Transmission Control Protocol (TCP) is the dominant transport protocol today and likely to be adopted in future high‐speed and optical networks. A number of literature works have been done to modify or tune the Additive Increase Multiplicative Decrease (AIMD) principle in TCP to enhance the network performance. In this work, to efficiently take advantage of the available high bandwidth from the high‐speed and optical infrastructures, we propose a Stratified TCP (STCP) employing parallel virtual transmission layers in high‐speed networks. In this technique, the AIMD principle of TCP is modified to make more aggressive and efficient probing of the available link bandwidth, which in turn increases the performance. Simulation results show that STCP offers a considerable improvement in performance when compared with other TCP variants such as the conventional TCP protocol and Layered TCP (LTCP).

The amount of packets on-the-fly in a wireless ad-hoc network increases when the size of congestion window and the number of hop count increase. It is possible that packets may have to retransmit because large amount of on-the-fly packets may increase the media contention. Besides, packet delay can grow rapidly when the wireless network becomes congested or the channel interference remains unresolved. This thesis presents a cross-layer TCP congestion window control mechanism for multihop ad-hoc networks to dynamically adjust the size of congestion window according to the MAC-layer contention statistics measured at each hop along the routing path. With the proposed scheme, the congestion window of each traffic flow can be dynamically set to an appropriate size to reduce packet delay and increase flow throughput. For the purpose of evaluation, we perform simulations on NS-2. The simulation results have shown the advantage of our proposed scheme over the two previous works, NewReno and CWL(Congestion Window Limit), especially when the ad-hoc network is loaded with background traffic.

IETF proposes the Differentiated Service(DiffServ) architecture for next-generation QoS networks. The main features of a DiffServ network are high scalability and compatibility. However, lacking of supporting QoS for individual flows becomes a problem. To remedy this problem, we propose an effective TCP transmission (ETCP) scheme based on the characteristics of TCP. The goal of this thesis is to make TCP transmission more effective. Because of the network complexity, we need to modify both the Ingress and Egress nodes. At Ingress node, we measure TCP congestion window(cwnd) and Round Trip Time(RTT), while at Egress node we monitor the throughput and embed a feedback ratio into TCP header of the acknowledge packets. When the throughput is below or over the target throughput, we adjust DSCP dynamically. The adjustment depends on both measured TCP congestion window and RTT. Each DSCP value is mapped with a delay time through Delayed-FIFO. Changing the delay time in Ingress node, we are able to control TCP transmission associated with RTT more effectively. For the purpose of the demonstration, we implement our scheme on Linux platform. We discuss the performance difference between the theoretical and practical results through experiments. Besides, from the implementation, we have shown that our proposed architecture is workable and compatible with today¡¦s network environment.

Transmission Control Protocol (TCP) designed to deliver seamless and reliable end-to-end data transfer across unreliable networks works impeccably well in wired environment. In fact, TCP carries the around 90% of Internet traffic, so performance of Internet is largely based on the performance of TCP. However, end-to-end throughput in TCP degrades notably when operated in wireless networks. In wireless networks, due to high bit error rate and changing level of congestion, retransmission timeouts for packets lost in transmission is unavoidable. TCP misinterprets these random packet losses, due to the unpredictable nature of wireless environment, and the subsequent packet reordering as congestion and invokes congestion control by triggering fast retransmission and fast recovery, leading to underutilization of the network resources and affecting TCP performance critically. This thesis reviews existing approaches, details two proposed systems for better handling in networks with random loss and delay. Evaluation of the proposed systems is conducted using OPNET simulator by comparing against standard TCP variants and with varying number of hops.

Transmission Control Protocol (TCP) designed to deliver seamless and reliable end-to-end data transfer across unreliable networks works impeccably well in wired environment. In fact, TCP carries the around 90% of Internet traffic, so performance of Internet is largely based on the performance of TCP. However, end-to-end throughput in TCP degrades notably when operated in wireless networks. In wireless networks, due to high bit error rate and changing level of congestion, retransmission timeouts for packets lost in transmission is unavoidable. TCP misinterprets these random packet losses, due to the unpredictable nature of wireless environment, and the subsequent packet reordering as congestion and invokes congestion control by triggering fast retransmission and fast recovery, leading to underutilization of the network resources and affecting TCP performance critically. This thesis reviews existing approaches, details two proposed systems for better handling in networks with random loss and delay. Evaluation of the proposed systems is conducted using OPNET simulator by comparing against standard TCP variants and with varying number of hops.

Transmission Control Protocol (TCP) designed to deliver seamless and reliable end-to-end data transfer across unreliable networks works impeccably well in wired environment. In fact, TCP carries the around 90% of Internet traffic, so performance of Internet is largely based on the performance of TCP. However, end-to-end throughput in TCP degrades notably when operated in wireless networks. In wireless networks, due to high bit error rate and changing level of congestion, retransmission timeouts for packets lost in transmission is unavoidable. TCP misinterprets these random packet losses, due to the unpredictable nature of wireless environment, and the subsequent packet reordering as congestion and invokes congestion control by triggering fast retransmission and fast recovery, leading to underutilization of the network resources and affecting TCP performance critically. This thesis reviews existing approaches, details two proposed systems for better handling in networks with random loss and delay. Evaluation of the proposed systems is conducted using OPNET simulator by comparing against standard TCP variants and with varying number of hops.

Besides the two classical techniques used to evaluate the performance of a protocol, computer simulation and experimental measurements, mathematical modeling has been used to study the performance of the TCP protocol. This technique gives an elegant way to gain insights when studying the behavior of a protocol, while providing useful information about its performance. This thesis presents an analytical model for the SF-SACK protocol, a TCP SACK based protocol conceived to be appropriate for data and streaming applications. SF-Sack modifies the multiplicative part of the Additive Increase Multiplicative Decrease of TCP to provide good performance for data and streaming applications, while avoiding the TCP-friendliness problem of the Internet. The modeling of the SF-SACK protocol raises new challenges compared to the classical TCP modeling in two ways: first, the model needs to be adapted to a more complex dynamism of the congestion window, and second, the model needs to incorporate the scheduler that SF-SACK makes use of in order to maintain a periodically updated value of the congestion window. Presented here is a model that is progressively built in order to consider these challenges. The first step is to consider only losses detected by triple-duplicate acknowledgments, with the restriction that one such loss happens each scheduler interval. The second step is to consider losses detected via triple-duplicate acknowledgments, while eliminating the above restriction. Finally, the third step is to include losses detected via time-outs. The result is an analytical characterization of the steady-state send rate and throughput of a SF-SACK flow as a function of the loss probability, the round-trip time (RTT), the time-out interval, and the scheduler interval. The send rate and the throughput of SF-SACK were compared against available results for TCP Reno. The obtained graphs showed that SF-SACK presents a better performance than TCP. The analytical model of the SF-SACK follows the trends of the results that are presently available, using both the ns-2 simulator and experimental measurements.