Optimizing IETF multimedia signaling protocols and architectures in 3GPP networks : an evolutionary approach

Signaling in Next Generation IP-based networks heavily relies in the family of multimedia signaling protocols defined by IETF. Two of these signaling protocols are RTSP and SIP, which are text-based, client-server, request-response signaling protocols aimed at enabling multimedia sessions over IP networks. RTSP was conceived to set up streaming sessions from a Content / Streaming Server to a Streaming Client, while SIP was conceived to set up media (e.g.: voice, video, chat, file sharing, …) sessions among users. However, their scope has evolved and expanded over time to cover virtually any type of content and media session. As mobile networks progressively evolved towards an IP-only (All-IP) concept, particularly in 4G and 5G networks, 3GPP had to select IP-based signaling protocols for core mobile services, as opposed to traditional SS7-based protocols used in the circuit-switched domain in use in 2G and 3G networks. In that context, rather than reinventing the wheel, 3GPP decided to leverage Internet protocols and the work carried on by the IETF. Hence, it was not surprise that when 3GPP defined the so-called Packet-switched Streaming Service (PSS) for real-time continuous media delivery, it selected RTSP as its signaling protocol and, more importantly, SIP was eventually selected as the core signaling protocol for all multimedia core services in the mobile (All-)IP domain. This 3GPP decision to use off-the-shelf IETF-standardized signaling protocols has been a key cornerstone for the future of All-IP fixed / mobile networks convergence and Next Generation Networks (NGN) in general. In this context, the main goal of our work has been analyzing how such general purpose IP multimedia signaling protocols are deployed and behave over 3GPP mobile networks. Effectively, usage of IP protocols is key to enable cross-vendor interoperability. On the other hand, due to the specific nature of the mobile domain, there are scenarios where it might be possible to leverage some additional “context” to enhance the performance of such protocols in the particular case of mobile networks. With this idea in mind, the bulk of this thesis work has consisted on analyzing and optimizing the performance of SIP and RTSP multimedia signaling protocols and defining optimized deployment architectures, with particular focus on the 3GPP PSS and the 3GPP Mission Critical Push-to-Talk (MCPTT) service. This work was preceded by a detailed analysis work of the performance of underlying IP, UDP and TCP protocol performance over 3GPP networks, which provided the best baseline for the future work around IP multimedia signaling protocols. Our contributions include the proposal of new optimizations to enhance multimedia streaming session setup procedures, detailed analysis and optimizations of a SIP-based Presence service and, finally, the definition of new use cases and optimized deployment architectures for the 3GPP MCPTT service. All this work has been published in the form of one book, three papers published in JCR cited International Journals, 5 articles published in International Conferences, one paper published in a National Conference and one awarded patent. This thesis work provides a detailed description of all contributions plus a comprehensive overview of their context, the guiding principles beneath all contributions, their applicability to different network deployment technologies (from 2.5G to 5G), a detailed overview of the related OMA and 3GPP architectures, services and design principles. Last but not least, the potential evolution of this research work into the 5G domain is also outlined as well.Els mecanismes de Senyalització en xarxes