The emergence of real-time applications, like Voice over IP and video conferencing, in IP networks implies a challenge to the underlying infrastructure. Several real-time applications have requirements on timeliness as well as on reliability and are accompanied by signaling applications to set up, tear down and control the media sessions. Since neither of the traditional transport protocols responsible for end-to-end transfer of messages was found suitable for signaling traffic, the Stream Control Transmission Protocol (SCTP) was standardized. The focus for the protocol was initially on telephony signaling applications, but it was later widened to serve as a general purpose transport protocol. One major new feature to enhance robustness in SCTP is multihoming, which enables for more than one path within the same association. In this thesis we evaluate some of the mechanisms affecting transmission performance in case of a switchover between paths in a multihomed SCTP session. The major part of the evaluation concerns a failure situation, where the current path is broken. In case of failure, the endpoint does not get an explicit notification, but has to react upon missing acknowledgements. The challenge is to distinguish path failure from temporary congestion to decide  when to switch to an alternate path. A too fast switchover may be spurious, which could reduce transmission performance, while a too late switchover also results in reduced transmission performance. This implies a tradeoff which involves several protocol as well as network parameters and we elaborate among these to give a coherent view of the parameters and their interaction. Further, we present a recommendation on how to tune the parameters to meet  telephony signaling requirements, still without violating fairness to other traffic. We also consider another angle of switchover performance, the startup on the alternate path. Since the available capacity is usually unknown to the sender, the transmission on a new path is started at a low rate and then increased as acknowledgements of successful transmissions return. In case of switchover in the middle of a media session the startup phase after a switchover could cause problems to the application. In multihomed SCTP the availability of the alternate path makes it feasible for the end-host to estimate the available capacity on the alternate path prior to the switchover. Thus, it would be possible to implement a more efficient startup scheme. In this thesis we combine different switchover scenarios with relevant traffic. For these combinations, we analytically evaluate and quantify the potential performance gain from utilizing an ideal startup mechanism as compared to the traditional startup procedure.

To provide Internet applications with good performance, the transport protocol TCP is designed to optimize the throughput of data transfers. Today, however, more and more applications rely on low latency rather than throughput. Such applications can be referred to as data-limited and are not appropriately supported by TCP. Another emerging problem is associated with the use of novel networking techniques that provide infrastructure-less networking. To improve connectivity and performance in such environments, multi-path routing is often used. This form of routing can cause packets to be reordered, which in turn hurts TCP performance. To address timeliness issues for data-limited traffic, we propose and experimentally evaluate several transport protocol adaptations. For instance, we adapt the loss recovery mechanisms of both TCP and SCTP to perform faster loss detection for data-limited traffic, while preserving the standard behavior for regular traffic. Evaluations show that the proposed mechanisms are able to reduce loss recovery latency with 30-50%. We also suggest modifications to the TCP state caching mechanisms. The caching mechanisms are used to optimize new TCP connections based on the state of old ones, but do not work properly for data-limited flows. Additionally, we design a SCTP mechanism that reduces overhead by bundling several packets into one packet in a more timely fashion than the bundling normally used in SCTP. To address the problem of packet reordering we perform several experimental evaluations, using TCP and state of the art reordering mitigation techniques. Although the studied mitigation techniques are quite good in helping TCP to sustain its performance during pure packet reordering events, they do not help when other impairments like packet loss are present. / <p>Paper V was in manuscript form at the time of the defense.</p>

In this paper, we introduce a novel congestion control algorithm for TCP over multihop IEEE 802.11 wireless networks implementing rate-based scheduling of transmissions within the TCP congestion window. We show how a TCP sender can adapt its transmission rate close to the optimum using an estimate of the current 4-hop propagation delay and the coefficient of variation of recently measured round-trip times. The novel TCP variant is denoted as TCP with Adaptive Pacing (TCP-AP). Opposed to previous proposals for improving TCP over multihop IEEE 802.11 networks, TCP-AP retains the end-to-end semantics of TCP and does neither rely on modifications on the routing or the link layer nor requires cross-layer information from intermediate nodes along the path. A comprehensive simulation study using ns-2 shows that TCP-AP achieves up to 84% more goodput than TCP NewReno, provides excellent fairness in almost all scenarios, and is highly responsive to changing traffic conditions.

