18 research outputs found
SIP COMPRESSION
The wired line network has been well studied and widely used for a long
time. Most of its protocols are so successful that passed the test of
time. There are many
similar tasks in mobile and wired line environment, and we would like
to achieve compatible, inter-working solutions. So it is a plausible
idea to use the protocols of the wired line network in mobile
environment too. However, the mobile and wired line environment differ
significantly; mainly the bandwidth is different in the two networks.
Although the difference is going to be smaller with the help of new
generation of mobile networks, it will
still remain significant. An acceptable solution is to compress these
protocols. We have not found such a solution in the literature
so our opinion is that this article is the first dealing with SIP compression.
We have created a demonstration
system, which connects two SIP user agents to each
other and ensures the compression and decompression of the
messages between them.
In this article we show our development about adapting various compressing
algorithms for SIP compression, and we evaluate them
Supporting group mobility in mission-critical wireless networks for SIP-based applications
DiplomityÜssä tarkastellaan viiveherkkien SIP-sovellusten verkkoalueiden välistä ryhmäliikkuvuutta langattomissa, IEEE 802.11x -pohjaisissa IPv4/IPv6 verkkoympäristÜissä. Nykyaikaisissa kriisinhallintatehtävissä reaaliaikaisen viestinnän merkitys on viime vuosina vahvasti korostunut. Tähän tarkoitukseen käytetyt viestintäjärjestelmät ovat olleet tavallisesti erittäin kalliita. Langattomien teknologioiden nopea kehitys on kuitenkin suunnannut mielenkiinnon edullisiin, kaupallisiin siviilipuolen valmisratkaisuihin.
Pitkät yhteydensiirtoviiveet ovat tärkeä ongelma reaaliaikaliikenteen yhteydensiirron kannalta. VoIP-pohjaisen puheliikenteen on todettu kestävän enimmillään suuruusluokkaa 100 ms olevia viiveaikoja palvelunlaadun ratkaisevasti kärsimättä. Linkkitason yhteydensiirron ohella duplikaattiosoitteiden tarkistuksella DHCP-osoitteenhaun aikana ja SIP-yhteyden uudelleenmuodostuksella on saumattoman yhteydensiirron kannalta olennainen merkitys.
Ryhmäliikkuvuus on saanut osakseen paljon huomiota ad hoc -verkkojen tutkimuksessa. TyÜssä tutkitaan mandollisesti saavutettavia hyÜtyjä, joita ryhmäliikkuvuusmalli pystyisi perinteiseen yhteydensiirtotapaan nähden tuomaan hierarkkisissa infrastruktuurisissa SIP-verkoissa.
Sovellustason liikkuvuutta ja signaloinnin tehokkuutta tarkastellaan kaistankäytÜn ja tietoturvallisuuden näkÜkulmasta. Kokeellisessa osiossa pyritään mallintamaan ryhmäyhteydensiirtoja yksinkertaisessa, simuloidussa ympäristÜssä. Päätelmien tueksi yhteydensiirtojen suorituskykyä arvioidaan lisäksi numeerisella analyysilla.This thesis studies the provision of group mobility during inter-domain hand-offs for delay-sensitive SIP applications over wireless IPv4/IPv6 network environment, based on the IEEE 802.11x platform. In contemporary disaster relief operations, the role of real-time communications has been strongly escalating over the recent years. The communication systems used for these ends have been conventionally very expensive. The rapid evolution of wireless technologies has brought the focus of interest to the affordable Common-Off-the-Shelf civilian applications.
Long latencies during hand-offs for real-time traffic are a very important problem. As the studies have pointed out, the VoIP-based voice traffic can withstand maximum approximate disruption times of 100Â ms, without too high degradation in the quality of service. Along with the link-layer hand-off, the duplicate address detection procedure during DHCP address acquisition and the SIP connection re-establishment both have a major impact on the hand-off latency.
The group mobility has gained high attention in the research of ad-hoc networks. The work studies the benefits that this scheme could possibly bring over the conventional hand-offs in hierarchical infrastructured SIP networks.
