Connection Robustness for Wireless Moving Networks Using Transport Layer Multi-homing

Given any form of mobility management through wireless communication, one useful enhancement is improving the reliability and robustness of transport-layer connections in a heterogeneous mobile environment. This is particularly true in the case of mobile networks with multiple vertical handovers. In this thesis, issues and challenges in mobility management for mobile terminals in such a scenario are addressed, and a number of techniques to facilitate and improve efficiency and the QoS for such a handover are proposed and investigated. These are initially considered in an end-to-end context and all protocols and changes happened in the middleware of the connection where the network is involved with handover issues and end user transparency is satisfied. This thesis begins by investigating mobility management solutions particularly the transport layer models, also making significant observation pertinent to multi-homing for moving networks in general. A new scheme for transport layer tunnelling based on SCTP is proposed. Consequently a novel protocol to handle seamless network mobility in heterogeneous mobile networks, named nSCTP, is proposed. Efficiency of this protocol in relation to QoS for handover parameters in an end-to-end connection while wired and wireless networks are available is considered. Analytically and experimentally it has been proved that this new scheme can significantly increase the throughput, particularly when the mobile networks roam frequently. The detailed plan for the future improvements and expansion is also provided

Connection robustness for wireless moving networks using transport layer multi-homing

Given any form of mobility management through wireless communication, one useful enhancement is improving the reliability and robustness of transport-layer connections in a heterogeneous mobile environment. This is particularly true in the case of mobile networks with multiple vertical handovers. In this thesis, issues and challenges in mobility management for mobile terminals in such a scenario are addressed, and a number of techniques to facilitate and improve efficiency and the QoS for such a handover are proposed and investigated. These are initially considered in an end-to-end context and all protocols and changes happened in the middleware of the connection where the network is involved with handover issues and end user transparency is satisfied. This thesis begins by investigating mobility management solutions particularly the transport layer models, also making significant observation pertinent to multi-homing for moving networks in general. A new scheme for transport layer tunnelling based on SCTP is proposed. Consequently a novel protocol to handle seamless network mobility in heterogeneous mobile networks, named nSCTP, is proposed. Efficiency of this protocol in relation to QoS for handover parameters in an end-to-end connection while wired and wireless networks are available is considered. Analytically and experimentally it has been proved that this new scheme can significantly increase the throughput, particularly when the mobile networks roam frequently. The detailed plan for the future improvements and expansion is also provided.EThOS - Electronic Theses Online ServiceGBUnited Kingdo

On SIP Server Clusters and the Migration to Cloud Computing Platforms

This thesis looks in depth at telephony server clusters, the modern switchboards at the core of a packet-based telephony service. The most widely used de facto standard protocols for telecommunications are the Session Initiation Protocol (SIP) and the Real Time Protocol (RTP). SIP is a signaling protocol used to establish, maintain, and tear down communication channel between two or more parties. RTP is a media delivery protocol that allows packets to carry digitized voice, video, or text. SIP telephony server clusters that provide communications services, such as an emergency calling service, must be scalable and highly available. We evaluate existing commercial and open source telephony server clusters to see how they differ in scalability and high availability. We also investigate how a scalable SIP server cluster can be built on a cloud computing platform. Elasticity of resources is an attractive property for SIP server clusters because it allows the cluster to grow or shrink organically based on traffic load. However, simply deploying existing clusters to cloud computing platforms is not good enough to take full advantage of elasticity. We explore the design and implementation of clusters that scale in real-time. The database tier of our cluster was modified to use a scalable key-value store so that both the SIP proxy tier and the database tier can scale separately. Load monitoring and reactive threshold-based scaling logic is presented and evaluated. Server clusters also need to reduce processing latency. Otherwise, subscribers experience low quality of service such as delayed call establishment, dropped calls, and inadequate media quality. Cloud computing platforms do not guarantee latency on virtual machines due to resource contention on the same physical host. These extra latencies from resource contention are temporary in nature. Therefore, we propose and evaluate a mechanism that temporarily distributes more incoming calls to responsive SIP proxies, based on measurements of the processing delay in proxies. Availability of SIP server clusters is also a challenge on platforms where a node may fail anytime. We investigated how single component failures in a cluster can lead to a complete system outage. We found that for single component failures, simply having redundant components of the same type are enough to mask those failures. However, for client-facing components, smarter clients and DNS resolvers are necessary. Throughout the thesis, a prototype SIP proxy cluster is re-used, with variations in the architecture or configuration, to demonstrate and address issues mentioned above. This allows us to tie all of our approaches for different issues into one coherent system that is dynamically scalable, is responsive despite latency varations of virtual machines, and is tolerant of single component failures in cloud platforms

