A content-based switch makes forwarding decisions (server selections) based on an application layer information and forwards data in the application layer. After making forwarding decisions, existing content-based switches improve their forwarding performance by TCP splicing, which releases them from maintaining TCP endpoints and allows them to forward data by packet forwarding. However, once content-based switches invoke TCP splicing, they are unable to use the application layer information for forwarding decisions. Thus the existing content-based switches cannot perform a handoff of pipelined HTTP transactions, which can greatly reduce client perceived latencies. This paper proposes an asymmetric TCP splicing and a method to perform the handoff of HTTP transactions between servers. Asymmetric TCP splicing allows the content-based switches to use all the application layer information in the TCP data stream from clients to servers, although it allows the switches to forward the TCP data stream from servers to clients by packet forwarding. The proposed handoff method content-based switches support pipelined HTTP transactions in combination with asymmetric TCP splicing. In the proposed method, the content-based switch utilizes the common function of TCP (TCP half-close) to detect the end of the series of responses from the currently selected server, and it changes the forwarding destination after the client finishes receiving a series of responses from the server. Our evaluation validates that the content-based switch which supports pipelined HTTP transactions by our method can reduce client-perceived latencies when there is a large correlation between destinations of any two consecutive requests.

We proposed a new buffer management scheme for GFR services through FIFO queuing discipline. Proposed scheme can provide minimum bandwidth guarantee for GFR VCs as well as improve the fairness among the competing GFR VCs on a single FIFO queue. From simulation result, we demonstrate the proposed scheme fulfills the requirements of GFR services as well as improves the TCP throughput

TCP/IP is a key technology in the next generation mobile communication networks. A significant amount of wireless traffic will be carried in the Internet, and wireless connections will have to share network resources with wired connections. However, in a wireless network environment, TCP suffers significant throughput degradation due to the lossy characteristic of a wireless link. Therefore, to design the next generation mobile networks, it is necessary to know how much the wireless connection suffers from the degradation in comparison to the wired connection. In this paper, we discuss the fairness issue between TCP connections over wireless and wired links, and theoretically analyze the fairness of throughput between TCP over wireless link with ARQ (Automatic Repeat reQuest)-based link layer error recovery and TCP over error-free wired link. We validate our analysis by comparing the numerical results obtained from the analysis with the results obtained from computer simulation. The numerical results show that the fairness is sensitive to network propagation delay and variation rapidity of wireless link characteristic. We also obtain the theoretical lower bound of fairness.

In this paper, we propose a new IEEE 802.11 based multichannel MAC protocol, which satisfies a single transceiver constraint and which is still compatible with the IEEE 802.11 standard. Our proposed protocol does not need any additional transceivers, and we can make use of the current IEEE 802.11 hardware for the implementation of our multichannel protocol. We propose and investigate two kinds of channel selection algorithms, these are, the sender based channel selection scheme and the receiver based channel selection scheme. We evaluate the TCP performance over our proposed multichannel MAC protocol by the simulations taking account of the overhead, the time taken by changing the frequency channel and the time taken by the carrier sense in each channel. It is shown that our proposed scheme improves the performance significantly with a single transceiver. It is shown that the overhead costs are quite large in the multichannel MAC protocol, and the sender based channel selection scheme achieves better performance than that of the receiver based channel selection scheme in most cases. It is also shown that there exists a break-even value of the PLL synthesizer lockup time if we compare with the single channel IEEE 802.11 MAC protocol.

In this paper, we propose the window-based congestion control mechanism using explicit feedback in TCP over ATM ABR service. The proposed scheme is based on notifying the network status as the free buffer length at the congested link to the IP station which informs the window rate by modifying the receivers' advertised window field in TCP ACK returning to the source. Results from simulation show that our proposed algorithm improves the fairness and stability of TCP connections in general network environments even if they have the different round trip times.

Although TCP is widely used in the Internet, new specifications are still proposed and implemented. In the circumstance above, it is highly possible that some errors are detected on the communication between new and old implementations. Several test tools were developed so far. However, they do not have enough functions to allow test operators to modify test sequences suitable for their test purposes. We have developed a TCP tester which generates test sequence using test scenario. The tester performs exceptional TCP protocol behavior only when the condition specified in the test scenario is satisfied. Otherwise, it performs normal TCP behavior. The tester is implemented by modifying TCP module of NetBSD with SACK code developed by Pittsburgh Supercomputing Center. We have also evaluated implementations of congestion control and SACK algorithms using the tester.

