Supporting SIP-based end-to-end Data Distribution Service QoS in WANs

Assuring end-to-end QoS in enterprise distributed real-time and embedded (DRE) systems is hard due to the heterogeneity and transient behavior of communication networks, the lack of integrated mechanisms that schedule communication and computing resources holistically, and the scalability limits of IP multicast in wide-area networks (WANs). This paper makes three contributions to research on overcoming these problems in the context of enterprise DRE systems that use the OMG Data Distribution Service (DDS) quality-of-service (QoS)-enabled publish/subscribe (pub/sub) middleware over WANs. First, it codifies the limitations of conventional DDS implementations deployed over WANs. Second, it describes a middleware component called Proxy DDS that bridges multiple, isolated DDS domains deployed over WANs. Third, it describes the NetQSIP framework that combines multi-layer, standards-based technologies including the OMG-DDS, Session Initiation Protocol (SIP), and IP DiffServ to support end-to-end QoS in a WAN and shield pub/sub applications from tedious and error-prone details of network QoS mechanisms. The results of experiments using Proxy DDS and NetQSIP show how combining DDS with SIP in DiffServ networks significantly improves dynamic resource reservation in WANs and provides effective end-to-end QoS management.     

基于DDS的发布/订阅中间件设计

基于OMG组织颁布的实时系统中数据分发服务最新标准设计了一个新的发布/订阅中间件，实现了分布式环境中以数据为中心的数据实时传输，解决了分布式实时系统中服务质量约束和单点失效等问题。提出一种全局数据空间管理策略，解决了通信节点动态增删问题。该模型与CORBA规范和其他发布/订阅模型相比，具有较好的灵活性、实时性、可扩展性和可靠性。     

End-to-end support for statistical quality of service in heterogeneous mobile ad hoc networks

In heterogeneous mobile ad hoc networks (MANETs), different types of mobile devices with diverse capabilities may coexist in the same network. The heterogeneity of MANETs makes end-to-end support for quality of service (QoS) guarantees more difficult than in other types of networks, not to mention the limited bandwidth and frequent topology changes of these networks. Since QoS routing is the first step toward achieving end-to-end QoS guarantees in heterogeneous MANETs, we propose a QoS routing protocol for heterogeneous MANETs. The proposed protocol, called virtual grid architecture protocol (VGAP), uses a cross-layer approach in order to provide end-to-end statistical QoS guarantees. VGAP operates on a fixed virtual rectilinear architecture (virtual grid), which is obtained using location information obtained from global positioning system (GPS). The virtual grid consists of a few, but possibly more powerful, mobile nodes known as ClusterHeads (CHs) that are elected periodically. CHs discover multiple QoS routes on the virtual grid using an extended version of the open shortest path first (OSPF) routing protocol and an extended version of WFQ scheduling policy that takes into account the wireless channel state. Moreover, VGAP utilizes a simple power control algorithm at the physical layer that provides efficient energy savings in this heterogeneous setting. Simulation experiments show that VGAP has a good performance in terms of packet delivery ratio, end-to-end packet delay, call blocking probability, and network scalability.     

大规模泛在RFID网络的构造方法*

依据RFID网络特征和行为的形式化描述,基于发布/订阅系统构建了RFID网络模型。RFID网络是构建在IP网络层之上的重叠网,借助代理(broker)的消息存储转发、匹配和路由机制,实现阅读事件的多对多异步通信,阅读事件与应用之间无须预先确定相关关系。实验验证了这种网络模型的有效性,可以满足将来大规模泛在RFID应用的需求。     

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.     

Model driven middleware: A new paradigm for developing distributed real-time and embedded systems

Distributed real-time and embedded (DRE) systems have become critical in domains such as avionics (e.g., flight mission computers), telecommunications (e.g., wireless phone services), tele-medicine (e.g., robotic surgery), and defense applications (e.g., total ship computing environments). These types of system are increasingly interconnected via wireless and wireline networks to form systems of systems. A challenging requirement for these DRE systems involves supporting a diverse set of quality of service (QoS) properties, such as predictable latency/jitter, throughput guarantees, scalability, 24x7 availability, dependability, and security that must be satisfied simultaneously in real-time. Although increasing portions of DRE systems are based on QoS-enabled commercial-off-the-shelf (COTS) hardware and software components, the complexity of managing long lifecycles (often ∼15-30 years) remains a key challenge for DRE developers and system integrators. For example, substantial time and effort is spent retrofitting DRE applications when the underlying COTS technology infrastructure changes.This paper provides two contributions that help improve the development, validation, and integration of DRE systems throughout their lifecycles. First, we illustrate the challenges in creating and deploying QoS-enabled component middleware-based DRE applications and describe our approach to resolving these challenges based on a new software paradigm called Model Driven Middleware (MDM), which combines model-based software development techniques with QoS-enabled component middleware to address key challenges faced by developers of DRE systems — particularly composition, integration, and assured QoS for end-to-end operations. Second, we describe the structure and functionality of CoSMIC (Component Synthesis using Model Integrated Computing), which is an MDM toolsuite that addresses key DRE application and middleware lifecycle challenges, including partitioning the components to use distributed resources effectively, validating software configurations, assuring multiple simultaneous QoS properties in real-time, and safeguarding against rapidly changing technology.     

