A critical design issue of Transmission Control Protocol (TCP) is its congestion control that allows the protocol to adjust the end-to-end communication rate based on the detection of packet loss. However, TCP congestion control may function poorly during its slow start and congestion avoidance phases. This is because TCP sends bursts of packets with the fast window increase and the ACK-clock based transmission in slow start, and respond slowly with large congestion windows especially in high bandwidth-delay product (BDP) networks during congestion avoidance. In this article, we propose an improved version of TCP, TCP-Ho, that uses an efficient congestion window control algorithm for a TCP source. According to the estimated available bandwidth and measured round-trip times (RTTs), the proposed algorithm adjusts the congestion window size with a rate between exponential growth and linear growth intelligently. Our extensive simulation results show that TCP-Ho significantly improves the performance of connections as well as remaining fair and stable when the BDP increases. Furthermore, it is feasible to implement because only sending part needs to be modified.

ACK thinning refers to the technique to discard or reduce TCP acknowledgements (ACKs) for the purpose of diverting scarce bandwidth to TCP data traffic. It has been shown that under some circumstances the technique is effective to boost the TCP throughput on wireless links, in particular the IEEE 802.11 wireless LAN (WLAN). In this letter, however, we show that ACK thinning backfires under congestion due to its cross-layer impact on the 802.11 MAC dynamics. With the ACK filtering example, we demonstrate the phenomenon and analyze the cause. Based on the analysis, we show how the IEEE 802.11 contention window size control solves the problem.

In order to improve TCP performance, the use of a PEP (Performance Enhancing Proxy) has been proposed. The PEP operates on a router along a TCP connection. When a data packet arrives at the PEP, it forwards the packet to the destination host, transmits the corresponding ACK (premature ACK) to the source host on behalf of the destination host, and stores a copy of the packet in a local buffer (PEP buffer) in case the packet needs to be retransmitted. In this paper, in accordance with a strategy that keeps the number of prematurely acknowledged packets in the PEP buffer below a fixed threshold (watermark) value, we investigate the relation between the watermark value and the average throughput. Extensive simulations show that the results can be roughly classified into two cases. In the first case, the average throughput becomes larger for larger watermark values and becomes a constant value when the watermark value is over a certain value. In the second case, although the average throughput becomes larger for lager watermark value in the same way, it decreases when the watermark value is over a certain value. We also show that the latter (former) case can occur more easily as the propagation delay in the input side network of the PEP becomes smaller (larger) and the propagation delay in the output side network of the PEP becomes larger (smaller), and also show that the latter (former) case can occur more easily as the transmission speed in the input side network becomes larger (smaller) and the transmission speed in the output side network becomes smaller (larger) while the PEP buffer capacity becomes smaller (larger).

Recently, the number of multi-homed hosts is getting large, which are equipped with multiple network interfaces to support multiple IP addresses. Although there are several proposals that aim at bandwidth aggregation for multi-homed hosts, few of them support mobility. This paper proposes a new framework called AMS: Aggregate-bandwidth Multi-homing Support. AMS provides functions of not only bandwidth aggregation but also mobility by responding to the changes of the number of connections during communication without the support of underlying infrastructure. To achieve efficient data transmission, AMS introduces a function called address pairs selection to select an optimal combination of addresses of the peer nodes. We implemented AMS in the kernel of NetBSD and evaluated it in our test network, in which dummynet was used to control bandwidth and delay. The measured results showed that AMS achieved ideal bandwidth aggregation in three TCP connections by selecting optimal address pairs.

Active Queue Management (AQM) based on nonlinear difference equations is proposed to solve the end-to-end TCP network congestion problem. The proposed AQM scheme can guarantee the stability of the multiple bottleneck network by nonlinear control of dropping probability of the routers by imposing some restrictions on the AQM parameter. Nonlinear control often relies on some heuristics and network traffic controllers that appear to be highly correlated with the multiple bottleneck network status. Based on the proposed nonlinear difference equations for TCP flows control across the network, this paper reveals the reasons of congestion of multiple bottleneck AQM, and provides a theorem for avoiding network congestion. Moreover, we give simulations to verify the results for nonlinear control of the multiple bottleneck network congestion.

