• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.11 s.353
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• pp.149-158
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• 2006

We proposed a method for improving a performance of TCP downstream between a desktop PC as a fixed host and a PDA as a mobile host in a wired and wireless network based on IEEE 802.11x wireless LAN. With data transmission between these heterogeneous terminals a receiving time during downstream is slower than that during upstream by 20% at maximum. The reason is that their congestion window size will be oscillated due to a significantly lower packet processing rate at receiver compared to a packet sending rate at sender. Thus it will cause to increase the number of control packets to negotiate their window size. To mitigate these allergies, we proposed two distinct methods. First, by increasing a buffer size of a PDA at application layer an internal processing speed of a socket receive buffer of TCP becomes faster and then the window size is more stable. However, a file access time in a PDA is kept nearly constant as the buffer size increases. With the buffer size of 32,768bytes the receiving time is faster by 32% than with that of 512bytes. Second, a delay between packets to be transmitted at sender should be given. With an inter-packet delay of 5ms at sender a resulting receiving time is faster by 7% than without such a delay.

• Journal of the Korea Society of Computer and Information
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• v.7 no.4
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• pp.127-132
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• 2002

This paper aims to adjust the window size according to the network condition that the sender determines by making the receiver participating in the congestion levels. TCP-RLDM has the measurement-based transmission strategy based on the data-receiving rate complementing TCP with the property of Additive Increase / Multiplicative Decrease. The protocol can make an performance improvement by responding differently according to the property of errors-whether congestion losses or transient transmission errors - to confront dynamically in heterogeneous environments with wired or wireless networks and delay-sensitive or -tolerant applications. By collecting data-receiving rate and the cause of errors from the receiver and by enabling sender to use the congestion avoidance strategy before occuring congestion possibly, the protocol works well at variable network environments.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.1
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• pp.45-57
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• 2005

In this paper, we extend the performance of bidirectional TCP connection over end-to-end network that uses transfer rate-based flow and congestion control. The sharing of a common buffer by TCP packets and acknowledgement has been known to result in an effect called ack compression, where acks of a connection arrive at the source bunched together, resulting in unfairness and degraded throughput. The degradation in throughput due to bidirectional traffic can be significant. Even in the simple case of symmetrical connections with adequate window size, the connection efficiency is improved about 20% for three levels of background traffic 2.5Mbps, 5.0Mbps and 7.5Mbps. Otherwise, the throughput of jitter is reduced about 50% because round trip delay time is smaller between source node and destination node. Also, we show that throughput curve is improved with connection rate algorithm which is proposed for TCP congetion avoidance as a function of aggressiveness threshold for three levels of background traffic 2.5Mbps, 5Mbps and 7.5Mbps. By analyzing the periodic bursty behavior of the source IP queue, we derive estimated for the maximum queue size and arrive at a simple predictor for the degraded throughput, applicable for relatively general situations.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.6B
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• pp.387-395
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• 2005

In this paper, we propose an end-to-end mobility management and TCP flow control scheme which considers different link characteristics for vertical handover environments. The end-to-end mobility management is performed by using SIP protocol. When a mobile node moves to a new network, it informs its movement of the correspondent node by sending SIP INFO message containing a new IP address which will be used in the new network. And then the corresponding node encapsulates all packets with the new IP address and sends them to the mobile node. in general, RTT of WLAN is shorter than RTT of cdma2000. when the MN moves from WLAN network to cdma2000 network, TCP retransmission timeout will be occurred in spite of non congestion situations. Thus, TCP congestion window size will be decreased and TCP throughput will be also decreased. To prevent this phenomenon, we propose a method using probe packets after handover to estimate a link delay of the new network. We also propose a method using bandwidth ratio of each network to update RTT. It is shown through NS-2 simulations that the proposed schemes can have better performance than the previous works.

• The Journal of the Korea institute of electronic communication sciences
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• v.11 no.3
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• pp.303-308
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• 2016

Recently, as the rapid development of network technology and the increase of services required high bandwidth such as multimedia service, the network traffic dramatically increases. This massive increase of network traffic causes some problems such as the degradation of QoS and the lack of network resources and, to solve these problems, various research to guarantee QoS have been performing. Currently, The most representative method to guarantee the QoS is the DiffServ(: Differentiated Service). The DiffServ defines the AF(: Assured Forwarding) PHB(: Per Hop Behavior) and statistically ensures the throughput over the certain level of data rate. However, the TCP congestion control method that make up the majority of the Internet traffic is not fundamentally suitable to the DiffServ that guarantees the throughput without managing the individual flow. Therefore, in this paper, we present this mismatch through the simulation as an example and propose the solution by controlling the TCP of the terminal in the network. The proposed scheme utilizes the information of the reception window size included in the ACK frame and does not require any modification of the TCP algorithms currently in use.

