Criteri di valutazione del protocollo TCP

Tesi relativa ai criteri di valutazione del protocollo TCP

Modifiche al protocollo di trasporto LTP per migliorarne le prestazioni in presenza di perdite elevate

La presente tesi si pone come obiettivo quello di analizzare il protocollo LTP (in particolare in ION) e proporre dei miglioramenti utili al caso in cui siano presenti perdite elevate. Piu in dettaglio, una prima parte introduttiva motiva l'inefficacia del TCP/IP in ambito interplanetario e introduce l'architettura DTN Bundle Protocol (Cap.1). La tesi prosegue con la descrizione delle specifiche del protocollo LTP (Cap.2), in particolar modo evidenziando come un bundle venga incapsulato in un blocco LTP, come questo sia successivamente diviso in tanti segmenti LTP e come questi vengano successivamente inviati con il protocollo UDP o con un protocollo analogo. Viene quindi presentata un'approfondita analisi delle penalizzazioni dovute alle perdite dei segmenti LTP, sia di tipo dati che di segnalazione (Cap. 3). Quest'analisi permette di dimostrare la criticita degli effetti delle perdite, in particolare per quello che riguarda i segmenti LTP di segnalazione. Mentre in presenza di perdite basse tali effetti hanno in media un impatto minimo sul tempo di consegna di un blocco LTP (quindi del bundle in esso contenuto), in quanto avvengono raramente, in presenza di perdite elevate rappresentano un collo di bottiglia per il tempo di consegna di un blocco LTP. A tal proposito sono state proposte alcune modifiche che permettono di migliorare le prestazioni di LTP (Cap. 4) compatibilmente con le specifiche RFC in modo da garantire l'interoperabilita con le diverse implementazioni del protocollo. Successivamente nel Cap. 5 viene mostrato come sono state implementate le modifiche proposte in ION 3.4.1. Nel capitolo finale (Cap. 6) sono presenti i risultati numerici relativi ad alcuni test preliminari eseguiti confrontando la versione originale del protocollo con le versioni modificate contenenti i miglioramenti proposti. I test sono risultati molto positivi per elevate perdite, confermando cosi la validita dell'analisi e dei miglioramenti introdotti.

HYBRID TCP/UDP video transport for H.264/AVC content delivery in burst loss networks

In this paper we propose a hybrid TCP/UDP transport, specifically for H.264/AVC encoded video, as a compromise between the delay-prone TCP and the loss-prone UDP. When implementing the hybrid approach, we argue that the playback at the receiver often need not be 100% perfect, provided that a certain level of quality is assured. Reliable TCP is used to transmit and guarantee delivery of the most important packets. This allows use of additional features in the H.264/AVC standard which simultaneously provide an enhanced playback quality, in addition to a reduction in throughput. These benefits are demonstrated through experimental results using a test-bed to emulate the hybrid proposal. We compare the proposed system with other protection methods, such as FEC, and in one case show that for the same bandwidth overhead, FEC is unable to match the performance of the hybrid system in terms of playback quality. Furthermore, we measure the delay associated with our approach, and examine its potential for use as an alternative to the conventional methods of transporting video by either TCP or UDP alone. © 2011 IEEE.

Experimental studies of wireless communication and GNSS kinematic positioning performance in high-mobility vehicle environments

