Architecture for a Session Initiation Protocol Based Presence Client

Mitä on läsnäolo? Tämä työ määrittelee läsnäolon tietyn henkilön, laitteen tai palvelun halukkuudeksi kommunikoida. Nykyään on olemassa lukuisia läsnäolotietoa levittäviä sovelluksia, joista jokainen käyttää erilaista protokollaa tehtävän suorittamiseen. Vasta viime aikoina sovellusten kehittäjät ovat huomanneet tarpeen yhdelle sovellukselle, joka kykenee tukemaan lukuisia läsnäoloprotokollia. Session Initiation Protocol (SIP) voi levittää läsnäolotietoa muiden ominaisuuksiensa lisäksi. Kun muita protokollia käytetään vain reaaliaikaiseen viestintään ja läsnäolotiedon lähetykseen, SIP pystyy moniin muihinkin asioihin. Se on alunperin suunniteltu aloittamaan, muuttamaan ja lopettamaan osapuolien välisiä multimediaistuntoja. Arkkitehtuurin toteutus käyttää kahta Symbian –käyttöjärjestelmän perusominaisuutta: asiakas-palvelin rakennetta ja kontaktitietokantaa. Asiakaspalvelin rakenne erottaa asiakkaan protokollasta tarjoten perustan laajennettavalle usean protokollan arkkitehtuurille ja kontaktitietokanta toimii läsnäolotietojen varastona. Työn tuloksena on Symbianin käyttöjärjestelmässä toimiva läsnäoloasiakas.

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.

Realização de uma arquitectura diffServ com configuração transparente de QOS

Dissertação apresentada na Faculdade de Ciências e Tecnologia da Universidade Nova de Lisboa para a obtenção do grau de Mestre em Engenharia Electrotécnica e de Computadores

Análise de Desempenho e Recomendação para Soluções IVR

Mestrado em Engenharia Informática, Área de Especialização em Tecnologias do Conhecimento e da Decisão

Towards automated test and validation of SIP solutions

IP networks are currently the major communication infrastructure used by an increasing number of applications and heterogeneous services, including voice services. In this context, the Session Initiation Protocol (SIP) is a signaling protocol widely used for controlling multimedia communication sessions such as voice or video calls over IP networks, thus performing vital functions in an extensive set of public and enter- prise solutions. However, the SIP protocol dissemination also entails some challenges, such as the complexity associated with the testing/validation processes of IMS/SIP networks. As a consequence, manual IMS/SIP testing solutions are inherently costly and time consuming tasks, being crucial to develop automated approaches in this specific area. In this perspective, this article presents an experimental approach for automated testing/validation of SIP scenarios in IMS networks. For that purpose, an automation framework is proposed allowing to replicate the configuration of SIP equipment from the pro- duction network and submit such equipment to a battery of tests in the testing network. The proposed solution allows to drastically reduce the test and validation times when compared with traditional manual approaches, also allowing to enhance testing reliability and coverage. The automation framework comprises of some freely available tools which are conveniently integrated with other specific modules implemented within the context of this work. In order to illustrate the advantages of the proposed automated framework, a real case study taken from a PT Inovação customer is presented comparing the time required to perform a manual SIP testing approach with the one time required when using the proposed auto- mated framework. The presented results clearly corroborate the advantages of using the presented framework.

