• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.4
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• pp.957-966
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• 1998

In this paper an imrpoved Signal-to-Interference ratio(SIR)-based call admission control(CAC) algorithm for DS-CDMA cellular system is proposed and its performance is analyzed. This algorithm uses Residual-Capacity defined asthe additional number of initial calls that a base station can accept such that system-wide outage probability will guaranteed to remain below a certain level. the residual capcity at each cell is calculated according to the reverse-link SIR measured not only at the home cell but also the adjacent cells. Then the adjacent cell interference-coupling coefficient .betha. is used. In this work we propose an improved algorithm that .betha. varies according to the traffic load of the home cell. The influence of traffic condition on system performance, namely blocking probability and outage probability, is then examined via simulation. The performance of the improved algorithm is evaluated both under homogeneous and hot spot traffic loads. The results show that the improved algorithm outperforms conventional algorithms under all load values. Under over-load situation, especially, the improved algorithm gives almost constant outage performance the QoS(quality of service) can be guranted.

• The Journal of the Korea institute of electronic communication sciences
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• v.16 no.5
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• pp.825-834
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• 2021

With the development of various devices and Internet platforms, production and demand for content are increasing rapidly. However, effective response to exponentially increasing data usage is difficult in current TCP/IP schemes. As a result, with the emergence of an alternative called CCN, data can be transmitted and received from the host-centered to the content-oriented, and large-capacity content can be handled with more mitigated network resources. In this paper, we will address intermediate node mobility, which is one of the areas studied in CCN. In CCN, real-time services are a very important factor for information providers as well as requestors. However, when a node departure situation occurs, packet transmission and transmission speed decreases. Therefore, to ensure these numbers, even if the intermediate node deviates using a dual path, it provides a complete QoS (Quality of Service) using an alternative path.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2010.10a
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• pp.643-645
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• 2010

The Internet-based multimedia services such as IPTV is being expanded with the development of technology to support the convergence of broadcasting and telecommunications technology for the control seems to be growing larger. Especially for the real-time TV broadcast multicast control technology to support the authentication and resource control, in addition to the technology services that enhance the value of technology for a variety of services in both directions seems to be developed. And, Internet-based transmission system transmit the video content for the video content delivery using RTP(Real Time Transport Protocol). Standardization body, IETF(Internet Engineering Task Force) within the RTP, according to a variety of audio and video formats only transmission format(RTP Payload Format) Establish a separate standard and scalable video content "RTP Payload Format for SVC(Switched Virtual Connection) Video" the standardization is currently processing. In this paper we are improving the quality of broadcasting and telecommunication systems, so that the upper layer by the application can react adaptively to the existing MPEG2-TS and RTP who are provided by a variety of content applied to a variety of devices consumers ETE(End- to-End) QoS(Quality of Service) for enhance the system who was designed and implemented.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.3B
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• pp.72-84
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• 2005

Optical burst switching networks usually employ one-way reservation by sending a burst control packet with a specific offset time, before transmitting each data burst Same. Due to such a Property, burst-contentions occur when multiple bursts contend for the same wavelength in the same output link simultaneously in a node, leading to burst losses, eventually degrading the quality of service. Therefore, in this paper, we propose a dynamic routing algorithm using minimum local state information in order to decrease burst-contentions. In this proposed scheme, if burst loss rate exceeds a threshold value at a certain node, a new alternative routing path is chosen according to burst priority and location of burst generation, which enables the contending bursts to detour around the congested link. Moreover, for reducing the effect of sending bursts on the primary path due to the alternative path, we also apply a minimum path-cost based routing on link-cost concept. Our simulation results show that proposed scheme improves the network performance in terms of burst loss probability and throughput by comparing with conventional one.

