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.     

The Session Initiation Protocol: Internet-centric signaling

The Session Initiation Protocol (SIP) provides advanced signaling and control functionality for a wide variety of multimedia services. SIP can efficiently and scalably locate resources based on a location-independent name and then negotiate session characteristics. It can find use in applications ranging from Internet telephony and conferencing to instant messaging, event notification, and the control of networked devices. We summarize the main protocol features and describe a range of extensions currently being discussed within the Internet Engineering Task Force     

Unified Mobility Manager: enabling efficient SIP/UMTS mobile network control

Internet telephony is viewed as an emerging technology not only for wireline networks, but also for third-generation wireless networks. Although IP end to end is considered the ultimate approach to future wireless voice services, there is still a long way to go before IP voice packets can be effectively transported over the air. Therefore, Internet telephony and today's circuit-switched wireless network will coexist for years to come, and it is essential to effectively perform interworking between these networks. This article proposes the Unified Mobility Manager (UMM) that achieves efficient interworking between traditional wireless networks and Internet telephony networks. The main characteristic of the UMM is that it combines UMTS HLR and SIP proxy functionality in one logical entity, which helps eliminate the performance degradation due to interworking between SIP and UMTS. This article identifies seven potential network architectures with and without the UMM and with varying degrees of IP penetration in the wireless core networks, and performs comparative analysis in terms of their call setup signaling latency. Our performance results show that for SIP originated calls, the architecture with the UMM can achieve better performance than existing UMTS networks without the UMM. Our results further show that when the backbone network is fully IP-enabled, dramatic performance gains can be accomplished with the UMM for PSTN originated calls as well as for SIP originated calls. The article also demonstrates that the UMM allows graceful migration from today's circuit-switched wireless networks to hybrid SIP/circuit-switched wireless networks, and toward the IMS architecture for all-IP UMTS networks in the future.     

3G Service Control

The huge investment in 3G mobile licences is an incentive to develop extra services above and beyond basic telephony and data services. Service control provides an infrastructure to deliver services, particularly those of a real-time nature. This paper focuses on the progress in developing service control in the context of the 3GPP Release 4/5 network. To achieve the objectives for service evolution, mass service development and cost-effective services, three technologies are being pursued for service control. These are CAMEL, APIs and SIP service extensions.     

True number portability and advanced call screening in a SIP-basedIP telephony system

Custom local area signaling service features offered in the PSTN have certain limitations due to the closed nature of PSTN network signaling. The adoption of telephony over IP (IP telephony) will enable a new paradigm of services and features that are not possible to implement in today's PSTN. This is especially the case for services that make use of personal, trusted information, which can be provided by a user's personal digital assistant. We demonstrate how personal information can be coupled with an IP telephony service to provide user-customized call handling by the network. In particular, we describe a demonstration architecture that includes Ethernet-attached phones running SIP, with an interface to synchronize with PDAs that supply personal information. The proposed architecture is quite flexible; it can support enhanced versions of the current PSTN and private branch exchange services, in addition to many new features and services. We describe true number portability and advanced call screening as examples of new services in a hybrid PSTN/IP telephony environment     

基于开放业务的网络融合

文章以IP电话为应用背景论述了网络融合的两种典型模型,讨论了采用Parlay/OSA 4.0开放业务规范实现网络融合的可能性,提出了一种能提供跨越因特网与传统电话网之间IP电话业务的网络融合方案:一种基于开放业务的网络融合方案,并讲座了该方案在实现网络融合相关的电信业务方面的优点和不足.     

