• Journal of Communications and Networks
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• v.6 no.3
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• pp.235-244
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• 2004

There have been a lot of approaches to evaluate and predict transmission control protocol (TCP) performance in a numerical way. Especially, under the recent advance in wireless transmission technology, the issue of TCP performance over wireless links has come to surface. It is because TCP responds to all packet losses by invoking congestion control and avoidance algorithms, resulting in degraded end-to-end performance in wireless and lossy systems. By several previous works, although it has been already proved that overall TCP performance is largely dependent on its loss recovery performance, there have been few works to try to analyze TCP loss recovery performance with thoroughness. In this paper, therefore, we focus on analyzing TCP's loss recovery performance and have developed a simple model that facilitates to capture the TCP sender's behaviors during loss recovery period. Based on the developed model, we can derive the conditions that packet losses may be recovered without retransmission timeout (RTO). Especially, we have found that TCP Reno can retransmit three packet losses by fast retransmits in a specific situation. In addition, we have proved that successive three packet losses and more than four packet losses in a window always invoke RTO easily, which is not considered or approximated in the previous works. Through probabilistic works with the conditions derived, the loss recovery performance of TCP Reno can be quantified in terms of the number of packet losses in a window.

• Proceedings of the Korea Society of Information Technology Applications Conference
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• 2005.11a
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• pp.211-214
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• 2005

There are emerging many eScience applications. More and more scientists want to collaborate on their investigation with international partners without space limitation by using these applications. Since these applications have to analyze the massive raw data, scientists need to send and receive the data in short time. So today's network related requirement is high speed networking. The key point of network performance is transport protocol. We can use TCP and UDP as transport protocol but we use TCP due to the data reliability. However, TCP was designed under low bandwidth network, therefore, general TCP, for example Reno, cannot utilize the whole bandwidth of high capacity network. There are several TCP variants to solve TCP problems related to high speed networking. They can be classified into two groups: loss based TCP and delay based TCP. In this paper, I will compare two approaches of TCP variants and propose a hybrid approach for high speed networking.

• Journal of Communications and Networks
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• v.11 no.5
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• pp.518-528
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• 2009

IEEE 802.11 wireless local area network (WLAN) has become the prevailing solution for wireless Internet access while transport control protocol (TCP) is the dominant transport-layer protocol in the Internet. It is known that, in an infrastructure-based WLAN with multiple stations carrying long-lived TCP flows, the number of TCP stations that are actively contending to access the wireless channel remains very small. Hence, the aggregate TCP throughput is basically independent of the total number of TCP stations. This phenomenon is due to the closed-loop nature of TCP flow control and the bottleneck downlink (i.e., access point-to-station) transmissions in infrastructure-based WLANs. In this paper, we develop a comprehensive analytical model to study TCP dynamics in infrastructure-based 802.11 WLANs. We calculate the average number of active TCP stations and the aggregate TCP throughput using our model for given total number of TCP stations and the maximum TCP receive window size. We find out that the default minimum contention window sizes specified in the standards (i.e., 31 and 15 for 802.11b and 802.11a, respectively) are not optimal in terms of TCP throughput maximization. Via ns-2 simulation, we verify the correctness of our analytical model and study the effects of some of the simplifying assumptions employed in the model. Simulation results show that our model is reasonably accurate, particularly when the wireline delay is small and/or the packet loss rate is low.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.8 s.350
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• pp.84-90
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• 2006

With the widespread of the wireless Internet and wireless LAN, different wireless technologies such as 3G cellular networks and WLAN will cooperate to support more users and applications with higher data rate over wider areas. When a mobile node moves around in the hybrid networks, it needs to perform seamless vertical handoffs between different wireless networks to provide high performance data transmission. When an application with TCP connection in a mobile node performs a vertical handoff, TCP performance is degraded due to packet losses even though it maintains the previous TCP state information during handoff, because 3G and WLAN have different available bandwidth. In this paper, we propose a new congestion control algorithm for vertical handoff to improve the TCP performance by measuring the rough end-to-end available bandwidth and calculating the slow-start threshold. By ns-2 simulation, we show that the proposed algorithm enhances the TCP performance during vertical handoffs compared to the previous algorithms.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.1
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• pp.139-145
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• 2016

Dumbbell is the most basic topology that can be used in almost all kind of network experiment within it or just by little expansion. While Transmission Control Protocol TCP is the basic protocol that is used for the connectivity among networks and stations. TCP major and basic goal is to provide path and services to different applications for communication. For that reason TCP has to transfer a lot of data through a communication medium that cause serious congestion problem. To calculate the congestion problem, different kind of pre-cure solutions are developer which are Loss Based Variant and Delay Based Variant. While LBV keep track of the data that is going to be passed through TCP protocol, if the data packets start dropping that means congestion occurrence which notify as a symptom, TCP CUBIC use LBV for notifying the loss. Similarly the DBV work with the acknowledgment procedure that is used in when data ACK get late with respect to its set data rate time, TCP COMPOUND/VAGAS are examples of DBV. Many algorithms have been purposed to control the congestion in different TCP variants but the loss of data packets did not completely controlled. In this paper, the congestion control algorithms are implemented and corresponding results are analyzed in Dumbbell topology, it is typically used to analyze the TCP traffic flows. Fairness of throughput is evaluated for different TCP variants using network simulator (NS-2).