• KNOM Review
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• v.22 no.2
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• pp.12-21
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• 2019

There is a rapid development of experimental equipment and ICT technology in data-intensive scientific areas, thus, big data of more than exabyte size is being generated. However, the big data transmission technology does not satisfy the needs of the application researchers who utilize it. Various high-performance transmission technologies have been developed based on QoS(Quality of Service), but they also require changes in the clean slate method. On the other hand, ScienceDMZ technologies improve the performance of scientific big data transmission by bypassing the firewall that causes a big problem in transmission performance. In addition, it is possible to implement without changing the existing network. In this paper, we built ScienceDMZ in an international long-distance environment based on KREONET(Korea Research Environment Open NETwork), and we verified the performance. We also introduced how GPU platform could be linked in a distributed ScienceDMZ environment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.7B
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• pp.1252-1260
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• 1999

The HFC network has a architecture of a star topology in fiber optic trunks, and tree and branch topology is used in the part of coaxial cable. It is well known that the HFC upstream channel is noisy. Ingress, common mode distortion and impulse noise exist in the upstream channel. In order to provide Quality of Service(QoS) to users with real-time data such as voice, video and interactive services, the evolving IEEE 802.14 standard for HFC networks must include an effective priority scheme. The scheme separates and resolves collisions between stations in a priority order. It is important to simulate protocols under a practical environment. The proposed algorithm in this paper is simulated with the assumption that the collision detector made certain mistake due to noises. Simulation results show that the proposed algorithm is more efficient than existing tree-based algorithm under practical environment.

• Journal of Internet Computing and Services
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• v.3 no.3
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• pp.67-74
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• 2002

Asynchronous transfer mode (ATM) can be efficiently used to transport packet data services. The switching system will support voice and packet data services simultaneously from end to end applications. To guarantee quality of service (QoS) of the offered services, source rate to send packet data is needed to control the network overload condition. Most existing control algorithms are shown to provide the threshold-based feedback control technique. However, real-time voice calls can be dynamically connected and released during data services in the network. If the feedback control information delays, quality of the serviced voice can be degraded due to a time delay between source and destination in the high speed link, An adaptive algorithm based on the optimal least mean square error technique is presented for the predictive feedback control technique. The algorithm attempts to predict a future buffer size from weight (slope) adaptation of unknown functions, which are used for feedback control. Simulation results are presented, which show the effectiveness of the algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.2B
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• pp.44-52
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• 2005

This paper proposes a neural network prediction model to improve the valid packet transmission rate for the QoS(Quality of Service) of multimedia transmission. The Round Trip Time(RTT) and Packet Loss Rate(PLR) are predicted using a neural network and then the transmission rate is decided based on the predicted RTT and the PLR. The suggested method will improve the transmission rate since it uses the rate control factors corresponding to time of data is being transmitted, while the conventional one uses the transmission rate determined based on the past informations. An experimental set-up has been established using a Linux PC system, and the multimedia data are transmitted using UDP protocol in real time. The valid transmitted packets are about 5% higher than the one in the conventional TCP-Friendly congestion control method when the suggested algorithm was applied.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.3B
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• pp.325-338
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• 2004

As the recent rapid development of internet technology and the wide spread of multimedia communication, 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 traffic control agent that can perform the dynamic resource allocation by controlling traffic flows on a DiffServ network. In addition, this paper presents a router that can support DiffServ on Linux to support selective QoS in IP network environment. To implement a method for selective traffic transmission based on priority on a DiffServ router, this paper changes the queuing discipline in Linux, and presents the traffic control agent so that it can efficiently control routers, efficiently allocates network resources according to service requests, and relocate resources in response to state changes of the network. Particularly for the efficient processing of Assured Forwarding(AF) Per Hop Behavior(PHB), this paper proposes an ACWF$^2$Q$^{+}$ packet scheduler on a DiffServ router to enhance the throughput of packet transmission and the fairness of traffic services.s.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.6B
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• pp.997-1003
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• 1999

It is required that synchronization model can describe various multimedia objects flexibly in the view point of time relationship and also respond quality of service requirement about jitter which is time difference between intramedia and skew which is time difference between intermedia. es desirable quality of service requirement. Proposed model applies maximum jitter and skew values which can be allowed, and then it presents high quality of service and real applies maximum jitter and skew values which can be allowed, and then it presents high quality of service and real time characteristics. In this paper we expend Petri-net and propose new synchronization specification model and apply two analysis method of Petri-net to prove oIn this paper we suggest multimedia synchronization model that is based on the Petri-net and servicur model suggested. We compared with other models and showed high QoS.