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     

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     

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.     

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     

Internet protocol support for telephony

In this paper, we consider the evolution of telephone networks from time-division multiplexing circuit switching to packet switching and, in particular, to packet switching-based on Internet Protocol (IP-supported telephony). We analyze IP-supported telephony design solutions by proposing a layered reference model in which each layer is associated to a subset of the functions that support telephony. We use the reference model to establish a terminology and a framework for the comparison of the design solutions. We group the design solutions in scenarios and compare them in terms of the reference model proposed. We then focus on IP telephony, in which IP is used in telephone company networks, and on Internet telephony, in which the Internet is used to support telephony. We show that they both can be seen as implementations of the same architecture, which consists of a set of components, associated to functions, and of the interactions among these components. We then consider the issue of voice-data integration and analyze the variety of design solutions that can be adopted to integrate voice and data.     

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: 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 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.     

Computer telephony architectures: MVIP, H-MVIP, and SCbus

The author discusses the objectives of a computer-telephony integration (CTI) bus for the PC, TDM, network synchronization, digital switching, multichassis connections, MVIP bus technologies, SCbus technology, and CTI client server architectures     

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     

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     

UMTS-to-IP QoS mapping for voice and video telephony services

Quality of service mapping between UMTS services and IP transport is crucial for maintaining a suitable end-to-end delay for emerging UMTS multimedia telephony. However, due to incompatibilities in QoS classifications within these two technologies, straightforward mapping is impossible and current proposals within the 3GPP could lead to unpredictable and undesirable behavior for certain services. In this article we focus on two very important UMTS services, voice and video telephony, and establish the QoS issues that exist for these services. We then propose a refined QoS mapping that differentiates between the transmission of voice and video-telephony and a weighted fair queuing scheduler to schedule the transmissions. Through a simulation study, we show the effect on the queuing delays of both traffic types when their WFQ weights vary and then derive an optimal weight that provides the best overall delays for multimedia telephony services.     

Teleaction services: standards and architectures

The article provides a review of the standards and architectures that can be used to implement teleaction services. Although they are already penetrating the market and are recognized as having substantial potential for growth, the work on standardization of these services is not complete. The article analyzes the variety of implementation architectures and standards that may be deployed; illustrates the cost benefits that can be achieved by deploying new technologies; and guides the standardization process by identifying the critical areas where standards will have maximum impact on business development. Drawing an architectural framework for the provision of teleaction services, the authors identify the generic capabilities for the support of teleaction services and analyze the main architectural implementations for teleaction services. Several types of networks based on existing technology are reviewed. ISDN-based implementations and their associated standards are described, and aspects such as protocols, interconnection of networks, and migration from current arrangements to ISDN implementations are analyzed. Quality of service (QOS) is of central importance. Requirements for QOS are reviewed mainly by reviewing some of the European standards. The significant progress in standardization of home systems is considered and a review of the major products and technical trends in home systems and their use as customer premises equipment for teleaction services is then presented. Finally, the authors review a system under development in Denmark and draw some conclusions     

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     

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     

Internet2 QBone: building a testbed for differentiated services

The Internet2 project is a partnership of over 130 U.S. universities, 40 corporations, and 30 other organizations. Since its inception, one of the primary technical objectives of Internet2 has been to engineer scalable, interoperable, and administrable interdomain QoS to support an evolving set of new advanced networked applications. Applications like distance learning, remote instrument access and control, advanced scientific visualization, and networked collaboratories will allow universities to fulfill their research and education missions into the future, but only if the network QoS these applications require can be ensured. To meet this challenge, the Internet2 QBone initiative has brought together a dedicated group of U.S. university and federal agency networks, international research networks, engineers, researchers, and applications developers to build a testbed for interdomain IP differentiated services. This article presents the engineering motivations behind DiffServ and its adoption by Internet2, provides an overview of the QBone architecture, and describes its anticipated deployment, including plans for a trial inter-domain bandwidth brokering architecture. Security aspects are considered togethered together with an inter-bandwidth broker reservation signaling protocol