What’s the Hang Up? The Future of VoIP Regulation and Taxation in New Hampshire

Alice in Austria wishes to call her friend Bob in Boston, using a Boston area code to avoid charges for an international call. Using VoIP, Alice may initiate her call from any location in Austria where she may find Internet access. Once Alice connects to the Internet, she can transmit her call with the aid of a VoIP service provider, such as Skype. In order to hear and communicate with Bob, Alice can rely on a microphone and a headset that she can plug into her computer. Through VoIP, not only may Alice carry on a telephone conversation, but most service providers also allow her to record conversations and manage other information, such as voice mail. The rise of Voice over Internet Protocol (“VoIP”) services “means nothing less than the death of the traditional telephone business,” as the ability to make free calls over a high-speed Internet connection in the future “undermines the existing pricing model for telephony.” This disruptive, convergent technology is blurring the boundary between Internet services and telephone services because VoIP functions like the traditional telephone system, but travels as ones and zeros through a broadband Internet connection. As a result, the Federal Communications Commission (“FCC”) has questioned whether to classify VoIP as an information service, generally free from FCC regulation under the Telecommunications Act of 1996, or as a telecommunication service, subject to a comprehensive regulatory regime and common carrier obligations. This note discusses why most VoIP services, with the exception of phone-to-phone Internet Protocol (“IP”) telephony, should be classified as information services and, as such, should remain free from state taxation – focusing specifically on the taxation in New Hampshire. Part II focuses on the technology of VoIP and how it differs from traditional telephony. Part III discusses the distinction between information and telecommunication services in the Telecommunications Act of 1996, whether VoIP may qualify as Internet access in light of the Internet Tax Freedom Act (“ITFA”) of 1998, and the federal regulation of VoIP. Finally, Part IV addresses the debate over taxation of VoIP in New Hampshire and discusses why VoIP services should not yet be taxed by the New Hampshire Department of Revenue Administration in light of federal law and the best interests of local businesses and consumers

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

VoIP security - attacks and solutions

Voice over IP (VoIP) technology is being extensively and rapidly deployed. Flexibility and cost efficiency are the key factors luring enterprises to transition to VoIP. Some security problems may surface with the widespread deployment of VoIP. This article presents an overview of VoIP systems and its security issues. First, we briefly describe basic VoIP architecture and its fundamental differences compared to PSTN. Next, basic VoIP protocols used for signaling and media transport, as well as defense mechanisms are described. Finally, current and potential VoIP attacks along with the approaches that have been adopted to counter the attacks are discussed

The Beginnings and Prospective Ending of “End-to-End”: An Evolutionary Perspective On the Internet’s Architecture

The technology of “the Internet” is not static. Although its “end-to- end” architecture has made this “connection-less” communications system readily “extensible,” and highly encouraging to innovation both in hardware and software applications, there are strong pressures for engineering changes. Some of these are wanted to support novel transport services (e.g. voice telephony, real-time video); others would address drawbacks that appeared with opening of the Internet to public and commercial traffic - e.g., the difficulties of blocking delivery of offensive content, suppressing malicious actions (e.g. “denial of service” attacks), pricing bandwidth usage to reduce congestion. The expected gains from making “improvements” in the core of the network should be weighed against the loss of the social and economic benefits that derive from the “end-to-end” architectural design. Even where technological “fixes” can be placed at the networks’ edges, the option remains to search for alternative, institutional mechanisms of governing conduct in cyberspace.

Improving the robustness of CELP-like speech decoders using late-arrival packets information : application to G.729 standard in VoIP

L'utilisation de la voix sur Internet est une nouvelle tendance dans Ie secteur des télécommunications et de la réseautique. La paquetisation des données et de la voix est réalisée en utilisant Ie protocole Internet (IP). Plusieurs codecs existent pour convertir la voix codée en paquets. La voix codée est paquetisée et transmise sur Internet. À la réception, certains paquets sont soit perdus, endommages ou arrivent en retard. Ceci est cause par des contraintes telles que Ie délai («jitter»), la congestion et les erreurs de réseau. Ces contraintes dégradent la qualité de la voix. Puisque la transmission de la voix est en temps réel, Ie récepteur ne peut pas demander la retransmission de paquets perdus ou endommages car ceci va causer plus de délai. Au lieu de cela, des méthodes de récupération des paquets perdus (« concealment ») s'appliquent soit à l'émetteur soit au récepteur pour remplacer les paquets perdus ou endommages. Ce projet vise à implémenter une méthode innovatrice pour améliorer Ie temps de convergence suite a la perte de paquets au récepteur d'une application de Voix sur IP. La méthode a déjà été intégrée dans un codeur large-bande (AMR-WB) et a significativement amélioré la qualité de la voix en présence de <<jitter » dans Ie temps d'arrivée des trames au décodeur. Dans ce projet, la même méthode sera intégrée dans un codeur a bande étroite (ITU-T G.729) qui est largement utilise dans les applications de voix sur IP. Le codeur ITU-T G.729 défini des standards pour coder et décoder la voix a 8 kb/s en utilisant 1'algorithme CS-CELP (Conjugate Stmcture Algebraic Code-Excited Linear Prediction).Abstract: Voice over Internet applications is the new trend in telecommunications and networking industry today. Packetizing data/voice is done using the Internet protocol (IP). Various codecs exist to convert the raw voice data into packets. The coded and packetized speech is transmitted over the Internet. At the receiving end some packets are either lost, damaged or arrive late. This is due to constraints such as network delay (fitter), network congestion and network errors. These constraints degrade the quality of speech. Since voice transmission is in real-time, the receiver can not request the retransmission of lost or damaged packets as this will cause more delay. Instead, concealment methods are applied either at the transmitter side (coder-based) or at the receiver side (decoder-based) to replace these lost or late-arrival packets. This work attempts to implement a novel method for improving the recovery time of concealed speech The method has already been integrated in a wideband speech coder (AMR-WB) and significantly improved the quality of speech in the presence of jitter in the arrival time of speech frames at the decoder. In this work, the same method will be integrated in a narrowband speech coder (ITU-T G.729) that is widely used in VoIP applications. The ITUT G.729 coder defines the standards for coding and decoding speech at 8 kb/s using Conjugate Structure Algebraic Code-Excited Linear Prediction (CS-CELP) Algorithm