Quality of Service (QoS) in MPLS networks

Telecommunications networks have been always expanding and thanks to it, new services have appeared. The old mechanisms for carrying packets have become obsolete due to the new service requirements, which have begun working in real time. Real time traffic requires strict service guarantees. When this traffic is sent through the network, enough resources must be given in order to avoid delays and information losses. When browsing through the Internet and requesting web pages, data must be sent from a server to the user. If during the transmission there is any packet drop, the packet is sent again. For the end user, it does not matter if the webpage loads in one or two seconds more. But if the user is maintaining a conversation with a VoIP program, such as Skype, one or two seconds of delay in the conversation may be catastrophic, and none of them can understand the other. In order to provide support for this new services, the networks have to evolve. For this purpose MPLS and QoS were developed. MPLS is a packet carrying mechanism used in high performance telecommunication networks which directs and carries data using pre-established paths. Now, packets are forwarded on the basis of labels, making this process faster than routing the packets with the IP addresses. MPLS also supports Traffic Engineering (TE). This refers to the process of selecting the best paths for data traffic in order to balance the traffic load between the different links. In a network with multiple paths, routing algorithms calculate the shortest one, and most of the times all traffic is directed through it, causing overload and packet drops, without distributing the packets in the other paths that the network offers and do not have any traffic. But this is not enough in order to provide the real time traffic the guarantees it needs. In fact, those mechanisms improve the network, but they do not make changes in how the traffic is treated. That is why Quality of Service (QoS) was developed. Quality of service is the ability to provide different priority to different applications, users, or data flows, or to guarantee a certain level of performance to a data flow. Traffic is distributed into different classes and each of them is treated differently, according to its Service Level Agreement (SLA). Traffic with the highest priority will have the preference over lower classes, but this does not mean it will monopolize all the resources. In order to achieve this goal, a set policies are defined to control and alter how the traffic flows. Possibilities are endless, and it depends in how the network must be structured. By using those mechanisms it is possible to provide the necessary guarantees to the real-time traffic, distributing it between categories inside the network and offering the best service for both real time data and non real time data. Las Redes de Telecomunicaciones siempre han estado en expansión y han propiciado la aparición de nuevos servicios. Los viejos mecanismos para transportar paquetes se han quedado obsoletos debido a las exigencias de los nuevos servicios, que han comenzado a operar en tiempo real. El tráfico en tiempo real requiere de unas estrictas garantías de servicio. Cuando este tráfico se envía a través de la red, necesita disponer de suficientes recursos para evitar retrasos y pérdidas de información. Cuando se navega por la red y se solicitan páginas web, los datos viajan desde un servidor hasta el usuario. Si durante la transmisión se pierde algún paquete, éste se vuelve a mandar de nuevo. Para el usuario final, no importa si la página tarda uno o dos segundos más en cargar. Ahora bien, si el usuario está manteniendo una conversación usando algún programa de VoIP (como por ejemplo Skype) uno o dos segundos de retardo en la conversación podrían ser catastróficos, y ninguno de los interlocutores sería capaz de entender al otro. Para poder dar soporte a estos nuevos servicios, las redes deben evolucionar. Para este propósito se han concebido MPLS y QoS MPLS es un mecanismo de transporte de paquetes que se usa en redes de telecomunicaciones de alto rendimiento que dirige y transporta los datos de acuerdo a caminos preestablecidos. Ahora los paquetes se encaminan en función de unas etiquetas, lo cual hace que sea mucho más rápido que encaminar los paquetes usando las direcciones IP. MPLS también soporta Ingeniería de Tráfico (TE). Consiste en seleccionar los mejores caminos para el tráfico de datos con el objetivo de balancear la carga entre los diferentes enlaces. En una red con múltiples caminos, los algoritmos de enrutamiento actuales calculan el camino más corto, y muchas veces el tráfico se dirige sólo por éste, saturando el canal, mientras que otras rutas se quedan completamente desocupadas. Ahora bien, esto no es suficiente para ofrecer al tráfico en tiempo real las garantías que necesita. De hecho, estos mecanismos mejoran la red, pero no realizan cambios a la hora de tratar el tráfico. Por esto es por lo que se ha desarrollado el concepto de Calidad de Servicio (QoS). La calidad de servicio es la capacidad para ofrecer diferentes prioridades a las diferentes aplicaciones, usuarios o flujos de datos, y para garantizar un cierto nivel de rendimiento en un flujo de datos. El tráfico se distribuye en diferentes clases y cada una de ellas se trata de forma diferente, de acuerdo a las especificaciones que se indiquen en su Contrato de Tráfico (SLA). EL tráfico con mayor prioridad tendrá preferencia sobre el resto, pero esto no significa que acapare la totalidad de los recursos. Para poder alcanzar estos objetivos se definen una serie de políticas para controlar y alterar el comportamiento del tráfico. Las posibilidades son inmensas dependiendo de cómo se quiera estructurar la red. Usando estos mecanismos se pueden proporcionar las garantías necesarias al tráfico en tiempo real, distribuyéndolo en categorías dentro de la red y ofreciendo el mejor servicio posible tanto a los datos en tiempo real como a los que no lo son.