Different approaches to application-level mobility and the signaling efficiency are examined from the viewpoint of bandwidth usage and network security. In the experimental part, group hand-offs are modeled in a simple, simulated environment. In addition, a numerical analysis is used to assess the hand-off performance to support the made conclusions
Analysis of Session Establishment Signaling Delay in IP Multimedia Subsystem
Abstract. This paper investigates and analyzes SIP delay in the session establishment signaling procedure in the IMS system. We investigate the delay for end-to-end link scenarios such as WiMAX-to-WiMAX, UMTS-to-UMTS, UMTS-to-WiMAX and vice versa. The analyses consider three types of delays: transmission delay, processing delay and queuing delay. The obtained results show that the main delay of session establishment signaling process is due to the processing delay. In addition, the lower channel rate in the UMTS network as well as IMS service rate has significant impact to the session establishment signaling delay
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, âŚ) 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 âcontextâ 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 SenyalitzaciĂł en xarxes de nova generaciĂł es fonamenten en protocols de senyalitzaciĂł definits per IETF. En particular, SIP i RTSP sĂłn dos protocols extensibles basats en missatges de text i paradigma peticiĂł-resposta. RTSP va ser concebut per a establir sessions de streaming de continguts, mentre SIP va ser creat inicialment per a facilitar lâestabliment de sessions multimèdia (veu, vĂdeo, xat, comparticiĂł) entre usuaris. Tot i aixĂ, el seu Ă mbit dâaplicaciĂł sâha anat expandint i evolucionant fins a cobrir virtualment qualsevol tipus de contingut i sessiĂł multimèdia. A mesura que les xarxes mòbils han anat evolucionant cap a un paradigma âAll-IPâ, particularment en xarxes 4G i 5G, 3GPP va seleccionar els protocols i arquitectures destinats a gestionar la senyalitzaciĂł 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 generaciĂł 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 senyalitzaciĂł. Encara mĂŠs significatiu, el protocol SIP va ser escollit com a mecanisme de senyalitzaciĂł per a IMS (IP Multimedia Subsystem), lâarquitectura de nova generaciĂł que substituirĂ la xarxa telefònica tradicional i permetrĂ el desplegament de nous serveis multimèdia. La decisiĂł per part de 3GPP de seleccionar protocols estĂ ndards 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 senyalitzaciĂł multimèdia, dissenyats per a xarxes IP genèriques, es comporten sobre xarxes mòbils 3GPP. Efectivament, lâĂşs de protocols IP ĂŠs fonamental de cara a facilitar la interoperabilitat de solucions diferents. Per altra banda, hi ha escenaris a on ĂŠs possible aprofitar informaciĂł de âcontextâ addicional per a millorar el comportament dâaquests protocols en al cas particular de xarxes mòbils. El cos principal del treball de la tesi ha consistit en lâanĂ lisi i optimitzaciĂł del rendiment dels protocols de senyalitzaciĂł multimèdia SIP i RTSP, i la definiciĂł dâarquitectures 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 dâanĂ lisi detallada del comportament dels protocols IP, TCP i UDP sobre xarxes 3GPP, que va proporcionar els fonaments adequats per a la posterior tasca dâanĂ lisi de protocols de senyalitzaciĂł sobre xarxes mòbils. Les contribucions inclouen la proposta de noves optimitzacions per a millorar els procediments dâestabliment de sessions de streaming multimèdia, lâanĂ lisi detallat i optimitzaciĂł del servei de Presència basat en SIP i la definiciĂł de nous casos dâĂşs i exemples de desplegament dâarquitectures optimitzades per al servei 3GPP MCPTT. Aquestes contribucions ha quedat reflectides en un llibre, tres articles publicats en Revistes Internacionals amb Ăndex JCR, 5 articles publicats en Conferències Internacionals, un article publicat en CongrĂŠs Nacional i lâadjudicaciĂł dâuna patent. La tesi proporciona una descripciĂł detallada de totes les contribucions, aixĂ com un exhaustiu repĂ s 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), aixĂ una presentaciĂł detallada de les arquitectures associades definides per organismes com OMA o 3GPP. Finalment tambĂŠ es presenta lâevoluciĂł potencial de la tasca de recerca cap a sistemes 5G.Postprint (published version
Contributions to presence-based systems for deploying ubiquitous communication services
Next-Generation Networks (NGNs) will converge the existing fixed and wireless networks. These networks rely on the IMS (IP Multimedia Subsystem), introduced by the 3GPP. The presence service came into being in instant messaging applications. A userÂżs presence information consists in any context that is necessary for applications to handle and adapt the user's communications. The presence service is crucial in the IMS to deploy ubiquitous services. SIMPLE is the standard protocol for handling presence and instant messages. This protocol disseminates users' presence information through subscriptions, notifications and publications. SIMPLE generates much signaling traffic for constantly disseminating presence information and maintaining subscriptions, which may overload network servers. This issue is even more harmful to the IMS due to its centralized servers. A key factor in the success of NGNs is to provide users with always-on services that are seamlessly part of their daily life. Personalizing these services according to the users' needs is necessary for the success of these services. To this end, presence information is considered as a crucial tool for user-based personalization.
This thesis can be briefly summarized through the following contributions:
We propose filtering and controlling the rate of presence publications so as to reduce the information sent over access links. We probabilistically model presence information through Markov chains, and analyzed the efficiency of controlling the rate of publications that are modeled by a particular Markov chain. The reported results show that this technique certainly reduces presence overload.
