Abstract Due to advances of technologies and growth of Internet usage, demand for larger and larger network capacity remains the major challenge for network operators. To meet the increasing demand, optical network has become the key technology in the current and next-generation Internet. In terms of network architecture, optical packet switching (OPS) is a promising up-star in achieving high efficiency just as the electronic counterpart. However, it is currently far reached because of the difficulty in making optical random access memory and ultra-high cost in making fast optical switches that can handle more than 10^9 packets per second. Optical burst switching (OBS), on the other hand, is a more achievable, economical alternative. In OBS networks, packets are aggregated into much larger sized bursts before entering the core network. It thus does not require fast optical switches. And by incorporating one-way delayed reservation scheme, OBS avoids using optical RAM. There have been many research activities toward OBS. However, for Internet with 90% of TCP traffic, the effect of packet aggregation introduced by OBS on TCP performance is still not well understood. Detti and Listanti derived a model for it and the model was verified in simulation [2]. Yet, we found many of the assumptions in their study are not realistic. The obtained result is therefore questionable. In this thesis, we relax their assumptions and design two new models accordingly in order to get deeper understanding on the effects of packet aggregation on TCP performance. From our simulation results, we conclude three affecting factors: burst assembly, assembly delay and assembler buffer size. Burst assembly shows positive effect, while the other two demonstrate negative effects, on TCP throughput.

In this thesis, a modified routing protocol, MAODV (Modified Ad Hoc On-Demand Distance Vector), is proposed for wireless Ad Hoc networks. MAODV uses ACK paths for rerouting data traffic whenever there is a route failure. We create new flags in the routing information of AODV to timely notify TCP layer the network status. After a route failure, the source node monitors ACK packets from the destination node. Every ACK packet when it is received from the destination node, its carried information is used to update the routing tables. Thus, without any interruption, TCP can take the reverse route of the ACK forwarding paths to continue to transmit the rest of data packets. In our proposed scheme, without decreasing the size of TCP congestion window (CWND), a lost packet can be immediately retransmitted as soon as the first duplicate ACK is received. We use NS-2 to simulate the proposed MAODV. From our simulation results, we have shown that when the position of route failures is near the source node, it may take more time to find new data paths. Hence, MAODV has the advantages by taking the reverse ACK paths to transmit data packets before new routing paths are found. Finally, we have demonstrated that MAODV performs better than TCP-BuS (BUffering capability and Sequence information) and the original TCP, when there are many hops between the source node and the destination node. On the other hand, when the position of route failures is near the destination node, the advantages of MAODV become not so significant because there exists some local repair mechanisms to be applied.

