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.     

Internet telephony in India

Internet telephony is a novel and cheaper method of communication and conducting business over the Internet. The paper presents an overview of Internet telephony, its methods, viz. PC-to-PC, PC-to-telephone, telephone-to-telephone and telephone-to-PC; benefits in cost advantage, simplification, consolidation, higher efficiency and reliability, etc., quality issues, protocols and drivers; challenges and regulatory framework; and status of Internet telephony in Asia Pacific region. Further, highlights its potentiality for India, implications of guidelines of Internet telephony, issues of concern, etc. Concludes that Internet telephony cannot make compromises in voice quality, reliability, scalability and manageability, and work seamlessly with telephone systems all over the world. Internet telephony will prove to be a boon for a price-sensitive market like India and rural telephony will receive an impetus. The Government of India may further deregulate the market and allow phone-to-phone telephony through the Internet and open long distance calling within the country for ISPs to realize “telecom for the common man” or “telecom for all” a reality.     

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     

An architecture for residential Internet telephony service

A new architecture that can be used for offering an Internet telephony service to residential customers is introduced. The architecture addresses scalability and availability requirements of mass-market deployment of carrier-grade services and supports interconnection with SS7 for Internet telephony calls to the public switched telephone network. The architecture is based on the concept of a gateway decomposition that separates the media transformation function of today's H.323 gateways from the gateway control function of the gateways and centralizes the intelligence in a call agent. The media gateway control protocol is introduced as the protocol between the call agent that assumes the gateway control function and the gateway that provides just the media transformation function. Interworking between the architecture and the public switched telephone network, the session initiation protocol, and H.323 are also discussed     

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     

Internet telephony and modem delay

The vast majority of consumer Internet access is via dialup modems. These modems are a primary source of the delay experienced on voice over IP cells. The author focuses on understanding the causes of delay within analog modems, with the objective of developing recommendations to minimize delay for VoIP applications. First, the relative importance of modem delay is assessed versus other causes of VolP delay (e.g., PC client, IP network), and delay from other access types (e.g., ISDN, cable modems). Second, the characteristics of VoIP data streams are examined as a key determinant of modem delay. Third, the internal operation of modems is examined with respect to delay when transmitting VoIP data streams. Finally, some recommendations and conclusions are presented     

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     

Advanced services architectures for Internet telephony: a criticaloverview

Advanced services are differentiating factors and crucial to service providers' survival and success. Examples are credit card calling, call forwarding, and toll-free calling. In classical telephony's early days their implementation was embedded in switching software, and this hindered fast deployment. A more modern architecture known as the intelligent network (IN) was born in the 1980s, allowing implementation in separate nodes, resulting in faster deployment of new services. Two tracks are emerging for Internet telephony: one from the ITU-T and the other from the IETF. As far as advanced services are concerned, the ITU-T track offers a rather archaic architecture, reminiscent of the early days of classical telephony. On the other hand, the IETF architecture, although more modern, does have a few pitfalls. There is plenty of room for improvement to both. This article scrutinizes the ITU-T and IETF advanced services architectures for Internet telephony. Salient features are reviewed and weaknesses pinpointed. Although these architectures are constantly evolving, alternatives may emerge. We provide a discussion of two potential alternatives: IN-based architectures and mobile-agent-based architectures     

Terminating telephony services on the Internet

We propose a general purpose service architecture for realizing services which start in the Public Switched Telephone Network (PSTN) but terminate and execute on the Internet. We discuss the needs for such services, our early research efforts in this direction which lead to prototyping certain benchmark services, and the current state of work in this area. We demonstrate the feasibility of the architecture by focusing on services which involve wireline PSTN as well as the wireless aspects (2 G, 2.5 G) of the PSTN. Our methodology is attractive since it keeps each of the domains (PSTN and Internet) unaware as to where the service is executing with respect to which domain actually requested the service. Individual entities participating in the service do not have any knowledge that external entities from another domain also contributed in the execution and fulfillment of such services. Our approach, as embodied in the service architecture, is to leverage the best of the Internet protocols (SIP, XML, HTTP) and technologies (instant messaging, presence) to provide a general framework for personalized service specification and execution.     

Packet telephony support for the IEEE 802.11 wireless LAN

Packet telephony is one of the most promising applications in the Internet. In this paper, we propose a modified MAC protocol supporting voice traffic over the IEEE 802.11 WLAN. The proposed scheme adapts the power-saved mode of the IEEE 802.11 specifications in such a way that it approaches the TDM access mode carrying voice traffic, and is compatible with the IEEE 802.11 standard. Simulation results show that the proposed scheme does not degrade the performance of the IEEE 802.11 WLAN using the DCF and also provides good voice quality     

The 3G-324M protocol for conversational video telephony

As mobile operators worldwide migrate to third-generation (3G) networks, conversational video-telephony services are becoming a key differentiator between new 3G offerings and existing 2G/2.5G services. Although it's possible to have limited video-based services - such as a multimedia messaging service - that deliver pictures and video clips over 2.5G services, these are delay-insensitive applications that could run over a packet-based wireless network like general packet radio service (GPRS) or code division multiple access (CDMA)'s 1XRTT. For delay-sensitive applications such as conversational video telephony, present 3G packet bearers are inadequate, and the Third Generation Partnership Project (3GPP; http://www.3gpp.org) mandates using the 3G bandwidth-guaranteed circuit-switched bearer and the 3G-324M system. The 3G-324M system is a derivative of the International Telecommunication Union (ITU) H.324 protocol standard for low-bitrate multimedia communication, which ITU-T developed for the public switched telephone network (PSTN). This article describes the 3G-324M system, which has been adopted by both 3GPP and 3GPP2 (htpp://www.3gpp2.org), as well as its H.324 roots.     

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     

Regulatory challenges for the transition of public telephony to the Internet

Fixed and mobile telephony with other telecommunication services are moving to a fundamentally different infrastructure as the transition proceeds from the public telephone network to the Internet. There are many opportunities to make significant progress on major developments such as separation of operator independent functions, modular design, and the integration of other desirable features of sustainable Internet solutions. This progress also means that the impacts of some shortcomings of currently competitive telephony services will decline??another benefit of the transition. However, because telephony services are essentially being ??emulated?? over the Internet infrastructure, other aspects of these services will be changed in ways that could affect the public good. In particular, trust-related issues such as authentication and validation, along with their business implications, present significant challenges. This paper explores these issues, including the opportunity that industry and the regulators now have to address them, and suggests approaches to sustainable solutions that can benefit all the stakeholders in this transition.     

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     

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     

Integrated circuits for telephony

Many integrated circuits for large-scale application to telephone networks have been designed, including subscriber line interfaces, antialiasing filters, analog-to-digital-to-analog converters, and tone receivers. This paper summarizes in a unified fashion both interfacing and functional requirements for these devices, as well as the related circuit and technology approaches which have been utilized.     

一种新的适合于Internet环境的密钥分配协议

提出了一个基于椭圆曲线数字签名密钥分配的新协议，该协议去掉了引起争议的事件标记，同时增加了可认证性，并在该协议基础上提出了适合于Internet的分级密钥分配协议。