834 resultados para wireless TCP
Resumo:
O crescimento dos serviços de banda-larga em redes de comunicações móveis tem provocado uma demanda por dados cada vez mais rápidos e de qualidade. A tecnologia de redes móveis chamada LTE (Long Term Evolution) ou quarta geração (4G) surgiu com o objetivo de atender esta demanda por acesso sem fio a serviços, como acesso à Internet, jogos online, VoIP e vídeo conferência. O LTE faz parte das especificações do 3GPP releases 8 e 9, operando numa rede totalmente IP, provendo taxas de transmissão superiores a 100 Mbps (DL), 50 Mbps (UL), baixa latência (10 ms) e compatibilidade com as versões anteriores de redes móveis, 2G (GSM/EDGE) e 3G (UMTS/HSPA). O protocolo TCP desenvolvido para operar em redes cabeadas, apresenta baixo desempenho sobre canais sem fio, como redes móveis celulares, devido principalmente às características de desvanecimento seletivo, sombreamento e às altas taxas de erros provenientes da interface aérea. Como todas as perdas são interpretadas como causadas por congestionamento, o desempenho do protocolo é ruim. O objetivo desta dissertação é avaliar o desempenho de vários tipos de protocolo TCP através de simulações, sob a influência de interferência nos canais entre o terminal móvel (UE User Equipment) e um servidor remoto. Para isto utilizou-se o software NS3 (Network Simulator versão 3) e os protocolos TCP Westwood Plus, New Reno, Reno e Tahoe. Os resultados obtidos nos testes mostram que o protocolo TCP Westwood Plus possui um desempenho melhor que os outros. Os protocolos TCP New Reno e Reno tiveram desempenho muito semelhante devido ao modelo de interferência utilizada ter uma distribuição uniforme e, com isso, a possibilidade de perdas de bits consecutivos é baixa em uma mesma janela de transmissão. O TCP Tahoe, como era de se esperar, apresentou o pior desempenho dentre todos, pois o mesmo não possui o mecanismo de fast recovery e sua janela de congestionamento volta sempre para um segmento após o timeout. Observou-se ainda que o atraso tem grande importância no desempenho dos protocolos TCP, mas até do que a largura de banda dos links de acesso e de backbone, uma vez que, no cenário testado, o gargalo estava presente na interface aérea. As simulações com erros na interface aérea, introduzido com o script de fading (desvanecimento) do NS3, mostraram que o modo RLC AM (com reconhecimento) tem um desempenho melhor para aplicações de transferência de arquivos em ambientes ruidosos do que o modo RLC UM sem reconhecimento.
Resumo:
The Transmission Control Protocol (TCP) has been the protocol of choice for many Internet applications requiring reliable connections. The design of TCP has been challenged by the extension of connections over wireless links. We ask a fundamental question: What is the basic predictive power of TCP of network state, including wireless error conditions? The goal is to improve or readily exploit this predictive power to enable TCP (or variants) to perform well in generalized network settings. To that end, we use Maximum Likelihood Ratio tests to evaluate TCP as a detector/estimator. We quantify how well network state can be estimated, given network response such as distributions of packet delays or TCP throughput that are conditioned on the type of packet loss. Using our model-based approach and extensive simulations, we demonstrate that congestion-induced losses and losses due to wireless transmission errors produce sufficiently different statistics upon which an efficient detector can be built; distributions of network loads can provide effective means for estimating packet loss type; and packet delay is a better signal of network state than short-term throughput. We demonstrate how estimation accuracy is influenced by different proportions of congestion versus wireless losses and penalties on incorrect estimation.
