Pricing network resources for adaptive applications

The Differentiated Services framework (DiffServ) has been proposed to provide multiple Quality of Service (QoS) classes over IP networks. A network supporting multiple classes of service also requires a differentiated pricing structure. In this work, we propose a pricing algorithm in a DiffServ environment based on the cost of providing different levels of services, and on long-term average user resource demand of a service class. We integrate the proposed service-dependent pricing scheme with a dynamic pricing and service negotiation environment by considering a dynamic and congestion-sensitive pricing component. Pricing network services dynamically based on the level of service, usage, and congestion allows a more competitive price to be offered, allows the network to be used more efficiently, and provides a natural and equitable incentive for applications to adapt their service requests according to network conditions. We also develop the demand behavior of adaptive users based on a physically reasonable user utility function. Simulation results show that a congestion-sensitive pricing policy coupled with user rate adaptation is able to control congestion and allows a service class to meet its performance assurances under large or bursty offered loads, even without explicit admission control. Users are able to maintain a stable expenditure, and allowing users to migrate between service classes in response to price increases further stabilizes the individual service prices. When admission control is enforced, congestion-sensitive pricing still provides an advantage in terms of a much lower connection blocking rate at high loads.     

异构无线网络中新的成本感知网络切换方案

针对异构无线网络中用户希望通过较低的花费获取最满意的网络服务的需求,本文提出了一种以用户为中心的成本感知网络切换方案.首先,根据用户能够获取到的网络传输速率建立用户服务满足度模型,并结合归一化的网络费用,将网络切换问题转化为一个多目标优化问题.考虑到网络阻塞是由于大量用户同时选择同一个网络导致的,将这种背景下的用户决策行为转化为EI Farol酒吧问题,通过求解该问题纳什均衡状态下的用户获取网络服务的概率来得到用户服务满足度的期望值.最后,通过考虑用户服务满足度回报率将所构建的多目标优化问题转化为一个最大化问题.仿真结果表明本文算法能够提高网络总吞吐量,降低用户切换阻塞率,使网络负载更加均衡.     

Cross-Layer optimization for LDPC-coded multirate multiuser systems with QoS constraints

In this paper, we propose a new multirate multiple-access wireless system implemented by variable spreading gain and chip-level random interleaving. The receiver employs a flexible chip-level iterative multiuser detection scheme where the variable spreading gain affects only the despreading parameters. Optimization across the physical and network layers in the uplink of such a system is treated. It is assumed that each user employs an low-density parity-check (LDPC) code to protect its data. At the physical layer, the quality of service (QoS) requirement is specified in terms of the target bit error rate (BER) of each user. Optimal user transmit powers are dynamically adjusted according to the current system load and the corresponding rate requirements. At the network layer, the QoS requirements include the call blocking probabilities, call connection delays, packet congestion probabilities and packet loss rates. To maximize the average revenue of the network subject to both call-level and packet-level QoS constraints, a multicriterion reinforcement learning (MCRL)-based adaptive call admission control (CAC) method is proposed that can easily handle multiple average QoS requirements. Unlike existing model-based approaches, the MCRL-based technique does not require the explicit knowledge of the state transition probabilities to derive the optimal policy. This feature is important when the number of states is so large that model-based optimization algorithms become infeasible, which is typically the case for a large integrated service network supporting a number of different service types.     

Network Functionality for Supporting Adaptive Applications over Varying Radio Conditions

The purpose of this paper is to analyze the concept of the Bandwidth Mediation Module (BMM). The BMM is a proposed network functionality which records the status of the mobile network (in terms of congestion, noise levels, etc.) and provides input to the requesting Application Server. This network status information is processed in order for the available network resources to be identified and presented in a uniform way. The outcome is further utilized by the provided services which are then adapted accordingly so as to be efficiently delivered over the varying network traffic conditions. The term “Adaptive Applications” means that the applications adjust their functionalities according to the available network resources. The BMM communicates network status information to the Application Server in order to constantly enhance the end user service perception and relieve the network from high congestion.     

