• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.3A
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• pp.250-258
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• 2011

In this paper we propose a load balancing scheme for heterogeneous multiple AS's (Application Server) in IMS (IP Multimedia Subsystem) based network. In IMS network, to perform load balancing among multiple ASs with different registration pattern, different weight value should be set for each AS. In previous systems, there exists an inconvenience that the weight value should be set manually by the operator after monitoring the result. In this paper we propose a method to calculate optimal weight in automatic manner and to perform load balancing simultaneously. We also propose a simplified algorithm to reduce calculation in specific situation and present a way to apply our proposed scheme in adaptive manner according to the situation. Through simulation result, we verify that our proposing scheme outperforms previous schemes in load balancing and adjusts well to the change of the system in automatic manner with fast convergence.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.4
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• pp.696-706
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• 2007

Multiprocessor system on chip (MPSoC) platform has set a new innovative trend for the System on Chip (SoC) design. With the rapidly approaching billion transistors era, some of the main problem in deep sub-micron technologies characterized by gate lengths in the range of 60-90 nm will arise from non scalable wire delays, errors in signal integrity and un-synchronized communication. These problems may be addressed by the use of Network on Chip (NOC) architecture for future SoC. Most future SoCs will use network architecture and a packet based communication protocol for on chip communication. This paper presents an adaptive wormhole routing with proactive turn prohibition to guarantee deadlock free on chip communication for NOC architecture. It shows a simple muting architecture with five full-duplex, flit-wide communication channels. We provide simulation results for message latency and compare results with those of dimension ordered techniques operating at the same link rates.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.6
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• pp.1104-1114
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• 2007

Digital video compression technique has played an important role that enables efficient transmission and storage of multimedia data where bandwidth and storage space are limited. The new video coding standard, H.264/AVC, developed by Joint Video Team(JVT) significantly outperforms previous standards in compression performance. Especially, variable length code(VLC) plays a crucial pun in video and image compression applications. H.264/AVC standard adopted Context-based Adaptive Variable Length Coding(CAVLC) as the entropy coding method. CAVLC of H.264/AVC requires a large number of the memory accesses. This is a serious problem for applications such as DMB and video phone service because of the considerable amount of power that is consumed in accessing the memory. In order to overcome this problem in this paper, we propose a variable length technique that implements memory-free coeff_token, level, and run_before decoding based on arithmetic operations and using only 70% of the required memory at total_zero variable length decoding.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.16 no.6
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• pp.1892-1912
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• 2022

With the massive demand and growth of cloud computing, virtualization plays an important role in providing services to end-users efficiently. However, with the increase in services over Cloud Computing, it is becoming more challenging to manage and run multiple Virtual Machines (VMs) in Cloud Computing because of excessive power consumption. It is thus important to overcome these challenges by adopting an efficient technique to manage and monitor the status of VMs in a cloud environment. Reduction of power/energy consumption can be done by managing VMs more effectively in the datacenters of the cloud environment by switching between the active and inactive states of a VM. As a result, energy consumption reduces carbon emissions, leading to green cloud computing. The proposed Efficient Dynamic VM Scheduling approach minimizes Service Level Agreement (SLA) violations and manages VM migration by lowering the energy consumption effectively along with the balanced load. In the proposed work, VM Scheduling for Efficient Dynamically Migrated VM (VMS-EDMVM) approach first detects the over-utilized host using the Modified Weighted Linear Regression (MWLR) algorithm and along with the dynamic utilization model for an underutilized host. Maximum Power Reduction and Reduced Time (MPRRT) approach has been developed for the VM selection followed by a two-phase Best-Fit CPU, BW (BFCB) VM Scheduling mechanism which is simulated in CloudSim based on the adaptive utilization threshold base. The proposed work achieved a Power consumption of 108.45 kWh, and the total SLA violation was 0.1%. The VM migration count was reduced to 2,202 times, revealing better performance as compared to other methods mentioned in this paper.

