• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.10A
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• pp.1512-1520
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• 2000

The wireless communication network is evolving toward IMT-2000 for providing various multimedia services. In order to accomplish this ultimate goal the effective schemes are required which can dynamically utilize the limited wireless resources based on different traffic characteristics of various services. This paper proposes a novel bandwidth allocation and call admission control scheme to transmit multimedia traffic based on the bandwidth reservation procedure using direction estimation in the IMT-2000 This scheme estimates the position of mobiles based on the mliticriteria decision making in which uncertain parameters such as RSS(Received Signal Strength), the distance between mobile and base station the moving direction and the previous location are participated in the decision process using aggregation function in fuzzy set theory. Its effectiveness is investigated by simulation.

• Proceedings of the Korea Information Processing Society Conference
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• 2014.11a
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• pp.203-206
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• 2014

무선 LAN(Wireless Local Area Networks 의 DCF(Distributed Coordination Function) 방식은 랜덤 백 오프 방식으로 매체에 접근하기 때문에 지연이 발생하여 정해진 시간 내에 전송을 보장할 수 없다는 단점이 있다. [1] 이는 곧 실시간 멀티미디어 트래픽(비디오, 음성 등)의 QoS(Quality Of Service)를 보장할 수 없다는 것을 뜻한다. 또한 IEEF 802.11e 표준 [2]에서 제공하는 QoS 를 위한 EDCA(Enhanced Distributed Channel Access)라는 프로토콜은 제시되어있으나 실제로 구현되어있는 디바이스의 부재로 QoS 를 지원하기가 어렵다. 따라서 무선 랜에서 IEEE 802.11e 를 지원하지 않는 망내 디바이스, 즉 큐가 1 개인 STA, 즉 기본적인 802.11 표준 기술인 DCF 를 사용하는 STA 을 위해서 멀티미디어 트래픽의 실시간 전송을 보장하기 위한 기법을 제시한다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2014.05a
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• pp.469-472
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• 2014

S-MAC is hybrids of CSMA and TDMA approaches that use local sleep-wake schedules to coordinate packet exchanges and reduce idle listening. In this method, all the nodes are considered with equal priority which may lead to increased delay during heavy traffic. The method introduced in this paper provides high throughput and small end-to-end delay suitable for applications such as real-time voice streaming and its functionality is independent of underlying synchronization protocol. The novel idea behind our scheme is that it uses the priority concept with (m,k)-firm scheduling in order to achieve its objectives. The performance of our scheme is obtained through simulations for various packet sizes, traffic loads which show significant improvements in packet delivery ratio, and delay compared to existing protocols.

• The Journal of the Korea institute of electronic communication sciences
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• v.16 no.6
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• pp.1069-1074
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• 2021

Recently, mobile communication technology has new eras with supply of 5G commercial communication. 5G mobile communication service is currently supplied with city area, services will be quickly expanded to entire area of country. This mmWave based 5G mobile communication is under spreading for human communication with voice and Internet service. After completion of construction of this human communication, this technology will be expanded to industrial communications. Railway communication system is an example of this industrial communications. In this, performance of traffic transmission for railway communication network based on this 5G railway communication networks will be analyzed with computer simulations. Construction requirements of 5G railway communication networks will be suggested with this analysis results.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.3B
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• pp.398-407
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• 2010

Due to increase of user requirement for various traffics and the advance of network technology, each distinct network has converge into FMC(Fixed Mobile Convergence) networks. However, we need to research the performance analysis of VoIP(Voice over Internet Protocol) in the FMC network to provide QoE for the voice user of FMC network. Therefore, this paper introduces the scenario which is the situation of voice quality degradation when a user uses VoIP to communicate with other users in the FMC network. Especially, this paper presents scenario in terms of the component of the network and finds the improvement point of voice quality. In the simulation results, three improvement points of voice quality are found as following: voice quality degradation by packet loss in the physical layer of the HSDPA network, by utilizing GGSN without QoS parameter mapping mechanism which is gateway between 3GPP and IP backbone, and by using non-QoS AP in the WLAN network.

