Non-Repudiation in Internet Telephony

We present a concept to achieve non-repudiation for natural language conversations over the Internet. The method rests on chained electronic signatures applied to pieces of packet-based, digital, voice communication. It establishes the integrity and authenticity of the bidirectional data stream and its temporal sequence and thus the security context of a conversation. The concept is close to the protocols for Voice over the Internet (VoIP), provides a high level of inherent security, and extends naturally to multilateral non-repudiation, e.g., for conferences. Signatures over conversations can become true declarations of will in analogy to electronically signed, digital documents. This enables binding verbal contracts, in principle between unacquainted speakers, and in particular without witnesses. A reference implementation of a secure VoIP archive is exhibited.Comment: Accepted full research paper at IFIP sec2007, Sandton, South Africa, 14-16 May 200

Signaling for Internet Telephony

Internet telephony must offer the standard telephony services.However, the transition to Internet-based telephony services also provides an opportunity to create new services more rapidly and with lower complexity than in the existing public switched telephone network(PSTN). The Session Initiation Protocol (SIP) is a signaling protocol that creates, modifies and terminates associations between Internet end systems, including conferences and point-to-point calls. SIP supports unicast, mesh and multicast conferences, as well as combinations of these modes. SIP implements services such as call forwarding and transfer, placing calls on hold, camp-on and call queueing by a small set of call handling primitives. SIP implementations can re-use parts of other Internet service protocols such as HTTP and the Real-Time Stream Protocol (RTSP). In this paper, we describe SIP, and show how its basic primitives can be used to construct a wide range of telephony services

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 email, 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 paper, 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 (CGI) that allows trusted users to develop services, and the Call Processing Language (CPL) that allows untrusted users to develop services

Quality aspects of Internet telephony

Internet telephony has had a tremendous impact on how people communicate. Many now maintain contact using some form of Internet telephony. Therefore the motivation for this work has been to address the quality aspects of real-world Internet telephony for both fixed and wireless telecommunication. The focus has been on the quality aspects of voice communication, since poor quality leads often to user dissatisfaction. The scope of the work has been broad in order to address the main factors within IP-based voice communication. The first four chapters of this dissertation constitute the background material. The first chapter outlines where Internet telephony is deployed today. It also motivates the topics and techniques used in this research. The second chapter provides the background on Internet telephony including signalling, speech coding and voice Internetworking. The third chapter focuses solely on quality measures for packetised voice systems and finally the fourth chapter is devoted to the history of voice research. The appendix of this dissertation constitutes the research contributions. It includes an examination of the access network, focusing on how calls are multiplexed in wired and wireless systems. Subsequently in the wireless case, we consider how to handover calls from 802.11 networks to the cellular infrastructure. We then consider the Internet backbone where most of our work is devoted to measurements specifically for Internet telephony. The applications of these measurements have been estimating telephony arrival processes, measuring call quality, and quantifying the trend in Internet telephony quality over several years. We also consider the end systems, since they are responsible for reconstructing a voice stream given loss and delay constraints. Finally we estimate voice quality using the ITU proposal PESQ and the packet loss process. The main contribution of this work is a systematic examination of Internet telephony. We describe several methods to enable adaptable solutions for maintaining consistent voice quality. We have also found that relatively small technical changes can lead to substantial user quality improvements. A second contribution of this work is a suite of software tools designed to ascertain voice quality in IP networks. Some of these tools are in use within commercial systems today

Competition in a Pure World of Internet Telephony

From the angle of competition policy, Voice over IP looks like a panacea. It not only brings better service, but it also increases competitive pressure on former telecommunications monopolists. This paper points to the largely overlooked downside. In a pure world of Internet telephony, there would be no charge for individual calls, nor for telephony, as distinct from other services running over the uniform network. Specifically, establishing property rights for either of these would be costly, whereas these property rights were automatic and free of charge in switched telephony. Giving voice over IP providers classic telephone numbers would enhance systems competition with switched telephony. But this would make it more difficult for clients to swap providers. The anti-competitive caller pays principle would extend to IP telephony.property right, non-linear pricing, pure bundling, club good, cross-subsidisation, packet switched telephony

Why the Government Should Not Regulate Internet Telephony?

The Federal Communications Commission has requested comments on the regulation of voice telephone services delivered over the Internet, dubbed "VoIP" or Voice over Internet Protocol. This paper examines whether there is a need to regulate VoIP. We conclude that there is no economic rationale for regulating VoIP and that consumers will likely be worse off if VoIP is regulated. Furthermore, the emergence of new technologies, such as VoIP, is rapidly eroding the rationale for continuing to regulate local telephone services.

Service Learning in Internet Telephony

Internet telephony can introduce many novel communication services, however, novelty puts learning burden on users. It will be a great help to users if their desired services can be created automatically. We developed an intelligent communication service creation environment which can handle automatic service creation by learning from users' daily communication behaviors. The service creation environment models communication services as decision trees and uses the Incremental Tree Induction (ITI) algorithm for decision tree learning. We use Language for End System Services (LESS) scripts to represent learned results and implemented a simulation environment to verify the learning algorithm. We also noticed that when users get their desired services, they may not be aware of unexpected behaviors that the serivces could introduce, for example, mistakenly rejecting expected calls. In this paper, we also did a comprehensive analysis on communication service fail-safe handling and propose several approaches to create fail-safe services

Interactive Real-Time Embedded Systems Education Infused with Applied Internet Telephony

The transition from traditional circuit-switched phone systems to modern packet-based Internet telephony networks demands tools to support Voice over Internet Protocol (VoIP) development. In this paper, we introduce the XinuPhone, an integrated hardware/software approach for educating users about VoIP technology on a real-time embedded platform. We propose modular course topics for design-oriented, hands-on laboratory exercises: filter design, timing, serial communications, interrupts and resource budgeting, network transmission, and system benchmarking. Our open-source software platform encourages development and testing of new CODECs alongside existing standards, unlike similar commercial solutions. Furthermore, the supporting hardware features inexpensive, readily available components designed specifically for educational and research users on a limited budget. The XinuPhone is especially good for experimenting with design trade-offs as well as interactions between real-time software and hardware components