Efecto del tamaño de la ventana de promediado y la histéresis variable en algoritmos de handover horizontal en redes convergentes

En las redes convergentes inalámbricas, el traspaso horizontal entre distintos puntos de acceso de la red WLAN es una gran fuente de degradación de la calidad de la VoIP y otros servicios conversacionales en tiempo real. Esto es debido a que este tipo de redes no fueron concebidas originalmente para soportar este tipo de servicios, y los traspasos siguen un protocolo ¿cortar antes de realizar¿, produciéndose interrupciones en la comunicación motivadas por el tiempo que necesitan los terminales en volver a asociarse a la red. En este artículo se estudia el efecto que tienen el tamaño de la ventana de promediado de la señal, la histéresis variable y el retardo del handover por parecido entre potencia de puntos de acceso de destino sobre el número de traspasos y las caídas de la potencia de señal por debajo del valor de sensibilidad del terminal, causantes principales de las interrupciones en la comunicación, y, con ello, de la degradación de la calidad de las comunicaciones.

Aprendizaje, simulación y desarrollo sobre las tecnologías de voz y datos

El objetivo del Proyecto Fin de Carrera (PFC) es el de conocer, simular y crear una red VoIP sobre una red de datos en un entorno docente, más concretamente, en la asignatura Redes y Servicios de telecomunicación en Grado en Ingeniería de Telecomunicaciones en la Universidad Politécnica de Madrid (UPM). Una vez se adquieran los conocimientos necesarios, se propondrán una serie de prácticas para que los alumnos se vayan familiarizando con el software y hardware utilizados, de manera que, se irá subiendo el grado de dificultad hasta que puedan realizar una auténtica red VoIP por sí mismos. A parte de la realización de las prácticas, los alumnos deberán pasar una prueba de los conocimientos adquiridos al final de cada práctica mediante preguntas tipo test. Los sistemas elegidos para la implantación de una red VoIP en los módulos de laboratorio son: 3CX System Phone y Asteisk-Trixbox. Los cuales, son capaces de trabajar mediante gestores gráficos para simplificar el nivel de dificultad de la configuración. 3CX es una PBX que trabaja sobre Windows y se basa exclusivamente en el protocolo SIP. Esto facilita el manejo para usuarios que solo han usado Windows sin quitar funcionalidades que tienen otras centralitas en otros sistemas operativos. La versión demo activa todas las opciones para poder familiarizarse con este sistema. Por otro lado, Asterisk trabaja en todas las plataformas, aunque se ha seleccionado trabajar sobre Linux. Esta selección se ha realizado porque el resto de plataformas limitan la configuración de la IP PBX, esta es de código abierto y permite realizar todo tipo de configuraciones. Además, es un software gratuito, esto es una ventaja a la hora de configurar novedades o resolver problemas, ya que hay muchos especialistas que dan soporte y ayudan de forma gratuita. La voz sobre Internet es habitualmente conocida como VoIP (Voice Over IP), debido a que IP (Internet Protocol) es el protocolo de red de Internet. Como tecnología, la VoIP no es solo un paso más en el crecimiento