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.     

A Study on a Differentiated Service Queuing Scheme Based on Transmitted Bytes Within a UMTS Core Network Gateway

Universal mobile telecommunications system (UMTS) is a popular 3G system to support network applications full of variety. According to the quality of service (QoS) features of four types of UMTS traffic, this study proposes a BBQ (byte-based queuing) scheme to handle a UMTS packet forwarding process in differentiation. With the BBQ scheme, four types of UMTS traffic depend on their QoS features to receive differentiated packet forwarding performance within a UMTS core network. Several scenarios are simulated to realize the packet transmission performance of the BBQ scheme. Moreover, a packet transmission performance comparison between the IETF RIO scheme and the BBQ scheme is discussed in this study. The simulation results show that a differentiated packet forwarding behavior among UMTS traffic can be provided by the BBQ scheme with low cost operation/architecture; this demonstrate the effectiveness of the proposed queuing scheme.     

A Study on Adaptive Time Token Priority-Based Queuing Scheme

Several famous priority-based queuing schemes operated in a gateway to support differentiated services among internet traffic. Examining packet forwarding operations in these queueing schemes, they only support a priority-based service either in a packet enqueuing process or in a packet dequeuing process. If a queuing scheme can support priority-based services in both packet enqueuing/dequeuing processes; it would enhance differentiated service performance for internet traffic. This study proposes a priority-based queuing scheme with an adaptive time token allotment measure to support a differentiated packet forwarding process for different types of IP traffic both in packet enqueuing/dequeuing processes. Depending on packet sizes and packet forwarding priorities of IP traffic, the proposed queuing scheme assigns fix and adaptive time token thresholds dynamically to logical queuing buffers separately. With assigned time tokens, logical queuing buffers allow arrival IP packets to be enqueued in a differentiated way. Moreover, the proposed queuing scheme uses a transferred WRR dequeuing measure to enhance a differentiated packet forwarding process. The simulation results show that the proposed queuing scheme supports a differentiated packet forwarding process for different types of IP traffic. The differentiated packet forwarding performance supported by the proposed scheme is close to the IETF DiffServ scheme; this result shows that the proposed scheme can support differentiated packet forwarding performance for different types of IP traffic with a lower operation cost.     

Core-stateless fair queueing: a scalable architecture to approximate fair bandwidth allocations in high-speed networks

Router mechanisms designed to achieve fair bandwidth allocations, such as fair queueing, have many desirable properties for congestion control in the Internet. However, such mechanisms usually need to maintain state, manage buffers, and/or perform packet scheduling on a per-flow basis, and this complexity may prevent them from being cost-effectively implemented and widely deployed. We propose an architecture that significantly reduces this implementation complexity yet still achieves approximately fair bandwidth allocations. We apply this approach to an island of routers - that is, a contiguous region of the network - and we distinguish between edge routers and core routers. Edge routers maintain per-flow state; they estimate the incoming rate of each flow and insert a label into each packet based on this estimate. Core routers maintain no per-flow state; they use first-in-first-out packet scheduling augmented by a probabilistic dropping algorithm that uses the packet labels and an estimate of the aggregate traffic at the router. We call the scheme core-stateless fair queueing. We present simulations and analysis on the performance of this approach.     

论业务量相关性对排队性能的影响

探讨业务量的相关性对排队系统性能究竟会带来什么样的影响,对于高速网络中流量控制机制和资源分配算法具有非常重要的意义。该文采用了一种独特的方法,来分析业务量的相关性在一定的网络环境下对于排队系统性能的影响。仿真的结果显示:一定时间尺度范围内的长期相关性导致排队性能的恶化,而且这种影响对于增加缓冲区的大小并不十分敏感。而且,业务量序列的边际分布和Hurst参数仍不足以充分决定排队性能的好坏。     

On the characteristics of queueing and scheduling at encoding nodes for network coding

Synchronization can greatly influence the performance of network coding. In this paper, we shall investigate the synchronization issue based on the use of queueing theory. We shall first propose a queueing model, referred to as the classic model, to investigate the characteristics of the encoding process. It will be proved that given the packet arrival processes are stationary, i.e. the distribution of the arrival processes does not depend on time, and obey independently and identically Poisson distribution and that the encoding time is exponentially distributed, the output flow will be an asymptotically Poisson flow with the same parameter. Through simulation we shall show that the network is sensitive to the arrival rate of input flows and becomes unstable with the input queue size increasing to infinity. This indicates that the classic coding scheme would impose strict requirements on synchronization over the whole network. In order to address this, we shall propose a combined opportunistic scheduling and encoding (COSE) strategy, in which the classic coding scheme and the traditional forwarding algorithm are well integrated. Theoretical analysis and simulation will demonstrate that the COSE strategy is able to control the input queue sizes and keep the network operating in a stable state while maintaining a relatively high throughput, low blocking probability and small waiting delay under various levels of traffic load. Copyright © 2009 John Wiley & Sons, Ltd.     

