Design and validation of a methodology for distributed relay service for NAT traversal in a peer-to-peer VoIP network

Voice-over-IP (VoIP) practices are widely diffused. The traditional and mostly deployed architecture is based on the IETF SIP protocol: User Agents connect to centralized servers (usually called SIP Proxies), which provide, among other features, user location service and call routing. On another side, the peer-to-peer paradigm has proven to be very scalable and have been widely accepted by the Internet community. This graduation thesis is going firstly to investigate the current protocols for doing VoIP and in particular the Session Initiation Protocol. Then peer-to-peer overlays are examined, devoting particular care to how integration with SIP can be made. Afterwards, the focus will move on Network Address Translation (NAT). NAT is largely employed in SOHO networks as well as in big networks installations, because it reduces the need of public IP addresses and is believed to increase network security. However it requires many protocols to be modified to work correctly. NAT traversal techniques will be analyzed, along with the issues that NAT creates for SIP and P2P protocols. In order to perform NAT traversal, a public rendez-vous point is needed. A methodology to build a distributed relay service over a pure peer-to-peer network will be proposed and validated by means of statistical analysis and simulation

Managing ClientInitiated Connections

The Session Initiation Protocol (SIP) allows proxy servers to initiate TCP connections or to send asynchronous UDP datagrams to User Agents in order to deliver requests. However, in a large number of real deployments, many practical considerations, such as the existence of firewalls and Network Address Translators (NATs) or the use of TLS with server-provided certificates, prevent servers from connecting to User Agents in this way. This specification defines behaviors for User Agents, registrars, and proxy servers that allow requests to be delivered on existing connections established by the User Agent. It also defines keep-alive behaviors needed to keep NAT bindings open and specifies the usage of multiple connections from the User Agent to its registrar. Status of This Memo This document specifies an Internet standards track protocol for the Internet community, and requests discussion and suggestions for improvements. Please refer to the current edition of the "Internet Official Protocol Standards " (STD 1) for the standardization state and status of this protocol. Distribution of this memo is unlimited. Copyright Notice Copyright (c) 2009 IETF Trust and the persons identified as th

A mobile toolkit and customised location server for the creation of cross-referencing location-based services

Although there are several Software Development kits and Application Programming Interfaces for client-side location-based services development, they mostly involve the creation of self-referencing location-based services. Self-referencing location-based services include services such as geocoding, reverse geocoding, route management and navigation which focus on satisfying the location-based requirements of a single mobile device. There is a lack of open-source Software Development Kits for the development of client-side location-based services that are cross-referencing. Cross-referencing location-based services are designed for the sharing of location information amongst different entities on a given network. This project was undertaken to assemble, through incremental prototyping, a client-side Java Micro Edition location-based services Software Development Kit and a Mobicents location server to aid mobile network operators and developers alike in the quick creation of the transport and privacy protection of cross-referencing location-based applications on Session Initiation Protocol bearer networks. The privacy of the location information is protected using geolocation policies. Developers do not need to have an understanding of Session Initiation Protocol event signaling specifications or of the XML Configuration Access Protocol to use the tools that we put together. The developed tools are later consolidated using two sample applications, the friend-finder and child-tracker services. Developer guidelines are also provided, to aid in using the provided tools

CIAO: A Component Model and its OSGi Framework for Dynamically Adaptable Telephony Applications

International audienceIn recent years, thanks to new IP protocols like SIP, telephony applications and services have evolved to other and combine a variety of communication forms including presence status, instant messaging and videoconference. As a result, advanced telephony applications now consist of distributed entities that are involved into multiple heterogeneous, stateful and long-running interactions (sessions). This evolution complicated significantly applications development and calls for more effective solutions. In this paper, we explore the adoption of components for addressing this issue, focusing specifically on the management and coordination of the numerous and various sessions occurring in such applications. The paper presents CIAO, a domain-specific and hierarchical component model for SIP applications. CIAO combines three kinds of component that are Actor, SessionPart and Role and manage them dynamically in accordance with real SIP sessions. By using these features, we are able to break the complexity of SIP entities and provide flexibility for their development. CIAO is implemented above OSGi to experiment the building of concrete SIP applications and enable their dynamic adaptation

Study, Analysis and Modeling of IP Multimedia systems on next generation networks providing mobile and fixed multimedia services

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Estudio de la movilidad en redes de siguiente generación