Transmission Control Protocol (TCP) is the dominant transport protocol in the Internet and supports many of the most popular Internet applications, such as the World Wide Web (WWW), file transfer and e-mail. TCP congestion control algorithms dynamically learn the network bandwidth and delay characteristics of a network and adapt its performance to changes in traffic so as to avoid network collapse. TCP is designed to perform well in traditional wireline networks with the assumptions that packet losses are mainly due to network congestion and random bit error rate (BER) is negligible. However, networks with wireless links suffer from significant packet losses due to random bit errors and handoffs. Hence TCP performs poorly in networks with wireless links because it treats any packet loss in the network to be a result of network congestion and slows down its transmission rate, or even cause the TCP sender to experience unnecessary timeouts, further reducing its performance. The development of advance wireless networks, such as WiFi, UMTS and WiMAX, make it necessary to find ways to improve TCP's efficiency and resource utilization, as well as improve the user's experience and reduce latency times. In order to find effective solutions to this effect, packet losses across wireless links should be distinguished from congestion related packet losses. In this thesis, we concentrate on two main strategies for enabling the TCP congestion control mechanism to determine the cause for a packet loss. One is a proxy-based mechanism that monitors the radio network interface and sends radio network feedback (RNF) to the TCP sender with the status of the wireless link. The other one is an end-to-end mechanism, in which the packet error pattern is used as the system metric to fine-tune the congestion control mechanism. It also presents an analytical model of TCP with enhanced recovery mechanism for wireless environments. In a proxy-based mechanism, TCP sender is explicitly informed of any effects caused by wireless links. However, the implementation technique is network dependent. We have proposed and developed three proxy-based schemes; the radio network feedback (RNF) scheme over an 802.11 WLAN network, the radio network controller (RNC) feedback over a UMTS network and a wireless enhancement proxy (WENP) over both the 802.11 WLAN and UMTS networks. The RNF scheme is introduced at the 802.11 WLAN base station that monitors the TCP packet flows over the wireless links, detects wireless packet losses and provides feedback to the TCP sender using one of the TCP header reserved control bits, called RNF flag. TCP Reno is modified to utilize the radio network feedback to distinguish the losses due to wireless effects form the congestion and fine-tuned to perform wireless enhanced fast retransmit and fast recovery mechanisms. The RNF scheme is implemented using the OPNET tool, and the simulation results show that the TCP performance is significantly improved. The RNC feedback mechanism, similar to the RNF scheme, is developed and implemented in a UMTS network. The GPRS Tunneling Protocol (GTP) layer of the UMTS Radio Network Control (RNC) protocol stack was modified to detect and notify the TCP sender of the wireless packet losses, which is the main difference between the RNF and RNC mechanisms. The simulation results shows that the RNC feedback mechanism significantly improves the TCP performance compared to that of standard TCP over UMTS. The wireless enhancement proxy (WENP) is developed to minimize spurious TCP timeouts over wireless networks and implemented in both 802.11 WLAN and UMTS networks. WENP extends the proposed RNF and RNC feedback mechanisms to detect both wireless packet losses and large delays across the wireless link, and to notify the TCP sender of these events with the aid of two reserved bits in the TCP header. TCP Reno is further modified to utilize the WENP feedback to distinguish both wireless packet losses from congestion losses and spurious timeouts from normal timeouts. It is also fine-tuned to perform both the wireless enhanced fast retransmit and fast recovery mechanism and the timeout mechanism. The simulation results demonstrate that the proposed scheme markedly improves the TCP performance compared to that of standard WLAN and UMTS implementations. An end-to-end early packet loss recovery (EPLR) mechanism that modifies the TCP Reno fast retransmit algorithm to detect packet losses early and to speed up the packet recovery process to reduce the number of TCP timeouts over networks with heavy packet losses, such as wireless networks is also presented. TCP Reno with EPLR scheme is implemented in a UMTS network and its performance is compared with that of TCP Reno and New Reno. Simulation results shows that Reno with EPLR improves the TCP performance and application response time significantly compared to that of both Reno and New Reno by reducing the TCP timeouts, which is the main cause of degradation of the TCP performance in a wireless environment. Finally, we develop an analytical TCP throughput model with enhanced TCP Reno fast retransmit algorithm to avoid timeouts. The model captures the TCP fast retransmit mechanism and expresses the steady state congestion window and throughput as a function of network utilization factor, round trip time (RTT) and loss rate. Another new feature added to the model is dynamic adjustment of the congestion window size depending on the packet drop rates. This speeds up the packet recovery process and reduces the number of TCP timeouts over networks with heavy packet losses. The proposed model is implemented over a UMTS network and its performance is compared with that of TCP Reno. Simulation results show that the proposed model reduces the TCP timeouts and improves the TCP performance compared to that of TCP Reno. It is also found that the model provides a very good match to the steady-state congestion window behavior.

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Expanding the global Internet to include mobile devices is an exciting area of current research. Because of the vast size of the Internet, and because the protocols in it are already widely deployed, mobile devices must inter-operate with those protocols. Although most of the incompatiblities with mobiles have been solved, the protocols that deliver data reliably, and that account for the majority of Internet traffic, perform very poorly. A change in location causes a disruption in traffic, and disruption is dealt with by algorithms tailored only for stationary hosts. The Transmission Control Protocol (TCP) is the predominant transport-layer protocol in the Internet. In this thesis, we look at the performance of TCP in mobile environments. We provide a complete explanation for poor performance; we conduct a large number of experiments, simulations, and analyses that prove and quantify poor performance;and we propose simple and scalable solutions that address the limitations.

Expanding the global Internet to include mobile devices is an exciting area of current research. Because of the vast size of the Internet, and because the protocols in it are already widely deployed, mobile devices must inter-operate with those protocols. Although most of the incompatiblities with mobiles have been solved, the protocols that deliver data reliably, and that account for the majority of Internet traffic, perform very poorly. A change in location causes a disruption in traffic, and disruption is dealt with by algorithms tailored only for stationary hosts. The Transmission Control Protocol (TCP) is the predominant transport-layer protocol in the Internet. In this thesis, we look at the performance of TCP in mobile environments. We provide a complete explanation for poor performance; we conduct a large number of experiments, simulations, and analyses that prove and quantify poor performance;and we propose simple and scalable solutions that address the limitations.

Ad hoc network is a new tendency of data transmission in the future. Because of the convenience and necessity of mobile phone and/or portable computer coupled with wireless data services, TCP/IP has become an important topic for the study in wireless networks. However, there were a few difficulties in data transmission that must be overcome due to ad hoc environments and the characteristics of the IEEE 802.11 protocols. In these protocols, MAC layer is our primary research topic. Based on the RTS/CTS signal of dynamic retransmission [4], this study presented an improvement to solve the problems: (1) media resources was easily robbed due to the RTS/CTS signal competition, (2) data frame would be dropped by the IEEE 802.11 protocol due to too many times collisions. In addition, this study modifies the CWND in TCP layer in accordance with congestion conditions. Sender can transmit data packets to the network, in which resources can be completely utilized without any waste or loss. We also use the related parameters from the IP and TCP header to calculate flow numbers. The calculated parameters are recorded in IP and TCP header, and then instantly forwarded to the receiver via routers. The receiver can forward these parameters back to the sender by using back transmission method. The simulation result shows that the proposed methods can effectively improve TCP performance, such as packet loss rate, and fastly increase the CWND, the buffer utilization, and so forth. Thus, the network can perform more effectively while using the MAC-layer RTS/CTS signal.