We mathematically study the amount of presence traffic exchanged between domains, and analyze the efficiency of several strategies for reducing this traffic.
We propose an strategy, which we call Common Subscribe (CS), for reducing the presence traffic exchanged between federated domains. We compare this strategy traffic with that generated by other optimizations. The reported results show that CS is the most efficient at reducing presence traffic.
We analyze the load in the number of messages that several inter-domain traffic optimizations cause to the IMS centralized servers. Our proposed strategy, CS, combined with an RLS (i.e., a SIMPLE optimization) is the only optimization that reduces the IMS load; the others increase this load.
We estimate the efficiency of the RLS, thereby concluding that the RLS is not efficient under certain circumstances, and hence this optimization is discouraged.
We propose a queuing system for optimizing presence traffic on both the network core and access link, which is capable to adapt the publication and notification rate based on some quality conditions (e.g, maximum delay). We probabilistically model this system, and validate it in different scenarios.
We propose, and implement a prototype of, a fully-distributed platform for handling user presence information. This approach allows integrating Internet Services, such as HTTP or VoIP, and optimizing these services in an easy, user-personalized way.
We have developed SECE (Sense Everything, Control Everything), a platform for users to create rules that handle their communications and Internet Services proactively. SECE interacts with multiple third-party services for obtaining as much user context as possible. We have developed a natural-English-like formal language for SECE rules.
We have enhanced SECE for discovering web services automatically through the Web Ontology Language (OWL). SECE allows composing web services automatically based on real-world events, which is a significant contribution to the Semantic Web.
The research presented in this thesis has been published through 3 book chapters, 4 international journals (3 of them are indexed in JCR), 10 international conference papers, 1 demonstration at an international conference, and 1 national conferenceNext-Generation Networks (NGNs) son las redes de prĂłxima generaciĂłn que soportaran la convergencia de redes de telecomunicaciĂłn inalĂĄmbricas y fijas. La base de NGNs es el IMS (IP Multimedia Subsystem), introducido por el 3GPP. El servicio de presencia naciĂł de aplicaciones de mesajerĂa instantĂĄnea. La informaciĂłn de presencia de un usuario consiste en cualquier tipo de informaciĂłn que es de utilidad para manejar las comunicaciones con el usuario. El servicio de presencia es una parte esencial del IMS para el despliegue de servicios ubicuos. SIMPLE es el protocolo estĂĄndar para manejar presencia y mensajes instantĂĄneos en el IMS. Este protocolo distribuye la informaciĂłn de presencia de los usuarios a travĂŠs de suscripciones, notificaciones y publicaciones. SIMPLE genera mucho trĂĄfico por la diseminaciĂłn constante de informaciĂłn de presencia y el mantenimiento de las suscripciones, lo cual puede saturar los servidores de red. Este problema es todavĂa mĂĄs perjudicial en el IMS, debido al carĂĄcter centralizado de sus servidores. Un factor clave en el ĂŠxito de NGNs es proporcionar a los usuarios servicios ubicuos que esten integrados en su vida diaria y asi interactĂşen con los usuarios constantemente. La personalizaciĂłn de estos servicios basado en los usuarios es imprescindible para el ĂŠxito de los mismos. Para este fin, la informaciĂłn de presencia es considerada como una herramienta base. La tesis realizada se puede resumir brevemente en los siguientes contribuciones: Proponemos filtrar y controlar el ratio de las publicaciones de presencia para reducir la cantidad de informaciĂłn enviada en la red de acceso. Modelamos la informaciĂłn de presencia probabilĂsticamente mediante cadenas de Markov, y analizamos la eficiencia de controlar el ratio de publicaciones con una cadena de Markov. Los resultados muestran que este mecanismo puede efectivamente reducir el trĂĄfico de presencia. Estudiamos matemĂĄticamente la cantidad de trĂĄfico de presencia generada entre dominios y analizamos el rendimiento de tres estrategias para reducir este trĂĄfico. Proponemos una estrategia, la cual llamamos Common Subscribe (CS), para reducir el trĂĄfico de presencia entre dominios federados. Comparamos el trĂĄfico generado por CS frente a otras estrategias de optimizaciĂłn. Los resultados de este anĂĄlisis muestran que CS es la estrategia mĂĄs efectiva. Analizamos la carga en numero de mensajes introducida por diferentes optimizaciones de trĂĄfico de presencia en los servidores centralizados del