The explosive growth of the Internet during the last few years, has been propelled by the TCP/IP protocol suite and the best effort packet forwarding service. However, quality of service (QoS) is far from being a reality especially for multimedia services like video streaming and video conferencing. In the case of wireless and mobile networks, the problem becomes even worse due to the physics of the medium, resulting into further deterioration of the system performance. Goal of this dissertation is the systematic development of comprehensive models that jointly characterize the performance of transport protocols and media delivery in heterogeneous wireless networks. At the core of our novel methodology, is the use of analytical models for driving the design of media transport algorithms, so that the delivery of conversational and non-interactive multimedia data is enhanced in terms of throughput, delay, and jitter. More speciffically, we develop analytical models that characterize the throughput and goodput of the transmission control protocol (TCP) and the transmission friendly rate control (TFRC) protocol, when CBR and VBR multimedia workloads are considered. Subsequently, we enhance the transport protocol models with new parameters that capture the playback buffer performance and the expected video distortion at the receiver. In this way a complete end-to-end model for media streaming is obtained. This model is used as a basis for a new algorithm for rate-distortion optimized mode selection in video streaming appli- cations. As a next step, we extend the developed models for the aforementioned protocols, so that heterogeneous wireless networks can be accommodated. Subsequently, new algorithms are proposed in order to enhance the developed media streaming algorithms when heterogeneous wireless networks are also included. Finally, the aforementioned models and algorithms are extended for the case of concurrent multipath media transport over several hybrid wired/wireless links.

Thesis (Ph. D.)--OGI School of Science & Engineering at OHSU, 2002. / Includes bibliographical references.

Thesis (M. Phil.)--Hong Kong University of Science and Technology, 2003. / Includes bibliographical references (leaves 68-71). Also available in electronic version. Access restricted to campus users.

Thesis (M.S.)--Worcester Polytechnic Institute. / Keywords: TCP; RED; satellite; PEP; STP; performance enhancing proxy; segment caching; IP-ABR; Internet; bandwidth reservation; IP-VBR; congestion avoidance; bandwidth sharing. Includes bibliographical references (p. 98-99).

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Despite the persistent change and growth that characterizes the Internet, the Transmission Control Protocol (TCP) still dominates at the transport layer, carrying more than 90\% of the global traffic. Despite its astonishing success, it has been observed that TCP can cause poor end-to-end performance, especially for large transfers and in network paths with high bandwidth-delay product. In this thesis, we focus on mechanisms that can address key problems in TCP performance, without any modification in the protocol itself. This evolutionary approach is important in practice, as the deployment of clean-slate transport protocols in the Internet has been proved to be extremely difficult. Specifically, we identify a number of TCP-related problems that can cause poor end-to-end performance. These problems include poorly dimensioned socket buffer sizes at the end-hosts, suboptimal buffer sizing at routers and switches, and congestion unresponsive TCP traffic aggregates. We propose solutions that can address these issues, without any modification to TCP. <br> <br> In network paths with significant available bandwidth, increasing the TCP window till observing loss can result in much lower throughput than the path's available bandwidth. We show that changes in TCP are {em not required} to utilize all the available bandwidth, and propose the application-layer SOcket Buffer Auto-Sizing (SOBAS) mechanism to achieve this goal. SOBAS relies on run-time estimation of the round trip time (RTT) and receive rate, and limits its socket buffer size when the receive rate approaches the path's available bandwidth. In a congested network, SOBAS does not limit its socket buffer size. Our experiment results show that SOBAS improves TCP throughput in uncongested network without hurting TCP performance in congested networks. <br> <br> Improper router buffer sizing can also result in poor TCP throughput. Previous research in router buffer sizing focused on network performance metrics such as link utilization or loss rate. Instead, we focus on the impact of buffer sizing on end-to-end TCP performance. We find that the router buffer size that optimizes TCP throughput is largely determined by the link's output to input capacity ratio. If that ratio is larger than one, the loss rate drops exponentially with the buffer size and the optimal buffer size is close to zero. Otherwise, if the output to input capacity ratio is lower than one, the loss rate follows a power-law reduction with the buffer size and significant buffering is needed. The amount of buffering required in this case depends on whether most flows end in the slow-start phase or in the congestion avoidance phase. <br> <br> TCP throughput also depends on whether the cross-traffic reduces its send rate upon congestion. We define this cross-traffic property as {em congestion responsiveness}. Since the majority of Internet traffic uses TCP, which reduces its send rate upon congestion, an aggregate of many TCP flows is believed to be congestion responsive. Here, we show that the congestion responsiveness of aggregate traffic also depends on the flow arrival process. If the flow arrival process follows an open-loop model, then even if the traffic consists exclusively of TCP transfers, the aggregate traffic can still be unresponsive to congestion. TCP flows that arrive in the network in a closed-loop manner are always congestion responsive, on the other hand. We also propose a scheme to estimate the fraction of traffic that follows the closed-loop model in a given link, and give practical guidelines to increase that fraction with simple application-layer modifications.