• Journal of KIISE:Computer Systems and Theory
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• v.26 no.8
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• pp.984-998
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• 1999

최근 개인 휴대 통신에 대한 관심도가 증가하면서 B-ISDN (Broadband Integrated Services Digital Network)과 같은 기존의 유선 망에서 제공하던 다양한 멀티미디어 응용 지원을 무선 망으로 확장시키기 위한 연구가 활발히 진행되고 있다. 그러나 기존의 유선 망에서는 멀티미디어 응용 지원을 위해 QoS (Quality of Service) Provisioning에 관한 많은 연구가 되어 있으나 무선 망에서는 이동성과 무선 전파의 열악한 전송으로 인해 새로운 QoS Provisioning 방법에 관한 연구가 필수적이다. 본 논문에서는 이러한 무선 망의 특수성으로 인해 발생할 수 있는 서비스의 질 저하와 강제 종료를 줄임으로써 지속적인 QoS를 보장해 주고 한정된 무선 자원을 효율적으로 사용하며 처리에 의한 오버헤드를 줄이기 위하여 다음과 같은 세 가지 방법을 제안하였다. 첫째, 핸드오프 강제 종료율을 줄이기 위하여 대역폭 예약 방법을 사용하되 특정 셀의 트래픽 특성에 맞게 또한 시간대에 따른 트래픽 특성에 따라 예약 대역폭의 양을 조절하는 적응적 대역폭 예약 방법이다. 둘째, 많은 경우 각 셀의 트랙픽 변화는 일정한 주기로 변화한다는 특성에 따라 과거의 트래픽 정보를 이용하는 예측 기반의 대역폭 예약 방법이다. 마지막으로 호의 종류, 트래픽 특성, 단말기의 이동 속도에 따라 다른 우선 순위에 의해 호 수락 제어를 수행하는 우선 순위 기반의 호 수락 제어를 제안하였다. 시뮬레이션을 통하여 기존에 제안된 방법과 성능 비교하여, 요구되는 수준의 QoS 보장과 효율적인 자원의 사용, 요구되는 처리비용의 최소화를 통해 전체 시스템의 성능 향상을 입증하였다.Abstract As interest in wireless hand-held terminals and in personal communications services increases recently, there have been broad studies on the ways to support multimedia applications provided in wired networks such as B-ISDN (Broadband Integrated Services Digital Network) in wireless networks. However, since many studies have focused on Quality of Service (QoS) Provisioning in wired networks to provide multimedia applications, new methods of QoS Provisioning are needed in wireless networks to resolve the problem of wireless channel fading and the difficulty of mobility occurred in wireless networks. This paper proposes three schemes of QoS Provisioning in wireless networks which will make continuous QoS guarantee and efficient use of limited wireless resources possible. The first scheme reserves bandwidth in proportion to the amount of real-time traffic in the neighbor cells to decrease the handoff dropping rate of delay sensitive real-time connections, adapting reserved bandwidth for efficient resource utilization. The second scheme is predictive bandwidth reservation scheme that utilizes the past handoff information. It can decrease overheads required to adapt bandwidth reservation. The last scheme is priority-based call admission control prioritizing traffic type (real-time traffic/ non-real-time traffic), connection type (new connection /handoff connection), and mobile terminal speed (fast mobile/slow mobile). Simulation results show that the proposed QoS Provisioning schemes improve the total system performance by achieving three goals - required QoS guarantee, higher bandwidth utilization and less overhead.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.9A
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• pp.1276-1284
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• 1999

We have evaluated the BER's and CLP's of Wireless ATM (WATM) cells employing the concatenated FEC code with pilot symbols for fading compensation through the simulation in indoor wireless channel modeled as a Rayleigh and a Rician fading channel, respectively. The results of the performance evaluation are compared with those obtained by employing the convolutional code in the same condition. In Rayleigh fading channel, considering the maximum tolerance BER ( $10^-3$) as a criterion of the voice service, it is blown that the performance improvement of about 4 dB is obtained in terms of $E_b/N_o$ by employing the concatenated FEC code with pilot symbols rather than the convolutional code with pilot symbols.When the values of K parameter which means the ratio of the direct signal to scattered signal power in Rician fading channel are 6 and 10, it is shown that the performance improvement of about 4 dB and 2 dB is obtained, respectively, in terms of $E_b/N_o$ by employing the concatenated FEC code with pilot symbols considering the maximum tolerance BER of the voice service. Also in Rician fading channel of K=6 and K= 10, considering CLP = $10^-3$ as a criterion, it is observed that the performance improvement of about 3.5 dB and1.5 dB is obtained, respectively, in terms of $E_b/N_o$ by employing the concatenated FEC code with pilot symbols.

