• Journal of the Korea Society of Computer and Information
• /
• v.17 no.7
• /
• pp.97-106
• /
• 2012

Management of biosignal data in mobile devices causes many problems in real-time transmission of large volume of multimedia data or storage devices. Therefore, this research paper intends to suggest an m-Health system, a clinical data processing system using mobile in order to provide quick medical service. This system deployed health system on IP network, compounded outputs from many bio sensing in remote sites and performed integrated data processing electronically on various bio sensors. The m-health system measures and monitors various biosignals and sends them to data servers of remote hospitals. It is an Android-based mobile application which patients and their family and medical staff can use anywhere anytime. Medical staff access patient data from hospital data servers and provide feedback on medical diagnosis and prescription to patients or users. Video stream for patient monitoring uses a scalable transcoding technique to decides data size appropriate for network traffic and sends video stream, remarkably reducing loads of mobile systems and networks.

• Proceedings of the Korean Society of Computer Information Conference
• /
• 2010.07a
• /
• pp.173-179
• /
• 2010

Currently, the communication systems are progressing two ways as the wireless and multimedia and these need big transmission capacity then before. In these circumstance, communication services existed as two different service forms which have different rates and characteristics. For example Voice/Video Services accept some errors but transmit on realtime, but Date Services don't need to transmit on realtime but have to retransmit if these have only one bit error. In Voice/Date Integrated traffics, it has big throughput that realtime voice/video data which could have some errors if integrated traffic is increased rapidly have transmission priority, then Data traffics which delay is accepted is sent after that. In this paper, I introduce the calculation method for various traffic when voice/data mixed traffics is transmitted to asynchronous unslotted ALOHA CDMA system proposed and the result is presented. And We can easily theoretical analysis for the system traffic and changing traffic using proposed solution in this paper.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.44 no.4
• /
• pp.17-27
• /
• 2007

The enhanced distributed channel access (EDCA) of IEEE 802.11e has been standardized for supporting Quality of Service (QoS) in wireless LANs. In the EDCA, support of QoS can be achieved statistically by reducing the probability of medium access for lower priority traffics. In other words, it provides statistical channel access rather than deterministically prioritized access to high priority traffic. Therefore, lower priority traffics affect the performance of higher priority traffics. Consequently, at the high loads, the EDCA does not guarantee the QoS of multimedia applications such as voice and video even though it provides higher priority. In this paper, we propose a simple and effective scheme, called deterministic priority channel access (DPCA), for improving the QoS performance of the EDCA mechanism. In order to provide guaranteed priority channel access to multimedia applications, the proposed scheme uses a busy tone for limiting the transmissions of lower priority traffics when higher priority traffic has data packets to send. Performance of the proposed scheme is investigated by numerical analysis and simulation. Our results show that the proposed scheme outperforms the EDCA in terms of throughput, delay, jitter, and drop under a wide range of contention levels.

• The Journal of The Korea Institute of Intelligent Transport Systems
• /
• v.10 no.1
• /
• pp.76-83
• /
• 2011

The IEEE WAVE-based communication systems do not provide handover services since most of the application layer messages of a small amount containing text data that are related to safe driving. Multimedia data service such as web pages and CCTV video clips, however, require a seamless handover for continuation of a session via multiple RSUs. In this paper, we propose a new proactive handover protocol based on IEEE WAVE. According to the proposed handover protocol, the OBU notifies the old RSU of its departure from the coverage such that the old RSU forwards to the new RSU the data heading towards the OBU to be cached for the further delivery upon its entry into the new RSU's coverage. The simulation results are presented which shows the performance of the proposed protocol in terms of throughput, delivery ratio and handover delay.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.41 no.5
• /
• pp.117-124
• /
• 2004

Various multimedia systems have been developed and their application areas widely proliferate. Thus, the interoperability is getting important among various networks and devices. The video transcoding is a technology to solve this interoperability problem among various coding standards. Transcoding can be defined as the conversion of one compressed coded data to another. In this paper, MPEG-2 to MPEG-4 transcoder in the spatial domain is compared with that in the DCT domain. The transcoder is very useful when a video sequence that is originally encoded for digital TV, DVD or satellite broadcasting is served in mobile environment. In order to compare two transcoders, all modules except motion compensation and down sampling are implemented identically. In addition, both transcoders do not search for motion vector. Instead, the decoded information is reused to the encoder. The experimental results show that the transcoder in the spatial domain is usually better than that in the DCT domain with respect to PSNR (Peak Signal-to-Noise Ratio), bitrate and execution time.

