• BMB Reports
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• v.49 no.6
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• pp.325-330
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• 2016

T-complex protein 10A homolog 2 (TCP10L) was previously demonstrated to be a potential tumor suppressor in human hepatocellular carcinoma (HCC). However, little is known about the molecular mechanism. MAX dimerization protein 1 (MAD1) is a key transcription suppressor that is involved in regulating cell cycle progression and Myc-mediated cell transformation. In this study, we identified MAD1 as a novel TCP10L-interacting protein. The interaction depends on the leucine zipper domain of both TCP10L and MAD1. TCP10L, but not the interaction-deficient TCP10L mutant, synergizes with MAD1 in transcriptional repression, cell cycle G1 arrest and cell growth suppression. Mechanistic exploration further revealed that TCP10L is able to stabilize intracellular MAD1 protein level. Consistently, the MAD1-interaction-deficient TCP10L mutant exerts no effect on stabilizing the MAD1 protein. Taken together, our results strongly indicate that TCP10L stabilizes MAD1 protein level through direct interaction, and they cooperatively regulate cell cycle progression.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.3 s.357
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• pp.33-40
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• 2007

In this paper, low-rate TCP attack as one of shrew attacks is considered and the scaling based dynamic time warping (S-DTW) algorithm is introduced. The low-rate TCP attack can not be detected by the detection method for the previous flooding DoS/DDoS (Denial of Service/Distirbuted Denial of Service) attacks due to its low average traffic rate. It, however, is a periodic short burst that exploits the homogeneity of the minimum retransmission timeout (RTO) of TCP flows and then some pattern matching mechanisms have been proposed to detect it among legitimate input flows. A DTW mechanism as one of detection approaches has proposed to detect attack input stream consisting of many legitimate or attack flows, and shown a depending method as well. This approach, however, has a problem that legitimate input stream may be caught as an attack one. In addition, it is difficult to decide a threshold for separation between the legitimate and the malicious. Thus, the causes of this problem are analyzed through simulation and the scaling by maximum auto-correlation value is executed before computing the DTW. We also discuss the results on applying various scaling approaches and using standard deviation of input streams monitored.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.12
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• pp.34-43
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• 2007

Recently, in addition to the sharp increase of mobile nodes, various kinds of wireless technologies are available for mobile nodes. If IPv6 technology is applied to the network, multi-homing terminals which have several public IP addresses on one interface will be common. Accordingly, there are many research activities on mobility management for multi-interface, multi-homming nodes. In this paper we propose an extended fast handover mechanism for multi-interface MIPv6 environments that uses multi-interface FBU (MFBU) message instead of the existing FBU message. The MFBU message has the "tunnel destination" mobility option that points a specific tunnel destination other than NAR, and "T" flag that indicates the existence of tunnel destination option. The proposed mechanism can improve the TCP performance by mitigating packet reordering during FMIPv6 handover that can cause unnecessary congestion control due to 3 duplicate ACKs. In this paper, we implemented a multi-Interface MIPv6 simulator by extending a single-interface MIPv6 simulator in NS-2, and showed that the performance of TCP traffic is improved by using the proposed multi-interface fast MIPv6.

• Smart Media Journal
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• v.2 no.4
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• pp.51-59
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• 2013

With increasing transmission speed of backbone networks, it is getting to provide enough available bandwidth. However, the bandwidth is not effectively utilized in volumetric data transfer. This mainly comes from the transmission protocol, TCP, which is used for most applications. TCP is inherently difficult to adapt the available bandwidth because of it's own characteristic of transfer mechanism. UDT is a prominent application level data transfer protocol which is targeting high speed network. In this paper, we propose UDT parallel transfer technologies which is adaptive to network status and then evaluate their performance in two points of view. Firstly, we measure data transfer rate of UDT with rate congestion control methods, and compare them with basic UDT. Secondly, we apply parallel transfer technologies adapted to network status, and measure their performance. Experimental results showed that UDT rate congestion control method outperforms UDT with 106% improvement in RTT 100ms section set with jitter 30ms. In addition, performance of parallel transfer with rate congestion control method showed 107% improvement than that of parallel transfer in RTT 400ms section set with jitter 20ms.

• 제어로봇시스템학회:학술대회논문집
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• 2000.10a
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• pp.118-118
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• 2000

This paper suggests the methodology to guarantee the rea1-time service over Ethernet and TCP/IP and to solve the problems such as re-transmission due to collision and platform-transparency and independence. This paper proposes a new mechanism called Real-time Ethernet protocol combined the centralized medium control mechanism with the token passing mechanism. The centralized medium control mechanism is used to get the token, the right to transmit the data, and the token passing mechanism is used to return the token to the token controller and to transmit data within the specified time interval. The proposed Real-Time Ethernet Protocol is based on Java and Java RMI(Remote Method Invocation). The presented protocol is believed to work enough in real-time applications considering latency occurred due to the Java RMI, which is less small than data transmission time.

