• The Transactions of the Korea Information Processing Society
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• v.6 no.9
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• pp.2512-2523
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• 1999

In this paper, we present the design and implementation results of RTP(Real-time Transport Protocol) which is introduced to provide end-to-end delivery service for multimedia data with real-time characteristics and to provide QoS(quality of service) monitoring services. Conventional RTP communication modules have some problems such as poor reusability and inefficiency. Reusability problem stems from the inherent characteristics of RTP that is framework is deliberately not complete, and so, RTP should have been designed and implemented in application-imbedded from. The RTP communication module, proposed in this paper, is designed and implemented in application-independent from in order to be used in all kinds of high level applications independent of their architectures and functions.

• Journal of the Korea Institute of Military Science and Technology
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• v.16 no.2
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• pp.160-168
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• 2013

In tactical environment, tactical wireless networks are generally comprised of Tactical MANETs(T-MANETs) or Tactical WMNs(T-WMNs). The most important services in tactical network are voice and low rate data such as command control and situation awareness. These data should be forwarded via multi-hop in tactical wireless networks. Urgent and mission-critical data should be protected in this environment, so QoS(Quality of Service) must be guaranteed for specific type of traffic for satisfying the requirement of a user. In IEEE 802.11s, TDMA-based MAC protocol, MCCA(MCF Controlled Channel Access), has a function of resource reservation. But 802.11s protocol can not guarantee the end-to-end QoS, because it only supports reservation with neighbors. In this paper, we propose the routing protocol, R-HWMP(Reservation-based HWMP) which has the resource reservation to support the end-to-end QoS. The proposed protocol can reserve the channel slots and find optimal path in T-WMNs. We analyzed the performance of the proposed protocol and showed that end-to-end QoS is guaranteed using NS-2 simulation.

• Journal of Internet Computing and Services
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• v.8 no.1
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• pp.25-31
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• 2007

The development of efficient Quality of Service (QoS) routing algorithms in high speed networks is very difficult since divergent services require various quality conditions, If the QoS parameter we concern is to measure the delay on that link, then the routing algorithm obtains the Least Delay (LD) path, Meanwhile, if the parameter is to measure of the link cast, then it calculates the Least Cost (LC) path. The Delay Constrained Least Cast (DCLC) path problem of the mixed issues on LD and LC has been shown to be NP-hard. The path cost of LD path is relatively mere expensive than that of LC path, and the path delay of LC path is relatively higher than that of LD path in DCLC problem. In this paper. we propose the algorithm based on estimated path for the DCLC problem and investigate its performance, It employs a new parameter which is probabilistic combination of cost and delay, We have performed empirical evaluation that compares our proposed algorithm with the DCUR in various network situations.

• Proceedings of the Korean Society of Computer Information Conference
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• 2008.06a
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• pp.55-62
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• 2008

최근 네트워크 고도화로 유무선 통합된 협업 서비스를 고객에게 제공한다. 서비스 제공자의 다양한 네트워크 서비스 상품이 앞으로 추가 출시될 예정이며 이에 따른 서비스 품질은 매우 중요한 이슈가 될 것이다. 특히 멀티미디어 서비스는 프리미엄 서비스로 서비스 제공자는 고객에게 서비스 품질을 보증해야만 된다. 초고속 인터넷 서비스 상품으로 인터넷 전화기, 인터넷 다운 속도 등 다양한 인터넷 서비스 품질 측정을 제공하고 있으나 전용선에 대한 인터넷 실시간 멀티미디어 서비스는 서비스 품질 측정이 초기 상태이다. 따라서 본 논문에서는 멀티미디어 서비스를 위한 서비스 품질 측정 지표, 종단간 서비스 품질 측정 방안, 고객에게 원활한 서비스를 제공하기 위한 에이전트 시스템 설계, 서비스 제공자가 이미 보유한 NMS/EMS에 에이전트 시스템 연동 방안을 제안한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38B no.5
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• pp.344-354
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• 2013

In this paper, In order to solve the traffic of the SNS to be executed on smartphone, we propose the design of the SNS integrated gateway server. Also, we capture the payload size in a variety of methods to send the MQTT protocol and QoS level of MQTT protocol for communication between the client and the Gateway. And we present the results and analysis of packet loss and end-to-end delay. By using analysis proposed, we present the results of the integrated SNS Gateway push notification protocol design.

• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• 2004.04a
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• pp.43-48
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• 2004

RTP that is proposed supplement of real-time services on internet environment, as Real-time Transport Protocol, is the protocol that for the purpose of sending data of stream type. RTP and RTCP(Real-time Transport Control Protocol) basically work at the same time, RTCP serves with state information of network at present. RTP has important properties of a transport protocol that runs on end-to-end systems and provides demultiplexing. It also offer reliability and protocol-defined flow／congestion control that transport protocol like TCP can not provides. In this paper, we look around concept and construction of Differentiated sen1ice tint run on RTP and by setting parameters of packet transfer method be used CBQ(Class-Based Queuing) for packet transfer on Differentiated service, each service queue controls properly through packet scheduling method, such as WRR(Weighted Round Robin) and PRR(Packet-by-packet Round Robin) all service classes do not experience the starvation and confirm the performance through computer simulation to achieve fairly scheduling.

• 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.

• 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.

• Journal of the Korea Society of Computer and Information
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• v.12 no.1 s.45
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• pp.137-144
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• 2007

Advances in networking technology digital media, and codecs have made it possible for the Internet evolves into a Broadband convergence Network (BcN) and provides various services including Voice over Internet Protocol (VoIP) and IPTV over their high-speed IP networks. In order for the Internet to make a profit as traditional Public Switched Telephone Network (PSTN), it must provide high qualify VoIP services. Therefore, real time qualify measurement framework is the most important requisite to provide VoIP service. For this, IETF (Internet Engineering Task Force) defined RTCP-Extended Reports (RTCP-XR) that extend RTCP (Real-Time Transport Protocol Control Protocol). However, procedure and method tot actually VoIP qualify measurement did not recommended nothing but defined item to measure voice quality. Our objective in this paper is to describes a practical measuring framework for end-to-end QoS of switched voice packet in an IP environment. It includes concepts as well as step-by-step procedures for measuring packetized voice streams. It also proposes new formats that extend RTCP-XR's concept.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.3B
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• pp.274-287
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• 2009

In the Internet environment in which best-effort services are provided, we need to guarantee end-to-end performance rather than hop performance to provide services with a variety of QoS(Quality of Service) requirements. The end-to-end performance is affected by many factors along the path of traffic flow. Most existing simulation studies are concentrated on a single node or a few nodes. We need much different approach considering the simulation execution time to simulate a large-scale network. In this paper, we derive requirements and present methodologies for the implementation of the simulator to simulate it. Then, we design and implement our simulator using OPNET. Performance evaluation is carried out using the simulator for large-scale KII (National Information Infrastructure in Korea). We use a self-similar traffic model and present some results on the end-to-end performance metric for the networks.