de nova generació es fonamenten en protocols de senyalització definits per IETF. En particular, SIP i RTSP són dos protocols extensibles basats en missatges de text i paradigma petició-resposta. RTSP va ser concebut per a establir sessions de streaming de continguts, mentre SIP va ser creat inicialment per a facilitar l’establiment de sessions multimèdia (veu, vídeo, xat, compartició) entre usuaris. Tot i així, el seu àmbit d’aplicació s’ha anat expandint i evolucionant fins a cobrir virtualment qualsevol tipus de contingut i sessió multimèdia. A mesura que les xarxes mòbils han anat evolucionant cap a un paradigma “All-IP”, particularment en xarxes 4G i 5G, 3GPP va seleccionar els protocols i arquitectures destinats a gestionar la senyalització dels serveis mòbils presents i futurs. En un moment determinat 3GPP decideix que, a diferència dels sistemes 2G i 3G que fan servir protocols basats en SS7, els sistemes de nova generació farien servir protocols estandarditzats per IETF. Quan 3GPP va començar a estandarditzar el servei de Streaming sobre xarxes mòbils PSS (Packet-switched Streaming Service) va escollir el protocol RTSP com a mecanisme de senyalització. Encara més significatiu, el protocol SIP va ser escollit com a mecanisme de senyalització per a IMS (IP Multimedia Subsystem), l’arquitectura de nova generació que substituirà la xarxa telefònica tradicional i permetrà el desplegament de nous serveis multimèdia. La decisió per part de 3GPP de seleccionar protocols estàndards definits per IETF ha representat una fita cabdal per a la convergència del sistemes All-IP fixes i mòbils, i per al desenvolupament de xarxes NGN (Next Generation Networks) en general. En aquest context, el nostre objectiu inicial ha estat analitzar com aquests protocols de senyalització multimèdia, dissenyats per a xarxes IP genèriques, es comporten sobre xarxes mòbils 3GPP. Efectivament, l’ús de protocols IP és fonamental de cara a facilitar la interoperabilitat de solucions diferents. Per altra banda, hi ha escenaris a on és possible aprofitar informació de “context” addicional per a millorar el comportament d’aquests protocols en al cas particular de xarxes mòbils. El cos principal del treball de la tesi ha consistit en l’anàlisi i optimització del rendiment dels protocols de senyalització multimèdia SIP i RTSP, i la definició d’arquitectures de desplegament, amb èmfasi en els serveis 3GPP PSS i 3GPP Mission Critical Push-to-Talk (MCPTT). Aquest treball ha estat precedit per una feina d’anàlisi detallada del comportament dels protocols IP, TCP i UDP sobre xarxes 3GPP, que va proporcionar els fonaments adequats per a la posterior tasca d’anàlisi de protocols de senyalització sobre xarxes mòbils. Les contribucions inclouen la proposta de noves optimitzacions per a millorar els procediments d’establiment de sessions de streaming multimèdia, l’anàlisi detallat i optimització del servei de Presència basat en SIP i la definició de nous casos d’ús i exemples de desplegament d’arquitectures optimitzades per al servei 3GPP MCPTT. Aquestes contribucions ha quedat reflectides en un llibre, tres articles publicats en Revistes Internacionals amb índex JCR, 5 articles publicats en Conferències Internacionals, un article publicat en Congrés Nacional i l’adjudicació d’una patent. La tesi proporciona una descripció detallada de totes les contribucions, així com un exhaustiu repàs del seu context, dels principis fonamentals subjacents a totes les contribucions, la seva aplicabilitat a diferents tipus de desplegaments de xarxa (des de 2.5G a 5G), així una presentació detallada de les arquitectures associades definides per organismes com OMA o 3GPP. Finalment també es presenta l’evolució potencial de la tasca de recerca cap a sistemes 5G.Postprint (published version

