• Proceedings of the KIEE Conference
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• 2008.11a
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• pp.379-381
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• 2008

TETRA(Terrestrial Trunked Radio) is a digital trunked radio standard developed by the ETSI(European Telecommunications Standards Institute). Currently, TETRA was set Digital TRS in electric power If wireless backbone network. In this time, we use many company's TETRA modem. So, TETRA modem performance evaluation is very important. TETRA modem use two type of Data transfer mode. One is Packet Data using UDP/IP. and the other is SDS(Short Data Service). In this paper, We generate Packet Data using Traffic Generator module. Packet Data transfer 1000 times each 10 bytes to 400 bytes. We analyze transmission delay time, success rate and standard deviation.

• Journal of Broadcast Engineering
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• v.4 no.1
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• pp.1-11
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• 1999

This paper introduces orthogonally multiplexed modulation and demodulation methods based on Wavelet Packet Bases and particularly describes Wavelet Packet Modulation (WPM) techniques that provide the designer of transmission signal set in time-frequency domain with tree structural information which can be adapted to given channel characterristics. Multi-dimensional signaling methods are also contrasted to common and different characteristics of conventional QAM. multi-tone modulation methods. The paper addresses the mothod how to find a best tree structure that has more adaptivity to impulse and narrowband tone pulse noises using a tunning algorithm which arbitrarily partitions the time-frequency space and makes a suitable orthogonal signaling waveforms. Simulation results exhibits a favorable performance over existing mod/demod methods specially for narrowband tone pulse and impulse interferences.

• Proceedings of the KIEE Conference
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• 2006.04a
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• pp.105-107
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• 2006

As internet streaming data increase, transport protocol such as TCP, TGP-Friendly is important to study control transmission rate and share of Internet bandwidth. In this paper, we propose a TCP-Friendly protocol using Neural Network for media delivery over wired Internet which has various traffic size(PTFRC). PTFRC can effectively send streaming data when occur congestion and predict one-step ahead round trip time and packet loss rate. A multi-layer perceptron structure is used as the prediction model, and the Levenberg-Marquardt algorithm is used as a traning algorithm. The performance of the PTFRC was evaluated by the share of Bandwidth and packet loss rate with various protocols.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.14 no.5
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• pp.453-461
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• 1989

Mean waiting time for each priority packet of each node in packet based priority token ring local area networks(LAN) was approximately analyzed using Bux's token ring LAN results which have not considered priority and Cohbam's head of line(HOL) priority results. In this paper, priority reservation method suggested in the IEEE 802.5 standard was not used. Relative error between numerical results which was presented in this paper and simulation results was identified by +-5%. For traffic intenity, number of node, packet length, transmission speed, line length, token latency, number of priority class and traffic percentage to some heavy trafficd node, mean waiting time of each priority was analyzed.

• The KIPS Transactions:PartC
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• v.13C no.3 s.106
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• pp.331-338
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• 2006

Most fair packet schedulers supporting quality-of-services of real-time multimedia applications are based on the finish time design scheme in which the expected transmission finish time of each packet is used as its timestamp. This scheme can adjust the latency of a flow with raising the flow's scheduling rate but it may suffer from severe bandwidth loss due to the coupled rate and delay allocation. This paper first introduces the concept of delay resource, and then proposes a scheduling method to improve the bandwidth utilization in which delay resource being lost due to the coupled allocation is transformed into bandwidth one. The performance evaluation shows that the proposed method gives higher bandwidth utilization by up to 50%.

• Journal of the Korea Society of Computer and Information
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• v.21 no.6
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• pp.47-53
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• 2016

This paper deals with the capacity constrained time slot assignment problem(CTSAP) that a satellite switches to traffic between $m{\times}n$ ground stations using on-board $k{\leq}_{min}\{m,n\}$ k-transponders switching modes in SS/TDMA time-division technology. There was no polynomial time algorithm to solve the optimal solution thus this problem classified by NP-hard. This paper suggests a heuristic algorithm with O(mn) time complexity to solve the optimal solution for this problem. Firstly, the proposed algorithm selects maximum packet lengths of $\({mn \atop c}\)$ combination and transmits the cut of minimum packet length in each switching mode(MSMC). In the case of last switching mode with inefficient transmission, we applies a compensation strategy to obtain the minimum number of switching modes and the minimum makespan. The proposed algorithm finds optimal solution in polynomial time for all of the experimental data.

• Journal of Industrial Technology
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• v.23 no.A
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• pp.41-46
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• 2003

Because the quality of a radio link in real environment is generally varied with time, there is a difference between the delay in the real environment and one obtained from the analytic model where a time-varying link model is not used as a link model for a Bluetooth. This paper analyzes the transmission delay of a message in the time-varying radio link model for the Bluetooth. The time-varying radio link is modeled with a two-state Markov model. The mean transmission delay of the message is analytically obtained in terms of the arrival rate of the message, the state transition probability in the Markov model, and the packet error rate.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.8A
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• pp.858-870
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• 2007

TCP is popular protocol for reliable data delivery in the Internet. In recent years, wireless environments with transmission errors are becoming more common. Therefore, there is significant interest in using TCP over wireless links. Previous works have shown that, unless the protocol is modified, TCP may perform poorly on paths that include a wireless link subject to transmission errors. The reason for this is the implicit assumption in TCP that all packet losses are due to congestion which causes unnecessary reduction of transmission rate when the cause of packet losses are wireless transmission errors. In this paper, we propose a new LDA that monitors the network congestion level using ROTT. And we evaluate the performance of our scheme and compare with TCP Veno, Spike scheme with NS2(Network Simulator 2). In the result of our experiment, our scheme reduces the packet loss misclassification to maximum 55% of other schemes. And the results of another simulation show that our scheme raise its transmission rate with the fairness preserved.

• Journal of applied mathematics & informatics
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• v.17 no.1_2_3
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• pp.547-557
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• 2005

In this paper, we investigate the packet delay distribution of a dynamic bandwidth allocation(DBA) scheme in an Ethernet passive optical network(EPON). We focus on the gated-exhaustive vacation system. We assume that input packets arrive at an optical network unit(ONU) according to general interarrival distribution. We use a discrete time queueing model in order to find the packet delay distribution of the gated-exhaustive system with the primary transmission queue and the secondary input queue. We give some numerical examples to investigate the mean packet delays of the proposed queueing model to analyze the DBA scheme in an EPON.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.6 no.4
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• pp.1006-1025
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• 2012

Improving the quality of service in IP networks is a major challenge for real-time voice communications. In particular, packet arrival-delay variation, so-called "jitter," is one of the main factors that degrade the quality of voice in mobile devices with the voice-over Internet protocol (VoIP). To resolve this issue, a receiver-based enhanced timing recovery algorithm combined with active jitter estimation is proposed. The proposed algorithm copes with the effect of transmission jitter by expanding or compressing each packet according to the predicted network delay and variations. Additionally, the active network jitter estimation incorporates rapid detection of delay spikes and reacts to changes in network conditions. Extensive simulations have shown that the proposed algorithm delivers high voice quality by pursuing an optimal trade-off between average buffering delay and packet loss rate.