Opening up networks with JAIN Parlay

The desire for new business growth has been a major driving force towards the development of open network APIs, such as the Parlay API, within telecommunications networks. The Parlay API enables both third parties (external companies, operating outside the security domain of the network operator) and network operators to build new applications that rely on real-time control of network resources. The Java APIs for integrated networks (JAIN^TM) Community is defining a Java version of the Parlay API to bring the benefits of the Java language to the Parlay API, and to promote industry-wide adoption of the Parlay API. This article describes the background and rationale behind the work of the Parlay Group, together with the characteristics, structure, and capabilities of the Parlay API. The benefits that Java and the JAIN Community bring to the Parlay API are then explored. A technical overview of the Java version of the Parlay API, referred to as the JAIN Parlay Edit Group API, is given and illustrated using example sequence diagrams. The article concludes by taking a look at what additional features may be added to the API and the implementation activities that lay ahead     

基于JAIN SIP的多媒体会议系统中发言权控制

发言权控制(Floor Control)是多媒体会议中会议控制的一部分,用来协调参与者对底层共享资源的使用,它直接关系到会议的服务质量.文章利用会话初始协议(SIP)的可扩展性,在SUN公司为实现SIP应用而提供的一套Java接口(JAIN SIP)架构下设计并实现了SIP多媒体会议系统的发言权控制系统,该系统具有控制方式多样化,适用于分布式系统等特点.测试结果表明系统设计合理,并具有延迟较小的特点,满足分布式系统的要求.     

Ubiquitous Information Services with JAIN Platform

To provide responsive information services in a ubiquitous computing environment, service software and system development are indispensable. A component-based ubiquitous information system with a JAIN (Java APIs for Integrated Networks) platform to achieve seamless transmissions and reach-everywhere communications was designed in our research. In this development, numerous ubiquitous service modules were identified: location management, roaming, mobile IP and WAP networking. These component-based network modules were constructed using a component composition language with component specification and interface definition, running on top of a distributed service architecture using a JAIN platform to distribute the ubiquitous information services to mobile users. Two applications, Wireless-Application-Protocol (WAP) Mail and Video-on-Demand (VoD) services, show that the JAIN-like platform with the developed networking components effectively fills the gap for application developers between mobile appliances and various kinds of ubiquitous information services.     

JAIN——21世纪的网络接口

介绍了JAIN网络接口的基本结构并着重论述了JAIN协议层和JAIN呼叫控制应用层。由于JAVA语言是新一代面向对象的程序设计语言,具有平台的独立性和程序的可移植性等独特的优点,应用于通信领域后,为通信网的发展产生了巨大的推动作用。因此采用JAVA语言的JAIN网络系统,可以为21世纪通信环境的迅速变化和业务的迅速更新提供新的途径和方法。     

用于呼叫控制的JAIN SIP

综合网络的Java应用编程接口(JAIN,Java API for Integrated Networks)是一组基于Java技术的API,这些接口将业务可移植性、网络聚合以及安全的网络接入引进电话网和数据网络,使得在Java平台上快速开发下一代电信产品和服务成为可能。本文在简要介绍了JAIN之后,重点介绍了JAIN SIP API。     

JAIN: a new approach to services in communication networks

JAIN^TM, a set of integrated network APIs for the Java ^TM platform, provides a framework to build and integrate solutions (or “services”) that span across packet (e.g., IP or ATM), wireless, and PSTN networks. The objective of JAIN is to provide service portability, convergence, and secure access (by services residing outside of the network) to such integrated networks. JAIN is defined and specified by a large number of participating communication companies (the JAIN Community), and according to a well-documented process (the Java Community Process or JCP). The objective of the JAIN Community is to create an open market for services across integrated networks using Java technology. The authors provide the JAIN business case. They then summarize how the JAIN Community works, and introduce how the JAIN Community is organized. They also explain how separate JAIN work items fit together. In particular, they provide the rationale for the currently supported levels of abstraction (in terms of session/call signaling models) in JAIN, and look at possible implementation scenarios     

基于JAIN SIP的多媒体会议系统设计与实现

会话初始化(SIP)协议作为下一代网络中应用层的核心控制协议,正得到越来越广泛的关注.JAIN SIP是SUN公司用于实现SIP应用而提供的一套标准JAVA接口.本文基于JAIN SIP的体系结构和机制,设计并实现了SIP多媒体会议系统.该系统支持多种SIP终端接入,具有灵活的会议召开和加入方式.     

