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

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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

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

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

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

Unified Messaging using SIP and RTSP

Traditional answering machines and voice mail services are closed systems, tightly coupled to a single end system,the local PBX or local exchange carrier. Even simple services, such as forwarding voice mail to another user outside the local system, are hard to provide. With the advent of Internet telephony, we need to provide voice and video mail services. This also offers the opportunity to address some of the shortcomings of existing voice mail systems. We list general requirements for a multimedia mail system for Internet telephony. We then propose an architecture using SIP (Session Initiation Protocol) and RTSP (Real-Time Streaming Protocol) and compare various alternative approaches to solving call forwarding, reclaiming and retrieval of messages. We also briefly describe our prototype implementation

On the development of Voice over IP

This record of study documents the experience acquired during my internship at Sonus Networks, Inc. for the Doctor of Engineering Program. In this record of study, I have surveyed and analyzed the current standardization status of Voice over Internet Protocol (VoIP) security and proposed an Internet draft on secure retargeting and response identity. The draft provides a simple and comprehensive solution to the response identity, call recipient identity and intermediate server retargeting problems in the Session Initiation Protocol (SIP) call setup process. To support product line development and enable product evolution in the quickly growing VoIP market, I have proposed a generic development framework for SIP application servers. The common and open architecture of the framework supports multiple products development and facilitates integration of new service modules. The systematical reuse of proven software design and implementation enables companies to reduce the development cost and shorten the time-to-market. As the development and diffusion of VoIP can never be isolated from the social sphere, I have investigated the current status, influence and interaction of three most important factors: standardization, market forces and government regulation on the development and diffusion of VoIP. The worldwide deregulation and market privatization have caused the transition of the standards development model. This transition in turn influences the market diffusion. Other than standardization, market forces including customer needs, the revenue pressure on carriers and vendors, competitive and economic environment, social culture and regulation uncertainties create both threats and opportunities. I have examined market drivers and obstacles in the current VoIP adoption stage, analyzed current VoIP market players and their strategies, and predicted the direction of VoIP business. The regulation creates the macro environment in which VoIP develops and diffuses. I have explored modern telecommunications regulation principles based on which government makes decisions on most current issues, including 911 support, mergers and acquisitions, interconnection obligation and leasing rights, rate structure and universal service fees

Developing a cross platform IMS client using the JAIN SIP applet phone

Since the introduction of the IP Multimedia Subsystem (IMS) by the Third Generation Partnership Project (3GPP) in 2002, a lot of research has been conducted aimed at designing and implementing IMS capable clients and network elements. Though considerable work has been done in the development of IMS clients, there is no single, free and open source IMS client that provides researchers with all the required functionality needed to test the applications they are developing. For example, several open and closed source SIP/IMS clients are used within the Rhodes University Conver- gence Research Group (RUCRG) to test applications under development, as a result of the fact that the various SIP/IMS clients support different subsets of SIP/IMS features. The lack of a single client and the subsequent use of various clients comes with several problems. Researchers have to know how to deploy, configure, use and at times adapt the various clients to suit their needs. This can be very time consuming and, in fact, contradicts the IMS philosophy (the IMS was proposed to support rapid service creation). This thesis outlines the development of a Java-based, IMS compliant client called RUCRG IMS client, that uses the JAIN SIP Applet Phone (JSAP) as its foundation. JSAP, which originally offered only basic voice calling and instant messaging (IM) capabilities, was modified to be IMS compliant and support video calls, IM and presence using XML Configuration Access Protocol (XCAP)

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

A Decentralized Session Management Framework for Heterogeneous Ad-Hoc and Fixed Networks

