116 research outputs found
Quality-Oriented Mobility Management for Multimedia Content Delivery to Mobile Users
The heterogeneous wireless networking environment determined by the latest developments in wireless access technologies promises a high level of communication resources for mobile
computational devices. Although the communication resources provided, especially referring to bandwidth, enable multimedia streaming to mobile users, maintaining a high user perceived quality is still a challenging task. The main factors which affect quality in multimedia streaming over wireless networks are mainly the error-prone nature of the wireless channels and the user mobility. These factors determine a high level of dynamics of wireless communication resources, namely variations in throughput and packet loss as well as network availability and delays in delivering the data packets. Under these conditions maintaining a high level of quality, as perceived by the user, requires a quality oriented mobility management scheme. Consequently we propose the Smooth Adaptive Soft-Handover Algorithm, a novel quality oriented handover management scheme which unlike other similar solutions, smoothly transfer the data traffic from one network to another using multiple simultaneous connections. To estimate the capacity of each connection the novel Quality of Multimedia Streaming (QMS) metric is proposed. The QMS metric aims at offering maximum flexibility and efficiency allowing the applications to fine tune the behavior of the handover algorithm. The current simulation-based performance evaluation clearly shows the better
performance of the proposed Smooth Adaptive Soft-Handover Algorithm as compared with other handover solutions. The evaluation was performed in various scenarios including
multiple mobile hosts performing handover simultaneously, wireless networks with variable overlapping areas, and various network congestion levels
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Effective video multicast over wireless internet
With the rapid growth of wireless networks and great success of Internet video, wireless video services are expected to be widely deployed in the near future. As different types of wireless networks are converging into all IP networks, i.e., the Internet, it is important to study video delivery over the wireless Internet. This paper proposes a novel end-system based adaptation protocol calledWireless Hybrid Adaptation Layered Multicast (WHALM) protocol for layered video multicast over wireless Internet. In WHALM the sender dynamically collects bandwidth distribution from the receivers and uses an optimal layer rate allocation mechanism to reduce the mismatches between the coarse-grained layer subscription levels and the heterogeneous and dynamic rate requirements from the receivers, thus maximizing the degree of satisfaction of all the receivers in a multicast session. Based on sampling theory and theory of probability, we reduce the required number of bandwidth feedbacks to a reasonable degree and use a scalable feedback mechanism to control the feedback process practically. WHALM is also tuned to perform well in wireless networks by integrating an end-to-end loss differentiation algorithm (LDA) to differentiate error losses from congestion losses at the receiver side. With a series of simulation experiments over NS platform, WHALM has been proved to be able to greatly improve the degree of satisfaction of all the receivers while avoiding congestion collapse on the wireless Internet
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Multimedia delivery in the future internet
The term “Networked Media” implies that all kinds of media including text, image, 3D graphics, audio
and video are produced, distributed, shared, managed and consumed on-line through various networks,
like the Internet, Fiber, WiFi, WiMAX, GPRS, 3G and so on, in a convergent manner [1]. This white
paper is the contribution of the Media Delivery Platform (MDP) cluster and aims to cover the Networked
challenges of the Networked Media in the transition to the Future of the Internet.
Internet has evolved and changed the way we work and live. End users of the Internet have been confronted
with a bewildering range of media, services and applications and of technological innovations concerning
media formats, wireless networks, terminal types and capabilities. And there is little evidence that the pace
of this innovation is slowing. Today, over one billion of users access the Internet on regular basis, more
than 100 million users have downloaded at least one (multi)media file and over 47 millions of them do so
regularly, searching in more than 160 Exabytes1 of content. In the near future these numbers are expected
to exponentially rise. It is expected that the Internet content will be increased by at least a factor of 6, rising
to more than 990 Exabytes before 2012, fuelled mainly by the users themselves. Moreover, it is envisaged
that in a near- to mid-term future, the Internet will provide the means to share and distribute (new)
multimedia content and services with superior quality and striking flexibility, in a trusted and personalized
way, improving citizens’ quality of life, working conditions, edutainment and safety.
