A distributed end-to-end overload control mechanism for networks of SIP servers.

The Session Initiation Protocol (SIP) is an application-layer control protocol standardized by the IETF for creating, modifying and terminating multimedia sessions. With the increasing use of SIP in large deployments, the current SIP design cannot handle overload effectively, which may cause SIP networks to suffer from congestion collapse under heavy offered load. This paper introduces a distributed end-to-end overload control (DEOC) mechanism, which is deployed at the edge servers of SIP networks and is easy to implement. By applying overload control closest to the source of traf?c, DEOC can keep high throughput for SIP networks even when the offered load exceeds the capacity of the network. Besides, it responds quickly to the sudden variations of the offered load and achieves good fairness. Theoretic analysis and extensive simulations verify that DEOC is effective in controlling overload of SIP networks.

Probe-based end-to-end overload control for networks of SIP servers

The Session Initiation Protocol (SIP) has been adopted by the IETF as the control protocol for creating, modifying and terminating multimedia sessions. Overload occurs in SIP networks when SIP servers have insufficient resources to handle received messages. Under overload, SIP networks may suffer from congestion collapse due to current ineffective SIP overload control mechanisms. This paper introduces a probe-based end-to-end overload control (PEOC) mechanism, which is deployed at the edge servers of SIP networks and is easy to implement. By probing the SIP network with SIP messages, PEOC estimates the network load and controls the traffic admitted to the network according to the estimated load. Theoretic analysis and extensive simulations verify that PEOC can keep high throughput for SIP networks even when the offered load exceeds the capacity of the network. Besides, it can respond quickly to the sudden variations of the offered load and achieve good fairness.

Contribution To Signalling Of 3d Video Streams In Communication Systems Using The Session Initiation Protocol

Las tecnologías de vídeo en 3D han estado al alza en los últimos años, con abundantes avances en investigación unidos a una adopción generalizada por parte de la industria del cine, y una importancia creciente en la electrónica de consumo. Relacionado con esto, está el concepto de vídeo multivista, que abarca el vídeo 3D, y puede definirse como un flujo de vídeo compuesto de dos o más vistas. El vídeo multivista permite prestaciones avanzadas de vídeo, como el vídeo estereoscópico, el “free viewpoint video”, contacto visual mejorado mediante vistas virtuales, o entornos virtuales compartidos. El propósito de esta tesis es salvar un obstáculo considerable de cara al uso de vídeo multivista en sistemas de comunicación: la falta de soporte para esta tecnología por parte de los protocolos de señalización existentes, que hace imposible configurar una sesión con vídeo multivista mediante mecanismos estándar. Así pues, nuestro principal objetivo es la extensión del Protocolo de Inicio de Sesión (SIP) para soportar la negociación de sesiones multimedia con flujos de vídeo multivista. Nuestro trabajo se puede resumir en tres contribuciones principales. En primer lugar, hemos definido una extensión de señalización para configurar sesiones SIP con vídeo 3D. Esta extensión modifica el Protocolo de Descripción de Sesión (SDP) para introducir un nuevo atributo de nivel de medios, y un nuevo tipo de dependencia de descodificación, que contribuyen a describir los formatos de vídeo 3D que pueden emplearse en una sesión, así como la relación entre los flujos de vídeo que componen un flujo de vídeo 3D. La segunda contribución consiste en una extensión a SIP para manejar la señalización de videoconferencias con flujos de vídeo multivista. Se definen dos nuevos paquetes de eventos SIP para describir las capacidades y topología de los terminales de conferencia, por un lado, y la configuración espacial y mapeo de flujos de una conferencia, por el otro. También se describe un mecanismo para integrar el intercambio de esta información en el proceso de inicio de una conferencia SIP. Como tercera y última contribución, introducimos el concepto de espacio virtual de una conferencia, o un sistema de coordenadas que incluye todos los objetos relevantes de la conferencia (como dispositivos de captura, pantallas, y usuarios). Explicamos cómo el espacio virtual se relaciona con prestaciones de conferencia como el contacto visual, la escala de vídeo y la fidelidad espacial, y proporcionamos reglas para determinar las prestaciones de una conferencia a partir del análisis de su espacio virtual, y para generar espacios virtuales durante la configuración de conferencias.

Videoconference capacity Leasing on Hybrid Clouds.