Langattoman Internet-puhelupalvelun tarjoaminen

Langattomalla Internetpuhelupalvelulla tarkoitetaan Internet-puheluiden (Voice over Internet Protocol, VoIP) siirtoa langattoman tiedonsiirtoverkon ylitse. Tälläisia langattomia verkkoja voivat olla esimerkiksi langattomat lähiverkot, WiMAX-verkot tai 450 megahertsin (MHz) taajuusalueella toimivat tiedonsiirtoverkot. VoIP-yhteyden toiminta voidaan jakaa kolmeen eri toiminta-alueeseen: yhteydenmuodostusprotokollaan, äänen koodaukseen sekä siirtotiehen. Yhteydenmuodostusprotokollia ovat esimerkiksi SIP (Session Initiation Protocol) sekä H.323. Yhteydenmuodostusprotokollantehtävänä on muodostaa yhteys käyttäjien välille sekä sopia yhteydessäkäytettävistä ominaisuuksista. Äänen koodauksessa ääni pakataan paketteihin, joita lähetetään siirtotietä pitkin eri käyttäjien välillä. Normaalissa Internetpuheluyhteydessä siirtotienä käytetään langallisia siirtoteitä. Tässä työssä on keskitytty langattomiin siirtoteihin ja niidentuomiin haasteisiin, kuten yhteyden luotettavuuteen ja laatuun, yhteyskapasiteetin riittävyyteen sekä siirtymiseen saman verkon eri tukiasemien sekä eri verkkojen välillä. Työssä rakennettiin yksinkertainen, mutta toimiva langaton Internetpuhelujärjestelmä sekä verrattiin sen ominaisuuksia normaaliin Internetpuhelujärjestelmään. Järjestelmää koekäytettiin oikeassa toimintatilanteessa varsinaisen puhelinjärjestelmän rinnalla tavallisessa toimistoympäristössä. Testaustulosten ja käyttäjäkokemusten perusteella on periaatteessa mahdollista rakentaa yksinkertainen langaton Internetpuhelujärjestelmä ja käyttää sitä puhelupalveluiden tarjoamiseen. Palvelun tarjoaminen vaatii kuitenkin vielä tiettyjen viestintäviraston määräysten täyttämistä ennen tuotantokäyttöön ottamista.

Investigation of interface between Multimedia IP Network and Call Processing Server in 3G IP Telephony Core Network

The thesis presents an overview of third generation of IP telephony. The architecture of 3G IP Telephony and its components are described. The main goal of the thesis is to investigate the interface between the Call Processing Server and Multimedia IP Networks. The interface functionality, proposed protocol stack and a general description are presented in the thesis. To provide useful services, 3G IP Telephony requires a set of control protocols for connection establishment, capabilities exchange and conference control. The Session Initiation Protocol (SIP) and the H.323 are two protocols that meet these needs. In the thesis these two protocols are investigated and compared in terms of Complexity, Extensibility, Scalability, Services, Resource Utilization and Management.

Plan de negocio - Beep

Beep ofrece seis servicios: Beep Móvil, Beep PBX Virtual, Beep SMS Masivo, Beep Callblasting, Beep Email y Beep SIPtrunk. Cada uno de ellos busca llegarle a un mercado objetivo diferente, pero siempre buscando ofrecerle la mejor calidad al consumidor. Para efectos de profundización en los productos que a través del tiempo pueden brindarle un mayor valor agregado al cliente y que son diferenciables de la competencia, este Plan de Negocio solamente se centrará en los dos primeros productos, es decir, en Beep Móvil y Beep PBX Virtual.

Uma arquitetura baseada em SCTP e SIP para suporte a aplicações VoIP móveis e a especificação formal do seu módulo de controle

New versions of SCTP protocol allow the implementation of handover procedures in the transport layer, as well as the supply of a partially reliable communication service. A communication architecture is proposed herein, integrating SCTP with the session initiation protocol, SIP, besides additional protocols. This architecture is intended to handle voice applications over IP networks with mobility requirements. User localization procedures are specified in the application layer as well, using SIP, as an alternative mean to the mechanisms used by traditional protocols, that support mobility in the network layer. The SDL formal specification language is used to specify the operation of a control module, which coordinates the operation of the system component protocols. This formal specification is intended to prevent ambiguities and inconsistencies in the definition of this module, assisting in the correct implementation of the elements of this architecture

Sicurezza di rete, analisi del traffico e monitoraggio.

