16 research outputs found
Supporting group mobility in mission-critical wireless networks for SIP-based applications
Diplomityössä tarkastellaan viiveherkkien SIP-sovellusten verkkoalueiden välistä ryhmäliikkuvuutta langattomissa, IEEE 802.11x -pohjaisissa IPv4/IPv6 verkkoympäristöissä. 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. Työssä tutkitaan mandollisesti saavutettavia hyötyjä, joita ryhmäliikkuvuusmalli pystyisi perinteiseen yhteydensiirtotapaan nähden tuomaan hierarkkisissa infrastruktuurisissa SIP-verkoissa.
Sovellustason liikkuvuutta ja signaloinnin tehokkuutta tarkastellaan kaistankäytön ja tietoturvallisuuden näkökulmasta. Kokeellisessa osiossa pyritään mallintamaan ryhmäyhteydensiirtoja yksinkertaisessa, simuloidussa ympäristössä. 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 100 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
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
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
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 user¿s 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 próxima generación que soportaran la convergencia de redes de telecomunicación inalámbricas y fijas. La base de NGNs es el IMS (IP Multimedia Subsystem), introducido por el 3GPP. El servicio de presencia nació de aplicaciones de mesajería instantánea. La información de presencia de un usuario consiste en cualquier tipo de información 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 estándar para manejar presencia y mensajes instantáneos en el IMS. Este protocolo distribuye la información de presencia de los usuarios a través de suscripciones, notificaciones y publicaciones. SIMPLE genera mucho tráfico por la diseminación constante de información de presencia y el mantenimiento de las suscripciones, lo cual puede saturar los servidores de red. Este problema es todavía más perjudicial en el IMS, debido al carácter centralizado de sus servidores. Un factor clave en el éxito de NGNs es proporcionar a los usuarios servicios ubicuos que esten integrados en su vida diaria y asi interactúen con los usuarios constantemente. La personalización de estos servicios basado en los usuarios es imprescindible para el éxito de los mismos. Para este fin, la información 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 información enviada en la red de acceso. Modelamos la información de presencia probabilísticamente 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 tráfico de presencia. Estudiamos matemáticamente la cantidad de tráfico de presencia generada entre dominios y analizamos el rendimiento de tres estrategias para reducir este tráfico. Proponemos una estrategia, la cual llamamos Common Subscribe (CS), para reducir el tráfico de presencia entre dominios federados. Comparamos el tráfico generado por CS frente a otras estrategias de optimización. Los resultados de este análisis muestran que CS es la estrategia más efectiva. Analizamos la carga en numero de mensajes introducida por diferentes optimizaciones de tráfico de presencia en los servidores centralizados del IMS. Nuestra propuesta, CS, combinada con un RLS (i.e, una optimización de SIMPLE), es la unica optimización 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 optimización. Proponemos un sistema de colas para optimizar el tráfico de presencia tanto en el núcleo de red como en la red de acceso, y que puede adaptar el ratio de publicación y notificación en base a varios parametros de calidad (e.g., maximo retraso). Modelamos y analizamos este sistema de colas probabilísticamente en diferentes escenarios. Proponemos una arquitectura totalmente distribuida para manejar las información 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 optimizacón 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 interactúa 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 través del lenguaje OWL (Web Ontology Language)
VoLTE service implementation in EPS-IMS networks
Diplomová práce popisuje VoLTE službu, vývoj a nasazení LTE (zaváděcí fázi, skutečný LTE stav a výhledy do budoucna atd.), EPC-IMS architekturu (popis funkce uzlu, rozhraní atd.) Komunikace mezi uzly a funkce, rozhraní a protokoly jsou používány v průběhu signalizace (SIP SDP) a datový tok (RTCP RTP). Práce stručně popisuje základní toky hovorů, typy nosičů (GBR and N-GBR), a to vytvoření / mazaní nosičů během komunikace. Další část diplomové práce o implementaci volte, instalace a konfigurace IMS. Závěrečná část diplomové práce popisuje zkoušky sítě a, analýzu protokolu.The master's thesis describes VoLTE service, LTE evolution and deployment (deployment phases, actual LTE state and future perspectives etc.), EPC-IMS architecture (functional node description, interfaces etc.). Communications between nodes and functions, interfaces and protocols which are used during signaling (SIP-SDP) and data flow (RTCP RTP). Thesis briefly describe basic call flows, bearers types (GBR and N-GBR) and their establishment/delete during communication. The next part of master's thesis is about VoLTE implementation solutions, IMS installation and configuration. The final part of master's thesis describes the network and protocols tests, analyzes.
