一种基于DiffServ的IP电话QoS实现方案

如何为IP网络中的业务提供QoS保证正成为lP技术所要解决的关键问题。文中基于Diffserv提出了一种IP电话QoS方案，并对其基本思想及实现方法作了详细介绍。该方案将不同的分组数据包设置成不同的优先级，其中系统控制分组数据包优先级最高，语音分组数据包次之，普通数据分组数据包最低，使得系统控制分组和语音分组数据包的平均等待时间缩短。研究结果表明，对不同数据包进行优先级设置是改善IP电话QoS的一种可行方法。     

Investigation of the M2/G2/1/∞,N queue with restricted admission of priority customers and its application to HSDPA mobile systems

This paper investigates a queuing system for QoS optimization of multimedia traffic consisting of aggregated streams with diverse QoS requirements transmitted to a mobile terminal over a common downlink shared channel. The queuing system, proposed for buffer management of aggregated single-user traffic in the base station of High-Speed Downlink Packet Access (HSDPA), allows for optimum loss/delay/jitter performance for end-user multimedia traffic with delay-tolerant non-real-time streams and partially loss tolerant real-time streams. In the queuing system, the real-time stream has non-preemptive priority in service but the number of the packets in the system is restricted by a constant. The non-real-time stream has no service priority but is allowed unlimited access to the system. Both types of packets arrive in the stationary Poisson flow. Service times follow general distribution depending on the packet type. Stability condition for the model is derived. Queue length distribution for both types of customers is calculated at arbitrary epochs and service completion epochs. Loss probability for priority packets is computed. Waiting time distribution in terms of Laplace–Stieltjes transform is obtained for both types of packets. Mean waiting time and jitter are computed. Numerical examples presented demonstrate the effectiveness of the queuing system for QoS optimization of buffered end-user multimedia traffic with aggregated real-time and non-real-time streams.     

Delivering relative differentiated services in future high-speed networks using hierarchical dynamic deficit round robin

With the increasing deployment of real-time audio/video services over the Internet, provision of quality of service (QoS) has attracted much attention. When the line rate of future networks upgrades to multi-terabits per second, if routers/switches intend to deliver differentiated services through packet scheduling, the reduction of computational overhead and elimination of bottleneck resulting from memory latency will both become important factors. In addition, the decrease of average queueing delay and provision of small delays for short packets are two further critical factors influencing the delivery of better QoS for real-time applications. The advanced waiting time priority (AWTP) is a timestamp-based packet scheduler which is enhanced from the well-known WTP. Although AWTP considers the effect of packet size, the latency resulting from timestamp access and a great quantity of computational overhead may result in bottlenecks for AWTP being deployed over high-speed links. Many existing schedulers have the same problems. We propose a multi-level hierarchical dynamic deficit round-robin (MLHDDRR) scheduling scheme which is enhanced from the existing dynamic deficit round-robin scheduler. The new scheme can resolve these issues and efficiently provide relative differentiated services under a variety of load conditions. Besides, MLHDDRR can also protect the highest priority traffic from significant performance degradation due to bursts of low-priority traffic. We compare the performance of AWTP with the proposed scheme. Extensive simulation results and complexity analysis are presented to illustrate the effectiveness and efficiency of MLHDDRR.     

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。     

A More Practical Approach for Single-Packet IP Traceback using Packet Logging and Marking

Tracing IP packets to their origins is an important step in defending Internet against denial-of-service attacks. Two kinds of IP traceback techniques have been proposed as packet marking and packet logging. In packet marking, routers probabilistically write their identification information into forwarded packets. This approach incurs little overhead but requires large flow of packets to collect the complete path information. In packet logging, routers record digests of the forwarded packets. This approach makes it possible to trace a single packet and is considered more powerful. At routers forwarding large volume of traffic, the high storage overhead and access time requirement for recording packet digests introduce practicality problems. In this paper, we present a novel scheme to improve the practicality of log-based IP traceback by reducing its overhead on routers. Our approach makes an intelligent use of packet marking to improve scalability of log-based IP traceback. We use mathematical analysis and simulations to evaluate our approach. Our evaluation results show that, compared to the state-of-the-art log-based approach called hash-based IP traceback, our approach maintains the ability to trace single IP packet while reducing the storage overhead by half and the access time overhead by a factor of the number of neighboring routers.     

