MPEG, FTP, WWW) is usually achieved by appropriate classification of scheduling services and grant/request mechanism. In this paper, we firstly discuss the QoS issue in IEEE 802.16 wireless MANs and propose a dynamic admission control scheme for scheduling services defined in the 802.16 specification. The proposed scheme provides the highest priority for Unsolicited Grant Service (UGS) connections and maximizes the bandwidth utilization by employing bandwidth borrowing and degradation. We develop an approximate analytical model to evaluate the system performance by assuming that the traffic processes of all scheduling services are Poisson processes. In fact, the self-similarity of non-voice traffic makes its traffic process far from Poisson process and should be modeled by Poisson Pareto Burst Process (PPBP). Therefore, in the later part of the paper, we analyze upper bound blocking probabilities of all scheduling services above the packet level using PPBP model for fractal traffic and Gaussian model for aggregated traffic in large wireless network as well as using the Chernoff bound method. Based on the analytical results, we give another admission control and bandwidth allocation mechanism above the packet level so as to minimize the blocking probability of each type of service in IEEE 802.16 wireless MAN. Analytical and simulation results are obtained and compared to demonstrate the effectiveness of proposed schemes and validate our analytical models.     

A scheduler ASIC for a programmable packet switch

While the Internet is successful in supporting traditional data-only traffic, an integrated services Internet is inevitable with the emergence of new applications such as voice, video, multimedia, and interactive video conferencing. Such an integrated services network should support a wide range of applications with diverse quality of service requirements and traffic characteristics. Provision for quality of service in packet networks in general, and in the Internet in particular, is the focus of most of the recent developments in switching and routing system design. We designed a generic, single-queue scheduler engine for use in a programmable packet switch/router to handle IP packets, ATM cells, or a combination of both. Comprising 275,000 gates, the 0.35-micron ASIC is incorporated into a prototype programmable packet switch     

Packet video transport in ATM networks with single-bit feedback

The promise of a broadband integrated service digital network has led to the design of mechanisms for efficient transport of real-time compressed video in packet switching networks. We examine feedback control for video transport in ATM networks where the available feedback is a single bit of information carried in the cell header. We investigate the performance of three single-bit schemes for source rate adaptation. Two were originally for congestion control of bursty data traffic and are modified to control video traffic. The third scheme conveys more information about the state of queue(s) at the bottleneck. The simulation results show that all three schemes for feedback control of VBR video streams work remarkably well. During severe network congestion, the signal quality degrades gracefully, but not uniformly across all connections. Based on insights from the initial simulations, we propose a scheme to improve the fairness of service and demonstrate its effectiveness.     

PCN-based marked flow termination

Pre-congestion notification (PCN) uses packet metering and marking to notify boundary nodes of a Differentiated Services IP network if configured rate thresholds have been exceeded on some links. This feedback is used for PCN-based admission control and flow termination. While admission control is rather well understood, flow termination is a new flow control function and useful especially in case of failures or during flash crowds. We present marked flow termination as a new class of termination algorithms which terminate overload traffic gradually and that work well with multipath routing. We study their termination behavior, give recommendation for their configuration, and discuss their benefits and shortcomings.     

无线自组网络QoS流的准入控制和流量控制

作者曾提出一个QoS路由和准入控制机制－QDSR,并且在小规模实时视频传输实验床上实现了这些方案。虽然QDSR的准入控制只保证每个节点有足够的可用带宽给准入的所有流使用,但并不保证每个流在较小的时间尺度上得到请求的带宽。为此,作者对QDSR的准入控制机制进行了改进并增加了流量控制机制,以满足实时应用的QoS需求。在NS2仿真环境中实现了QDSR以及对QDSR的改进。仿真结果表明,改进机制改善了QDSR的视频流的传输质量,较好地提高了视频流的吞吐率和投递率,减小了数据包的端到端延迟,并且能很好地满足包括视频传输在内的多种业务的传输需要。     