Conveying and Handling Location Information in the IP Multimedia Subsystem

The IP Multimedia Subsystem (IMS), specified by the 3rd Generation Partnership Project (3GPP), is a key element in the next-generation network (NGN) converged architecture. Extending the IMS towards provisioning support for location based services (LBS) will enable enhanced services and offer new revenues to the operator. Conveying location information in the IMS and connecting the IMS with a positioning system are still open issues. This paper presents the design and implementation of an IMS Location Server (ILS) integrating IMS with a positioning system. From the IMS perspective, the ILS serves as a service enabler for LBS. In order to demonstrate proof-of-concept in enhancing IMS-based services, two prototype service scenarios have been implemented: Location-aware Messaging (LaM), and Location-aware Push-to-Talk over cellular (LaPoC). Some work has been done by the IETF in the area of location information transport based on the Session Initiation Protocol (SIP). This paper proposes improvements in this area, primarily related to reducing the necessary amount of signaling with the specification of a new type of location filter. We have conducted measurements in a laboratory environment in order to illustrate our proposed solution and verify the benefits compared to existing solutions in terms of traffic load and session establishment time. Furthermore, we present a case study integrating the ILS with the Ericsson Mobile Positioning System (MPS)

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

Not availabl

Serviços multimédia multicast de próxima geração

Mestrado em Engenharia Electrónica e TelecomunicaçõesUma das mais recentes conquistas na evolução móvel foi o 3G, permitindo o acesso a serviços multimédia com qualidade de serviço assegurada. No entanto, a tecnologia UMTS, tal como definida na sua Release ’99, é apenas capaz de transmitir em modo unicast, sendo manifestamente ineficiente para comunicações multimédia almejando grupos de utilizadores. A tecnologia IMS surge na Release 5 do 3GPP que começou a responder já a algumas necessidades, permitindo comunicações sobre IP oferecendo serviços Internet a qualquer momento e em qualquer lugar sobre tecnologias de comunicação móveis fornecendo pela primeira vez sessões multimédia satisfatórias. A Release 6 por sua vez trouxe a tecnologia MBMS que permite transmissões em broadcast e multicast para redes móveis. O MBMS fornece os serviços de aplicações multimédia que todos estavam à espera, tanto para os utilizadores como para os prestadores de serviços. O operador pode agora fazer uso da tecnologia existente aumentando todo o tipo de benefícios no serviço prestado ao cliente. Com a possível integração destas duas tecnologias passa a ser possível desenvolver serviços assentes em redes convergentes em que os conteúdos são entregues usando tecnologias unicast, multicast ou broadcast. Neste contexto, o principal motivo deste trabalho consiste essencialmente em fazer uso dos recursos da rede terminando com o desperdício dos mesmos e aumentando a eficiência dos serviços através da integração das tecnologias IMS e MBMS. O trabalho realizado começa com o estudo do estado da arte das telecomunicações móveis com referência às tecnologias referidas, seguindo-se a apresentação da possível integração IMS-MBMS e terminando com o projecto de uma plataforma de demonstração que no futuro possa ser uma implementação de serviço multimédia multicast. O objectivo principal é mostrar os benefícios de um serviço que era normalmente executado em unicast relativamente ao modo multicast, fazendo uso da nova convergência de tecnologias IMS e MBMS. Na conclusão do trabalho são referidas as vantagens do uso de portadoras multicast e broadcast, tendo como perspectiva de que este trabalho possa ser um ponto de partida para um novo conjunto de serviços poupando recursos de rede e permitindo uma eficiência considerável em serviços inovadores.3G is bang up to date in the mobile phone industry. It allows access to multimedia services and gives a guarantee of quality of service. The UMTS technology, defined in 3GPP Release ’99, provides an unicast transmission, but it is completely inefficient when it comes to multimedia group communications. The IMS technology first appeared in Release 5 that has already started to consider the interests of the clients. It provides communications over IP, offering Internet services anytime, anywhere on mobile communication technologies. Also, it offers for the first time satisfactory multimedia sessions. On the other hand, Release 6 gave rise to the MBMS technology that provides broadcast and multicast transmissions for mobile networks. The MBMS provides multimedia applications services that everyone was waiting, including users and service providers. Now the operator makes use of existing technology in order to provide better costumer services. The possible integration of these two technologies will contribute to develop services based on converged networks in which contents are delivered through the unicast, multicast or broadcast technologies. Therefore, the objective of this work is basically to make use of network resources avoiding wastes and improving customer services through the integration of the IMS and the MBMS technologies. The executed work starts with the mobile telecommunications state of the art with reference to the referred technologies, followed by the IMS-MBMS convergence presentation and finishing with the proposal for implementation of a service platform that can be used for a multimedia multicast service. The main point is to show the benefits of a service that has been normally executed in unicast mode over the multicast mode, making use of the new IMS and MBMS technologies integration. To closure the work it is referred the advantages to use multicast and broadcast bearers, with the perspective that this work could be a starting point to a new set of services, saving network resources and allowing for innovate services a considerable efficency

Parlay X Web Services for Policy and Charging Control in Multimedia Networks

The paper investigates the capabilities of Parlay X Web Services for Policy and Charging Control (PCC) in managing all Internet-protocol-based multimedia networks (IMSs). PCC is one of the core features of evolved packet networks. It comprises flow-based charging including charging control and online credit control, gating control, and Quality of Service (QoS) control. Based on the analysis of requirements for PCC, the functionality for open access to QoS management and advanced charging is identified. Parlay X Web Services are evaluated for the support of PCC, and some enhancements are suggested. Implementation aspects are discussed, and Parlay X interfaces are mapped onto IMS control protocols. Use cases of Parlay X Web Services for PCC are presented

Optimizing IETF multimedia signaling protocols and architectures in 3GPP networks : an evolutionary approach

