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.     

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     

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.     

SIP重定向攻击实现及防范

SIP是最流行的多媒体会话控制协议,它是一个应用层的控制协议,可以用来建立、修改和终止多媒体会话（或者会议）,SIP采用文本形式表示消息的词法和语法,而对文本形式的分析比较简单,因此SIP消息容易被攻击者模仿、篡改,从而加以非法利用。论文研究并实现了一种较新的攻击方法——重定向攻击,最后提出防范此种攻击的安全饥制。     

基于SIP的Internet电话

文章先给出了基于ＳＩＰ（会话初始化协议）的Ｉｎｔｅｒｎｅｔ电话的协议结构,随后对其中传输数据的ＲＴＰ（实时传输协议）和ＲＴＣＰ（实时传输控制协议）进行了介绍和分析,并对会话初始化协议的组成实体、命名、寻址、操作进行了详细的论述。     

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     

SIP在移动Internet中的应用

移动Internet是目前学术界研究的热点问题之一,SIP是IETF提出的IP电话信令协议。结合移动Internet技术介绍了一些扩展SIP以提供移动性支持的机制,主要讨论注册、定位和切换等方面的问题。     

Voice and Internet Multimedia in UMTS Networks

Voice telephony is the predominant service on today's cellular mobile networks, in terms of number of customers, revenues and network usage. However, it is difficult to predict how long this will be the case given the rising demand for new Internet multimedia services. It is therefore essential that 3rd generation (3G) mobile networks support a voice telephony service, but also that these networks are also capable of providing Internet multimedia services using the same technology.This paper provides an overview of how voice telephony is provided in the initial phase of the universal mobile telecommunications system (UMTS). It then describes how this is expected to evolve in later phases — so that voice telephony becomes one of a large number of multimedia services provided from a common Internet protocol-based mobile network.     

基于SIP协议的IP电话语音系统设计与实现

本文基于SIP协议的IP电话作为主要研究内容,探讨了IP电话的相关议、相关标准和关键技术,对SIP电话协议进行了研究分析,设计提出一套结构合理的VoIP系统,并对系统组成、系统流程等作了详细的规划,安装配置OpenSER服务器,并能够使用MySql数据库系统来存储用户信息,基于分层设计的思想设计了客户端软件,最后利用该软件进行了系统测试,结果表明系统性能优良。     

新兴电信运营商的战略选择——统一的IP核心网

在无连接的IP网上,基于SIP的VoIP系统用软件交换机提供VoIP业务,既可完成固定电话的交换传输,也可完成移动电话的交换传输,差别仅仅是接入方式不同。它不但可以提供传统电话智能网的全部功能,还可以提供传统电话智能网和互联网特征融合的新功能。对于新兴运营商而言,采用这样的系统不仅可以以价格优势、更重要的是可以以功能的优势与传统电信运营商竞争。     

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.     

Handoff with DSP Support: Enabling Seamless Voice Communications across Heterogeneous Telephony Systems on Dual-Mode Mobile Devices

In this paper we investigate the problem of voice communications across heterogeneous telephony systems on dual-mode (WiFi and GSM) mobile devices. Since GSM is a circuit-switched telephony system, existing solutions that are based on packet-switched network protocols cannot be used. We show in this paper that an enabling technology for seamless voice communications across circuit-switched and packet-switched telephony systems is the support of digital signal processing (DSP) techniques during handoffs. To substantiate our argument, we start with a framework based on the Session Initiation Protocol (SIP) for vertical handoffs on dual-mode mobile devices. We then identify the key obstacle in achieving seamless handoffs across circuit-switched and packet-switched systems, and explain why DSP support is necessary in this context. We propose a solution that incorporates time alignment and time scaling algorithms during handoffs for supporting seamless voice communications across heterogeneous telephony systems. We conduct testbed experiments using a GSM-WiFi dual-mode notebook and evaluate the quality of speech when the call is migrated from WiFi to GSM networks. Evaluation results show that such a cross-disciplinary solution involving signal processing and networking can effectively support seamless voice communications across heterogeneous telephony systems.     