de las comunicaciones por voz, sino que supone integrar las comunicaciones de datos y las de voz en una misma red, y en concreto, en la red con mayor cobertura mundial: Internet. La mayor importancia y motivación de este Proyecto Fin de Carrera es que el alumno sea capaz de llegar a un entorno laboral y pueda tener unos conocimientos capaces de afrontar esta tecnología que esta tan a la orden del día. La importancia que estas redes tienen y tendrán en un futuro muy próximo en el mundo de la informática y las comunicaciones. Cabe decir, que se observa que estas disciplinas tecnológicas evolucionan a pasos agigantados y se requieren conocimientos más sólidos. ABSTRACT. The objective of my final project during my studies in university was, to simulate and create a VoIP network over a data network in a teaching environment, more specifically on the subject of telecommunications networks and services in Telecommunication Engineering Degree in Polytechnic University of Madrid (UPM). Once acquiring the necessary knowledge a number of practices were proposed to the students to become familiar with the software and hardware used, so that it would rise to the level of difficulty that they could make a real VoIP network for themselves. Parts of the experimental practices were that students must pass a test of knowledge acquired at the end of each practice by choice questions. The systems chosen for the implementation of a VoIP network in the laboratory modules are: 3CX Phone System and Asteisk - Trixbox. Which were able to work with graphics operators to simplify the difficulty level of the configuration. 3CX is a PBX that works on Windows and is based solely on the SIP protocol. This facilitates handling for users who have only used Windows without removing functionality with other exchanges in other operating systems. Active demo version all options to get to grips with this system. Moreover, Asterisk works on all platforms, but has been selected to work on Linux. This selection was made because other platforms limit the IP PBX configuration, as this is open source and allows all kinds of configurations. Also, Linux is a free software and an advantage when configuring new or solve problems, as there are many specialists that support and help for free. Voice over Internet is commonly known as VoIP (Voice Over IP), because IP (Internet Protocol) is the Internet protocol network. As technology, VoIP is not just another step in the growth of voice communications, but communications of integrating data and voice on a single network, and in particular, in the network with the largest global coverage: Internet. The increased importance and motivation of this Thesis is that the student is able to reach a working environment and may have some knowledge to deal with these technologies that is so much the order of the day. The importances of these networks have and will be of essences in the very near future in the world of computing and communications. It must be said it is observed that these technological disciplines evolve by leaps and bounds stronger knowledge required.

Diseño y configuración de una red de comunicaciones en un Centro de Contactos - Contact Center

