Language independent on-off voice over IP source model with lognormal transitions

The recent explosive growth of voice over IP (VoIP) solutions calls for accurate modelling of VoIP traffic. This study presents measurements of ON and OFF periods of VoIP activity from a significantly large database of VoIP call recordings consisting of native speakers speaking in some of the world's most widely spoken languages. The impact of the languages and the varying dynamics of caller interaction on the ON and OFF period statistics are assessed. It is observed that speaker interactions dominate over language dependence which makes monologue-based data unreliable for traffic modelling. The authors derive a semi-Markov model which accurately reproduces the statistics of composite dialogue measurements. © The Institution of Engineering and Technology 2013.

Voice over IP 2.0: an analysis of limits and potential of IP2IP telecommunication

Internet Telephony (VoIP) is changing the telecommunication industry. Oftentimes free, VoIP is becoming more and more popular amongst users. Large software companies have entered the market and heavily invest into it. In 2011, for instance, Microsoft bought Skype for 8.5bn USD. This trend increasingly impacts the incumbent telecommunication operators. They see their main source of revenue – classic telephony – under siege and disappear. The thesis at hand develops a most-likely scenario in order to determine how VoIP is evolving further and it predicts, based on a ten-year forecast, the impact it will have on the players in the telecommunication industry.The paper presents a model combining Rogers’ diffusion and Christensen’s innovation research. The model has the goal of explaining the past evolution of VoIP and to isolate the factors that determine the further diffusion of the innovation. Interviews with industry experts serve to assess how the identified factors are evolving.Two propositions are offered. First, VoIP operators are becoming more important in international, corporate, and mobile telephony. End-to-end VoIP (IP2IP) will exhibit strong growth rates and increasingly cannibalize the telephony revenues of the classic operators. Second, fix-net telephony in SMEs and at home will continue to be dominated by the incumbents. Yet, as prices for telephony fall towards zero also they will implement IP2IP in order to save costs. By 2022, up to 90% of the calls will be IP2IP. The author recommends the incumbents and VoIP operators to proactively face the change, to rethink their business strategies, and to even be open for cooperation.

Studio ed analisi della sicurezza del protocollo SIP nelle reti Voice over IP

Nell'era di Internet e della digitalizzazione, anche la telefonia ha avuto la possibilità di evolversi, e grazie alle tecnologie Voice-over-IP è stato possibile realizzare servizi di comunicazione avanzata su reti di dati. Anche se la comunicazione vocale è l'aspetto chiave di questi sistemi, le reti VoIP supportano altri tipi di servizi, tra cui video, messaggistica istantanea, condivisione di file, ecc. Il successo di questa nuova tipologia di rete è dovuto ad una migliore flessibilità rispetto ai vecchi sistemi analogici, grazie ad architetture aperte e implementazioni a livello software, e soprattutto ad un minor costo legato alle apparecchiature ed ai collegamenti utilizzati, ed ai nuovi modelli di business e di consumo sempre più orientati allo sfruttamento della connettività a banda larga. Tuttavia, l'implementazione dei sistemi VoIP rappresenta anche un grado di complessità maggiore in termini di architetture di rete, di protocolli, e di implementazione, e con questo ne segue un incremento delle possibili vulnerabilità. Una falla nella sicurezza in questi sistemi può portare a disservizi e violazione della privacy per gli utenti con conseguenti ripercussioni economiche per i relativi gestori. La tesi analizza la sicurezza delle reti VoIP concentrandosi sul protocollo che sta alla base dei servizi multimediali, il protocollo SIP. SIP è un protocollo di livello applicativo realizzato per creare, modificare e terminare delle sessioni multimediali tra due o più utenti. Dopo un'introduzione alle generalità del protocollo, vengono esaminate le classi di vulnerabilità delle reti VoIP e gli attacchi a SIP, e vengono presentate alcune contromisure attuabili. Viene mostrato un esempio di come vengano attuati alcuni dei principali attacchi a SIP tramite l'utilizzo di appositi strumenti. L'eborato conclude con alcune considerazioni sulle minacce al protocollo e sugli obiettivi futuri che la comunità scientifica dovrebbe perseguire.