In today’s technology industry, cloud refers to servers which can be accessed viainternet, and the software and database applications run on the servers [22]. Whereas cloud computing is a concept of delivering the IT resources via internet which is accessed by the users and companies. In these scenarios OpenStack is the widely used cloud software which controls large pool of IT resources like compute, storage and networking which are managed and provisioned through APIs [11]. In this technology the underlying parts are the transport protocols and web servers used for authenticating and provisioning mechanisms. When stating about its underlying transport protocols, TCP is the default (standard) protocol used behind the functioning of cloud, and HTTP/1.1 version towards actions between Web servers (apache2 and nginx) and browsers [10]. The scope of the thesis is to observe the complexity of replacing TCP with QUIC (Quick UDP Internet Connection) in OpenStack and to observe the performance difference in OpenStack using HTTP/1.1 and HTTP/3. To observe this complexity, Performance Evaluation has been considered the best way in which the performance can be observed from the terminals. The thesis deals with performance of OpenStack with transport protocols from a Web server supporting HTTP/3 feature. We prove that its possible to provide the Keystone API via both HTTP/1.1 and HTTP/3. From our results we see that for simple API access HTTP/3 is faster than the HTTP/1.1 and also when the network is subjected to packet loss. The resultstates that there is a path obtained for OpenStack’s Keystone Service to interact with HTTP/3 and the average request-response time (total time) of HTTP/3 is less than that of HTTP/1.1 for accessing Keystone and Token generation even at defined packet loss rates.

The evolution of computer communications and the Internet has led to the emergence of a large number of communication technologies with widely different capabilities and characteristics. While this multitude of technologies provides a wide array of possibilities it also creates a complex and heterogeneous environment for higher-layer communication protocols. Specific link technologies, as well as overall network heterogeneity, can hamper user-perceived performance or impede end-to-end throughput. In this thesis we examine two transport layer centered approaches to improve performance. The first approach addresses the decrease in user satisfaction that occurs when web waiting times become too long. Increased transport layer flexibility with regards to reliability, together with error-resilient image coding, is used to enable a new trade-off. The user is given the possibility to reduce waiting times, at the expense of image fidelity. An experimental examination of this new functionality is provided, with a focus on image-coding aspects. The results show that reduced waiting times can be achieved, and user studies indicate the usefulness of this new trade-off. The second approach concerns the throughput degradations that can occur as a consequence of link and transport layer interactions. An experimental evaluation of the GSM environment shows that when negative interactions do occur, they are coupled to large variability in link layer round-trip times rather than simply to poor radio conditions. Another type of interaction can occur for link layers which expose higher layers to residual bit errors. Residual bit errors create an ambiguity problem for congestion controlled transport layer protocols which cannot correctly determine the cause for a loss. This ambiguity leads to an unnecessary throughput degradation. To mitigate this degradation, loss differentiation and notification mechanisms are proposed and experimentally evaluated from both performance and fairness perspectives. The results show that considerable performance improvements can be realized. However, there are also fairness implications that need to be taken into account since the same mechanisms that improve performance may also lead to unfairness towards flows that do not employ loss differentiation.

The revolutionary advances in the wireless communication technologies are inspiring the researchers to envision the next generation wireless networking architectures, i.e., Next Generation Wireless Internet (NGWI), InterPlaNetary (IPN) Internet, and Wireless Sensor Networks (WSN). There exist significant technological challenges for the realization of these envisioned next generation network architectures. NGWI will be the convergence of the Internet and heterogeneous wireless architectures, which have diverse characteristics and hence pose different sets of research challenges, to achieve anywhere, anytime seamless service to the mobile users. Similarly, the unique characteristics and challenges posed by deep space communications call for novel networking protocols to realize the IPN Internet objective. Furthermore, in order to realize the potential gains of WSN, it is imperative that communication challenges imposed by resource constraints of sensor nodes must be efficiently addressed with novel solutions tailored to the WSN paradigm. The objective of this research is to develop new advanced transport protocols for reliable data transport and real-time multimedia delivery in the next generation heterogeneous wireless network architectures. More specifically, the analytical rate control (ARC) protocol for real-time multimedia delivery is first proposed for wired/wireless hybrid networks. Next, a new rate control scheme (RCS) is proposed to achieve high throughput performance and fairness for real-time multimedia traffic over the satellite links. The unified adaptive transport layer (ATL) suite and its protocols for both reliable data transport (TCP-ATL) and real-time multimedia delivery (RCP-ATL) are introduced for the NGWI. A new reliable transport protocol for data transport in the IPN Internet (TP-Planet) is then proposed to address the unique challenges of the IPN Internet backbone links. A new integrated tranmission protocol (ITP) is then proposed for reliable data transport over multihop IPN Internet paths. Finally, the event-to-sink reliable transport (ESRT) protocol is proposed to achieve reliable event transport with minimum energy expenditure in WSN.

