• The Journal of Korean Institute of Communications and Information Sciences
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• v.42 no.3
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• pp.604-611
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• 2017

In wireless sensor networks, an energy efficient operation is important since the energy of the sensors is limited. This paper proposes an energy efficient method that reduces a packet generation with Matrix Completion method where sensor value matrix has low-rank and decreases a collision rate and an overhead by transmitting only sensor ID to a time slot corresponding to the sensor value. Computer simulations demonstrates that the proposed method shows 17% of transmission failure and 73% of the packet generation compared to a conventional CSMA/CS. Delay time of transmitting information of the proposed method exhibits 22% of the CSMA/CA and the MSE error after reconstructing sensor values by Singular Value Thresholding(SVT) in Fusion Center is 87% of the CSMA/CA.

• Journal of KIISE:Information Networking
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• v.33 no.4
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• pp.333-342
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• 2006

The cdma2000 lxEV - DO mobile communication system provides broadcast and multicast services (BCMCS) to meet an increasing demand from multimedia data services. The servicing of video streams over a BCMCS network must, however, face a challenge from the unreliable and error-prone nature of the radio channel. The BCMCS network uses Reed-Solomon coding integrated with the MAC protocol for error recovery. We analyze this coding technique and show that it is not effective in the case of slowly moving mobiles. To improve the playback quality of an MPEG-4 FGS video stream, we propose the Hybrid error recovery scheme, which combines Reed-Solomon with ARQ, using slots which are saved by reducing the Reed-Solomon coding overhead. The target packets to be retransmitted are prioritized by a utility function to reduce the packet error rate in the application layer within a fixed retransmission budget. This is achieved by considering of the map of the error control block at each mobile node. The proposed Hybrid error recovery scheme also uses the characteristics of MPEG-4 FGS (fine granularity scalability) to improve the video quality even when conditions are adverse: slow-moving nodes and a high error rate in the physical channel.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.3B
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• pp.85-94
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• 2005

In this paper, we propose a reliable high-speed UDP-based media transport with an adaptive error control. The proposed adaptive transport scheme controls the amount of redundancy by monitoring the network in order to adapt to network fluctuations efficiently. The feedback of receiver enables the sender to be aware of current reception status (i.e., rate and type of packet loss) and to estimate the expected network status. Based on this, the proposed transport attempts to enable reliable transport by adaptively controlling the amount of both whole sending rate and the ratio for adaptive FEC code. Experiment with high-speed network has been conducted to verify the performance of the proposed system that demonstrates the enhanced reliability of the proposed transport at the speed of up to several hundred Mbps.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.9 no.5
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• pp.589-602
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• 1998

While ATM technology results in considerable advantages(less overhead, increased throughput) in an optical networks, It causes severe problems(single and burst error) when ATM is transmitted over an error-prone channel, such as wireless link. In this paper we describe our investigation on ATM over wireless network. we first evaluate performance of ATM data transmission over a wireless link. Secondly we analyze performance improvements of Forward Error Correction(FEC) or ARQ(Automatic Repeat reQuest) scheme applied to wireless ATM link and provide performance comparison between FEC and ARQ through the use of packet error rate and throughput. Lastly we suggest error control architecture to overcome the impact of the bit error characteristics of a wireless link on wireless ATM network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.6A
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• pp.820-831
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• 1999

In this paper, we have compared the performance of a D-TDMA protocol with that of a CDMA protocol, in radio access system for integrated voice/data services.The D-TDMA protocol is based on a generic dynamic channel assignment approach to be followed a combination of “circuit mode” reservation for voice calls, along with dynamic first-come-first served assignment of remaining capacity for data messages. In the CDMA protocol, we have used the voice activity detection to reduce the interface power of other mobiles in internal and external cells, and analyzed the interference power ratio. Also we have computed BER(Bit Error Rate) by using this interference power ratio and evaluated voice blocking probability(voice packet loss probability) and data transmission delay, according to average data length and average data arrival rate.We have found the CDMA protocol achieves comparatively higher performance for short data length, regardless of data arrival rate. Otherwise, the data transmission delay of D-TDMA protocol is shorter than that of the CDMA protocol for long data message.

