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

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

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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

Push-to-Talk över Bluetooth

Push-to-Talk over Cellular (PoC) är en teknologi som möjliggör en radiotelefonlik service över GPRS vilken väckt ökande popularitet. I skrivande stund pågår specifiering av en öppen PoC-standard inom Open Mobile Alliance (OMA). OMA planerar att baser PoC på en IP/UDP/RTP protokollstack samt en server-clientarkitektur. Systemet utnyttjar även SIP-signaleringsegenskaperna hos IP Multimedia Subsystem (IMS). PoC-nätelement handhar bl.a. gruppförvaltning och taltursfördelning. Forskningsproblemet för denna avhandling är: "Hur kan en PoC-liknande service erbjudas gratis åt mobiltelefonsanvändare med hjälp av Bluetooth-teknologi?" Den primära målsättningen för detta arbete är därmed att skissa upp ett förslag för hur man kunde utveckla en Push-to-Talk (PTT)-funktion som utnyttjar ett Bluetooth scatternet-nät samt PAN-profilen för att överföra data. En måttlig räckvidd kan uppnås med hjälp av Bluetooth apparater av effektklass 1 vars räckvidd kan vara t.o.m. 100 m. En sekundär målsättning är att beskriva PoC samt de protokoll PoC utnyttjar (t.ex. SIP och SDP). Denna beskrivning utgör både en utgångspunkt för att uppnå den primära målsättningen och erbjuder även en introduktion till OMA PoC som lämpar sig för både studeranden och yrkesmän. Det uppskissade förslaget för Push-to-Talk över Bluetooth (PoB) innefattar metoder för skapande av grupper och nät, dataöverföring samt taltursfördelning. Metoden för nätskapande (som kan vara användbar även för andra ändamål) baserar sig på att skapa ett scatternet emellan apparater som tillhör en på förhand specifierad grupp av apparater samt på att undvika slingor. Detta möjliggör enkel kommunikation genom att skicka data till alla apparater inom nätet, förutsatt att de apparater som sammanbinder piconet-näten till ett scatternet fungerar som repeterare. Ytterligare uppskissas en metod för att kombinera PoB och PoC. Avsikten med detta är att möjliggöra PTT-kommunikation med både lokalt och avlägset belägna gruppmedlemmar med hjälp av Bluetooth respektive GPRS.Push-to-Talk over Cellular (PoC) is an emerging technology enabling a walkie-talkie-like service over GPRS. At the time of writing, an open standard for PoC is being specified by the Open Mobile Alliance (OMA). As specified by the OMA standard drafts, PoC is based on an IP/UDP/RTP protocol stack and a client-server based architecture. The systems exploits the SIP signalling capabilities of the the IP Multimedia Subsystem (IMS). Group management, floor control etc are administered by the network elements of PoC. The research problem of this thesis is: "How can mobile phone users be provided with a free-of-charge PTT-feature with PoC-like user experience by means of Bluetooth technology?" The primary objective of the study is thus to propose an outline for developing a Push-to-Talk (PTT) feature that utilizes a Bluetooth scatternet and the PAN profile for data communications. A reasonable range can be obtained with Bluetooth class 1 devices, which provide a range of up to 100 m. A subsidiary objective is to provide a description of OMA PoC and the protocols it relies upon. The description serves both as a basis for pursuing the primary objective and as a tutorial, which is suitable for students or professionals desiring to acquaint themselves with OMA PoC. The proposed outline for Push-to-Talk over Bluetooth (PoB) comprises e.g. methods for group formation, network formation, communication, and floor control. The network formation method, which can be utilized in other applications as well, is based on creating a scatternet among a predefined set of devices and on avoiding loops. This approach enables usage of a simple broadcasting based communication method, in which the devices bridging the piconets into a scatternet act as repeaters. A method for combining PoB and PoC is also outlined. It is intended for enabling PTT-communication with both local and distant group members over Bluetooth and GPRS respectively

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

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

Tutkielma ryhmitellyistä konferensseista ja Binary Floor Control Protocol:n toteutuksesta keskitettyyn konferenssijärjestelmään

The introduction of the third generation (3G) in the mobile telecommunication world offers the possibility for a wide range of new applications and services that operators can offer to their customers. One of these services is multimedia conferencing. There is ongoing work to provide conferencing services in the IP Multimedia Subsystem (IMS) environment as one of the most significant services. This thesis focuses on providing a comprehensive overview of conferencing systems, especially of the Binary Floor Control Protocol (BFCP) and cascade conferences. The Master's Thesis consisted of two parts: The first part is a theoretical part, which provides the concepts of the centralized conferencing, known as tightly coupled conferences, and reviews the current specifications stage of the different standardization bodies. In contrast, the study of the applicability of the current centralized conferencing specifications in a cascaded conferencing environment is presented, as well as the strengths and weaknesses of them. The second part is a practical implementation of the Binary Floor Control Protocol (BFCP). BFCP is implemented in MiniSip, an existing secure open-source SIP User Agent (UA), and in Asterisk, an open source Private Branch Exchange (PBX) replacement system. BFCP is built using the specification defined by the XCON working group within the Internet Engineering Task Force (IETF). Finally, BFCP is evaluated and based on this evaluation, some conclusions are given.Kolmannen sukupolven matkapuhelinverkot mahdollistavat laajan uusien ohjelmien ja palveluiden kirjon, joita operaattorit voivat tarjota asiakkailleen. Eräs tämänlainen palvelu on multimedia konferenssi. Tällä hetkellä tehdään työtä, jonka tarkoituksena on mahdollistaa konferenssinpalvelun tarjoaminen IP Multimedia Subsystem (IMS) ympäristössä. Tämä diplomityö keskittyy konferenssijärjestelmän perusteelliseen kuvaukseen, painottuen Binary Floor Control Protocol:aan (BFCP) sekä ryhmiteltyihin konferensseihin. Työ koostuu kahdesta osasta: Ensimmäinen osa keskittyy teoriaan, joka käsittelee keskitettyjä konferenssijärjestelmiä sekä aiheen nykyistä tilaa eri standardointiorganisaatioissa. Vastakohtana tarkastellaan nykyisen keskitetyn konferenssijärjestelmän heikkouksia ja vahvuuksia. Toinen osa käsittelee käytännön toteutusta BFCP:sta, joka on toteutettu MiniSip- sekä Asterisk-ohjelmistoihin. MiniSip on avoimeen lähdekoodiin perustuva SIP käyttäjäagentti, ja Asterisk paikallisvaihdeohjelmiston (PBX) avoin korvaaja. BFCP perustuu spesifikaatioon, jonka on määritellyt XCON työryhmä IETF:ssa. Lopuksi BFCP protokollaa on arvioitu tämän toteutuksen avulla