In this evolving environment, new transport protocols, new multimedia encoding schemes, cross-layer inthe
network adaptation, machine-to-machine communication (including RFIDs), rich 3D content as well as
community networks and the use of peer-to-peer (P2P) overlays are expected to generate new models of
interaction and cooperation, and be able to support enhanced perceived quality-of-experience (PQoE) and
innovative applications “on the move”, like virtual collaboration environments, personalised services/
media, virtual sport groups, on-line gaming, edutainment. In this context, the interaction with content
combined with interactive/multimedia search capabilities across distributed repositories, opportunistic P2P
networks and the dynamic adaptation to the characteristics of diverse mobile terminals are expected to
contribute towards such a vision.
Based on work that has taken place in a number of EC co-funded projects, in Framework Program 6 (FP6)
and Framework Program 7 (FP7), a group of experts and technology visionaries have voluntarily
contributed in this white paper aiming to describe the status, the state-of-the art, the challenges and the way
ahead in the area of Content Aware media delivery platforms
Content-Aware Multimedia Communications
The demands for fast, economic and reliable dissemination of multimedia
information are steadily growing within our society. While people and
economy increasingly rely on communication technologies, engineers still
struggle with their growing complexity.
Complexity in multimedia communication originates from several sources. The
most prominent is the unreliability of packet networks like the Internet.
Recent advances in scheduling and error control mechanisms for streaming
protocols have shown that the quality and robustness of multimedia delivery
can be improved significantly when protocols are aware of the content they
deliver. However, the proposed mechanisms require close cooperation between
transport systems and application layers which increases the overall system
complexity. Current approaches also require expensive metrics and focus on
special encoding formats only. A general and efficient model is missing so
far.
This thesis presents efficient and format-independent solutions to support
cross-layer coordination in system architectures. In particular, the first
contribution of this work is a generic dependency model that enables
transport layers to access content-specific properties of media streams,
such as dependencies between data units and their importance. The second
contribution is the design of a programming model for streaming
communication and its implementation as a middleware architecture. The
programming model hides the complexity of protocol stacks behind simple
programming abstractions, but exposes cross-layer control and monitoring
options to application programmers. For example, our interfaces allow
programmers to choose appropriate failure semantics at design time while
they can refine error protection and visibility of low-level errors at
run-time.
Based on some examples we show how our middleware simplifies the
integration of stream-based communication into large-scale application
architectures. An important result of this work is that despite cross-layer
cooperation, neither application nor transport protocol designers
experience an increase in complexity. Application programmers can even
reuse existing streaming protocols which effectively increases system
robustness.Der Bedarf unsere Gesellschaft nach kostengünstiger und
zuverlässiger
Kommunikation wächst stetig. Während wir uns selbst immer mehr von modernen
Kommunikationstechnologien abhängig machen, müssen die Ingenieure dieser
Technologien sowohl den Bedarf nach schneller Einführung neuer Produkte
befriedigen als auch die wachsende Komplexität der Systeme beherrschen.
Gerade die Übertragung multimedialer Inhalte wie Video und Audiodaten ist
nicht trivial. Einer der prominentesten Gründe dafür ist die
Unzuverlässigkeit heutiger Netzwerke, wie z.B.~dem Internet. Paketverluste
und schwankende Laufzeiten können die Darstellungsqualität massiv
beeinträchtigen. Wie jüngste Entwicklungen im Bereich der
Streaming-Protokolle zeigen, sind jedoch Qualität und Robustheit der
Übertragung effizient kontrollierbar, wenn Streamingprotokolle
Informationen über den Inhalt der transportierten Daten ausnutzen.
Existierende Ansätze, die den Inhalt von Multimediadatenströmen
beschreiben, sind allerdings meist auf einzelne Kompressionsverfahren
spezialisiert und verwenden berechnungsintensive Metriken. Das reduziert
ihren praktischen Nutzen deutlich. Außerdem erfordert der
Informationsaustausch eine enge Kooperation zwischen Applikationen und
Transportschichten. Da allerdings die Schnittstellen aktueller
Systemarchitekturen nicht darauf vorbereitet sind, müssen entweder die
Schnittstellen erweitert oder alternative Architekturkonzepte geschaffen
werden. Die Gefahr beider Varianten ist jedoch, dass sich die Komplexität
eines Systems dadurch weiter erhöhen kann.