Internet applications make use of the services provided by a transport protocol, such as TCP (a reliable, in-order stream protocol). We use this term Transport Service to mean the end-to- end service provided to application by the transport layer.   That service can only be provided correctly if information about the intended usage is supplied from the application. The application may determine this information at the design time, compile time, or run time, and it may include guidance on whether a feature is required, a preference by the application, or something in between. Multipath TCP (MPTCP) adds the capability of using multiple paths to a regular TCP session. Even though it is designed to be totally backward compatible to applications. The data transport differs compared to regular TCP, and there are several additional degrees of freedom that the particular application may want to exploit.   Multipath TCP is particularly useful in the context of wireless networks using both Wi-Fi and a mobile network is a typical use case. In addition to the gains in throughput from inverse multiplexing, links may be added or dropped as the user moves in or out of coverage without disrupting the end-to-end TCP connection. The problem of link handover is thus solved by abstraction in the transport layer, without any special mechanisms at the network or link level.   Handover functionality can then be implemented at the endpoints without requiring special functionality in the sub-networks according to the Internet's end-to-end principle. Multipath TCP can balance a single TCP connection across multiple interfaces and reach very high throughput.

Internet applications make use of the services provided by a transport protocol, such as TCP (a reliable, in-order stream protocol). We use the term Transport Service to mean the endtoend service provided to application by the transport layer. That service can only be provided correctly if information about the intended usage is supplied from the application. The application may determine this information at the design time, compile time, or run time, and it may include guidance on whether a feature is required, a preference by the application, or something in between. Multipath TCP (MPTCP) adds the capability of using multiple paths to a regular TCP session. Even though it is designed to be totally backward compatible to applications. The data transport differs compared to regular TCP, and there are several additional degrees of freedom that the particular application may want to exploit. Multipath TCP is particularly useful in the context of wireless networks using both Wi-Fi and a mobile network is a typical use case. In addition to the gains in throughput from inverse multiplexing, links may be added or dropped as the user moves in or out of coverage without disrupting the end-to-end TCP connection. The problem of link handover is thus solved by abstraction in the transport layer, without any special mechanisms at the network or link level. Handover functionality can then be implemented at the endpoints without requiring special functionality in the sub-networks according to the Internet's end-to-end principle. Multipath TCP can balance a single TCP connection across multiple interfaces and reach very high throughput.

A key objective for 3G wireless networks, such as the Universal Mobile Telecommunication System (UMTS), is the explicit support for data communications for mobile users. Today, the dominant transport protocol in the Internet is the Transport Control Protocol (TCP). Since TCP was not tailored for wireless networks, there are some performance issues occurring when TCP traffic is transferred over a UMTS radio link. In this paper, the characteristics of TCP and UMTS are specified and the problems of TCP over UMTS dedicated channels (DCHs) are analyzed. The problem is high delays implying low utilization of allocated resources for small file transfers or due to packet losses. High Speed Downlink Packet Access (HSDPA)’s potential to solve the problem is studied. Since in HSDPA, High-Speed Downlink shared channel (HS-DSCH) is introduced as an alternative of DCH on downlink packet access, a model of HS-DSCH is built and simulations are performed in order to compare its TCP performance with DCH. The focus is on studying retransmission delay and Block Error Rate (BLER) targets. Some scheduling methods are also compared. The results indicate that HS-DSCH gives better TCP performance than a DCH, and that advanced scheduling methods gives similar result as round robin if there are packet losses. Moreover, a somewhat surprising result is found regarding fast retransmission and channel utilization for increasing BLER targets. / I denna rapport specificeras karaktäristiken för TCP and UMTS, och problemet med TCP över dedikerade UMTS kanaler analyseras. Problemet är höga fördröjningar som medför låg utnyttjande grad av allokerade resurser vid överföring av små filer eller vid paket förluster. High Speed Downlink Packet Access (HSDPA)’s potential att lösa problemet studeras. En modell av HS-DSCH konstrueras och simuleringar utförs för att jämföra dess TCP prestanda med DCH. Fokus är på att studera återsändningsfördröjning och BLER riktvärden. Några scheduleringsmetoder jämförs också. Resultatet indikerar att HS-DSCH ger bättre TCP prestanda än DCH, och att avancerad scheduleringsmetoder ger liknande resultat som round robin vid paket förluster. Dessutom har ett ganska överraskande resultat funnits kring snabb återsändning och kanal utnyttjande grad då BLER riktvärdet ökar.