• Journal of the Korean Institute of Telematics and Electronics A
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• v.31A no.7
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• pp.17-28
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• 1994

A combined random/reservation multiple access protocol is proposed which can provide services for packet transmission in mobile satellite systems between mobile statons, or between mobile stations and fixed stations. Random multiple access protocol and reservation multiple access protocol which are currently employed in most satellite communication systems have some strengthes and some weeknesses in according to the kind of user and traffic. In this paper, a combined random/reservation multiple access protocol with better characteristics is proposed. The models of the modified random access protocol and the proposed access protocol is setted and analyzed. The performance of the PDAMA protocol, the random access protocol and the proposed access protocol is compared using simulation. For small packet arrival rate, the performance of the proposed access protocol is close at that of the modified random access protocol, and better than that of the PDAMA protocol. As the packet arrival rate is increased, the modified random access protocol is saturated and unstable at 0.23, and the performance of the proposed access protocol is better than that of the PDAMA protocol.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.2A
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• pp.187-194
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• 2004

We propose priority-based reservation code multiple access (P-RCMA) which can enhance voice traffic quality of the previous RCMA. The proposed protocol maintains two power levels and consider traffic characteristics in contending shared available codes to transmit packets. P-RCMA gives priority to the voice request packets rather than data packets by capture effect at the receiver part of base station. We show numerical results from EPA (equilibrium point analysis) analysis and simulation study in terms of voice packet dropping probability and average data packet transmission delay.

• The Journal of the Acoustical Society of Korea
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• v.18 no.6
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• pp.71-76
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• 1999

In digital cellular communication systems, tandem coding occurs in communications between mobile phones with different speech codecs, resulting in poor voice quality, high computational load, and long transmission delay. In this paper, voice packet conversion technique is proposed to solve the tandem coding problems, and packet conversion algorithm from 13kbps QCELP to 8kbps QCELP is developed. Simulations using various speech data show that the proposed packet conversion method produces voice quality which is equivalent to that by the conventional tandem coding method with shorter transmission delay using about 33% computational load.

• International Journal of Computer Science & Network Security
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• v.22 no.3
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• pp.1-8
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• 2022

Supporting real-time flows with delay and throughput constraints is an important challenge for future wireless networks. In this paper, we develop an optimal scheduling scheme to optimally choose the packets to transmit. The optimal transmission strategy is based on an observable Markov decision process. The novelty of the work focuses on a priority-based probabilistic packet scheduling strategy for efficient packet transmission. This helps in providing guaranteed services to real time traffic in Heterogeneous Wireless Networks. The proposed scheduling mechanism is able to optimize the desired performance. The proposed scheduler improves the overall end-to-end delay, decreases the packet loss ratio, and reduces blocking probability even in the case of congested network.

• Journal of Korea Multimedia Society
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• v.7 no.2
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• pp.204-210
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• 2004

This paper proposes a transmission probability control scheme and MAC protocol for packet data services in slotted CDMA system. In slotted CDMA system, multiple access interference is the major factor of unsuccessful packet transmissions. Therefore, in order to obtain the optimal system throughput, the number of simultaneously transmitted packets should be kept at a proper level. In the proposed protocol, the base station calculates the packet transmission probability of mobile stations according to the offered load and then broadcasts it. Mobile stations, which have a packet, attempt to transmit packet with the received probability. Numerical analysis and simulation results show that the proposed scheme can offer better system throughput than the conventional one, and guarantee a good fairness among all mobile stations regardless of the offered load.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.10B
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• pp.1824-1831
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• 1999

In this paper, the data recovery performance of systematic erasure codes in burst loss environments is analyzed and the estimation method of redundant data according to loss characteristics is suggested. The burstness of packet loss is modeled by Gilbert model, and the performance of proposed packet loss recovery method in the case of using systematic erasure code is analyzed based on previous study on the loss recovery in the case of using erasure code. The required redundancy data fitting method for systematic erasure code in the condition of given loss property is suggested in the consideration of packet loss characteristics such as average packet loss rate and average loss length.

• Journal of Advanced Navigation Technology
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• v.8 no.1
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• pp.65-72
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• 2004

A basic access method using for IEEE 802.11a wireless LANs is the DCF method that is based on the CSMA/CA. But, Since IEEE 802.11 MAC layer uses original backoff algorithm (Exponential backoff method), when collision occurs, the size of contention windows increases the double size. Hence, packet transmission delay time increases and efficiency is decreased by original backoff scheme. In this paper, we have analyzed TCP packet transmission time of IEEE 802.11 MAC DCF protocol for wireless LANs using a proposed enhanced backoff algorithm. From the results, in OFDM/quadrature phase shift keying channel (QPSK), we can achieve that the transmission time in wireless channel decreases as the TCP packet size increases and based on the data collected, we can infer the correlation between TCP packet size and total message transmission time, allowing for an inference of the optimal packet size in the TCP layer.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.1B
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• pp.34-44
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• 2003

As the recent rapid development of internet technology and the wide spread of multimedia communications, massive increase of network traffic causes some problems such as the lack of network paths and the bad quality of service To resolve these problems, this paper presents a web-based traffic control system which supports QoS of realtime packet transmission for the multimedia communication The traffic control system presented in this paper applies the integrated service model and provides QoS of packet transmission by means of determining the packet transmission rate according to the policies of network manager and the optimal resource allocation considering the end-to-end traffic load It also provides QoS for the realtime packet transmission through the admission controller and the packet scheduler by the modified $WF^2Q^+$ algorithm support asynchronous and class-based queuing.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4
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• pp.349-360
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• 2002

In ITU H.323 teleconference system, the RTP/RTCP protocol is offered to transfer real-time multimedia stream. Both sender and receiver hate experience in packet loss and jitter which result from network congestion over Internet. Audio quality over Internet depends on the number of lost packets and on jitter between successive packets. The goal of our study is to improve the speech quality over Internet by checking the packet loss characteristics of the network and adopting the but for control management mechanism at the receiver. We suggest a dynamic redundant audio transmission mechanism which examines the packet loss rate and uses the feedback information through RTCP.

• 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.