A neural-based technique for estimating self-similar trafficaverage queueing delay

Estimating buffer latency is one of the most important challenges in the analysis and design of traffic control algorithms. In this paper a novel approach for estimating average queueing delay in multiple source queueing systems is introduced. The approach relies on the modeling power of neural networks in predicting self-similar traffic patterns in order to determine the arrival rate and the packet latency of low loss, moderately loaded queueing systems accommodating such traffic patterns     

Admission control in time-slotted multihop mobile networks

The emergence of nomadic applications have generated a lot of interest in next-generation wireless network infrastructures which provide differentiated service classes. So it is important to study how the quality of service (QoS), such as packet loss and bandwidth, should be guaranteed. To accomplish this, we develop am admission control scheme which can guarantee bandwidth for real-time applications in multihop mobile networks. In our scheme, a host need not discover and maintain any information of the network resources status on the routes to another host until a connection request is generated for the communication between the two hosts, unless the former host is offering its services as an intermediate forwarding station to maintain connectivity between two other hosts. This bandwidth guarantee feature is important for a mobile network to interconnect wired networks with QoS support. Our connection admission control scheme can also work in a stand-alone mobile ad hoc network for real-time applications. This control scheme contains end-to-end bandwidth calculation and bandwidth allocation. Under such a scheme, the source is informed of the bandwidth and QoS available to any destination in the mobile network. This knowledge enables the establishment of QoS connections within the mobile network and the efficient support of real time applications. In the case of ATM interconnection, the bandwidth information can be used to carry out an intelligent handoff between ATM gateways and/or to extend the ATM virtual circuit service to the mobile network with possible renegotiation of QoS parameters at the gateway. We examine via simulation the system performance in various QoS traffic flows and mobility environments     

面向软件定义核心网的OpenFlow分组转发优先制排队模型研究

软件定义网络（Software-Defined Networking,SDN）作为一种数据转发与控制逻辑相解耦、并开放底层编程接口的创新网络架构,为降低核心网的部署运营成本、提升应用业务性能提供了全新的解决思路.然而,在SDN架构下,逻辑上集中的控制平面容易出现性能瓶颈,进而加大分组转发时延,因此有必要理解其分组转发性能特性.为此,本文首先介绍了软件定义核心网的典型部署场景,分析了控制平面的Packet-in消息到达过程和数据平面的分组到达过程,进而应用M/M/n/m和M/M/1/m排队模型分别刻画控制器集群的Packet-in消息处理过程和OpenFlow交换机的分组处理过程.在此基础上,建立OpenFlow分组转发优先制排队模型,进而推导出不同优先级的分组转发时延及其累积分布函数CDF.最后,借助控制器性能测量工具OFsuite_Performance进行实验评估,结果表明：与现有模型相比,本文所提的M/M/n/m模型更能准确估计控制器集群的实际性能.同时,采用数值分析的方法对比了多种情况下不同优先级的分组转发时延及CDF曲线,为软件定义核心网的实际应用部署提供有效参考.     

