Vulnerabilities of signaling system number 7 (SS7) to cyber attacks and how to mitigate against these vulnerabilities.

As the mobile network subscriber base exponentially increases due to some attractive offerings such as anytime anywhere accessibility, seamless roaming, inexpensive handsets with sophisticated applications, and Internet connectivity, the mobile telecommunications network has now become the primary source of communication for not only business and pleasure, but also for the many life and mission critical services. This mass popularisation of telecommunications services has resulted in a heavily loaded Signaling System number 7 (SS7) signaling network which is used in Second and Third Generations (2G and 3G) mobile networks and is needed for call control and services such as caller identity, roaming, and for sending short message servirces. SS7 signaling has enjoyed remarkable popularity for providing acceptable voice quality with negligible connection delays, pos- sibly due to its circuit-switched heritage. However, the traditional SS7 networks are expensive to lease and to expand, hence to cater for the growing signaling demand and to provide the seamless interconnectivity between the SS7 and IP networks a new suite of protocols known as Signaling Transport (SIGTRAN) has been designed to carry SS7 signaling messages over IP. Due to the intersignaling between the circuit-switched and the packet-switched networks, the mo- bile networks have now left the “walled garden”, which is a privileged, closed and isolated ecosystem under the full control of mobile carriers, using proprietary protocols and has minimal security risks due to restricted user access. Potentially, intersignaling can be exploited from the IP side to disrupt the services provided on the circuit-switched side. This study demonstrates the vulnerabilities of SS7 messages to cyber-attacks while being trans- ported over IP networks and proposes some solutions based on securing both the IP transport and SCTP layers of the SIGTRAN protocol stack

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

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

An investigation into intelligent network congestion control strategies

This thesis examines the congestion control issues that arise in Intelligent Networks, when it is necessary to support multiple service types with different load requirements and priorities. The area of Intelligent Network (IN) congestion control has been under investigation for over a decade, but in general, the models used in this research were over-simplified and all service types were assumed to have the same priority levels and load requirements at the various IN physical elements. However, as the IN is a dynamic network that must process many different service types that have radically different call load profiles and are based on different service level agreements and charging schemes, the validity of the above assumptions is questionable. The aim of this work, therefore, is to remove a number of the classic assumptions made in IN congestion control research, by: • developing a detailed model of an IN, catering for multiple traffic types, • using this model to establish the shortcomings of classic congestion control strategies, • devising a new IN congestion control strategy and verifying its superiority on the model. To achieve these aims, an IN model (both simulation and analytic) is developed to reflect the physical and functional architecture of the network and model the information flows required between network entities in order to execute services. The effectiveness of various classic active and reactive congestion control strategies are then investigated using this model and it is established that none of these strategies are capable of protecting both the Service Control Point and Service Switching Points under all possible traffic mixes and loads. This is partially due to the fact that all of these strategies are based on the use of fixed parameters (and are therefore not flexible enough to deal with IN traffic) and partially because none of these strategies take into account the different load requirements of the different service types. A new, flexible strategy is then devised to facilitate global IN congestion control and cater for service types with different characteristics. This strategy maximises IN performance by protecting all network elements from overload while maximising network revenue and preserving fairness between service types during overload. A number of factors determining the relative importance or weight of different traffic types are also identified and used by the strategy to maintain call importance during overload. The efficiency of this strategy is demonstrated by comparing its operation to that of the best classic IN overload controls and also to a new strategy, which has scalable and dynamic behaviour (and which was devised for the purpose of providing a fair comparison to the optimisation strategy). The optimisation-based strategy and dynamic strategy are found to be equally effective and far superior to the classic strategies. However, the optimisation algorithm also preserves relative importance and fairness, while maximising network revenue - but at the cost of a not insignificant processing overhead

A time dependent performance model for multihop wireless networks with CBR traffic

In this paper, we develop a performance modeling technique for analyzing the time varying network layer queueing behavior of multihop wireless networks with constant bit rate traffic. Our approach is a hybrid of fluid flow queueing modeling and a time varying connectivity matrix. Network queues are modeled using fluid-flow based differential equation models which are solved using numerical methods, while node mobility is modeled using deterministic or stochastic modeling of adjacency matrix elements. Numerical and simulation experiments show that the new approach can provide reasonably accurate results with significant improvements in the computation time compared to standard simulation tools. © 2010 IEEE