下一代网络开放业务接口的研究

下一代网络是基于业务驱动的网络,开放性是下一代网络的重要特征。在分析研究一些典型的开放接口技术基础上,针对Parlay X业务呼叫接口存在的缺陷,提出Parlay X增强型业务呼叫接口的改进方案,并给出其实现模型。通过使用增强型呼叫接口,应用开发商不需要掌握详细的电信知识,就能对呼叫进行持续并且有效的监控,并且以更为灵活的交互方式为用户提供丰富多彩的特殊资源。     

Formalizing REST APIs for web-based communication and SIP interworking

This paper proposes an instantaneous recovery route design scheme using multiple coding-aware link protection scenarios to achieve higher link cost reduction in the network. In this scheme, two protection scenarios, namely, (i) traffic splitting (TS), and (ii) two sources and a common destination (2SD) are used to integrate multiple sources and a common destination. The proposed scheme consists of two phases. In the first phase, the proposed scheme determines routes for 2SD and TS scenarios of all possible source-destination pairs to minimize the total link cost. In this phase, the network coding is applied to the common path within each scenario, individually. In the second phase, network coding is applied to the common path between two scenarios (or a scenario pair) in order to enhance the resource saving. This phase develops conditions that select the most appropriate combination of scenario pairs, such as TS–TS, 2SD–2SD, and 2SD–TS for network coding, including their proofs. Using these conditions, a heuristic algorithm is introduced in order to select the most appropriate combination of scenario pairs for further enhancing of resource saving. Simulation results show that the proposed scheme outperforms the conventional 1 + 1 protection scheme, the TS scenario, and the 2SD scenario in terms of link cost reduction in the network.     

基于JAIN SLEE的业务逻辑执行环境的设计与实现

本文简要分析了下一代业务逻辑执行环境(SLEE,Service Logic Execution Environment)JAIN SLEE规范,在此基础上给出了基于规范的SLEE设计与实现.     

一种利用Win32 API函数实现串口实时通迅的方法

介绍了一种在Windows 9x及以上操作系统下利用Win32 API函数实现串行通讯的方法，引入多线程、异步I／O操作及事件驱动机制，成功地解决了在32位视窗操作系统下计算机监控系统实时串行通讯问题，并给出了部分串口通讯的源代码。     

Opening the networks with Parlay/OSA: standards and aspects behind the APIs

Recently there has been an enormous increase in efforts to open up telecommunication networks for application development. In opening up the network, new business models emerge where applications can be developed and provided by enterprises outside the traditional network operator domain. This new business model, combined with the fact that applications can be built using standardized API with off-the-shelf IT technology and tools, will result in new innovative applications that will hit the market with drastically reduced development cycles. This article provides an overview of the Parlay/OSA initiatives concerning the specification of a set of open standardized API. Furthermore, the article outlines some architectural aspects that are implied and implicitly contained in the Parlay/OSA specifications. These are critical for understanding the implications when allowing access to the network via Parlay/OSA. The article then outlines different types of applications that can be built using Parlay/OSA. Finally, we conclude with an analysis of the future of Parlay/OSA.     

GPRS手机协议栈中LLC协议的研究

现介绍了GPRS和GPRS手机协议栈中逻辑链路控制LLC层协议的基本概况，并对LLC层协议中的功能、LLC帧的结构、操作规程、参数协商、流量控制进行了阐述。     

Internetwork protocol

Application of Open Systems Interconnection (OSI) protocols requires cost-effective interconnection of a wide variety of existing and future networks. Differences in underlying technologies, in administrative control, in available qualifies of service, and in other important factors complicate the task of achieving interconnection This paper discusses a variety of the technical issues related to interconnection within the OSI Network Layer.     

