Understanding the impact of TFRC feedbacks frequency over long delay links

TFRC is a transport protocol specifically designed to carry multimedia streams. TFRC does not enable a reliable and in order data delivery services. However the mechanism is designed to be friendly with TCP flows and thus, enables a control congestion algorithm. This congestion control relies in a feedback mechanism allowing receivers to communicate to the senders an experienced drop rate. Several studies attempted to adapt TFRC to a wide range of network conditions and topologies. Although the current TFRC RFC writes that there is little gain from sending a large number of feedback messages per RTT, recent studies have shown that in long-delay contexts, such as satellite-based networks, the performance of TFRC can be greatly improved by increasing the feedback frequency. Nevertheless, currently it is not clear how and why this increase may improve the performance of TFRC. Therefore, in this paper, we aim at understanding the impact that multiple feedback per RTT may have (i) on the key parameters of TFRC (RTT, drop rate, and sending rate) and (ii) on the network parameters (reactiveness and link utilization).We also provide a detailed description of the micro-mechanisms at the origin of the improvements of the TFRC behaviour when multiple feedback per RTT are delivered, and determine the context where such feedback frequencies should be applied

Rate Control State-of-the-art Survey

The majority of Internet traffic use Transmission Control Protocol (TCP) as the transport level protocol. It provides a reliable ordered byte stream for the applications. However, applications such as live video streaming place an emphasis on timeliness over reliability. Also a smooth sending rate can be desirable over sharp changes in the sending rate. For these applications TCP is not necessarily suitable. Rate control attempts to address the demands of these applications. An important design feature in all rate control mechanisms is TCP friendliness. We should not negatively impact TCP performance since it is still the dominant protocol. Rate Control mechanisms are classified into two different mechanisms: window-based mechanisms and rate-based mechanisms. Window-based mechanisms increase their sending rate after a successful transfer of a window of packets similar to TCP. They typically decrease their sending rate sharply after a packet loss. Rate-based solutions control their sending rate in some other way. A large subset of rate-based solutions are called equation-based solutions. Equation-based solutions have a control equation which provides an allowed sending rate. Typically these rate-based solutions react slower to both packet losses and increases in available bandwidth making their sending rate smoother than that of window-based solutions. This report contains a survey of rate control mechanisms and a discussion of their relative strengths and weaknesses. A section is dedicated to a discussion on the enhancements in wireless environments. Another topic in the report is bandwidth estimation. Bandwidth estimation is divided into capacity estimation and available bandwidth estimation. We describe techniques that enable the calculation of a fair sending rate that can be used to create novel rate control mechanisms.Peer reviewe

Improved algorithms for TCP congestion control

Reliable and efficient data transfer on the Internet is an important issue. Since late 70’s the protocol responsible for that has been the de facto standard TCP, which has proven to be successful through out the years, its self-managed congestion control algorithms have retained the stability of the Internet for decades. However, the variety of existing new technologies such as high-speed networks (e.g. fibre optics) with high-speed long-delay set-up (e.g. cross-Atlantic links) and wireless technologies have posed lots of challenges to TCP congestion control algorithms. The congestion control research community proposed solutions to most of these challenges. This dissertation adds to the existing work by: firstly tackling the highspeed long-delay problem of TCP, we propose enhancements to one of the existing TCP variants (part of Linux kernel stack). We then propose our own variant: TCP-Gentle. Secondly, tackling the challenge of differentiating the wireless loss from congestive loss in a passive way and we propose a novel loss differentiation algorithm which quantifies the noise in packet inter arrival times and use this information together with the span (ratio of maximum to minimum packet inter arrival times) to adapt the multiplicative decrease factor according to a predefined logical formula. Finally, extending the well-known drift model of TCP to account for wireless loss and some hypothetical cases (e.g. variable multiplicative decrease), we have undertaken stability analysis for the new version of the model