基于SIP协议的IP电话增值业务实现技术

讨论了SIP协议以及基于SIP协议的IP电话增值业务实现技术 ,并对SIPCGI、CPL、SIPServlets、JAINAPIs等几种SIP编程技术进行了分析与比较 ,归纳总结了开发IP电话增值业务的一般方法     

Parlay X Web Services

The Parlay X specification describes a number of Web Services that will provide a simple interface to telephony and other systems. Other telephony interface initiatives have addressed the needs of complex and niche applications. The Parlay X initiative has set out to address the needs of the wider population of application developers. These application developers should be able to create telephony applications with little or no knowledge of the underlying network. Version 1 of the specification is complete and should be released to the public in the near future. This version covers third-party call control, SMS, multimedia messaging, payment, account management, user status and user location. This paper discusses the motivation for these new initiatives and presents an overview of the specifications and also gives an insight into the implementations available. This revised version was published online in July 2006 with corrections to the Cover Date.     

SIP协议下Parlay业务生成环境的研究及实现

开放式API（如Parlay、JAIN等）将网络资源向第三方开发商开放，电信新业务的开发变得快速、有效、方便灵活，然而使用更高级的抽象业务生成环境（SCE）将会使业务的开发变得更快更有效。给出了SIP（会话初始化协议）协议下，Parlay API业务生成环境的总体结构和映射，并在此结构上应用一个业务实例来分析它的相关过程。尽管它仅支持发起呼叫业务和生成业务的范围仍然有限，但它是非常有应用前景的。     

VoIP信令协议SIP和H.323的比较

VoIP系统凭借其低廉的话费和较好的语音质量,已经成为重要的电信业务,并有取代传统长途业务的趋势.许多组织研究并制定了IP网络上呼叫的协议标准,但有两种IP电话信令和控制标准最具有影响力.一种是ITU推荐的H.323协议,另一种是IETF的SIP.这两种协议代表了解决同一问题的两种不同的方法:H.323是信令基于ISDN Q.931和早期推荐的H系列协议的传统的电路交换的方法,而SIP是一种支持基于HTTP的IP网络的超轻量协议标准.本文,我们主要针对SIP和H.323的体系结构,可靠性,复杂性,可扩展性,可伸缩性以及支持业务类型方面进行比较.     

IP电话：创新者的追求

早在１９９５年就有人预测Ｉｎｔｅｒｎｅｔ在某种意义上可能会对电话技术产生重要影响。短短几年间,ＩＰ电话技术的屡次重大突破为这一预言提供了不可辨驳的佐证,也使人们不再小看这一事物。目前,ＩＰ电话供应商已开发出大量应用：ＩＰ增强型服务、因特网呼叫等待／电话倍频器、基于ＩＰ的ＰＢＸ和基于Ｗｅｂ的呼叫中心等,ＩＰ电话正逐步改变着这个世界的交流方式。     

Parlay‐based service provision in circuit‐ and packet‐switched telecommunications networks

The spectrum of potential value added services over Internet telephony is wide, but the current service provision solutions are inadequate or proprietary. The nature of Internet differs significantly from that of circuit switched network, however, VoIP architectures can capitalize service control architectures in the PSTN world. We describe such an architecture based on the intelligent network and the Parlay, employing distributed objects and mobile agents as enabling technologies. This architecture has been implemented in the PSTN and the Internet and it has provided a framework for service provisioning, augmenting the space of supported services. Copyright © 2004 John Wiley & Sons, Ltd.     

Internet telephony signaling

The term “multimedia session” refers to the integration of data coming from various sources, such as sound, video and text, within a computer application. Telephony over the Internet is among the more exciting current developments. The signaling of a telephone call consists of the set of messages and procedures used to establish a connection, to request changes in communication bandwidth, to obtain the message status for the end points participating in the conversation, and to close the link. At present there exist two competing signaling protocols for Internet telephony, viz., the H.323 protocol sponsored by the ITU and the Session Invitation Protocol (SIP) sponsored by the IETF. Each of them supplies its own signaling mechanisms.

In this paper, these two protocols in terms of their main functionalities are compared. Based on the results of this comparison, a Client/Server architecture for the development of an application that supports a basic SIP implementation, as well as the formulation of requests allowing the establishment and the disconnection of communications between a number of users in a multimedia session are then defined.     

