• 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 Korea Multimedia Society
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• v.15 no.8
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• pp.1038-1047
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• 2012

The SIP(Session Initation Protocol) is an application layer protocol that is used to establish, release, and change the call session of the IP telephony. In the SIP signalling networks, when the number of the UA(User Agent) requested the call session increase, the number of messages to be processed by SIP proxy server increase. It often will be caused the overload of the SIP proxy server. In this paper, we proposed the overload control method with a normal queue and a delay queue in the SIP proxy server. When it is estimated the overload of the server by the excess of the high threshold in the normal queue, new INVITE messages will be put into the delay queue to reduce the load of the server. It results in some delay of the call session from the INVITE message. Subsequently when the number of messages in the normal queue is reduced below the low threshold, the INVITE messages in the delay queue is processed. The simulation results showed that the number of the retransmission messages by our proposed method was 45% less than the one by the method with single queue. The results also showed that the average call success rate by the proposed method was 2% higher than the one by the method with single queue.

• The KIPS Transactions:PartC
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• v.9C no.4
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• pp.531-542
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• 2002

An input and output buffering ATM switch conventionally operates in either Queueloss mode or Backpressure mode. Recently, a new mode, which is called Hybrid mode, was proposed to overcome the drawbacks of Queueloss mode and Backpressure mode. In Hybrid mode, when both the destined output buffer and the originfted input buffer are full, a cell is dropped. This thesis analyzes the cell loss rate and the cell delay of Queueloss, Backpressure and Hybrid modes in a switch adopting output-port expansion scheme under uniform traffic. Output-port expansion scheme allows only one cell from an input buffer to be switched during one time slot. If several cells switch to a same destined output port, the number of maximum transfer cells is restricted to K (Output-port expansion ratio). The simulation results show that if an offered load is less than 0.9, Hybrid mode has lower cell loss rate than the other modes; otherwise, Queueloss mode illustrates the lowest cell loss rate, which is a different result from previous researches. However, the difference between Hybrid and Queueloss modes is comparably small. As expected, the average cell delay in Backpressure mode is lower than those of Queueloss mode and Hybrid mode, since the cell delay due to the retransmission of higher number of dropped cells in Backpressure mode is not considered.

• Journal of KIISE:Computing Practices and Letters
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• v.14 no.4
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• pp.369-377
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• 2008

Today, Ethernet technology is rapidly developing to have a bandwidth of 10Gbps beyond 1Gbps. In such high-speed networks, the existing method that host CPU processes TCP/IP in the operating system causes numerous overheads. As a result of the overheads, user applications cannot get the enough computing power from the host CPU. To solve this problem, the TCP/IP Offload Engine(TOE) technology was emerged. TOE is a specialized NIC which processes the TCP/IP instead of the host CPU. In this paper, we implemented a high-performance, lightweight TCP/IP(HL-TCP) for the TOE and applied it to an embedded system. The HL-TCP supports existing fundamental TCP/IP functions; flow control, congestion control, retransmission, delayed ACK, processing out-of-order packets. And it was implemented to utilize Ethernet MAC's hardware features such as TCP segmentation offload(TSO), checksum offload(CSO) and interrupt coalescing. Also we eliminated the copy overhead from the host memory to the NIC memory when sending data and we implemented an efficient DMA mechanism for the TCP retransmission. The TOE using the HL-TCP has the CPU utilization of less than 6% and the bandwidth of 453Mbps.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.45 no.4
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• pp.31-37
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• 2008

Conventional cooperative communications can attain gain of spatial diversity and path loss reduction because destination node independently received same signal from source node and relay node located between source node and destination node. However, these techniques bring about decreased spectral efficiency with relay node and increased complexity of receiver by using maximal ratio combining (MRC). This paper has proposed cooperative ARQ protocol that can improve the above problems and can get the better performance. This method can increase the spectral efficiency than conventional cooperative communication because if the received signal from source node is satisfied by the destination preferentially, the destination transmits ACK message to both relay node and source node and then recovers the received signal. In addition, if ARQ message indicates NACK relay node operates selective retransmission and we can increase reliability of system compared with that of general ARQ protocol in which source node retransmits data. In the proposed protocol, the selective retransmission and ARQ message are to be determined by comparing log-likelihood ratio (LLR) computation of received signal from source node with predetermined threshold values. Therefore, this protocol don't waste redundant bandwidth with CRC code and can reduce complexity of receiver without MRC. We verified spectral efficiency and BER performance for the proposed protocol through Monte-Carlo simulation over Rayleigh fading plus AWGN.

