• Journal of KIISE:Computing Practices and Letters
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• v.9 no.1
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• pp.95-99
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• 2003

In addition to wired communication technology, wireless communication technology has had communication revolution nowadays. Bluethooth technology carries out data/voice communication within pico-net. Nowadays the various services are supported by access network connected to public network. This paper presents implementation of bluetooth protocol analyser which simulates bluetooth protocol. MS window98 and visual C are used for development environment and application program is operated over the firmware loaded on the bluetooth device connected to the PC through UART which of the maximum transmission rate is 115kbps because transmission rate less than 20kbps affects rarely the performance. The performance analysis on the propose system is carried out as simulating the signalling information for the voice test and the traffics between two bluetooth systems for file transfer. The throughput analysis for file transfer service and call processing capacity for voice service are considered as performance analysis parameters. File access time is very important parameter and throughput is 13 kbps in case breakpoint time to file access is 0.04sec. Also call processing time is about 16.6ms in case of communication with the headset. The performance analysis of simulation results satisfies with bluetooth device development.

• The KIPS Transactions:PartD
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• v.13D no.7 s.110
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• pp.901-908
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• 2006

With the development of wireless communication techniques and mobile environment we are able to transmit data between mobile systems without restriction of time and space. Recently, researches on the data communication between mobile systems have focused on a small amount of sending out or receiving data and data synchronization at a fixed server and mobile clients in mobile environment. However, two more servers should be able to move mutual independently, information is shared with other systems, and data is synchronized in the special environment like a battlefield situation. Therefore, we propose a data synchronization method between systems moving mutual independently in mobile environment. The proposed method is an optimization solution to data propagation path between servers that considers limited bandwidth and process of data for disconnection communication. In addition, we propose a data reduction method that considers importance and sharing of information in order to reduce data transmission between huge servers. We verified the accuracy of data after accomplishing our data synchronization method by applying it in the real world environment. Additionally, we showed that our method could accomplish data synchronization normally within an allowance tolerance when we considered data propagating delay time by server extension.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.14 no.12
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• pp.1239-1247
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• 2003

OFDM(orthogonal frequency division multiplexing) communications system is very attractive for the high data rate transmission in the frequency selective lading channel. Since OFDM has high PAPR(peak-to-average power ratio), OFDM signal may be distorted by the nonlinear HPA(high power amplifier). In this paper, we propose the DSI(dummy sequence insertion) method for OFDM communication system. Some sub-carriers are inserted for PAPR reduction. They carry the specified dummy data sequence which are used for only PAPR reduction and do not work as side information like the conventional PTS(partial transmit sequence) or SLM(selected mapping) method. We use the complementary sequence and the combination of the correlation sequence as the dummy sequence. Flipping technique is used for the DSI method to get the effective PAPR reduction. It is important that BER of the proposed method is independent of the damage of the dummy data sequence. And DSI method has better spectral efficiency than the conventional block coding. On the other hand, threshold PAPR method is applied to cut down the processing time. However, this DSI method is not better than the conventional PTS method in the respect of the PAPR reduction performance. The DSI method includes the threshold PAPR lower than the PAPR of the OFDM signal, reduces the processing time and improves the BER performance.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.15 no.6
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• pp.187-193
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• 2015

The ICT healing platform sets a couple of goals including preventing chronic diseases and sending out early disease warnings based on personal information such as bio-signals and life habits. The 2-step open system(TOS) had a relay designed between the healing platform and the storage of personal health data. It also took into account a publish/subscribe(pub/sub) service based on large-scale connections to transmit(monitor) the data processing process in real time. In the early design of TOS pub/sub, however, the same buffers were allocated regardless of connection idling and type of message in order to encode connection messages into a deflate algorithm. Proposed in this study, the dynamic buffer allocation was performed as follows: the message transmission type of each connection was first put to queuing; each queue was extracted for its feature, computed, and converted into vector through tf-idf, then being entered into a k-means cluster and forming a cluster; connections categorized under a certain cluster would re-allocate the resources according to the resource table of the cluster; the centroid of each cluster would select a queuing pattern to represent the cluster in advance and present it as a resource reference table(encoding efficiency by the buffer sizes); and the proposed design would perform trade-off between the calculation resources and the network bandwidth for cluster and feature calculations to efficiently allocate the encoding buffer resources of TOS to the network connections, thus contributing to the increased tps(number of real-time data processing and monitoring connections per unit hour) of TOS.

• The Transactions of the Korea Information Processing Society
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• v.5 no.12
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• pp.3047-3062
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• 1998

This paper deseribes the desihn and the implementation of the Realtime Transport Protocol(RTP)/ Rdaltime Control Protocol(RTCP) (RFC 1889,1890) that is used to transmit the audio/video data to any destination and to feedback the Quality of Service (QoS) information of the received media data to the sender, in the teleconferencing systems proposed by ITU-T. These protocols are implemented with multi thead technique and run on top of UDP/IP-Multicast through the socket interface as the underlying protocol. The upper layer is impelmented such that in can be accessed by the H245 comference control protocol. The RTP packetizes the digitized audio/video data from the encoder info a fixed format, and multieast to the participants. The RTCP monitors RTP packets and extracts the QoS values from it such as round-trip delay, jiter and packet loss to form RTCP packets and non periokically sends them to the sender site. In this Paper, we also descritx the study of measurement and analysis for QoS factors that observed on performing teleconferencing system over Internet. The results from this experiment is indicate that RTT and Jitter value are acceptable even entwork load is high. However, it appears that packet loss rate is high in daytime and most losses periods have length one or two.