Este proyecto muestra una solución de red para una empresa que presta servicios de Contact Center desde distintas sedes distribuidas geográficamente, utilizando la tecnología de telefonía sobre IP. El objetivo de este proyecto es el de convertirse en una guía de diseño para el despliegue de soluciones de red utilizando los actuales equipos de comunicaciones desarrollados por el fabricante Cisco Systems, Inc., los equipos de seguridad desarrollados por el fabricante Fortinet y los sistemas de telefonía desarrollados por Avaya Inc. y Oracle Corporation, debido a su gran penetración en el mercado y a las aportaciones que cada uno ha realizado en el sector de Contact Center. Para poder proveer interconexión entre las sedes de un Contact Center se procede a la contratación de un acceso a la red MPLS perteneciente a un operador de telecomunicaciones, quien provee conectividad entre las sedes utilizando la tecnología VPN MPLS con dos accesos diversificados entre sí desde cada una de las sedes del Contact Center. El resultado de esta contratación es el aprovechamiento de las ventajas que un operador de telecomunicaciones puede ofrecer a sus clientes, en relación a calidad de servicio, disponibilidad y expansión geográfica. De la misma manera, se definen una serie de criterios o niveles de servicio que aseguran a un Contact Center una comunicación de calidad entre sus sedes, entendiéndose por comunicación de calidad aquella que sea capaz de transmitirse con unos valores mínimos de pérdida de paquetes así como retraso en la transmisión, y una velocidad acorde a la demanda de los servicios de voz y datos. Como parte de la solución, se diseña una conexión redundante a Internet que proporciona acceso a todas las sedes del Contact Center. La solución de conectividad local en cada una de las sedes de un Contact Center se ha diseñado de manera general acorde al volumen de puestos de usuarios y escalabilidad que pueda tener cada una de las sedes. De esta manera se muestran varias opciones asociadas al equipamiento actual que ofrece el fabricante Cisco Systems, Inc.. Como parte de la solución se han definido los criterios de calidad para la elección de los Centros de Datos (Data Center). Un Contact Center tiene conexiones hacia o desde las empresas cliente a las que da servicio y provee de acceso a la red a sus tele-trabajadores. Este requerimiento junto con el acceso y servicios publicados en Internet necesita una infraestructura de seguridad. Este hecho da lugar al diseño de una solución que unifica todas las conexiones bajo una única infraestructura, dividiendo de manera lógica o virtual cada uno de los servicios. De la misma manera, se ha definido la utilización de protocolos como 802.1X para evitar accesos no autorizados a la red del Contact Center. La solución de voz elegida es heterogénea y capaz de soportar los protocolos de señalización más conocidos (SIP y H.323). De esta manera se busca tener la máxima flexibilidad para establecer enlaces de voz sobre IP (Trunk IP) con proveedores y clientes. Esto se logra gracias a la utilización de SBCs y a una infraestructura interna de voz basada en el fabricante Avaya Inc. Los sistemas de VoIP en un Contact Center son los elementos clave para poder realizar la prestación del servicio; por esta razón se elige una solución redundada bajo un entorno virtual. Esta solución permite desplegar el sistema de VoIP desde cualquiera de los Data Center del Contact Center. La solución llevada a cabo en este proyecto está principalmente basada en mi experiencia laboral adquirida durante los últimos siete años en el departamento de comunicaciones de una empresa de Contact Center. He tenido en cuenta los principales requerimientos que exigen hoy en día la mayor parte de empresas que desean contratar un servicio de Contact Center. Este proyecto está dividido en cuatro capítulos. El primer capítulo es una introducción donde se explican los principales escenarios de negocio y áreas técnicas necesarias para la prestación de servicios de Contact Center. El segundo capítulo describe de manera resumida, las principales tecnologías y protocolos que serán utilizados para llevar a cabo el diseño de la solución técnica de creación de una red de comunicaciones para una empresa de Contact Center. En el tercer capítulo se expone la solución técnica necesaria para permitir que una empresa de Contact Center preste sus servicios desde distintas ubicaciones distribuidas geográficamente, utilizando dos Data Centers donde se centralizan las aplicaciones de voz y datos. Finalmente, en el cuarto capítulo se presentan las conclusiones obtenidas tras la elaboración de la presente memoria, así como una propuesta de trabajos futuros, que permitirían junto con el proyecto actual, realizar