非饱和状态下时隙CSMA/CA机制改进与性能分析

针对无线传感器网络中有重要信息的高优先级数据包需要尽快传输,且IEEE 802.15.4协议本身不支持任何优先级机制的情况,结合优先级调度策略和差分服务机制,对具有优先级级的时隙CSMA/CA机制进行全面数学建模,包括节点马尔科夫模型和信道马尔科夫模型,据此提出了一种非饱和状态下具有优先级的IEEE 802.15.4时隙CSMA/CA机制性能的分析方法。通过比较分析,改进的机制对提高网络中高优先级数据包的传输性能具有积极作用。     

LTE系统中提高TCP性能的资源调度算法研究

LTE可以提供真正无处不在基于IP的移动宽带业务,但随着承载网的IP化,网络拥塞、丢包、抖动、延时等质量问题将影响到LTE业务层的QoS质量.作为LTE无线资源管理的核心,研究并设计一个良好的资源调度算法是提高数据业务的性能和终端用户的体验是一个亟待解决的重要任务.本文通过借鉴LTE对VoIP数据分组的半持续调度算法的思想,提出了一种LTE的无线资源调度的改进方案.方案将TCP确认包映射到具有更高优先级的空闲逻辑信道,从而降低了ACK包在无线信道中发生丢弃和拥塞的概率,避免了TCP拥塞控制机制的频繁开启.仿真结果表明,本文提出的TCP确认包映射转换方案在RTT时延、吞吐量等方面均有一定的提升,具有一定的稳定性和性能优势.     

Availability of end-to-end ideal QoS in IP packet networks

Network processor technology has advanced to the point where high-precision time-based store-and-forward logic is readily incorporated into packet switches and routers. With appropriate scheduling, packets from multiple flows can be serviced without contending for link resources. Accordingly, packet flows traversing a network of switching elements can have both path and time determinacy attributes which support ideal end-to-end QoS (zero jitter, zero loss, acceptable end-to-end latency) for real-time UDP packet flows and guaranteed goodput for TCP flows. One approach to packing a network with a relatively large number of such deterministic flows, i.e. achieving high availability of the ideal QoS service in a network, uses precise buffering of packets at each switch, which introduces latency. This paper describes analysis methods for quantifying how much buffering may be necessary to achieve high (99.999%) availability. For typical network topologies the analysis shows that buffering latency requirements are very small compared to transport delays, even when the network is highly utilized with heterogeneous (e.g. voice, video, circuit emulation, and data) traffic. Actual physical implementations have empirically validated the analysis results as well as the scalability of the end-to-end, time-based forwarding approach and the end-to-end availability of ideal QoS services in IP packet networks.     

DiffServ网络中基于流量负载的动态带宽分配

在DiffServ网络中,流量以聚集类的形式存在,聚集类中的流量负载是随时间不断地发生动态变化的.当不同聚集类中的流量负载与调度算法(如WFQ)为其分配的资源(如带宽)不成比例时,即使两个聚集类的优先级相同,它们中的数据包也会得到不公平的待遇.为此,DiffServ网络中面向公平的动态带宽分配引起了广泛的研究.本文中为了实现公平的带宽分配,提出了一种基于流量负载的动态带宽分配的方法,其中在动态计算各个聚集类应分得的新带宽时主要考虑了当前分得的带宽和聚集类的队列长度增量这两个因素.仿真实验结果说明了该方法的有效性.     

A traffic-class burst-polling based delta DBA scheme for QoS in distributed EPONs

Ethernet passive optical network (EPON) preserves the merits of traditional Ethernet network while reducing complexities and improving quality of service (QoS). In this paper, a traffic-class burst-polling based delta dynamic bandwidth allocation (TCBP-DDBA) scheme is presented to provide better QoS to expedited forwarding packet and maximize channel utilization service to assure forwarding and best effort packets. The network resources are efficiently utilized and adaptively allocated to the three traffic classes by guaranteeing the requested QoS. Simulation results using OPNET show that the TCBP-DDBA scheme performs well in comparison to the conventional allocation scheme for a set of given parameters such as: packet delay, queue size, packet delay variation and channel utilization. This work considers system-wide DBA development in contrast to unit-based approach. It is concluded that the algorithm can be used for many types of EPON-based practical distributed networks.     

Admission control by implicit signaling in support of voice over IP over ADSL

Voice over DSL (VoDSL) is a technology that enables the transport of data and multiple voice calls over a single copper-pair. Voice over ATM (VoATM) and Voice over IP (VoIP) are the two main alternatives for carrying voice over DSL. ATM is currently the preferred technology, since it offers the advantage of ATM’s built-in Quality of Service (QoS) mechanisms. IP QoS mechanisms have been maturing only in recent years. However, if VoIP can achieve comparable performance to that of VoATM in the access networks, it would facilitate end-to-end IP telephony and could result in major cost savings. In this paper, we propose a VoIP-based VoDSL architecture that provides QoS guarantees comparable to those offered by ATM in the DSL access network. Our QoS architecture supports Premium and Regular service categories for voice traffic and the Best-Effort service category for data traffic. The Weighted Fair Queuing algorithm is used to schedule voice and data packets for transmission over the bottleneck link. Fragmentation of large data packets reduces the waiting time for voice packets in the link. We also propose a new admission control mechanism called Admission Control by Implicit Signaling. This mechanism takes advantage of application layer signaling by mapping it to the IP header. We evaluate the performance of our QoS architecture by means of a simulation study. Our results show that our VoIP architecture can provide QoS comparable to that provided by the VoATM architecture.     

