• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.3B
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• pp.130-137
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• 2005

Wireless communication environment is changing rapidly as we use new wireless communication technology such as WiBro to access high speed Internet. As a result, reliable data transmission using TCP is also expected to increase. Since TCP assumes that it is used in wired network, TCP suffers significant performance degradation over wireless network where packet losses are related to non-congestion loss. Especially RTO imposes a great performance degradation of TCP. In this paper, we analyze the loss recovery probabilities based on previous researches, and use simulation results of our algorithm to show that it prevents performance degradation by quickly detecting and recovery losses without RTO during fast recovery.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.6B
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• pp.331-337
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• 2005

The optical Internet is considered as a feasible solution for transporting huge amount of traffic volume in the future Internet. Among optical switching technology for the optical Internet, OBS becomes one of the most promoting solution. Recently, a lebeled OBS(LOBS) architecture is considered for an efficient control on OBS network. Given that a data burst may contain few thousands of IP packets, a single loss of data burst results in a serious throughput degradation in LOBS network. In this paper, we improve the performance of LOBS network by introducing the burst congestion control mechanism. More specifically, the OBS router at the network core detects the network congestion by measuring the loss probability of burst control packet. The OBS router at the network edge reduces the burst generation according to the network condition repored by the OBS router at the network core. Through the simulations, it is shown that the proposed congestion control mechanism can reduce the burst loss probability and improve the LOBS network throughput.

• The Journal of the Acoustical Society of Korea
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• v.37 no.1
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• pp.31-38
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• 2018

Based on the measured data in the south western sea of Jeju Island during the SAVEX15(Shallow Water Acoustic Variability EXperiment 2015), the effect of internal waves on the PPD (Predictive Probability of Detection) of a sonar system was analyzed. The southern west sea of Jeju Island has complex flows due to internal waves and USC (Underwater Sound Channel). In this paper, sonar performance is predicted by probabilistic approach. The LFM (Linear Frequency Modulation) and MLS (Maximum Length Sequence) signals of 11 kHz - 31 kHz band of SAVEX15 data were processed to calculate the TL (Transmission Loss) and NL (Noise Level) at a distance of approximately 2.8 km from the source and the receiver. The PDF (Probability Density Function) of TL and NL is convoluted to obtain the PDF of the SE (Signal Excess) and the PPD according to the depth of the source and receiver is calculated. Analysis of the changes in the PPD over time when there are internal waves such as soliton packet and internal tide has confirmed that the PPD value is affected by different aspects.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7B
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• pp.675-684
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• 2004

There have been several efforts to avoid unnecessary retransmission timeouts (RTOs), which is the main cause for TCP throughput degradation. Unnecessary RTOs can be classified into three groups according to their cause. RTOs due to multiple packet losses in the same window for TCP Reno, the most prevalent TCP version, can be avoided by TCP NewReno or using selective acknowledgement (SACK) option. RTOs occurring when a packet is lost in a window that is not large enough to trigger fast retransmit can be avoided by using the Limited Transmit algorithm. In this Paper, we comparatively analyze these schemes to cope with unnecessary RTOs by numerical analysis and simulations. On the basis of the results in this paper, TCP performance can be quantitatively predicted from the aspect of loss recovery probability. Considering that overall performance of TCP is largely dependent upon the loss recovery performance, the results shown in this paper are of great importance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38A no.1
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• pp.68-78
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• 2013

In this paper, we propose an early spectrum sensing(ESS) as an advance preparation for radio-access trial, which enables multi-mode terminals to access the most appropriate radio-access system in a cloud-conceptual base station system where multiple radio access technologies(RATs) coexist. Prior to a random access to one of RATs, a multi-mode terminal conducts a spectrum sensing over entire frequency bands of whole RATs, then select the RAT with the lowest sensing power, that is likely to have the most available spectrum. Thus, an access failure caused by that the selected RAT has no available radio spectrum could be avoidable in advance. In computer simulation, we consider as various RATs as possible. First, circuit and packet systems are taken into consideration. In addition, the packet systems are classified according to the feasibility of carrier aggregation(CA). In case of terminal, three modes are considered with circuit-only, packet-only, and multi-mode. Subsequently, packet traffic is classified into real-time and non-real-time traffic with three different tolerable delay levels. The simulation includes a call process starting with a call generation and ending up with a resource allocation reflecting individual user's QoS requirements and evaluates the proposed scheme in terms of the successful access probability, system access time, system balancing factor and packet loss probability.

