• The Journal of the Acoustical Society of Korea
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• v.34 no.1
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• pp.82-91
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• 2015

In this paper, a novel codebook design method for data modem over voice channel is presented. Proposed method searches the symbols which have the maximum probability distribution overlap in the symbol space and minimizes the overlap to improve the symbol error rate via minimax optimization. We present numerical simulations and an example implementation. We also give the results of the experiment tests.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4
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• pp.394-400
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• 2002

In digital speech communications, the quality of service can be increased by speech coding scheme that is adaptive to the error rate of voice packet transmission. However, current communication protocol in cellular and internet communications does not provide the function that transmits the channel error information. To solute this problem, in this paper, new method for real-time transmission of channel error information is proposed, where channel error information is embedded in voice packet. The proposed method utilizes the pulse positions of codevector in ACELP speech codec, which results in little degradation in speech quality and low false alarm rate. The simulations with various speech data show that the proposed method meets the requirement in speech quality, detection rate, and false alarm rate.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.8B
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• pp.780-786
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• 2002

In this paper several access schemes for voice and data packet over CPCH with waiting time signalling are proposed. Waiting time signalling is used for reducing the excessive attempt of UE when all the CPCH channel resources is being used by other UEs. The proposed schemes in this paper are better than other schemes in view of the voice/data packets? dropping probability, average access attempt to transmit voice/data packets successfully and the average throughput.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39C no.1
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• pp.90-96
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• 2014

IEEE802.15.4 that is a standard for sensor networks is mainly used the wireless personal area networks such as ZigBee networks and it features low-power, low-speed data communication. However, recently research for interworking sensor network based voice communication and Session Initiation Protocol (SIP) for long-range, multi-user support has been actively conducted. In this paper, we designed a integrated base station based existing systems for interworking sensor networks based voice communication and SIP. We measured number of packet and delay according to increase the number of users to evaluate the performance of designed Base Station.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.8B
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• pp.1021-1027
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• 2011

VHF is generally used for data or voice communications between ships and land stations. Although there exist many analytical and empirical channel models for radio waves on land, over-the-sea models are very limited to basic description. In this paper, a VHF channel is measured using vertical and/or horizontal whip antennas on shore and a moving ship for various ranges. The results show that the Doppler shift and changes in power for various ranges are observed. We can also find that the statistical analysis results in that the statistical parameters are rapidly changes, especially for V/H and H/V channels.

• Journal of Communications and Networks
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• v.10 no.3
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• pp.287-296
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• 2008

The IEEE 802.11 wireless standard uses the carrier sense multiple access with collision avoidance (CSMA/CA) as its MAC protocol (during the distributed coordination function period). This protocol is an adaptation of the CSMA/CD of the wired networks. CSMA/CA mechanism cannot guarantee quality of service (QoS) required by the application because orits random access method. In this study, we propose a new MAC protocol that considers different types of traffic (e.g., voice and data) and for each traffic type different priority levels are assigned. To improve the QoS of IEEE 802.11 MAC protocols over a multi-channel CSMA/CA, we have developed a new admission policy for both voice and data traffics. This protocol can be performed in direct sequence spread spectrum (DSSS) or frequency hopping spread spectrum (FHSS). For voice traffic we reserve a channel, while for data traffic the access is random using a CSMA/CA mechanism, and in this case a selective reject and push-out mechanism is added to meet the quality of service required by data traffic. To study the performance of the proposed protocol and to show the benefits of our design, a mathematical model is built based on Markov chains. The system could be represented by a Markov chain which is difficult to solve as the state-space is too large. This is due to the resource management and user mobility. Thus, we propose to build an aggregated Markov chain with a smaller state-space that allows performance measures to be computed easily. We have used stochastic comparisons of Markov chains to prove that the proposed access protocol (with selective reject and push-out mechanisms) gives less loss rates of high priority connections (data and voices) than the traditional one (without admission policy and selective reject and push-out mechanisms). We give numerical results to confirm mathematical proofs.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.5B
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• pp.879-888
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• 2000

