• The Transactions of the Korean Institute of Electrical Engineers D
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• v.55 no.5
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• pp.236-239
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• 2006

A wireless MAC scheduler that provides a high level of quality-of-service (QoS) for video-on-demand (VOD) applications while achieving a reasonable level of system throughput is proposed. The proposed scheduler considers both channel qualities of mobiles and the urgency of real-time packets coming from VOD applications in a cross-layer approach between application and MAC layers.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.6
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• pp.1496-1502
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• 1998

In this paper, a real-time audio broadcasting service system which is robust to loaded traffic on the Internet is developed. For implementing reliable real-time data transfer, the transfer characteristics of TCP/IP and UDP/IP was compared and analyzed. For lost packet recovery, redundant audio data algorithm was used and interleaving technique was applied for scattering consecutive packet loss. Test results showed, when using TCP/IP, pause occurred during playback, and when using UDP/IP, a stable receive rate was noticeable but the quality of the sound was lower than that of uisng TCP/IP. The recovery rate using redundant audio data and interleaving technique is shown in Fig. 9 and the delay is shown in Fig 4.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.11B
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• pp.969-978
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• 2006

Recently, there has been explosive growth in Internet service contents, and we have witnessed the emergence of many real-time applications on the Internet. These types of applications require special support from the underlying network such as reliability, timeless, and guaranteed delivery, as well as different levels of service quality. Estimation network status and transcoding data rate are researching for QoS of multimedia service by using the RTP/RTCP. We propose two schemes that the way to calculate the one-way delay by expanding the structure of RTP/RTCP in Internet that provides us 'best effort' service and to estimate the network condition by using this in this paper. And we propose the dynamic RTCP mechanism based on one-way delay to deal with rapidly changed network situation immediately. In simulation, proposed monitoring schemes using the one-way delay has similar in performance to monitoring schemes using the GPS, proposed dynamic RTCP scheme obtain the desired result of decreased packet loss rate by sending RTCP RR packet lower than other schemes.

• Journal of Korea Multimedia Society
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• v.22 no.12
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• pp.1438-1446
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• 2019

Recently, the number of mobile users as well as high-speed Internet user has been increasing rapidly. Moreover, traffic is growing fast as services that provide real-time content such as Youtube and Netflix become popular. The problem of traffic control in real-time content services is important because many people use cell phones to receive real-time content. In this regard, the field of CCN is currently being studied. We studied the mobility of nodes among CCN research fields. Node mobility can be divided into three categories : consumer mobility, intermediate node, and provide mobility. In this paper, we propose Mobility Node Support(MD-INS) to support the intermediate-node mobility in CCN real-time services. Experimental results show that the proposed scheme shows better performance than CCN in terms of service disconnection time and packet loss.

• 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.

• Proceedings of the Korean Society of Computer Information Conference
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• 2018.07a
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• pp.45-46
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• 2018

본 논문에서는 시간에 엄격한(Time critical) 산업용 IoT(Industrial IoT) 환경의 무선 센서 네트워크 시스템 상의 효율적인 패킷 전달과 정확도(Accuracy) 향상을 위해 강화학습과 EDF 알고리즘을 혼합한 스케쥴링 기법을 제안한다. 이 방식은 다중 대기열(Multiple queue) 환경에서 각 대기열의 요구 정확도(Accuracy Requirement)를 기준으로 최대한 패킷 처리를 미룸으로써 효율적인 CPU자원 분배와 패킷 손실율(Packet Loss)을 조절한다. 제안하는 기법은 무선 센서 네트워크 상의 가변적이고 예측 불가능한 환경에 대한 사전지식이 없이도 요구하는 서비스의 질(Quality of service)를 만족할 수 있도록 한다. 또한 정확도를 요구조건으로 제시하여 마감시간이 중요시되는 작업에서도 효율을 최대화한다.

• ETRI Journal
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• v.34 no.6
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• pp.827-837
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• 2012

We propose a flow admission control (FAC) for setting up a wire-speed connection for new flows based on their negotiated bandwidth. It also terminates a flow that does not have a packet transmitted within a certain period determined by the users. The FAC can be used to provide a reliable transmission of user datagram and transmission control protocol applications. If the period of flows can be set to a short time period, we can monitor active flows that carry a packet over networks during the flow period. Such powerful flow management can also be applied to security systems to detect a denial-of-service attack. We implement a network processor called a flow management network processor (FMNP), which is the second generation of the device that supports FAC. It has forty reduced instruction set computer core processors optimized for packet processing. It is fabricated in 65-nm CMOS technology and has a 40-Gbps process performance. We prove that a flow router equipped with an FMNP is better than legacy systems in terms of throughput and packet loss.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.37 no.1
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• pp.32-43
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• 2000

In this paper, we investigate the existing medium access control (MAC) protocols to integrate the voice and data services in packet-based CDMA networks and furthermore, propose a new approach to circumvent the operational limits inherent in them. We propose the $P^2R$-CDMA (Prioritized Packet Reservation Code Division Multiple Access) protocol for the uplink in the synchronous multi-code CDMA system, which employs the centralized frame-based slot reservation along with the dynamic slot assignment in the base station using the QoS-oriented dynamic priority of individual terminal. The simulation results show that, as compared with the existing scheme based on the adaptive permission probability control (APC), the proposed approach can significantly improve the system capacity while guaranteeing the real-time requirement of voice service.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.1A
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• pp.16-24
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• 2010

High Speed Uplink Packet Access(HSUPA) is a WCDMA Release 6 technology which corresponds to High Speed Downlink Packet Access(HSDPA). Node B Supports fast scheduling, Hybrid ARQ(HARQ), short Transmission Time Interval(TTI) for high rate uplink packet data. It is very important to detect Iub congestion to improve end user's Quality of Service(QoS). This paper proposes Node B Congestion Detection(BCD) mechanism and suggests to use the hybrid of Transport Network Layer(TNL) congestion detection and BCD. It is shown that HSUPA user throughput performance can be improved by the proposed method even with small Iub bandwidth.

• 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.