Enabling Multipath and Multicast Data Transmission in Legacy and Future Internet

The quickly growing community of Internet users is requesting multiple applications and services. At the same time the structure of the network is changing. From the performance point of view, there is a tight interplay between the application and the network design. The network must be constructed to provide an adequate performance of the target application. In this thesis we consider how to improve the quality of users' experience concentrating on two popular and resource-consuming applications: bulk data transfer and real-time video streaming. We share our view on the techniques which enable feasibility and deployability of the network functionality leading to unquestionable performance improvement for the corresponding applications. Modern mobile devices, equipped with several network interfaces, as well as multihomed residential Internet hosts are capable of maintaining multiple simultaneous attachments to the network. We propose to enable simultaneous multipath data transmission in order to increase throughput and speed up such bandwidth-demanding applications as, for example, file download. We design an extension for Host Identity Protocol (mHIP), and propose a multipath data scheduling solution on a wedge layer between IP and transport, which effectively distributes packets from a TCP connection over available paths. We support our protocol with a congestion control scheme and prove its ability to compete in a friendly manner against the legacy network protocols. Moreover, applying game-theoretic analytical modelling we investigate how the multihomed HIP multipath-enabled hosts coexist in the shared network. The number of real-time applications grows quickly. Efficient and reliable transport of multimedia content is a critical issue of today's IP network design. In this thesis we solve scalability issues of the multicast dissemination trees controlled by the hybrid error correction. We propose a scalable multicast architecture for potentially large overlay networks. Our techniques address suboptimality of the adaptive hybrid error correction (AHEC) scheme in the multicast scenarios. A hierarchical multi-stage multicast tree topology is constructed in order to improve the performance of AHEC and guarantee QoS for the multicast clients. We choose an evolutionary networking approach that has the potential to lower the required resources for multimedia applications by utilizing the error-correction domain separation paradigm in combination with selective insertion of the supplementary data from parallel networks, when the corresponding content is available. Clearly both multipath data transmission and multicast content dissemination are the future Internet trends. We study multiple problems related to the deployment of these methods.Internetin nopeasti kasvava käyttäjäkunta vaatii verkolta yhä enemmän sovelluksia ja palveluita. Samaan aikaan verkon rakenne muuttuu. Suorituskyvyn näkökulmasta on olemassa selvä vuorovaikutussovellusten ja verkon suunnittelun välillä. Verkko on rakennettava siten, että se pystyy takaamaan riittävän suorituskyvyn halutuille palveluille. Tässä väitöskirjassa pohditaan, miten verkon käyttökokemusta voidaan parantaa keskittyen kahteen suosittuun ja resursseja vaativaan sovellukseen: tiedonsiirtoon ja reaaliaikaiseen videon suoratoistoon. Esitämme näkemyksemme tekniikoista, jotka mahdollistavat tarvittavien verkkotoiminnallisuuksien helpon toteuttavuuden sekä kiistatta parantavat sovelluksien suorityskykyä. Nykyaikaiset mobiililaitteet monine verkkoyhteyksineen, kuten myös kotitietokoneet, pystyvät ylläpitämään monta internet-yhteyttä samanaikaisesti. Siksi ehdotamme monikanavaisen tiedonsiirron käyttöä suorituskyvyn parantamiseksi ja etenkin vaativien verkkosovelluksien, kuten tiedostonsiirron, nopeuttamiseksi. Tässä väitöskirjassa suunnitellaan Host Identity Protocol (mHIP) -laajennus, sekä esitetään tiedonsiirron vuorotteluratkaisu, joka hajauttaa TCP-yhteyden tiedonsiirtopaketit käytettävissä oleville kanaville. Protokollamme tueksi luomme myös ruuhkautumishallinta-algoritmin ja näytämme sen pystyvän toimimaan yhteen nykyisien verkkoprotokollien kanssa. Tämän lisäksi tutkimme peliteoreettista mallinnusta käyttäen, miten monikanavaiset HIP-verkkopäätteet toimivat muiden kanssa jaetuissa verkoissa. Reaaliaikaisten sovellusten määrä kasvaa nopeasti. Tehokas ja luotettava multimediasisällön siirto on olennainen vaatimus nykypäivän IP-verkoissa. Tässä työssä ratkaistaan monilähetyksen (multicast) jakelustruktuurin skaalautuvuuteen liittyviä ongelmia. Ehdotamme skaalautuvaa monilähetysarkkitehtuuria suurille peiteverkoille. Ratkaisumme puuttuu adaptiivisen virhekorjauksen (Adaptive Hybrid Error Correction, AHEC) alioptimaalisuuteen monilähetystilanteissa. Luomme hierarkisen monivaiheisen monilähetyspuutopologian parantaaksemme AHECin suorituskykyä, sekä taataksemme monilähetysasiakkaiden palvelun laadun. Valitsimme evoluutiomaisen lähestymistavan, jolla on potentiaalia keventää multimediasovelluksien verkkoresurssivaatimuksia erottamalla virhekorjauksen omaksi verkkotunnuksekseen, sekä käyttämällä valikoivaa täydentävää tiedonlisäystä rinnakkaisverkoista vastaavan sisällön ollessa saatavilla. Sekä monikanava- että monilähetystiedonsiirto ovat selvästi osa internetin kehityssuuntaa. Tässä väitöskirjassa tutkimme monia ongelmia näiden tekniikoiden käyttöönottoon liittyen