MP-CFM: MPTCP-Based communication functional module for next generation ERTMS

184 p. El contenido de los capítulos 4,5,6,7,8 y 9 está sujeto a confidencialidadEl Sistema Europeo de Gestión del Tráfico Ferroviario (ERTMS, por sus siglasen inglés), fue originalmente diseñado para los ferrocarriles europeos. Sinembargo, a lo largo de las dos últimas décadas, este sistema se ha convertidoen el estándar de-facto para los servicios de Alta Velocidad en la mayoría depaíses desarrollados.El sistema ERTMS se compone de tres subsistemas principales: 1) el Sistemade Control Ferroviario Europeo (ETCS, por sus siglas en inglés), que actúacomo aplicación de señalización; 2) el sistema Euroradio, que a su vez estádividido en dos subsistemas, el Módulo de Seguridad Funcional (SFM, porsus siglas en inglés), y el Módulo de Comunicación Funcional (CFM, porsus siglas en inglés); y 3) el sistema de comunicaciones subyacente, GSM-R,que transporta la información intercambiada entre el sistema embarcado enel tren (OBU, por sus siglas en inglés) y el Centro de Bloqueo por Radio(RBC, por sus siglas en inglés). El sistema de señalización ETCS soporta tresniveles dependiendo del nivel de prestaciones soportadas. En el nivel 3 seintroduce la posibilidad de trabajar con bloques móviles en lugar de bloquesfijos definidos en la vía. Esto implica que la distancia de avance entre dos trenesconsecutivos puede ser reducida a una distancia mínima en la que se garanticela seguridad del servicio, aumentando por tanto la capacidad del corredorferroviario. Esta distancia de seguridad viene determinada por la combinaciónde la distancia de frenado del tren y el retraso de las comunicaciones deseñalización. Por lo tanto, se puede afirmar que existe una relación directaentre los retrasos y la confiabilidad de las transmisiones de las aplicaciones deseñalización y la capacidad operacional de un corredor ferroviario. Así pues,el estudio y mejora de los sistemas de comunicaciones utilizados en ERTMSjuegan un papel clave en la evolución del sistema ERTMS. Asimismo, unaoperatividad segura en ERTMS, desde el punto de vista de las comunicacionesimplicadas en la misma, viene determinada por la confiabilidad de lascomunicaciones, la disponibilidad de sus canales de comunicación, el retrasode las comunicaciones y la seguridad de sus mensajes.Unido este hecho, la industria ferroviaria ha venido trabajando en ladigitalización y la transición al protocolo IP de la mayor parte de los sistemasde señalización. Alineado con esta tendencia, el consorcio industrial UNISIGha publicado recientemente un nuevo modelo de comunicaciones para ERTMSque incluye la posibilidad, no solo de operar con el sistema tradicional,basado en tecnología de conmutación de circuitos, sino también con un nuevosistema basado en IP. Esta tesis está alineada con el contexto de migraciónactual y pretende contribuir a mejorar la disponibilidad, confiabilidad yseguridad de las comunicaciones, tomando como eje fundamental los tiemposde transmisión de los mensajes, con el horizonte puesto en la definición deuna próxima generación de ERTMS, definida en esta tesis como NGERTMS.En este contexto, se han detectado tres retos principales para reforzar laresiliencia de la arquitectura de comunicaciones del NGERTMS: 1) mejorarla supervivencia de las comunicaciones ante disrupciones; 2) superar laslimitaciones actuales de ERTMS para enviar mensajes de alta prioridad sobretecnología de conmutación de paquetes, dotando a estos mensajes de un mayorgrado de resiliencia y menor latencia respecto a los mensajes ordinarios; y3) el aumento de la seguridad de las comunicaciones y el incremento de ladisponibilidad sin que esto conlleve un incremento en la latencia.Considerando los desafíos previamente descritos, en esta tesis se proponeuna arquitectura de comunicaciones basada en el protocolo MPTCP, llamadaMP-CFM, que permite superar dichos desafíos, a la par que mantener laretrocompatibilidad con el sistema de comunicaciones basado en conmutaciónde paquetes recientemente propuesto por UNISIG. Hasta el momento, esta esla primera vez que se propone una arquitectura de comunicaciones completacapaz de abordar los desafíos mencionados anteriormente. Esta arquitecturaimplementa cuatro tipos de clase de servicio, los cuales son utilizados porlos paquetes ordinarios y de alta prioridad para dos escenarios distintos; unescenario en el que ambos extremos, el sistema embarcado o OBU y el RBC,disponen de múltiples interfaces de red; y otro escenario transicional en el cualel RBC sí tiene múltiples interfaces de red pero el OBU solo dispone de unaúnica interfaz. La arquitectura de comunicaciones propuesta para el entornoferroviario ha sido validada mediante un entorno de simulación desarrolladopara tal efecto. Es más, dichas simulaciones demuestran que la arquitecturapropuesta, ante disrupciones de canal, supera con creces en términos derobustez el sistema diseñado por UNISIG. Como conclusión, se puede afirmarque en esta tesis se demuestra que una arquitectura de comunicaciones basadade MPTCP cumple con los exigentes requisitos establecidos para el NGERTMSy por tanto dicha propuesta supone un avance en la evolución del sistema deseñalización ferroviario europeo

