VoIP mobility in IP/cellular network internetworking

This article explores VoIP mobility in the context of IP and cellular networks interworking. ITU-T Rec. H.323 gateways provide the interconnection between IP networks and switched circuit networks. They allow a call originating from an SCN phone to be transmitted over an IP network to an H.323 terminal, or bridged to another SCN phone. While H.323 provides interoperability with other SCN terminals, the major efforts have been focused on IP/wired SCN (PSTN, ISDN, etc.) interworking. In this article we discuss the challenges associated with the interworking between IP networks and cellular networks through H.323 gateways, and propose an innovative approach using the existing call transfer supplementary service to provide VoIP mobility in the H.323 IP telephony networks. The proposed approach uses existing components in the H.323 standard, thereby allowing VoIP mobility service in hybrid IP/cellular networks to be a value-added feature in the existing H.323-compliant Internet telephony systems     

Mobile Internet telephony: mobility extension to H.323

This paper studies mobility extensions to ITU-T Rec. H.323 for the support of mobile Internet telephony. Internet telephony, also known as voice-over Internet protocol (IP) (VoIP), requires the transmission of two-way and real-time traffic over IP-based networks. The current version of H.323 allows IP telephony and the interoperability of the Internet with switched circuit networks (SCN). However, VoIP mobility has not been previously widely considered, where VoIP mobility refers to the mobility within the scope of IP telephony. We focus on terminal mobility for VoIP. We investigate the influence of mobility on the H.323 layer and propose an H.323 mobility solution to be implemented over the IP layer. Two approaches to mobility extensions to H.323 are described: using ad hoc multipoint conference expansion and using IP multicasting to emulate mobility. Besides, we have also shown that the proposed ad hoc expansion approach shares many properties with the alternative of using IP multicasting for mobility. Hence, the call signaling procedure for the ad hoc expansion approach is also applicable to the multicasting approach. Since ad hoc multipoint expansion has been defined in H.323, our solution introduces no additional entities to H.323 and requires minimal modifications to the existing H.323 protocol. Such mobility extensions can serve as a value-added feature for the Internet telephony systems compliant to the H.323 standard     

软交换中的用户网关设计

介绍了一种基于DSP的应用于软交换的用户网关设计方案。该用户网关支持H.323协议标准,完成包括音频处理、语音编解码、回声抵消、信令音的产生与检测等功能。网关DSP采用TI公司的TMS320 VC5409作为语音编解码器,实现了ITU-G.723.1的全部功能和回声抵消功能,同时采用Silicon Laboratories公司的SI3220实现了各种呼叫进程音的产生和检测。普通双音频电话机可通过用户网关接入Internet,实现IP电话呼叫。对本设计方案的各个系统硬件模块进行了详细描述。     

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     

软交换系统七号信令设计

基于公共交换电话网络大规模应用七号信令的实际情况,对软交换体系结构、信令网关、初始会话协议(SIP)进行了描述。并针对下一代网络的分层思想、软交换的互联互通结构和呼叫流程,提出了利用软交换交换核心平台和信令网关及SIP相互配合实现七号信令的解决方案。描述了软交换系统的七号信令软件结构、信令网关的软件结构,阐述了相应呼叫流程的实现。     

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     

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.     

Talk about a change [VoIP replacing PSTN]

The rapid rise of Internet telephony services means that existing telecoms networks could soon be redundant. This paper examines the technological and financial issues concerning voice over Internet protocol (VoIP) and the probability of it substantially replacing existing public switched telephone networks (PSTN) within the next few years.     

A programmable architecture for the provision of hybrid services

The success of new service provision platforms will largely depend on their ability to blend with existing technologies. The advent of Internet telephony, although impressive, is unlikely to make telephone customers suddenly turn in favor of computers. Rather, customers display increasing interest in services that span multiple networks (especially Internet protocol-based networks and the telephone and cellular networks) and open new vistas. We refer to these services as hybrid services and propose an architecture for their provision. This architecture allows for programming the service platform elements (i.e., network nodes, gateways, control servers, and terminals) in order to include new service logics. We identify components that can be assembled to build these logics by considering a service as a composition of features such as address translation, security, call control, connectivity, charging, and user interaction. Generic service components are derived from the modeling of these features. We assure that our proposal can be implemented even in existing systems in return for slight changes. These systems are required to generate an event when a special service is encountered. The treatment of this event is handled by an object at a Java service layer. Java has been chosen for its platform-neutrality property and its embedded security mechanisms. Using our architecture, we design a hybrid closed user group service     

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.     

Real-time multimedia over ATM: RMOA

RMOA is a new ATM Forum standard addressing the transport of H.323 VoIP traffic over ATM-based Internet backbones. It defines a new H.323 gateway devised to carry H.323 real-time media streams by taking advantage of the quality of service features of ATM. The approach is extremely efficient in that it reduces the protocol overhead on the ATM transport     

Internet电话技术及其与PSTN的接口

本文首先介绍了Ｉｎｔｅｒｎｅｔ电话技术以及为什么要与ＰＳＴＮ接口的原因，接着对现有的几种方案进行了比较和说明，然后讨论实现Ｉｎｔｅｒｎｅｔ电话实时传送所面临的问题，介绍了现有关于Ｉｎｔｅｒｅｔ电话技术在国际标准和相关协议，最后提出了一种实现Ｉｎｔｅｒｎｅｔ电话技术（ＰＳＴＮ接入网关）的硬件和软件结构。     

VTOA/VolP/N-ISDN Telephony Gateway

This article presents the architecture and implementation of a telephony gateway for interworking between N- ISDN, ATM and IP telephony. In this way, interworking is achieved both within private networks and with the PSTN, address translation being performed according to both the vtoa (atm interface) and H .323 (ip interface) specifications. The gateway implementation is based on a PC, presenting a cost- effective alternative to the equipment currently available on the market. Moreover, its highly modular software architecture allows new telephony interfaces to be easily added.     