• The KIPS Transactions:PartC
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• v.10C no.7
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• pp.943-954
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• 2003

The mobile telecommunication system has been growing exponentially after 1990s due to the high population in a city and the growth of mobile user. In this time, the current mobile system mainly concentrates on the voice communication. However, in the next generation, mobile users want to get very diverse services via mobile terminal such as the Internet access, web access, multimedia communication, and etc. For this reason, the next generation system, such as the UMTS (Universal Mobile Telecommunication Services) system, has to support the packet data service and it will play the major role in the system. By the way, since the Web service is based on TCP, most of the Internet traffic TCP traffic. Therefore, efficient transmission of TCP traffic will take very important role in the performance of packet data service. There are many researches about improving TCP performance over wireless network. In those schemes, the UMTS system adapts the link layer retransmission scheme. However, there are rarely studies about the exact performance of the link layer retransmission scheme in the face of dynamic changes of wireless environment over the UMTS system. The dynamic changes of wireless environment, such as wireless bandwidth, can degrade TCP performance directly. So, in this paper, we simulate and analyze the TCP performance in the UMTS system with dynamic wireless environments. Then, we propose a simple scheme for minimizing TCP performance degradation. As a result of simulation, we can find that when wireless environment is changed dynamically, the probability of TCP timeout is increased, and the TCP performance is degraded very much. In this situation, the proposed simple scheme shows good performance. It saves wireless resources and reduces the degradation of TCP performance without large overhead of the base station.

• Journal of KIISE:Information Networking
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• v.29 no.4
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• pp.447-458
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• 2002

In this paper, we design and implement Wireless Transaction Protocol(WTP) and evaluate it for wireless transaction processing in mobile computing environments. The design and implementation of WTP are based on the coroutine model that might be suitable for light-weight portable devices. We test the compatibility between our product and the other products such as Nokia, Kannel and WinWAP For the evaluation of WTP, we use an Internet simulator that can arbitrary generate random wireless errors based on the Gilbert model. In our experiment, the performance of WTP is measured and compared to those of Transmission Control Protocol(TCP) and TCP for Transactions. The experiment shows that WTP outperforms the other two protocols for wireless transaction processing in terms of throughput and delay. Especially, WTP shows much higher performance In ease of high error rate and high probability of burst errors. This comes from the fact that WTP uses a small number of packets to process a transaction compared to the other two protocols and introduces a fixed time interval for retransmission instead of the exponential backoff algorithm. The experiment also shows that the WTP performance is optimized when the retransmission counter is set to 5 or 6 in case of high burst error rate.

• KIPS Transactions on Computer and Communication Systems
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• v.10 no.2
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• pp.29-38
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• 2021

The Internet including mobile networks is developing to overcoming the limitation of physical distance and providing or acquiring information from remote locations. However, the systems that use video as primary information require higher bandwidth for recognizing the situation in remote places more accurately through high-quality video as well as lower latency for faster interaction between devices and users. The emergence of the 5th generation mobile network provides features such as high bandwidth and precise location recognition that were not experienced in previous-generation technologies. In addition, the Mobile Edge Computing that minimizes network latency in the mobile network requires a change in the traditional system architecture that was composed of the existing smart device and high availability server system. However, even with 5G and MEC, since there is a limit to overcome the mobile network state fluctuations only by enhancing the network infrastructure, this study proposes a high-definition video streaming system in ultra-low latency based on the SRT protocol that provides Forward Error Correction and Fast Retransmission. The proposed system shows how to deploy software components that are developed in consideration of the nature of 5G and MEC to achieve sub-1 second latency for 4K real-time video streaming. In the last of this paper, we analyze the most significant factor in the entire video transmission process to achieve the lowest possible latency.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.1
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• pp.83-88
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• 2009

Selection relaying(SR) is usually based on signal-to-noise ratio(SNR) to decide whether or not to forward recovered symbols. However, instantaneous noise at relay is ignored, leading to the risk of erroneous retransmission induced by the relay that can be detrimental to the eventual detection of symbols at destination. To overcome this problem, we propose using new symbol error probability(SEP) related directly to reliability of received symbols instead of SNR. Simulation results show that the proposed SEP-based SR is considerably better than the conventional SNR-based SR under any relay position and threshold.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.477-480
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• 2008

In this paper, we design the TCP-Light protocol for directly connecting a sensor network with Internet. When we are monitoring the data and controlling of the sensor node, the sensor network must be connected to Internet. But TCP in wireless networks has a number of performance problems which is high bit-error rates and a hardware constraint. Moreover, the end-to-end acknowledgment and retransmission scheme employed by TCP causes expensive retransmissions along every hop of the path between the sender and the receiver. This paper introduces The TCP-Light protocol which increases TCP performance in wireless sensor networks, decreases the number of end-to-end retransmissions and decreases memory consumption.