Resumo:
TCP performance degrades when end-to-end connections extend over wireless connections-links which are characterized by high bit error rate and intermittent connectivity. Such link characteristics can significantly degrade TCP performance as the TCP sender assumes wireless losses to be congestion losses resulting in unnecessary congestion control actions. Link errors can be reduced by increasing transmission power, code redundancy (FEC) or number of retransmissions (ARQ). But increasing power costs resources, increasing code redundancy reduces available channel bandwidth and increasing persistency increases end-to-end delay. The paper proposes a TCP optimization through proper tuning of power management, FEC and ARQ in wireless environments (WLAN and WWAN). In particular, we conduct analytical and numerical analysis taking into "wireless-aware" TCP) performance under different settings. Our results show that increasing power, redundancy and/or retransmission levels always improves TCP performance by reducing link-layer losses. However, such improvements are often associated with cost and arbitrary improvement cannot be realized without paying a lot in return. It is therefore important to consider some kind of net utility function that should be optimized, thus maximizing throughput at the least possible cost.
Resumo:
We discuss the design principles of TCP within the context of heterogeneous wired/wireless networks and mobile networking. We identify three shortcomings in TCP's behavior: (i) the protocol's error detection mechanism, which does not distinguish different types of errors and thus does not suffice for heterogeneous wired/wireless environments, (ii) the error recovery, which is not responsive to the distinctive characteristics of wireless networks such as transient or burst errors due to handoffs and fading channels, and (iii) the protocol strategy, which does not control the tradeoff between performance measures such as goodput and energy consumption, and often entails a wasteful effort of retransmission and energy expenditure. We discuss a solution-framework based on selected research proposals and the associated evaluation criteria for the suggested modifications. We highlight an important angle that did not attract the required attention so far: the need for new performance metrics, appropriate for evaluating the impact of protocol strategies on battery-powered devices.
Resumo:
End-to-End differentiation between wireless and congestion loss can equip TCP control so it operates effectively in a hybrid wired/wireless environment. Our approach integrates two techniques: packet loss pairs (PLP) and Hidden Markov Modeling (HMM). A packet loss pair is formed by two back-to-back packets, where one packet is lost while the second packet is successfully received. The purpose is for the second packet to carry the state of the network path, namely the round trip time (RTT), at the time the other packet is lost. Under realistic conditions, PLP provides strong differentiation between congestion and wireless type of loss based on distinguishable RTT distributions. An HMM is then trained so observed RTTs can be mapped to model states that represent either congestion loss or wireless loss. Extensive simulations confirm the accuracy of our HMM-based technique in classifying the cause of a packet loss. We also show the superiority of our technique over the Vegas predictor, which was recently found to perform best and which exemplifies other existing loss labeling techniques.
Resumo:
Modern cellular channels in 3G networks incorporate sophisticated power control and dynamic rate adaptation which can have a significant impact on adaptive transport layer protocols, such as TCP. Though there exists studies that have evaluated the performance of TCP over such networks, they are based solely on observations at the transport layer and hence have no visibility into the impact of lower layer dynamics, which are a key characteristic of these networks. In this work, we present a detailed characterization of TCP behavior based on cross-layer measurement of transport, as well as RF and MAC layer parameters. In particular, through a series of active TCP/UDP experiments and measurement of the relevant variables at all three layers, we characterize both, the wireless scheduler in a commercial CDMA2000 network and its impact on TCP dynamics. Somewhat surprisingly, our findings indicate that the wireless scheduler is mostly insensitive to channel quality and sector load over short timescales and is mainly affected by the transport layer data rate. Furthermore, we empirically demonstrate the impact of the wireless scheduler on various TCP parameters such as the round trip time, throughput and packet loss rate.
Resumo:
Modern cellular channels in 3G networks incorporate sophisticated power control and dynamic rate adaptation which can have significant impact on adaptive transport layer protocols, such as TCP. Though there exists studies that have evaluated the performance of TCP over such networks, they are based solely on observations at the transport layer and hence have no visibility into the impact of lower layer dynamics, which are a key characteristic of these networks. In this work, we present a detailed characterization of TCP behavior based on cross-layer measurement of transport layer, as well as RF and MAC layer parameters. In particular, through a series of active TCP/UDP experiments and measurement of the relevant variables at all three layers, we characterize both, the wireless scheduler and the radio link protocol in a commercial CDMA2000 network and assess their impact on TCP dynamics. Somewhat surprisingly, our findings indicate that the wireless scheduler is mostly insensitive to channel quality and sector load over short timescales and is mainly affected by the transport layer data rate. Furthermore, with the help of a robust correlation measure, Normalized Mutual Information, we were able to quantify the impact of the wireless scheduler and the radio link protocol on various TCP parameters such as the round trip time, throughput and packet loss rate.