一种改进的Woo-Lam协议

串空间是一种基于定理证明的,新兴的安全协议形式化分析模型.认证协议使用密码技术实现网络环境下的身份认证和信息保密.本文针对Woo-Lam协议的不足,对其进行改进,提出了一种新的Woo-Lam协议,使其成为双向认证协议,并基于串空间模型,然后对改进的Woo-Lam协议进行分析证明,结果证明了改进的Woo-Lam协议为双向认证协议.最后通过比较了改进前后的Woo-Lam协议,得出了结论.     

Flow synchronization protocol

Presents an adaptive flow synchronization protocol that permits synchronized delivery of data to and from geographically distributed sites. Applications include inter-stream synchronization, synchronized delivery of information in a multisite conference, and synchronization for concurrency control in distributed computations. The contributions of this protocol in the area of flow synchronization are the ability to synchronize over arbitrary topologies, the introduction of an adaptive synchronization delay, the flexibility to maintain multiple synchronization groups, and the use of a modular architecture that permits the application to tailor synchronization calculations to its service requirements. The authors take advantage of network protocols capable of maintaining network clock synchronization in the millisecond range     

ISDN protocol architecture

IT IS NOW widely recognized that the Integrated Services Digital Network (ISDN) concept is providing a useful framework for the development of future telecommunications networks and services. An ISDN can be regarded as a generalpurpose digital network capable of supporting (or integrating) a wide range of services (voice and non-voice) using a small set of standard multipurpose user-network interfaces. Relevant CCITT Recommendations (or standards) on ISDN interfaces were expected to be available in 1984, the final year of the current study period. This paper reviews CCITT progress on the architectural studies of protocols associated with ISDN's, in particular, the ISDN Protocol Reference Model being developed by Study Group XVIII. This Model, based on the concepts and principles of the ISO/CCITT Open Systems Interconnection (OSI) Reference Model, aims at providing a unified framework for modeling ISDN multiservice communications and capabilities. The new Model is applied to several possible ISDN communications configurations, enabling a number of technical issues to be easily identified.     

Using PARLAY APIs Over a SIP System in a Distributed Service Platform for Carrier Grade Multimedia Services

The implementation of new mobile communication technologies developed in the third generation partnership project (3GPP) will allow to access the Internet not only from a PC but also via mobile phones, palmtops and other devices. New applications will emerge, combining several basic services like voice telephony, e-mail, voice over IP, mobility or web-browsing, and thus wiping out the borders between the fixed telephone network, mobile radio and the Internet. Offering those value-added services will be the key factor for success of network and service providers in an increasingly competitive market. In 3GPP's service framework the use of the Parlay APIs is proposed that allow application development by third parties in order to speed up service creation and deployment. 3GPP has also adopted SIP for session control of multimedia communications in an IP network. This article proposes a mapping of SIP functionality to Parlay services and describes a prototype implementation using the SIP Servlet API. Furthermore, an architecture of a Service Platform is presented that offers a framework for the creation, execution and management of carrier grade multimedia services in heterogeneous networks.     

Dynamic RSVP protocol

RSVP is a resource reservation setup protocol that can be used by a host to request specific QoS for multicast multimedia flows on the Internet. Multiprotocol label switching (MPLS) architecture also needs RSVP. The fact that the resolutions of the display system used in different receiver nodes might have different, multi-resolution characteristics is supported in the MPEG-4 standard, and the EZW compression algorithm can cease decoding at any point in the bitstream. However, RSVP does not provide a more flexible mechanism. In this article we propose an extension of RSVP to provide the needed mechanism, coined dynamic RSVP (DRSVP), to dynamically adjust reserved resources on nodes without much effort. It provides different video resolutions to different receiver nodes with different needed reserved resources. Therefore, it does not waste precious Internet resources to transmit unnecessary multimedia packets.