Signaling in Next Generation IP-based networks heavily relies in the family of multimedia signaling protocols defined by IETF. Two of these signaling protocols are RTSP and SIP, which are text-based, client-server, request-response signaling protocols aimed at enabling multimedia sessions over IP networks. RTSP was conceived to set up streaming sessions from a Content / Streaming Server to a Streaming Client, while SIP was conceived to set up media (e.g.: voice, video, chat, file sharing, …) sessions among users. However, their scope has evolved and expanded over time to cover virtually any type of content and media session. As mobile networks progressively evolved towards an IP-only (All-IP) concept, particularly in 4G and 5G networks, 3GPP had to select IP-based signaling protocols for core mobile services, as opposed to traditional SS7-based protocols used in the circuit-switched domain in use in 2G and 3G networks. In that context, rather than reinventing the wheel, 3GPP decided to leverage Internet protocols and the work carried on by the IETF. Hence, it was not surprise that when 3GPP defined the so-called Packet-switched Streaming Service (PSS) for real-time continuous media delivery, it selected RTSP as its signaling protocol and, more importantly, SIP was eventually selected as the core signaling protocol for all multimedia core services in the mobile (All-)IP domain. This 3GPP decision to use off-the-shelf IETF-standardized signaling protocols has been a key cornerstone for the future of All-IP fixed / mobile networks convergence and Next Generation Networks (NGN) in general. In this context, the main goal of our work has been analyzing how such general purpose IP multimedia signaling protocols are deployed and behave over 3GPP mobile networks. Effectively, usage of IP protocols is key to enable cross-vendor interoperability. On the other hand, due to the specific nature of the mobile domain, there are scenarios where it might be possible to leverage some additional “context” to enhance the performance of such protocols in the particular case of mobile networks. With this idea in mind, the bulk of this thesis work has consisted on analyzing and optimizing the performance of SIP and RTSP multimedia signaling protocols and defining optimized deployment architectures, with particular focus on the 3GPP PSS and the 3GPP Mission Critical Push-to-Talk (MCPTT) service. This work was preceded by a detailed analysis work of the performance of underlying IP, UDP and TCP protocol performance over 3GPP networks, which provided the best baseline for the future work around IP multimedia signaling protocols. Our contributions include the proposal of new optimizations to enhance multimedia streaming session setup procedures, detailed analysis and optimizations of a SIP-based Presence service and, finally, the definition of new use cases and optimized deployment architectures for the 3GPP MCPTT service. All this work has been published in the form of one book, three papers published in JCR cited International Journals, 5 articles published in International Conferences, one paper published in a National Conference and one awarded patent. This thesis work provides a detailed description of all contributions plus a comprehensive overview of their context, the guiding principles beneath all contributions, their applicability to different network deployment technologies (from 2.5G to 5G), a detailed overview of the related OMA and 3GPP architectures, services and design principles. Last but not least, the potential evolution of this research work into the 5G domain is also outlined as well.Els mecanismes de Senyalització en xarxes de nova generació es fonamenten en protocols