• The Transactions of The Korean Institute of Electrical Engineers
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• v.62 no.6
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• pp.744-749
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• 2013

The protection switching technology is widely adopted in the fiber-optical transmission equipments based on TDM(Time Division Multiplexing), such as PDH, SDH/SONET. A variety of protection switching algorithms for Ethernet networks and the progress of standardization are summarized in the document. There are several kinds of protection switching algorithms for Ethernet networks, such as STP, RSTP, MSTP and etc. However, since Ethernet signal move through detour route, it causes much time to recover. Accordingly, it is difficult to secure a usability of Ethernet networks and QOS(Quality of Service). Also, if the protection switching protocol standardized by IEEE and ITU-T is used, it remains a inherent network switching time for protection. Therefore, a specific protection switching algorithm for Ethernet are needed for seamless and stable operation of Ethernet networks service for Distribution Automation System(DAS). A reliable protection algorithm with no switching delay time is very important to implement Self-healing service for DAS. This study of FPGA based protection switching algorithm for Ethernet networks shows that in case of faults occurrence on distribution power network, immediate fault isolation and restoration are conducted through interaction with distribution equipments using P2P(Peer to Peer) communication for protection coordination. It is concluded that FPGA based protection switching algorithm for Ethernet networks available 0ms switching time is crucial technology to secure reliability of DAS.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.8B
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• pp.751-759
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• 2009

Due to the advance of WLAN technology, the use of the multimedia service such as the video streaming service has been increased in the home network. However, we need to study the method which decreases the transmission delay and the frame loss rate to provide QoS of the video streaming service. Therefore, this paper proposes an active buffer management mechanism to guarantee QoS of the streaming service in IEEE 802.11e EDCA. The proposed protocol discards the frame in the HoL of the buffer based on the importance of each frame and the virtual transmission delay of frame newly arriving at the buffer. In the simulation results, the proposed algorithm not only decreases the frame loss probability of important I and P frames but also stabilizes the transmission delay. It may increase the QoS of video streaming services.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.8B
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• pp.707-715
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• 2003

In this paper, we propose a new MAC(Medium Access Control) protocol using Movable-boundary, which tries to guarantee Qos for multimedia traffic in the slotted CDMA system. In this scheme, the traffic scheduler assigns channel resource according to the packet priority per service class and adapts the Movable-boundary concept in which the minimum resource is assigned to each traffic class; the remaining resource if it is available can be assigned dynamically according to the temporal demand of other traffic classes. For performance analysis, we performed computer simulations to obtain throughput and packet loss rate and compared the results with Fixed-boundary system. We observed that the error rate of voice traffic could be maintained below a prescribed value while bursty traffic such as video source shares the same channel. In comparison with Fixed-boundary scheme, our protocol exhibits better throughput and packet loss rate performance.

• Journal of the Korean Institute of Intelligent Systems
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• v.15 no.5
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• pp.605-614
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• 2005

In this paper, we present the design and implementation of an agent knowledge and QoS tuning methodology to improve the flexibility of agent-based flexible video-conference system. In order to improve the flexibility during video-conferencing, we propose a new T-INTER(Tuning-INTER) architecture of knowledge part in video-conference manager (VCM) agent in which an automatic QoS parameter tuning method is imbedded. The flexible video-conference system structured based on the proposed architecture can cope with the changes in service quality required by users. The VCM agent cooperates with other agents by protocols and executes the automatic QoS parameter tuning task whenever needed. By the tuned parameters, the system is able to flexibly cope with the internal or external changes and the burden of users can be decreased. In the experimental section, it is shown that our proposed system outperforms the existing system.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.22 no.12
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• pp.1604-1610
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• 2018

The speech recognition service is used as an ancillary means of communication by converting and visualizing the speaker's voice into text to the hearing impaired persons. However, in open environments such as classrooms and conference rooms it is difficult to provide speech recognition service to many hearing impaired persons. For this, a method is needed to efficiently provide it according to the surrounding environment. In this paper, we propose a system that recognizes the speaker's voice and transmits the converted text to many hearing impaired persons as messages. The proposed system uses the MQTT protocol to deliver messages to many users at the same time. The end-to-end delay was measured to confirm the service delay of the proposed system according to the QoS level setting of the MQTT protocol. As a result of the measurement, the delay between the most reliable Qos level 2 and 0 is 111ms, confirming that it does not have a great influence on conversation recognition.

• Proceedings of the Korea Information Processing Society Conference
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• 2007.11a
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• pp.1014-1017
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• 2007