发布订阅系统在移动Ad Hoc网络中的三种路由算法

介绍了分布式发布/订阅系统的原型，为Ad Hoc网络中的发布/订阅系统设计了三种基于内容的路由算法，分别说明了它们的运行方式，并进行了仿真与比较。结果表明：基于内容的可靠容错路由算法能够100%保证信息的传输，有容错机制，可靠性较强，但出现了传输延迟；基于内容的路由算法的可靠性不强，适合有线网络中分布式发布订阅系统的信息传输，传输延迟小；基于内容的容错路由算法有容错机制，传输延迟小     

Interworking of heterogeneous access networks and QoS provisioning via IP multimedia core networks

Moving towards packet networks, where IP will have a prominent role, constitutes nowadays a widely accepted perception of future communications, the first instance of which has begun to materialise with the IP multimedia core network subsystem (IMS). By specification, IMS is the first implementation towards reaching converged communications which allows users to communicate with video, audio and multimedia content, via any fixed, mobile and wireless access network type, with controllable QoS. To enable IMS communications across heterogeneous networks, incorporating UMTS, WLAN and fixed IP access points, 3GPP and ETSI’s TISPAN currently work on schemes for controlling bandwidth allocation at the service level by employing logical interfaces that carry SIP messages. This article analyzes how interconnection between such heterogeneous networks may be performed on real platforms. In this effort, special attention is paid to the way the various interconnection possibilities can affect end-to-end QoS provisioning.     

An analytical QoS service model for delay-based differentiation

The increased need for Quality of Service (QoS) in today’s IP networks has concentrated a lot of research and implementation efforts. Carefully designed and managed priority services are essential for quality-demanding traffic, especially in large-scale IP-based environments where aggregation of flows is extensive and a variety of traffic types co-exist. Although individual mechanisms have been widely investigated, not much related work exists on integrated approaches to QoS provisioning that are also feasible to implement. This work presents a thorough approach to the design, dimensioning and provisioning of a high-priority service for high-quality-demanding traffic over an IP network. Our approach employs efficient scheduling and a dynamic admission control scheme while demonstrating an outstanding performance in terms of the quality offered to QoS-demanding traffic. A novel feature is also introduced: while other existing schemes only focus on provisioning of service rate guarantees, ours achieves in addition differentiation of the end-to-end delay perceived by IP flows.     

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.     

基于DDS规范的航行通告分发服务

航行通告是以电信方式发布,有关任何航空设施、状况或变化的情报通告,及时了解此类通告对与飞行活动有关的航务人员是至关重要的。数据分发服务（Data Distribution Service,DDS）是对象管理组织（Object Management Group,OMG）颁布数据发布服务规范,它定义了以数据为中心的发布／订阅（Data-Centric Publish-Subscribe,DCPS）服务。为提高情报交换的效率,确保飞行的正常、安全与效益,文中将DDS规范运用到航行通告数据交换中,对航行通告分发服务的主题结构进行设计,并利用DDS丰富的QoS策略,控制航行通告数据的质量,基于DDS规范的航行通告分发服务可有效地避免单点失败,提高系统可靠性。     

Network convergence and QoS for future multimedia services in the VISION project

The emerging use of real-time 3D-based multimedia applications imposes strict quality of service (QoS) requirements on both access and core networks. These requirements and their impact to provide end-to-end 3D videoconferencing services have been studied within the Spanish-funded VISION project, where different scenarios were implemented showing an agile stereoscopic video call that might be offered to the general public in the near future. In view of the requirements, we designed an integrated access and core converged network architecture which provides the requested QoS to end-to-end IP sessions. Novel functional blocks are proposed to control core optical networks, the functionality of the standard ones is redefined, and the signaling improved to better meet the requirements of future multimedia services. An experimental test-bed to assess the feasibility of the solution was also deployed. In such test-bed, set-up and release of end-to-end sessions meeting specific QoS requirements are shown and the impact of QoS degradation in terms of the user perceived quality degradation is quantified. In addition, scalability results show that the proposed signaling architecture is able to cope with large number of requests introducing almost negligible delay.     