In order to support intelligent transportation system (ITS) road safety applications such as collision avoidance, lane departure warnings and lane keeping, Global Navigation Satellite Systems (GNSS) based vehicle positioning system has to provide lane-level (0.5 to 1 m) or even in-lane-level (0.1 to 0.3 m) accurate and reliable positioning information to vehicle users. However, current vehicle navigation systems equipped with a single frequency GPS receiver can only provide road-level accuracy at 5-10 meters. The positioning accuracy can be improved to sub-meter or higher with the augmented GNSS techniques such as Real Time Kinematic (RTK) and Precise Point Positioning (PPP) which have been traditionally used in land surveying and or in slowly moving environment. In these techniques, GNSS corrections data generated from a local or regional or global network of GNSS ground stations are broadcast to the users via various communication data links, mostly 3G cellular networks and communication satellites. This research aimed to investigate the precise positioning system performances when operating in the high mobility environments. This involves evaluation of the performances of both RTK and PPP techniques using: i) the state-of-art dual frequency GPS receiver; and ii) low-cost single frequency GNSS receiver. Additionally, this research evaluates the effectiveness of several operational strategies in reducing the load on data communication networks due to correction data transmission, which may be problematic for the future wide-area ITS services deployment. These strategies include the use of different data transmission protocols, different correction data format standards, and correction data transmission at the less-frequent interval. A series of field experiments were designed and conducted for each research task. Firstly, the performances of RTK and PPP techniques were evaluated in both static and kinematic (highway with speed exceed 80km) experiments. RTK solutions achieved the RMS precision of 0.09 to 0.2 meter accuracy in static and 0.2 to 0.3 meter in kinematic tests, while PPP reported 0.5 to 1.5 meters in static and 1 to 1.8 meter in kinematic tests by using the RTKlib software. These RMS precision values could be further improved if the better RTK and PPP algorithms are adopted. The tests results also showed that RTK may be more suitable in the lane-level accuracy vehicle positioning. The professional grade (dual frequency) and mass-market grade (single frequency) GNSS receivers were tested for their performance using RTK in static and kinematic modes. The analysis has shown that mass-market grade receivers provide the good solution continuity, although the overall positioning accuracy is worse than the professional grade receivers. In an attempt to reduce the load on data communication network, we firstly evaluate the use of different correction data format standards, namely RTCM version 2.x and RTCM version 3.0 format. A 24 hours transmission test was conducted to compare the network throughput. The results have shown that 66% of network throughput reduction can be achieved by using the newer RTCM version 3.0, comparing to the older RTCM version 2.x format. Secondly, experiments were conducted to examine the use of two data transmission protocols, TCP and UDP, for correction data transmission through the Telstra 3G cellular network. The performance of each transmission method was analysed in terms of packet transmission latency, packet dropout, packet throughput, packet retransmission rate etc. The overall network throughput and latency of UDP data transmission are 76.5% and 83.6% of TCP data transmission, while the overall accuracy of positioning solutions remains in the same level. Additionally, due to the nature of UDP transmission, it is also found that 0.17% of UDP packets were lost during the kinematic tests, but this loss doesn't lead to significant reduction of the quality of positioning results. The experimental results from the static and the kinematic field tests have also shown that the mobile network communication may be blocked for a couple of seconds, but the positioning solutions can be kept at the required accuracy level by setting of the Age of Differential. Finally, we investigate the effects of using less-frequent correction data (transmitted at 1, 5, 10, 15, 20, 30 and 60 seconds interval) on the precise positioning system. As the time interval increasing, the percentage of ambiguity fixed solutions gradually decreases, while the positioning error increases from 0.1 to 0.5 meter. The results showed the position accuracy could still be kept at the in-lane-level (0.1 to 0.3 m) when using up to 20 seconds interval correction data transmission.

A reliable real-time transport protocol for networked control systems over wireless networks

Deploying wireless networks in networked control systems (NCSs) has become more and more popular during the last few years. As a typical type of real-time control systems, an NCS is sensitive to long and nondeterministic time delay and packet losses. However, the nature of the wireless channel has the potential to degrade the performance of NCS networks in many aspects, particularly in time delay and packet losses. Transport layer protocols could play an important role in providing both reliable and fast transmission service to fulfill NCS’s real-time transmission requirements. Unfortunately, none of the existing transport protocols, including the Transport Control Protocol (TCP) and the User Datagram Protocol (UDP), was designed for real-time control applications. Moreover, periodic data and sporadic data are two types of real-time data traffic with different priorities in an NCS. Due to the lack of support for prioritized transmission service, the real-time performance for periodic and sporadic data in an NCS network is often degraded significantly, particularly under congested network conditions. To address these problems, a new transport layer protocol called Reliable Real-Time Transport Protocol (RRTTP) is proposed in this thesis. As a UDP-based protocol, RRTTP inherits UDP’s simplicity and fast transmission features. To improve the reliability, a retransmission and an acknowledgement mechanism are designed in RRTTP to compensate for packet losses. They are able to avoid unnecessary retransmission of the out-of-date packets in NCSs, and collisions are unlikely to happen, and small transmission delay can be achieved. Moreover, a prioritized transmission mechanism is also designed in RRTTP to improve the real-time performance of NCS networks under congested traffic conditions. Furthermore, the proposed RRTTP is implemented in the Network Simulator 2 for comprehensive simulations. The simulation results demonstrate that RRTTP outperforms TCP and UDP in terms of real-time transmissions in an NCS over wireless networks.

