• The Transactions of the Korea Information Processing Society
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• v.6 no.3
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• pp.699-709
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• 1999

This paper is aimed at defining items of analysis using SNMP MIB-II for the purpose of managing the Internet based network running on TCP/IP protocol, and then utilize these items, in conjunction with various Web technology and JAVA to design and implement a Web based interface of a management system to analyze the performance and status of the Internet. Among the required items in order to accomplish this task are utilization, interface packet transmission rate, I/O traffic ratio, and so on. Based on these items, the actual analysis is carried out by the Web interface according to the type of analysis. For instance, the interface executes the function of real-time analysis, collection processing, elementary analysis and detailed analysis. The demand of the user is fed into the Web interface which carried out a real-time analysis with the client system which in turn will eventually produce the results of the analysis. In order words, the interface acts as a mediator server for the analysis system. Furthermore, a protocol for exchange of data and messages between the server and the analysis system, the MATP protocol, was also designed. Finally, the results obtained through the system presented in this paper were displayed on screen according to the type of analysis. The system realized in this paper uses We technology and is independent of platform and allows the user to determine the performance of Internet at his/her own host according to the selected items of analysis.

• KIPS Transactions on Computer and Communication Systems
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• v.2 no.9
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• pp.381-392
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• 2013

During the last decade, the multitude of advances attained in terminal computers, along with the introduction of mobile hand-held devices, and the deployment of high speed networks have led to a recent surge of interest in Quality of Service (QoS) for video applications. The main difficulty is that mobile devices experience disparate channel conditions, which results in different rates and patterns of packet loss. One way of making more efficient use of network resources in video services over wireless channels with heterogeneous characteristics to heterogeneous types of mobile device is to use a scalable video coding (SVC). An SVC divides a video stream into a base layer and a single or multiple enhancement layers. We have to ensure that the base layer of the video stream is successfully received and decoded by the subscribers, because it provides the basis for the subsequent decoding of the enhancement layer(s). At the same time, a system should be designed so that the enhancement layer(s) can be successfully decoded by as many users as possible, so that the average QoS is as high as possible. To accommodate these characteristics, we propose an efficient transmission scheme which incorporates SVC-aware dual-channel repetition to improve the perceived quality of services. We repeat the base-layer data over two channels, with different characteristics, to exploit transmission diversity. On the other hand, those channels are utilized to increase the data rate of enhancement layer data. This arrangement reduces service disruption under poor channel conditions by protecting the data that is more important to video decoding. Simulations show that our scheme safeguards the important packets and improves perceived video quality at a mobile device.

• The Journal of the Acoustical Society of Korea
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• v.37 no.1
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• pp.31-38
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• 2018

Based on the measured data in the south western sea of Jeju Island during the SAVEX15(Shallow Water Acoustic Variability EXperiment 2015), the effect of internal waves on the PPD (Predictive Probability of Detection) of a sonar system was analyzed. The southern west sea of Jeju Island has complex flows due to internal waves and USC (Underwater Sound Channel). In this paper, sonar performance is predicted by probabilistic approach. The LFM (Linear Frequency Modulation) and MLS (Maximum Length Sequence) signals of 11 kHz - 31 kHz band of SAVEX15 data were processed to calculate the TL (Transmission Loss) and NL (Noise Level) at a distance of approximately 2.8 km from the source and the receiver. The PDF (Probability Density Function) of TL and NL is convoluted to obtain the PDF of the SE (Signal Excess) and the PPD according to the depth of the source and receiver is calculated. Analysis of the changes in the PPD over time when there are internal waves such as soliton packet and internal tide has confirmed that the PPD value is affected by different aspects.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.5
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• pp.267-273
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• 2012

The short-range communication module was developed in this paper which is suitable for the point-to-multipoint circumstances of 40 terminals communicating frequently in about 50 meters. The media access control layer of this communication module using asynchronous time-division multiplexing provides a fast and robust performance even in the worst case of simultaneous transmission events, and low packet error rate was measured a in LOS (Line-of-sight) circumstance by adding the function of acknowledge response to media access control layer. The difference test was carried out in order to measure the performance of point-to-multipoint communication. Two communication modules are respectively measured and graphed in 10 branches to 40 branches. The communication module developed in this paper showed a faster performance than the commercial Zigbee module in the specific case presented in this paper. Especially, in over 20 branches showed wide differences of the transmission speed. This results is caused by more network overhead of Zigbee whose wider applications needs the network layer and applicaiton layer besides media access control layer. Also, the asynchronous time-division multiplexing proposed in this paper are more suitable than CSMA-CA of Zigbee module when a lot of module ought to be frequently communicated in small area.