This paper proposes a new methodology focused on implementing cost effective architectures on Cloud Computing systems. With this methodology the paper presents some disadvantages of systems that are based on single Cloud architectures and gives some advices for taking into account in the development of hybrid systems. The work also includes a validation of these ideas implemented in a complete videoconference service developed with our research group. This service allows a great number of users per conference, multiple simultaneous conferences, different client software (requiring transcodification of audio and video flows) and provides a service like automatic recording. Furthermore it offers different kinds of connectivity including SIP clients and a client based on Web 2.0. The ideas proposed in this article are intended to be a useful resource for any researcher or developer who wants to implement cost effective systems on several Clouds

Diffusion Tensor Imaging in Alzheimer's disease

Attentional control and Information processing speed are central concepts in cognitive psychology and neuropsychology. Functional neuroimaging and neuropsychological assessment have depicted theoretical models considering attention as a complex and non-unitary process. One of its component processes, Attentional set-shifting ability, is commonly assessed using the Trail Making Test (TMT). Performance in the TMT decreases with increasing age in adults, Mild Cognitive Impairment (MCI) and Alzheimer’s Disease (AD). Besides, speed of information processing (SIP) seems to modulate attentional performance. While neural correlates of attentional control have been widely studied, there are few evidences about the neural substrates of SIP in these groups of patients. Different authors have suggested that it could be a property of cerebral white matter, thus, deterioration of the white matter tracts that connect brain regions related to set-shifting may underlie the age-related, MCI and AD decrease in performance. The aim of this study was to study the anatomical dissociation of attentional and speed mechanisms. Diffusion tensor imaging (DTI) provides a unique insight into the cellular integrity of the brain, offering an in vivo view into the microarchitecture of cerebral white matter. At the same time, the study of ageing, characterized by white matter decline, provides the opportunity to study the anatomical substrates speeded or slowed information processing. We hypothesized that FA values would be inversely correlated with time to completion on Parts A and B of the TMT, but not the derived scores B/A and B-A.

Supporting eradication of maternal child malnutrition in developing countries: a case from Nicaragua scenario.

World Health Organization actively stresses the importance of health, nutrition and well-being of the mother to foster children development. This issue is critical in the rural areas of developing countries where monitoring of health status of children is hardly performed since population suffers from a lack of access to health care. The aim of this research is to design, implement and deploy an e-health information and communication system to support health care in 26 rural communities of Cusmapa, Nicaragua. The final solution consists of an hybrid WiMAX/WiFi architecture that provides good quality communications through VoIP taking advantage of low cost WiFi mobile devices. Thus, a WiMAX base station was installed in the health center to provide a radio link with the rural health post "El Carrizo" sited 7,4 km. in line of sight. This service makes possible personal broadband voice and data communication facilities with the health center based on WiFi enabled devices such as laptops and cellular phones without communications cost. A free software PBX was installed at "San José de Cusmapa" health care site to enable communications for physicians, nurses and a technician through mobile telephones with IEEE 802.11 b/g protocol and SIP provided by the project. Additionally, the rural health post staff (midwives, brigade) received two mobile phones with these same features. In a complementary way, the deployed health information system is ready to analyze the distribution of maternal-child population at risk and the distribution of diseases on a geographical baseline. The system works with four information layers: fertile women, children, people with disabilities and diseases. Thus, authorized staff can obtain reports about prenatal monitoring tasks, status of the communities, malnutrition, and immunization control. Data need to be updated by health care staff in order to timely detect the source of problem to implement measures addressed to alleviate and improve health status population permanently. Ongoing research is focused on a mobile platform that collects and automatically updates in the information system, the height and weight of the children locally gathered in the remote communities. This research is being granted by the program Millennium Rural Communities of the Technical University of Madrid.

A component library to improve the reusability in the development of converged services

The evolution of communications networks to Next Generation Networks (NGN) has encouraged the development of new services. Nowadays, several technologies are being integrated into telecommunications services in order to provide new functionalities, resulting in what are known as converged services. The objective is to adapt the behavior of the services to the necessities of different users, generating customized services. Some of the main technologies involved in their development are those related to the Web. But due to this type of services implies the combination of different technologies, their development is a very complex process that has to be improved to reduce the time and cost required, with the aim of promoting the success of such services. This paper proposes to apply software reuse through the utilization of a component library and presents one focused on ECharts for SIP Servlets (E4SS). It is a framework, based on the SIP Servlet specification, which uses finite state machines for the definition of converged communications services. Also, to promote the use of the library, a methodology is proposed in order to facilitate the integration between the library operations and the software development cycle.

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.