Moving toward the intra-protocol de-ossification of TCP in mobile networks: Start-up and mobility

182 p.El uso de las redes móviles de banda ancha ha aumentado significativamente los últimos años y se espera un crecimiento aún mayor con la inclusión de las futuras capacidades 5G. 5G proporcionará unas velocidades de transmisión y reducidos retardos nunca antes vistos. Sin embargo, la posibilidad de alcanzar las mencionadas cuotas está limitada por la gestión y rendimiento de los protocolos de transporte. A este respecto, TCP sigue siendo el protocolo de transporte imperante y sus diferentes algoritmos de control de congestión (CCA) los responsables finales del rendimiento obtenido. Mientras que originalmente los distintos CCAs han sido implementados para hacer frente a diferentes casos de uso en redes fijas, ninguno de los CCAs ha sido diseñado para poder gestionar la variabilidad de throughput y retardos de diferentes condiciones de red redes móviles de una manera fácilmente implantable. Dado que el análisis de TCP sobre redes móviles es complejo debido a los múltiples factores de impacto, nuestro trabajo se centra en dos casos de uso generalizados que resultan significativos en cuanto a afección del rendimiento: movimiento de los usuarios como representación de la característica principal de las redes móviles frente a las redes fijas y el rendimiento de la fase de Start-up de TCP debido a la presencia mayoritaria de flujos cortos en Internet. Diferentes trabajos han sugerido la importancia de una mayor flexibilidad en la capa de transporte, creando servicios de transporte sobre TCP o UDP. Sin embargo, estas propuestas han encontrado limitaciones relativas a las dependencias arquitecturales de los protocolos utilizados como sustrato (p.ej. imposibilidad de cambiar la configuración de la capa de transporte una vez la transmisión a comenzado), experimentando una capa de transporte "osificada". Esta tesis surge como respuesta a fin de abordar la citada limitación y demostrando que existen posibilidades de mejora dentro de la familia de TCP (intra-protocolar), proponiendo un marco para solventar parcialmente la restricción a través de la selección dinámica del CCA más apropiado. Para ello, se evalúan y seleccionan los mayores puntos de impacto en el rendimiento de los casos de uso seleccionados en despliegues de red 4G y en despliegues de baja latencia que emulan las potenciales latencias en las futuras capacidades 5G. Estos puntos de impacto sirven como heurísticas para decidir el CCA más apropiado en el propuesto marco. Por último, se valida la propuesta en entornos de movilidad con dos posibilidades de selección: al comienzo de la transmisión (limitada flexibilidad de la capa de transporte) y dinámicamente durante la transmisión (con una capa de transporte flexible). Se concluye que la propuesta puede acarrear importantes mejoras de rendimiento al seleccionar el CCA más apropiado teniendo en cuenta la situación de red y los requerimientos de la capa de aplicación

Contributions based on cross-layer design for quality-of-service provisioning over DVB-S2/RCS broadband satellite system