Il lavoro è stato suddiviso in tre macro-aree. Una prima riguardante un'analisi teorica di come funzionano le intrusioni, di quali software vengono utilizzati per compierle, e di come proteggersi (usando i dispositivi che in termine generico si possono riconoscere come i firewall). Una seconda macro-area che analizza un'intrusione avvenuta dall'esterno verso dei server sensibili di una rete LAN. Questa analisi viene condotta sui file catturati dalle due interfacce di rete configurate in modalità promiscua su una sonda presente nella LAN. Le interfacce sono due per potersi interfacciare a due segmenti di LAN aventi due maschere di sotto-rete differenti. L'attacco viene analizzato mediante vari software. Si può infatti definire una terza parte del lavoro, la parte dove vengono analizzati i file catturati dalle due interfacce con i software che prima si occupano di analizzare i dati di contenuto completo, come Wireshark, poi dei software che si occupano di analizzare i dati di sessione che sono stati trattati con Argus, e infine i dati di tipo statistico che sono stati trattati con Ntop. Il penultimo capitolo, quello prima delle conclusioni, invece tratta l'installazione di Nagios, e la sua configurazione per il monitoraggio attraverso plugin dello spazio di disco rimanente su una macchina agent remota, e sui servizi MySql e DNS. Ovviamente Nagios può essere configurato per monitorare ogni tipo di servizio offerto sulla rete.

Articulated Narrowcasting for Privacy and Awareness in Multimedia Conferencing Systems and Design for Implementation Within a SIP Framework

This article proposes a new focus of research for multimedia conferencing systems which allows a participant to flexibly select another participant or a group for media transmission. For example, in a traditional conference system, participants voices might by default be shared with all others, but one might want to select a subset of the conference members to send his/her media to or receive media from. We review the concept of narrowcasting, a model for limiting such information streams in a multimedia conference, and describe a design to use existing standard protocols (SIP and SDP) for controlling fine-grained narrowcasting sessions.

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.

Applying a disciplined approach to the development of a context-aware communication application

Pervasive computing applications must be engineered to provide unprecedented levels of flexibility in order to reconfigure and adapt in response to changes in computing resources and user requirements. To meet these challenges, appropriate software engineering abstractions and infrastructure are required as a platform on which to build adaptive applications. In this paper, we demonstrate the use of a disciplined, model-based approach to engineer a context-aware Session Initiation Protocol (SIP) based communication application. This disciplined approach builds on our previously developed conceptual models and infrastructural components, which enable the description, acquisition, management and exploitation of arbitrary types of context and user preference information to enable adaptation to context changes

An approach for design and analysis of User-centric Communication Middleware

Today, the development of domain-specific communication applications is both time-consuming and error-prone because the low-level communication services provided by the existing systems and networks are primitive and often heterogeneous. Multimedia communication applications are typically built on top of low-level network abstractions such as TCP/UDP socket, SIP (Session Initiation Protocol) and RTP (Real-time Transport Protocol) APIs. The User-centric Communication Middleware (UCM) is proposed to encapsulate the networking complexity and heterogeneity of basic multimedia and multi-party communication for upper-layer communication applications. And UCM provides a unified user-centric communication service to diverse communication applications ranging from a simple phone call and video conferencing to specialized communication applications like disaster management and telemedicine. It makes it easier to the development of domain-specific communication applications. The UCM abstraction and API is proposed to achieve these goals. The dissertation also tries to integrate the formal method into UCM development process. The formal model is created for UCM using SAM methodology. Some design errors are found during model creation because the formal method forces to give the precise description of UCM. By using the SAM tool, formal UCM model is translated to Promela formula model. In the dissertation, some system properties are defined as temporal logic formulas. These temporal logic formulas are manually translated to promela formulas which are individually integrated with promela formula model of UCM and verified using SPIN tool. Formal analysis used here helps verify the system properties (for example multiparty multimedia protocol) and dig out the bugs of systems.