• Journal of Korea Multimedia Society
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• v.14 no.7
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• pp.905-916
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• 2011

Recently, the W3C has developed the HTML 5 standard to provide various multimedia services as well as a web based streaming. HTML5 based web browsers with the built-in video codecs can be used directly for streaming services, though previous web browsers should contain playback plug-in for streaming services. In addition, an adaptive streaming technology using HTTP/TCP has been standardized, since an Internet firewall does not support RTP/UDP streaming protocol but HTTP/TCP streaming protocol. In this paper, we propose the service migration mechanism among various devices in the HTML5-based adaptive HTTP streaming environments. The proposed algorithm and architecture were modeled for migrating media flows, media flow control, and session control and implemented using JavaScript. Finally, we showed that the proposed mechanism was effective to migrate HTTP streaming service among various devices by performing several experiments.

• Journal of KIISE:Information Networking
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• v.34 no.3
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• pp.143-155
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• 2007

In video streaming applications over the Internet, TCP-friendly rate control schemes are useful for improving network stability and inter-protocol fairness. However it does not always guarantee a smooth quality for video streaming. To simultaneously satisfy both the network and application requirements, video streaming applications should be quality-adaptive. In this paper, we propose a new quality adaptation mechanism to adjust the quality of congestion controlled video stream by controlling the frame rate. Based on the current network condition, it controls the frame rate and sending rate of video stream. Through the simulation, we prove that our adaptation mechanism appropriately adjusts the quality of video stream while improving network stability.

• Journal of the Korea Society of Computer and Information
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• v.22 no.10
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• pp.83-92
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• 2017

Reconnaissance is performed gathering information from a series of scanning probes where the objective is to identify attributes of target hosts. Network reconnaissance of IP addresses and ports is prerequisite to various cyber attacks. In order to increase the attacker's workload and to break the attack kill chain, a few proactive techniques based on the network-based moving target defense (NMTD) paradigm, referred to as IP address mutation/randomization, have been presented. However, there are no commercial or trial systems deployed in real networks. In this paper, we propose a threat model and the request for requirements for developing NMTD techniques. For this purpose, we first examine the challenging problems in the NMTD mechanisms that were proposed for the legacy TCP/IP network. Secondly, we present a threat model in terms of attacker's intelligence, the intended information scope, and the attacker's location. Lastly, we provide seven basic requirements to develop an NMTD mechanism for the legacy TCP/IP network: 1) end-host address mutation, 2) post tracking, 3) address mutation unit, 4) service transparency, 5) name and address access, 6) adaptive defense, and 7) controller operation. We believe that this paper gives some insight into how to design and implement a new NMTD mechanism that would be deployable in real network.

• Journal of the Korea Society of Computer and Information
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• v.14 no.10
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• pp.101-110
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• 2009

The 6LoWPAN(IPv6 Low-power Wireless Personal Area Network) performs IPv6 header compression, TCP/UDP/IGMP header compression, packet fragmentation and re-assemble to transmit IPv6 packet over IEEE 802,15.4 MAC/PHY. However, from the point of view of security. It has the existing security threats issued by IP packet fragmenting and reassembling, and new security threats issued by 6LoWPAN packet fragmenting and reassembling would be introduced additionally. If fragmented packets are retransmitted by replay attacks frequently, sensor nodes will be confronted with the communication disruption. This paper analysis security threats introduced by 6LoWPAN fragmenting and reassembling, and proposes a re-transmission mechanism that could minimize re-transmission to be issued by replay attacks. Re-transmission procedure and fragmented packet structure based on the 6LoWPAN standard(RFC4944) are designed. We estimate also re-transmission delay of the proposed mechanism. The mechanism utilizes timestamp, nonce, and checksum to protect replay attacks. It could minimize reassemble buffer overflow, waste of computing resource, node rebooting etc., by removing packet fragmentation and reassemble unnecessary.

• Journal of Communications and Networks
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• v.7 no.3
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• pp.367-376
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• 2005

The smooth handoff supported by the route optimization extension to the mobile IP standard protocol should support a packet buffering mechanism at the base station (BS), in order to reduce the degradation in TCP performance caused by packet losses within mobile network environments. The purpose of packet buffering at the BS is to recover the packets dropped during intersubnetwork handoff by forwarding the packets buffered at the previous BS to the new BS. However, when the mobile host moves to a congested BS within a new foreign subnetwork, the buffered packets forwarded by the previous BS are likely to be dropped. This subsequently causes global synchronization to occur, resulting in the degradation of the wireless link in the congested BS, due to the increased congestion caused by the forwarded burst packets. Thus, in this paper, we propose an implicit priority forwarding (IPF) packet buffering scheme as a solution to this problem within mobile IP based networks. In the proposed IPF method, the previous BS implicitly marks the priority packets being used for inter-subnetwork handoff. Moreover, the proposed modified random early detection (M-RED) buffer at the new congested BS guarantees some degree of reliability to the priority packets. The simulation results show that the proposed IPF packet buffering scheme increases the wireless link utilization and, thus, it enhances the TCP throughput performance in the context of various intersubnetwork handoff cases.