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 security issues: the SIP authentication procedure and its processing load

Session Initiation Protocol (SIP) is currently receiving much attention and seems to be the most promising candidate as a signaling protocol for the current and future IP telephony services, also becoming a real competitor to the plain old telephone service. For the realization of such a scenario, there is an obvious need to provide a certain level of quality and security, comparable to that provided by the traditional telephone systems. While the problem of QoS mostly refers to the network layer, the problem of security is strictly related to the signaling mechanisms and the service provisioning model. For this reason, at present, a very hot topic in the SIP and IP telephony standardization track is security support. In this work, the security model used by SIP is described, and the different open issues are highlighted. We focus, in particular, on the problem of authentication providing a short tutorial on the solution under standardization. The architecture of a possible commercial IP telephony service including user authentication is also described. Finally, we focus on performance issues. By means of a real testbed implementation, we provide an experimental performance analysis of the SIP security mechanisms, based on our open source Java implementation of a SIP proxy server. The performance of the server has been compared with and without security support, under various scenarios.     

Internet telephony: going like crazy

Interet telephony is possibly the fastest-growing part of communications today. This article discusses what exactly it is, who needs it, and how it works. Internet telephony, or voice over Internet protocol (VoIP), is the provision of phone service over the Internet. But in sharp contrast with conventional telephony, it carries voice traffic as data packets over a packet-switched data network instead of as a synchronous stream of binary data over a circuit-switched, time-division multiplexed (TDM) voice network. There are some substantial benefits (as well as some sticky problems) to the scheme, which is why companies and individuals are finding it increasingly attractive     

Internet telephony: services, technical challenges, and products

The rapid proliferation of the Internet has given rise to a strong interest in carrying telephony over the Internet. Because the Internet supports data communications, a range of other services can be bundled together with Internet telephony. The Internet, however, was designed for non-real-time data communications, and hence it poses several technical challenges that must be overcome before the Internet can be successfully used for carrying telephone services. This article discusses new services we can expect from Internet telephony, the technical challenges and solutions, and the emerging products that promise to support Internet telephony     

Voice over Internet protocol (VoIP)

During the Internet stock bubble, articles in the trade press frequently said that, in the near future, telephone traffic would be just another application running over the Internet. Such statements gloss over many engineering details that preclude voice from being just another Internet application. This paper deals with the technical aspects of implementing voice over Internet protocol (VoIP), without speculating on the timetable for convergence. First, the paper discusses the factors involved in making a high-quality VoIP call and the engineering tradeoffs that must be made between delay and the efficient use of bandwidth. After a discussion of codec selection and the delay budget, there is a discussion of various techniques to achieve network quality of service. Since call setup is very important, the paper next gives an overview of several VoIP call signaling protocols, including H.323, SIP, MGCP, and Megaco/H.248. There is a section on telephony routing over IP (TRIP). Finally, the paper explains some VoIP issues with network address translation and firewalls     

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.     

Approach to stress tests in SIP environment based on marginal analysis

In the field of performance metrics and measurements of SIP (Session Initiation Protocol) Proxy and B2BUA (Back-to-Back User Agent) no standardized methodology has been presented yet. This gap results in a problematic determination of a hardware, the performance of which would be cost-effective and sufficient for the running the SIP Server in a given environment. Today practice relies on the administrator’s skills and experience with the needs of the telephony infrastructure. From this and the increasing usage of SIP based VoIP technologies come the main reasons for creating a methodology that would allow administrators to precisely measure the SIP Server performance and compare it to other software and hardware platform. This work also utilizes SIP Server performance measurements to comparison the results taken when transcoding was in use and when it was not and provides the means for comparison of B2BUAs platform independently.     

Integrating communication services

The need for communication services which span multiple communication technologies is growing. Communication services are being developed in three areas: in the public switched telephony networks, on the Internet in the form of integrated multimedia including voice-over-Internet, and in private switched telephony networks in the form of enterprise computer-telephony integration applications. This article shows it is plausible to create unified services which span the Internet and public switched telephony networks, and goes on to describe Nexus, an architecture and prototype for integrated communication services