基于Parlay X的下一代电信增值业务模型研究

Parlay X是一套简单易懂、比之Parlay在更高层次抽象电信网络基本功能的应用编程接口(API),它以Web Service的形式供IT应用开发者在开发的应用中调用电信网络功能.下一代网络业务的发展是一个渐进过程,对运营商来说,如何以更小的成本快速开发和部署新业务成了当务之急.文中介绍了Parlay X体系结构模型,着重提出了基于Parlay X的电信增值业务过渡模型,并对短信业务的开发流程进行了分析.     

The Application of WSFL in the Parlay X Based Services Creation

1　Introduction　NGN abstracts beneath protocols as a set of easy under standing interfaces through open Application ProgrammingInterfaces (API). These APIs are independent with net works, so the services created with these APIs are irrespec tive to special network details[1]　ParlayXAPI is a set of more single and higher abstractedAPIs than Parlay API. These ParlayXAPIs have the abilityto access the network functions, and are easy to understandby IT developers to use in …     

The IETF Internet telephony architecture and protocols

Internet telephony was first used as a simple way to provide point-to-point voice transport between two IP hosts. However, the growing interest in providing integrated voice, data, and video services has caused its scope to be extended. Internet telephony now encompasses a range of services, including not only traditional conferencing, call control, multimedia, and mobility services, but also new ones that integrate Web, e-mail, presence, and instant messaging applications with telephony. Internet telephony and traditional circuit-switched telephony will coexist for quite some time, requiring interworking between the two. In this article we present a suite of protocols, developed in the IETF, which provide a partial solution to this complex problem     

Programming Internet telephony services

Internet telephony enables a wealth of new service possibilities. Traditional telephony services such as call forwarding, transfer, and 800 number services, can be enhanced by interaction with e-mail, Web, and directory services. Additional media types, like video and interactive chat, can be added as well. One of the challenges in providing these services is how to effectively program them. Programming these services requires decisions regarding where the code executes, how it interfaces with the protocols that deliver the services, and what level of control the code has. In this article we consider this problem in detail. We develop requirements for programming Internet telephony services, and we show that at least two solutions are required-one geared for service creation by trusted users (such as administrators), and one geared for service creation by untrusted users (such as consumers). We review existing techniques for service programmability in the Internet and in the telephone network, and extract the best components of both. The result is a common gateway interface that allows trusted users to develop services, and the call processing language that allows untrusted users to develop services     

ISDN computer-aided telephony

It is argued that ISDN computer-aided telephony requires properly architected platforms to satisfy changing application needs during the 1990s. Proper architecting necessitates the use of functionally rich and consistent telephony application programming interfaces (APIs). Other APIs are also needed to support integrated applications. The coexistence of telephony and other APIs must be accommodated in the ISDN driver architecture to make efficient use of D-channel signaling and voice, data, or image communications on the associated B/H channels. This driver may support Open Systems Interconnection (OSI), Systems Network Architecture (SNA), X.25, or other protocol stacks in the same computer using a single ISDN access link. Applications being currently explored show that significant benefits can be realized using incoming call management and LAN-based image server access by means of ISDN. It is envisioned that by the year 2001, a common API will facilitate multimedia applications on multivendor platforms architected within the OSI framework. These platforms will support interconnections of public and private ISDNs and bridging to BISDNs     

Web服务及其在Parlay接口中的应用

电信网与Internet走向融合，而Parlay接口与Web服务作为各自领域开放技术的代表，也开始了互相结合。Web服务是一种基于可扩展标记语言（XML）、面向消息的分布式计算技术，与公共对象请求代理体系结构（CORBA）等分布式对象技术相比，在Internet范围内的互操作性更好。Web服务是实现面向服务体系结构（SOA）的最佳候选技术之一。基于Web服务的Parlay接口包括Parlay Web服务和Parlay X。其中，Parlay Web服务模拟面向对象的Parlay应用编程接口（API）定义，Parlay X的设计遵循Web服务面向消息的技术发展思路。基于Web服务的Parlay接口技术为构建电信网和Internet融合环境下的统一业务体系提供了基础。