• Journal of KIISE:Information Networking
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• v.33 no.5
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• pp.369-379
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• 2006

In recent years, there have been many researches about Ad-hoc Networks which is available to communicate freely between mobile devices by using multi-hop without any support of relay base or access point. TCP that used the most widely transport protocol in the Internet repeats packet loss and retransmission because it increases congestion window size by using reactive congestion control until packet loss occurs. As a result of this, energy of mobile device is wasted unnecessarily. In this paper, we propose TCP-New Veno in order to improve the energy efficiency of mobile device. According to the state of network, the scheme adjusts appropriate size of congestion window. Therefore, the energy efficiency of mobile device and utilization of bandwidth are improved by the scheme. From the simulation by using ns-2, we could see more improved energy efficiency with TCP-New Veno than those with TCP in Ad-hoc Networks.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.6B
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• pp.430-439
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• 2008

In this paper, we propose a sender-based TCP congestion control scheme for downward vertical handover (DVHO), in which mobile node moves from a cellular network to a wireless LAN. DVHO can give rise to severe performance problems in TCP throughput because it causes a drastic change of link characteristics. Particularly, TCP executes falsely congestion control by packet reordering, which is occurred from link delay difference between a cellular link and a wireless LAN link. Therefore, the congestion window is reduced. And unnecessary retransmissions wastes bandwidth. To solve these problems, we propose a method using estimated round-trip time in cellular link to process duplicated ACKs from reordering. Furthermore, the duplicated ACKs are used to the control congestion window size. Simulation result shows that the proposed scheme can solve problems. Moreover, the proposed scheme can have better performance than TCP New Reno and nodupack.

• 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.

• The Journal of the Korea Contents Association
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• v.7 no.1
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• pp.131-138
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• 2007

In wired networks, packet losses mostly occur due to congestion. TCP reacts to the congestion by decreasing its congestion window, thus to reduce network utilization. In wireless networks, however, losses may occur due to the high bit-error rate of the transmission medium or due to fading and mobility. Nevertheless, TCP still reacts to packet losses according to its congestion control scheme, thus to reduce the network utilization unnecessarily. This reduction of network utilization causes the performance of TCP to decrease. In this paper, we predict packet loss by using RSS(Received Signal Strengths) on the wireless and suggest adding RSS flag bit in ACK packet of MH. By using RSS flag bit in ACK, the FH(Fixed Host) decides whether it adopt congestion control scheme or not for the maximum throughput. The result of the simulation by NS-2 shows that the proposed mechanism significantly increases sending amount and receiving amount by 40% at maximum.

• 전자공학회논문지 IE
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• v.48 no.3
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• pp.16-19
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• 2011

Differentiated Service is an IP QoS ensuring method based on packet marking that allows packets to be prioritized according to user requirements. During the time of congestion, more low priority packets are dropped than high priority packets. Different policy models are used to determine how to mark the packet. This paper investigated the performance of Differentiated Service using time sliding window with 3 color marking (TSW3CM). Simulation results using NS-2 showed that Differentiated Service can provide the quality of service requirements.

• Journal of Communications and Networks
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• v.8 no.3
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• pp.351-359
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• 2006

In this article, we present a new slow-start variant, which improves the throughput of transmission control protocol (TCP) Vegas. We call this new mechanism Gallop-Vegas because it quickly ramps up to the available bandwidth and reduces the burstiness during the slow-start phase. TCP is known to send bursts of packets during its slow-start phase due to the fast window increase and the ACK-clock based transmission. This phenomenon causes TCP Vegas to change from slow-start phase to congestion-avoidance phase too early in the large bandwidth-delay product (BDP) links. Therefore, in Gallop-Vegas, we increase the congestion window size with a rate between exponential growth and linear growth during slow-start phase. Our analysis, simulation results, and measurements on the Internet show that Gallop-Vegas significantly improves the performance of a connection, especially during the slow-start phase. Furthermore, it is implementation feasible because only sending part needs to be modified.

• 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.

• The Journal of Engineering Research
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• v.6 no.2
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• pp.39-46
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• 2004

In this paper, we carry out a performance study related to the Broadband Network. For this network, it has been proposed to use the leaky bucket as a way of controlling congestion within the network. On the top of leaky bucket type rate based congestion control scheme for high speed networks, a user will typically operate an error control scheme for retransmitting lost and erroneous packets. We propose a performance model in order to study the interaction between a user's error control scheme and the leaky bucket congestion control scheme for high speed networks. Simulation results show that parameters such as the window size and the token generation rate in the leaky bucket are key factors affecting the end-to-end delay.

• Annual Conference of KIPS
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• 2007.05a
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• pp.1288-1290
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• 2007

Congestion in WSN increases energy dissipation rates of sensor nodes as well as loss of packets and thereby hinders fair and reliable event detections. We find that one of the key reasons of congestion in WSN is allowing sensing nodes to transfer as many packets as possible. This is due to the use of CSMA/CA that gives opportunistic media access control. In this paper, we propose an energy efficient congestion avoidance protocol that includes source count based hierarchical and load adaptive medium access control. Our proposed mechanism ensures load adaptive media access to the nodes and thus achieves fairness in event detection. The results of simulation show our scheme exhibits more than 90% delivery ratio with retry limit 1, even under bursty traffic condition which is good enough for reliable event perception.

• 대한공업교육학회지
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• v.32 no.2
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• pp.199-216
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• 2007

Stream Control Transmission Protocol(SCTP) is a transport layer protocol to support the data transmission. SCTP is similar to Transmission Control Protocol(TCP) in a variety of aspects. However, several features of SCTP including multi-homing and multi-streaming incur the performance difference from TCP. This paper highlights the data transfer during the initial slow start phase in SCTP congestion control composed of slow start phase and congestion avoidance phase. In order to compare the mean transfer time between SCTP and TCP, we experiment with different performance parameters including bandwidth, round trip time, and data length. By varying data length, we also measure the corresponding initial window size, which is one of factors affecting the mean transfer time. For the experiment, we have written server and client applications by C language using SCTP socket API and have measured the transfer time by ethereal program. We transferred data between client and server using round-robin method. Analysis of these experimental results from the testbed implementation shows that larger initial window size of SCTP than that of TCP brings the reduction in the mean transfer time of SCTP compared with TCP by 15 % on average during the initial slow start phase.