• Journal of Korea Society of Industrial Information Systems
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• v.16 no.5
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• pp.21-30
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• 2011

In this paper, we propose a real-time service based on retransmission slot in low rate WPAN. In the proposed scheme, during the communication period of the beacon-enabled mode in LR-WPAN standard, a special GTSs is dynamically assigned for retransmission of the packet that fails during a real-time service such as voice. This provides a time diversity in the severe channel error environments to support the required QoS. Analytical results show that this scheme achieves a much higher throughput and better transmission success rate per GTS slot than conventional schemes such as a common reserved scheme in LR WPAN.

• Convergence Security Journal
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• v.10 no.2
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• pp.1-9
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• 2010

Snort is an open source network intrusion prevention and detection system (IDS/IPS) developed by Sourcefire. Combining the benefits of signature, protocol and anomaly-based inspection, Snort is the most widely deployed IDS/IPS technology worldwide. With millions of downloads and approximately 300,000 registered users. Snort identifies network indicators by inspecting network packets in transmission. A process on a host's machine usually generates these network indicators. This means whatever the snort signature matches the packet, that same signature must be in memory for some period (possibly micro seconds) of time. Finally, investigate some security issues that you should consider when running a Snort system. Paper coverage includes: How an IDS Works, Where Snort fits, Snort system requirements, Exploring Snort's features, Using Snort on your network, Snort and your network architecture, security considerations with snort under digital forensic windows environment.

• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.49 no.6
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• pp.515-519
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• 2021

IEEE 1588 PTP is a precision time protocol in which two systems synchronize without the aid of GPS by exchanging packets including transmission/reception time information. In the time synchronization process, the propagation delay time can be calculated and the distance between the two systems can be measured using this. In this paper, we proposed a method to improve the distance measurement precision less than the modulation symbol period using the timing error information extracted from the preamble of the received packet. Computer simulations show that the distance measurement precision is proportional to the length of the preamble PN sequence and the signal-to-noise ratio.

• Journal of Korea Multimedia Society
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• v.15 no.5
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• pp.587-594
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• 2012

IEEE 802.15.6 standard technology is proposed for low-power wireless communication in, on and around body, where vital signs such as pulse, blood pressure, ECG, and EEG signals are transmitted as a type of data packet. Especially, these vital signs should be delivered in real time, so that the latency from slave node to hub node can be one of the pivotal performance requirements. However, in the case of IEEE 802.15.6 technology data retransmission caused by transmission failure can be done in the next superframe. In order to overcome this limitation, we propose an adaptive polling algorithm for IEEE 802.15.6 technology. The proposing algorithm makes the hub to look for an appropriate time period in order to make data retransmission within the superframe. Through the performance evaluation, the proposing algorithm achieves a 61% and a 73% latency reduction compared to those of IEEE 802.15.6 technology in the environment of 70% traffic offered load with 10ms and 100ms superframe period. In addition, the proposing algorithm prevents bursty traffic transmission condition caused by mixing retransmission traffic with the traffic reserved for transmission. Through the proposing adaptive polling algorithm, it will be possible to transmit time-sensitive vital signs without severe traffic delay.

• The Journal of the Korea Contents Association
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• v.11 no.2
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• pp.69-78
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• 2011

Recently, the bandwidth available to an end user has been dramatically increasing with the advancing of network technologies. This high-speed network naturally requires faster and/or stable data transmission techniques. The UDT(UDP based Data Transfer protocol) is a UDP based transport protocol, and shows more efficient throughput than TCP in the long RTT environment, with benefit of rate control for a SYN time. With a NAK event, however, it is difficult to expect an optimum performance due to the increase of fixed sendInterval and the flow control based on the previous RTT. This paper proposes a rate control method on following a NAK, by adjusting the sendInterval according to some degree of RTT period which calculated from a set of experimental results. In addition, it suggests an improved flow control method based on the TCP vegas, in order to predict the network congestion afterward. An experimental results show that the revised flow control method improves UDT's throughput about 20Mbps. With combining the rate control and flow control proposed, the UDT throughput can be improved up to 26Mbps in average.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37B no.9
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• pp.814-821
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• 2012

We propose a network processor called an OmniFlow processor capable of wire-speed flow management by a hardware-based flow admission control(FAC) in this paper. Because the OmniFlow processor can set up and release a wire-speed connection for flows, the update period of flows can be set to a short time, and only active flows can be effectively managed by terminating a flow that does not have a packet transmitted within this period. Therefore, the FAC can be used to provide a reliable transmission of UDP as well as TCP applications. This processor is fabricated in 65nm CMOS technology, and total gate count is 25 million. It has 40 Gb/s throughput performance in using the 32 RISC cores when maximum operating frequency is 555MHz.

