• The KIPS Transactions:PartC
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• v.8C no.4
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• pp.479-488
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• 2001

디지털 오디오나 비디오와 같은 멀티미디어 트래픽 QoS 제약 조건(종단간 지연, 대역폭 가용성, 패킷 분실률, 지터 등)을 요구하고 있다. 이러한 멀티미디어 트래픽의 제약 조건을 보장하기 위해서는 데이터 전송을 하기 전에 실시간 채널을 설정할 필요가 있으며 그러한 채널의 설정은 QoS 제약 조건을 고려한 효율적인 경로 설정 알고리즘을 필요로 한다. 일반적으로 네트워크에서 지연 제약 조건과 최소 비용을 만족하는 경로를 찾는 것은 NP-complete임이 증명되었다. 따라서 본 논문에서는 간단하고 분산된 루프 프리 라우팅 알고리즘(Loop-free Delay-constrained Routing algorithm with Load balancing : DRL)을 제안한다. 제안한 DRL은 지연 제약 조건을 만족하는 경로를 선택하며 네트워크의 상태에 따라 부하를 분산시키는 기능을 갖고 있다. 또한 지연 제약 라우팅을 위해 각 노드에서는 한정된 네트워크 상태 정보만을 필요로 한다. 시뮬레이션 결과는 DRL이 LDP와 마찬가지로 지연 제약조건을 만족하는 경로를 제공하면서 DCUR에 비해 루프가 발생하지 않는 경로 설정을 수행한다는 것을 보여주고 있다.

• Journal of KIISE
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• v.42 no.11
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• pp.1339-1348
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• 2015

In a distributed system environment based on a service-oriented architecture, separate systems collaborate to achieve the goals of the entire system by using services provided other systems. A service quality violation from using one service can cause runtime system failure in the environment. The existing self-adaptation methods follow fault tolerance mechanism that responds to a failure after a service quality violation. In other words, these methods are limited to responsive action. Therefore, a service-dependability-case based self-adaptation mechanism is necessary to preserve the dependability of the self-adaptive system. This paper demonstrates that the service-dependability-case based self-adaptation mechanism is better than QoS(quality of service)-based self-adaptation with fault tolerance to preserve the dependability of the self-adaptive system. Additionally, this paper suggests a method to present and analyze service dependability by using GSN(Goal Structuring Notation) which is the existing modeling method for the presentation of assurance cases, an action mechanism adapted using an analysis result of service-dependability-cases, a methods of leveraging the service-dependability-case based self-adaptation mechanism by following the service's life cycle, and the framework architecture including the major components and the interactions between the components in the control loop of the self-adaptation process.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.9A
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• pp.1285-1294
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• 1999

In this paper we consider the performance of TCP video traffic over ABR with Long-Range Dependent VBR traffic. As compressed coded video traffics are increasing rapidly over Internet, lots of studies are being done for transmitting those traffics efficiently using limited network resources. We consider here the transmitting video service over ABR service in ATM networks, especially satellite networks. CBR or VBR services are suggested in transmitting the video traffic in ATM Forum TM 4.0. But ABR service connection, which is considered as appropriate service for data traffic, can be established with a small amount of bandwidth, MCR (Minimum cell rate). Furthermore ABR service can control the source's transmitting rate using feedback mechanism. Using this feature ABR service can be used in some applications which can control their quality of services corresponding to network loads. Compressed video sources with MPEG-2 are used for Long-Range Dependent VBR traffic here. We model the compressed video source to resemble the MPEG-2 transport streams. These compressed video traffic streams are consisted of three different frames, I-frame, P-frame, and B-frame. So when a network are overloaded, we can control the quality of service using this traffic features. TCP Traffics over ABR need large buffers in ATM switch to satisfy their QoS with background VBR traffics, which have high deviations in bandwidth. Furthermore satellite ATM networks with large feedback delay need large buffers corresponding RTT delay. The performance comparisons among EFCI and ER switch (ERICA+) switches in the network circumstances described above were shown in this paper. We also considered the case with ON-OFF VBR traffics.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.6
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• pp.203-210
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• 2013

Majority of the wireless ad hoc routing protocols are proposed to find feasible routes without considering the network load, end-to-end link quality and bandwidth requirements of the application. Therefore, protocol may not provide sufficient quality of service (QoS) to a high speed traffic such as multimedia. In this paper, we propose a path-quality and bandwidth-estimation based routing protocol (PBBR) for the high quality multimedia stream that can meet the application's bandwidth requirements and find the best reliable route. The novelty of this protocol is to select a reliable path to respond the application's requirements based on available bandwidth at each intermediate node and end-to-end path loss ratio. Obtained results from the simulation demonstrates that our protocol can achieve sufficient performance in terms of throughput and end-to-end delay.

