• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.6
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• pp.1177-1186
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• 2003

The multimedia ad hoc wireless network is quite an attractive issue since it offers a flexible solution to enable delivery of multimedia services to mobile end users without fixed backbone networks. However, with the unique design challenges of ad hoc wireless networks, it is a non-trivial issue to provide bounded delay guarantee, with fair share of resources. In this paper, we implemented the delay guaranteed fair queueing (DGFQ) scheme distributively. Through the results of performance evaluation, we can conclude that DGFQ also performs well to control bounded delay in multimedia ad hoc wireless networks.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.4C
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• pp.308-315
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• 2002

Guaranteed frame rate (GFR) service has been defied to provide minimum cell rate (MCR) guarantees for virtual connections (VCs) carrying Internet traffic in ATM networks and allow them to fairly share residual bandwidth. The simplest switch implementation mechanism to support the GFR service in ATM networks consists of the frame-based generic cell rate algorithm (F-GCRA) frame classifier and the early packet discard (EPD)-like buffer acceptance algorithm in a single FIFO buffer. This mechanism is simple, but has foiled to guarantee the same bandwidth as an MCR to a VC that has reserved a relatively large MCR. This paper applies the packet spacing scheme to TCP traffic to alleviate its burstness, so as to guarantee a larger MCR to a VC. In addition, the random early detection (RED) scheme is added to the buffer acceptance algorithm in order to improve fairness in use of residual bandwidth. Simulation results show that the applied two schemes improve a quality of service (QoS) in the GFR service for the TCP traffic.

• The Transactions of the Korea Information Processing Society
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• v.5 no.11
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• pp.2943-2953
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• 1998

As communication networks become very fast, the scope of applications is varied and applications using multi-party communications in which more than one communication station participate, have been increased. In this paper, multicast servers are designed and implemented by two different schemes (centralized multicast server scheme and distributed multicast server scheme). Both scheme contain three operation for joining/leaving of a node, operation for transmitting the name of the group and messages to the server, and operation for a node to read messages and clear shared memory. Based on their implementations, two multicast server schemes are compared in terms of complexity, extendibility, efficiency and real-time aspects. As a result, the average transfer time of the distributed multicast server scheme is shorter than that of the centralized multicast server scheme. Finally, we designed the multi-party real-time communication method to guarantee QoS in multicast.

• Journal of the Institute of Convergence Signal Processing
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• v.7 no.3
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• pp.136-142
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• 2006

Existing buffer management schemes for shared-memory output queueing switches can be classified into two types: In the first type, some constant amount of memory space is guaranteed to each virtual queue using static queue thresholds. The static threshold method (ST) belongs to this type. On the other hand, the second type of approach tries to maximize the buffer utilization in 머 locating buffer memories. The complete sharing (CS) method is classified into this type. In the case of CS, it is very hard to protect regular traffic from mis-behaving traffic flows while in the case of ST the thresholds can not be adjusted according to varying traffic conditions. In this paper, we propose a new buffer management method called weighted dynamic thresholds (WDT) which can process packet flows based on loss priorities for quality-of-service (QoS) functionalities with fairly high memory utilization factors. We verified the performance of the proposed scheme through computer simulations.

• Journal of Korea Multimedia Society
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• v.14 no.2
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• pp.219-227
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• 2011

The superframe structure of IEEE 802.15.4, an international standard for low rate WPAN, is composed of CAP(Contention access period) and CFP(Contention free period). CAP is the contention-based access period, while CFP is contention-free access period for supporting QoS by allocating fixed bandwidth. The standard can support QoS for only a few devices, because the maximum number of GTSs is 7. Furthermore, as the value of BO (Beacon order) or SO (Superframe order) increases, the size of a time slot increases. This makes it difficult to precisely allocate bandwidth for any device, because the bandwidth is allocated by the unit of GTS. The proposed scheme of this paper can reduce the waste of BW in CFP by adaptively reducing the size of a time slot in CFP as the value of BO or SO increases and increase the number of GTSs to 127 by modifying the standard. The performance analysis shows that the proposed scheme can dramatically increase the bandwidth utilization during the CFP when comparing with IEEE 802.15.4.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.12
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• pp.83-93
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• 2003