Das zentrale Ziel dieser Dissertation ist es deshalb,
schichtenübergreifende Koordination bei gleichzeitiger Reduzierung der
Komplexität zu erreichen. Hier leistet die Arbeit zwei Beträge zum
aktuellen Stand der Forschung. Erstens definiert sie ein universelles
Modell zur Beschreibung von Inhaltsattributen, wie Wichtigkeiten und
Abhängigkeitsbeziehungen innerhalb eines Datenstroms. Transportschichten
können dieses Wissen zur effizienten Fehlerkontrolle verwenden. Zweitens
beschreibt die Arbeit das Noja Programmiermodell für multimediale
Middleware. Noja definiert Abstraktionen zur Übertragung und Kontrolle
multimedialer Ströme, die die Koordination von Streamingprotokollen mit
Applikationen ermöglichen. Zum Beispiel können Programmierer geeignete
Fehlersemantiken und Kommunikationstopologien auswählen und den konkreten
Fehlerschutz dann zur Laufzeit verfeinern und kontrolliere
LTE Optimization and Resource Management in Wireless Heterogeneous Networks
Mobile communication technology is evolving with a great pace. The development of the Long Term Evolution (LTE) mobile system by 3GPP is one of the milestones in this direction. This work highlights a few areas in the LTE radio access network where the proposed innovative mechanisms can substantially improve overall LTE system performance. In order to further extend the capacity of LTE networks, an integration with the non-3GPP networks (e.g., WLAN, WiMAX etc.) is also proposed in this work. Moreover, it is discussed how bandwidth resources should be managed in such heterogeneous networks. The work has purposed a comprehensive system architecture as an overlay of the 3GPP defined SAE architecture, effective resource management mechanisms as well as a Linear Programming based analytical solution for the optimal network resource allocation problem. In addition, alternative computationally efficient heuristic based algorithms have also been designed to achieve near-optimal performance
Smart PIN: performance and cost-oriented context-aware personal information network
The next generation of networks will involve interconnection of heterogeneous individual
networks such as WPAN, WLAN, WMAN and Cellular network, adopting the IP as common infrastructural protocol and providing virtually always-connected network. Furthermore,
there are many devices which enable easy acquisition and storage of information as pictures, movies, emails, etc. Therefore, the information overload and divergent content’s
characteristics make it difficult for users to handle their data in manual way. Consequently, there is a need for personalised automatic services which would enable data exchange across heterogeneous network and devices. To support these personalised services, user centric approaches
for data delivery across the heterogeneous network are also required.
In this context, this thesis proposes Smart PIN - a novel performance and cost-oriented context-aware Personal Information Network. Smart PIN's architecture is detailed including its network, service and management components. Within the service component, two novel schemes for efficient delivery of context and content data are proposed:
Multimedia Data Replication Scheme (MDRS) and Quality-oriented Algorithm for Multiple-source Multimedia Delivery (QAMMD).