Convergence: the next big step

Recently, web based multimedia services have gained popularity and have proven themselves to be viable means of communication. This has inspired the telecommunication service providers and network operators to reinvent themselves to try and provide value added IP centric services. There was need for a system which would allow new services to be introduced rapidly with reduced capital expense (CAPEX) and operational expense (OPEX) through increased efficiency in network utilization. Various organizations and standardization agencies have been working together to establish such a system. Internet Protocol Multimedia Subsystem (IMS) is a result of these efforts. IMS is an application level system. It is being developed by 3GPP (3rd Generation Partnership Project) and 3GPP2 (3rd Generation Partnership Project 2) in collaboration with IETF (Internet Engineering Task Force), ITU-T (International Telecommunication Union – Telecommunication Standardization Sector), and ETSI (European Telecommunications Standards Institute) etc. Initially, the main aim of IMS was to bring together the internet and the cellular world, but it has extended to include traditional wire line telecommunication systems as well. It utilizes existing internet protocols such as SIP (Session Initiation Protocol), AAA (Authentication, Authorization and Accounting protocol), and COPS (Common Open Policy Service) etc, and modifies them to meet the stringent requirements of reliable, real time communication systems. The advantages of IMS include easy service quality management (QoS), mobility management, service control and integration. At present a lot of attention is being paid to providing bundled up services in the home environment. Service providers have been successful in providing traditional telephony, high speed internet and cable services in a single package. But there is very little integration among these services. IMS can provide a way to integrate them as well as extend the possibility of various other services to be added to allow increased automation in the home environment. This thesis extends the concept of IMS to provide convergence and facilitate internetworking of the various bundled services available in the home environment; this may include but is not limited to communications (wired and wireless), entertainment, security etc. In this thesis, I present a converged home environment which has a number of elements providing a variety of communication and entertainment services. The proposed network would allow effective interworking of these elements, based on IMS architecture. My aim is to depict the possible advantages of using IMS to provide convergence, automation and integration at the residential level

Adaptive Voice Applications over Delay Tolerant Networks

Internet is predominantly based on best effort packet transmission. Performance of applications over internet suffers due to disconnections, delays, losses and dynamic nature of elements in the network. Voice communications, such as Voice over Internet Protocols (VoIP), over mobile networks has to deal with technical barriers such as delays and temporary disconnections. Delay tolerant networks provides communication based on asynchronous messaging that deals with delays and disconnections; which provides a mechanism to deliver the messages irrespective of instantaneous end-to-end path connectivity. In the thesis, delay tolerant adaptive media is proposed to allow DTN-based communication as a fall-back if real time end-to-end voice communication fails. We designed a system which adapts to delays and losses by switching between RTP/UDP and RTP/DTN-based voice packets transmission. The real time communication works fine as long as continuous end-to-end path exists. The continuous path might not exist when there are changes in the network topology of mobile users. So in the case of non-availability of end-to-end path, we swiftly adapt to RTP/DTN-based voice with variable length messaging mechanism. To assess the call quality in different modes of operation, we used R values of E model specified by ITU-T. The results show that the proposed delay tolerant adaptive media for adaptive voice over delay tolerant networks achieves better utility for the users when end-to-end connectivity is not available or when delays are higher

Voice Communication in Mobile Delay-Tolerant Networks

Push-to-talk (PTT) is one class of voice communication system generally employed in cellular phone services. Today's PTT services mainly rely on infrastructure and require stable end-to-end path for successful communication. But users with PTT enabled mobile devices may travel in challenged environments where infrastructure is not available or end-to-end path is highly unreliable. In such cases those PTT services may exhibit poor performance or may even fail completely. Even though some existing PTT solutions allow users to communicate in an ad-hoc fashion, they need sufficient node density to establish end-to-end path and eventually fail to communicate in sparse mobile ad-hoc environments. Delay-Tolerant Networking (DTN) is an emerging research area that addresses the communication requirements specfic to challenged networks. In this thesis we develop a voice communication system (DT-Talkie) which enables both individual and group users to communicate over infrastructure-less and challenged networks in the walkie-talkie fashion. The DTN concept of asynchronous message forwarding is applied to the DT-Talkie in order to transmit voice messages reliably. We employ variable-length fragmentation mechanism in the application layer with the vision to speed-up session interactivity in stable scenarios. Some approaches to resolve codec interoperability issues are implied in this thesis. To validate the concepts of the DT-Talkie, we implement an application for Maemo based Nokia Internet Tablets, leveraging the DTN reference implementation developed in the DTN Research Group. Moreover in this thesis we evaluate the performance of the DT-Talkie through conducting a set of simulations using several DTN routing protocols and using different mobility models