Unicasting video streams over TCP connections is a challenging problem, because video sources cannot normally adapt to delay and throughput variations of TCP connections. This paper describes a method of extending TCP so that TCP connections can effectively carry hierarchically-encoded layered video streams, while being friendly to other competing connections. We call the method Receiver-based Delay Control (RDC). Under RDC, a TCP connection can slow down its transmission rate to avoid congestion by delaying ACK packet generation at the TCP receiver based on congestion notifications from routers. We present the principle behind RDC, argue that it is TCP-friendly, describe an implementation that uses 1-bit congestion notification from routers, and demonstrate by simulations its effectiveness in streaming hierarchically-encoded layered video.

In the Internet, flow analysis and network monitoring have been studied by various methods. Some methods try to make TCP (Transport Control Protocol) traces more readable by showing them graphically. Others such as MRTG, NetScope, and NetFlow read the traffic counters of the routers and record the data for traffic engineering. Even if all of the above methods are useful, they are made only to perform a single task. This paper describes an improved TCP Protocol Machine, a multipurpose tool that can be used for flow analysis, intrusion detection and link congestion monitoring. It is developed based on a finite state machine (automaton). The machine separates the flows into two main groups. If a flow can be mapped to a set of input symbols of the automaton, it is valid, otherwise it is invalid. It can be observed that intruders' attacks are easily detected by the use of the protocol machine. Also link congestion can be monitored, by measuring the percentage of valid flows to the total number of flows. We demonstrate the capability of this tool through measurement and working examples.

The potential role of a network in improving end-to-end TCP control is considered. Communications in the high-speed network age are revealing the limitations of end-to-end TCP control. Especially, fairness among TCP connections is one such example. Solving these problems requires not only end-to-end control but also active network control. A brief overview of related work is given, followed by the proposal of a method for adjusting the Ack interval based on network information. The principle of our algorithm is based on the relationship between TCP transmission rate at TCP sources and Ack packets intervals from the bottleneck router. Our algorithm implicitly controls transmission rate of TCP sources. Special focus is given to a scenario in which a networks has a bottleneck at a router. Simulation based on the proposed interworking algorithm, called AckAdjust, showed a good end-to-end TCP performance as to fairness between multiple TCP connections in various cases.

TCP congestion control is receiving increased attention in recent years due to their usefulness for network stability, robustness use of network buffer and bandwidth resources on an end-to-end per-connection basis. The RED scheme was designed for a network where a single dropped packet is sufficient to signal the presence of congestion to the TCP protocol. This paper applies matrix-analytic approach to analyze both the long-term and the short-term drop behaviors of a queue with RED scheme and uses this model to quantify the benefits brought about by RED. The result shows that the drop probability between RED and Drop-Tail is very close under heavy load conditions. This indicates that RED not only can resolve the synchronization problem but also has the same loss performance with Drop-Tail scheme under the heavy load circumstances. Our findings also show that the rate oscillation behavior of RED is better than Drop-Tail when TCP applies the additive-increase and multiplication-decrease mechanism. As a consequence, it can help reduce the required buffer capacity in the RED router.

Due to the fast advances in wireless networking technology, there is an increasing number of hosts using TCP/IP to connect to the Internet via wireless links. However, it is known that TCP performs poorly on paths with wireless links. This paper presents an approach to address the problem. In the proposed approach, a sender controls the size of transmitted packets and observes the number of losses for each of the controlled packet sizes. It then estimates the number of congestion losses and random losses separately. The results are used to decide whether to reduce the window size or not when a packet loss is detected. We use ns-2 simulator to evaluate the proposed approach. The simulation results show that the proposed approach is able to achieve between 25% and 150% better throughput than FACK under the byte-error rate range of 510-5 to 2010-5.

To obtain a high throughput for transmission control protocol (TCP) connections over the wireless links, we previously proposed a novel transmission power control method for code division multiple access (CDMA) packet communication systems. By using this transmission power control method, we developed a transmission power control method and a packet multiplexing method to transmit constant bit rate (CBR) and TCP packets over CDMA wireless systems. Our methods can guarantee the quality of service (QoS) for CBR connections and utilize bandwidth effectively without modifying the TCP protocol or using slot assignments. Evaluation of our methods by computer simulation showed that the proposed methods provide a near-maximum throughput and guarantee the packet loss ratio of CBR connections regardless of the number of connections.

We propose a new adaptive FEC scheme combined with ELN (Explicit Loss Notification) that was proposed for improving TCP performance in wireless cellular networks. In our method, transmission errors on the wireless link are measured at the packet level and the error status is notified the TCP sender with ELN. According to this information, an appropriate FEC code is determined in order to maximize the TCP performance. We first compare the TCP performance using Snoop Protocol, ELN and the fixed FEC, through which we find the appropriate FEC code against given BER (bit error ratio). We then show how the adaptive FEC can be realized using our solution, and also examine the appropriate observation period of measuring BER enough for the fading speed on the noisy wireless link. We finally demonstrate that our method can achieve better performance than the conventional fixed FEC by using the Gilbert model as a wireless error model.