무선랜에서는 작은 셀 크기로 인해 노드들의 이동에 따른 빈번한 핸드오프가 이루어진다. 그래서 무선랜에서는 지속적인 통신 서비스를 제공하기 위한 방법으로 모바일 아이피와 같은 방법을 개발하고 있다. 모바일 아이피는 모바일 노드가 한 장소에서 다른 장소로 이동할 때 IP 주소의 변경 없이도 이동할 수 있도록 해준다. 그러나 모바일 아이피는 긴 시간의 등록과정과 연결 재설정 때문에 시간지연이나 패킷 손실과 같은 오버헤드를 발생시킨다. 따라서 무선랜의 QOS(Quality Of Service)를 향상시키기 위해서 Mobile IP 의 핸드오프 시간을 줄여야만 한다. 본 논문은 이러한 문제를 해결하기 위해 핸드오프 지연 시간을 감소시키는 기법을 제안한다. 제안기법의 기본 아이디어는 모바일 노드의 이동 네트워크를 예상하여, 미리 패킷 포워딩을 수행하는데 있다. 우리는 각 모바일 노드들에게 connection proxy table 정보를 추가하였다. 그리고 이 테이블 정보를 이용함으로써 모바일 노드들은 홈에이전트와 포린에이전트에 COA(Care of address)를 미리 등록하는 것이 가능해졌다. 그 결과로 이동 노드들은 홈 네트워크를 완전히 벗어나지 않고도 핸드오프가 이루어져 지속적인 서비스가 가능하였다. 본 논문에서 제안하는 기법이 기존 Mobile IP 와 비교해 볼 때 핸드오프가 이루어지는 동안 비 연결 시간을 줄일 수 있다는 것을 실험결과를 통해서 확인할 수 있었다.

• The KIPS Transactions:PartC
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• v.8C no.1
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• pp.60-67
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• 2001

Multicast Protocol for multimedia service on the Internet can be classified into two types, e.g., source based tree and shared tree according to difference of tree construction method. Shared tree based multicast is known to show outstanding results in the aspect of scalability than source based tree. Generally, There have been lots of researches on the method to satisfy QoS constraints through proper Rendezvous Point (RP) in the shared tree. In addition, as the multicast group members join and leave dynamically in the service time, RP of the shared tree should b be reselected for guranteeing Qos to new member, But, RP reselection method has not been considered generally as the solution to satisfy QoS C constraints. In this paper, new initial RP selection and RP reselection method are proposed, which utilize RTCP (Real Time Control Protocol) report packet fields. Proposed initial RP selection and RP reselection method use RTCP protocol which underlying multimedia application service So, the proposed method does not need any special process for collecting network information to calculate RP. New initial RP selection method s shows better performance than random and topology based one by 40-50% in simulation. Also, RP reselection method improves delay p performance by 50% after initial RP selection according to the member’s dynamicity.

• The KIPS Transactions:PartA
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• v.10A no.6
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• pp.691-700
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• 2003

Due to the fast development of wire&wireless internet and computer hardware, more and more internet services are being developed, such as Internet broadcast, VoD (Video On Demand), etc. So QoS (Qualify of Service) is essentially needed to guarantee the quality of these services. Traditional Internet is Best-Effort service in which all packets are transported in FIFO (First In First Out) style. However, FIFO is not suitable to guarantee the quality of some services, so more research in QoS router and QoS protocol are needed. Researched QoS router and protocol are high cost and inefficient because the existing infra is not used. To solve this problem, a new QoS control method, named Network Adaptive QoS, is introduced and applied to client/server-based streaming systems. Based on network bandwidth monitoring mechanism, network adaptive QoS control method can be used in wire&wireless networks to support QoS in real-time streaming system. In order to reduce application cost, the existing streaming service is used in NAQoS. A new module is integrated into the existing server and client. So the router and network line are not changed. By simulation in heavy traffic network conditions, we proved that stream cannot be seamless without network adaptive QoS method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.7B
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• pp.608-618
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• 2006

The increasing popularity of multimedia streaming services introduces new challenges in content distribution. Especially, it is important to provide the QoS guarantees as they are increasingly expected to support the multimedia applications. The service providers can improve the performance of multimedia streaming by caching the initial segment (prefix) of the popular streams at proxies near the requesting clients. The proxy can initiate transmission to the client while requesting the remainder of the stream from the server. In this paper, in order to apply the prefix caching service based on IETF's RTSP environment to the wireless networks, we propose the effective RTSP handling scheme that can adapt to the radio situation in wireless network and reduce the cutting phenomenon. Also, we propose the traffic based caching algorithm (TSLRU) to improve the performance of caching proxy. TSLRU classifies the traffic into three types, and improve the performance of caching proxy by reflecting the several elements such as traffic types, recency, frequency, object size when performing the replacement decision. In simulation, TSLRU and RTSP handling scheme performs better than the existing schemes in terms of byte hit rate, hit rate, startup latency, and throughput.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7B
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• pp.598-612
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• 2004

Client requests have increased with the improvement of network resources at client side, whereas network resources at server side could not keep pace with the increased client request. Therefore, it is primary factor of the Qos that efficiently utilize network resources at server side. In this paper, we proposed a new model that peer-to-peer transfer scheme for partial multimedia stream based on CIWP which it decrease server network bandwidth by utilizing client disk resources saves additional server network resources. Especially, adapting Threshold_Based Multicast scheme guarantees to do that data transfer within clients never exceed service time of previous peer by restriction of which data size transferring from previous peer less than data size transferring from server. Peer-to-peer transfer within clients is limited in same group classified as ISPs. Our analytical result shows that proposed scheme reduces appling network resources at server side as utilizing additional client disk resource. furthermore, we perform various simulation study demonstrating the performance gain through comparing delay time and proportion of waiting requesters. As a result, when we compared to Threshold_Based Multicast scheme, the proposed scheme reduces server network bandwidth by 35%.