Resumo:
This paper investigates a dynamic buffer man-agement scheme for QoS control of multimedia services in be-yond 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 multi-plexed 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 guar-antee the end-to-end QoS of the real-time streaming flow, whilst simultaneously protecting the non real-time flow from starva-tion resulting in improved end-to-end throughput performance
Resumo:
This paper presents and investigates a dynamic
buffer management scheme for QoS control of multimedia
services in a 3.5G wireless system i.e. the High Speed Downlink
Packet Access (HSDPA). HSDPA was introduced to enhance
UMTS for high-speed packet switched services. With HSDPA,
packet scheduling and HARQ mechanisms in the base station
require data buffering at the air interface thus introducing a
potential bottleneck to end-to-end communication. Hence, for
multimedia services with multiplexed parallel diverse flows
such as video and data in the same end-user session, buffer
management schemes in the base station are essential to support
end-to-end QoS provision. We propose a dynamic buffer management
scheme for HSDPA multimedia sessions with aggregated real-time and non real-time flows in the paper. The end-to-end performance impact of the scheme is evaluated with an example multimedia session comprising a real-time streaming
flow concurrent with TCP-based non real-time flow via extensive HSDPA simulations. Results demonstrate that the scheme can guarantee the end-to-end QoS of the real-time streaming flow, whilst simultaneously protecting non real-time flow from starvation resulting in improved end-to-end throughput performance
Resumo:
Interesting wireless networking scenarios exist wherein network services must be guaranteed in a dynamic fashion for some priority users. For example, in disaster recovery, members need to be able to quickly block other users in order to gain sole use of the radio channel. As it is not always feasible to physically switch off other users, we propose a new approach, termed selective packet destruction (SPD) to ensure service for priority users. A testbed for SPD has been created, based on the Rice University Wireless open-Access Research Platform and been used to examine the feasibility of our approach. Results from the testbed are presented to demonstrate the feasibility of SPD and show how a balance between performance and acknowledgement destruction rate can be achieved. A 90% reduction in TCP & UDP traffic is achieved for a 75% MAC ACK destruction rate.
Resumo:
Congestion control in wireless networks is an important and open issue. Previous research has proven the poor performance of the Transport Control Protocol (TCP) in such networks. The factors that contribute to the poor performance of TCP in wireless environments concern its unsuitability to identify/detect and react properly to network events, its TCP window based ow control algorithm that is not suitable for the wireless channel, and the congestion collapse due to mobility. New rate based mechanisms have been proposed to mitigate TCP performance in wired and wireless networks. However, these mechanisms also present poor performance, as they lack of suitable bandwidth estimation techniques for multi-hop wireless networks. It is thus important to improve congestion control performance in wireless networks, incorporating components that are suitable for wireless environments. A congestion control scheme which provides an e - cient and fair sharing of the underlying network capacity and available bandwidth among multiple competing applications is crucial to the definition of new e cient and fair congestion control schemes on wireless multi-hop networks. The Thesis is divided in three parts. First, we present a performance evaluation study of several congestion control protocols against TCP, in wireless mesh and ad-hoc networks. The obtained results show that rate based congestion control protocols need an eficient and accurate underlying available bandwidth estimation technique. The second part of the Thesis presents a new link capacity and available bandwidth estimation mechanism denoted as rt-Winf (real time wireless inference). The estimation is performed in real-time and without the need to intrusively inject packets in the network. Simulation results show that rt-Winf obtains the available bandwidth and capacity estimation with accuracy and without introducing overhead trafic in the network. The third part of the Thesis proposes the development of new congestion control mechanisms to address the congestion control problems of wireless networks. These congestion control mechanisms use cross layer information, obtained by rt-Winf, to accurately and eficiently estimate the available bandwidth and the path capacity over a wireless network path. Evaluation of these new proposed mechanisms, through ns-2 simulations, shows that the cooperation between rt-Winf and the congestion control algorithms is able to significantly increase congestion control eficiency and network performance.