• Journal of the Korea Convergence Society
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• v.7 no.4
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• pp.39-44
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• 2016

Due to the dazzling development of IT in this IT-oriented era, information delivering technology among objects, between objects and humans, and among humans has been actively performed. As information delivery technology has been actively performed, IoT became closely related to our daily lives and ubiquitous at any time and place. Therefore, IoT has become a part of our daily lives. CoAp, a web-based protocol, is mostly used in IoT environment. CoAp protocol is mostly used in the network where transmission speed is low along with the huge loss. Therefore, it is mostly used in IoT environment. However, there is a weakness on IoT that it is weak in security. If security issue occurs in IoT environment, there is a possibility for secret information of individuals or companies to be disclosed. If attackers infect the targeted device, and infected device accesses to the wireless frequently used in public areas, the relevant device sends arp spoofing to other devices in the network. Afterward, infected devices receive the packet sent by other devices in the network after occupying the packet flow in the internal network and send them to the designated hacker's server. This study suggests counter-attacks on this issues and a method of coping with them.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2000.10a
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• pp.284-287
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• 2000

Recently, developing medical devices have a tendency becoming the module for satisfying user's mutual complex needs. Because the most effective method for the observation of patients condition a diagnosis and a treatment is collecting data from various devices and controling operation following it. Module tendency is more popular due to manage easily totally many individual systems. This study implemented communication protocol to control by one control system connecting modular medical devices. Implemented system consist of one master module controlling all module and managing communication and many Slave modules. Communication between each modules introduced SPI(Serial Peripheral Interface) among many synchronous serial communication methods for the exact transmission and receipt of data. All communication executes by packet format. This can detect error. And, this protocol introduced PNP(Plug And Play) function that auto-detect connecting or removing module during running. This protocol exactly transmitted and received in faster speed more than 1Mbps. And in practical application to the ventilator this confirmed to give and take real-time data. And various functions by th central control system is implemented in this protocol.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.5B
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• pp.449-455
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• 2006

Due to advance in multimedia applications and integrated Internet services, the optical access networks have been actively studied. In particular, Ethernet passive optical network (EPON) has received much attention due to high bandwidth provision with low cost. In EPON system, the data transmission is carried out in two directions: downstream (from OLT to ONU) and upstream (from ONU to OLT). The downstream data is broadcasted to every ONUs, while the upstream data is point-to-point transmitted between each ONU and OLT, where the uplink is shared by all ONUs in the form of TDMA. The bandwidth allocation algorithm is required to efficiently manage the bandwidth on the uplink. The limited algorithm was proposed to enhance the capability of dynamic bandwidth allocation. In this paper, we propose the adaptive limited algorithm to enhance the shortcomings of limited algorithm. The adaptive limited algorithm enhances the dynamics on bandwidth allocation, and at the same time controls the fairness on packet delay. Through the computer simulations, it is shown that the adaptive limited algorithm achieves high dynamic on bandwidth allocation, maintains a good fairness on packet delay between ONUs, and keeps the fairness on the bandwidth on the demand basis.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.1A
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• pp.96-103
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• 2000

The voice traffic is one of the most important real-time objects in WATM(Wireless Asynchronous Transfer Mode) networks. In this paper, we propose a new MAC(Medium Access'Control) protocol for the efficienttransmission of voice traffic over WATM networks in the multimedia environment and compare the performanceto existing similar protocols. The new protocol separates the reservation slot period for voice and that for data toguarantee some level of QoS(Quality of Service) in voice traffic. This is denoted by a slot assignment functiondepending on the frame size. According to the characteristics of voice traffic which is repeatedly in silent states,the protocol allocates voice reservation request slots dynamically with respect to the number of silent(off state)voice sources and also sends the first block of talkspurt restarted after silent period with a reservation requestslot to reduce the access delay.The simulation results show that the proposed protocol has better performance than Slotted Aloha in bandwidthefficiency, and can serve a certain level of QoS by the given slot assignment function even when the number ofvoice terminals varies dynamically. This means we can observe that the new MAC protocol is much better thanthe NC-PRMA(None Collision-Packet Reservation Multiple Access) protocol.

• 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:Computing Practices and Letters
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• v.15 no.2
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• pp.114-124
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• 2009

In this paper, we propose a practical TCP friendly rate control scheme that considers the minimum channel bandwidth of the network when transporting SVC (scalable video coding) video over IP netowrks such as Internet. RTP and RTCP is mainly designed for use with UDP (User Datagram Protocol) for real-time video transport over the Internet. TCP-friendly rate control was proposed to satisfy the demands of multimedia applications while being reasonably fair when competing for bandwidth with conventional TCP applications. However the rate control model of the conventional TCP-friendly rate control scheme does not consider the minimum channel bandwidth of the network. Thus the estimated channel bandwidth by the conventional rate control model might be quite different from the real channel bandwidth when the packet loss ratio of the network is very large. In this paper, we propose a modified TCP-friendly rate control scheme that considers the minimum channel bandwidth of the network. Based on the modified TCP-friendly rate control, we assign the minimum channel bandwidth to the base layer bitstream of SVC video, and remaining available bandwidth is allocated to the enhancement layer of SVC video for the TCP friendly scalable video transmission. It is shown by simulations that the modified TCP-friendly rate control scheme can be effectively used for a wider range of controlled bit rates depending on the packet loss ratio than the conventional TCP-friendly control scheme. Furthermore, the effectiveness of the proposed scheme in terms of objective video quality is proved by comparing PSNR performance with the conventional scheme.