An Architecture for Global Distributed SIP Network Using IPv4 Anycast

Tato diplomová práce se zabývá metodami pro výběr nejbližší RTP proxy k VoIP klientům s použitím IP anycastu. RTP proxy servery jsou umístěny v síti Internetu a přeposílají RTP data pro VoIP klienty za síťovými překladači adres(NAT). Bez zeměpisně rozmístěných RTP proxy serverů a metod pro nalezení nejbližšího RTP proxy serveru by došlo ke zbytečnému poklesu kvality přenosu médialních dat a velkému zpoždení. Tento dokument navrhuje 4 metody a jejich porovnání s podrobnějšími rozbory metod s využitím DNS resolvování a přímo SIP protokolu. Tento dokument také obsahuje měření chování IP anycastu v porovnání mezi metrikami směrování a metrikami časovými. Nakonec dokumentu je také uvedena implemetace na SIP Express Router platformě.This thesis is about using IP anycast-based methods for locating RTP proxy servers close to VoIP clients. The RTP proxy servers are hosts on the public Internet that relay RTP media between VoIP clients in a way that accomplishes traversal over Network Address Translators (NATs). Without geographically-dispersed RTP proxy servers and methods to find one in client's proximity, voice latency may be unbearably long and dramatically reduce perceived voice quality. This document proposes four methods their comparison with further design of DNS-based and SIP-based methods. It includes IP anycast measurements that provides an overview of IP anycast behaviour in terms of routing metrics and latency metrics. It also includes implementation on SIP Express Router platform.

Reliable Server Pooling - Evaluierung, Optimierung und Erweiterung einer neuen IETF-Architektur

The Reliable Server Pooling (RSerPool) architecture currently under standardization by the IETF RSerPool Working Group is an overlay network framework to provide server replication and session failover capabilities to applications using it. These functionalities as such are not new, but their combination into one generic, application-independent framework is. Initial goal of this thesis is to gain insight into the complex RSerPool mechanisms by performing experimental and simulative proof-of-concept tests. The further goals are to systematically validate the RSerPool architecture and its protocols, provide improvements and optimizations where necessary and propose extensions if useful. Based on these evaluations, recommendations to implementers and users of RSerPool should be provided, giving guidelines for the tuning of system parameters and the appropriate configuration of application scenarios. In particular, it is also a goal to transfer insights, optimizations and extensions of the RSerPool protocols from simulation to reality and also to bring the achievements from research into application by supporting and contributing relevant results to the IETF's ongoing RSerPool standardization process. To achieve the described goals, a prototype implementation as well as a simulation model are designed and realized at first. Using a generic application model and appropriate performance metrics, the performance of RSerPool systems in failure-free and server failure scenarios is systematically evaluated in order to identify critical parameter ranges and problematic protocol behaviour. Improvements developed as result of these performance analyses are evaluated and finally contributed into the standardization process of RSerPool

QoE-Centric Control and Management of Multimedia Services in Software Defined and Virtualized Networks

Multimedia services consumption has increased tremendously since the deployment of 4G/LTE networks. Mobile video services (e.g., YouTube and Mobile TV) on smart devices are expected to continue to grow with the emergence and evolution of future networks such as 5G. The end user’s demand for services with better quality from service providers has triggered a trend towards Quality of Experience (QoE) - centric network management through efficient utilization of network resources. However, existing network technologies are either unable to adapt to diverse changing network conditions or limited in available resources. This has posed challenges to service providers for provisioning of QoE-centric multimedia services. New networking solutions such as Software Defined Networking (SDN) and Network Function Virtualization (NFV) can provide better solutions in terms of QoE control and management of multimedia services in emerging and future networks. The features of SDN, such as adaptability, programmability and cost-effectiveness make it suitable for bandwidth-intensive multimedia applications such as live video streaming, 3D/HD video and video gaming. However, the delivery of multimedia services over SDN/NFV networks to achieve optimized QoE, and the overall QoE-centric network resource management remain an open question especially in the advent development of future softwarized networks. The work in this thesis intends to investigate, design and develop novel approaches for QoE-centric control and management of multimedia services (with a focus on video streaming services) over software defined and virtualized networks. First, a video quality management scheme based on the traffic intensity under Dynamic Adaptive Video Streaming over HTTP (DASH) using SDN is developed. The proposed scheme can mitigate virtual port queue congestion which may cause buffering or stalling events during video streaming, thus, reducing the video quality. A QoE-driven resource allocation mechanism is designed and developed for improving the end user’s QoE for video streaming services. The aim of this approach is to find the best combination of network node functions that can provide an optimized QoE level to end-users through network node cooperation. Furthermore, a novel QoE-centric management scheme is proposed and developed, which utilizes Multipath TCP (MPTCP) and Segment Routing (SR) to enhance QoE for video streaming services over SDN/NFV-based networks. The goal of this strategy is to enable service providers to route network traffic through multiple disjointed bandwidth-satisfying paths and meet specific service QoE guarantees to the end-users. Extensive experiments demonstrated that the proposed schemes in this work improve the video quality significantly compared with the state-of-the- art approaches. The thesis further proposes the path protections and link failure-free MPTCP/SR-based architecture that increases survivability, resilience, availability and robustness of future networks. The proposed path protection and dynamic link recovery scheme achieves a minimum time to recover from a failed link and avoids link congestion in softwarized networks