El continuo avance de las redes de telecomunicaciones nos proporciona cada vez más facilidades en todos los ámbitos de nuestra vida. En este caso, nos hemos centrado en el estudio de la movilidad en Redes de Siguiente Generación. Una parte del presente proyecto se ha realizado en colaboración con Deutsche Telekom AG, durante una estancia de seis meses trabajando como colaboradora en sus laboratorios con emplazamiento en Berlín. El principal objetivo de este proyecto ha sido realizar un estudio sobre los diferentes estándares y tecnologías que facilitan la movilidad en Redes de Siguiente Generación. Por ello, en la primera parte se han estudiado los diferentes grupos de trabajo centrados en este aspecto, así como se ha recabado información sobre productos y soluciones disponibles en el mercado, para obtener una visión global de la situación actual. Como se puede comprobar más adelante, esta primera parte es la más extensa de todo el documento. Esto se debe a que es, probablemente, la parte más importante del trabajo, ya que contiene el estudio de los mecanismos que más tarde nos servirán para dar una solución teórica a los distintos escenarios que se plantean. En la segunda parte del proyecto, nos hemos centrado en desarrollar varios escenarios de interés en sistemas de Redes de Siguiente Generación y aportar, de forma posterior, posibles soluciones teóricas. Para finalizar, se han expuesto las conclusiones extraídas como resultado del trabajo y los aspectos que se podrán tratar sobre el mismo en un futuro próximo.Ingeniería de Telecomunicació

Sip-rtsp Convergence: Rtsp-c

Tez (Yüksek Lisans) -- İstanbul Teknik Üniversitesi, Fen Bilimleri Enstitüsü, 2008Thesis (M.Sc.) -- İstanbul Technical University, Institute of Science and Technology, 2008Bu çalışma ile SIP protokolü iletimde kullanılarak ve RTSP protokolü yetenekleri SIP mesaj gövdesine yerleştirilerek VOIP ağlarında yeni bir medya kontrol modeli öne sürülmüştür. RTSP-C olarak isimlendirilmiş olan bu yeni yakınlaşma modeli sadece SIP ve RTSP protokollerinin bir arada çalışmasını garantilemekle kalmamakta; aynı zamanda medya kontrol isteklerinin asıllanması ve oturum sunum bilgisi (SDP) alış verişindeki bazı açık noktalara çözüm getirmektedir. Bu yeni model RTSP protokolünün NAT geçirimie ait yöntemlere gereksinimini ortadan kaldırmakla beraber, SIP protokolünün NAT geçirim yöntemleri geçerliliğini korumaktadır. Bu modelin sağladığı asıl gelişme medya kontrolü bilgisi ve durum bilgisini SIP protokolüne açık hale getirmesidir. Bu sayede bu model medya yayınına dayalı yeni SIP servislerinin geliştirilmesine olanak sağlamaktadır. Bu proje kapsamında ortaya koyulan yeni modeli örneklendirmek amacıyla bir İsteğe Bağlı Görüntü Yayını (VoD) sistemi geliştirilmiştir. Bu uygulama ile RTSP-C yakınlaştırma modelinin çalışabilirliği doğrolanmıştır. Sonuçlar literatürdeki diğer örnekler ile karşılaştırıldığında modelin daha onceden belirlenen sorunlara uygun çözümleri sağladığı görülmüştür.In this study, using Session Initiation Protocol (SIP) as transport and placing Real Time Streaming Protocol (RTSP) capabilities in the SIP message body, a media control model has been introduced for Voice Over IP (VOIP) networks. This new convergence model does not only guarantee the interoperability of SIP and RTSP protocols but also resolve some open points on media control request authentication and session presentation (SDP) exchange. This new model is also valid for NAT Traversal methods applicable to SIP while it lifts the necessity of NAT Traversal methods for RTSP. The major advancement this model provides is: it makes the media control method/state information available to SIP. By doing that, this model enables the development of new streaming based SIP services. In this project content a Video on Demand (VoD) system is developped to instantiate the new convergence model. The implementation validated the operability of RTSP-C convergence model. The comparison of the results with other models on literature showed that the model provided adequate solutions on the pre-determined problems.Yüksek LisansM.Sc

