Delivering Live Multimedia Streams to Mobile Hosts in a Wireless Internet with Multiple Content Aggregators

We consider the distribution of channels of live multimedia content (e.g., radio or TV broadcasts) via multiple content aggregators. In our work, an aggregator receives channels from content sources and redistributes them to a potentially large number of mobile hosts. Each aggregator can offer a channel in various configurations to cater for different wireless links, mobile hosts, and user preferences. As a result, a mobile host can generally choose from different configurations of the same channel offered by multiple alternative aggregators, which may be available through different interfaces (e.g., in a hotspot). A mobile host may need to handoff to another aggregator once it receives a channel. To prevent service disruption, a mobile host may for instance need to handoff to another aggregator when it leaves the subnets that make up its current aggregator�s service area (e.g., a hotspot or a cellular network).\ud In this paper, we present the design of a system that enables (multi-homed) mobile hosts to seamlessly handoff from one aggregator to another so that they can continue to receive a channel wherever they go. We concentrate on handoffs between aggregators as a result of a mobile host crossing a subnet boundary. As part of the system, we discuss a lightweight application-level protocol that enables mobile hosts to select the aggregator that provides the �best� configuration of a channel. The protocol comes into play when a mobile host begins to receive a channel and when it crosses a subnet boundary while receiving the channel. We show how our protocol can be implemented using the standard IETF session control and description protocols SIP and SDP. The implementation combines SIP and SDP�s offer-answer model in a novel way

De-ossifying the Internet Transport Layer : A Survey and Future Perspectives

ACKNOWLEDGMENT The authors would like to thank the anonymous reviewers for their useful suggestions and comments.Peer reviewedPublisher PD

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

Efficient User Controlled Inter-Domain SIP Mobility: Authentication, Registration, and Call Routing

Over the past decade, multimedia services have gained significant acceptance and played an important role in the convergence of IP networks. Supporting mobility in IP (Internet Protocol) networks is a crucial step towards satisfying the nomadic communication paradigms on the current Internet. The Session Initiation Protocol (SIP) presents one approach towards supporting IP mobility. Additionally, SIP is increasingly gaining in popularity as the next generation multimedia signaling and session establishment protocol. It is anticipated that the SIP infrastructure will be extensively deployed all over the Internet. In this paper, we explore an efficient approach to inter-domain SIP mobility in an attempt to improve personal and terminal mobility schemes. We succeed in applying a persistent identification framework to application level SIP addressing by introducing a level of indirection on top of the traditional SIP architecture. We refer to our approach as the Handle SIP (H-SIP). H-SIP leverages the current SIP architecture abstracting any domain binding from users. Our approach to mobility is user-controlled. We experimentally prove the efficiency of H-SIP in achieving inter-domain authentication and call routing through modeling and real-time measurements

Linking session based services with transport plane resources in IP multimedia subsystems.

The massive success and proliferation of Internet technologies has forced network operators to recognise the benefits of an IP-based communications framework. The IP Multimedia Subsystem (IMS) has been proposed as a candidate technology to provide a non-disruptive strategy in the move to all-IP and to facilitate the true convergence of data and real-time multimedia services. Despite the obvious advantages of creating a controlled environment for deploying IP services, and hence increasing the value of the telco bundle, there are several challenges that face IMS deployment. The most critical is that posed by the widespread proliferation ofWeb 2.0 services. This environment is not seen as robust enough to be used by network operators for revenue generating services. However IMS operators will need to justify charging for services that are typically available free of charge in the Internet space. Reliability and guaranteed transport of multimedia services by the efficient management of resources will be critical to differentiate IMS services. This thesis investigates resource management within the IMS framework. The standardisation of NGN/IMS resource management frameworks has been fragmented, resulting in weak functional and interface specifications. To facilitate more coherent, focused research and address interoperability concerns that could hamper deployment, a Common Policy and Charging Control (PCC) architecture is presented that defines a set of generic terms and functional elements. A review of related literature and standardisation reveals severe shortcomings regarding vertical and horizontal coordination of resources in the IMS framework. The deployment of new services should not require QoS standardisation or network upgrade, though in the current architecture advanced multimedia services are not catered for. It has been found that end-to-end QoS mechanisms in the Common PCC framework are elementary. To address these challenges and assist network operators when formulating their iii NGN strategies, this thesis proposes an application driven policy control architecture that incorporates end-user and service requirements into the QoS negotiation procedure. This architecture facilitates full interaction between service control and resource control planes, and between application developers and the policies that govern resource control. Furthermore, a novel, session based end-to-end policy control architecture is proposed to support inter-domain coordination across IMS domains. This architecture uses SIP inherent routing information to discover the routes traversed by the signalling and the associated routes traversed by the media. This mechanism effectively allows applications to issue resource requests from their home domain and enable end-to-end QoS connectivity across all traversed transport segments. Standard interfaces are used and transport plane overhaul is not necessary for this functionality. The Common PCC, application driven and session based end-to-end architectures are implemented in a standards compliant and entirely open source practical testbed. This demonstrates proof of concept and provides a platform for performance evaluations. It has been found that while there is a cost in delay and traffic overhead when implementing the complete architecture, this cost falls within established criteria and will have an acceptable effect on end-user experience. The open nature of the practical testbed ensures that all evaluations are fully reproducible and provides a convenient point of departure for future work. While it is important to leave room for flexibility and vendor innovation, it is critical that the harmonisation of NGN/IMS resource management frameworks takes place and that the architectures proposed in this thesis be further developed and integrated into the single set of specifications. The alternative is general interoperability issues that could render end-to-end QoS provisioning for advanced multimedia services almost impossible

Roaming Real-Time Applications - Mobility Services in IPv6 Networks

Emerging mobility standards within the next generation Internet Protocol, IPv6, promise to continuously operate devices roaming between IP networks. Associated with the paradigm of ubiquitous computing and communication, network technology is on the spot to deliver voice and videoconferencing as a standard internet solution. However, current roaming procedures are too slow, to remain seamless for real-time applications. Multicast mobility still waits for a convincing design. This paper investigates the temporal behaviour of mobile IPv6 with dedicated focus on topological impacts. Extending the hierarchical mobile IPv6 approach we suggest protocol improvements for a continuous handover, which may serve bidirectional multicast communication, as well. Along this line a multicast mobility concept is introduced as a service for clients and sources, as they are of dedicated importance in multipoint conferencing applications. The mechanisms introduced do not rely on assumptions of any specific multicast routing protocol in use.Comment: 15 pages, 5 figure

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

A Comprehensive Survey of Voice over IP Security Research

We present a comprehensive survey of Voice over IP security academic research, using a set of 245 publications forming a closed cross-citation set. We classify these papers according to an extended version of the VoIP Security Alliance (VoIPSA) Threat Taxonomy. Our goal is to provide a roadmap for researchers seeking to understand existing capabilities and to identify gaps in addressing the numerous threats and vulnerabilities present in VoIP systems. We discuss the implications of our findings with respect to vulnerabilities reported in a variety of VoIP products. We identify two specific problem areas (denial of service, and service abuse) as requiring significant more attention from the research community. We also find that the overwhelming majority of the surveyed work takes a black box view of VoIP systems that avoids examining their internal structure and implementation. Such an approach may miss the mark in terms of addressing the main sources of vulnerabilities, i.e., implementation bugs and misconfigurations. Finally, we argue for further work on understanding cross-protocol and cross-mechanism vulnerabilities (emergent properties), which are the byproduct of a highly complex system-of-systems and an indication of the issues in future large-scale systems