基于IP网络的自适应QoS管理方案研究

目前实时音频、视频多媒体应用已经开始进入IP网络,但还有许多问题没有得到很好地解决,其中一个关键问题是多媒体服务的QoS问题、TCP／IP协议本身只提供一种“Best-effort”级别的服务,对QoS支持很少,“Best-effort”级别的服务往往会导致实时应用出现延迟抖动、分组丢失率高,从而极大的影响了实时应用的运行效果,因此必须研究可行的、高效的基于IP网络的QoS控制机制。IP网络QoS已成为分布式多媒体和网络通信的重要研究热点和难点课题。本文在IEＴＦ Intserv与Diffserv相结合的体系结构基础上,提出了一种适用于IP网络的自适应QoS管理框架,与Intserv或Diffserv模型不同的是,该QoS管理框架引入优先级节和自适应概念,QoS俦权处理采用基于多媒体优先级节的算法,在传输控制上采用了自适应QoS控制算法。它在应用层上完成,因此独立于底层网络协议。本文首先讨论了基本概念和函数,然后提出一种基于优先级节的自适应QoS管理框架和优先级调度流程。提出一种基于IP的自适应QoS协商机制,基本思想是基于RSVP（资源预留协议）。在提出QoS请求时进行QoS映射,然后启动适应性函数和资源管理函数进行协商,直到获得一组在当前资源状况下最佳的QoS指标。最后,本文还讨论了此方案的有效性并在自行开发的实验平台GUT上进行的QoS实验。实验表明应用该方法可以根据网络当前状态自适应地整多媒体实时应用的QoS需求。     

The probability relationship between video’s instantaneous and average aggregate rates

The growing demand for video applications and services has contributed substantially to the increase of video traffic on the Internet. Measurement-based admission control was proposed with the primary aim of eliminating or reducing the need of flow state information; also to control overhead for admission decision and maximize utilization at the potential cost of QoS degradation. Some of the admission algorithms depend on the instantaneous rate for its operation. On the other hand, the average aggregate rate has been proposed to better suit variable rate such as video traffic. In this paper, we investigate the probability relationship between the instantaneous and average aggregate rates for video traffic. A mathematical model has been developed to quantify the probability relationship between both rates and validated through extensive simulations using real video sequences. The average rate was found to be lower than instantaneous for a small number of flows, however there was no pronounced difference for a large number of flows. Furthermore, the difference between both rates increases for fast moving content such as sport or longer measurement time window.     

Joint In-network Video Rate Adaptation and Measurement-Based Admission Control: Algorithm Design and Evaluation

The important new revenue opportunities that multimedia services offer to network and service providers come with important management challenges. For providers, it is important to control the video quality that is offered and perceived by the user, typically known as the quality of experience (QoE). Both admission control and scalable video coding techniques can control the QoE by blocking connections or adapting the video rate but influence each other’s performance. In this article, we propose an in-network video rate adaptation mechanism that enables a provider to define a policy on how the video rate adaptation should be performed to maximize the provider’s objective (e.g., a maximization of revenue or QoE). We discuss the need for a close interaction of the video rate adaptation algorithm with a measurement based admission control system, allowing to effectively orchestrate both algorithms and timely switch from video rate adaptation to the blocking of connections. We propose two different rate adaptation decision algorithms that calculate which videos need to be adapted: an optimal one in terms of the provider’s policy and a heuristic based on the utility of each connection. Through an extensive performance evaluation, we show the impact of both algorithms on the rate adaptation, network utilisation and the stability of the video rate adaptation. We show that both algorithms outperform other configurations with at least 10 %. Moreover, we show that the proposed heuristic is about 500 times faster than the optimal algorithm and experiences only a performance drop of approximately 2 %, given the investigated video delivery scenario.     

一种基于节的视频服务器允许接纳控制算法

提出了一种基于“服务器节”的支持压缩多媒体流的服务器中CPU、磁盘、网络和内存等资源管理的方法和允许接纳控制算法。“服务器节”概念定义了一组客户视频服务特性,如播放、快进、慢进和暂停等,并且确定了视频服务所需资源的分配参。一个“服务器节”包括视频服务器、磁盘设备、网络设备和允许接纳控制。它不但能优化使用单个资源,对于给定系统支持最大数量的客户端,保证其服务质量（QoS),而且其允许接纳控制算法能根据系统所有资源的状况,在不影响原有的视频服务基础上,确定对客户端新提出的视频服务是否接受。     

Per-stream loss behavior of ∑MAP/M/1/K queuing system with a random early detection mechanism

In this paper, the matrix-analytic approach is applied to explore the per-stream loss behavior of the multimedia traffic under RED scheme. We constructed a ∑MAP/M/1/K queuing model for the RED mechanism with multimedia traffic which follows a continuous-time Markovian arrival process (MAP). In addition to evaluating the long-term per-stream packet drop probabilities, we examine the bursty nature of per-stream packet drops by means of conditional statistics with respect to dropped periods and the probability that the queuing system stays in the dropped period. The dropped period corresponds to having more than a certain number of packets in router buffer; non-dropped period corresponds to the opposite. These performance measures describe the quality of service provided by the router to particular multimedia traffic streams in the presence of background multimedia traffic.     

