• Journal of the Institute of Electronics Engineers of Korea TC
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• v.48 no.5
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• pp.16-24
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• 2011

The IEEE 802.11 wireless LAN (Local Area Network) is widely used for wireless access due to its easy deployment and low cost. Multicast in wireless LANs is very useful for transmitting data to multiple receivers compared to unicast to each receiver. In the IEEE 802.11 wireless LAN, multicast transmissions are unreliable since multicast data packets are transmitted without any feedback from receivers. Recently, various protocols have been proposed to enhance the reliability of multicast transmissions. They still have serious problems in reliability and efficiency due to the excessive control overhead by the use of a large number of control packets in the error recovery process, and due to a large number of retransmissions to satisfy all receivers. In this paper, we propose an effective scheme called PTRM(Proactive Transmission based Reliable Multicast). The proposed scheme uses a block erasure code to generate parity packets and to reduce the impact of independent packet error among receivers. After generating parity packets, the PTRM transmits data packets as many as receivers need to recover error, and then requests feedback from them. The simulation results show that the proposed scheme provides reliable multicast while minimizing the feedback overhead.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.7
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• pp.31-40
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• 2004

In this paper, we analyze mathematically the performance of the gated service scheduling in the Interleaved Polling with Adaptive Cycle Time(IPACT) was proposed to control upstream traffic for Gigabit Ethernet-PONs. In the analysis, we model EPON MAC protocol as a polling system and use mean value analysis. We divide arrival rate λ into three regions and analyze each region accordingly In the first region in which λ value is very small, there are very few ONUs' data to be transmitted. In the second region in which λ has reasonably large value, ONUs have enough data for continuous transmission. In the third region, ONUs' buffers are always saturated with data since λ value is very large. We obtain average packet delay, average Queue size, average cycle time of the gated service. We compare analysis results with simulation to verify the accuracy of the mathematical analysis. Simulation requires much time and effort to evaluate the performance of EPONs. On the other hand, mathematical analysis can be widely used in the design of EPON systems because system designers can obtain various performance results rapidly. We can design appropriate EPON systems for varioustraffic property by adjusting control parameters.

• 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 Society of Computer and Information
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• v.14 no.12
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• pp.157-167
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• 2009

Recently, the leakage of confidential information and personal information is taking place on the Internet more frequently than ever before. Most of such online security incidents are caused by attacks on vulnerabilities in web applications developed carelessly. It is impossible to detect an attack on a web application with existing firewalls and intrusion detection systems. Besides, the signature-based detection has a limited capability in detecting new threats. Therefore, many researches concerning the method to detect attacks on web applications are employing anomaly-based detection methods that use the web traffic analysis. Much research about anomaly-based detection through the normal web traffic analysis focus on three problems - the method to accurately analyze given web traffic, system performance needed for inspecting application payload of the packet required to detect attack on application layer and the maintenance and costs of lots of network security devices newly installed. The UTM(Unified Threat Management) system, a suggested solution for the problem, had a goal of resolving all of security problems at a time, but is not being widely used due to its low efficiency and high costs. Besides, the web filter that performs one of the functions of the UTM system, can not adequately detect a variety of recent sophisticated attacks on web applications. In order to resolve such problems, studies are being carried out on the web application firewall to introduce a new network security system. As such studies focus on speeding up packet processing by depending on high-priced hardware, the costs to deploy a web application firewall are rising. In addition, the current anomaly-based detection technologies that do not take into account the characteristics of the web application is causing lots of false positives and false negatives. In order to reduce false positives and false negatives, this study suggested a realtime anomaly detection method based on the analysis of the length of parameter value contained in the web client's request. In addition, it designed and suggested a WAF(Web Application Firewall) that can be applied to a low-priced system or legacy system to process application data without the help of an exclusive hardware. Furthermore, it suggested a method to resolve sluggish performance attributed to copying packets into application area for application data processing, Consequently, this study provide to deploy an effective web application firewall at a low cost at the moment when the deployment of an additional security system was considered burdened due to lots of network security systems currently used.

• Journal of KIISE:Information Networking
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• v.30 no.2
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• pp.282-292
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• 2003

This paper proposes an efficient data structure for forwarding IPv4 and IPv6 packets at the gigabit speed in backbone routers. The LPM(Longest Prefix Matching) search becomes a bottleneck of routers' performance since the LPM complexity grows in proportion to the forwarding table size and the address length. To speed up the forwarding process, this paper introduces a data structure named BMT(Bit-Map Tie) to minimize the frequent main memory accesses. All the necessary search computations in BMT are done over a small index table stored at cache. To build the small index table from the tie representation of the forwarding table, BMT represents a link pointer to the child node and a node pointer to the corresponding entry in the forwarding table with one bit respectively. To improve the poor performance of the conventional tries when their height becomes higher due to the increase of the address length, BMT adopts a binary search algorithm for determining the appropriate level of tries to start. The simulation experiments show that BMT compacts the IPv4 backbone routers' forwarding table into a small one less than 512-kbyte and achieves the average speed of 250ns/packet on Pentium II processors, which is almost the same performance as the fastest conventional lookup algorithms.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37B no.9
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• pp.806-813
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• 2012