Differentiated service networks, based on bandwidth broker, perform the control and management of QoS provisioning for the QoS guaranteed services, However, the centralized bandwidth broker model has a scalability problem since it has centralized resource management for the admission control function of all call attempts, Therefore, in this paper, we proposea novel adaptive admission control mechanism according to the attempted call status for enhancing the scalability under the centralized bandwidth broker model in IP differentiated service networks, The proposed mechanism decouples the function of admission control from the bandwidth broker, So, the ingress edge node performs the admission control and the bandwidth broker performs the resource management and QoS provisioning, We also introduce an edge-to-edge path concepts and the ingress edge node performs the admission control under the allocated bandwidth to eachpath. The allocated bandwidth per path adaptively varies according to the status of the attempted calls, This mechanism can significantly reduce the number of communication message between the bandwidth broker and each edge node in the network and increase the bandwidth utilization via adaptable resource allocation, In this paper we study the adaptive admission control operation and show the efficient and extensive improvement through the performance evaluation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.11A
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• pp.1106-1112
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• 2007

As a demand for high speed uplink packet services increases, the WCDMA enhanced uplink, also known as high speed uplink packet access (HSUPA), has been specified in release 6 by 3GPP. This HSUPA will provide various types of multimedia services, such as real-time video streaming, gaming, VoIP, and FTP. Generally, the performance of HSUPA is dominated by scheduling policy. Therefore, it is required to design a scheduling algorithm considering the traffic characteristics to provide QoS guaranteed services in various traffic environments. In this paper, we propose a scheduling algorithm considering the traffic characteristics to guarantee QoS in a mixed traffic environment. Finally, the performance of the proposed scheduling algorithm is evaluated in terms of average packet delay, packet delay jitter, and system throughput using a system level simulator.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.3 s.345
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• pp.78-83
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• 2006

In this paper, we propose Active bandwidth allocation scheme of multiplexed streamed MPEG video sequences over BcN network. In order to real time processing, multiplexed source is estimated by linear-prediction per measurement period. n the result target quality value were not sufficient, we proposed a over-allocation method and a reallocation one to guarantee QoS. We used two kinds of sources, one is random multiplexed source made of four different video sources, the other is the one considered the arrange of I frame in the sequence. With those sources, we analyzed the linear prediction, compared over-allocation with reallocation method. As a result, In both schemes, the objected target quality value is achieved, the sufficient valuce bandwidth under 10% when measurement period is over 1.8 sec, the utilization is over 0.9. Especially, the Target of quality value of the reallocation scheme is better at the same condition.

• Journal of the Korea Society for Simulation
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• v.22 no.2
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• pp.53-62
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• 2013

Cloud environment is one in the field of distributed computing and it consists of physical nodes and virtual nodes. In distributed cloud environment, an optimal path search is that each node to perform a search for an optimal path. Synchronization of each node is required for the optimal path search via fast data transmission because of real-time environment. Therefore, a quantization technique is required in order to guarantee QoS(Quality of Service) and search an optimal path. The quantization technique speeds search data transmission of each node. So a main server can transfer data of real-time environment to each node quickly and the nodes can perform to search optimal paths smoothly. In this paper, we propose the quantization technique to solve the search problem. The quantization technique can reduce the total data transmission. In order to experiment the optimal path search system which applied the quantized data transmission, we construct a simulation of cloud environment. Quantization applied cloud environment reduces the amount of data that transferred, and then QoS of an application for the optimal path search problem is guaranteed.

• Journal of the Korean Operations Research and Management Science Society
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• v.41 no.2
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• pp.53-65
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• 2016

Static wireless multi-hop networks, such as wireless mesh networks and wireless sensor networks have proliferated in recent years because of they are easy to deploy and have low installation cost. Two key measures are used to evaluate the performance of a multicast tree algorithm or protocol : end-to-end delay and the number of transmissions. End-to-end delay is the most important measure in terms of QoS because it affects the total throughput in wireless networks. Delay is similar to the hop count or path length from the source to each destination and is directly related to packet success ratio. In wireless networks, each node uses the air medium to transmit data, and thus, bandwidth consumption is related to the number of transmission nodes. A network has many transmitting nodes, which will cause many collisions and queues because of congestion. In this paper, we optimize two metrics through a guaranteed delay scheme. We provide an integer linear programming formulation to minimize the number of transmissions with a guaranteed hop count and preprocessing to solve the aforementioned problem. We extend this scheme not only with the guaranteed minimum hop count, but also with one or more guaranteed delay bounds to compromise two key metrics. We also provide an explanation of the proposed heuristic algorithm and show its performance and results.