SIP based IP-telephony network security analysis
Masteroppgave i informasjons- og kommunikasjonsteknologi 2004 - Høgskolen i Agder, GrimstadThis thesis evaluates the SIP Protocol implementation used in the Voice over IP (VoIP) solution at
the fibre/DSL network of Èlla Kommunikasjon AS. The evaluation focuses on security in the
telephony service, and is performed from the perspective of an attacker trying to find weaknesses
in the network.
For each type of attempt by the malicious attacker, we examined the security level and possible
solutions to flaws in the system.
The conclusion of this analysis is that the VoIP service is exploitable, and that serious
improvements are needed to achieve a satisfying level of security for the system
Feasibility study of IP Multimedia Subsystem (IMS) based Push-to-Talk over Cellular (PoC) for public safety and security communications
Feasibility of IMS based PoC over public cellular network, specifically over commercial GPRS network, has been studied for public safety and security communication. Both day-to-day routine work and emergency operation handling capability of the PoC service as well as of the network have been considered. The requirements of PSS communication have been taken as the basis for analysing the technical viability of PoC service for PSS communication. The study is based on publicly available specification documents and related literature. The technical data and factual information have been collected from technical papers, reports and news articles posted into the Internet. The thesis also included a case study comparing over-aged analogue PMR systems and possible PoC service over a nationwide commercial GSM/GPRS network for PSS communication.
The key findings in favour of using PoC over commercial GPRS networks include already existed nation-wide radio coverage and roaming across the nation boarder, ease of use, special group call functionality, advanced IMS based multimedia services. The study has identified some challenges of using PoC for PSS communication, which are mainly due to the inherent latency of GPRS access network and lack of prioritisation of special groups of users in commercial networks. Though the PoC service over commercial network does not fulfil all the requirements strictly but taking into account the suggested measures, it can be a good alternative to PMR networks. Use of PoC for PSS communication requires zero initial as well as less operational expenses for the respective PSS organizations. At the same time PSS sector is a potential revenue generating market segment for the operator as well
Distributed resource discovery: architectures and applications in mobile networks
As the amount of digital information and services increases, it becomes increasingly important to be able to locate the desired content. The purpose of a resource discovery system is to allow available resources (information or services) to be located using a user-defined search criterion. This work studies distributed resource discovery systems that guarantee all existing resources to be found and allow a wide range of complex queries. Our goal is to allocate the load uniformly between the participating nodes, or alternatively to concentrate the load in the nodes with the highest available capacity.
The first part of the work examines the performance of various existing unstructured architectures and proposes new architectures that provide features especially valuable in mobile networks. To reduce the network traffic, we use indexing, which is particularly useful in scenarios, where searches are frequent compared to resource modifications. The ratio between the search and update frequencies determines the optimal level of indexing. Based on this observation, we develop an architecture that adjusts itself to changing network conditions and search behavior while maintaining optimal indexing. We also propose an architecture based on large-scale indexing that we later apply to resource sharing within a user group. Furthermore, we propose an architecture that relieves the topology constraints of the Parallel Index Clustering architecture. The performance of the architectures is evaluated using simulation.
