基于JBoss实现Web服务与通信业务整合

作为面向服务体系( SOA)的最主要的实现方式,Web服务(Web Services)技术被认为是业务整合的主要基础之一.根据ParlayX的Web服务规范,在NGN/IMS网络环境中,提出了一种基于JBoss服务器的以Web服务的方式交付通信业务的下一代网络融合业务平台架构,详细分析了Web服务与通信业务之间的交互方式,并利用该融合业务平台架构实现了一个第三方呼叫控制示例业务,展示了如何整合Web服务和通信业务,验证了该融合业务平台架构的可行性.     

基于云计算及SOA架构的媒体应用聚合云平台＊

本文以云计算技术为基础、参考SOA技术架构,提出了一个适用于有线电视运营商的媒体应用聚合云平台架构。该平台聚合传统的电视业务及新媒体业务,支持域内业务运营及网络域间的业务合作运营;平台的底层基于云计算的虚拟技术进行资源整合,实现对资源的统一管理与调度,SaaS层和PaaS层基于SOA架构进行服务及应用设计。该云平台的架构对有线电视运营商云平台的搭建具有一定的参考价值。     

基于IMS的多媒体彩铃业务解决方案

一、IMS系统特点 IMS(IP多媒体子系统)最初是3GPP组织制定的3G网络核心技术标准,目前已被ITU-T和ETSI认可,纳入NGN的核心标准框架,并被认为是实现未来FMC(固定/移动网络融合)的重要技术基础.IMS是-个全IP的网络架构,采用SIP协议进行控制,实现移动性管理、多媒体会话信令和载体业务传输,实现端到端的IP业务.此外,IMS还具有普通业务采用归属服务控制,接入独立,业务、控制、承载层完全分离等特点.     

SIP协议在视频监控系统中的应用

首先介绍了视频监控业务的发展与现状，对采用SIP协议作为控制协议的意义和SIP多媒体体系架构进行介绍，然后介绍基于SIP协议视频监控系统，对SIP协议扩展支持监控业务提出探讨的想法，最后对SIP实现视频监控业务的优势进行分析。     

面向服务架构在SCADA中应用的研究与探讨

为了实现电力SCADA(数据采集与监控)系统与电力调度其他相关应用系统的集成,提出了基于SOA(面向服务架构)实现跨平台的集成的应用框架和体系架构。该应用框架以SOA思想为指导,借助Web Service和J2EE平台技术的特性优势来实现。将电力SCADA系统分为用户层、Web层、业务逻辑层和企业信息系统层,对每一层做了具体的设计说明。该基于SOA的集成方法较之其他的集成方法,展现了更广泛的软硬件兼容性、灵活的功能部署和简易的系统扩性。     

基于SOA的教学管理系统web服务的设计与实现

面向服务的SOA架构能通过将教学管理系统需要共享的信息封装成服务发布和调用,可以解决原有教学管理系统的信息孤岛问题。本文研究了用SOA架构对教学管理系统进行web服务封装的设计和实现,有效地实现了基于SOA架构的教学管理系统应用层次的设计与实现。     

基于SOA的NGOSS构架的研究

本文通过对移动运营商下一代业务运营支撑系统（NGOSS）的建模分析,阐述了面向服务架构（SOA）框架的设计理念,提出基于SOA的NGOSS的架构,并通过设计举例说明如何实现该架构。     

基于IMS的多媒体彩铃业务解决方案

IMS（IP多媒体子系统）最初是3GPP组织制定的3G网络核心技术标准，目前已被ITU-T和ETSI认可，纳入NGN的核心标准框架，并被认为是实现未来FMC（固定／移动网络融合）的重要技术基础。IMS是一个全IP的网络架构，采用SIP协议进行控制，实现移动性管理、多媒体会话信令和载体业务传输，实现端到端的IP业务。此外，IMS还具有普通业务采用归属服务控制，接入独立，业务、控制、承载层完全分离等特点。[第一段]     

基于IMS的网络融合技术

由于IP多媒体子系统(IMS)具有采用SIP协议进行呼叫控制、与接入无关和能够灵活提供多种业务的特点，ETSI和ITU-T已确定将其作为NGN的核心结构。在介绍IMS体系结构的基础上，分析了基于IMS实现固定和移动网络融合的可行性。并提出了融合网络的参考架构。     

基于Android的SIP远程视频监控系统的设计

本文利用SIP协议建立组呼来控制监控摄像头,从而建立多路远程视频监控系统。论文介绍了基于Android系统的SIP协议栈JAIN-SIP和远程视频监控系统架构,阐述了利用SIP扩展消息完成组呼控制,从而实现监控中心对远端监控摄像头的呼叫控制。     

An Approach to Integrating SIP in Converged Multimodal/Multimedia Communication Services

In this paper, we present an approach of integrating SIP (Session Initiation Protocol) in converged multimodal/multimedia communication services. An extensible VoIPTeleserver for VoIP in SIP environment is described. It is based on the concept of dialogue system and Web convergence that separates the channel dependent media resources from the application dependent service creation and hosting environment. It supports XML based service applications for multiple channels including voice, DTMF, IM and chat over IP. The loosely coupled open architecture in our approach is highly extensible. We describe the concept and structure of VoIPTeleServer used in our approach in detail, which interfaces to the VoIP world through SIP signaling and works as a broker between the VoIP SIP environment and MTIP to deliver converged communication services. A prototype of VoIPTeleServer was implemented, and services and applications based on SIP and MTIP convergence are constructed. Special attention is given to the adverse effect of delay, jitter and packet loss for voice portal services over IP. In particular, case studies of DTMF service in voice portal under adverse channel conditions are performed. The compounding effects of multiple channel impairments to DTMF in voice portal services over IP are characterized. The potential high error rate of the DTMF service indicates that the data redundancy method as proposed in RFC 2198 is needed for DTMF in order to achieve reliable voice portal services over IP.     