una solución técnica completa incluyendo otras áreas tecnológicas necesarias en una empresa de Contact Center. Todas las ilustraciones y tablas de este proyecto son de elaboración propia a partir de mi experiencia profesional y de la información obtenida en diversos formatos de la bibliografía consultada, excepto en los casos en los que la fuente es mencionada. ABSTRACT This project shows a network solution for a company that provides Contact Center services from different locations geographically distributed, using the Telephone over Internet Protocol (ToIP) technology. The goal of this project is to become a design guide for performing network solutions using current communications equipment developed by the manufacturer Cisco Systems, Inc., firewalls developed by the manufacturer Fortinet and telephone systems developed by Avaya Inc. and Oracle Corporation, due to their great market reputation and their contributions that each one has made in the field of Contact Center. In order to provide interconnection between its different sites, the Contact Center needs to hire the services of a telecommunications’ operator, who will use the VPN MPLS technology, with two diversified access from each Contact Center’s site. The result of this hiring is the advantage of the benefits that a telecommunications operator can offer to its customers, regarding quality of service, availability and geographical expansion. Likewise, Service Level Agreement (SLA) has to be defined to ensure the Contact Center quality communication between their sites. A quality communication is understood as a communication that is capable of being transmitted with minimum values of packet loss and transmission delays, and a speed according to the demand for its voice and data services. As part of the solution, a redundant Internet connection has to be designed to provide access to every Contact Center’s site. The local connectivity solution in each of the Contact Center’s sites has to be designed according to its volume of users and scalability that each one may have. Thereby, the manufacturer Cisco Systems, Inc. offers several options associated with the current equipment. As part of the solution, quality criteria are being defined for the choice of the Data Centers. A Contact Center has connections to/from the client companies that provide network access to teleworkers. This requires along the access and services published on the Internet, needs a security infrastructure. Therefore is been created a solution design that unifies all connections under a single infrastructure, dividing each services in a virtual way. Likewise, is been defined the use of protocols, such as 802.1X, to prevent unauthorized access to the Contact Center’s network. The voice solution chosen is heterogeneous and capable of supporting best-known signaling protocols (SIP and H.323) in order to have maximum flexibility to establish links of Voice over IP (IP Trunk) with suppliers and clients. This can be achieved through the use of SBC and an internal voice infrastructure based on Avaya Inc. The VoIP systems in a Contact Center are the key elements to be able to provide the service; for this reason a redundant solution under virtual environment is been chosen. This solution allows any of the Data Centers to deploy the VoIP system. The solution carried out in this project is mainly based on my own experience acquired during the past seven years in the communications department of a Contact Center company. I have taken into account the main requirements that most companies request nowadays when they hire a Contact Center service. This project is divided into four chapters. The first chapter is an introduction that explains the main business scenarios and technical areas required to provide Contact Center services. The second chapter describes briefly the key technologies and protocols that will be used to carry out the design of the technical solution for the creation of a communications network in a Contact Center company. The third chapter shows a technical solution required that allows a Contact Center company to provide services from across geographically distributed locations, using two Data Centers where data and voice applications are centralized. Lastly, the fourth chapter includes the conclusions gained after making this project, as well as a future projects proposal, which would allow along the current project, to perform a whole technical solution including other necessary technologic areas in a Contact Center company All illustrations and tables of this project have been made by myself from my professional experience and the information obtained in various formats of the bibliography, except in the cases where the source is indicated.