Computer networks used in Telecommunication, particularly the Internet, have been used to carry computer data only, but now they carry voice and/or video also. Because each type of this traffic has specific flow characteristics, each type has to be handled with a certain level of guaranteed quality. So based on that, IETF introduced a Quality of Service tool, called Differentiated Service. It offers different levels of service to different classes of traffic. Even then, the problem of congestion arises due to sharing of a finite bandwith. In the case of real time multi media traffic, congestion due to inadequate bandwith contributes heavily to the quality, whereas in non-real time traffic the effect of congestion is to make data transfer take a longer time. In contrast, real time data become become obsolete if they do not arrive on time. Therefore what is needed is some way of ensuring that during periods of congestion, real time traffic is not affected at all, or is at least given a higher priority than non-real time. / The motivation for the research that has been carried out was therefore to develop a rule based traffic management scheme for DiffServ networks with a view to introducing QoS (Quality of Service). This required definition of rules for congestion management/control based on the type and nature of IP traffic encountered, and then constructing and storing these rules to enable future access for application and enforcement. We first developed the required rule base and then developed the software based mobile agents using the Java (RMI) application package, for accessing these rules for application and enforcement. Consequently, these mobile agents act as smart traffic managers at nodes/routers in the computer based communication network and manage congestion. The rule base as well as the mobile agent software developed in Java (RMI), were validated using computer simulation. The contents of the research carried out have been presented in the thesis. / Thesis (PhD)--University of South Australia, 2006.

Thesis (PhDTelecommunications)--University of South Australia, 2002.

In recent times, the imminent lack of public IPv4 addresses has attracted the attention of both the research community and industry. The cellular industry has decided to combat this problem by using IPv6 for all new terminals. However, the success of 3G network deployment will depend on the services offered to end users. Currently, almost all services reside in the IPv4 address space, making them inaccessible to users in IPv6 networks. Thus, an intermediate translation mechanism is required. Previous studies on network address translation methods have shown that Realm Base Kluge Address Heuristic-IP, REBEKAH-IP supports all types of services that can be offered to IPv6 hosts from the public IPv4 based Internet, and provides excellent scalability. However, the method suffers from an ambiguity problem which may lead to call blocking. This thesis presents an improvement to REBEKAH-IP scheme in which the side effect is removed, creating a robust and fully scalable system. The improvement can be divided into two major tasks including a full investigation on the scalability of addressing and improvements to the REBEKAH-IP scheme that allow it to support important features such as ICMP and IP mobility. To address the first task a method called REBEKAH-IP with Port Extension (RPX) is introduced. RPX is extended from the original REBEKAH-IP scheme to incorporate centralised management of both IP address and port numbers. This method overcomes the ambiguity problem, and improves scalability. We propose a priority queue algorithm to further increase scalability. Finally, we present extensive simulation results on the practical scalability of RPX with different traffic compositions, to provide a guideline of the expected scalability in large-scale networks. The second task concerns enabling IP based communication. Firstly, we propose an ICMP translation mechanism which allows the RPX server to support important end-toend control functions. Secondly, we extend the RPX scheme with a mobility support scheme based on Mobile IP. In addition, we have augmented Mobile IP with a new tunneling mechanism called IP-in-FQDN tunneling. The mechanism allows for unique mapping despite the sharing of IP addresses while maintaining the scalability of RPX. We examine the viability of our design through our experimental implementation.

Thesis (M.E.E.)--University of Delaware, 2007. / Principal faculty advisor: Gonzalo Arce, Dept. of Electrical and Computer Engineering. Includes bibliographical references.