• The KIPS Transactions:PartB
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• v.12B no.2 s.98
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• pp.167-172
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• 2005

A rapid growth in the number of Internet users and in massive media contents tends to cause server overload and high network traffic. As a consequence, it can be safely assumed that the quality of service is on the brink of deterioration. A solution of this problem is the development of the Content Distribution Network. Therefore, in this paper, we propose a transmission model using CDN. This model uses regional media server which closes to client, and uses MDC coding and multi-channel to reduce packet loss and delay. In conclusion, this study improved performance of multimedia service by reducing packet loss and delay.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.3A
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• pp.254-266
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• 2006

Recently, researches about downlink resource allocation algorithms applying SDMA to enhance the system throughput and cell coverage have begun. Most OFDM/SDMA based resource allocation algorithms have some limitations such that those only concentrate on maximizing the system throughput or can be applied in single cell environment. In this paper, we propose an OFDM/SDMA based downlink resource allocation algorithm which considers high layer QoS parameters suitable for the required data traffic and it also minimizes the system throughput loss and considers inter-cell interference from adjacent cells. so it can be adopted in multi-cell environment. We manifest the performance of the proposed algorithm in Ped A and SCME MIMO Channel Model.

• 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.

• Journal of Advanced Navigation Technology
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• v.16 no.1
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• pp.96-102
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• 2012

The goal of mobile internet system is to provide a high-data-rate, low-latency and optimized packet radio access technology supporting flexible bandwidth deployments. Therefore, network architecture is designed with the goal to support packet-switched traffic with seamless mobility, quality of service and minimal latency. An important requirement for the mobile internet system is improved cell-edge BER performance and data throughput. This is to provide some level of service consistency in terms of geographical coverage as well as in terms of available data throughput within the communication coverage area. In a cellular system, however, the signal to interference plus noise power ratio gap between cell-center and cell-edge users can be of the order of 20 [dB]. The disparity can be even higher in a communication coverage limited cellular system. This leads to vastly lower data throughputs for the cell-edge users relative to cell-center users creating a large QoS gap. This paper proposes a analytical approach that tries to reduce inter-cell interference, and shows the SIR and BER performance according to the OFDM system parameters in mobile Internet environment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.9A
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• pp.1013-1021
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• 2004

In the WCDMA mobile multimedia communication system, the adaptive array antenna IS adopted to improve the performance of the system by reducing inter-user interference using antenna beam control. Usually, the interference resulted from the higher data rate users is much more significant to the lower data rate users than the other way around, so the overall performance can be enhanced by reducing the interference from higher data rate users. In order to maximize the efficiency of adaptive antenna operation, an optimal call admission control, especially during handoff, adaptive to the data rates is a critical problem. In this paper, We propose a call admission control algorithm based on the Soft QoS concept for the efficient processing of the handoff of higher data rate calls, and an adaptive handoff control mechanism according to the data rates. The proposed algorithm has been evaluated by computer simulation that it accommodates high data rate users among many lower data rate users much better, and the average call blocking probability for lower rate users becomes much lower than the conventional call admission control algorithm.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.14 no.6
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• pp.2464-2479
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• 2020

This paper presents a detailed subjective quality assessment for the ultra-high definition (UHD) videos having frame rates of 30fps and 60 fps. The subjective assessment is based on the ITU-R BT-500 recommendations, where double stimulus continuous quality scale (DSCQS-type II) test is performed for the evaluation of the perceived quality of the user's in terms of differential mean opinion score (DMOS). Encoding of the UHD videos by opting encoders i.e. H.264/AVC, H.265/HEVC, and VP9 at five different quantization parameter (QP) levels is done to investigate the perceived user's quality of experience (QoE) given as DMOS. Moreover, the encoding efficiency as the encoding time for each encoder and qualitative performance by employing full-reference (FR) quality metrics are presented in this work.

• The Transactions of the Korea Information Processing Society
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• v.3 no.6
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• pp.1386-1396
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• 1996

In the distributed multimedia application, high throughput and low delay is one of the most important QoS(Quality of Service) requirement. Emerging high speed communication offers transmission rate above Gbps, but it can not be utilized efficiently by the performance bottlenecks of communication protocols. To overcom the problem, we propose a horizontal processing architecture that processes data as soon as it arrive from the network. Unfortunately, the OSI protocol stack often imposes ordering constraints that prevent concurrent processing of the protocol layers. By using a fixed packet format, the network layer through the presentation layer are processed in parallel. Our prototype shows that the proposed model has performance improvement up to 61% more advantage than the conventional approach.

• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.601-606
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• 2008

When sharing real-time media in the ubiquitous computing environments, due to heterogeneous performance of devices, conflict to users' media service requirements might be occurred. To address the problems, there are extensive researches about media streaming QoS control schemes in the aspects of network or application. However, their deployment has met with difficulty because of critical reasons such as high development cost and system complexity. In this paper, in the level of media services, we propose a negotiation approach to offer improved quality of media services. The proposed approach Constructs video distribution group between producer and consumer services, which target to globally minimize the concession of users' quality requirements Consequently, users can be provided video services in the level of conformation to the users' expectation.

• The Transactions of the Korea Information Processing Society
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• v.7 no.11S
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• pp.3723-3731
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• 2000

Recently, the growth of applications and services over high-speed Internet increase, ATM networks as wide area back-bone has been a major solution. The conventional TCP suite is still the standard protocol used to support upper application on current Internet and uses a window based protocol for flow control in the transport layer. When TCP data uses the UBR service in ATM layer, the control method is also buffer management. If a cell is discarded in ATM layer. one whole packet of TCP will be lost. Which is responsible for most TCP performance degradation and do not offer sufficiently QoS. To solve this problem, Several dropping strategies, such as Tail Drop, EPD, PPO, SPD, FBA, have been proposed to improve the TCP performance over ATM. In this paper, to improve the TCP fairness of end to end, we propose a packet dropping scheme algorithm using two fixed threshold. Under similar condition, we compared our proposed scheme with other dropping strategies. Although the number of VC is increased, simulation results showed that the proposed scheme can allocate more fairly each VC than other schemes.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.7A
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• pp.659-665
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• 2006

When using the Mobile IP which is the representative technology to secure the mobility in general IP networks, the packet loss during the handover is inescapable. To remedy the packet loss problem, the smooth handover was introduced. However, the smooth handover causes the packets sequence disruption during the packet forwarding procedure and it may result in the degradation of the network performance. The same problem also occurs in the WiBro (High-speed Portable Internet) system that is the next generation portable IP service system. The WiBro system, which provides the high speed data service just like xDSL and leased line in wired internet, aims to guarantee the portability, mobility, and the differentiated service based on IEEE 802.16. So, the handover mechanisms that solve the problems of packet loss and packet sequence distribution are required in the WiBro system. In this paper, we propose the handover mechanism and the packet sequence control algorithm that provide the reliability and the differentiated service for the unicast service in the WiBro system.