Optimizing Multimedia Experience in a Thin Client Environment for a Resource Constrained Processor

In this paper, we study how TCP and UDP flows interact with each other when the end system is a CPU resource constrained thin client. The problem addressed is twofold, 1) the throughput of TCP flows degrades severely in the presence of heavily loaded UDP flows 2) fairness and minimum QoS requirements of UDP are not maintained. First, we identify the factors affecting the TCP throughput by providing an in-depth analysis of end to end delay and packet loss variations. The results obtained from the first part leads us to our second contribution. We propose and study the use of an algorithm that ensures fairness across flows. The algorithm improves the performance of TCP flows in the presence of multiple UDP flows admitted under an admission algorithm and maintains the minimum QoS requirements of the UDP flows. The advantage of the algorithm is that it requires no changes to TCP/IP stack and control is achieved through receiver window control.

Implementation of Selective Packet Destruction on Wireless Open-Access Research Platform

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.

Evaluating and improving the performance of video content distribution in lossy networks

The contributions in this research are split in to three distinct, but related, areas. The focus of the work is based on improving the efficiency of video content distribution in the networks that are liable to packet loss, such as the Internet. Initially, the benefits and limitations of content distribution using Forward Error Correction (FEC) in conjunction with the Transmission Control Protocol (TCP) is presented. Since added FEC can be used to reduce the number of retransmissions, the requirement for TCP to deal with any losses is greatly reduced. When real-time applications are needed, delay must be kept to a minimum, and retransmissions not desirable. A balance, therefore, between additional bandwidth and delays due to retransmissions must be struck. This is followed by the proposal of a hybrid transport, specifically for H.264 encoded video, as a compromise between the delay-prone TCP and the loss-prone UDP. It is argued that the playback quality at the receiver often need not be 100% perfect, providing a certain level is assured. Reliable TCP is used to transmit and guarantee delivery of the most important packets. The delay associated with the proposal is measured, and the potential for use as an alternative to the conventional methods of transporting video by either TCP or UDP alone is demonstrated. Finally, a new objective measurement is investigated for assessing the playback quality of video transported using TCP. A new metric is defined to characterise the quality of playback in terms of its continuity. Using packet traces generated from real TCP connections in a lossy environment, simulating the playback of a video is possible, whilst monitoring buffer behaviour to calculate pause intensity values. Subjective tests are conducted to verify the effectiveness of the metric introduced and show that the results of objective and subjective scores made are closely correlated.

QUIC tramite udp: un’indagine sul protocollo che intende velocizzare il web

In questo documento ho analizzato lo scenario della passata e dell’odierna Internet, dal classico protocollo HTTP, al protocollo sperimentale QUIC, argomento di questa tesi. In primis ho analizzato gli attuali protocolli utilizzati nella rete e ricercato i motivi che hanno portato a crearne di nuovi, successivamente ho effettuato un analisi teorica del protocollo affidandomi ai documenti forniti dall'IETF, poi in un capitolo a sé ho descritto l'handshake crittografato tipico di questo protocollo ed infine nell'ultimo capitolo ho mostrato graficamente e praticamente come lavora il protocollo in una reale implementazione. Dopo aver completato questa tesi, mi sono potuto rendere conto di quanto sia necessario un cambio di rotta verso protocolli di rete più veloci, sicuri ed affidabili. I classici protocolli oramai non sono più sufficienti a soddisfare le migliaia di richieste di connessione e presentano, come si vedrà, delle lacune a cui bisogna porre rimedio. Gran parte della popolazione mondiale ha accesso al web,ed è uno strumento ormai alla portata di tutti e non più privilegio di pochi e ci si augura per il bene della rete Internet che tale protocollo o protocolli simili possano prendere presto piede per una migliore esperienza di navigazione a livello globale. Probabilmente saranno necessari molti anni, ma l’idea che già si pensi ad un futuro non tanto prossimo fa ben sperare su quello che ci aspetta. Nella lettura di questa tesi si vedrà come queste ultime affermazioni possano diventare realtà.

Introduzione al Protocollo TCP

Diapositive utilizate a lezione sugli argomenti: - connessione TCP - formato segmento TCP - macchina a stati finiti TCP - timer TCP

Analisi per via simulativa del protocollo tcp a ritrasmissione asimmetrica anticipata su wifi

Viene implementato un algoritmo di ritrasmissione dei pacchetti nel protocollo TCP. Lo studio viene fatt utilizzando il simulatore OMNET++ e il framework INET