• Journal of KIISE:Information Networking
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• v.31 no.1
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• pp.1-10
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• 2004

This study discusses Application-layer FEC using erasure codes. Because of the simple decoding process, erasure codes are used effectively in Application-layer FEC to deal with Packet-level errors. The large number of parity packets makes the loss rate to be small, but causes the network congestion to be worse. Thus, a redundancy control algorithm that can adjust the number of parity packets depending on network conditions is necessary. In addition, it is natural that high-priority frames such as I frames should produce more parity packets than low-priority frames such as P and B frames. In this paper, we propose a redundancy control algorithm that can adjust the amount of redundancy depending on the network conditions and depending on data priority, and test the performance in simple links and congestion links.

• Journal of Advanced Marine Engineering and Technology
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• v.40 no.7
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• pp.629-634
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• 2016

This paper describes the implementation of a ZigBee sensor node that can be utilized as a multiband and machine to machine (M2M) communication gateway. The IEEE 802.15.4-2003 standard was used as the wireless network frequency band. Ember's Type EM357 SoC was used as the transmission and reception device to perform the communication function, and it was also used for both the main M2M gateway and the sensor node. For the implementation of the operating protocol, EmberZNet Stack 4.5.4 from the Ember Corporation was used. The measurement of the reception sensitivity in the receiving module and the actual output signal from the reference were obtained from the transmission of a packet, and the packet included the M2M gateway within the attached ZigBee sensor. The packet error rate was measured as 0% with a -98 dBm reception sensitivity at the ZigBee frequency. In addition, excellent current characteristics of the ZigBee modules were shown by the implementation of the low-power circuit.

• Journal of KIISE:Information Networking
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• v.31 no.2
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• pp.188-195
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• 2004

We have analyzed the performance of TCP in the CDMA mobile communications systems with the adaptive modulation and coding(AMC). The wireless channel using AMC is characterized with not high error rate but highly varying bandwidth. Due to time-varying bandwidth, timeout events of TCP occurs more frequently, which leads to the throughput degradation. The analysis model is composed of the two parts. In the first part, we divide TCP packet stream into ‘packet groups’and derive the probability distribution of the wireless transmission time of each Packet group that reflects the time varying characteristics of AMC. In the second part, we formulate embedded Markov chain by making use of the results of the first part to model TCP timer mechanism and wireless transmission. Since our system model is characterized by the forward link high speed data transmission using AMC, the results reported in this paper can be used as a guideline for the design and operation of HSDPA, 1xEV-DO, and 1xEV-DV.

• Journal of Internet Computing and Services
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• v.3 no.3
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• pp.67-74
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• 2002

Asynchronous transfer mode (ATM) can be efficiently used to transport packet data services. The switching system will support voice and packet data services simultaneously from end to end applications. To guarantee quality of service (QoS) of the offered services, source rate to send packet data is needed to control the network overload condition. Most existing control algorithms are shown to provide the threshold-based feedback control technique. However, real-time voice calls can be dynamically connected and released during data services in the network. If the feedback control information delays, quality of the serviced voice can be degraded due to a time delay between source and destination in the high speed link, An adaptive algorithm based on the optimal least mean square error technique is presented for the predictive feedback control technique. The algorithm attempts to predict a future buffer size from weight (slope) adaptation of unknown functions, which are used for feedback control. Simulation results are presented, which show the effectiveness of the algorithm.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.1
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• pp.625-629
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• 2005

According to the rapid development of the information and communication technology, the demand on the efficient compression technology for the multimedia data is increased magnificently. In this Paper, we designed new compression algorithm structure using wavelet base for the compression of ECG signal and audible signal data. We examined the efficiency of the compression between 2-band structure and wavelet packet structure, and investigated the efficiency and reconstruction error by wavelet base function using Daubechies wavelet coefficient and Coiflet coefficient for each structure. Finally, data were compressed further more using Huffman code, and resultant Compression Rate(CR) and Percent Root Mean Square difference(PRD) were compared with those of existent DCT.