Le support de VoIP dans les réseaux maillés sans fil WiMAX en utilisant une approche de contrôle et d'assistance au niveau MAC

Les réseaux maillés sans fil (RMSF), grâce à leurs caractéristiques avantageuses, sont considérés comme une solution efficace pour le support des services de voix, vidéo et de données dans les réseaux de prochaine génération. Le standard IEEE 802.16-d a spécifié pour les RMSF, à travers son mode maillé, deux mécanismes de planifications de transmission de données; à savoir la planification centralisée et la planification distribuée. Dans ce travail, on a évalué le support de la qualité de service (QdS) du standard en se focalisant sur la planification distribuée. Les problèmes du système dans le support du trafic de voix ont été identifiés. Pour résoudre ces problèmes, on a proposé un protocole pour le support de VoIP (AVSP) en tant qu’extension au standard original pour permettre le support de QdS au VoIP. Nos résultats préliminaires de simulation montrent qu’AVSP offre une bonne amélioration au support de VoIP.

Quality of experience evaluation of voice communication: an affect-based approach

Voice communication systems such as Voice-over IP (VoIP), Public Switched Telephone Networks, and Mobile Telephone Networks, are an integral means of human tele-interaction. These systems pose distinctive challenges due to their unique characteristics such as low volume, burstiness and stringent delay/loss requirements across heterogeneous underlying network technologies. Effective quality evaluation methodologies are important for system development and refinement, particularly by adopting user feedback based measurement. Presently, most of the evaluation models are system-centric (Quality of Service or QoS-based), which questioned us to explore a user-centric (Quality of Experience or QoE-based) approach as a step towards the human-centric paradigm of system design. We research an affect-based QoE evaluation framework which attempts to capture users' perception while they are engaged in voice communication. Our modular approach consists of feature extraction from multiple information sources including various affective cues and different classification procedures such as Support Vector Machines (SVM) and k-Nearest Neighbor (kNN). The experimental study is illustrated in depth with detailed analysis of results. The evidences collected provide the potential feasibility of our approach for QoE evaluation and suggest the consideration of human affective attributes in modeling user experience.

Implementation and evaluation of open source unified communications for SMBs

Unified Communication (UC) is the integration of two or more real time communication systems into one platform. Integrating core communication systems into one overall enterprise level system delivers more than just cost saving. These real-time interactive communication services and applications over Internet Protocol (IP) have become critical in boosting employee accessibility and efficiency, improving customer support and fostering business agility. However, some small and medium-sized businesses (SMBs) are far from implementing this solution due to the high cost of initial deployment and ongoing support. In this paper, we will discuss and demonstrate an open source UC solution, viz. “Asterisk” for use by SMBs, and report on some performance tests using SIPp. The contribution from this research is the provision of technical advice to SMBs in deploying UC, which is manageable in terms of cost, ease of deployment and support.

Design and implementation of unified communications as a service based on the OpenStack cloud environment

Cloud Computing, based on early virtual computer concepts and technologies, is now itself a maturing technology in the marketplace and it has revolutionized the IT industry, being the powerful platform that many businesses are choosing to migrate their in-premises IT services onto. Cloud solution has the potential to reduce the capital and operational expenses associated with deploying IT services on their own. In this study, we have implemented our own private cloud solution, infrastructure as a service (IaaS), using the OpenStack platform with high availability and a dynamic resource allocation mechanism. Besides, we have hosted unified communication as a service (UCaaS) in the underlying IaaS and successfully tested voice over IP (VoIP), video conferencing, voice mail and instant messaging (IM) with clients located at the remote site. The proposed solution has been developed in order to give advice to bussinesses that want to build their own cloud environment, IaaS and host cloud services and applicatons in the cloud. This paper also aims at providing an alternate option for proprietary cloud solutions for service providers to consider.