Wireless technologies are continuously evolving. Second generation cellular networks have gained worldwide acceptance. Wireless LANs are commonly deployed in corporations or university campuses, and their diffusion in public hotspots is growing. Third generation cellular systems are yet to affirm everywhere; still, there is an impressive amount of research ongoing for deploying beyond 3G systems. These new wireless technologies combine the characteristics of WLAN based and cellular networks to provide increased bandwidth. The common direction where all the efforts in wireless technologies are headed is towards an IP-based communication. Telephony services have been the killer application for cellular systems; their evolution to packet-switched networks is a natural path. Effective IP telephony signaling protocols, such as the Session Initiation Protocol (SIP) and the H 323 protocol are needed to establish IP-based telephony sessions. However, IP telephony is just one service example of IP-based communication. IP-based multimedia sessions are expected to become popular and offer a wider range of communication capabilities than pure telephony. In order to conjoin the advances of the future wireless technologies with the potential of IP-based multimedia communication, the next step would be to obtain ubiquitous communication capabilities. According to this vision, people must be able to communicate also when no support from an infrastructured network is available, needed or desired. In order to achieve ubiquitous communication, end devices must integrate all the capabilities necessary for IP-based distributed and decentralized communication. Such capabilities are currently missing. For example, it is not possible to utilize native IP telephony signaling protocols in a totally decentralized way. This dissertation presents a solution for deploying the SIP protocol in a decentralized fashion without support of infrastructure servers. The proposed solution is mainly designed to fit the needs of decentralized mobile environments, and can be applied to small scale ad-hoc networks or also bigger networks with hundreds of nodes. A framework allowing discovery of SIP users in ad-hoc networks and the establishment of SIP sessions among them, in a fully distributed and secure way, is described and evaluated. Security support allows ad-hoc users to authenticate the sender of a message, and to verify the integrity of a received message. The distributed session management framework has been extended in order to achieve interoperability with the Internet, and the native Internet applications. With limited extensions to the SIP protocol, we have designed and experimentally validated a SIP gateway allowing SIP signaling between ad-hoc networks with private addressing space and native SIP applications in the Internet. The design is completed by an application level relay that permits instant messaging sessions to be established in heterogeneous environments. The resulting framework constitutes a flexible and effective approach for the pervasive deployment of real time applications.The invention of the phone has radically changed the way people communicate, as it allowed persons to get in contact instantly no matter of their location. However, phone communication has been confined for decades to a fixed location, be it one's own house or a phone boot. The widespread affirmation of cellular technologies has had for fixed telephony a similar impact that the invention of the phone has had on communications years before. With mobile phones, people are enabled to talk with each other anytime and anywhere. Internet has also revolutionized the way people communicate. E-mails have soon become one of the Internet killer applications. Later on, instant messaging, popularly known as chatting, has gained huge consensus among net surfers. Only recently, the use of the Internet for voice communication is becoming mainstream, and the so called Voice over IP (VoIP) applications (Skype is probably the most famous for the masses) are becoming common use. Despite its popularity, Internet still suffers from the inherent limitations that affected early telephony: it is fixed. The usage of Internet on the move still does not constitute the easiest and most satisfactory user experience, due to capabilities and limitations of the access technology, terminals, services and applications. Efforts for mobilizing the Internet are ongoing both in the industrial and in the academic worlds, but several bricks are needed to build the wall of mobile Internet. This dissertation provides one of these bricks, describing a solution that allows the deployment of multimedia applications (chat, VoIP, gaming) in mobile environments. In other words, this dissertation gives solutions for facilitating ubiquitous Internet-based communication, anytime and anywhere. The vision that we want to become true is that Internet must become mobile in the same way as fixed telephony has become mobile thanks to the cellular technology. More than this, we do not want that users are limited by the presence of an infrastructure to communicate with each other. In order to achieve this, we present solutions to deploy Internet-based services and applications in environments where no support from servers is available. In other words, we enable direct device-to-device, user-to-user Internet communication. Our contribution is mainly focused on the steps needed to establish the communication, the so called session establishment or signaling phase. We have validated our signaling framework by building a chat application that utilizes its features and works in server-less environments. The custom server-less solution does not prohibit to connect at the same time with the Internet, so that one can engage in a chess game using direct communication with a person in the proximity while having a chat in progress with a friend using standard Internet services. The challenge that we had to face is that Internet services and applications are usually built implying support from a centralized server. In order to deploy direct user-to-user Internet services, while maintaining interoperability with mainstream services, we had to enhance native Internet services to work without infrastructure support, without sacrificing interoperability with standard Internet applications. To conclude, we have placed our brick on the still yet to be completed wall of mobile Internet. Our hope is that one day, thanks also to this brick, everybody will be able to enjoy Internet-based applications as easily as now it is possible to use mobile telephony services