Contributions based on cross-layer design for Quality-of-Service provisioning over DVB-S2/RCS Broadband Satellite Systems Nowadays, geostationary (GEO) satellite infrastructure plays a crucial role for the provisioning of IP services. Such infrastructure can provide ubiquity and broadband access, being feasible to reach disperse populations located worldwide within remote areas where terrestrial infrastructure can not be deployed. Nevertheless, due to the expansion of the World Wide Web (WWW), new IP applications such as Voice over IP (VoIP) and multimedia services requires considering different levels of individual packet treatment through the satellite network. This differentiation must include not only the Quality of Service (QoS) parameters to specify packet transmission priorities across the network nodes, but also the required amount of bandwidth assignment to guarantee its transport. In this context, the provisioning of QoS guarantees over GEO satellite systems becomes one of the main research areas of organizations such as the European Space Agency (ESA). Mainly because, their current infrastructures require continuous exploitation, as launching a new communication satellite is associated with excessive costs. Therefore, the support of IP services with QoS guarantees must be developed on the terrestrial segment to enable using the current assets. In this PhD thesis several contributions to improve the QoS provisioning over DVB-S2/RCS Broadband Satellite Systems have been developed. The contributions are based on cross-layer design, following the layered model standardized in the ETSI TR 102 157 and 462. The proposals take into account the drawbacks posed by GEO satellite systems such as delay, losses and bandwidth variations. The first contribution proposes QoSatArt, an architecture defined to improve QoS provisioning among services classes considering the physical layer variations due to the presence of rain events. The design is developed inside the gateway, including the specification of the main functional blocks to provide QoS guarantees and mechanisms to minimize de delay and jitter values experienced at the application layer. Here, a cross-layer design between the physical and the network layer has been proposed, to enforce the QoS specifications based on the available bandwidth. The proposed QoSatArt architecture is evaluated using the NS-2 simulation tool. In addition, the performance analysis of several standard Transmission Control Protocol (TCP) variants is also performed. This is carry out to find the most suitable TCP variant that enhances TCP transmission over a QoS architecture such as the QoSatArt. The second contribution proposes XPLIT, an architecture developed to enhance TCP transmission with QoS for DVB-S2/RCS satellite systems. Complementary to QoSatArt, XPLIT introduces Performance Enhanced Proxies (PEPs), which breaks the end-to-end semantic of TCP connections. However, it considers a cross-layer design between the network layer and the transport layer to enhance TCP transmission while providing them with QoS guarantees. Here, a modified TCP variant called XPLIT-TCP is proposed to send data through the forward and the return channel. XPLIT-TCP uses two control loops (the buffer occupancy and the service rate to provide optimized congestion control functions. The proposed XPLIT architecture is evaluated using the NS-2 simulation tool. Finally, the third contribution of this thesis consists on the development of a unified architecture to provide QoS guarantees based on cross-layer design over broadband satellite systems. It adopts the enhancements proposed by the QoSatArt architecture working at the network layer, in combination with the enhancements proposed by the XPLIT architecture working at the transport layer.Actualmente, los satélites Geoestacionarios (GEO) juegan un papel muy importante en la provisión de servicios IP. Esta infraestructura permite proveer ubicuidad y acceso de banda ancha, haciendo posible alcanzar poblaciones dispersas en zonas remotas donde la infraestructura terrestre es inexistente. Sin embargo, en la provisión de aplicaciones como Voz sobre IP (VoIP) y servicios multimedia, es importante considerar el tratamiento diferenciado de paquetes a través de la red satelital. Esta diferenciación debe considerar no solo los requerimientos de Calidad de Servicio (QoS) que especifican las prioridades de los paquetes a través de los nodos de red, si no también el ancho de banda asignado para garantizar su transporte. En este contexto, la provisión de garantías de QoS sobre satélites GEO es una de las Principales áreas de investigación de organizaciones como la Agencia Espacial Europea (ESA) persiguen. Esto se debe principalmente ya que dichas organizaciones requieren la explotación continua de sus activos, dado que lanzar un nuevo satélite al espacio representa costos excesivos. Como resultado, el soporte de servicios IP con calidad de servicio sobre la infraestructura satelital actual es de vital importancia. En esta tesis doctoral se presentan varias contribuciones para el soporte a la Calidad de Servicio en redes DVB-S2/RCS satelitales de banda ancha. Las contribuciones propuestas se basan principalmente en el diseño ”cross-layer” siguiendo el modelo de capas definido y estandarizado en las especificaciones ETSI TR 102 157 [ETS03] y 462 [10205]. Las contribuciones propuestas consideran las limitaciones presentes de los sistemas satelitales GEO como lo son el retardo de propagación, la perdida de paquetes y las variaciones de ancho de banda causados por eventos atmosféricos. La primera contribución propone QoSatArt, una arquitectura definida para mejorar el soporte a la QoS. Esta arquitectura considera las variaciones en la capa física debido a la presencia de eventos de lluvia para priorizar los niveles de QoS. El diseño se desarrolla en el gateway e incluye las especificaciones de los principales elementos funcionales y mecanismos para garantizar la QoS y minimizar el retardo presente en la capa de aplicación. Aquí, se propone un diseño ”cross-layer” entre la capa física y la capa de red, con el objetivo de reforzar las especificaciones de QoS considerando el ancho de banda disponible. La arquitectura QoSatArt es simulada y evaluada empleando la herramienta de simulación NS-2. Adicionalmente, un análisis de desempeño de diversas variantes de TCP (Transmission Control Protocol) es realizado con el objetivo de encontrar la variante de TCP más adecuada para trabajar en un ambiente con QoS como QoSatArt. La segunda contribución propone XPLIT, una arquitectura desarrollada para mejorar las transmisiones TCP con QoS en un sistema satelital DVB-S2/RCS. Complementario a QoSatArt, XPLIT emplea PEPs (Performance Enhanced Proxies), afectando la semántica end-to-end de las conexiones TCP. Sin embargo, XPLIT considera un diseño ”cross-layer” entre la capa de red y la capa de transporte con el objetivo de mejorar las transmisiones TCP considerando los parámetros de QoS como la ocupación de la cola y la tasa de transmisión (_i, _i). Aquí, se propone el uso de una nueva variante de TCP es propuesta llamada XPLIT-TCP, que usa dos bucles para proveer funciones mejoradas en el control de congestión. La arquitectura XPLIT es simulada y evaluada empleando la herramienta de simulación NS-2. Finalmente, la tercera contribución de esta tesis consiste en el desarrollo de un arquitectura unificada para el soporte a la QoS en redes satelitales de banda ancha basada en técnicas ”cross-layer”. Esta arquitectura adopta las mejoras propuestas por QoSatArt en la capa de red en combinación con las mejoras propuestas por XPLIT en la capa de transporte