MDRS supports efficient data accessibility among distributed devices using data replication which is based on a utility function and a minimum data set. QAMMD employs a buffer underflow avoidance scheme for streaming, which achieves high multimedia quality without content adaptation to network conditions. Simulation models for MDRS and
QAMMD were built which are based on various heterogeneous network scenarios. Additionally a multiple-source streaming based on QAMMS was implemented as a prototype and tested in an emulated network environment. Comparative tests show that MDRS and QAMMD perform significantly better than other approaches
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
Techniques for End-to-End Tcp Performance Enhancement Over Wireless Networks
Today’s wireless network complexity and the new applications from various user devices call for an in-depth understanding of the mutual performance impact of networks and applications. It includes understanding of the application traffic and network layer protocols to enable end-to-end application performance enhancements over wireless networks. Although Transport Control Protocol (TCP) behavior over wireless networks is well known, it remains as one of the main drivers which may significantly impact the user experience through application performance as well as the network resource utilization, since more than 90% of the internet traffic uses TCP in both wireless and wire-line networks. In this dissertation, we employ application traffic measurement and packet analysis over a commercial Long Term Evolution (LTE) network combined with an in-depth LTE protocol simulation to identify three critical problems that may negatively affect the application performance and wireless network resource utilization: (i) impact of the wireless MAC protocol on the TCP throughput performance, (ii) impact of applications on network resource utilization, and (iii) impact of TCP on throughput performance over wireless networks. We further propose four novel mechanisms to improve the end-to-end application and wireless system performance: (i) an enhanced LTE uplink resource allocation mechanism to reduce network delay and help prevent a TCP timeout, (ii) a new TCP snooping mechanism, which according to our experiments, can save about 20% of system resources by preventing unnecessary video packet transmission through the air interface, and (iii) two Split-TCP protocols: an Enhanced Split-TCP (ES-TCP) and an Advanced Split-TCP (AS-TCP), which significantly improve the application throughput without breaking the end-to-end TCP semantics. Experimental results show that the proposed ES-TCP and AS-TCP protocols can boost the TCP throughput by more than 60% in average, when exercised over a 4G LTE network. Furthermore, the TCP throughput performance improvement may be even superior to 200%, depending on network and usage conditions. We expect that these proposed Split-TCP protocol enhancements, together with the new uplink resource allocation enhancement and the new TCP snooping mechanism may provide even greater performance gains when more advanced radio technologies, such as 5G, are deployed. Thanks to their superior resource utilization efficiency, such advanced radio technologies will put to greater use the techniques and protocol enhancements disclosed through this dissertation
Handoff management for infotainment services over vehicular networks
Intelligent Transportation Systems (ITS) has impulsed the vehicular communications at the present time. The vehicular communications field is a hot research topic and is attracting a great interest in the automotive industry and telecommunications. There are essentially two main lines of work: (1) communication services related to road safety and traffic information; and (2) information and entertainment services, also named infotainment services. These latter services include both transmitting multimedia (voice over IP, streaming, on-line gaming, etc.) and classic data services (e-mail, access to private networks, web browsing, file sharing, etc.). In this thesis we will focus on these infotainment services because further research in this immature research field is necessary and, until nowadays, the main effort of the research community regarding vehicular communication has been focused on road safety and traffic information.
Vehicular nodes need to be reached from the Internet and vice versa to be able to access to infotainment services. While vehicles move along the road infrastructure, they change their wireless point of attachment to the network. During this process, connectivity breaks down until the vehicle is connected again to a new road side unit in its area. This disconnection causes a disruption in the communications. Fast handoffs are a crucial requirement for vehicular networks to avoid long disruption times, since the high speed of vehicular nodes involves suffering a lot of handoffs during an Internet connection.
This thesis is focused on Vehicular-to-Infrastructure (V2I) real-time infotainment services. The main contributions of this thesis are: i) a new testing framework for V2I communications to be able to test infotainment services in an easy way; ii) the analysis of the deployability of infotainment video services in vehicular networks using mobility protocols; and iii) the development of a new TCP architecture that will provide a better performance for all TCP-based infotainment services in a vehicular scenario with handoffs.
In this thesis, firstly, we propose a new testing framework for vehicular infotainment applications. This framework is a vehicular emulation platform that allows testing real applications installed on Linux virtual machines. Using emulation, we are able to evaluate the performance of real applications with real-time requirements, so we can test multimedia applications used to offer infotainment services in vehicular scenarios in a straightforward way.
Secondly, using the testing framework implemented in the first part of the thesis, we have done a performance evaluation of an infotainment service. Among these services, we think that video on demand services on highways will be interesting for users, and generate revenue to network operators. So we evaluated how network-layer handoffs can limit the deployment of a video streaming service. According to the results obtained, driving at high speeds will be an issue for a correct playback of video content, even using fast handoffs techniques.