Protocol security for third generation telecommunication systems

In this thesis, a novel protocol stack architecture is presented. The Future Core Networks System (FCNS) forms a secure reference model for use in packet-switched structures, with its applicability ranging from computer to telecommunication networks. An insight on currently used network protocol systems is given, analysing standardised sets of communication rules with respect to the security they afford to the messages exchanged. The lack of protection schemes for the internal protocol stack messages and the implementation pitfalls of their security architectures are described, in relation to the effects they have on the communication process. The OSI security model is also considered, with disadvantages identified in the placement of security functionality and its management. The drawbacks depicted for currently used systems form the motivation behind this work. The analysis of the FCNS follows, which is composed of three parts. In the first part, the FCNS communication layers are examined, with respect to the mechanisms used to establish, maintain and tear down a connection between peer entities. In the second part, the security mechanisms of the proposed reference architecture are given, including details on the FCNS keystream generator used for the security of the internal FCNS messages. Finally, the FCNS Error Protocol is depicted, illustrating the modes of operation and advantages it exhibits over currently used systems. The work then moves into presenting details of the software FCNS implementation, followed by the presentation of the results and measurements obtained by the case studies created. Comparisons are given in relation to the TCP/IP suite, to provide the means of identifying the FCNS applicability in various network environments. The work is concluded by presenting the FCNS functionality in delivering information for the UMTS, together with further work that may enhance the flexibility and use of the proposed architecture

Priority Communications for Critical Situations on Mobile Networks

[ES] En la actualidad, las redes públicas de comunicación están ampliamente desplegadas en todo el territorio. Como las redes públicas no contemplan un uso priorizado de los recurso, los cuerpos de seguridad tienden a utilizar redes privadas de uso específico. Estas redes privadas satisfacen los requisitos marcados pero, a cambio, los costes de despliegue y mantenimiento son muy elevados, lo cual limita su despliegue y disponibilidad. Además, la interconexión entre distintas redes privadas no siempre es posible, lo que supone un gran problema cuando la emergencia se produce en zonas fronterizas. Estos grandes inconvenientes justifican un estudio minucioso sobre nuevos mecanismos de priorización en la gestión de recursos radio que permitan hacer uso de las redes públicas por parte de los cuerpos de seguridad y emergencias. Para ello se ha analizado el marco tecnológico actual, se ha contactado con distintos cuerpos de seguridad para averiguar los requisitos de comunicación actuales y los deseables. Caracterizado el sistema, se han definido distintos escenarios realistas utilizados en simulación masivas para finalmente demostrar cómo una red pública es capaz de cursar todo el tráfico que actualmente cursa una red privada en una situación de emergencia.[EN] Technical evaluation for enhancement and priorization of calls during a emergency situation over 2G and 3G networksDíaz Sendra, S. (2012). Priority Communications for Critical Situations on Mobile Networks. http://hdl.handle.net/10251/27446.Archivo delegad

Voice over IP

The area that this thesis covers is Voice over IP (or IP Telephony as it is sometimes called) over Private networks and not over the Internet. There is a distinction to be made between the two even though the term is loosely applied to both. IP Telephony over Private Networks involve calls made over private WANs using IP telephony protocols while IP Telephony over the Internet involve calls made over the public Internet using IP telephony protocols. Since the network is private, service is reliable because the network owner can control how resources are allocated to various applications, such as telephony services. The public Internet on the other hand is a public, largely unmanaged network that offers no reliable service guarantee. Calls placed over the Internet can be low in quality, but given the low price, some find this solution attractive. What started off as an Internet Revolution with free phone calls being offered to the general public using their multimedia computers has turned into a telecommunication revolution where enterprises are beginning to converge their data and voice networks into one network. In retrospect, an enterprise\u27s data networks are being leveraged for telephony. The communication industry has come full circle. Earlier in the decade data was being transmitted over the public voice networks and now voice is just another application which is/will be run over the enterprises existing data networks. We shall see in this thesis the problems that are encountered while sending Voice over Data networks using the underlying IP Protocol and the corrective steps taken by the Industry to resolve these multitudes of issues. Paul M. Zam who is collaborating in this Joint Thesis/project on VoIP will substantiate this theoretical research with his practical findings. On reading this paper the reader will gain an insight in the issues revolving the implementation of VoIP in an enterprises private network as well the technical data, which sheds more light on the same. Thus the premise of this joint thesis/project is to analyze the current status of the technology and present a business case scenario where an organization will be able to use this information