We previously proposed a new version of TCP, called Inline measurement TCP (ImTCP), in [2],[3]. The ImTCP sender adjusts the transmission intervals of data packets and then utilizes the arrival intervals of ACK packets for available bandwidth estimation. This type of active measurement is preferred because the obtained results are as accurate as those of other conventional types of active measurement, even though no extra probe traffic is injected onto the network. In the present research, we develop a new capacity measurement function and combine it with ImTCP in order to enable simultaneous measurement of both capacity and available bandwidth in ImTCP. The capacity measurement algorithm is a new packet-pair-based measurement technique that utilizes the estimated available bandwidth values for capacity calculation. This new algorithm promises faster measurement than current packet-pair-based measurement algorithms for various situations and works well for high-load networks, in which current algorithms do not work properly. Moreover, the new algorithm provides a confidence interval for the measurement result.

Overlay networks are expected to be a promising technology for the realization of QoS (Quality of Service) control. Overlay networks have recently attracted considerable attention due to the following advantages: a new service can be developed in a short duration and it can be started with a low cost. The definition and necessity of the overlay network is described, and the classification of various current and future overlay networks, particularly according to the QoS feature, is attempted. In order to realize QoS control, it is considered that routing overlay and session overlay are promising solutions. In particular, session and overlay networks are explained in detail since new TCP protocols for QoS instead of current TCP protocols that control congestion in the Internet can be used within overlay networks. However, many open issues such as scalability still need further research and development although overlay networks have many attractive features and possess the potential to become a platform for the deployment of new services.

Fairness among uplink and downlink TCP flows can not be supported in infrastructure IEEE 802.11 Wireless LANs using the distributed coordination function (DCF). In order to resolve this problem, we propose the Distributed Access Time Control (DATC) scheme, in which each station controls the rates of its TCP flows based on channel access time. The DATC scheme can provide per-station fairness even when the number and direction of flows in each station are different. This scheme can be simply implemented in mobile stations without having to modify the access points. The properties of the proposed scheme are investigated and its effectiveness is verified through simulations.

An overlay traffic control is a way to provide flexible and deployable QoS mechanisms over existing networks, such as the Internet. While most of QoS mechanisms proposed so far require router supports, overlay QoS mechanisms rely on traffic control at transport layer without modifying existing routers in the network. Thus, traffic control algorithms, which are implemented at traffic sources or PEPs (Performance Enhancement Proxies), play a key role in an overlay QoS mechanism. In this paper, we propose an end-to-end prioritization scheme using TCP-Westwood Low-Priority (TCPW-LP), a low-priority traffic control scheme that maximizes the utilization of residual capacity without intrusion on coexisting foreground flows. Simulation and Internet measurement results show that TCPW-LP appropriately provides end-to-end low-priority service without any router supports. Under a wide range of buffer capacity and link error losses, TCPW-LP appropriately defers to foreground flows and better utilizes the residual capacity than other proposed priority schemes or even TCP Reno.

In the present paper, ImTCP-bg, a new background TCP data transfer mechanism that uses an inline network measurement technique, is proposed. ImTCP-bg sets the upper limit of the congestion window size of the sender TCP based on the results of the inline network measurement, which measures the available bandwidth of the network path between the sender and receiver hosts. ImTCP-bg can provide background data transfer without affecting the foreground traffic, whereas previous methods cannot avoid network congestion. ImTCP-bg also employs an enhanced RTT-based mechanism so that ImTCP-bg can detect and resolve network congestion, even when reliable measurement results cannot be obtained. The performance of ImTCP-bg is investigated through simulations, and the effectiveness of ImTCP-bg in terms of the degree of interference with foreground traffic and the link bandwidth utilization is also investigated.