• Journal of KIISE:Software and Applications
• /
• v.31 no.10
• /
• pp.1320-1331
• /
• 2004

A video adaptation system translates the source video stream into appropriate video stream while satisfying the network and client constraints and maximizing the video quality as much as possible. This paper proposes a semi-dynamic video adaptation scheme, in which several intermediate video streams and the information for the measuring of video quality are generated statically. The intermediate video streams are generated by reducing the resolution of the video stream by a power of two several times, and they are stored as the intermediate video streams on the video server. The statically generated information for the input video stream consists of the degrees of smoothness for each frame rate and the degree of frame definition for each pixel bit rate. It helps to dynamically generate the target video stream according to the client's QoS at run-time as quickly as possible. Experimental result shows that the proposed adaptation scheme can generate the target video stream about thirty times faster while keeping the quality degradation as less than 2% comparing to the target video stream that is totally dynamically generated, although the extra storages for the intermediate video streams are required.

• Journal of the Institute of Convergence Signal Processing
• /
• v.2 no.2
• /
• pp.62-67
• /
• 2001

In the recent mobile communication systems, the performance of Turbo Code using the error correction coding depends on the interleaver influencing the free distance determination and the recursive decoding algorithms that is executed in the turbo decoder. However, performance depends on the interleaver depth that need a large time delay over the reception process. Moreover, Turbo Code has been known as the robust ending method with the confidence over the fading channel. The International Telecommunication Union(ITU) has recently adopted as the standardization of the channel coding over the third generation mobile communications such as IMT-2000. Therefore, in this paper, we proposed of the method to improve the conventional performance with the parallel concatenated 4-New Turbo Decoder using MAP a1gorithm in spite of complexity increasement. In the real-time video and video service over the third generation mobile communications, the performance of the proposed method was analyzed by the reduced decoding delay using the variable decoding method by computer simulation over AWGN and fading channels.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.21 no.6
• /
• pp.1490-1500
• /
• 1996

One of the important issues for multimedia communications over ATM netowrks is efficient use of net-work resources since transmitting video at low cost requires high utilization of channel bandwidth. In this paper, we propose an effective bandwidth estimation scheme for MPEG sources in ATM networks. It has been known difficult to allocate effective transmission bandwidth to MPEG sources due to its bursty charactersitics. The MPEG traffic stream can bemodeled using five traffic parameters:service rate, burst load ratio, interburst load ratio, burst period, and interburst period. Using these parameters, a transmission bandwidth for each source can be estimated, and the estimated bandwidth is applied synchronously to reduce the queue size in a buffer. For a simple network modelof a high speed link that multiplexes a number of virtual-circuit connections, simulations on adaptive bandwidth allocation were performed, and the results show that the queueing delay is significantly reduced, when compared to a fixed bandwidth allocation.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2013.10a
• /
• pp.1033-1036
• /
• 2013

A study about collision resolution is needed because huge file, video and multimedia information are transmitted to wireless LAN. CSMA/CR protocol guarantees fairness, but doesn't consider QoS. To practicalize CSMA/CR protocol in saturation fully-connected single hop network environments, this paper proposes method of collision compensation. previous CSMA/CR protocol chooses CR Slot only considering fairness, but this paper chooses front CR Slot for compensation of collision to station experiencing collisions. The station transmits jam signal and has channel priority of transmission. So transmitting efficiency and successful transmission probability rise. This proposed simple method can offer benefit to station experiencing collisions without additional hardware cost.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.7 no.8
• /
• pp.1854-1863
• /
• 2003

In the recent digital communication systems, the performance of Turbo Code using the mr correction coding depends on the interleaver influencing the free distance determination and the recursive decoding algorithms that is executed in the huh decoder. However, performance depends on the interleaver depth that needs many delays over the reception process. Moreover, turbo code has been blown as the robust coding methods with the confidence over the fading channel. International Telecommunication Union(ITU) has recently adopted it as the standardization of the channel coding over the third generation mobile communications(IMT­2000). Therefore, in this paper, we preposed the interleaver that has the better performance than existing block interleaver, and modified turbo decoder that has the parallel concatenated structure using MAP algorithm. In the real­time voice and video service over third generation mobile communications, the performance of the proposed two methods was analyzed and compared with the existing methods by computer simulation in terms of reduced decoding delay using the variable decoding method over AWGN and fading channels for CDMA environments.