Developing applications for Internet telephony: a case study on the use of parlay call control APIs in SIP networks

Applications offered to end users as value-added services, or more simple, services, are crucial for the survival and success of service providers. Two main sets of standards have emerged for Internet telephony: H.323 from the ITU-T and SIP from the IETF. Unfortunately, the related application development frameworks are rather weak. Parlay, a set of standard object-oriented and signaling protocol-neural APIs, is an alternative. It allows applications to access network functionality, including call control, in a controller manner. Call control makes it possible to establish, modify, and tear down calls. It is the main functionality offered by Internet telephony networks. We have built a call control application in a SIP environment, using the call control APIs offered by Parlay. The application is a multiparty game. This article describes the case study. The mapping of the APIs onto SIP is presented, and its implementation is described. Related work reviewed, and the lessons learned are discussed. Parlay call control APIs are suitable for application development in Internet telephony. However, well isolated extensions are needed to realize their full potential.     

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.     

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。     

The Distributed Gateway Interworking with SIP/H.323

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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.     

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.     

Integrated service creation environment for open network services

The service creation scheme is changing with the advent of open network service architecture for next-generation network. New requirements should be considered for IT domain developers to create telecom and Internet combined services more easily. This paper describes an integrated service creation environment (SCE) to reflect the trend of network evolution toward an open network environment. The SCE provides multiple service programming tools to support various users’ background, a mash-up toolkit for IT domain, a simulation-based validation tool, a run-time adaptation tool, and a personalized service provisioning environment. Several example services were implemented to verify the features of an integrated SCE. Our approach is very promising because it supports various requirements and background of service developers on the full service creation process. Furthermore, it provides a means for personalized service creation driven by end-user, which is a new trend of future network.     

QoS enabled voice support in the next generation Internet: issues,existing approaches and challenges

The Internet is under rapid growth and continuous evolution in order to accommodate an increasingly large number of applications with diverse service requirements. In particular, Internet telephony, or voice over IP is one of the most promising services currently being deployed. Besides the potentially significant cost reduction, Internet telephony can offer many new features and easier integration with widely adopted Web-based services. Despite these advantages, there still exist a number of barriers to the widespread deployment of Internet telephony. The most prominent one, however, is how to ensure the QoS needed for voice conversation. The purpose of this article is to survey the state-of-the-art technologies in enabling the QoS support for voice communications in the next-generation Internet. In this article, we first review the existing technologies in supporting voice over IP networks, including the basic mechanisms in the IETF Internet telephony architecture and ITU-T H.323-related Recommendations. We then discuss the IETF QoS framework, specifically the Intserv and Diffserv framework. Finally, we present two leading companies' (Cisco and Lucent) solutions to offering IP telephony services as examples to illustrate how real systems are implemented     

IN services for converged (Internet) telephony

Given the convergence of the PSTN and IP-based networks, it would be advantageous to transparently support access to the existing installed base of intelligent network services from packet endpoints, while simultaneously providing newer, more advanced services to said endpoints from within the IN infrastructure. In this article we describe the INSeCT (IN Services for Converged [Internet] Telephony) prototype, aimed at achieving these very goals in networks using H.323. It presents background material on VoIP and IN, then focuses on the prototype implementation     

Policy-based distributed management architecture for large-scale enterprise conferencing service using SIP

Recent advances in broadband communication and computing technology have accelerated the proliferation of Internet protocol-based multimedia conferencing services in large-scale enterprises. Most of the research on session initial protocol (SIP)-based multimedia conferencing, however, has been limited in scalability due to the centralized management of conference control by a single server. In order to overcome this limitation, we have designed policy-based distributed management architecture for a large-scale enterprise conferencing service by extending the Internet Engineering Task Force's (IETF's) approach. The salient feature of the proposed management architecture is that in addition to the distribution of media processing, both participant membership control and authorization functions are dynamically distributed in accordance with the management policy in order to improve scalability. In order to implement these distributed management functions, we have extended both SIP and conference policy control protocols of the IETF. We also show the procedures for the distributed conference management using the extended SIP signaling methods. Finally, we have evaluated by simulation the performances of the proposed architecture in comparison with the centralized architecture of the IETF. The simulation results show that the proposed architecture greatly improves scalability.     

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。     

SIP协议及其应用

由IETF的MMUSIC工作组提出的会话初始协议(SIP)采用与H.323不同的设计理念来实现Internet多媒体通信的信令功能,它借鉴了其它Internet的标准和协议的设计思想,坚持简练、开放、兼容和可扩展等原则,在复杂性、可扩展性以及呼叫建立过程等方面都优于H.323。SIP主要采用三种呼叫方式建立连接:直接呼叫、重定向呼叫和代理呼叫。     

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.     

Evolutionary trends in intelligent networks

A number of groups are currently developing technologies aimed at evolving and enhancing the capabilities of intelligent networks. In this article we discuss three of these initiatives: PINT, Parlay, and IN/CORBA interworking. The IETF PINT work addresses how Internet applications can request and enrich telecommunications services. The Parlay consortium is specifying an object-oriented service control API that facilitates the access, control, and configuration of IN services by enterprise IT systems. The OMG's IN/CORBA interworking specification enables CORBA-based systems to interwork with existing IN infrastructure, thereby promoting the adoption of CORBA for the realization of IN functional entities. We address how all three of these technologies could be deployed together in order to provide a basis for a more flexible and open IN architecture. We also identify a number of common trends and potential pitfalls highlighted by current work on the evolution of IN     

SIP与H.323的比较与互通

文中从实际出发,参考了RFC3261和H.323文献,详细阐述了SIP和H.323的区别,提出了两者在网络中互通的基本概念。在此基础上,通过网际协作功能(IWF)提出了SIP和H.323的互通模型。最后,在软交换网络体系中,提出了SIP和H.323互通的网络构架,以满足未来网络和业务发展的需要。     

NGN(下一代网络)电信业务开发模型研究

提出了应用多视图的方法研究基于Web services的下一代网络(NGN：next generation network)电信业务的开发，并对下一代网络电信业务开发中的业务模式模型、开发模型和过渡模型进行了详细的介绍。     

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.     

基于Parlay和OGSA的NGN业务生成框架模型

提出了一种基于开放网络业务能力标准Parlay和开放网格服务体系结构(OGSA)的NGN(下一代网络)业务生成框架模型,不仅实现了对异构网络业务能力的开放,而且在此基础上建立了有效的分布式集成机制。为了实现该模型,进一步提出了Parlay和OGSA间的一种面向服务(service-oriented)的通信连接方法。将网格技术应用在电信业务生成领域是一种全新且有益的尝试。通过仿真实验验证了该框架模型在跨越网络快速生成业务上的有效性,同时,该框架模型在部署上不需要现有终端设备更新或升级就能支持。     

SIP下一代网的关键技术

SIP,会话发起协议。未来的信息网是一个基于全IP的网络平台,在这个平台上运营商能够为用户提供丰富的综合性新业务,需要一个公共的协议来进行多设备供应商之间、多协议之间的翻译和互通,SIP框架思想是实现下一代网络解决方案的正确手段。本文对SW进行了概括性的介绍并说明如何建立SIP会话,并与ITU-T提出的H.323比较,说明SW的优越性。