• The KIPS Transactions:PartC
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• v.11C no.2
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• pp.229-234
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• 2004

In this study, the performance is compared between RFC compatible normal TCP and several speed constraints Ignored TCP. To do these, the main algorithms that constraints the transmit rate of TCP are modified. We have modified TCP protocol stack in a Linux kernel to compare the speeds between the standard TCP and our modified TCP. We find that if the destination is short distance away from the source and packet error is scarce then the speed differences between normal and modified TCP nay be negligible. However, if the destination is far away from the source and slow start algorithm is not adopted then the transfer time for small file is different greatly In addition, if packet error occurred frequently, our modified TCP is faster than the standard TCP regardless of distance.

• The KIPS Transactions:PartC
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• v.14C no.2
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• pp.163-170
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• 2007

The next generation communication environment consists of various wireless access networks with distinct features that are configured as an overlay topology. In the network environments, the frequency of hand overs should be minimized and the error propagation should be solved in order to provide high-quality multimedia services to mobile users. Therefore, we propose an performance enhancement approach, based on mSCTP, that provides high quality multimedia services to mobile users by ameliorating the error propagation problem. We utilizes the following four functions: 1) the separation of transmission paths according to the types of frames. 2) retransmission strategy to minimize the loss rate of frames, 3) Foced vertical handover execution by utilizing bicasting, 4) using the stability period in order to reduce the effect of the ping pong phenomenon. The simulation results show that the proposed approach provides seamless multimedia service to mobile users by achieving error resilience.

• Journal of KIISE:Information Networking
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• v.28 no.3
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• pp.347-354
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• 2001

This paper proposes and analyzes an error control scheme which tries to recover the transmission error within the deadline of a message on FDDI networks. The error control procedure does not interfere other normal message transmissions by delivering retransmission request via asynchronous traffic as well as by delivering retarnsmitted message via overallocated bandwidth which is inevitably produced by the bandwidth allocation scheme for hard real-time guarantee. The receiver counts the number of tokens which it meets, determines the completion of message transmission, and finally sends error report. The analysis results along with simulation performed via SMPL show that the proposed scheme is able to enhance the deadline meet ratio of messages by overcoming the network errors. Using the proposed error control scheme, the hard real-time network can be built at cost lower than, but performacne comparable to the dual link network.

• Journal of KIISE:Information Networking
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• v.31 no.6
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• pp.595-607
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• 2004

Networks often evolve to provide a host with multiple access points to the Internet. In this paper, we propose a transport layer load distribution mechanism utilizing the multiple network interfaces simultaneously. We specifically propose an extension of Stream Control Transmission Protoco1 (SCTP) to have load sharing over multiple network interfaces. We named the particular service provided by the Proposed load sharing mechanism to be LS (Load Sharing) mode service. LS mode service is based on the following four key elements: (i) the separation of flow control and congestion control, (ii) congestion window based striping, (iii) redundant packet retransmission for fast packet loss recovery, (iv) a novel mechanism to keep track of the receiver window size with the SACKS even if they arrive out-of-order. Through simulations, it is shown that the proposed LS mode service can aggregate the bandwidth of multiple paths almost ideally despite of the disparity in their bandwidth. When a path with a delay of 100% greater is utilized as the second path, the throughput is enhanced about 20%.

• Journal of KIISE:Information Networking
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• v.31 no.1
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• pp.62-69
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• 2004

The cluster structure reduces the number of retransmission messages, when a broadcast to all hosts in ad hoc network is needed. A cluster maintenance scheme is employed to preserve this advantage from time to time. However, most of the cluster maintenance schemes require additional control messages for cluster reformation as well as control messages for acquiring neighbor information. This mitigates the advantages of employing cluster structure in ad hoc network. In this paper, a cluster maintenance scheme which forces only clusterheads to broadcast control messages during hello time is proposed. When the cluster reformation is needed, the proposed scheme employs a strategy to reduce the control messages to a minimum. In these processes, the proposed scheme doesn't violate the definition of 2-cluster and produces the clusters in fully distributed method. The simulation results prove that our scheme is better than LCC(1).