End-to-End Adaptive QoS Provisioning over GPRS Wireless Mobile Network

The General Packet Radio Service (GPRS) offers performance guaranteed packet data services to mobile users over wireless frequency-division duplex links with time division multiple access, and core packet data networks. This paper presents a dynamic adaptive guaranteed Quality-of-Service (QoS) provisioning scheme over GPRS wireless mobile links by proposing a guaranteed QoS media access control (GQ-MAC) protocol and an accompanying adaptive prioritized-handoff call admission control (AP-CAC) protocol to maintain GPRS QoS guarantees under the effect of mobile handoffs. The GQ-MAC protocol supports bounded channel access delay for delay-sensitive traffic, bounded packet loss probability for loss-sensitive traffic, and dynamic adaptive resource allocation for bursty traffic with peak bandwidth allocation adapted to the current queue length. The AP-CAC protocol provides dynamic adaptive prioritized admission by differentiating handoff requests with higher admission priorities over new calls via a dynamic multiple guard channels scheme, which dynamically adapts the capacity reserved for dealing with handoff requests based on the current traffic conditions in the neighboring radio cells. Integrated services (IntServ) QoS provisioning over the IP/ATM-based GPRS core network is realized over a multi-protocol label switching (MPLS) architecture, and mobility is supported over the core network via a novel mobile label-switching tree (MLST) architecture. End-to-end QoS provisioning over the GPRS wireless mobile network is realized by mapping between the IntServ and GPRS QoS requirements, and by extending the AP-CAC protocol from the wireless medium to the core network to provide a unified end-to-end admission control with dynamic adaptive admission priorities.     

A low-state packet marking framework for approximate fair bandwidth allocation

Misbehaving, non-congestion-reactive traffic is on the rise in the Internet. One way to control misbehaving traffic is to enforce local fairness among flows. Locally fair policies, such as fair-queueing and other fair AQM schemes, are inadequate to simultaneously control misbehaving traffic and provide high network utilization. We thus need to enforce globally fair bandwidth allocations. However, such schemes have typically been stateful and complex to implement and deploy. In this letter, we present a low state, lightweight scheme based on stateless fair packet marking at network edges followed by RIO queueing at core nodes, to control misbehaving flows with more efficient utilization of network bandwidth. Additionally, with low-state feedback from bottleneck routers, we show that, in practice, we can approximate global max-min fairness within an island of routers. We show, using simulations, that we can indeed control misbehaving flows and provide more globally fair bandwidth allocation.     

Performance analysis of hierarchical cellular networks with queueing and user retrials

How to efficiently utilize the scarce radio channel resource while maintaining the desired user‐perceived quality level and improved network performance is a major challenge to a wireless network designer. As one solution to meet this challenge in cellular mobile networks, a network architecture with hierarchical layers of cells has been widely considered. In this paper, we study the performance of a hierarchical cellular network that allows the queueing of both overflow slow‐mobility calls (from the lower layer microcells) and macrocell handover fast‐mobility calls that are blocked due to lack of free resources at the macrocell. Further, to accurately represent the wireless user behaviour, the impact of call repeat phenomenon is considered in the analysis of new call blocking probability. Performance analysis of the hierarchical cellular structure with queueing and call repeat phenomenon is performed using both analytical and simulation techniques. Numerical results show that queueing of calls reduces forced call termination probability and increases resource utilization with minimal call queueing delay. It is also shown that ignoring repeat calls leads to optimistic estimates of new call blocking probability especially at high offered traffic. Copyright © 2005 John Wiley & Sons, Ltd.     

A survey and comparative study of QoS aware broadcasting techniques in VANET

Mobile operators currently encounter numerous challenges caused by the centralized architecture of mobile networks. A single mobility anchor placed at the network core maintains the entire mobility and data traffic forwarding in the existing centralized mobility management (CMM) solutions. The CMM approach confronts several issues in scalability, reliability, signaling overhead, and non-optimal routing due to the increasing number of mobile devices and the volume of data traffic. To overcome these issues, a new architectural paradigm called distributed mobility management (DMM) is proposed to flatten the network architecture by moving mobility anchors closer to users and separating the control and data planes at the network edge. Two DMM solutions are developed: partially distributed mobility management (partial-DMM) in which only the data plane is distributed and fully distributed mobility management (full-DMM) where both control and data planes are distributed, which can be potentially applied for future mobile networks. This paper presents a network-based full-DMM scheme that was developed and implemented using NS2 network simulator by removing any dedicated centralized mobility anchor from the architecture. Extensive simulations were conducted to evaluate and compare the performance of the full-DMM model with that of the traditional CMM model. The simulation results show that the full-DMM provides lower end-to-end delay performance than CMM. However, the full-DMM generates higher handover latency and packet loss than CMM at high MN speeds. Moreover, simulation results clearly show the benefits of dynamic mobility activation in the full-DMM model.     

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.     