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.2
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• pp.304-309
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• 2004

Recently, the needs for multimedia stream such as digital audio and video in the Internet has increased tremendously. Unlike TCP, the UDP protocol, which has been used to transmit streaming traffic thorough the Internet, does not apply any congestion control mechanism to regulate the data flow thorough the shared network And it leads to congestion collapse of the Internet and results in long-term transmission delay. To avoid any adverse effect on the current Internet functionality, a now protocol of modification or addition of some functionality to perform congestion control and to reduce huge transmission delay in transmitting of multimedia stream are in study. TCP-friendly congestion control mechanism is classified into two : one is window-based congestion control scheme using general window management functionalities, the other is rate-based congestion control scheme using TCP modeling equation. In this paper, we propose an algorithm for improving the transmitting rate on a hybrid TCP-friendly congestion control scheme combined with widow-based and late-based congestion control for multimedia stream. And we also simulate the performance of improved TEAR implementation using NS. With He simulation results, we show that the improved TEAR can provide better fairness and lower rate fluctuations than TCP.

• International Journal of Internet, Broadcasting and Communication
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• v.16 no.1
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• pp.29-35
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• 2024

TCP is considered as one of the major candidate transport protocols even for constrained IoT networks..In our previous work, we investigated the congestion control mechanism of the uIP TCP. Since the uIP TCP sets the window size to one segment by default, managing the retransmission timer is the primary approach to congestion control. However, the original uIP TCP sets the retransmission timer based on the fixed RTO, it performs poorly when a radio duty cycling mechanism is enabled and the hidden terminal problem is severe. In our previous work, we proposed a TCP retransmission timer adjustment scheme for uIP TCP which adopts the notion of weak RTT estimation of CoCoA, exponential backoffs with variable limits, and dithering. Although our previous work showed that the proposed retransmission timer adjustment scheme can improve performance, we observe that the scheme often causes a node to set the retransmission timer for an excessively too long time period. In this work, we show that slightly modifying the dithering mechanism of the previous scheme is effective for improving TCP fairness.

• The KIPS Transactions:PartC
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• v.9C no.2
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• pp.197-212
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• 2002

Transmission Control Protocol (TCP) [1] has been tuned as a reliable transfer protocol for traditional networks comprising wired links and stationary hosts with same link characteristics. TCP assumes that congestion in the network be a primary cause for packet losses and unusual delays. TCP performs welt over such networks adapting to end-to-end delays and congestion losses, by standard congestion control mechanisms, such as slow-start, congestion avoidance, fast retransmit and recovery. However, networks with wireless and other lossy links suffer from significant losses due to bit errors and handoffs. An asymmetry network such as ADSL has different bandwidth for both directions. As a result, TCP's standard mechanisms incur end-to-end performance degradation in various links. In this paper, we analyze the TCP problems in wireless, satellite, and asymmetry links, and measure the new TCP mechanisms that are recommended by IETF Performance Implications of Link Characteristics (PILC) WG[2], by using Network Simulator 2 (NS-2).

• 대한공업교육학회지
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• v.33 no.1
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• pp.233-248
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• 2008

Stream Control Transmission Protocol(SCTP) is a new transport protocol that is known to provide improved performance than Transmission Control Protocol(TCP) in multi-homing environment that is having two and more IP addresses. But currently single-homed computer is used primarily that is having one IP address. To identify whether mean transfer time for SCTP is faster that for TCP in single-homed environment considering packet loss, we make up real testbed regulating the bandwidth, delay time and packet loss rate on router and observe the transfer time. We write server and client applications to measure SCTP and TCP mean transfer time by C language. Analysis of these experimental results from the testbed implementation shows that mean transfer time of SCTP is not better than performance of TCP in single homed environment exceptional case. Main reasons of performance are that SCTP compared to TCP stops transmitting data by timeout and data transmission is often delayed when SACK congestion happens. The result of study shows that elaborate performance tuning is required in developing a new SCTP module or using a implemented SCTP module.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7B
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• pp.660-666
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• 2004

Overall TCP performance represented by end-to-end throughput is largely dependent upon its loss recovery performance. In particular non-congestion packet losses caused by transmission errors degrade TCP performance seriously. Using Markov process, we analyze TCP loss recovery performance for correlated packet losses caused by multipath fading. The results show that loss recovery performance can be severely affected by burstiness in packet losses, even if overall packet loss ratio is very low.