In this paper, we present laboratory and field experimental results using High Speed Downlink Packet Access (HSDPA) test-beds in order to reveal the actual HSDPA performance based on key technologies such as base station (BS) scheduling, adaptive modulation and coding, hybrid automatic repeat request, and advanced receiver design. First, this paper evaluates the effects of advanced user equipment capabilities such as the maximum number of multi-codes, transmit diversity, receive diversity, and a chip equalizer. Increases in throughput of 60% and 85% due to using 10 and 15 codes were observed compared to 5 codes, respectively. The gain of 22% was obtained by applying closed-loop transmit diversity to the HSDPA network. Receive diversity improves the throughput in the region from low to high signal-to-interference ratio, and the gain of 45% was obtained by applying receive diversity to the conventional RAKE receiver. A throughput gain of approximately 17% due to the use of the chip equalizer was obtained and it was observed mainly in the high Ior/Ioc region and under multi-path conditions. Second, field experiments are conducted to elucidate the effects of multi-user diversity using a BS scheduling algorithm, and reveal that proportional fairness scheduling provides both the increase in sector throughput of 18% and a sufficient degree of fairness among users. The transmit control protocol (TCP)-level throughput performance is also investigated in order to reveal the actual end-user throughput. The results show that the throughput rate of approximately 90% of the throughput of the MAC-hs layer is achieved in the TCP layer in the laboratory experiments and in the field experiments.

In this letter, we suggest a rate-based supervisory congestion control scheme for the ad hoc networks that use TCP as the transport protocol. This scheme makes it possible for the TCP sender to distinguish the causes of packet loss. In addition, this scheme guarantees the fair sharing of the available bandwidth among the connections. We show the reliability of our scheme by using the supervisory control framework and simulations confirm the effectiveness of our scheme.

Two congestion control schemes designed specifically to handle changes in the datalink interface of a mobile host are presented. The future mobile environment is expected to involve multimode connectivity to the Internet and dynamic switching of the connection mode depending on network conditions. The conventional Transmission Control Protocol (TCP), however, is unable to maintain stable and efficient throughput across such interface changes. The two main issues are the handling of the change in host Internet Protocol (IP) address, and the reliability and continuity of TCP flow when the datalink interface changes. Although existing architectures addressing the first issue have already been proposed, the problem of congestion control remains. In this paper, considering a large change in bandwidth when the datalink interface changes, two new schemes to address these issues are proposed. The first scheme, Immediate Expiration of Timeout Timer, detects interface changes and begins retransmission immediately without waiting for a retransmission timeout as in existing architectures. The second scheme, Bandwidth-Aware Slow Start Threshold, detects the interface change and estimates the new bandwidth so as to set an appropriate slow start threshold for retransmission. Through simulations, the proposed schemes are demonstrated to provide marked improvements in performance over existing architectures.

Satellite Internet is one of the most important networks for emergency communications because of its tolerant of disasters such as earthquake. Therefore, satellite Internet has received considerable attention over recent years. However, most standard implementations of TCP congestion control method perform poorly in satellite Internet due to its high bit error rate and long propagation delay. This paper proposes a new TCP congestion control method called TCP-STAR to improve the throughput over satellite Internet. TCP-STAR has three new mechanisms, namely Congestion Window Setting (CWS) based on available bandwidth, Lift Window Control (LWC), and Acknowledgment Error Notification (AEN). CWS can resist the reduction of the transmission rate when data losses are caused by bit error. LWC is able to increase the congestion window quickly based on the estimated available bandwidth. AEN can avoid the reduction of the throughput by mis-retransmission of data. The mis-retransmission is caused by ack losses or delay. Simulations show that TCP-STAR can obtain the best throughput comparing with other TCP variants (TCP-J and TCP-WestwoodBR). Furthermore, we found that the fairness of TCP-STAR is a little lower than that of TCP-WestwoodBR. However, the fairness of TCP-STAR is equal to TCP-J.

In this paper, we propose Split Multi-path Routing protocol with Load Balancing policy (SMR-LB) to improve TCP performance in mobile ad hoc networks. In SMR-LB, each intermediate node records how many primary paths are attempted to construct as well as which source nodes attempt to construct the primary path. Each intermediate node decides which primary path should be constructed by using the primary path and the source node ID information. As a result, SMR-LB can balance the loads and so reduce the probability of congestion and avoid the continuous link breakage time between the specific source and destination pair. Computer simulation results show that SMR-LB can improve TCP performance compared with the conventional protocols.