报文分流最优策略研究

在向下一代互联网络演进的过程中,一个重要的趋势是IP网络将成为语音、视频等应用的主要承栽.VoIP是一个重要的语音应用.然而,IP网络的丢包造成了VoIP的服务质量不能得到保证,并且对于VoIP而言,连续丢包对其服务质量的影响要远大于分散丢包.报文分流是近年来学术界讨论的一种提高VoIP服务质量的方法,其基本思想是把1个VoIP会话的报文分散到多个网络链路传输,从而把连续丢包转化为分散丢包,缓解丢包对VoIP服务质量的影响.然而,目前的研究只局限于用一种特定的分流策略(平均分流)说明报文分流的潜力.报文分流的理论基础,比如报文分流能在多大程度上提高VoIP的服务质量,什么是最优的分流策略等,并不明了.对报文分流的理论基础进行了研究,首次给出了分流策略与VoIP服务质量的定量关系描述,给出并证明了Bernoulli网络丢包模型下的最优分流策略.同时,以ns-2仿真实验验证了该最优分流策略在Gilbert网络丢包模型下的有效性.     

Reservation arbitrated access — a bandwidth on demand protocol for multiplexing variable bit rate isochronous traffic over dual bus metropolitan area networks

With the advent of multimedia communication services, transport of real-time traffic over metropolitan area networks (MANs) is becoming an important problem. We present a novel reservation arbitrated (RA) access protocol for multiplexing variable bit rate isochronous (VBRI) traffic such as packet voice and video over dual bus MANs in general and IEEE 802.6 MANs in particular. In combination with a cyclic release mechanism, RA access allows variable bit rate traffic sources (VBRSs) to capture and reserve some isochronous channels on a bandwidth on demand basis in a round robin fashion. For a reasonable bus length suitable for metropolitan coverage, it is possible to select operation parameters which enable contention free access in the reservation process. Bandwidth utilization can be further improved by employing a movable boundary option to efficiently integrate VBRI traffic with other traffic. System performances including packet loss ratio, packet delay, delay jitter, probability distribution of consecutive packet loss and channel utilization are analyzed by both theoretical computations and computer simulations for voice, video conference and motion video traffic. Results indicate that the protocol is fair and provides a nearly isochronous transport service while ensuring efficient bandwidth utilization, yielding substantial capacity improvements over pre-arbitrated (PA) access. Compared to queue-arbitrated (QA) access, RA access not only provides variable bit rate isochronous channels but also allows VBRSs to adapt to the reserved bandwidth during network congestion so that performance degradation can be minimized. RA access complements existing PA and QA access methods in 802.6 MANs to provide a complete traffic transport solution for all types of BISDN services.     

Available bandwidth estimation for the network paths with multiple tight links and bursty traffic

Available bandwidth (ABW) estimation is useful for various applications such as network management, traffic engineering, and rate-based multimedia streaming. Most of the ABW estimation methods are based on the fluid cross-traffic model. Inevitably, their estimation accuracy is limited in the network environments with bursty cross-traffic. In this paper, we apply packet trains (a series of probing packets) and a modified Ping to probe the ABW of a network path. Our proposed probing method can identify several tight links along a path and can infer their individual ABWs. The ABW estimation algorithm developed in this study, GNAPP, is also based on the fluid traffic model, but it can effectively filter out probing noise incurred in networks that carry bursty traffic. The algorithm employs not only the gaps of any two consecutive probing packets but also those of nonadjacent probing packets for ABW estimation. Thus, the number of samples for ABW estimation increases significantly without resorting to sending more probing packets and the estimation efficiency and accuracy are improved. In addition, two-stage filtering and moving averages are used in GNAPP for reducing estimation errors. Numerical results demonstrate that the estimation scheme based on GNAPP can achieve good accuracy even when the traffic is bursty and there are multiple tight links on the path being observed. Thus, it outperforms other well-known ABW estimation tools.     

基于车辆载带中继的路边单元突发业务分组调度最优策略

高速公路车联网场景中,路边单元(Roadside units,RSUs)可作为多种周边监测数据的汇入网关,其业务具有突发特性,且可通过移动车辆以"存储—载带—转发"方式传输到与骨干网络互联的RSU.针对RSU间业务传输问题,源RSU可根据实时业务到达率按需匹配资源,以应对业务突发性对分组端到端时延的影响.本文首先针对R...