基于OMAP5912和SIP的IP电话终端

以车载汽车电子多媒体视听系统为研发背景，在OMAP5912平台上采用SIP（会话发起协议）设计了IP电话终端。详细说明了IP电话终端的主要硬件组成部分，介绍了SIP和oSIP协议栈，给出了嵌入式系统IP电话软件的程序设计框架和效果，并对在该基础上实现无线IP电话终端提出了进一步的设想。     

SIP-based VoIP network and its interworking with the PSTN

The Session Initiation Protocol (SIP) is considered to be a future powerful alternative to the H.323 standard as the signalling system for the dominant Voice over IP (VoIP) communications. This paper provides an in-depth analysis of SIP by describing the SIP protocol stack, summarising the main features of the protocol, and illustrating its architecture, message and operation. The paper also explains the architecture and the two key aspects of signalling interworking when SIP is interconnected with the PSTN.     

IP多媒体子系统的安全问题剖析

IP多媒体子系统(IMS)是一种非常重要的全新体系框架,同时也是全IP网络发展的终极目标。这种框架为移动应用融合语音、视频和数据服务。IMS采用SIP作为主要信令机制,并覆盖于IP网络之上。鉴于其所处的全IP环境以及以SIP为基础的设计架构,IMS在核心网层和应用层面都呈现出了多种安全挑战。本文对IMS的体系框架和由此产生的安全挑战做了深入探讨和研究。     

Seamless SIP-based mobility for multimedia applications

Application-level protocol abstraction is required to support seamless mobility in next-generation heterogeneous wireless networks. Session initiation protocol (SIP) provides the required abstraction for mobility support for multimedia applications in such networks. However, the handoff procedure with SIP suffers from undesirable delay and hence packet loss in some cases, which is detrimental to applications like voice over IP (VoIP) or streaming video that demand stringent quality of service (QoS) requirements. In this article we present a SIP-based architecture that supports soft handoff for IP-centric wireless networks. Soft handoff ensures that there is no packet loss and that the end-to-end delay jitter is kept under control.     

An IPv4-IPv6 translation mechanism for SIP overlay network in UMTS all-IP environment

Both IPv6 and session initiation protocol (SIP) are default protocols for Universal Mobile Telecommunications System (UMTS) all-Internet protocol (IP) network. In the existing mobile telecommunications environments, an IPv6-based UMTS all-IP network needs to interwork with other Internet protocol version 4 (IPv4)-based SIP networks. Therefore, mobile SIP applications are typically offered through an overlay structure over the IPv4-Internet protocol version 6 (IPv6) interworking environments. Based on 3GPP 23.228, we propose an IPv4-IPv6 translation mechanism (i.e., SIPv6 translator) that integrates different IP infrastructures (i.e., IPv4 and IPv6) to provide an overlay network for transparent SIP application deployment. In this paper, we present the architecture and the call flows of the SIPv6 translator. An analytic model is proposed to investigate the fault tolerance issue of our approach. Our study provides guidelines to select appropriate number of processors for fault tolerance.     

P2PSIP原理和应用

SIP协议是当前VoIP和IM等多媒体协议的呼叫建立协议。P2P技术提供了分布式的网络架构中通信节点之间得对等通信能力,P2P网络架构经历了集中目录式网络架构、纯P2P网络架构、混合式P2P网络架构和结构化P2P网络架构。通过引入P2P技术,SIP网络的能力可以得到增强。P2PSIP网络主要有两个重要的操作:对P2P节点的操作和用户层面的操作。     

STEM: Secure Telephony Enabled Middlebox

Dynamic applications, including IP telephony, have not seen wide acceptance within enterprises because of problems caused by the existing network infrastructure. Static elements, including firewalls and network address translation devices, are not capable of allowing dynamic applications to operate properly. The Secure Telephony Enabled Middlebox (STEM) architecture is an enhancement of the existing network design to remove the issues surrounding static devices. The architecture incorporates an improved firewall that can interpret and utilize information in the application layer of packets to ensure proper functionality. In addition to allowing dynamic applications to function normally, the STEM architecture also incorporates several detection and response mechanisms for well-known network-based vulnerabilities. This article describes the key components of the architecture with respect to the SIP protocol.     

基于SOA的软件体系架构测评研究

现在,众多运营商或IT厂商都越来越关注面向服务的体系结构的应用。本文以研究SOA测评为目标,从架构及技术层面明确和阐述SOA的结构,将评测指标对应并落实到具体的系统功能及性能等质量特性,使SOA从结构级别向下延伸为可测量的一套评估模型,以协助企业了解并实施应用SOA。     

会话的初始化协议

会话初始化协议(SIP),它的主要目的是为了解决IP网中的信令控制,以及同软交换机(SoftSwitch)的通信,从而构成下一代的增值业务平台,对电信、银行、金融等行业提供更好的增值业务.本文简单介绍了SIP的系统结构、主要功能、消息机制、优越性和发展现状,最后通过介绍了它的应用实例,模拟了SIP的呼叫建立及终止过程.