In this paper, the performance of IS-95B CDMA system adopting multi-code scheme for data service on the forward link is obtained by theoretical analysis and computer simulation. The wireless channel is assumed to include AWGN, MUI(Multi-User Interference), and MPI(Multi-Path Interference), and multipath Rayleigh fading modeled by TDL(Tapped Delay Line). To obtain actual performance, IS-95B system using PN codes(M-sequence) concatenated with Walsh codes is considered. The BER performance of voice/data service satisfying QoS requirement and the maximum capacity of voice users is derived in terms of various system parameters(voice activity factor, required SNR of voice/data users, the number of multi-codes, the number of voice/data users.,etc.). As a result, under the condition that the maximum allowable number of voice user is 38 in voice only system, as the number of multi-code is decreased by 1, the maximum allowable number of voice user is increased by 5% for system with 1 data user and 22.3% for system with 4 data users. Therefore, it is possible to increase maximum allowable number of users in IS-95B CDMA system by appropriate control of the number of multi-codes and required SNR according to channel environment and traffic characteristic.

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.245-248
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• 2005

Wireless LANs based on the IEEE 802.11 standard are widely spread for use nowadays. Traffic which are conveyed over the WLANs change rapidly from normal data such as Email and Web pages, to multimedia data of high resolution video and voice. To meet QoS (Quality of Service) required by these multimedia traffic, the IEEE 802 committee recently has developed a new standard, IEEE 802.11e. Current IEEE 802.11e, however, is not sufficient to support service differentiations and network performance enhancements, under a varying network environment experiencing as varying channel characteristics and high network loads. Recently, there have been much research to complement this deficiency of 802.11e standard. This paper surveys these research efforts.

• Journal of The Institute of Information and Telecommunication Facilities Engineering
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• v.8 no.4
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• pp.201-206
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• 2009

The third generation mobile communications system requiring the reliable multimedia data transmission has provided with the reliable voice, data and video services over the variable propagation environment. However the broadband wireless multiple access technologies cause Inter Symbol Interference (ISI) or Multiple Access Interference (MAI) to degrade the performance of W-CDMA (Code Division Multiple Access) system. Constant Modulus Algorithm (CMA) which is frequently used as the adaptive blind equalizers to remove the interfering signal has ill-convergence phenomenon without proper initialization. In this paper, new blind equalization method based on conventional CMA is proposed to improve the channel efficiency, and through computer simulation this is tested over the time varying fading environment of mobile communication system. Consequently, new blind equalization method into concatenated Kalman filter with CMA is verified better than conventional CMA through adopting minimum mean square errors and eye-pattern obtained from algorithm are compared.

• Journal of the Korea Institute of Military Science and Technology
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• v.13 no.2
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• pp.243-251
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• 2010

The military has been putting great efforts into applying data communication on existing voice communication systems being used in NCW(Network Centric Warfare). Data communication will be an effective choice in one of many effort to yield a minimum kill chain, comparing to legacy voice communications, when tactical units conduct their missions. However, the required budget will be enormous, in case of the replacement of a lot of legacy communication systems with new one. As a cost-effective alternative, the tactical data communication systems using the conventional radio systems instead of the development of new radio systems has been proposed. It is mandatory, though, to ensure QoS while maintaining data communication by making use of legacy radio systems already in use. This paper focuses on the performance issues experimented and analyzed for tactical data communication through the legacy radio systems as the first step towards guaranteed QoS. We have conducted various experiments such as the transmission error rate on certain tactical messages, performance evaluation of redundant transfers, the relationship between the transmission frame size and rate of error, the identification of error points in the transmission frame, and techniques to reduce the errors in both hopping and non-hopping modes. As a result of the performance experiments, The adaptive communication module which decides the redundant transmission or the Forward Error Correction(FEC) technique by analyzing channel status and current transmission status(hopping/non-hopping) of the legacy radio should be designed. the FEC technique in non-hopping, and the redundant transmission technique in hopping mode was recommended from the result of experiment with the frame size is 20bytes in non-hopping and 10Bytes frame size in hopping mode.