• Journal of Internet of Things and Convergence
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• v.1 no.1
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• pp.1-4
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• 2015

This paper aims to present the mathematical model to find the mean web object transfer latency in the slow-start phase of TCP congestion control mechanism, which is one of the main control techniques of Internet. Mean latency is an important service quality measure of end-user in the network. The application area of the proposed latency model is the narrowband environment including multi-hop wireless network and Internet of Things(IoT), where packet loss occurs in the slow-start phase only due to small window. The model finds the latency considering initial window size and the packet loss rate. Our model shows that for a given packet loss rate, round trip time and initial window size mainly affect the mean web object transfer latency. The proposed model can be applied to estimate the mean response time that end user requires in the IoT service applications.

• Journal of Navigation and Port Research
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• v.27 no.3
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• pp.283-288
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• 2003

Recently, the growth of applications and services over high-speed Internet increases, ATM networks as wide area back-bone has been a major solution. As the conventional TCP/IP suite is still the standard protocol used to support upper application on current. Internet, the issues regarding whether TCP/IP will operate efficiently on top of an ATM infrastructure and how to control its QoS still remain for studies. TCP uses a window-based protocol for flow control in the transport layer. When TCP uses the UBR service in ATM layer, the control method is only buffer management. If a cell is discarded in ATM layer, one whole packet of TCP will be lost; this fact occur the most TCP performance degradation. Several dropping strategies, such as Tail Drop, EPD, PPD, SPD, FBA, have been proposed to improve the TCP performance over ATM. In this paper, to improve the TCP performance, we propose a packet dropping scheme that is based on comparison with EPD, SPD and FBA. Our proposed scheme is applied to schemes discussed in the previous technology. Our proposed scheme does not need to know each connection's mean packet size. When the buffer exceeds the given threshold, it is based on comparison between the number of dropped packet and the approved packet. Our results are reported and discussed for comparing these discarding schemes under similar conditions. Although the number of virtual channel (VC) is increased, the simulation results showed that the proposed scheme can allocate more fairly each VC than other scheme.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.21 no.6
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• pp.1127-1136
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• 2017

In wireless networks, the packet loss due to the bit error is misinterpreted as loss due to the congestion state, so TCP congestion control occurs frequently and performance degradation occurs. This degradation also occurs in MPTCP(Multipath TCP), which is an extension protocol of original TCP. In MPTCP, the overall performance of the multipath is degraded. In this paper, we propose a congestion control scheme which measures the bandwidth on each path of MPTCP and reduces the congestion window size by the measured bandwidth when packet loss occurs, in order to solve the MPTCP performance degradation in the wireless environment. We also implemented the proposed congestion control in the Linux kernel and compared it with the original MPTCP in the testbed and real wireless networks. Experimental results show that the proposed congestion control has better throughput performance than original MPTCP congestion control in the wireless environment.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.04b
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• pp.1483-1486
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• 2002

Multicast protocols are efficient methods of group communication, but they do not support the various transmission protocol services like a reliability guarantee, FTP, or Telnet that TCPs do. The purpose of this dissertation is to find a method to utilize sewer routers to form multicasts that can simultaneously transport multicast packets and TCP packets. For multicast network scalability and error recovery the existing SRM method has been used. Three packets per TCP transmission control window size are used for transport and an ACK is used for flow control. A CBR and a SRM is used for UDP traffic control. Divided on whether a UDP multicast packet and TCP unicast packet is used simultaneously or only a UDP multicast packet transport is used, the multicast receiver with the longest delay is measured on the number of packets and its data receiving rate. It can be seen that the UDP packet and the TCP's IP packet can be simultaneously used in a server router.

• The Journal of the Korea institute of electronic communication sciences
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• v.1 no.2
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• pp.102-109
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• 2006

In this paper, we improve the performance of bidirectional TCP connection over end-to-end network that uses transfer rate-based flow and congestion control. The sharing of a common buffer by TCP packets and acknowledgement has been known to result in an effect called ack compression, where acks of a connection arrive at the source bunched together, resulting in unfairness and degraded throughput. The degradation in throughput due to bidirectional traffic can be significant. For example, even in the simple case of symmetrical connections with adequate window size, the connection efficiency is improved about 20% for three levels of background traffic 2.5Mbps, 5.0Mbps and 7.5Mbps. Otherwise, the throughput of jitter is reduced about 50% because round trip delay time is smaller between source node and destination node. Also, we show that throughput curve is improved with connection rate algorithm which is proposed for TCP congestion avoidance as a function of aggressiveness threshold for three levels of background traffic 2.5Mbps, 5Mbps and 7.5Mbps.