Comparison of different congestion control strategies for low priority controllable traffic in packet switched backbone networks

It is expected that a significant part of the data flows of future multi‐service packet switched backbone networks will use low priority, non‐real‐time data transmission services of the networks. The common benefit for both user applications and network operators is that the data flows of the low priority services could use the free capacity of the networks, after the load of higher priority data flows. Congestion control methods are needed for these low priority data flows to reach an optimal utilisation level of the networks, high throughput and low packet loss ratios. This kind of low priority data transmission service which adjusts the data rates of the data flows according to the data rate changes of higher priority data flows, but does not guarantee any specific service for these data flows, is called a controlled load service. In this paper, we have compared the performance, efficiency and scalability of four different congestion control methods designed for the controlled load service. Two of these methods were based on very simple congestion control algorithms and the other two used relatively complex control algorithms based on control methods utilising computational intelligence. The principal aim of this study was to research how remarkable were the effects that the different complexities of the congestion control methods had on the achieved level of service. The simulation tests indicate that the complexity of the methods clearly affects the performance and efficiency of the methods. Copyright © 2001 John Wiley & Sons, Ltd.     

Cross‐layer and cognitive QoS management system for next‐generation networking

Owing to limited wireless network resources, network applications must provide an adaptive quality‐guaranteed service to satisfy user requirements. Different applications are associated with different quality of service (QoS) concerns, as well as different QoS control parameters. This work presents an adaptive QoS algorithm by discussing the QoS specifications of three wireless access technologies, i.e. 3G, WiMAX and WiFi. Based on cross‐layer and cognition concepts, these environmental parameters are integrated with the sensing of spectral and received signal strength from a cognitive radio paradigm. An adaptive QoS algorithm is then proposed to select the optimal access network for services. Simulation results indicate that the proposed adaptive QoS algorithm outperforms available ones in real‐time applications. Compared with traditional algorithms, the proposed algorithm reduces not only the average delay time and jitter for VoIP services to 0.16 s and 0.09 ms, respectively, but also the packet loss ratio for high‐definition video streaming by 3.4%. Copyright © 2011 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。     

基于Q学习的网间移动号码携带拥塞控制方法(英文)

Inter-networking Mobile Number Portability(MNP) is an important way to promote the development of new telecom services.For the congestion that always occurs on the MNP query nodes,this paper proposes a Q-learning-based inter-networking mobile number portability congestion control mechanism,taking the node queue delay and query service stability as the control targets,based on the optimal decision which is made over mobile user number portability query through self-adaptive connection admission control.The s...     

Characterizing user‐perceived impairment events using end‐to‐end measurements

Measures of quality of service (QoS) must correlate to end‐user experience. For multimedia services, these metrics should focus on the phenomena that are observable by the end‐user. Metrics such as delay and loss may have little direct meaning to the end‐user because knowledge of specific coding and/or adaptive techniques is required to translate delay and loss to the user‐perceived performance. Impairment events, as defined in this paper, are observable by the end‐users independent of coding, adaptive playout or packet loss concealment techniques employed by their multimedia applications. Time between impairments and duration of impairments are metrics that are easily understandable by a network user. Methods to detect these impairment events using end‐to‐end measurements are developed here. In addition, techniques to identify Layer 2 route changes and congestion events using end‐to‐end measurements are also developed. These are useful in determining what caused the impairments. End‐to‐end measurements were conducted for about 26 days on 9 different node pairs to evaluate the developed techniques. Impairments occurred at a high rate on the two paths on which congestion events were detected. On these two paths, congestion occurred for 6–8 hours during the day on weekdays. Impairments caused by route changes were rare but lasted for several minutes. Copyright © 2005 John Wiley & Sons, Ltd.     

Cross-Layer design in HSDPA system to reduce the TCP effect

This paper focuses on the interaction between the transport control protocol (TCP) layer and the radio interface in the high-speed downlink packet access (HSDPA) wireless system. In the literature, studies of the interaction between TCP and wireless networks are focused on the evaluation of user bit rate in the case of dedicated channels. In this paper, the interaction between TCP, hybrid automatic repeat request (HARQ), and scheduling techniques (especially, proportional fair scheduling) is conducted. Analytical models to evaluate HSDPA cell capacity, user bit rate, and interaction with TCP layer are presented. Even if as expected the bit rate per flow decreases strongly with the congestion frequency in the wired network, it is shown that the overall capacity achieved by HSDPA is not as affected by the TCP layer. Using this result, a method to reduce the effect of TCP on wireless network without losing much cell capacity is proposed. This method has the advantage of modifying the scheduling algorithm only and of not requiring any change to the TCP protocol.     