In a cloud computing environment, various solutions were introduced to provide the service to users such as Infrastructure as a Service (IaaS), Platform as a Service (PaaS), Software as a Service (SaaS) and Desktop as a Service (DaaS). Nowadays, Mobile as a Service (MaaS) to provide the mobility in a cloud environment. In other words, users must have access to data and applications even when they are moving. Thus, to support the mobility to a mobile Thin-Client is the key factor. Related works to support the mobility for mobile devices were Mobile IPv6 and Proxy Mobile IPv6 which showed performance drawbacks such as packet loss during hand-over which could be very critical when collaborating with cloud computing environment. The proposed model in this paper deploys middleware and replica servers to support the data transmission among cloud and PMIPv6 domain. It supports efficient mobility during high-speed movement as well as high-density of mobile nodes in local mobility anchor. In this paper, through performance evaluation, the proposed scheme shows the cost comparison between previous PMIPv6 and verifies its significant efficiency.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12A
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• pp.2027-2035
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• 2001

In this paper, we present an audio stream delivery using the AMR (Adaptive Multi-Rate) coder that was adopted by ETSI and 3GPP as a standard vocoder for next generation IMT-2000 service in which includes combined sender (FEC) and receiver reconstruction technique in the Internet. By use of the media-specific FEC scheme, the possibility to recover lost packets can be much increased due to the addition of repair data to a main data stream, by which the contents of lost packets can be recovered. The AMR codec is based on the code-excited linear predictive (CELP) coding model. So we use a frame erasure concealment for CELP-based coders. The proposed scheme is evaluated with ITU-T G.729 (CS-ACELP) coder and AMR - 12.2 kbit/s through the SNR (Signal to Noise Ratio) and the MOS (Mean Opinion Score) test. The proposed scheme provides 1.1 higher in Mean Opinion Score value and 5.61 dB higher than AMR - 12.2 kbit/s in terms of SNR in 10% packet loss, and maintains the communicab1e quality speech at frame erasure rates lop to 20%.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.10
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• pp.2181-2190
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• 1997

The synchronous stream cipher has the problem of synchronization loss by cycle slip. Synchronization loss make the state which sender and receiver can't communicate and it may make the receiving system disordered. To lessen the risk, we usually use a continuous resynchronization which achieve resynchronization at fixed timesteps by inserting synchronization pattern and session key. While we can get effectively resynchronizationby continuous resynchronization, there are some problems. In this paper, we proposed an adaptive resynchronization algorithm for cipher system using LAPB protocol. It is able to solve the problem of the continunous resynchronization.The proposed adaptive algorithm make resynchronization only in the case that the resynchronization is occurred by analyzing the address field of LAPB. It measure the receiving rate of the address field in the decesion duration. If the receiving rate is smaller than threshold value, it make resynchronization or not. By using adaptively resynchronization, it solves the problems of continunous resynchronization. When the proposed adaptive algorithm is applied to the synchronous stream cipher system which is used in X.25 packet network, it reduced the time for resynchronization by ten times. It means that 11.3% of total data for transmit is compressed.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.15 no.12 s.91
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• pp.1152-1160
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• 2004

OFDM technique was adopted as the standard of IEEE 802.1 la and it has been widely used for wireless LAN, European DVB/DAB system, Korean DMB system. In the standard of IEEE 802.11a the data packet is composed of two parts, preamble and data. Preamble is composed of short pilots and long pilots, which are used for synchronization and estimation of frequency offset and channel. We can also compensate phase noise effect in the transceiver by using above pilots. The phase noise is more complicate than frequency offset and seriously affects system performance. In this paper, we newly propose CPE and ICI simultaneous compensation method to compensate phase noise generated by transceiver oscillator and compare with previous studies. As results, phase noise effect can be significantly compensated by CPE cancellation method, PNS algorithm and our proposed CPE and ICI compensation method. Especially, the proposed CPE and ICI compensation method can achieve the best BER performance compared with original OFDM, CPE cancellation method and PNS algorithm.

• Journal of the Institute of Electronics and Information Engineers
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• v.53 no.4
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• pp.3-11
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• 2016

In the environment of Internet of Things (IoT), IoT devices are limited by physical components such as power supply and memory, and also limited to their network performance in bandwidth, wireless channel, throughput, payload, etc. Despite these limitations, resources of IoT devices are shared with other IoT devices. Especially, remote management of the information of devices and patients are very important for the IoT healthcare service, moreover, providing the interoperability between the healthcare device and healthcare platform is essential. To meet these requirements, format of the message and the expressions for the data information and data transmission need to comply with suitable international standards for the IoT environment. However, the ISO/IEEE 11073 PHD (Personal Healthcare Device) standards, the existing international standards for the transmission of health informatics, does not consider the IoT environment, and therefore it is difficult to be applied for the IoT healthcare service. For this matter, we have designed and implemented the IoT healthcare system by applying the oneM2M, standards for the Internet of Things, and ISO/IEEE 11073 DIM (Domain Information Model), standards for the transmission of health informatics. For the implementation, the OM2M platform, which is based on the oneM2M standards, has been used. To evaluate the efficiency of transfer syntaxes between the healthcare device and OM2M platform, we have implemented comparative performance evaluation between HTTP and CoAP, and also between XML and JSON by comparing the packet size and number of packets in one transaction.