New Cross-Layer Channel Switching Policy for TCP Transmission on 3G UMTS Downlink

In 3G UMTS, two main transport channels have been provided for downlink data transmission: a common FACH channel and a dedicated DCH channel. The performance of TCP in UMTS depends much on the channel switching policy used. In this paper, we propose and analyze three new basic threshold-based channel switching policies for UMTS that we name as QS (Queue Size), FS (Flow Size) and QSFS (QS & FS combined) policy. These policies significantly improve over a modified threshold policy in [1] by about 17% in response time metrics. We further propose and evaluate a new improved switching policy that we call FS-DCH (at-least flow-size threshold on DCH) policy. This policy is biased towards short TCP flows of few packets and is thus a cross-layer policy that improves the performance of TCP by giving priority to the initial few packets of a flow on the fast DCH channel. Extensive simulation results confirm this improvement for the case when number of TCP connections is low

Final report on the evaluation of RRM/CRRM algorithms

Deliverable public del projecte EVERESTThis deliverable provides a definition and a complete evaluation of the RRM/CRRM algorithms selected in D11 and D15, and evolved and refined on an iterative process. The evaluation will be carried out by means of simulations using the simulators provided at D07, and D14.Preprin

Quality of service optimization of multimedia traffic in mobile networks

Mobile communication systems have continued to evolve beyond the currently deployed Third Generation (3G) systems with the main goal of providing higher capacity. Systems beyond 3G are expected to cater for a wide variety of services such as speech, data, image transmission, video, as well as multimedia services consisting of a combination of these. With the air interface being the bottleneck in mobile networks, recent enhancing technologies such as the High Speed Downlink Packet Access (HSDPA), incorporate major changes to the radio access segment of 3G Universal Mobile Telecommunications System (UMTS). HSDPA introduces new features such as fast link adaptation mechanisms, fast packet scheduling, and physical layer retransmissions in the base stations, necessitating buffering of data at the air interface which presents a bottleneck to end-to-end communication. Hence, in order to provide end-to-end Quality of Service (QoS) guarantees to multimedia services in wireless networks such as HSDPA, efficient buffer management schemes are required at the air interface. The main objective of this thesis is to propose and evaluate solutions that will address the QoS optimization of multimedia traffic at the radio link interface of HSDPA systems. In the thesis, a novel queuing system known as the Time-Space Priority (TSP) scheme is proposed for multimedia traffic QoS control. TSP provides customized preferential treatment to the constituent flows in the multimedia traffic to suit their diverse QoS requirements. With TSP queuing, the real-time component of the multimedia traffic, being delay sensitive and loss tolerant, is given transmission priority; while the non-real-time component, being loss sensitive and delay tolerant, enjoys space priority. Hence, based on the TSP queuing paradigm, new buffer managementalgorithms are designed for joint QoS control of the diverse components in a multimedia session of the same HSDPA user. In the thesis, a TSP based buffer management algorithm known as the Enhanced Time Space Priority (E-TSP) is proposed for HSDPA. E-TSP incorporates flow control mechanisms to mitigate congestion in the air interface buffer of a user with multimedia session comprising real-time and non-real-time flows. Thus, E-TSP is designed to provide efficient network and radio resource utilization to improve end-to-end multimedia traffic performance. In order to allow real-time optimization of the QoS control between the real-time and non-real-time flows of the HSDPA multimedia session, another TSP based buffer management algorithm known as the Dynamic Time Space Priority (D-TSP) is proposed. D-TSP incorporates dynamic priority switching between the real-time and non-real-time flows. D-TSP is designed to allow optimum QoS trade-off between the flows whilst still guaranteeing the stringent real-time component’s QoS requirements. The thesis presents results of extensive performance studies undertaken via analytical modelling and dynamic network-level HSDPA simulations demonstrating the effectiveness of the proposed TSP queuing system and the TSP based buffer management schemes

Overview of UMTS network evolution through radio and transmission feature validation