Towards Robust Multi-Layer Traffic Engineering: Optimization of Congestion Control and Routing

In the Internet today, traffic engineering is performed assuming that the offered traffic is inelastic. In reality, end hosts adapt their sending rates to network congestion, and network operators adapt the routing to the measured traffic. This raises the question of whether the joint system of congestion control (transport layer) and routing (network layer) is stable and optimal. Using the established optimization models for TCP and traffic engineering as a basis, we find the joint system can be stabilized and often maximizes aggregate user utility. We prove that both stability and optimality of the joint system can be guaranteed for sufficiently elastic traffic simply by tuning the cost function used for traffic engineering. Then, we present a new algorithm that adapts on a smaller timescale to changes in traffic distribution and is more robust to large traffic bursts. Uniting the network and transport layers in a multi-layer approach, this algorithm, distributed adaptive traffic engineering (DATE), jointly optimizes the goals of end users and network operators and reacts quickly to avoid bottlenecks. Simulations demonstrate that DATE converges quickly     

Service convergence using MPLS multiservice networks

Enterprises are increasingly using virtual private networks to interconnect remote sites. Traditionally, service providers have used ATM core networks to deliver layer 2 services such as frame relay, ATM, or TDM private lines, which enterprise customers have then used to build their corporate network infrastructure. Such services account for the majority of data service revenues today. However, pressure has increased on service providers to combine increased flexibility with reduced costs in the context of a highly dynamic telecommunications market. Service providers also need to generate new revenues from their IP network infrastructure, through new opportunities such as IP VPNs and virtual private LAN services, while simultaneously achieving operational efficiencies through the convergence of all of their services on a common MPLS backbone. New access and metro network technologies, such as Ethernet, are also emerging that can be used to deliver these new services to enterprise customers alongside ATM and frame relay access. This must be achieved while also supporting existing technologies such as ATM, which continue to deliver highly profitable services. This article discusses the technical challenges in meeting the often conflicting requirements of delivering both traditional layer 2 services and new layer 3 services on a converged MPLS network. We show how both network and service interworking are required, and how these must operate at the user, control, and management planes to enable profitable services to be delivered over the new converged network. The different solutions being defined in the standards bodies are described, and the distinct scenarios they address are explained.     

业务优先级感知的网络流量调度机制

针对网络中业务数据流过大、分布不均匀所造成的网络拥塞,提出一种优先级感知的动态网络流量调度机制.利用令牌桶算法,根据业务优先级的不同为不同业务分配不同速率的令牌,以实现业务优先级的划分;综合考虑业务的优先级及用户节点剩余缓存空间,对不同的业务采取不同的处理方式;同时,以流量到达因素、服务因素及节点缓存为指标定义了一种网络流量调度机制性能指标——分组丢失率.数值结果表明,所提机制可以对网络中业务优先级进行合理的划分,从而有效利用网络资源,预防网络拥塞,提升网络性能,为用户提供更加稳定可靠的网络服务.     

QoS control in wireless ATM

This paper introduces a 3-level multiagent architecture for QoS control in WATM. The ultimate aim of the proposed architecture is to provide a self-regulating network congestion control management by means of global network state awareness and agent interactions. The agents dynamically manage the buffer space at the level of a switch and interact to reduce the cell loss ratio while guaranteeing a bounded transit delay. We particularly address video transmission over UBR services using a per-VP queuing approach and an adaptive cell discarding congestion control scheme. Furthermore, a dynamic reconfiguration of the agents is performed during handoffs in order to continue meeting user end-to-end QoS requirements. The handoff delay absorption is also addressed. This revised version was published online in June 2006 with corrections to the Cover Date.     

Script MIB在虚拟主动网络服务管理中的应用

主动网络是一种新型的可计算网络体系结构，在国内外已作为下一代网络可能模型的研究热点。首先介绍主动网络及虚拟主动网络，利用虚拟主动网络，能够有效地加快网络服务的开发和部署，与此同时，用户需要灵活地管理自己提供的服务。因此，提出一种使用Script MIB对虚拟主动网络的服务进行有效的、灵活的管理模型，能够动态地实现对主动网络服务的管理。     