During TCP/IP communications, MobileIP routing optimization functions causes out-of-order TCP packet sequences. To solve this problem, we propose a dynamic delayed ACK control scheme in which the wireless link-state management part notifies the upper TCP/IP layer of base-station hand-over, and at this time the TCP/IP layer sends dynamic delayed ACKs in response by using two-level-timer (i.e., hard-timer and soft-timer) processing. Simulation results confirm that applying dynamic delayed ACK control to MobileIP networks improves average throughput.

In mobile computing environments, a problem may exist between loss recovery mechanisms employed by the TCP (transmission control protocol) and RLP (radio link protocol). It is because that local retransmissions performed by the RLP could interfere with the TCP end-to-end error recovery when there are long and correlated packet losses due to bursty channel errors. That is, a spurious timeout would occur at the transport layer. In this paper, a new method is proposed to effectively suppress the occurrence of TCP spurious timeouts. In this new method a small number of ACKs (acknowledgements) is buffered at the base station prior to the emergence of every bad state period in the wireless channel, and these ACKs are henceforth released by the base station one at a time to reset the TCP sender's retransmission timer. Comprehensive comparisons between the proposed method and a baseline method are conducted through simulations to show that the improvement in throughput performance can be as large as 22%.

This paper proposes an idea for modeling aggregated TCP/IP traffic arriving at a bottleneck link by focusing on its scaling behavior. Here, the aggregated TCP/IP traffic means the IP packet traffic from many TCP connections sharing the bottleneck link. The model is constructed based on the outcomes of our previous works investigating how the TCP/IP networking mechanism affects the self-similar scaling behavior of the aggregated TCP/IP traffic in a LAN/WAN environment. The proposed traffic model has been examined from the perspective of application to network performance estimation. The examinations have shown that it models the scaling behavior and queueing behavior of actual traffic, though it neglects the interaction among TCP connections that compete with each other for the single bottleneck link bandwidth.

We developed a distributed control algorithm to solve the problem of a trade-off between transient response and stability. We applied it to a congestion control algorithm for transmitting best-effort packets such as transmission control protocol (TCP) packets over the Internet. A new transmission power control algorithm suitable for transmitting best-effort packets over the wireless Internet was also developed using the distributed control algorithm. We showed that in a steady state, TCP connections can use the bandwidth efficiently over both wired and wireless Internet when the proposed control algorithms are used. The transient response was also evaluated and it was found that the packet transmission rate and the transmission power adjusted by the proposed control algorithms converge to a steady state faster than when adjusted by conventional control algorithms while maintaining the stability of network systems.

We derived the design requirements that wireless systems and congestion control algorithms must satisfy to transmit best-effort Internet protocol (IP) packets over wireless systems. We proved that, if these requirements are satisfied, congestion control algorithms are robust against unfairness in the systems and can provide near-maximum throughputs in various environments. From the viewpoint of the design requirements, we investigated the effect of automatic repeat request (ARQ) on the throughputs of best-effort IP connections, and showed why ARQ can improve the throughputs while too large a number of retransmissions degrade them. We also investigated the effect of variance in packet transmission rates and clarified what kind of congestion control algorithm degrades the throughputs.

We propose a new analytical model of Transmission Control Protocol (TCP) in wireless environments where transmission errors occur frequently. In our proposed model, we consider the exponential increase of a congestion window and the exponential increase of a timeout back-off. Finally, we have clarified the behavior of TCP mechanisms of different versions and TCP throughput characteristics analytically. From our result, the behavior of TCP mechanisms is different in each implementation version. These differences mean that the required characteristics of wireless links are different in each implementation version. Therefore, our proposed model is a base analysis of designing wireless link mechanisms.

Mobile IP is a solution to support host mobility in the Internet. But, packets can be lost during the movement detection and registration periods of the Mobile IP. Regular TCP interprets theses packet losses as signs of network congestion, so it reduces its transmission rate by reducing its window size and slow start threshold. Besides, the multiple packet losses occurring during handoffs trigger successive retransmission timeouts at the TCP sender, causing a long communication pause even after handoff is completed. These together lead to significant throughput degradation. In this paper, we propose two new TCP schemes to reduce packet losses and to alleviate the effects of handoffs on TCP performance. TCP-MD (Movement Detection) is proposed to reduce packet losses by detecting a handoff earlier, and TCP-R (Registration) is designed to prevent packet losses by freezing data transmission during registration. The proposed schemes maintain end-to-end TCP semantics, making it possible to fully interoperate with the existing infrastructure. Only a small change is required in the mobile host, plus the implementation is simple because some Mobile IP messages are used to notify the handoff, eliminating the need for any additional messages. Simulations confirmed that the proposed schemes can give an excellent performance in an environment where the mobile host experiences frequent handoffs.