In the second part of the work we apply the architectures to two types of mobile networks: cellular networks and ad hoc networks. In the cellular network, we first consider scenarios where multiple commercial operators provide a resource sharing service, and then a scenario where the users share resources without operator support. We evaluate the feasibility of the mobile peer-to-peer concept using user opinion surveys and technical performance studies. Based on user input we develop access control and group management algorithms for peer-to-peer networks. The technical evaluation is performed using prototype implementations. In particular, we examine whether the Session Initiation Protocol can be used for signaling in peer-to-peer networks. Finally, we study resource discovery in an ad hoc network. We observe that in an ad hoc network consisting of consumer devices, the capacity and mobility among nodes vary widely. We utilize this property in order to allocate the load to the high-capacity nodes, which serve lower-capacity nodes. We propose two methods for constructing a virtual backbone connecting the nodes
Prospects of peer-to-peer SIP for mobile operators
Tämän diplomityön tarkoituksena on esitellä kehitteillä oleva Peer-to-Peer Session Initiation Protocol (P2PSIP), jonka avulla käyttäjät voivat itsenäisesti ja helposti luoda keskenään puhe- ja muita multimediayhteyksiä vertaisverkko-tekniikan avulla. Lisäksi tarkoituksena on arvioida P2PSIP protokollan vaikutuksia ja mahdollisuuksia mobiilioperaattoreille, joille sitä voidaan pitää uhkana. Tästä huolimatta, P2PSIP:n ei ole kuitenkaan tarkoitus korvata nykyisiä puhelinverkkoja.
Työn alussa esittelemme SIP:n ja vertaisverkkojen (Peer-to-Peer) periaatteet, joihin P2PSIP-protokollan on suunniteltu perustuvan. SIP mahdollistaa multimedia-istuntojen luomisen, sulkemisen ja muokkaamisen verkossa, mutta sen monipuolinen käyttö vaatii keskitettyjen palvelimien käyttöä. Vertaisverkon avulla käyttäjät voivat suorittaa keskitettyjen palvelimien tehtävät keskenään hajautetusti. Tällöin voidaan ylläpitää laajojakin verkkoja tehokkaasti ilman palvelimista aiheutuvia ylläpito-kustannuksia.
Mobiilioperaattorit ovat haasteellisen tilanteen edessä, koska teleliikennemaailma on muuttumassa yhä avoimemmaksi. Tällöin operaattoreiden asiakkaille aukeaa mahdollisuuksia käyttää kilpailevia Internet-palveluja (kuten Skype) helpommin ja tulevaisuudessa myös itse muodostamaan kommunikointiverkkoja P2PSIP:n avulla.
Tutkimukset osoittavat, että näistä uhista huolimatta myös operaattorit pystyvät näkemään P2PSIP:n mahdollisuutena mukautumisessa nopeasti muuttuvan teleliikennemaailman haasteisiin. Nämä mahdollisuudet sisältävät operaattorin oman verkon optimoinnin lisäksi vaihtoehtoisten ja monipuolisempien palveluiden tarjoamisen asiakkailleen edullisesti. Täytyy kuitenkin muistaa, että näiden mahdollisuuksien toteuttamisten vaikutusten ei tulisi olla ristiriidassa operaattorin muiden palveluiden kanssa. Lisäksi tulisi muistaa, että tällä hetkellä keskeneräisen P2PSIP-standardin lopullinen luonne ja ominaisuudet voivat muuttaa sen vaikutuksia.The purpose of this thesis is to present the Peer-to-Peer Session Initiation Protocol (P2PSIP) being developed. In addition, the purpose of this thesis is to evaluate the impacts and prospects of P2PSIP to mobile operators, to whom it can be regarded as a threat. In P2PSIP, users can independently and easily establish voice and other multimedia connections using peer-to-peer (P2P) networking. However, P2PSIP is not meant to replace the existing telephony networks of the operators.
We start by introducing the principles of SIP and P2P networking that the P2PSIP is intended to use. SIP enables to establish, terminate and modify multimedia sessions, but its versatile exploitation requires using centralized servers. By using P2P networking, users can decentralize the functions of centralized servers by performing them among themselves. This enables to maintain large and robust networks without maintenance costs resulted of running such centralized servers.
Telecommunications market is transforming to a more open environment, where mobile operators and other service providers are challenged to adapt to the upcoming changes. Subscribers have easier access to rivalling Internet-services (such as Skype) and in future they can form their own communication communities by using P2PSIP.
The results show that despite of these threats, telecom operators can find potential from P2PSIP in concurrence in adaptation to the challenges of the rapidly changing telecom environment. These potential roles include optimization of the network of the operator, but as well roles to provide alternative and more versatile services to their subscribers at low cost. However, the usage of P2PSIP should not conflict with the other services of the operator. Also, as P2PSIP is still under development, its final nature and features may change its impacts and prospects