de senyalització definits per IETF. En particular, SIP i RTSP són dos protocols extensibles basats en missatges de text i paradigma petició-resposta. RTSP va ser concebut per a establir sessions de streaming de continguts, mentre SIP va ser creat inicialment per a facilitar l’establiment de sessions multimèdia (veu, vídeo, xat, compartició) entre usuaris. Tot i així, el seu àmbit d’aplicació s’ha anat expandint i evolucionant fins a cobrir virtualment qualsevol tipus de contingut i sessió multimèdia. A mesura que les xarxes mòbils han anat evolucionant cap a un paradigma “All-IP”, particularment en xarxes 4G i 5G, 3GPP va seleccionar els protocols i arquitectures destinats a gestionar la senyalització dels serveis mòbils presents i futurs. En un moment determinat 3GPP decideix que, a diferència dels sistemes 2G i 3G que fan servir protocols basats en SS7, els sistemes de nova generació farien servir protocols estandarditzats per IETF. Quan 3GPP va començar a estandarditzar el servei de Streaming sobre xarxes mòbils PSS (Packet-switched Streaming Service) va escollir el protocol RTSP com a mecanisme de senyalització. Encara més significatiu, el protocol SIP va ser escollit com a mecanisme de senyalització per a IMS (IP Multimedia Subsystem), l’arquitectura de nova generació que substituirà la xarxa telefònica tradicional i permetrà el desplegament de nous serveis multimèdia. La decisió per part de 3GPP de seleccionar protocols estàndards definits per IETF ha representat una fita cabdal per a la convergència del sistemes All-IP fixes i mòbils, i per al desenvolupament de xarxes NGN (Next Generation Networks) en general. En aquest context, el nostre objectiu inicial ha estat analitzar com aquests protocols de senyalització multimèdia, dissenyats per a xarxes IP genèriques, es comporten sobre xarxes mòbils 3GPP. Efectivament, l’ús de protocols IP és fonamental de cara a facilitar la interoperabilitat de solucions diferents. Per altra banda, hi ha escenaris a on és possible aprofitar informació de “context” addicional per a millorar el comportament d’aquests protocols en al cas particular de xarxes mòbils. El cos principal del treball de la tesi ha consistit en l’anàlisi i optimització del rendiment dels protocols de senyalització multimèdia SIP i RTSP, i la definició d’arquitectures de desplegament, amb èmfasi en els serveis 3GPP PSS i 3GPP Mission Critical Push-to-Talk (MCPTT). Aquest treball ha estat precedit per una feina d’anàlisi detallada del comportament dels protocols IP, TCP i UDP sobre xarxes 3GPP, que va proporcionar els fonaments adequats per a la posterior tasca d’anàlisi de protocols de senyalització sobre xarxes mòbils. Les contribucions inclouen la proposta de noves optimitzacions per a millorar els procediments d’establiment de sessions de streaming multimèdia, l’anàlisi detallat i optimització del servei de Presència basat en SIP i la definició de nous casos d’ús i exemples de desplegament d’arquitectures optimitzades per al servei 3GPP MCPTT. Aquestes contribucions ha quedat reflectides en un llibre, tres articles publicats en Revistes Internacionals amb índex JCR, 5 articles publicats en Conferències Internacionals, un article publicat en Congrés Nacional i l’adjudicació d’una patent. La tesi proporciona una descripció detallada de totes les contribucions, així com un exhaustiu repàs del seu context, dels principis fonamentals subjacents a totes les contribucions, la seva aplicabilitat a diferents tipus de desplegaments de xarxa (des de 2.5G a 5G), així una presentació detallada de les arquitectures associades definides per organismes com OMA o 3GPP. Finalment també es presenta l’evolució potencial de la tasca de recerca cap a sistemes 5G