• Journal of Korea Multimedia Society
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• v.14 no.1
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• pp.54-75
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• 2011

PoC(Push-to-talk Over Cellular) is an integrated technology of group voice calls, video calls and internet based multimedia services. If a PoC user can not participate in the PoC session for various reasons such as an emergency situation, lack of battery capacity, then the user can use the PoC Box which has a similar functionality to the MM Box in the MMS(Multimedia Messaging Service). The RTSP(Real-Time Streaming Protocol) method is recommended to be used when there is a transmission session between the PoC box and a terminal. Since the existing VOD service uses a wired network, the packet size of RTSP-based VOD service is huge, however, the PoC service has wireless communication environments which have general characteristics to be used in RTSP method. Packet loss in a wired communication environments is relatively less than that in wireless communication environment, therefore, a buffering latency occurs in PoC service due to a play-out delay which means an asynchronous play of audio & video contents. Those problems make a user to be difficult to find the information they want when the media contents are played-out. In this paper, the following techniques and methods were proposed and their performance and superiority were verified through testing: cross-over dual reception buffering technique, advance partition multi-reception buffering technique, and on-demand multi-reception buffering technique, which are designed for effective picking up of information in media content being transmitted in short amount of time using RTSP when a user searches for media, as well as for reduction in playback delay; and same-priority packetization transmission method and priority-based packetization transmission method, which are media data packetization methods for transmission. From the simulation of functional evaluation, we could find that the proposed multiple receiving buffering and packetizing methods are superior, with respect to the media retrieval inclination, to the existing single receiving buffering method by 6-9 points from the viewpoint of effectiveness and excellence. Among them, especially, on-demand multiple receiving buffering technology with same-priority packetization transmission method is able to manage the media search inclination promptly to the requests of users by showing superiority of 3-24 points above compared to other combination methods. In addition, users could find the information they want much quickly since large amount of informations are received in a focused media retrieval period within a short time.

• Journal of the Korea Computer Industry Society
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• v.2 no.4
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• pp.455-470
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• 2001

This paper is focused on the traffic characterization for the efficient transmission of the VBR video source in the ATM network. For the traffic characterization, low traffic monitoring technique is applied and the dynamic VBR characterization method is suggested to satisfy the delay requirement. The dynamic VBR method uses the token bucket algorithm buffering though Cumulative Constraint Function. According to the Cumulative Constraint Function, the packet initially started transferring at the peak rate and the token bucket provided proper amount of buffer for traffic after a certain period of monitoring. It also reduced the network resource bandwidth through renewal of the cumulative frame and changed the rate from the previous frame information. It requires only small amount of monitoring and causes little overhead. In addition, it lowered the complexity of Deterministic Constraint Function to 0(n) and mapped the token rate and token depth to the token bucket. This study shows less network resource consumed than the previous method, comparing and analyzing the result of simulations.

• Journal of IKEEE
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• v.22 no.2
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• pp.289-301
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• 2018

IIoT, the IoT technology applied to the industrial field, is being used as a monitoring technology for increasing in production rate and safety of workers. However, monitoring through the construction of IIoT network using Ethernet and RS485 in production lines where dozens to hundreds of machine tools are manufacturing components, have difficulties of infrastructure cost and network flexibility and fluidity. Therefore, in this paper, using IEEE 802.15.4 standard WSN device to construct a Ad-Hoc WSN in the production line. In addition, the transmission period and order of the sensor nodes are set by applying the EtherCAT communication method in which the payload frames are shared by all the sensor nodes. From this, we have overcome the problem of reliability decline and real-time issue due to the packet collision of wireless network and confirmed that it is a wireless network routing method that can be used in the actual industrial field.

• Journal of KIISE:Information Networking
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• v.35 no.1
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• pp.46-55
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• 2008

As a mobility support for IP have studied, Internet Engineering Task Force(IETF) standardized the Mobile IPv6(MIPv6) protocol. When a mobile node moves between subnets, MIPv6 maintains connectivity to network and supports seamless communication, and these processes are called a Handover. Whenever the mobile node moves between subnets, the Handover is performed. The mobile node can not communicate during the Handover. This period is Galled Handover latency. To reduce this latency, mipshop working group standardizes Fast Handovers for Mobile IPv6(FMIPv6), but latency which the mobile node registers its new care-of address to a home agent and a correspondent node is still long. To solve this problem, we propose a scheme that the mobile node registers the new care-of address to the home agent and initiates Return Routability procedure in advance during layer 2 handover, based on FMIPv6 and IEEE 802.11. We analyze MIPv6, FMIPv6 and the proposed scheme in term of packet transmission cost during the Handover. Compared to MIPv6 the proposed scheme gains 79% improvement, while it gains 31% improvement compared to FMIPv6.