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.10 no.5
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• pp.424-434
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• 2017

In industrial wireless sensor networks, many applications require integrated QoS supporting. This paper proposes a reliable protocol for real-time monitoring in industrial wireless sensor networks. Retransmission is well-known to recover the transmission failure, however, this might cause the time delay to violate the real-time requirement. The proposed protocol exploits broadcasting feature of wireless networks and the temporal opportunity concept. The opportunities to relay the data packets are shared by the broadcasting feature and the temporal opportunity concept maximize the number of candidates in communication. Simulation results show that the proposed protocol is superior to the existing real-time protocols in term of real-time service and reliability.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.11 no.3
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• pp.259-269
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• 2018

Multipath TCP (MPTCP) can be expected to provide improved network performance because it transmits data through multiple paths. However, Bufferbloat, which unexpectedly occurs in the transmission path, degrades not only the performance of the corresponding path but also the performance of other paths, so that the performance is worse than that of a single TCP. In this paper, we propose the transmission status monitoring method at the sender's MPTCP level and also HoL packets retransmission algorithm in order to solve the Bufferbloat problem. The proposed algorithm enables Bufferbloat detection by the sender side independently, and it can resolve the HoL blocking problem by identifying the HoL packet in the proposed transmission status monitoring buffer and retransmitting it to the normal path. Simulation results based on NS-3 show that the proposed algorithm achieves the improved throughput performance up to 22.8% compared to the existing MPTCP, and the average number of queued packets in the sender and receiver's buffers is decreased to 44.3% and 9.2%, respectively.

• The Journal of the Korea Contents Association
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• v.8 no.6
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• pp.43-53
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• 2008

Hyper text transfer protocol (HTTP) over transmission control protocol (TCP) is currently used to transfer objects in the Internet. Stream control transmission protocol (SCTP), an alternative to TCP, which allows for independent delivery among streams, and can thus reduce the mean response delay of web object. We present an analytical model to find the mean response delay for HTTP over SCTP, therefore, estimate the effectiveness of SCTP over TCP. Typical TCP delay models assume the wired environment. On the contrary, the proposed model in this paper assumes the multiple packet losses and wireless environment where fast retransmission is not possible due to small window. The estimated mean response time can be used the benchmark to meet quality of service (QoS) at end-user. We validate the accuracy of our model using experiments. It is shown that the differences between the results from model and those from experimental are very small below 6 % on average. We also find that the mean response delay for HTTP over SCTP is less than that for HTTP over TCP.

• The Journal of the Acoustical Society of Korea
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• v.21 no.2
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• pp.93-101
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• 2002

This paper presents the Internet audio broadcasting technology based on the streaming technology. In this paper, we choose the MPEG-2 AAC for multimedia data, and for the streaming of this data we use RTP/RTCP protocol. We use RTSP protocol for the control of streaming data and TCP/IP for the exchange of information between server and client. By using all of these protocols and MPEBG-2 AAC, we explain the implementation method for the unicast/multicast streaming server/client system. Our system was tested by ETRI intranet, which is connected by 2000 researchers. Experimental result show that our system can be process the packet loss and jitter by retransmission and variable length buffer. Multicast streaming server can be used for the audio broadcasting service inside the company, unicast streaming server can be used for the AOD (Audio On Demand) service.

• Journal of Satellite, Information and Communications
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• v.10 no.3
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• pp.53-56
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• 2015

In this paper, we want provide improved services with faster transmission, IEEE 802.11n was standardized. A-MPDU (Aggregation MAC Protocol Data UNIT) is a vital function of the IEEE 802.11n standard, which was proposed to improve transmission rate by reducing frame transmission overhead. In this paper, we show the problems of TCP retransmission with A-MPDU and propose a solution utilizing the property of TCP cumulative ACK. If the transmission of an MPDU subframe fails, A-MPDU mechanism allows selective re-transmission of failed MPDU subframe in the MAC layer. In TCP traffic transmission, however, a failed MPDU transmission causes TCP Duplicate ACK, which causes unnecessary TCP re-transmission. Furthermore, congestion control of TCP causes reduction in throughput. By supressing unnecessary duplicate ACKs the proposed mechanism reduces the overhead in transmitting redundant TCP ACKs, and transmitting only a HS-ACK with the highest sequence number. By using the RACK mechanism, through the simulation results, it was conrmed that the RACK mechanism increases up to 20% compared the conventional A-MPDU, at the same time, it tightly assures the fairness among TCP flows.