Incorporating global positioning data in real time P2P audio/video streams for mobile devices

We propose an original method to geoposition an audio/video stream with multiple emitters that are at the same time receivers of the mixed signal. The achieved method is suitable for those comes where a list of positions within a designated area is encoded with a degree of precision adjusted to the visualization capabilities; and is also easily extensible to support new requirements. This method extends a previously proposed protocol, without incurring in any performance penalty.

P2P audio/video protocol with global positioning data in real time for mobile devices

In this paper, we propose an original method to geoposition an audio/video stream with multiple emitters that are at the same time receivers of the mixed signal. The obtained method is suitable when a list of positions within a known area is encoded with precision tailored to the visualization capabilities of the target device. Nevertheless, it is easily adaptable to new precision requirements, as well as parameterized data precision. This method extends a previously proposed protocol, without incurring in any performance penalty.

Protocolo multiplataforma no centralizado para comunicaciones multimedia seguras

En este trabajo se propone y desarrolla una topología en k-hipercubos que resuelve los principales inconvenientes asociados a la topología en hipercubo convencional. Los resultados obtenidos son muy prometedores, con aplicaciones tanto en el campo de la voz sobre IP, como en muchos otros campos que precisen de un intercambio de información muchos a muchos. Sobre la topología propuesta se define el protocolo Darkcube, que es una propuesta de protocolo totalmente distribuido basado en el concepto de darknet, posibilitando la realización de conversaciones muchos a muchos incluyendo audio, vídeo, texto y datos de geoposicionamiento, entre otros. También se propone un método de codificación de coordenadas de geoposicionamiento que resulta especialmente eficiente en el aprovechamiento del ancho de banda sobrante en las comunicaciones muchos a muchos que proporciona Darkcube. Durante el desarrollo de este trabajo, se ha implementado el simulador DarkcubeEmu; herramienta que posibilita la obtención de resultados relevantes en términos de la calidad de la comunicación. Finalmente, utilizando como base el protocolo Darkcube, se propone un protocolo de seguridad que traslada un esquema de infraestructura de clave pública a un protocolo totalmente distribuido, como es Darkcube; garantizando, de esta forma, la confidencialidad en las comunicaciones y la legitimidad de la identidad asociada a cada uno de sus miembros.

Digital Classics Outside the Echo-Chamber : Teaching, Knowledge Exchange and Public Engagement /

The international perspectives on these issues are especially valuable in an increasingly connected, but still institutionally and administratively diverse world. The research addressed in several chapters in this volume includes issues around technical standards bodies like EpiDoc and the TEI, engaging with ways these standards are implemented, documented, taught, used in the process of transcribing and annotating texts, and used to generate publications and as the basis for advanced textual or corpus research. Other chapters focus on various aspects of philological research and content creation, including collaborative or community driven efforts, and the issues surrounding editorial oversight, curation, maintenance and sustainability of these resources. Research into the ancient languages and linguistics, in particular Greek, and the language teaching that is a staple of our discipline, are also discussed in several chapters, in particular for ways in which advanced research methods can lead into language technologies and vice versa and ways in which the skills around teaching can be used for public engagement, and vice versa. A common thread through much of the volume is the importance of open access publication or open source development and distribution of texts, materials, tools and standards, both because of the public good provided by such models (circulating materials often already paid for out of the public purse), and the ability to reach non-standard audiences, those who cannot access rich university libraries or afford expensive print volumes. Linked Open Data is another technology that results in wide and free distribution of structured information both within and outside academic circles, and several chapters present academic work that includes ontologies and RDF, either as a direct research output or as essential part of the communication and knowledge representation. Several chapters focus not on the literary and philological side of classics, but on the study of cultural heritage, archaeology, and the material supports on which original textual and artistic material are engraved or otherwise inscribed, addressing both the capture and analysis of artefacts in both 2D and 3D, the representation of data through archaeological standards, and the importance of sharing information and expertise between the several domains both within and without academia that study, record and conserve ancient objects. Almost without exception, the authors reflect on the issues of interdisciplinarity and collaboration, the relationship between their research practice and teaching and/or communication with a wider public, and the importance of the role of the academic researcher in contemporary society and in the context of cutting edge technologies. How research is communicated in a world of instant- access blogging and 140-character micromessaging, and how our expectations of the media affect not only how we publish but how we conduct our research, are questions about which all scholars need to be aware and self-critical.

Statistical study of talk spurts and silence lengths with relation to various languages and speakers for advanced silence suppression in Next Generation Wireless Communication Devices

As wireless network technologies evolve towards an All-IP framework, Next Generation Wireless Communication Devices demand better use of spectral resources by employing advanced techniques of silence suppression. This paper presents an analysis of VoIP call data and compares the statistical results based on observed patterns of talk spurts and silence lengths to those achieved by a modified on-off voice model for silence suppression in wireless networks. As talk spurts and silence lengths are sensitive to varying word lengths, temporal structure and other prosodic aspects of speech, the impact of the use of various languages, dialects and gender of speakers on these results is also assessed.