Finally, we developed a new TCP architecture to enhance performance during handoffs. Most of the non-safety services on ITS rely on the Transport Control Protocol (TCP), one of the core protocols of the Internet Protocol Suite. However there exists several issues related to TCP and mobility that can affect to TCP performance, and these issues are particularly important in vehicular networks due to its high mobility. Using new IEEE 802.21 MIH services, we propose a new TCP architecture that is able to anticipate handoffs, permitting to resume the communication after a handoff, avoiding long delays caused by TCP issues and adapting the TCP parameters to the new characteristics of the network. Using the architecture proposed, the performance of TCP is enhanced, getting a higher overall throughput and avoiding TCP fairness issues between users.Els Sistemes de Transport Intel·ligents (ITS) han impulsat les comunicacions vehiculars en l'actualitat. Les comunicacions vehiculars és un camp d'investigació de moda, i està atraient un gran interès en la indústria automobilística i de les telecomunicacions. En el camp de les comunicacions vehiculars, hi ha principalment dues línies de treball: (1) serveis de comunicacions relacionats amb la seguretat viària i la informació del trànsit; i (2) serveis d'informació i entreteniment, també anomenats serveis d'infotainment. Aquests últims inclouen tant serveis multimèdia (veu sobre IP, streaming, jocs on-line, etc.), com serveis clàssics de dades (correu electrònic, accés a xarxes privades, navegació web, compartir arxius, etc.). En aquesta tesi ens centrarem en aquests serveis d'infotainment ja que és necessari aprofundir en la investigació per aquests tipus de serveis, ja que, fins avui, els esforços de la comunitat científica en el camp de les comunicacions vehiculars s'ha centrat en els serveis relacionats amb la seguretat viària i la informació del trànsit. Els nodes vehiculars necessiten tenir connexió a Internet per a poder tenir accés als serveis d'infotainment. Mentre els vehicles estan en moviment a través de la xarxa viària, els vehicles han d'anar canviant el punt de connexió sense fils amb la xarxa. Durant aquest procés de canvi de punt de connexió, anomenat handoff, es perd la connectivitat fins que el vehicle es reconnecta a un altre punt de connexió viària prop de la seva àrea. Aquesta desconnexió causa interrupcions en les comunicacions. Uns handoffs ràpids són bàsics a les xarxes vehiculars per a evitar llargs períodes d'interrupció durant les comunicacions, ja que la gran velocitat a la que es mouen els nodes vehiculars significa un gran nombre de handoffs durant una connexió a Internet. Aquesta tesi es centra en serveis d'infotaiment en temps real per a comunicacions Vehicle-a-Infraestructura (V2I). Les principals contribucions d'aquesta tesi son: i) un nou marc de proves per a les comunicacions (V2I) per a poder provar serveis d'infotainment d'una manera fàcil; ii) l'anàlisi de la viabilitat del desplegament de serveis d'infotainment de vídeo en xarxes vehiculars utilitzant protocols de mobilitat IP; i iii) el desenvolupament d'una nova arquitectura TCP que proporciona un millor funcionament per a tots aquells serveis d'infotainment basats en el protocol TCP en un escenari vehicular amb handoffs. En aquesta tesi, primer proposem un nou marc de proves per a aplicacions vehiculars d'infotainment. Aquest marc és una plataforma d'emulació vehicular que permet provar aplicacions reals instal·lades en màquines virtuals Linux. Utilitzant l'emulació, som capaços d'avaluar el rendiment d'aplicacions reals amb característiques de temps real. D'aquesta manera es poden avaluar aplicacions multimèdia utilitzades per oferir serveis d'infotainment d'una forma senzilla en escenaris vehiculars. Segon, utilitzant el marc de prova implementat en la primera part de la tesi, hem avaluat el rendiment d'un servei d'infotainment. Entre aquest tipus de servei, creem que els serveis de vídeo sota demanda en autopistes/autovies serà interessant pels usuaris i generarà beneficis per als operadors de la xarxa. Per tant, hem avaluat com els handoffs a nivell de la capa de xarxa poden limitar el desplegament d'un servei de streaming de vídeo sota demanda. D'acord amb els resultats obtinguts, conduir a grans velocitats podria ser un problema per a poder reproduir un vídeo correctament, tot i utilitzar tècniques de handoffs ràpids. Finalment, hem desenvolupat una nova arquitectura TCP per a millorar el rendiment del protocol durant els handoffs. La majoria dels serveis d'infotainment utilitzen el Protocol de Control de Transport (TCP), un dels principals protocols de la pila de protocols d'Internet. Però existeixen forces problemes relacionats amb l'ús de TCP i la mobilitat que n'afecta el rendiment, i aquests problemes són particular
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