• Journal of Broadcast Engineering
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• v.25 no.4
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• pp.609-619
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• 2020

In accordance with a demand for high quality video streaming, it needs high data rate in limited bandwidth and more traffic congestion occurs. In particular, when providing real time video service, packet loss rate and bit error probability increase significantly. To solve these problems, a raptor code, which is one of FEC(Forward Error Correction) techniques, is pervasively used in the application layers as a method for improving real-time service quality. In this paper, we propose a method of determining image transmission parameters based on various deep neural networks to increase transmission efficiency at a similar level of image quality by using raptor codes. The proposed neural network uses the packet loss rate, video encoding rate and data rate as inputs, and outputs raptor FEC parameters and packet sizes. The results of the proposed method present that the throughput is 1.2% higher than that of the existing multimedia transmission technique by optimizing the transmission efficiency at a PSNR(Peak Signal-to-Noise Ratio) level similar to that of the existing technique.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.12
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• pp.99-108
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• 2014

WBAN(Wireless Body Area Network) is network to support medical and non-medical services. It is susceptible to loss and delay of data. WBAN is required to satisfy many kinds of demands such as a variety of data rate and a data priority for providing various service. In this paper scheduling algorithm, considering a data priority and transmission delay time, is proposed to improve service quality of WBAN. The proposed algorithm operates by allocating a channel to a flow with longer transmission delay. When a packet, in a queue of herb, is left within a certain period, the packet is assigned a channel and transmitted according to a data priority. Through the comparison with other existing scheduling algorithms, it is confirmed that QoS is improved due to higher arrival probability of medical data and less delay time in the proposed algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7B
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• pp.650-659
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• 2004

Domestic wireless network environment is changing rapidly while adapting to meet service requirements of users and growth of market. As a result, reliable data transmission using TCP is also expected to increase. Since TCP assumes that it is used in wired networt TCP suffers significant performance degradation over wireless network where packet losses are not always result of network congestion. Especially RTO imposes a great performance degradation of TCP. In this paper, we propose DAC$^{＋}$ and EFR in order to prevent performance degradation by quickly detecting and recovering loss without RTO during fast recovery. Compared with TCP NewReno, proposed scheme shows improvements in steady-state in terms of higher fast recovery Probability and reduced response time.

• KIPS Transactions on Computer and Communication Systems
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• v.2 no.3
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• pp.117-124
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• 2013

In this paper we propose a TFRC(TCP Friendly Rate Control) which guarantees a minimum rate in order to improve the efficiency of the previous TFRC which cannot distinguish congestion losses and wireless losses and decreases throughput both in wired and wireless networks. This TFRC technique is able to guarantee a minimum rate for video by restricting a loss event rate with packet loss probability about existing TFRC and constraining a rate reduction from the feedback timeout. When we experimented both the existing TFRC and the new one with TCP in the same network, we found that the latter is better than the former. Consequently, it shows that the proposed TFRC can improve video streaming quality using a guaranteed minimum transmission rate.

• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.229-232
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• 2000

AAL2 has been adopted for bandwidth-efficient trans-mission of low bit tate traffic over ATM networks in ITUT and ATM Forum. Since ATM/AAL2 is expected to be used as a switching technology in third-generation mobile access networks and mobile data traffic is expected to increase rapidly in near future, there must be a need for efficient scheduling scheme satisfying the QoS requirement of ow bit rate voice as well as the one of high bit rate data. In this paper, we propose a new class-scheduling scheme to improve data packet loss probability, while Qos of voice traffic is guaranteed, when data traffic is multiplexed together with mobile voice traffic into a single ATM VCC. The proposed scheme can efficiently support data traffic by assigning a time threshold value to voice traffic. Through simulation study, we show that the proposed scheme does not only achieve better efficiency for providing both mobile voice and data services than HOL class-scheduling scheme and normal FIFO scheme, but also guarantees mean voice packet delay under a certain criteria.