Emerging wireless networks are characterized by increased heterogeneity in wireless access technologies as well as increased peer-to-peer communication among wireless hosts. The heterogeneity among wireless access interfaces mainly exists because of the fact that different wireless technologies deliver different performance trade-offs. Further, more and more infrastructure-less wireless networks such as ad-hoc networks are emerging to address several application scenarios including military and disaster recovery. These infrastructure-less wireless networks are characterized by the peer-to-peer communication model. In this thesis, we propose transport protocols that tackle the challenges that arise due to the above-mentioned properties of state-of-the-art wireless data networks. The main contributions of this work are as follows: 1. We determine the ideal nature and granularity of transport adaptation for efficient operation in heterogeneous wireless data networks by performing comprehensive experimental analysis. We then design and implement a runtime adaptive transport framework, *TP, which accommodates the capabilities of the ideal transport adaptation solution. 2. We prove that conversational transport protocols are not efficient under peer-to-peer wireless data networks. We then design and implement NCTP which is a non-conversational transport protocol.

Cette thèse traite de l’amélioration du transfert de données, d’une part sur les réseaux sans-fils et d’autre part pour des données continues telles que la vidéo. Pour améliorer les transmissions sur les réseaux sans-fils nous nous sommes intéressés au contrôle de congestion des protocoles de transport mais nous avons également proposé une méthode pratique d’adaptation de la vidéo aux conditions du réseau.Cette thèse contient donc deux volets. La première porte sur la différenciation de pertes entre les pertes de congestion et les pertes sur le réseau sans fil. Il est connu que lors d’une perte, les protocoles de transport actuels réduisent le débit (par deux par exemple). Or, pour les pertes sans fil, cela n’a pas d’intérêt. Pour différencier ces pertes sur l’émetteur des données, nous proposons une méthode originale qui utilise à la fois ECN (Explicit Congestion Notification) et le changement sur le RTT du paquet qui suit la perte. La seconde propose une méthode originale d’adaptation vidéo au niveau de la couche application sur l’émetteur. Avec l’arrivée des vidéos à bitrate élevés (HD, 3D) et l’augmentation constante mais irrégulière des bandes passantes réseau, la qualité vidéo à l’utilisateur reste à la traîne : elle est non-optimale (bitrate beaucoup plus petit ou plus grand que le débit disponible) et non adaptable (aux conditions dynamiques du réseau). Nous proposons une méthode très simple à implémenter, puisqu’elle ne requiert qu’une modification côté émetteur au niveau de la couche application. Elle adapte en permanence le bitrate de la vidéo aux conditions du réseau, autrement dit elle fait un contrôle de congestion sur l’émetteur. La visioconférence est un cas d’application idéal. Cette méthode fonctionne au-dessus de tout protocole de transport avec contrôle de congestion (TCP, DCCP), ce qui lui confère aussi la propriété de TCP-friendliness. / This thesis deals in improving the data transfer on wireless networks and for the continuous data such as video. To improve transmission over wireless networks, we were interested in congestion control transport protocols and we also proposed a practical method for adjusting the video rate to network conditions.This thesis composes of two parts. The first part concerns the loss differentiation between congestion losses and losses on the wireless network. It is known that when there is a loss, transport protocols reduce the current sending rate (e.g. by two). However, for wireless losses, it has no interest in reducing the rate. To differentiate these losses on the data senders side, we propose a novel method that uses both the ECN (Explicit Congestion Notification) and the change of RTT of the packet following the loss. The second part proposes a novel method for video adaptation at the application layer of the sender. With the advent of high bitrate video (e.g. HD, 3D) and steadily increasing but irregular network bandwidth, video quality to the user lags. It is non-optimal (bitrate is highly smaller or larger than the available bandwidth) and not adaptable (to the dynamic conditions of the network). We propose a simple method to implement, since it requires a change only at the application layer of the sender. It adapts the bitrate of the video to the network conditions, i.e. it is a congestion control on the transmitter. Videoconferencing is an ideal case for the application of adaptation. This method works over any transport protocol with congestion control (e.g. TCP, DCCP), which also confers the property of TCP-friendliness.