• The Transactions of the Korea Information Processing Society
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• v.4 no.9
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• pp.2378-2390
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• 1997

We, in this paper, have implemented E1 HDSL(high-bit-rate digital subscriber line) function over an ATM switching system. The maximum loop lengths for subscriber service and cell loss rates to meet the bit error rate of $10^{-7}$ at transmission of 2B1Q HDSL data of E1 rate over existing telephone copper wires in the presense of the significant impairments such as crosstalk, impulse noise, power line noise and longitudinal over the CSAs environment consisting of 26 gauge and 24 gauge unloaded copper telephone lines has assessed. We have confirmed the typical media services function such as video on demand(VOD) service for MPEG-1, image conference service and high-speed Internet access service over ATM switching system.

• The Transactions of the Korea Information Processing Society
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• v.7 no.12
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• pp.3896-3909
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• 2000

The MPEG Video Service of hilving il property of continuity and large cilPilcity needs lilrge network capilcity. Because most of users have a heterogeneous network environment, it's not efficient way that all users have same size of video data to provide them with real time video service. Therefore, for the pUl1Xlse of an efficient and appropriate utilization of network resources, it requires to develop and deploy a new scalable transmission technique in consideration of respective network environment and individual clients computing power. The purpose of this paper is to develop a technology that can adjust the amount of dilta transmitted as an M1'EG video stream according to its gi yen communication bandwidth, and a technique that can reflect dynamic bilndwidth while playing a video stream. For this purpose, we propose a TFS (Temporal-Fidelity Scaling) technique that splits the MPEG video stream into various substream according to picture type or resolution. Those methods proposed her can filcilitilte an effective use of network resources, and provide multimedia MPEG video services in real- time with respect to individual client computing environment

• The Transactions of the Korea Information Processing Society
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• v.3 no.4
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• pp.938-946
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• 1996

Hierarchical subband codec has been widely promoted in the field of data compression/decompression because of their simplicity and modular nature. Over the past years, the study has received great attention to the perfect reconstruction (PR)system which perfectly recovers the original input signal at the reconstructed output. However, in the actual subband codec system, the signals that passed through the analysis filter bank are quantized before transmission to the receiver side and reconstructed by the synthesis filter bank. Thus the PR system is impossible and the quantization effects must be carefully considered in the system design such that the system recovers the reconstructed output as possible to the the original input signal with minimum quantization error.In this paper, we propose an optimum hierarchical subband codec structure in the presence of quantizer. The optimality criteria of the code is given to the deign of the hierarchical analysis/synthesis subband filter bank and the quantizer that minimize then output mean square error due to the quantizer in the codec. Specific opti-mum design esamples are shown with level-1, level-2 hierarchical structure. The optimal designs are verified by computer simulation.

• Journal of Multimedia Information System
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• v.7 no.3
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• pp.221-230
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• 2020

Used optimization techniques as solution for mobile network have been implemented as a reference systems for various applications against fading and signals perturbation, in addition each transition to 5th generation telecommunication standards require a deep studies in order to park an applied instantaneous process. The paper describes a preliminary planning and a careful preparation to update both subscriber radio access network as well as data transmission network this approach conducts to make network resource updates invisible for customers and with minimal costs for mobile operators basically in terms of delay. In addition, network operators transit to mobile networks, multimedia services efficient delivery are considered the challenging application and the most promising for mobile network operators today, this work conduct to optimize video consumption of mobile users which are exponentially increasing. The interference is a complex phenomenon in mobile radio telecommunication system, and a mobile phone can be a source of interference to another one. Actual advances in technology necessitate the need for the complicated software solution that can take several unexpected phenomena in consideration to rise to a level higher than ever. The capability needs today require the use of Drive test which is used to take the performance of network in the field by using a special software called TEMS investigation, it have been implemented as standalone systems for various applications. The paper focuses on considering as the best technical for optimization of mobile networks, analysis and processing of signal, a Drive Test is the method used to take the performance of network in the field by using a special software called TEMS investigation. Most used in the world, this software is reputed to detect and analyze many problems of mobile network between the mobile phone and the transmitter: BTS in case of GSM and Node B for UMTS. An example of that is interference in radio communication. It exists permanently and it degrades considerably the quality of received signal when it exceeds certain levels.

• Journal of Korea Multimedia Society
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• v.12 no.2
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• pp.257-271
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• 2009

In this paper, we propose a protocol compression scheme for enhancing the performance of call processing of PTT(Push-to-Talk) which is one of the important services in IMS(IP Multimedia Subsystem), a next generation integrated wired/wireless packet communication network. To service the PTT on an IMS network, it should use the same call setup procedure as legacy Mobile and TRS(Trunked Radio System) networks and have a fast call setup time and enough communication bandwidth because a number of terminals should be able to exchange same data in real time. The proposed A+SigComp scheme reduces the initial call setup delay of SIP by about 10%, which is used by PTT service for call setup. In addition, the A+ROHC scheme is proposed to compress the header of RTP packets transferred during PTT voice transmission and, as a result, about 5% of increase in communication efficiency is observed.