A Decentralized Session Management Framework for Heterogeneous Ad-Hoc and Fixed Networks

Wireless technologies are continuously evolving. Second generation cellular networks have gained worldwide acceptance. Wireless LANs are commonly deployed in corporations or university campuses, and their diffusion in public hotspots is growing. Third generation cellular systems are yet to affirm everywhere; still, there is an impressive amount of research ongoing for deploying beyond 3G systems. These new wireless technologies combine the characteristics of WLAN based and cellular networks to provide increased bandwidth. The common direction where all the efforts in wireless technologies are headed is towards an IP-based communication. Telephony services have been the killer application for cellular systems; their evolution to packet-switched networks is a natural path. Effective IP telephony signaling protocols, such as the Session Initiation Protocol (SIP) and the H 323 protocol are needed to establish IP-based telephony sessions. However, IP telephony is just one service example of IP-based communication. IP-based multimedia sessions are expected to become popular and offer a wider range of communication capabilities than pure telephony. In order to conjoin the advances of the future wireless technologies with the potential of IP-based multimedia communication, the next step would be to obtain ubiquitous communication capabilities. According to this vision, people must be able to communicate also when no support from an infrastructured network is available, needed or desired. In order to achieve ubiquitous communication, end devices must integrate all the capabilities necessary for IP-based distributed and decentralized communication. Such capabilities are currently missing. For example, it is not possible to utilize native IP telephony signaling protocols in a totally decentralized way. This dissertation presents a solution for deploying the SIP protocol in a decentralized fashion without support of infrastructure servers. The proposed solution is mainly designed to fit the needs of decentralized mobile environments, and can be applied to small scale ad-hoc networks or also bigger networks with hundreds of nodes. A framework allowing discovery of SIP users in ad-hoc networks and the establishment of SIP sessions among them, in a fully distributed and secure way, is described and evaluated. Security support allows ad-hoc users to authenticate the sender of a message, and to verify the integrity of a received message. The distributed session management framework has been extended in order to achieve interoperability with the Internet, and the native Internet applications. With limited extensions to the SIP protocol, we have designed and experimentally validated a SIP gateway allowing SIP signaling between ad-hoc networks with private addressing space and native SIP applications in the Internet. The design is completed by an application level relay that permits instant messaging sessions to be established in heterogeneous environments. The resulting framework constitutes a flexible and effective approach for the pervasive deployment of real time applications.

Analysis of information leakage from encrypted Skype conversations

Voice over IP (VoIP) has experienced a tremendous growth over the last few years and is now widely used among the population and for business purposes. The security of such VoIP systems is often assumed, creating a false sense of privacy. This paper investigates in detail the leakage of information from Skype, a widely used and protected VoIP application. Experiments have shown that isolated phonemes can be classified and given sentences identified. By using the dynamic time warping (DTW) algorithm, frequently used in speech processing, an accuracy of 60% can be reached. The results can be further improved by choosing specific training data and reach an accuracy of 83% under specific conditions. The initial results being speaker dependent, an approach involving the Kalman filter is proposed to extract the kernel of all training signals.

On the Privacy of Encrypted Skype Communications

The privacy of voice over IP (VoIP) systems is achieved by compressing and encrypting the sampled data. This paper investigates in detail the leakage of information from Skype, a widely used VoIP application. In this research, it has been demonstrated by using the dynamic time warping (DTW) algorithm, that sentences can be identified with an accuracy of 60%. The results can be further improved by choosing specific training data. An approach involving the Kalman filter is proposed to extract the kernel of all training signals.

