Conventional TCP is window based, which exploits the sliding window mechanism to conduct the flow control. It increases the sending window additively and decreases the sending window multiplicatively in response to successful transmission and, packet loss/timeout events respectively. While the mechanism works quite well in normal networks, TCP can hardly reach the ideal bandwidth utilization in long fat networks (LFNs) due to long delay and bursts of packet losses. Besides, as wireless and mobile computing has become popular today, packet loss in such networks may occur due to noise, interference and handoff across different domains. TCP could not react to different situations effectively as it sees all packet losses as an indication of network congestion. In this thesis, we proposed a new transmission control mechanism called Active Rate Anchoring TCP (ARCH-TCP). In ARCH-TCP, we explicitly integrate bandwidth measurement into TCP to solve the aforementioned problem. Specifically, we exploit packet-pair measurement to quickly estimate bandwidth share and then RTT variation is observed to compensate measurement error. We built the model in J-Sim network simulator to evaluate the effectiveness of our proposal. We found that ARCH-TCP can react to network conditions quickly and precisely in both wired and wireless networks and both in the normal networks and LFNs.

Many applications in modern science need to transmit extremely massive amount of data over wide area networks. These data usually do not need stringent real-time requirements but require large bandwidth to finish transmission with unreasonable time. High-energy physics experiments and climate modeling and analysis are typical examples of such applications. As TCP is known to perform inefficiently over networks of large delay-bandwidth product, efficient transmission of this kind of massive, non-real-time data has been heavily studied in the past. The previous results work well in dedicated networks but will compete for fair share of bandwidth with normal TCP connections if they operate in the public networks. The objective of this thesis is to design a new transmission protocol for the above applications that can operate in the public networks without affecting normal TCP connections. The new protocol is called Rate Control Transmission Protocol (RCTP). The idea is to apply the packet-pair measurement technique to measure the bandwidth share in the network for the transmission. The sending rate is based on that measurement and is precisely compensated by the RTT variance measurement. Due to the RTT compensation, RCTP can efficiently utilize the unused bandwidth in the network while not affecting the normal TCP transmissions, making it perfect for transmitting massive, non-real-time data in the public networks.

Due to its ubiquitous availability, Short Message Service (SMS), first introduced in the 1980s, became not only the most popular way of communication, but also stimulated the development of SMS-based value added services. This application-to-person traffic is delivered to end users through SMS aggregators who provide the link between service providers and mobile carriers. In order to perform optimal traffic routing, the aggregators need to estimate the quality of each potential international route to the specified destination. The evaluation criteria include end-to-end delivery time, as well as correct verification of delivered data. This thesis suggests a method of quality of service (QoS) assessment for international SMS service which combines two types of tests, end-to-end delay measurements and various verification tests. A prototype of the testing system for international SMS service was developed to generate SMS traffic, collect and analyze results, and evaluate the experienced QoS of the SMS route used in accordance with the proposed approach. As a part of end-to- end delay measurement tests, SMS traffic was sent to Singtel network in Singapore along two routes. The verification tests were executed via different routes to two mobile networks: Singtel and Tele2 (Sweden). The results of the performed measurements determined the route with the highest QoS, i.e. the one with bigger bottleneck bandwidth and lower data loss rate. The prototype of the SMS testing system can be used by SMS aggregators to verify delivery of a SMS message, check the integrity of the message, figure out interconnection type of the route supplier with the destination carrier and to identify the presence of load balancers in the path. The prototype also makes it possible to compare end-to-end delay times of several routes and compute bottleneck values for each of the tested routes.

As the Internet continues to evolve, without providing any performance guarantees or explicit feedback to applications, the only way to infer the state of the network and to dynamically react to congestion is through end-to-end measurements. The emph{available bandwidth} (avail-bw) is an important metric that characterizes the dynamic state of a network path. Its measurement has been the focus of significant research during the last 15 years. However, its estimation remained elusive for several reasons. The main contribution of this thesis is the development of the first estimation methodology for the avail-bw in a network path using end-to-end measurements. In more detail, our first contribution is an end-to-end methodology, called SLoPS, to determine whether the avail-bw is larger than a given rate based on the sequence of one-way delays experienced by a periodic packet stream. The second contribution is the design of two algorithms, based on SLoPS, to estimate the mean and the variation range, respectively, of the avail-bw process. These algorithms have been implemented in two measurement tools, referred to as PathLoad and PathVar. We have validated the accuracy of the tools using analysis, simulation, and extensive experimentation. Pathload has been downloaded by more than 6000 users since 2003. We have also used PathVar to study the variability of the avail-bw process as a function of various important factors, including traffic load and degree of multiplexing. Finally, we present an application of avail-bw estimation in video streaming. Specifically, we show that avail-bw measurements can be used in the dynamic selection of the best possible overlay path. The proposed scheme results in better perceived video quality than path selection algorithms that rely on jitter or loss-rate measurements.