Sip Based Mobile Voice Over Ip Client For Wireess Networks

Tez (Yüksek Lisans) -- İstanbul Teknik Üniversitesi, Fen Bilimleri Enstitüsü, 2008Thesis (M.Sc.) -- İstanbul Technical University, Institute of Science and Technology, 2008Bu tez SIP tabanlı mobile bir VoIP istemcisinin tasarımını ve gerçeklenmesini tanımlar. Bu tez temelde çoktürel ağlar üzerinde çalışabilen bir VoIP istemcisi tasarımının çözülmesi gereken iki sorununun üzerinde yogunlaşır. Birinci ve en zorlu sorun farklı erişim teknolojileri arasında kullanıcıya fark ettirmeden yer değişim desteği sağlanmasıdır. Bu tezde, kullanıcıya fark ettirmeden el değiştirme yönetimi, uygulama katmanında, multimedya oturumunu başlatmak, sonlandırmak ve değiştirmek için kullanılan Oturum Başlatma Protokolü (SIP) kullanılarak ele alınmıştır. SIP yaygın bir şekilde kabul edilmekte olan bir VoIP standartıdır. Kullanıcıya fark ettirmeden el değiştirmeyi destekleyebilmek için, VoIP istemcisi üzerinde çalışan SIP tabanlı bir bağlantı yöneticisi önerilmiştir. Bağlantı yöneticisi yeni ağlar keşfettiğinde, adaylar listesinden bir ağ seçer ve hali hazırda yürütülmekte olan iletişimi kullanıcıya fark ettirmeden yeni ağ arayüzüne aktarır. Dolayısı ile, bu birim Wi-Fi, 3G gibi çoktürel ağlar arasında dolaşmayı sağlar. İkinci sorun ise, en kaliteli çağrı (arama) desteğini sağlamaktır. En kaliteli çağrı desteği, iletişim kurmak isteyen tarafların farklı türden ağlara bağlı olmaları durumunda, VoIP uygulamasının iletişim tipine (yarı-çift yönlü yada tam-çift yönlü) karar vermebilmesi demektir. Örneğin, eğer iletişim kurmak isteyen taraflardan biri bir GSM ağındaysa, en iyi çağrı kalitesini yakalayabilmek için, iletişim yarı-çift yönlü olarak kurulmalıdır. Bu tez, bahsedilen özelliği desteklemek için, istemci tabanlı bir karar mekanizması önerir. Bu karar mekanizması, iletişim kurulmak istenen tarafa, istemcinin içinde bulunduğu ağa göre belirlenmiş iletişim tipini içeren bir davet iletisi gönderir. Diğer istemci bu davet iletisini aldıktan sonra, aynı karar mekanizması, iletişimi “bas-konuş VoIP” yada “tam-çift yönlü VoIP” olarak ayarlar.This thesis describes the design and the implementation of a SIP-based mobile VoIP client. It mainly focuses on two challenges of designing a VoIP client which works on heterogeneous network environments. One and the most challenging problem is the provision of seamless mobility support among different access technologies. In this thesis, seamless handover management is handled at the application layer by using Session initiation protocol (SIP), which is used to initiate, terminate, and modify multimedia session. SIP is becoming a widely accepted standard for VoIP. To support seamless handover, a SIP based connection manager is proposed on VoIP client application. As new networks are discovered by the connection manager, it selects a new network from the candidate list and transfers the current communication to the new network interface seamlessly. Therefore, this module provides roaming across heterogeneous networks such as Wi-Fi, 3G. Second problem is providing the best effort call quality support. It means that if the communication parties are in dissimilar networks, the VoIP application should decide the communication type (half-duplex or full-duplex). For instance, if one of the communication parties is in a GSM network, then the communication should be established as a half-duplex manner to achieve best call quality. This thesis proposes a client-based decision mechanism to support this property. This decision mechanism sends an invite message including the communication type (half-duplex or full-duplex) of the client according to the network in which it operates to the other communication party. After the other client receives this invite message, same decision mechanism adjusts the communication as either a “push to talk VoIP” or a “full-duplex VoIP”.Yüksek LisansM.Sc

