Signaling for Internet Telephony

Internet telephony must offer the standard telephony services.However, the transition to Internet-based telephony services also provides an opportunity to create new services more rapidly and with lower complexity than in the existing public switched telephone network(PSTN). The Session Initiation Protocol (SIP) is a signaling protocol that creates, modifies and terminates associations between Internet end systems, including conferences and point-to-point calls. SIP supports unicast, mesh and multicast conferences, as well as combinations of these modes. SIP implements services such as call forwarding and transfer, placing calls on hold, camp-on and call queueing by a small set of call handling primitives. SIP implementations can re-use parts of other Internet service protocols such as HTTP and the Real-Time Stream Protocol (RTSP). In this paper, we describe SIP, and show how its basic primitives can be used to construct a wide range of telephony services

Designing and Implementation of SIP-ALG with State Recovery Signaling

SIP (Session Initiation Protocol)による通信では，シグナリングがエンティティの主要な動作を決定する．そのため，シグナリングがないままに通信途中でエラーが発生すると，ユーザ端末を含めパス上に存在するサーバは，適切に初期状態へ移行することができない．したがって，これらは復旧までの間，様々なリソースを浪費することになる．従来までこの問題に対する解決策は，エンティティに対して独自にタイマを持たせることでのみ実現されていた．通信セッションに異常が発生しても，エンティティはリソースを解放できないことがあった．本論文では，この問題を解決をするために，状態正常化シグナリングに基づくSIP-ALGを提案する．提案方式に基づいて実装を行い，その動作を検証する．またエンティティに対して定義されているオートマトンを利用して，状態正常化シグナリングの有効性について検討する．修士論

Network convergence and QoS for future multimedia services in the VISION project

The emerging use of real-time 3D-based multimedia applications imposes strict quality of service (QoS) requirements on both access and core networks. These requirements and their impact to provide end-to-end 3D videoconferencing services have been studied within the Spanish-funded VISION project, where different scenarios were implemented showing an agile stereoscopic video call that might be offered to the general public in the near future. In view of the requirements, we designed an integrated access and core converged network architecture which provides the requested QoS to end-to-end IP sessions. Novel functional blocks are proposed to control core optical networks, the functionality of the standard ones is redefined, and the signaling improved to better meet the requirements of future multimedia services. An experimental test-bed to assess the feasibility of the solution was also deployed. In such test-bed, set-up and release of end-to-end sessions meeting specific QoS requirements are shown and the impact of QoS degradation in terms of the user perceived quality degradation is quantified. In addition, scalability results show that the proposed signaling architecture is able to cope with large number of requests introducing almost negligible delay

An interoperable and secure architecture for internet-scale decentralized personal communication

Interpersonal network communications, including Voice over IP (VoIP) and Instant Messaging (IM), are increasingly popular communications tools. However, systems to date have generally adopted a client-server model, requiring complex centralized infrastructure, or have not adhered to any VoIP or IM standard. Many deployment scenarios either require no central equipment, or due to unique properties of the deployment, are limited or rendered unattractive by central servers. to address these scenarios, we present a solution based on the Session Initiation Protocol (SIP) standard, utilizing a decentralized Peer-to-Peer (P2P) mechanism to distribute data. Our new approach, P2PSIP, enables users to communicate with minimal or no centralized servers, while providing secure, real-time, authenticated communications comparable in security and performance to centralized solutions.;We present two complete protocol descriptions and system designs. The first, the SOSIMPLE/dSIP protocol, is a P2P-over-SIP solution, utilizing SIP both for the transport of P2P messages and personal communications, yielding an interoperable, single-stack solution for P2P communications. The RELOAD protocol is a binary P2P protocol, designed for use in a SIP-using-P2P architecture where an existing SIP application is modified to use an additional, binary RELOAD stack to distribute user information without need for a central server.;To meet the unique security needs of a fully decentralized communications system, we propose an enrollment-time certificate authority model that provides asserted identity and strong P2P and user-level security. In this model, a centralized server is contacted only at enrollment time. No run-time connections to the servers are required.;Additionally, we show that traditional P2P message routing mechanisms are inappropriate for P2PSIP. The existing mechanisms are generally optimized for file sharing and neglect critical practical elements of the open Internet --- namely link-level security and asymmetric connectivity caused by Network Address Translators (NATs). In response to these shortcomings, we introduce a new message routing paradigm, Adaptive Routing (AR), and using both analytical models and simulation show that AR significantly improves message routing performance for P2PSIP systems.;Our work has led to the creation of a new research topic within the P2P and interpersonal communications communities, P2PSIP. Our seminal publications have provided the impetus for subsequent P2PSIP publications, for the listing of P2PSIP as a topic in conference calls for papers, and for the formation of a new working group in the Internet Engineering Task Force (IETF), directed to develop an open Internet standard for P2PSIP

A service-enabling framework for the session initiation protocol (SIP)

In this dissertation, we propose a framework to provide multimedia communication services. Our proposed framework is based on SIP (Session Initiation Protocol) and has four fundamental properties: it is available, secure, high performing, and oriented to innovations. The framework is not an architecture with a rigid structure. Instead, the framework is a toolkit made up of a set of tools that can be combined in different ways. The combination of these tools provides applications and services with functionality needed to implement a wide variety of multimedia communication services. Applications and services built on top of the framework use different tools within the toolkit in order to provide their desired overall functionality. The functionality provided by the framework includes a number of primitives to be used by applications and services. These primitives mostly relate to multiparty communications and include floor control. The framework also offers support functions that relate to PSTN (Public Switched Telephony Network) interworking, policy control, and consent-based communications. Additionally, the framework contains functions that relate to signalling transport, multihoming, mobility, security, and NAT (Network Address Translation) traversal. The framework also allows building overlay networks when a SIP network infrastructure is not available. In order to test and refine the ideas presented in this dissertation, we have implemented most of them in proof-of-concept prototypes. We have used experiments and simulations to validate our assumptions and obtain new insights

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