• The Transactions of the Korea Information Processing Society
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• v.4 no.7
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• pp.1770-1780
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• 1997

Our problem is improvement of image quality because it is inevitable cell loss of image data when traffic congestion occurs. If cells are discarded indiscriminately in transmission of MPEG video data, it occurs severe degradation in quality of service(QOS). In this paper, to solve this problem, we propose two method. The first, we analyze the traffic characteristics of an MPEG encoder and generate high priority and low priority data stream. During network congestion, only the least low priority cells are dropped, and this ensures that the high priority cells are successfully transmitted, which, in turn, guarantees satisfactory QoS. In this case, the prioritization scheme for the encoder assigns components of the data stream to each priority level based on the value of a parameter ${\beta}$. The second, Number of high priority cells are increased when value of ${\beta}$ is large. It occurs the loss of high priority cell in the congestion. To prevent it, this paper is regulated to data stream rate as buffer occupancy with UPC controller. Therefore, encoder's bandwidth can be calculated renegotiation of the encoder and networks. In this paper, the encoder's bandwidth requirements are characterized by a usage parameter control (UPC) set consisting of peak rate, burstness, and sustained rate. An adaptive encoder rate control algorithm at the Networks Interface Card(NIC) computes the necessary UPC parameter to maintain the user specified quality of service. Simulation results are given for a rate-controlled VBR video encoder operating through an ATM network interface which supports dynamic UPC. These results show that dynamic bandwidth renegotiation of prioritized data stream could provided bandwidth saving and significant quality gains which guarantee high priority data stream.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.12C
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• pp.1251-1264
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• 2002

Currently, audio streaming services such as on-demand service and live service support only a limited number of clients. They also suffer from a lack of stability and degradation of service quality due to their inefficient use of network resources. Futhermore, since the streaming services usually do not consider dynamic services, they are very inconvenience to use. In order to resolve these drawbacks, we propose a novel audio streaming system based on MPEG-2 AAC file format which are facilitated with the network bandwidths efficiently. The proposed system supports QoS for audio streaming as well as guarantees a stability while streaming service is undergoing. Moreover, the system provides a dynamic interface which enables us to use the streaming service more easily and to manage streaming servers with convenient manner. On the contrary, most of the current available static interface streaming services are mainly depending only on a single fixed web page between client and server, which in consequence lead us to use unflexible static service environment. Our implementation results show the proposed system improves the performance compared to those of the currently existing systems that use MP3 file format. It also provides some benefits such as a stability of service and a easy to management of streaming servers.

• The Transactions of the Korean Institute of Electrical Engineers P
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• v.64 no.4
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• pp.321-326
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• 2015

Smart metering is one of the most implementable internet-of-thing service. In order to implement the smart metering, a wireless communication network should be newly designed and evaluated so as to satisfy quality-of-service of smart metering. In this paper, we consider a wireless network for the smart metering implemented with multi-purpose wireless chips and channel coding-functioned micro controllers. Especially, channel coding is newly adopted to improve successful frame transmission probability. Based on the successful frame transmission probability, average transmission delay and delay violation probability are analyzed. Using the analytical results, service coverage expansion is evaluated. Through the delay analysis, service feasibility can be verified. According to our results, channel coding needs not to be utilized to improve the delay performance if the smart metering service coverage is several tens of meters. However, if more coverage is required, chanel coding adoption definitely reduces the delay time and improve the service feasibility.

• The Journal of the Korea institute of electronic communication sciences
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• v.6 no.3
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• pp.433-438
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• 2011

The purpose of this paper is to define WiFi LDT(Location Determination Technology) and Service available for mobile wireless network. This paper introduces positioning technology such as Basic Technology Element and QoS(Quality of Service) etc. of WPS(WiFi Positioning System) for mobile wireless network. The LDT and wireless positioning technology in order to determine the position of terminal when mobile based positioning service is provided, and by providing service with postioning technology, it will not only provide convenience to the users or subscribers but also contribute to the activation of LBS(Location Based Services) industries.

• Journal of Industrial Technology
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• v.20 no.B
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• pp.139-146
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• 2000

To support Quality of Service (QoS) in the existing Internet, Differentiated Service (Diff-Serv) has been proposed. But, the unfairness between TCP connections remains as a serious problem not only in the conventional best-effort service Internet but also in new Diff-Serv network. In this paper, we propose the Balancing Marker Algorithm (BMA) improving the fairness between individual connections of aggregated sources in a Diff-Serv network. This algorithm is based on the 3-level priority marking method. We compared the 2-level packet priority marker with the Balancing Marker proposed in this paper. And we showed that the BMA improved the fairness and the throughputs between the individual connections with different delays in an aggregated source.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.6 no.9
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• pp.2285-2301
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• 2012

Peer-to-Peer (P2P) networks are becoming more and more popular in video content delivery services, such as Video on Demand (VoD). Scalability feature of P2P allows a higher number of simultaneous users at a given server load and bandwidth to use stream service. However, the quality of service (QoS) in these networks is difficult to be guaranteed because of the free-riding problem that nodes download the recourses while never uploading recourses, which degrades the performance of P2P VoD networks. In this paper, a distributed trust model is designed to reduce node's free-riding phenomenon in P2P VoD networks. In this model, the P2P network is abstracted to be a super node hierarchical structure to monitor the reputation of nodes. In order to calculate the reputation of nodes, the Hidden Markov Model (HMM) is introduced in this paper. Besides, a distinction algorithm is proposed to distinguish the free-riders and malicious nodes. The free-riders are the nodes which have a low frequency to free-ride. And the malicious nodes have a high frequency to free-ride. The distinction algorithm takes different measures to response to the request of these two kinds of free-riders. The simulation results demonstrate that this proposed trust model can improve QoS effectively in P2P VoD networks.