This project is based on several UMTS network feature validation with the aim to provide an end-to-end in-depth knowledge overview gained in parallel in the areas of radio network mobility processes (cell camping and inter-system handover), Quality of Service improvement for HSPA data users and transport network evolution towards the All-IP era.Hardware and software validation is a key step in the relationship between the mobile network operator and the vendor. Through this verification process, while executing that functionality or testing a specific hardware, the difference between the actual result and expected result can be better understood and, in turn, this in-depth knowledge acquisition is translated into a tailored usage of the product in the operator’s live network. As a result, validation helps in building a better product as per the customer’s requirement and helps satisfying their needs, which positively impacts in the future evolution of the vendor product roadmap implementation process for a specific customer. This project is based on several Universal Mobile Telecommunication Services (UMTS) network feature validation with the aim to provide an end-to-end in-depth knowledge overview gained in parallel in the areas of radio network mobility processes (cell camping and inter-system handover), Quality of Service improvement for High Speed Downlink Packet Access (HSPA) data users and transport network evolution towards the All-IP era.Las campañas de validación hardware y software son un paso clave en las relaciones comerciales establecidas entre un operador de telecomunicaciones y su proveedor de equipos de red. Durante los procesos de certificación, mientras se ejecuta una funcionalidad software o se valida un determinado hardware, se obtiene un conocimiento profundo de la diferencia entre el resultado obtenido y el esperado, repercutiendo directamente en un uso a medida de dicha funcionalidad o hardware en la propia red del cliente. Como consecuencia de lo anterior, podemos aseverar que los procesos de validación permiten en gran medida al proveedor adaptarse mejor a los requerimientos del cliente, ayudando a satisfacer realmente sus necesidades. Esto implica directamente un impacto positivo en la futura evolución del portfolio que el fabricante ofrece a un determinado cliente. Este proyecto está basado en la validación de diferentes funcionalidades de red UMTS, cuyo objetivo es proporcionar un conocimiento global de distintos aspectos que conforman el funcionamiento de una red de telecomunicaciones 3G, como son los procesos de movilidad de acceso radio (acampado de red y handover inter-sistema), las mejoras en la calidad de servicio para usuarios de datos HSPA y la convergencia de la red de transporte hacia la era IP.Els processos de validació hardware i software són un punt clau en les relacions comercials establertes entre un operador de telecomunicaciones i el proveïdor d'equipament de la xarxa. En el transcurs dels processos de certificació, a la mateixa vegada que s'executa una funcionalitat software o es valida un determinat hardware, s'obtenen grans coneixements respecte la diferència entre el resultat obtingut i l'esperat, que són d'aplicació directa a l'hora d'establir un ús adpatat a la xarxa del client. En conseqüència, podem asseverar que les campanyes de validació permeten en gran mesura al proveïdor adaptar-se millor als requeriments del client, ajudant a satisfer realment les seves necessitats. Això implica directament un impacte positiu en la futura evol.lució del portfoli que el fabricant ofereix a un determinat client. Aquest projecte es basa en la presentació d'un procès de validació de diferents funcionalitats relacionades amb la xarxa UMTS, amb l'objectiu de proporcionar un coneixement global de la varietat d'aspectes que conformen el funcionament d'una xarxa de telecomunicacions 3G, com són els processos de mobilitat en accès radio (acampat de l'usuari i handover inter-sistema), millores en la qualitat de servei per a usuaris de dades HSPA i la convergència de la xarxa de transport cap a l'era IP

Dynamic Buffer Management for Multimedia QoS in Beyond 3G Wireless Networks

This paper investigates a dynamic buffer management scheme for QoS control of multimedia services in beyond 3G wireless systems. The scheme is studied in the context of the state-of-the-art 3.5G system i.e. the High Speed Downlink Packet Access (HSDPA) which enhances 3G UMTS to support high-speed packet switched services. Unlike earlier systems, UMTS-evolved systems from HSDPA and beyond incorporate mechanisms such as packet scheduling and HARQ in the base station necessitating data buffering at the air interface. This introduces a potential bottleneck to end-to-end communication. Hence, buffer management at the air interface is crucial for end-to-end QoS support of multimedia services with multiplexed parallel diverse flows such as video and data in the same end-user session. The dynamic buffer management scheme for HSDPA multimedia sessions with aggregated real-time and non real-time flows is investigated via extensive HSDPA simulations. The impact of the scheme on end-to-end traffic performance is evaluated with an example multimedia session comprising a real-time streaming flow concurrent with TCP-based non real-time flow. Results demonstrate that the scheme can guarantee the end-to-end QoS of the real-time streaming flow, whilst simultaneously protecting the non real-time flow from starvation resulting in improved end-to-end throughput performance