A QoS-enhanced intelligent stochastic real-time packet scheduler for multimedia IP traffic

A re-configurable, QoS-enhanced intelligent stochastic real-time optimal fair packet scheduler, QUEST, for IP routers is proposed and investigated. The objective is to maximize the system QoS subject to the constraint that the processor utilization is kept at 100%. All past work on router schedulers for multimedia traffic were of earlier generation, in that they focused on maximizing utilization whereas being QoS-aware but without explicitly maximizing the QoS. Keeping utilization fixed at nearly 100%, QoS is dynamically maximized, thus moving to the next generation. QUEST’s other unique advantages are three-fold. First, it solves the challenging problem of starvation for low priority processes; second, it solves the major bottleneck of Earliest Deadline First scheduler’s failure at heavy traffic loads. Finally, QUEST offers the benefit of arbitrarily pre-programming the process utilization ratio. Three classes of multimedia IP traffic, namely, VoIP, IPTV and HTTP have been considered. Two most important QoS metrics, namely, packet loss rate (PLR) and mean waiting time, are addressed. All claims are supported by discrete event and Monte Carlo simulations. The proposed scheduler outperforms benchmark schedulers and offers 37% improvement in packet loss rate and 23% improvement in mean waiting time over the best competing current scheduler Accuracy-aware EDF. The proposed scheduler was validated in a test-bed platform of a NetFPGA® router and results were observed with Paessler® PRTG network monitor.

     

Parallel Switch System with QoS Guarantee for Real-Time Traffic

This paper studies the load-balancing algorithm and quality of service （QoS） control mechanism in a 320Gb/s switch system, which incorporates four packet-level parallel switch planes. Eight priorities for both unicast and multicast traffic are implemented, and the highest priority with strict QoS guarantee is designed for real-time traffic. Through performance analysis under multi-prlorlty burst traffic, we demonstrate that the load-balancing algorithm is efficient, and the switch system not only provides excellent performance to real-time traffic, but also efficiently allocates bandwidth among other traffic of lower priorities. As a result, this parallel switch system is more scalable towards next generation core routers with QoS guarantee, as well as ensures in-order delivery of IP packets.     

A novel path protection scheme for MPLS networks using multi-path routing

Multi-protocol label switching (MPLS) is an evolving network technology that is used to provide traffic engineering (TE) and high speed networking. Internet service providers, which support MPLS technology, are increasingly demanded to provide high quality of service (QoS) guarantees. One of the aspects of QoS is fault tolerance. It is defined as the property of a system to continue operating in the event of failure of some of its parts. Fault tolerance techniques are very useful to maintain the survivability of the network by recovering from failure within acceptable delay and minimum packet loss while efficiently utilizing network resources.In this paper, we propose a novel approach for fault tolerance in MPLS networks. Our approach uses a modified (k, n) threshold sharing scheme with multi-path routing. An IP packet entering MPLS network is partitioned into n MPLS packets, which are assigned to node/link disjoint LSPs across the MPLS network. Receiving MPLS packets from k out of n LSPs are sufficient to reconstruct the original IP packet. The approach introduces no packet loss and no recovery delay while requiring reasonable redundant bandwidth. In addition, it can easily handle single and multiple path failures.     

Management of Multi-Queue Switches in QoS Networks

The concept of Quality of Service (QoS) networks has gained growing attention recently, as the traffic volume in the Internet constantly increases, and QoS guarantees are essential to ensure proper operation of most communication-based applications. A QoS switch serves m incoming queues by transmitting packets arriving to these queues through one output port, one packet per time step. Each packet is marked with a value indicating its priority in the network. Since the queues have bounded capacities and the rate of arriving packets can be much higher than the transmission rate, packets can be lost due to insufficient queue space. The goal is to maximize the total value of transmitted packets. This problem encapsulates two dependent questions: buffer management, namely which packets to admit into the queues, and scheduling, i.e. which queue to use for transmission in each time step. We use competitive analysis to study online switch performance in QoS-based networks. Specifically, we provide a novel generic technique that decouples the buffer management and scheduling problems. Our technique transforms any single-queue buffer management policy (preemptive or non-preemptive) to a scheduling and buffer management algorithm for our general m queues model, whose competitive ratio is at most twice the competitive ratio of the given buffer management policy. We use our technique to derive concrete algorithms for the general preemptive and non-preemptive cases, as well as for the interesting special cases of the 2-value model and the unit-value model. We also provide a 1.58-competitive randomized algorithm for the unit-value case. This case is interesting by itself since most current networks (e.g. IP networks) do not yet incorporate full QoS capabilities, and treat all packets equally.     