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     

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.     

TOPS: an architecture for telephony over packet networks

Packet telephony is of increasing interest in both the telecommunications and Internet communities. The emergence of packet telephony will create new services, and presents an opportunity to rethink how conventional telephony services are implemented. In this paper, we present an architecture for telephony over packet networks (TOPS). TOPS allows users to move between terminals or to use mobile terminals while being reachable by the same name. TOPS users can have multiple terminals and control how calls are routed to them. TOPS allows for terminals with a range of capabilities such as support for video, whiteboard, and other media with a variety of coding formats. TOPS retains the necessary information on terminal capabilities to determine the appropriate type of communication to be established with the remote terminal. The architecture assumes that the underlying network supports the establishment of end-to-end connectivity between terminals, with an appropriate quality of service. The components of TOPS are a directory service, an application layer signaling protocol, and a logical channel abstraction for communication between end-systems. The directory service maps a user's name to a set of terminals where the user may be reached. A user can control the translation operation by specifying profiles that customize how his name is mapped to a set of terminals where he can be reached. Terminal capabilities are also stored in the directory service. The application layer signaling protocol establishes and maintains call state between communicating terminals. The logical channel abstraction provides a shared end-to-end context for a call's constituent media and control streams, while isolating the applications from the details of the network transport mechanisms. In addition to supporting simple point-to-point calls, the architecture supports both centralized and decentralized conferencing. We also introduce a simple encapsulation format for voice     

基于MANET接入Internet的动态 网关布局与选取规划模型

 本文以实现MANET接入Internet为应用背景,考虑到动态网关在整个网络接入过程中的重要作用,提出了一种新的适用于MANET接入Internet的动态网关的布局与选取的三层规划模型.该模型通过引入具有协调控制功能的节点(即决策节点)来对接入网络中的网关和路径的使用情况进行调控,通过上下层的交互决策协调网络整体利益和节点局部利益之间的关系,并以此作为配置网关的依据,从而优化网关布局.仿真结果表明,引入该模型后网络的延迟、开销以及吞吐率等性能指标得到了改善,验证了该模型的有效性.     

H.323: the multimedia communications standard for local areanetworks

This article provides an overview of ITU-T Recommendation H.323, “Visual telephone systems and equipment for local area networks which provide a non-guaranteed quality of service.” This recommendation applies to multimedia communications over packet-switched networks, such as Ethernet, which run TCP/IP, IXP/SPX, or other related protocols. In addition to the multimedia terminal, other H.323 components are defined which provide for conference admissions (gatekeeper), multipoint communications (multipoint controller, multipoint processor), and interoperability (gateway) with terminals on other types of networks. H.323 has application to a variety of network media, including local area networks, enterprise networks, metropolitan area networks, wide area networks, dial-up line connections to LANs, and the Internet. This provides the capability to have global multimedia communications from the desktop using existing network infrastructures     

Supplementary services in the H.323 IP telephony network

Traditionally, different networks were developed to handle voice, data, and video. The circuit-switched telephone network carried voice and the packet network carried data. Due to different deployment of these networks, different services were developed, such as voice mail in the telephone network and electronic mail on the Internet. With the revolution of multimedia in the computer industry, voice, video, and data are now being carried on both networks. Supplementary services, such as transfer and forwarding (which were originally developed for private telephone networks and later migrated to public telephone networks) are now being developed for packet networks. The standards for packet networks are being defined in the H.323-based series of ITU-T recommendations. This article provides the H.323 architecture for supplementary services, the differences in deployment of these services between the circuit-switched and packet-switched networks, and interworking of these services across hybrid networks     

QoS routing in ad hoc wireless networks

The emergence of nomadic applications have generated much interest in wireless network infrastructures that support real-time communications. We propose a bandwidth routing protocol for quality-of-service (QoS) support in a multihop mobile network. The QoS routing feature is important for a mobile network to interconnect wired networks with QoS support (e.g., ATM, Internet, etc.). The QoS routing protocol can also work in a stand-alone multihop mobile network for real-time applications. This QoS routing protocol contains end-to-end bandwidth calculation and bandwidth allocation. Under such a routing protocol, the source (or the ATM gateway) is informed of the bandwidth and QoS available to any destination in the mobile network. This knowledge enables the establishment of QoS connections within the mobile network and the efficient support of real-time applications. In addition, it enables more efficient call admission control. In the case of ATM interconnection, the bandwidth information can be used to carry out intelligent handoff between ATM gateways and/or to extend the ATM virtual circuit (VC) service to the mobile network with possible renegotiation of QoS parameters at the gateway. We examine the system performance in various QoS traffic flows and mobility environments via simulation. Simulation results suggest distinct performance advantages of our protocol that calculates the bandwidth information. It is particularly useful in call admission control. Furthermore, “standby” routing enhances the performance in the mobile environment. Simulation experiments show this improvement