基于DDS的异构网络数据分发方法研究

在异构网络环境中,数据实现高效、可靠与安全的分发,面临着不同的应用环境对操作系统、通信方式、网络协议以及服务质量（ QoS）的多种要求。文中设计了一种基于DDS的,在异构网络通信中使用的数据分发系统。针对异构网络与DDS的特性,提出了异构网络数据分发系统的三层组织结构,并对每一层的具体内容,要实现的功能进行了研究。经过对异构网络中所传递的数据进行分析,结合OpenDDS本身的QoS策略推出该系统本身的QoS策略。最后,通过实验对该数据分发系统的网络延迟以及时延抖动进行了测试,表明了该系统能够有效满足异构网络之间的数据分发需求。     

Fast setup of end-to-end paths for bandwidth constrained applications in an IP/MPLS–ATM integrated environment

Transport networks are currently being moved towards a model of high performance Internet Protocol/Multiprotocol Label Switching (IP/MPLS) routers interconnected through intelligent core networks. Currently, Asynchronous Transfer Mode (ATM) technology has been widely deployed in several network backbones along with the Private Network-to-Network Interface (PNNI) protocols as the control plane. In order to cope with the increasing Internet traffic demands in the current context, fast setup of end-to-end paths with the required Quality of Service (QoS) is necessary.This paper analyzes the case of two IP/MPLS networks interconnected through an ATM core network, assuming MPLS as the mechanism to provide Traffic Engineering in the IP networks, and a PNNI-based control plane in the core network. This paper aims to define a mechanism needed to set up a fast end-to-end QoS Label Switched Path (LSP) between two Label Switched Routers (LSRs) belonging to different IP/MPLS domains. First, the fast end-to-end setup is achieved by modifying the network backbone control plane. Second, two different aggregation schemes are proposed to summarize the QoS network state information to be transported through the ATM core network. Therefore, both the efficient aggregation schemes and the fast mechanism allow source routing to set up a path faster than the existing methods and to reduce the blocking probability using a summary of the available resource information.     

QoS中间件中实时发布/订阅服务集成评估

支持QoS的中间件技术在构造分布式实时嵌入式系统中得到了广泛应用,已成为支持实时发布/订阅服务的关键技术.评估并分析了QoS中间件中3种集成实时发布/订阅服务方法.重点研究了容器管理方式的性能,并与面向对象的实时发布/订阅服务比较.研究结果表明,容器管理方式中,CIAO中间件的等待时间稍长,有可预测性,适用于DRE系统.     

一种基于时隙划分的三层卫星网络QoS路由协议

提出了一种基于低轨/中轨/高轨（LEO/MEO/GEO）卫星网络体系结构（TLSN）和时空分割思想的QoS路由协议（TDRP）,详细阐述了时空分割思想和时隙优化方法,重点研究了QoS路由的实现框架和路由算法（BDA）的实现原理.在此基础上,卫星网络体系结构和路由协议的性能得到了详细分析和仿真验证.相同负载情况下,TLSN中端到端时延和时延抖动性能明显好于LEO卫星网络;同Dijkstra算法相比,BDA算法在端到端时延、时延抖动、丢包率、吞吐量方面具有更好的QoS保证能力.     

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.     

Computational intelligence tools for next generation quality of service management

In this paper we explore the interest of computational intelligence tools in the management of heterogeneous communication networks, specifically to predict congestion, failures and other anomalies in the network that may eventually lead to degradation of the quality of offered services. We show two different applications based on neural and neuro-fuzzy systems for quality of service (QoS) management in next generation networks for voice and video service over heterogeneous Internet protocol (V2oIP) services. The two examples explained in this paper attempt to predict the communication network resources for new incoming calls, and visualizing the QoS of a communication network by means of self-organizing maps.     

High performance mobility without agent infrastructure for connection oriented service

Currently it is very difficult for connection oriented applications to use a mobile environment. One reason is that Mobile IP requires intermediate software agents to be deployed in the Internet. This infrastructure based mobility scheme offers connectivity to itinerant hosts but incurs significant handoff and tunneling delays along with deployment costs. These delays are particularly harmful for connection oriented applications. In this paper we investigate an alternate mobility scheme which does not require any such infrastructure but only uses an end-point technique and interestingly provides much faster loss-free handoff for connection oriented applications. This End-to-End scheme named Interactive Protocol for Mobile Networks (IPMN) intelligently performs handoff based on information provided by MAC Layer. The network address change is handled by renewing the existing connections by manipulating the TCP/IP stack at the end-points. Also, unlike several other recently proposed end-to-end techniques which require extensive modification of end-protocols, the proposed scheme does not require any functional change in the TCP/IP protocol software. Besides the difference in deployment scenarios, the IPMN offers blazingly fast event based handoff and much faster and simplified transport (no tunneling delay) than MIP. We have implemented IPMN over FreeBSD. In this paper we show the performance advantage of IPMN over MIP with real deployment for three interesting real-time traffic types — www, voice streaming and, steerable/interactive time critical video.