• Journal of KIISE:Information Networking
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• v.29 no.1
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• pp.1-14
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• 2002

The most important issue in providing multimedia traffic on a mobile computing environments is to guarantee the mobile host(client) with consistent QoS(Quality of Service). However, the QoS negotiated between the client and network in one cell may not be honored due to client mobility, causing hand-offs between cells. In this paper, a call admission control mechanism is proposed to provide consistent QoS guarantees for multimedia traffics in a mobile computing environment. Each cell can reserve fractional bandwidths for hand-off calls to its adjacent cells. It is important to determine the right amount of reserved bandwidth for hand-off calls because the blocking probability of new calls may increase if the amount of reserved bandwidth is more than necessary. An adaptive bandwidth reservation based on an MPP(Mobility Pattern Profile) and a 2-tier cell structure has been proposed to determine the amount of bandwidth to be reserved in the cell and to control dynamically its amount based on its network condition. We also propose a call admission control based on this bandwidth reservation and "next-cell prediction" scheme using an MPP. In order to evaluate the performance of our call admission control mechanism, we measure the metrics such as the blocking probability of our call admission control mechanism, we measure the metrics such as the blocking probability of new calls, dropping probability of hand-off calls, and bandwidth utilization. The simulation results show that the performance of our mechanism is superior to that of the existing mechanisms such as NR-CAT1, FR-CAT1, and AR-CAT1.

• 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.

• KIPS Transactions on Computer and Communication Systems
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• v.11 no.1
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• pp.23-34
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• 2022

With the development of network technology, the application area has also been diversified, and protocols for various purposes have been developed and the amount of traffic has exploded. Therefore, it is difficult for the network administrator to meet the stability and security standards of the network with the existing traditional switching and routing methods. Software Defined Networking (SDN) is a new networking paradigm proposed to solve this problem. SDN enables efficient network management by programming network operations. This has the advantage that network administrators can flexibly respond to various types of attacks. In this paper, we design a threat level management module, an attack detection module, a packet statistics module, and a flow rule generator that collects attack information through the controller and switch, which are components of SDN, and detects attacks based on these attributes of SDN. It proposes a method to block denial of service attacks (DoS) of advanced attackers by programming and applying honeypot. In the proposed system, the attack packet can be quickly delivered to the honeypot according to the modifiable flow rule, and the honeypot that received the attack packets analyzed the intelligent attack pattern based on this. According to the analysis results, the attack detection module and the threat level management module are adjusted to respond to intelligent attacks. The performance and feasibility of the proposed system was shown by actually implementing the proposed system, performing intelligent attacks with various attack patterns and attack levels, and checking the attack detection rate compared to the existing system.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.1
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• pp.48-59
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• 2009

Next generation mobile communication system will have various access networks and provide seamless multimedia data service to mobile nodes. The WiNGS system, one of these access network, has superior RAT(Radio Access Technology) and network connectivity other than previous ones. In this paper, we propose link-layer SR-ARQ between mobile node and WAGW to solve packet re-ordering problem during handover. In the proposed scheme, we apply the adaptive timer at the SR-ARQ sender to reduce unnecessary packet retransmission during handoff and delayed NAK algorithm at the SR-ARQ receiver for delaying NAK response due to temporary out-of-sequenced frames. We evaluate the performance of the proposed scheme by implementing the new SR-ARQ protocol at the link-layer using ns-2 simulator, and show the handover performance is improved greatly by preventing unnecessary retransmission.