Convergence du web et des services de communication

Les services de communication, du courrier postal à la téléphonie, en passant par la voix et la vidéo sur IP (Internet Protocol), la messagerie électronique, les salons de discussion sur Internet, les visioconférences ou les télécommunications immersives ont évolué au fil du temps. Un système de communication voix-vidéo sur IP est réalisé grâce à deux couches architecturales fondamentales : la couche de signalisation et la couche média. Le protocole de signalisation est utilisé pour créer, modifier et terminer des sessions multimédias entre des participants. La couche de signalisation est divisée en deux sous-couches - la couche de service et celle de contrôle - selon la spécification de l IP Multimedia Subsystem (IMS). Deux systèmes de communication largement utilisés sont l IMS et SIP Pair-à- Pair (P2P SIP). Les fournisseurs de services, qui se comportent en tant qu intermédiaires entre appelants et appelés, implémentent les systèmes de communication, contrôlant strictement la couche signalisation. Or ces fournisseurs de services ne prennent pas en compte la diversité des utilisateurs. Cette thèse identifie trois barrières technologiques dans les systèmes de communication actuels et plus précisément concernant la couche de signalisation. I. Un manque d ouverture et de flexibilité dans la couche de signalisation pour les utilisateurs. II. Un développement difficile des services basés sur le réseau et les sessions. III. Une complexification du la couche de signalisation lors d un très grand nombre d appels. Ces barrières technologiques gênent l innovation des utilisateurs avec ces services de communication. Basé sur les barrières technologiques listées cidessus, le but initial de cette thèse est de définir un concept et une architecture de système de communication dans lequel chaque individu devient un fournisseur de service. Le concept, "My Own Communication Service Provider" (MOCSP) et le système MOCSP sont proposés, accompagné d un diagramme de séquence. Ensuite, la thèse fournit une analyse qui compare le système MOCSP avec les systèmes de communication existants en termes d ouverture et de flexibilité. La seconde partie de la thèse présente des solutions pour les services basés sur le réseau ou les sessions, mettant en avant le système MOCSP proposé. Deux services innovants, user mobility et partial session transfer/retrieval (PSTR) sont pris comme exemples de services basés sur le réseau ou les sessions. Les services basés sur un réseau ou des sessions interagissent avec une session ou sont exécutés dans une session. Dans les deux cas, une seule entité fonctionnelle entre l appelant et l appelé déclenche le flux multimédia pendant l initialisation de l appel et/ou en cours de communication. De plus, la coopération entre le contrôle d appel réseau et les différents pairs est facilement réalisé. La dernière partie de la thèse est dédiée à l extension de MOCSP en cas de forte densité d appels, elle inclut une analyse comparative. Cette analyse dépend de quatre facteurs - limite de passage à l échelle, niveau de complexité, ressources de calcul requises et délais d établissement de session - qui sont considérés pour évaluer le passage à l échelle de la couche de signalisation. L analyse comparative montre clairement que la solution basée sur MOCSP est simple et améliore l usage effectif des ressources de calcul par rapport aux systèmes de communication traditionnelsDifferent communication services from delivery of written letters to telephones, voice/video over Internet Protocol(IP), email, Internet chat rooms, and video/audio conferences, immersive communications have evolved over time. A communication system of voice/video over IP is the realization of a two fundamental layered architecture, signaling layer and media layer. The signaling protocol is used to create, modify, and terminate media sessions between participants. The signaling layer is further divided into two layers, service layer and service control layer, in the IP Multimedia Subsystem (IMS) specification. Two widely used communication systems are IMS, and Peer-to-Peer Session Initiation Protocol (P2P SIP). Service providers, who behave as brokers between callers and callees, implement communication systems, heavily controlling the signaling layer. These providers do not take the diversity aspect of end users into account. This dissertation identifies three technical barriers in the current communication systems especially in the signaling layer. Those are: I. lack of openness and flexibility in the signaling layer for end users. II. difficulty of development of network-based, session-based services. III. the signaling layer becomes complex during the high call rate. These technical barriers hinder the end-user innovation with communication services. Based on the above listed technical barriers, the first part of this thesis defines a concept and architecture for a communication system in which an individual user becomes the service provider. The concept, My Own Communication Service Provider (MOCSP) and MOCSP system is proposed and followed by a call flow. Later, this thesis provides an analysis that compares the MOCSP system with existing communication systems in terms of openness and flexibility. The second part of this thesis presents solutions for network-based, session based services, leveraging the proposed MOCSP system. Two innovative services, user mobility and partial session transfer/retrieval are considered as examples for network-based, session-based services. The network-based, sessionbased services interwork with a session or are executed within a session. In both cases, a single functional entity between caller and callee consistently enables the media flow during the call initiation and/or mid-call. In addition, the cooperation of network call control and end-points is easily achieved. The last part of the thesis is devoted to extending the MOCSP for a high call rate and includes a preliminary comparative analysis. This analysis depends on four factors - scalability limit, complexity level, needed computing resources and session setup latency - that are considered to specify the scalability of the signaling layer. The preliminary analysis clearly shows that the MOCSP based solution is simple and has potential for improving the effective usage of computing resources over the traditional communication systemsEVRY-INT (912282302) / SudocSudocFranceF