Modelo estendido de adoção da tecnologia de comunicação pessoal de voz pela internet

Estudos científicos têm demonstrado a existência de diversos fatores de influência sobre o processo de adoção de Tecnologia da Informação. Vários aspectos potencializadores das intenções de uso das tecnologias parecem estar presentes nas decisões sobre a adoção de tecnologia como, por exemplo, vantagens relativas obtidas pelo uso da tecnologia, conhecimento requerido para operação dos sistemas, facilidade e utilidade uso da tecnologia, entre outros. No entanto, há estudos que demonstram que o processo de adoção de tecnologia pode não ser explicado, somente, por aspectos financeiros e mercadológicos, mas englobar fatores endógenos que interferem nas decisões dos usuários sobre o uso de tecnologias de informação. Teorias e modelos de adoção de tecnologias conseguem explicar uma parcela dos motivos que levam os indivíduos a se comportarem de acordo com o uso de uma determinada tecnologia. Por exemplo: hábito individual, conectividade e conveniência podem influenciar as preferências de uso de uma determinada tecnologia de informação. De acordo com estas teorias, é possível analisar as influências que os indivíduos percebem e consideram nas decisões como justificativas sobre o uso de tecnologias de comunicação pessoal, além da busca exclusiva por resultados econômicos. Observa-se que um dos segmentos de tecnologia que apresenta condições de diferentes ofertas e múltiplas funcionalidades se refere ao segmento de tecnologias de comunicação de voz no qual o usuário pode se comunicar por meio de linhas telefônicas fixas, móveis, Internet, entre outras formas. Para a operacionalização de uma análise de adoção de tecnologia que englobe múltiplas interações de influências ao usuário, optou-se pela contextualização do estudo com foco na tecnologia de comunicação de voz pela Internet – VOIP, pois quando comparada com outras tecnologias de comunicação, adiciona-se que a gratuidade de ligações entre usuários de um mesmo sistema pode ser apresentada como um dos benefícios econômicos principais, aliada a outros benefícios provindos da telefonia em si. Os resultados obtidos por esta pesquisa confirmam a influência de diversos fatores posicionados em diferentes dimensões e proporcionam conclusões relevantes à adoção das tecnologias de comunicação de voz sobre Internet. Conclui-se que as percepções individuais sobre as características da tecnologia, a rede de contatos do usuário, hábito de uso e incentivos comerciais destinados ao uso de outras tecnologias de comunicação podem formar uma rede de influências à adoção da telefonia VOIP frente às percepções sobre os benefícios que podem ser obtidos com o uso desta aplicação.

Sistema de conmutación y control de llamadas basado en OpenBTS y Asterisk

El presente trabajo empleó herramientas de hardware y software de licencia libre para el establecimiento de una estación base celular (BTS) de bajo costo y fácil implementación. Partiendo de conceptos técnicos que facilitan la instalación del sistema OpenBTS y empleando el hardware USRP N210 (Universal Software Radio Peripheral) permitieron desplegar una red análoga al estándar de telefonía móvil (GSM). Usando los teléfonos móviles como extensiones SIP (Session Initiation Protocol) desde Asterisk, logrando ejecutar llamadas entre los terminales, mensajes de texto (SMS), llamadas desde un terminal OpenBTS hacia otra operadora móvil, entre otros servicios.

Procedimiento para implementar QoS en la capa de acceso en redes de próxima generación enfocado en el servicio de voz

En la actualidad, todos los servicios convergen en una Red de Próxima Generación [NGN]. Asimismo, las exigencias de calidad de servicio [QoS], por los requerimientos de los usuarios, son más estrictas, lo que hace necesario plantear procedimientos de QoS que garanticen una operación eficaz en el transporte de los servicios más críticos y de tiempo real ¿como la voz¿, garantizando la disminución de los problemas de latencia, jitter, pérdida de paquetes y eco. Los operadores de Telecomunicaciones deben aplicar las regulaciones emitidas por la Comisión de Regulación de Comunicaciones de Colombia [CRC] y ajustarse a las recomendaciones Y.1540 y Y.1541 de la Unión Internacional de Telecomunicaciones [UIT]. Este documento presenta un procedimiento para aplicar mecanismos de QoS en una NGN en el acceso xDSL con el fin de mantener un nivel de QoS en Voz sobre IP (VoIP) que permita su provisión, con eficiencia económica y técnica, en favor tanto del cliente, como del operador de telecomunicaciones.