Modeling and performance analysis for IPv6 traffic with multiple QoS classes

This paper focuses on the modeling and performance analysis for IPv6 traffic with multi-class QoS in virtual private networks (VPN). The multi-class QoS is implemented on differentiated service basis using priority scheme of 4 bits defined in the packet header of IPv6. A VPN-enabled IP router is modeled as a tandem queuing system in which each output link consists of two parallel priority output queues. The high-priority queue is used to carry the delay sensitive traffic while the low-priority queue is used to carry the delay insensitive traffic. On the other hand, multiple thresholds are implemented in each queue, respectively, for packet loss priority control. The performance analysis is done using fluid flow techniques. The numerical results obtained from the analysis show that the differentiated service based on the priority schemes defined in IPv6 is able to effectively satisfy the multi-class QoS requirement for supporting multimedia services in VPN. The performance trade-off between the delay sensitive traffic and delay insensitive traffic in terms of traffic throughput, packet loss probability and end-to-end delay in VPN networks is presented.     

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.     

Quality-of-service mapping mechanism for packet video indifferentiated services network

This paper presents a futuristic framework for quality-of-service (QoS) mapping between practically categorized packet video and relative differentiated service (DiffServ or DS) network employing unified priority index and adaptive packet forwarding mechanism under a given pricing model (e.g., DiffServ level differentiated price/packet). Video categorization is based on the relative priority index (RPI), which represents the relative preference per each packet in terms of loss and delay. We propose an adaptive packet forwarding mechanism for a DiffServ network to provide persistent service differentiation. Effective QoS mapping is then performed by mapping video packets onto different DiffServ levels based on RPI. To verify the efficiency of proposed strategy, the end-to-end performance is evaluated through an error resilient packet video transmission using ITU-T H.263+ codec over a simulated DiffServ network. Results show that the proposed QoS mapping mechanism can exploit the relative DiffServ advantage and result in the persistent service differentiation among DiffServ levels and the enhanced end-to-end video quality with the same pricing constraint     

一种跨平台的实时TCP/IP协议栈设计

为提高异构系统下网络通信的实时性,提出一种跨平台的实时TCP/IP协议栈(RTTCP/IP)实现方法。运用操作系统适配技术屏蔽底层数据处理的差异性,增强协议栈的可移植性和可扩展性;通过简化TCP/IP协议栈的结构,减少协议栈对系统资源的占用;采用内存映射技术,将内核空间地址映射到用户空间,避免用户与内核间的数据拷贝操作;引入基于时分多址的介质访问机制和数据包优先级策略,防止网络传输冲突,解决数据包优先级倒置问题。测试结果表明,RTTCP/IP实现方法能够减少系统开销和通信延迟,提高系统实时性和稳定性。     

Improved WWW multimedia transmission performance in HTTP/TCP overATM networks

Transmission control protocol/Internet protocol (TCP/IP) is the de facto standard of the networking world. It dynamically adjusts routing of packets to accommodate failures in channels and allows construction of very large networks with little central management. But IP packets are based on the datagram model and are not really suited to real-time traffic. In order to overcome the drawbacks, a new network technology, ATM, is proposed. ATM provides quality of service (QOS) guarantees for various classes of applications and in-order delivery of packets via connection oriented virtual circuits. Unfortunately, when ATM is to be internetworked with the existing network infrastructure, some special signaling, addressing and routing protocols are needed. IP over ATM is one of the methods proposed by IETF. It allows existing TCP/IP applications to run on ATM end-stations and ATM networks to interconnect with legacy LAN/WAN technologies. But the performance of TCP/IP over ATM leaves something to be desired. Partial packet discard (PPD) and early packet discard (EPD) are two schemes to improve its performance. This paper proposes a “selective packet retransmission” scheme for improving HTTP/TCP performance when transmitting through ATM networks. In selective packet retransmission, we take advantage of the property of humans' perception tolerance for errors to determine whether to retransmit a corrupted TCP segment or not. For lossable data, such as images, when an error occurs because of cell losses, it will not be retransmitted. The simulations show that, for the same buffer size and traffic load, selective packet retransmission performs better than PPD, EPD, and plain TCP over ATM