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.

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.

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.