Designing and prototyping WebRTC and IMS integration using open source tools

WebRTC, or Web Real-time Communications, is a collection of web standards that detail the mechanisms, architectures and protocols that work together to deliver real-time multimedia services to the web browser. It represents a significant shift from the historical approach of using browser plugins, which over time, have proven cumbersome and problematic. Furthermore, it adopts various Internet standards in areas such as identity management, peer-to-peer connectivity, data exchange and media encoding, to provide a system that is truly open and interoperable. Given that WebRTC enables the delivery of multimedia content to any Internet Protocol (IP)-enabled device capable of hosting a web browser, this technology could potentially be used and deployed over millions of smartphones, tablets and personal computers worldwide. This service and device convergence remains an important goal of telecommunication network operators who seek to enable it through a converged network that is based on the IP Multimedia Subsystem (IMS). IMS is an IP-based subsystem that sits at the core of a modern telecommunication network and acts as the main routing substrate for media services and applications such as those that WebRTC realises. The combination of WebRTC and IMS represents an attractive coupling, and as such, a protracted investigation could help to answer important questions around the technical challenges that are involved in their integration, and the merits of various design alternatives that present themselves. This thesis is the result of such an investigation and culminates in the presentation of a detailed architectural model that is validated with a prototypical implementation in an open source testbed. The model is built on six requirements which emerge from an analysis of the literature, including previous interventions in IMS networks and a key technical report on design alternatives. Furthermore, this thesis argues that the client architecture requires support for web-oriented signalling, identity and call handling techniques leading to a potential for IMS networks to natively support these techniques as operator networks continue to grow and develop. The proposed model advocates the use of SIP over WebSockets for signalling and DTLS-SRTP for media to enable one-to-one communication and can be extended through additional functions resulting in a modular architecture. The model was implemented using open source tools which were assembled to create an experimental network testbed, and tests were conducted demonstrating successful cross domain communications under various conditions. The thesis has a strong focus on enabling ordinary software developers to assemble a prototypical network such as the one that was assembled and aims to enable experimentation in application use cases for integrated environments

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)

Towards a scalable video interactivity solution over the IMS

Includes bibliographical references (leaves 72-76).Rapid increase in bandwidth and the interactive and scalability features of the Internet provide a precedent for a converged platform that will support interactive television. Next Generation Network platforms such as the IP Multimedia Subsystem (IMS) support Quality of Service (QoS), fair charging and possible integration with other services for the deployment of IPTV services. IMS architecture supports the use of the Session Initiation Protocol (SIP) for session control and the Real Time Streaming Protocol (RTSP) for media control. This study aims to investigate video interactivity designs over the Internet using an evaluation framework to examine the performance of both SIP and RTSP protocols over the IMS over different access networks. It proposes a Three Layered Video Interactivity Framework (TLVIF) to reduce the video processing load on a server

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

Security Enhancements in Voice Over Ip Networks

Voice delivery over IP networks including VoIP (Voice over IP) and VoLTE (Voice over LTE) are emerging as the alternatives to the conventional public telephony networks. With the growing number of subscribers and the global integration of 4/5G by operations, VoIP/VoLTE as the only option for voice delivery becomes an attractive target to be abused and exploited by malicious attackers. This dissertation aims to address some of the security challenges in VoIP/VoLTE. When we examine the past events to identify trends and changes in attacking strategies, we find that spam calls, caller-ID spoofing, and DoS attacks are the most imminent threats to VoIP deployments. Compared to email spam, voice spam will be much more obnoxious and time consuming nuisance for human subscribers to filter out. Since the threat of voice spam could become as serious as email spam, we first focus on spam detection and propose a content-based approach to protect telephone subscribers\u27 voice mailboxes from voice spam. Caller-ID has long been used to enable the callee parties know who is calling, verify his identity for authentication and his physical location for emergency services. VoIP and other packet switched networks such as all-IP Long Term Evolution (LTE) network provide flexibility that helps subscribers to use arbitrary caller-ID. Moreover, interconnecting between IP telephony and other Circuit-Switched (CS) legacy telephone networks has also weakened the security of caller-ID systems. We observe that the determination of true identity of a calling device helps us in preventing many VoIP attacks, such as caller-ID spoofing, spamming and call flooding attacks. This motivates us to take a very different approach to the VoIP problems and attempt to answer a fundamental question: is it possible to know the type of a device a subscriber uses to originate a call? By exploiting the impreciseness of the codec sampling rate in the caller\u27s RTP streams, we propose a fuzzy rule-based system to remotely identify calling devices. Finally, we propose a caller-ID based public key infrastructure for VoIP and VoLTE that provides signature generation at the calling party side as well as signature verification at the callee party side. The proposed signature can be used as caller-ID trust to prevent caller-ID spoofing and unsolicited calls. Our approach is based on the identity-based cryptography, and it also leverages the Domain Name System (DNS) and proxy servers in the VoIP architecture, as well as the Home Subscriber Server (HSS) and Call Session Control Function (CSCF) in the IP Multimedia Subsystem (IMS) architecture. Using OPNET, we then develop a comprehensive simulation testbed for the evaluation of our proposed infrastructure. Our simulation results show that the average call setup delays induced by our infrastructure are hardly noticeable by telephony subscribers and the extra signaling overhead is negligible. Therefore, our proposed infrastructure can be adopted to widely verify caller-ID in telephony networks

A Cross Domain Next Generation Network IPTV Client for Media Center environments

Functions, which can be summarized to the keyword Internet Protocol Television (IPTV) describe the transmission of video services to users via Internet Protocol (IP). Accompanying to this new television transmission path Home Theatre PCs (HTPC) running a so called Media Center platform are more and more entering the living rooms as a companion for the popular LCD and Plasma displays. Perfect ease of use and the visual integration on the screen and also into the living room is raising their acceptance. These HTPCs are a central node for multimedia services such as TV, radio and email within the networked household. Thus, there are good preconditions for the use of a HTPC as end device for Telco operator driven IPTV and telecommunication services. In the context of this diploma thesis possibilities for the provisioning of IPTV and Next Generation Network (NGN) services on a converged multimedia home entertainment platform for the living room will be investigated, especially Vista Media Center platforms. For this reason, standardization activities will be investigated, which deal with the integration of IPTV and telecommunication services into NGN. The validation of the results will be achieved by the design and implementation of a Vista Media Center Add-In, which can be integrated as an IP Multimedia Subsystem (IMS) based User Agent (UA) in ETSI TISPAN Release 2 IPTV infrastructures. Additionally, a Cross Domain messaging service for IMS based UA is created, which enables a cross-network communication between users

Converged multimedia services in emerging Web 2.0 session mobility scenarios

n/

Creation of value with open source software in the telecommunications field

Tese de doutoramento. Engenharia Electrotécnica e de Computadores. Faculdade de Engenharia. Universidade do Porto. 200