• Journal of Korea Multimedia Society
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• v.6 no.7
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• pp.1266-1276
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• 2003

Multimedia information containing voice and image is transmitted on Internet, which is ten times or hundred times larger than ordinary information. Analysis types for network management in this environment consist of a real time analysis, a basic analysis and an intensive analysis. The intensive analysis is useful for gathering the trend information of specific objects periodically for certain period in order to monitor network status. When SNMP is applied to collect the trend information of intensive analysis, it brings on the increase of network load, the delay of response time and the decrease of data collection accuracy since an agent responds to manager's every polling. In this paper, an efficient SNMP is proposed and implemented to add time variables in the existing SNMP PDU. It minimizes unnecessary traffic in the intensive analysis between manager and agent, and collects trend information more accurately. The results of experiments show that it has compatibility with the existing SNMP, decreases the amount of network traffic greatly and increases the accuracy of data collection.

• The Transactions of the Korea Information Processing Society
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• v.4 no.1
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• pp.224-236
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• 1997

In order to control the flow of traffic in ATM networks and optimize the usage of network resources, an appropriate control mechanism is necessary to cope with congestion or prevent degradation of network performance caused by congestion. While the conventional UPC algorithm provides only unstable preventive function unrelated with the state of networks and is unable to recover congestion, the proposed adaptive UPC algorithm supervises the situation of ATM networks using the information from OAM cell. Then the monitor of the proposed adaptive UPC algorithm controls leaky rate and buffer threshold value according to QOS. Therefore, the proposed algorithm can cope with congestion as well as prevent and react sensitively to buffer delay. In proportion to the diversity of traffic and the increase of transmission rate in networks, the adaptive UPC-BT algorithm proposed in this paper can be effectively used in ATM networks with wide applications. This paper shows that the proposed algorithm efficiently uses in ATM networks with application. This paper shows that the proposed algorithm efficiently use network resources and provides QOS to users for various kinds of traffics by comparing with conventional UPC algorithms.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.2
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• pp.228-238
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• 2003

We believe that existing wire solutions such as HomePNA2.0 using phone lines and HomePlug using power line and wireless solution such as HomeRF are the most promising solutions, because of its cost-effectiveness. However, MAC protocols of these solutions provide only Class of Service(CoS) using priority mechanism like HomePNA and HomePlug or consider only voice among real-time traffics like HomeRF. For these reasons, we perceive the needs of the new MAC protocol which is no new wire solution and provides guaranteed Quality of Service (QoS) for not only voice but also video and audio. In light of this, we present the design and software implementation of a new MAC protocol for Home Networks. Our evaluation results of software implementation verify that proposed MAC protocol can provide low delay, low loss, and low jitter to real-time traffic by reserving bandwidth.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.2
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• pp.381-389
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• 1994

In the communication network, multimedia service such as high quality voice, high speed data, image etc. will be added to the existing service. This service generates new requirements for the communication networks. The priority control mechanism can be used to control multimedia traffics generated by many communication systems. The priority mechanism which assigns prioirities to generated cells according to service quality is one of the traffic control. The priority assignment can be divided by priority criterion for each traffic characteristics such as loss sensitivity and delay sensitivity. In this paper, we alnalyzed the partial buffur sharing (PBS) mechani느 as a traffic control reducing the cell loss, and proposed analysis method. We analyzed the PBS mechanism using classical approach as a Markov chain. In order to validata proposed analysis method, simulation is performed using simulation package SIMSCRIPT 11.5. From this results, we confirmed that proposed analysis method can be verified. Also, we presented cell loss probability of ATM network when this results are to be applied to ATM networks.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.2
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• pp.535-538
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• 2005

Wireless LANs based on the IEEE 802.11 standard are widely spread for use nowadays. Traffic which are conveyed over the WLANs change rapidly from normal data such a Email and Web pages, to multimedia data of high resolution video and voice. To meet QoS (Quality of Service) required by these multimedia traffic, the IEEE 802 committee recently has developed a new standard, IEEE 802.11e. IEEe 802.11.e contains two MAC mechanisms for providing QoS: EDCA(Enhanced Distributed Channel Access) and HCCA (HCF Controlled Channel Access). Using these standardized MAC mechanisms as a building platform, various admission control mechanisms can be combined to offer QoS gurantees for multimedia traffic. This paper surveys these research efforts.