基于软交换的个人监控通信

网络的融合及终端智能化程度的提高，为个人监控通信的实现提供了条件。借助于M2M技术，软交换网络可以为个人监控通信提供理想的实现平台。基于软交换的网络架构和实现原理，个人监控通信系统可分为网络侧和用户侧两部分。网络侧设备包括软交换机、现场业务中心等，位于核心控制层和业务层；用户侧设备包括监控设备、现场设备、网关和现场控制平台，位于接入层和现场设备层。通过基于软交换的个人监控通信系统可以实现家庭自动化、公共服务、资源调度、安全监测以及社群互助等多种人机通信业务。随着机器智能的提高和网络的融合，将来个人监控还会不断完善和发展，极有可能成为未来通信的主流业务之一。     

A congestion control framework for ATM networks

The authors propose a congestion control framework for ATM networks that requires simple protocols and that provides guaranteed grade of service The authors suggest that the network provides two different services: express service and first-class service. Express service is appropriate for real-time applications, whereas first-class service is appropriate for nonreal-time applications. To provide such services, the proposed congestion control scheme prevents congestion inside the network by controlling the congestion at two levels: VC (virtual circuit) level and cell level. The VC-level control takes place during the call setup in the form of (logical) bandwidth reservation for individual VCs. The cell-level control is exercised across the network interface in the form of the choking/relieving of only first-class VC traffic in case of congestion. The authors analyze the performance of the proposed scheme, and show how to engineer various parameters to satisfy cell-level performance requirements (e.g., cell delay and loss)     

Internet connection with UMTS

Mobile telecommunication new services are based on data networks specially Internet. These services include http, telnet, ftp, Simple Mail Transfer Protocol, etc. Besides, we recognize a mobile network as a multiuser network. Transmission Control Protocol (TCP)/Internet Protocol which is sensitive to link congestion in wireline data links is also used in wireless networks. In order to improve the system performance, the TCP layer uses flow control and congestion control. Besides, radio link control (RLC) and medium access control sublayers have been introduced to compensate the deficiency of TCP layer in wireless environment. RLC has an important role in quality of service enhancement of the Universal Mobile Telecommunications System (UMTS). In this paper, we review the protocol stack of UMTS Terrestrial Radio Access Network which is based on Third-Generation Partnership Project. Then, we evaluate its layer 2 error control mechanisms and verify TCP over automatic repeat request error control mechanism and finally quality of service improvement results from it in fading channels.     

Adaptive predictive congestion control of high-speed ATM networks

This paper proposes an auto regressive moving average (ARMAX)-based adaptive control methodology to prevent congestion in high-speed asynchronous transfer mode (ATM) networks. An adaptive controller is developed to control traffic where sources adjust their transmission rates in response to the feedback information from the network switches. Specifically, the buffer dynamics at a given switch is modeled as a nonlinear discrete-time system and an ARMAX controller is designed so as to predict the explicit values of the transmission rates of the sources so as to prevent congestion. Tuning methods are provided for the unknown coefficients of the ARMAX model to estimate the unpredictable and statistically fluctuating network traffic. Mathematical analysis is given to demonstrate the stability of the closed-loop system so that a desired quality of service (QoS) can be guaranteed. The QoS is defined in terms of cell loss ratio (CLR), transmission delay and buffer utilization. We derive design rules mathematically for selecting the parameters of the ARMAX algorithm such that the desired performance is guaranteed during congestion and potential tradeoffs are shown. Simulation results are provided to justify the theoretical conclusions for multiple source/single switch scenarios using both ON/OFF and MPEG data. The performance of the proposed congestion control scheme is also evaluated in the presence of feedback delays for robustness considerations.     

Max-utility subcarrier allocation for heterogeneous services in wireless OFDM system

In future wireless network, one user will require multiple homogeneous or heterogeneous services simultaneously. Then, the scheduling algorithm is not only responsible for assigning a resource block to different users but also sharing the assigned resource block among multiple services for one user. Most of the traditional scheduling algorithms are designed to serve one service per user, and cannot be applied directly to this scenario because of the fairness criterion. This article focuses on adaptive resource allocation for multiple services per user at the downlink of orthogonal frequency division multiplexing (OFDM) based system. This article addresses this integrative resource scheduling problem based on utility function. First, the optimal algorithm for dynamic subcarrier allocation and share is deduced for homogeneous best-effort service system. Then the algorithm is extended to heterogeneous services system by classifying the delay sensitive service according to the head-of-line packet delay. The design goal is to maximize aggregate utility function to exploit multiuser diversity gain to the greatest extent even as guaranteeing quality of service (QoS) for delay sensitive service.