Network delay analysis of a class of fair queueing algorithms

A self-clocked fair queueing (SCFQ) scheme has been proposed by Golestani (see Proc. IEEE INFOCOM, p. 636-636, 1994) as an easily implementable version of fair queueing. In this paper, the worst case network delay performance of a class of fair queueing algorithms, including the SCFQ scheme, is studied. We build upon and generalize the methodology developed by Parekh and Gallager (see ACM/IEEE Trans. Networking, vol.1, no.3, p.344-357, 1993, and vol.2, no.2, p.137-150, 1994) to study this class of algorithms based on the leaky-bucket characterization of traffic. Under modest resource allocation conditions, the end-to-end session delays and backlogs corresponding to this class of algorithms are shown to be bounded. For the SCFQ scheme, these bounds are larger, but practically as good as the corresponding bounds for the PGPS scheme. It is shown that the SCFQ scheme can provide adequate performance guarantees for the delay-sensitive traffic in ATM     

Byzantine robust trust establishment for mobile ad hoc networks

In a mobile ad hoc network (MANET), the nodes act both as traffic sources and as relays that forward packets from other nodes along multi-hop routes to the destination. Such networks are suited to situations in which a wireless infrastructure is unavailable, infeasible, or prohibitively expensive. However, the lack of a secure, trusted infrastructure in such networks make secure and reliable packet delivery very challenging. A given node acting as a relay may exhibit Byzantine behavior with respect to packet forwarding, i.e., arbitrary, deviant behavior, which disrupts packet transmission in the network. For example, a Byzantine node may arbitrarily choose to drop or misroute a certain percentage of the packets that are passed to it for forwarding to the next hop. In earlier work, we proposed a trust establishment framework, called Hermes, which enables a given node to determine the “trustworthiness” of other nodes with respect to reliable packet delivery by combining first-hand trust information obtained independently of other nodes and second-hand trust information obtained via recommendations from other nodes. A deficiency of the Hermes scheme is that a node can fail to detect certain types of Byzantine behavior, such as packet misforwarding directed at a particular source node. In this paper, we propose new mechanisms to make Hermes robust to Byzantine behavior and introduce a punishment policy that discourages selfish node behavior. We present simulation results that demonstrate the effectiveness of the proposed scheme in a variety of scenarios involving Byzantine nodes that are malicious both with respect to packet forwarding and trust propagation.     

Research on distributed mobile management of satellite network based on software define network

Distributed mobility management (DMM) was an effective method to solve the mobile address handover.Therefore,it was considered to be a technology that can be applied to satellite network mobility management.A distributed mobile management scheme which based on software definition network (SDN) was proposed to solve the traffic redirection problem in satellite network.Different from the traditional DMM application scenario which was network-based or terminal-based,the SDN-DMM scheme implements location management and address handover in SDN controllers.Therefore,SDN-based satellite network distributed mobile management scheme can realize packet forwarding path optimization compared with traditional scheme,and it shows significant advantages in managementcost and traffic management.     

Experimental queueing analysis with long-range dependent packettraffic

Traffic measurement studies from a wide range of working packet networks have convincingly established the presence of significant statistical features that are characteristic of fractal traffic processes, in the sense that these features span many time scales. Of particular interest in packet traffic modeling is a property called long-range dependence (LRD), which is marked by the presence of correlations that can extend over many time scales. We demonstrate empirically that, beyond its statistical significance in traffic measurements, long-range dependence has considerable impact on queueing performance, and is a dominant characteristic for a number of packet traffic engineering problems. In addition, we give conditions under which the use of compact and simple traffic models that incorporate long-range dependence in a parsimonious manner (e.g., fractional Brownian motion) is justified and can lead to new insights into the traffic management of high speed networks     

Load balancing geographic routing strategy for aeronautical ad hoc networks

In aeronautical ad hoc networks, the traditional greedy perimeter stateless routing (GPSR) protocol poses sev-eral issues. For example, it is difficult to adapt to the highly-dynamic network environment, and it is prone to cause con-gestions. In order to address the problems, a TTE (time to enter the communication range of the destination)-based load balancing geographic routing (LBGR) protocol was presented. Taking TTE as the main routing decision metrics, this pro-tocol included the TTE-based packet forwarding scheme, multi-path traffic allocation scheme, and local optimum han-dling scheme. Furthermore, the multi-path traffic allocation scheme employing the queueing theory was modeled, and the mathematical expressions of some metrics were derived, such as the mean queue size, mean number of packets waiting in the queue, and mean waiting time. Finally, the analysis of the OMNeT++ simulations shows LBGR protocol has advan-tages over GPSR and some other protocols in terms of the packet delivery ratio and end-to-end delay, and is more suitable for the highly-dynamic aeronautical environment.