Wireless communication has seen a tremendous growth in the last decades. Continuing on this trend, wireless multi-hop networks  are nowadays used or planned for use in a multitude of contexts, spanning from Internet access at home to emergency situations. The Transmission Control Protocol (TCP) provides reliable and ordered delivery of a data and is used by major Internet applications such as web browsers, email clients and file transfer programs. TCP traffic is also the dominating traffic type on the Internet. However, TCP performs less than optimal in wireless multi-hop networks due to packet reordering, low link capacity, packet loss and variable delay. In this thesis, we develop novel proposals for enhancing the network and transport layer to improve TCP performance in wireless multi-hop networks. As initial studies, we experimentally evaluate the performance of different TCP variants, with and without mobile nodes. We further evaluate the impact of multi-path routing on TCP performance and propose packet aggregation combined with aggregation aware multi-path forwarding as a means to better utilize the available bandwidth. The last contribution is a novel extension to multi-path TCP to  enable single-homed hosts to fully utilize the network capacity. / <p>Opponent changed. Prof. C. Lindeman from the University of Leipzig was substituted by Prof. Zhang.</p>

Submitted by Milena Rubi (milenarubi@ufscar.br) on 2017-06-01T14:58:47Z No. of bitstreams: 1 SILVA_Alan_2016.pdf: 14722594 bytes, checksum: 0c8346924c434318f6c349f7ed8112d9 (MD5) / Approved for entry into archive by Milena Rubi (milenarubi@ufscar.br) on 2017-06-01T14:58:54Z (GMT) No. of bitstreams: 1 SILVA_Alan_2016.pdf: 14722594 bytes, checksum: 0c8346924c434318f6c349f7ed8112d9 (MD5) / Approved for entry into archive by Milena Rubi (milenarubi@ufscar.br) on 2017-06-01T14:58:59Z (GMT) No. of bitstreams: 1 SILVA_Alan_2016.pdf: 14722594 bytes, checksum: 0c8346924c434318f6c349f7ed8112d9 (MD5) / Made available in DSpace on 2017-06-01T14:59:06Z (GMT). No. of bitstreams: 1 SILVA_Alan_2016.pdf: 14722594 bytes, checksum: 0c8346924c434318f6c349f7ed8112d9 (MD5) Previous issue date: 2016-12-06 / Não recebi financiamento / The Multipath TCP (MPTCP) protocol allows applications to better explore the network resources available to multi-connected devices such as mobile phones or multi-homed systems. Here, some advantages are envisioned: bandwidth aggregation, the ability to maintain the connection, if one of the network path fails and the use of multiple paths. To extend these capabilities to the application, RFC 6897 defines an API to better control each of MPTCP’s subflows, so that these can be added or removed as needed. This work presents an initial API implementation as defined in RFC 6897. We implemented some functions described in the document, such as protocol on/o, check existent subflows and add new subflows. To test the API and validate our implementation we built an HTTP application that detects elephant flows and uses the API for open new subflows using the original TCP connection. Some tests were performed in a network using a cubic topology and showed that the API utilization decreased the Flow Completion time of TCP connections. / O protocolo Multipath TCP (MPTCP) permite que as aplicações possam explorar melhor os recursos de rede disponíveis para dispositivos multiconectados como os telefones móveis ou sistemas multi-homed. Aqui, algumas vantagens são previstas: agregação de banda, a habilidade de manter a conexão estabelecida se houver falha em um dos caminhos de rede e a utilização de múltiplos caminhos. Para estender essas capacidades para a aplicação, a RFC 6897 define uma API que permite um melhor controle de cada subfluxo MPTCP, de modo que esses possam ser adicionados ou removidos conforme necessário. Este trabalho apresenta uma implementação inicial da API descrita na RFC 6897 para o protocolo MPTCP. Sendo assim, implementamos algumas das funções de manipulação do protocolo MPTCP descritas no documento, quais sejam: ligar e desligar o protocolo, verificar subfluxos existentes e adicionar novos subfluxos. Para testar a API e validar a nossa implementação, nós desenvolvemos uma aplicação HTTP que detecta fluxos elefantes e utiliza a API para abrir novos subfluxos a partir da conexão TCP original. Testes de desempenho foram realizados em uma topologia cúbica e mostraram que a utilização da API pela aplicação diminuiu o Flow Completion Time das conexões TCP.