Peer-to-peer television for the IP multimedia subsystem

Peer-to-peer (P2P) video streaming has generated a significant amount of interest in both the research community and the industry, which find it a cost-effective solution to the user scalability problem. However, despite the success of Internet-based applications, the adoption has been limited for commercial services, such as Internet Protocol Television (IPTV). With the advent of the next-generation-networks (NGN) based on the IP Multimedia Subsystem (IMS), advocating for an open and inter-operable architecture, P2P emerges as a possible alternative in situations where the traditional mechanisms are not possible or economically feasible. This work proposes a P2P IPTV architecture for an IMS-based NGN, called P2PTV, which allows one or more service providers to use a common P2P infrastructure for streaming the TV channels to their subscribers. Instead of using servers, we rely on the uploading capabilities of the user equipments, like set-top boxes, located at the customers’ premise. We comply with the existing IMS and IPTV standards from the 3rd Generation Partnership Project (3GPP) and the Telecommunications and Internet converged Services and Protocols for Advanced Networking (TISPAN) bodies, where a centralized P2PTV application server (AS) manages the customer access to the service and the peer participation. Because watching TV is a complex and demanding user activity, we face two significant challenges. The first is to accommodate the mandatory IMS signaling, which reserves in the network the necessary QoS resources during every channel change, establishing a multimedia session between communicating peers. The second is represented by the streaming interruptions, or churn, when the uploading peer turns off or changes its current TV channel. To tackle these problems, we propose two enhancements. A fast signaling method, which uses inactive uploading sessions with reserved but unused QoS, to improve the tuning delay for new channel users. At every moment, the AS uses a feedback based algorithm to compute the number of necessary sessions that accommodates well the demand, while preventing the over-reservation of resources. We approach with special care mobility situations, where a proactive transfer of the multimedia session context using the IEEE 802.21 standard offers the best alternative to current methods. The second enhancement addresses the peer churn during channel changes. With every TV channel divided into a number of streams, we enable peers to download and upload streams different from their current channel, increasing the stability of their participation. Unlike similar work, we benefit from our estimation of the user demand and propose a decentralized method for a balanced assignment of peer bandwidth. We evaluate the performance of the P2PTV through modeling and large-scale computer simulations. A simpler experimental setting, with pure P2P streaming, indicates the improvements over the delay and peer churn. In more complex scenarios, especially with resource-poor peers having a limited upload capacity, we envision P2P as a complementary solution to traditional approaches like IP multicast. Reserving P2P for unpopular TV channels exploits the peer capacity and prevents the necessity of a large number of sparsely used multicast trees. Future work may refine the AS algorithms, address different experimental scenarios, and extend the lessons learned to non-IMS networks. ----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------La transmisión de vídeo con tecnologías peer-to-peer (P2P) ha generado un gran interés, tanto en la industria como en la comunidad científica, quienes han encontrado en dicha unión la solución para afrontar los problemas de escalabilidad de la transmisión de vídeo, reduciendo al mismo tiempo sus costes. A pesar del éxito de estos mecanismos en Internet, la transmisión de vídeo mediante técnicas P2P no se ha utilizado en servicios comerciales como puede ser el de televisión por IP (IPTV). Con la aparición de propuestas de redes de próxima generación basadas en el IP Multimedia Subsystem (IMS), que permite una arquitectura abierta e interoperable, los mecanismos basados en P2P emergen como posibles alternativas en situaciones donde los mecanismos tradicionales de transmisión de vídeo no se pueden desplegar o no son económicamente viables. Esta tesis propone una arquitectura de servicio de televisión peer-to-peer para una red de siguiente generación basada en IMS, que abreviaremos como P2PTV, que permite a uno o más proveedores de servicio utilizar una infraestructura P2P común para la transmisión de canales de TV a sus suscriptores. En vez de utilizar varios servidores, proponemos utilizar la capacidad de envío de los equipos de usuario, como los set-top boxes, localizados en el lado del cliente. En esta tesis extendemos los trabajos de estandarización sobre IMS IPTV de los organismos 3rd Generation Partnership Project (3GPP) y del Telecommunications and Internet converged Services and Protocols for Advanced Networking (TISPAN), donde un servidor de aplicación (AS) central de P2PTV administra el acceso de los clientes al servicio y permite compartir los recursos de los equipos. Debido a que el acceso a los canales de TV por parte de los usuarios es una actividad compleja, nos enfrentamos a dos retos importantes. El primero es administrar la señalización de IMS, con la cual se reservan los recursos de QoS necesarios durante cada cambio de canal, estableciendo una sesión multimedia entre los diferentes elementos de la comunicación. El segundo está representado por las interrupciones de la reproducción de video, causado por los equipos que sirven dicho vídeo cuando estos se desconectan del sistema o cuando cambian de canal. Para afrontar estos retos, proponemos dos mejoras al sistema. La primera mejora introduce el método de señalización rápida, en la cual se utilizan sesiones multimedia inactivas pero con recursos reservados para acelerar las conexiones entre usuarios. En cada momento, el AS utiliza la información extraída del algoritmo propuesto, que calcula el número de sesiones necesarias para administrar la demanda de conexiones, pero sin realizar una sobre-estimación, manteniendo bajo el uso de los recursos. Hemos abordado con especial cuidado la movilidad de los usuarios, donde se ha propuesto una transferencia de sesión pro-activa utilizando el estándar IEEE 802.21, el cual brinda una mejor alternativa que los métodos propuestos hasta la fecha. La segunda mejora se enfoca en las desconexiones de usuarios durante cambios de canal. Dividiendo los canales de TV en varios segmentos, permitimos a los equipos descargar o enviar diferentes partes de cualquier canal, aumentando la estabilidad de su participación. A diferencia de otros trabajos, nuestra propuesta se beneficia de la estimación de la demanda futura de los usuarios, proponiendo un método descentralizado para una asignación balanceada del ancho de banda de los equipos. Hemos evaluado el rendimiento del sistema P2PTV a través de modelado y de simulaciones de ordenador en sistemas IPTV de gran escala. Una configuración simple, con envío P2P puro, indica mejoras en el retardo y número de desconexiones de usuarios. En escenarios más complejos, especialmente con equipos con pocos recursos en la subida, sugerimos el uso de P2P como una solución complementaria a las soluciones tradicionales de multicast IP. Reservando el uso de P2P para los canales de TV poco populares, se permite explotar los recursos de los equipos y se previene la necesidad de un alto número de árboles multicast dispersos. Como trabajo futuro, se propone refinar los algoritmos del AS, abordar diferentes escenarios experimentales y también extender las lecciones aprendidas en esta tesis a otros sistemas no basados en IMS