Cooperative End-to-end Congestion Control in Heterogeneous Wireless Networks

Sharing the resources of multiple wireless networks with overlapped coverage areas has a potential of improving the transmission throughput. However, in the existing frameworks, the improvement cannot be achieved in congestion scenarios because of independent congestion control procedures among the end-to-end paths. Although various network characteristics make the congestion control complex, this variety can be useful in congestion avoidance if the networks cooperate with each other. When congestion happens in an end-to-end path, it is inevitable to have a packet transmission rate less than the minimum requested rate due to congestion window size adjustments. Cooperation among networks can help to avoid this problem for better service quality. When congestion is predicted for one path, some of the on-going packets can be sent over other paths instead of the congested path. In this way, the traffic can be shifted from a congested network to others, and the overall transmission throughput does not degrade in a congestion scenario. However, cooperation is not always advantageous since the throughput of cooperative transmission in an uncongested scenario can be less than that of non-cooperative transmission due to cooperation costs such as cooperation setup time, additional signalling for cooperation, and out-of-order packet reception. In other words, a trade-off exists between congestion avoidance and cooperation cost. Thus, cooperation should be triggered only when it is beneficial according to congestion level measurements. In this research, our aim is to develop an efficient cooperative congestion control scheme for a heterogeneous wireless environment. To this end, a cooperative congestion control algorithm is proposed, in which the state of an end-to-end path is provided at the destination terminal by measuring the queuing delay and estimating the congestion level. The decision on when to start/stop cooperation is made based on the network characteristics, instantaneous traffic condition, and the requested quality of service (QoS). Simulation results demonstrate the throughput improvement of the proposed scheme over non-cooperative congestion control.1 yea