Resumo:
In IP networks, most of packets, that have been dropped, are recovered after the expiration of retransmission timeouts. These can result in unnecessary retransmissions and needless reduction of congestion window. An inappropriate retransmission timeout has a huge impact on TCP performance. In this paper we have proved that CSMA/CA mechanism can cause TCP retransmissions due to CSMA/CA effects. For this we have observed three wireless connections that use CSMA/CA: with good link quality, poor link quality and in presence of cross traffic. The measurements have been performed using real devices. Through tracking of each transmitted packet it is possible to analyze the relation between one-way delay and packet loss probability and the cumulative distribution of distances between peaks of OWDs. The distribution of OWDs and the distances between peaks of OWDs are the most important parameters of tuning TCP retransmission timeout on CSMA/CA networks. A new perspective through investigating the dynamical relation between one-way delay and packet loss ratio depending on the link quality to enhance the TCP performance has been provided.
Resumo:
In dealing with computer networks, these allow the flow of information through the resources of various equipment's. This work describes the implementation through the encapsulation of Protocol DNP3, usually employed in Smart Grid communication, in a simulator of discrete events. The NS-2 is a simulator in open source of network events, that facilitate the development of communication networks scenarios considering the protocols involved, in wireless or wired technologies. The objective of this work is to develop the DNP3 protocol encapsulation over a TCP/IP in the in the discrete event Simulator NS-2, allowing an analysis of behavior of a middle or large network sized in Smart Grid applications. © 2013 IEEE.
Resumo:
Nell'ambito dell'architettura RWMA per l'aumento della qualità del servizio in ambiente wireless, si colloca la ritrasmissione asimmetrica anticipata su wifi. Lo scopo è migliorare ed ottimizzare la comunicazione nel livello datalink tra Access Point e Nodo Mobile per quanto riguarda le connessioni TCP. Il progetto è sviluppato nell'ambiente simulativo OMNET++/INET.
Resumo:
La perdita di pacchetti durante una trasmissione su una rete Wireless influisce in maniera fondamentale sulla qualità del collegamento tra due End-System. Lo scopo del progetto è quello di implementare una tecnica di ritrasmissione asimmetrica anticipata dei pacchetti perduti, in modo da minimizzare i tempi di recupero dati e migliorare la qualità della comunicazione. Partendo da uno studio su determinati tipi di ritrasmissione, in particolare quelli implementati dal progetto ABPS, Always Best Packet Switching, si è maturata l'idea che un tipo di ritrasmissione particolarmente utile potrebbe avvenire a livello Access Point: nel caso in cui la perdita di pacchetti avvenga tra l'AP e il nodo mobile che vi è collegato via IEEE802.11, invece che attendere la ritrasmissione TCP e Effettuata dall'End-System sorgente è lo stesso Access Point che e effettua una ritrasmissione verso il nodo mobile per permettere un veloce recupero dei dati perduti. Tale funzionalità stata quindi concettualmente divisa in due parti, la prima si riferisce all'applicazione che si occupa della bufferizzazione di pacchetti che attraversano l'AP e della loro copia in memoria per poi ritrasmetterli in caso di segnalazione di mancata acquisizione, la seconda riguardante la modifica al kernel che permette la segnalazione anticipata dell'errore. E' già stata sviluppata un'applicazione che prevede una ritrasmissione anticipata da parte dell'Access Point Wifi, cioè una ritrasmissione prima che la notifica di avvenuta perdita raggiunga l'end-point sorgente e appoggiata su un meccanismo di simulazione di Error Detection. Inoltre è stata anche realizzata la ritrasmissione asincrona e anticipata del TCP. Questo documento tratta della realizzazione di una nuova applicazione che fornisca una più effciente versione del buffer di pacchetti e utilizzi il meccanismo di una ritrasmissione asimmetrica e anticipata del TCP, cioè attivare la ritrasmissione su richiesta del TCP tramite notifiche di validità del campo Acknowledgement.