Modelling data and voice traffic over IP networks using continuous-time Markov models

Common approaches to IP-traffic modelling have featured the use of stochastic models, based on the Markov property, which can be classified into black box and white box models based on the approach used for modelling traffic. White box models, are simple to understand, transparent and have a physical meaning attributed to each of the associated parameters. To exploit this key advantage, this thesis explores the use of simple classic continuous-time Markov models based on a white box approach, to model, not only the network traffic statistics but also the source behaviour with respect to the network and application. The thesis is divided into two parts: The first part focuses on the use of simple Markov and Semi-Markov traffic models, starting from the simplest two-state model moving upwards to n-state models with Poisson and non-Poisson statistics. The thesis then introduces the convenient to use, mathematically derived, Gaussian Markov models which are used to model the measured network IP traffic statistics. As one of the most significant contributions, the thesis establishes the significance of the second-order density statistics as it reveals that, in contrast to first-order density, they carry much more unique information on traffic sources and behaviour. The thesis then exploits the use of Gaussian Markov models to model these unique features and finally shows how the use of simple classic Markov models coupled with use of second-order density statistics provides an excellent tool for capturing maximum traffic detail, which in itself is the essence of good traffic modelling. The second part of the thesis, studies the ON-OFF characteristics of VoIP traffic with reference to accurate measurements of the ON and OFF periods, made from a large multi-lingual database of over 100 hours worth of VoIP call recordings. The impact of the language, prosodic structure and speech rate of the speaker on the statistics of the ON-OFF periods is analysed and relevant conclusions are presented. Finally, an ON-OFF VoIP source model with log-normal transitions is contributed as an ideal candidate to model VoIP traffic and the results of this model are compared with those of previously published work.

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.

Google - Security Testing Tool

Using Google as a security testing tool, basic and advanced search techniques using advanced google search operators. Examples of obtaining control over security cameras, VoIP systems, web servers and collecting valuable information as: Credit card details, cvv codes – only using Google.

Traffic Offered Behaviour Regarding Target QOS Parameters in Network Dimensioning

We consider a model of overall telecommunication network with virtual circuits switching, in stationary state, with Poisson input flow, repeated calls, limited number of homogeneous terminals and 8 types of losses. One of the main problems of network dimensioning/redimensioning is estimation of traffic offered in network because it reflects on finding of necessary number of circuit switching lines on the basis of the consideration of detailed users manners and target Quality of Service (QoS). In this paper we investigate the behaviour of the traffic offered in a network regarding QoS variables: “probability of blocked switching” and “probability of finding B-terminals busy”. Numerical dependencies are shown graphically. A network dimensioning task (NDT) is formulated, solvability of the NDT and the necessary conditions for analytical solution are researched as well. International Journal "Information Technologies and Knowledge" Vol.2 / 2008 174 The received results make the network dimensioning/redimensioning, based on QoS requirements easily, due to clearer understanding of important variables behaviour. The described approach is applicable directly for every (virtual) circuit switching telecommunication system e.g. GSM, PSTN, ISDN and BISDN. For packet - switching networks, at various layers, proposed approach may be used as a comparison basis and when they work in circuit switching mode (e.g. VoIP).

Quality of Service System Approximation in IP Networks

This paper is sponsored by the Ministry of Education and Research of the Republic of Bulgaria in the framework of project No 105 “Multimedia Packet Switching Networks Planning with Quality of Service and Traffic Management”.

Primary and Secondary Empirical Values in Network Redimensioning

A model of an overall telecommunication network with virtual circuits switching, in stationary state, with Bernoulli-Poisson-Pascal (BPP) input flow, repeated calls, limited number of homogeneous terminals and 8 types of losses is considered. One of the main problems of network redimensioning is estimation of the traffic offered in the network because it reflects on finding of necessary number of equivalent switching lines on the basis of the consideration of detailed users behavior and target Quality of Service (QoS). The aim of this paper is to find a new solution of Network Redimensioning Task (NRDT) [4], taking into account the inconvenience of necessary measurements, not considered in the previous research [5]. The results are applicable for redimensioning of every (virtual) circuit switching telecommunication system, both for wireline and wireless systems (GSM, PSTN, ISDN and BISDN). For packet - switching networks proposed approach may be used as a comparison basis and when they work in circuit switching mode (e.g. VoIP).