IMS. Nuestra propuesta, CS, combinada con un RLS (i.e, una optimizaciĂłn de SIMPLE), es la unica optimizaciĂłn que reduce la carga en el IMS. Estimamos la eficiencia del RLS, deduciendo que un RLS no es eficiente en ciertas circunstancias, en las que es preferible no usar esta optimizaciĂłn. Proponemos un sistema de colas para optimizar el trĂĄfico de presencia tanto en el nĂşcleo de red como en la red de acceso, y que puede adaptar el ratio de publicaciĂłn y notificaciĂłn en base a varios parametros de calidad (e.g., maximo retraso). Modelamos y analizamos este sistema de colas probabilĂsticamente en diferentes escenarios. Proponemos una arquitectura totalmente distribuida para manejar las informaciĂłn de presencia del usuario, de la cual hemos implementado un prototipo. Esta propuesta permite la integracion sencilla y personalizada al usuario de servicios de Internet, como HTTP o VoIP, asi como la optimizacĂłn de estos servicios. Hemos desarrollado SECE (Sense Everything, Control Everything), una plataforma donde los usuarios pueden crear reglas para manejar todas sus comunicaciones y servicios de Internet de forma proactiva. SECE interactĂşa con una multitud de servicios para conseguir todo el contexto possible del usuario. Hemos desarollado un lenguaje formal que parace como Ingles natural para que los usuarios puedan crear sus reglas. Hemos mejorado SECE para descubrir servicios web automaticamente a travĂŠs del lenguaje OWL (Web Ontology Language)
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, âŚ) 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 âcontextâ 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 SenyalitzaciĂł en xarxes de nova generaciĂł es fonamenten en protocols de senyalitzaciĂł definits per IETF. En particular, SIP i RTSP sĂłn dos protocols extensibles basats en missatges de text i paradigma peticiĂł-resposta. RTSP va ser concebut per a establir sessions de streaming de continguts, mentre SIP va ser creat inicialment per a facilitar lâestabliment de sessions multimèdia (veu, vĂdeo, xat, comparticiĂł) entre usuaris. Tot i aixĂ, el seu Ă mbit dâaplicaciĂł sâha anat expandint i evolucionant fins a cobrir virtualment qualsevol tipus de contingut i sessiĂł multimèdia. A mesura que les xarxes mòbils han anat evolucionant cap a un paradigma âAll-IPâ, particularment en xarxes 4G i 5G, 3GPP va seleccionar els protocols i arquitectures destinats a gestionar la senyalitzaciĂł 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 generaciĂł 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 senyalitzaciĂł. Encara mĂŠs significatiu, el protocol SIP va ser escollit com a mecanisme de senyalitzaciĂł per a IMS (IP Multimedia Subsystem), lâarquitectura de nova generaciĂł que substituirĂ la xarxa telefònica tradicional i permetrĂ el desplegament de nous serveis multimèdia. La decisiĂł per part de 3GPP de seleccionar protocols estĂ ndards 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 senyalitzaciĂł multimèdia, dissenyats per a xarxes IP genèriques, es comporten sobre xarxes mòbils 3GPP. Efectivament, lâĂşs de protocols IP ĂŠs fonamental de cara a facilitar la interoperabilitat de solucions diferents. Per altra banda, hi ha escenaris a on ĂŠs possible aprofitar informaciĂł de âcontextâ addicional per a millorar el comportament dâaquests protocols en al cas particular de xarxes mòbils. El cos principal del treball de la tesi ha consistit en lâanĂ lisi i optimitzaciĂł del rendiment dels protocols de senyalitzaciĂł multimèdia SIP i RTSP, i la definiciĂł dâarquitectures 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 dâanĂ lisi detallada del comportament dels protocols IP, TCP i UDP sobre xarxes 3GPP, que va proporcionar els fonaments adequats per a la posterior tasca dâanĂ lisi de protocols de senyalitzaciĂł sobre xarxes mòbils. Les contribucions inclouen la proposta de noves optimitzacions per a millorar els procediments dâestabliment de sessions de streaming multimèdia, lâanĂ lisi detallat i optimitzaciĂł del servei de Presència basat en SIP i la definiciĂł de nous casos dâĂşs i exemples de desplegament dâarquitectures optimitzades per al servei 3GPP MCPTT. Aquestes contribucions ha quedat reflectides en un llibre, tres articles publicats en Revistes Internacionals amb Ăndex JCR, 5 articles publicats en Conferències Internacionals, un article publicat en CongrĂŠs Nacional i lâadjudicaciĂł dâuna patent. La tesi proporciona una descripciĂł detallada de totes les contribucions, aixĂ com un exhaustiu repĂ s 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), aixĂ una presentaciĂł detallada de les arquitectures associades definides per organismes com OMA o 3GPP. Finalment tambĂŠ es presenta lâevoluciĂł potencial de la tasca de recerca cap a sistemes 5G
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