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

Private Communication Detection via Side-Channel Attacks

Private communication detection (PCD) enables an ordinary network user to discover communication patterns (e.g., call time, length, frequency, and initiator) between two or more private parties. Analysis of communication patterns between private parties has historically been a powerful tool used by intelligence, military, law-enforcement and business organizations because it can reveal the strength of tie between these parties. Ordinary users are assumed to have neither eavesdropping capabilities (e.g., the network may employ strong anonymity measures) nor the legal authority (e.g. no ability to issue a warrant to network providers) to collect private-communication records. We show that PCD is possible by ordinary users merely by sending packets to various network end-nodes and analyzing the responses. Three approaches for PCD are proposed based on a new type of side channels caused by resource contention, and defenses are proposed. The Resource-Saturation PCD exploits the resource contention (e.g., a fixed-size buffer) by sending carefully designed packets and monitoring different responses. Its effectiveness has been demonstrated on three commercial closed-source VoIP phones. The Stochastic PCD shows that timing side channels in the form of probing responses, which are caused by distinct resource-contention responses when different applications run in end nodes, enable effective PCD despite network and proxy-generated noise (e.g., jitter, delays). It was applied to WiFi and Instant Messaging for resource contention in the radio channel and the keyboard, respectively. Similar analysis enables practical Sybil node detection. Finally, the Service-Priority PCD utilizes the fact that 3G/2G mobile communication systems give higher priority to voice service than data service. This allows detection of the busy status of smartphones, and then discovery of their call records by correlating the busy status. This approach was successfully applied to iPhone and Android phones in AT&T's network. An additional, unanticipated finding was that an Internet user could disable a 2G phone's voice service by probing it with short enough intervals (e.g., 1 second). PCD defenses can be traditional side-channel countermeasures or PCD-specific ones, e.g., monitoring and blocking suspicious periodic network traffic

Optimization and Performance Analysis of High Speed Mobile Access Networks

The end-to-end performance evaluation of high speed broadband mobile access networks is the main focus of this work. Novel transport network adaptive flow control and enhanced congestion control algorithms are proposed, implemented, tested and validated using a comprehensive High speed packet Access (HSPA) system simulator. The simulation analysis confirms that the aforementioned algorithms are able to provide reliable and guaranteed services for both network operators and end users cost-effectively. Further, two novel analytical models one for congestion control and the other for the combined flow control and congestion control which are based on Markov chains are designed and developed to perform the aforementioned analysis efficiently compared to time consuming detailed system simulations. In addition, the effects of the Long Term Evolution (LTE) transport network (S1and X2 interfaces) on the end user performance are investigated and analysed by introducing a novel comprehensive MAC scheduling scheme and a novel transport service differentiation model

Application priority framework for fixed mobile converged communication networks

This thesis was submitted for the degree of Doctor of Philosophy and awarded by Brunel University.The current prospects in wired and wireless access networks, it is becoming increasingly important to address potential convergence in order to offer integrated broadband services. These systems will need to offer higher data transmission capacities and long battery life, which is the catalyst for an everincreasing variety of air interface technologies targeting local area to wide area connectivity. Current integrated industrial networks do not offer application aware context delivery and enhanced services for optimised networks. Application aware services provide value-added functionality to business applications by capturing, integrating, and consolidating intelligence about users and their endpoint devices from various points in the network. This thesis mainly intends to resolve the issues related to ubiquitous application aware service, fair allocation of radio access, reduced energy consumption and improved capacity. A technique that measures and evaluates the data rate demand to reduce application response time and queuing delay for multi radio interfaces is proposed. The technique overcomes the challenges of network integration, requiring no user intervention, saving battery life and selecting the radio access connection for the application requested by the end user. This study is split in two parts. The first contribution identifies some constraints of the services towards the application layer in terms of e.g. data rate and signal strength. The objectives are achieved by application controlled handover (ACH) mechanism in order to maintain acceptable data rate for real-time application services. It also looks into the impact of the radio link on the application and identifies elements and parameters like wireless link quality and handover that will influence the application type. It also identifies some enhanced traditional mechanisms such as distance controlled multihop and mesh topology required in order to support energy efficient multimedia applications. The second contribution unfolds an intelligent application priority assignment mechanism (IAPAM) for medical applications using wireless sensor networks. IAPAM proposes and evaluates a technique based on prioritising multiple virtual queues for the critical nature of medical data to improve instant transmission. Various mobility patterns (directed, controlled and random waypoint) has been investigated and compared by simulating IAPAM enabled mobile BWSN. The following topics have been studied, modelled, simulated and discussed in this thesis: 1. Application Controlled Handover (ACH) for multi radios over fibre 2. Power Controlled Scheme for mesh multi radios over fibre using ACH 3. IAPAM for Biomedical Wireless Sensor Networks (BWSN) and impact of mobility over IAPAM enabled BWSN. Extensive simulation studies are performed to analyze and to evaluate the proposed techniques. Simulation results demonstrate significant improvements in multi radios over fibre performance in terms of application response delay and power consumption by upto 75% and 15 % respectively, reduction in traffic loss by upto 53% and reduction in delay for real time application by more than 25% in some cases