Tools and Technologies for Enabling Characterisation in Synthetic Biology

Synthetic Biology represents a movement to utilise biological organisms for novel applications through the use of rigorous engineering principles. These principles rely on a solid and well versed understanding of the underlying biological components and functions (relevant to the application). In order to achieve this understanding, reliable behavioural and contextual information is required (more commonly known as characterisation data). Focussing on lowering the barrier of entry for current research facilities to regularly and easily perform characterisation assays will directly improve the communal knowledge base for Synthetic Biology and enable the further application of rational engineering principles. Whilst characterisation remains a fundamental principle for Synthetic Biology research, the high time costs, subjective measurement protocols, and ambiguous data analysis specifications, deter regular performance of characterisation assays. Vitally, this prevents the valid application of many of the key Synthetic Biology processes that have been derived to improve research yield (with regards to solving application problems) and directly prevent the intended goal of addressing the ad hoc nature of modern research from being realised. Designing new technologies and tools to facilitate rapid ‘hands off’ characterisation assays for research facilities will improve the uptake of characterisation within the research pipeline. To achieve this two core problem areas have been identified that limit current characterisation attempts in conventional research. Therefore, it was the primary aim of this investigation to overcome these two core problems to promote regular characterisation. The first issue identified as preventing the regular use of characterisation assays was the user-intensive methodologies and technologies available to researchers. There is currently no standardised characterisation equipment for assaying samples and the methodologies are heavily dependent on the researcher and their application for successful and complete characterisation. This study proposed a novel high throughput solution to the characterisation problem that was capable of low cost, concurrent, and rapid characterisation of simple biological DNA elements. By combining in vitro transcription-translation with microfluidics a potent solution to the characterisation problem was proposed. By utilising a completely in vitro approach along with excellent control abilities of microfluidic technologies, a prototype platform for high throughput characterisation was developed. The second issue identified was the lack of flexible, versatile software designed specifically for the data handling needs that are quickly arising within the characterisation speciality. The lack of general solutions in this area is problematic because of the increasing amount of data that is both required and generated for the characterisation output to be considered as rigorous and of value. To alleviate this issue a novel framework for laboratory data handling was developed that employs a plugin strategy for data submission and analysis. Employing a plugin strategy improves the shelf life of data handling software by allowing it to grow with the needs of the speciality. Another advantage to this strategy is the increased ability for well documented processing and analysis standards to arise that are available for all researchers. Finally, the software provided a powerful and flexible data storage schema that allowed all currently conceivable characterisation data types to be stored in a well-documented manner. The two solutions identified within this study increase the amount of enabling tools and technologies available to researchers within Synthetic Biology, which in turn will increase the uptake of regular characterisation. Consequently, this will potentially improve the lateral transfer of knowledge between research projects and reduce the need to perform ad hoc experiments to investigate facets of the fundamental biological components being utilised.Open Acces

On the development of Voice over IP

This record of study documents the experience acquired during my internship at Sonus Networks, Inc. for the Doctor of Engineering Program. In this record of study, I have surveyed and analyzed the current standardization status of Voice over Internet Protocol (VoIP) security and proposed an Internet draft on secure retargeting and response identity. The draft provides a simple and comprehensive solution to the response identity, call recipient identity and intermediate server retargeting problems in the Session Initiation Protocol (SIP) call setup process. To support product line development and enable product evolution in the quickly growing VoIP market, I have proposed a generic development framework for SIP application servers. The common and open architecture of the framework supports multiple products development and facilitates integration of new service modules. The systematical reuse of proven software design and implementation enables companies to reduce the development cost and shorten the time-to-market. As the development and diffusion of VoIP can never be isolated from the social sphere, I have investigated the current status, influence and interaction of three most important factors: standardization, market forces and government regulation on the development and diffusion of VoIP. The worldwide deregulation and market privatization have caused the transition of the standards development model. This transition in turn influences the market diffusion. Other than standardization, market forces including customer needs, the revenue pressure on carriers and vendors, competitive and economic environment, social culture and regulation uncertainties create both threats and opportunities. I have examined market drivers and obstacles in the current VoIP adoption stage, analyzed current VoIP market players and their strategies, and predicted the direction of VoIP business. The regulation creates the macro environment in which VoIP develops and diffuses. I have explored modern telecommunications regulation principles based on which government makes decisions on most current issues, including 911 support, mergers and acquisitions, interconnection obligation and leasing rights, rate structure and universal service fees