It is well known that TCP does not fully utilize the available bandwidth in fast long-distance networks. To solve this scalability problem, several high speed transport protocols have been proposed. They include HighSpeed TCP (HS-TCP), Scalable TCP (S-TCP), Binary increase control TCP (BIC-TCP), and H-TCP. These protocols increase (decrease) their window size more aggressively (slowly) compared to standard TCP (STD-TCP). This paper aims at evaluating and comparing these high speed transport protocols through computer simulations. We select six metrics that are important for high speed protocols; scalability, buffer requirement, TCP friendliness, TCP compatibility, RTT fairness, and responsiveness. Simulation scenarios are carefully designed to investigate the performance of these protocols in terms of the metrics. Results clarify that each high speed protocol successfully solves the problem of STD-TCP. In terms of the buffer requirement, S-TCP and BIC-TCP have better performance. For TCP friendliness and compatibility, HS-TCP and H-TCP offer better performance. For RTT fairness, BIC-TCP and H-TCP are superior. For responsiveness, HS-TCP and H-TCP are preferred. However, H-TCP achieves a high degree of fairness at the expense of the link utilization. Thus, we understand that all the proposed high speed transport protocols have their own shortcomings. Thus, much more research is needed on high speed transport protocols.

As computer hardware components are achieving greater speeds, network link bandwidths are becoming wider. A number of enhancements to TCP have been developed in order to fully exploit these improvements in network infrastructures, including TCP window scale option, SACK option, and HighSpeed TCP (HSTCP) modifications. However, even with these enhancements, TCP cannot provide satisfactory performance in high-speed long-delay networks. As a means addressing this problem, gentle HighSpeed TCP (gHSTCP) has been proposed in [1]. However, its effectiveness has only been demonstrated in simulation experiments. In the present paper, a refined gHSTCP algorithm is proposed for application to real networks. The performance of the refined gHSTCP algorithm is then assessed experimentally. The refined gHSTCP algorithm is based on the original algorithm, which uses two modes (Reno mode and HSTCP mode) in the congestion avoidance phase and switches modes based on RTT increasing trends. The refined gHSTCP algorithm compares two RTT thresholds and judges which mode will be used. The performance of gHSTCP is compared with TCP Reno/HSTCP and parallel TCP mechanisms. The experimental results demonstrate that gHSTCP can provide a better tradeoff in terms of utilization and fairness against co-existing traditional TCP Reno connections, whereas HSTCP and parallel TCP suffer from the trade-off problem.

We propose a probing-based channel adaptive video streaming method to provide seamless video transmission over the wireless 3G network. In the proposed method, the available bandwidth (AB) of the end-to-end path is estimated by analyzing RTP packets at the streaming client and the network/client buffer status (NCBS) is predicted by examining the RTCP receiver report (RR) and the application defined packet (APP) including the AB information. Using the estimated AB, the NCBS, and the stored multirate bitstream information, the proposed streaming method determines the next transmission bitrate precisely. Experimental results indicate that the proposed method can be effectively used for video streaming.

It is well known that deploying a proxy at the boundary of wireless networks and the Internet is able to improve the performance of transmission control protocol (TCP) over wireless links. Snoop protocol, acting like a transport layer proxy, performs local retransmissions for packets corrupted by wireless channel errors. In this letter, an improvement for the Snoop protocol is proposed to shorten the time spent on local recovery by sending extra copies in every local retransmission attempt. This enables TCP to quickly return to normal, effectively eliminating several of the problems that may cause throughput degradation.

IETF MANET (Mobile Ad Hoc Network) WG has standardized OLSR (Optimized Link State Routing) as its proactive routing protocol. In addition, TCP (Transmission Control Protocol) is still needed for MANET thanks to its suitability for smooth integration with the fixed Internet. In particular, TCP-Vegas is a well-known transport protocol that can efficiently take account of network condition. However, TCP-Vegas that requires an accurate BaseRTT estimation cannot be directly applied to MANET because a route change makes the estimated BaseRTT obsolete. We propose a technique to improve the performance of TCP-Vegas by considering the route change, and show the performance improvement through simulation study using the ns-2 simulator.