• The KIPS Transactions:PartC
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• v.12C no.6 s.102
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• pp.773-780
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• 2005

Recently, the telecommunication network is in a transition toward the BcN (Broadband convergence Networks) that integrates the Internet, the telecommunication and the broadcasting. The BcN makes it possible to provide of 4he third-party application by using the Open API The Parlay X API is an example of the Open API and the Parlay X API is grafted in the Web Service and is abstracted from the Parlay API. So, the Parlay X API does not only have advantages of a Web Service, but also disadvantages of the Web Service. The main disadvantage of the Web Service is that it does not support the any QoS mechanism. In this paper, we propose the architecture that solves the reliability among the QoS issues for the Web Service. The proposed mechanism adopts the Asynchronous architecture for the reliable Web Service and improves the performance of the retransmission between the client and the server using the expected response time. We perform the performance evaluation through the simulation in older to compare with the existing mechanism. Consequently, we can confirm that the performance of the proposed mechanism performs better than the existing mechanism.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2001.06a
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• pp.21-24
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• 2001

For very large multimedia traffic to be supported successfully in wireless network environment, it is necessary to provide Quality-of-Service(QoS) guarantees between mobile hosts(clients). In order to guarantee the Qos, we have to keep the call blocking probability below target value during handoff session. However, the QoS negotiated between the client and the network may not be guaranteed due to lack of available channels for traffic in the new cell, since mobile clients should be able to continue their on-going sessions. In this paper we propose a efficient load-balancing algorithm based on the adaptive bandwidth reservation scheme for enlarging available channels in a cell. We design a new method to predict the mobility of clients using MPT(mobility profile table). This method is then used to reserve a part of bandwidths for handoff calls to its adjacent cells and this reserved bandwidth can be used for handoff call prior to new connection requests. If the number of free channels is also under a low threshold value, our scheme use a load-balancing algorithm with a adaptive bandwidth reservation. In order to evaluate the performance of our algorithm, we measure the metrics such as the blocking probability of new calls and dropping probability of handoff calls, and compare with other existing schemes.

• The Journal of the Korea institute of electronic communication sciences
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• v.16 no.5
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• pp.807-816
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• 2021

With the technology of the 4th industrial revolution, network traffic is increasing due to an increase in supply, an increase in demand, and an increase in the complexity of traffic patterns. SDN, a concept in which H/W and S/W are separated in order to efficiently manage such massive traffic, is attracting attention as a next-generation network. A lot of research is being conducted on the merits of applying flexible policies by avoiding the problem of rigid vendor dependency by using the SDN controller implemented with S/W Opensource. Therefore, in this paper, we propose an efficient load balancing technique by grouping through the packet structure of the network layer using the control layer and infrastructure layer of SDN and analyzing the packet delay and reception rate.

• Journal of Korea Multimedia Society
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• v.17 no.9
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• pp.1084-1091
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• 2014

When demand for transmitting multimedia data increases, network congestion is more likely to occur and users will suffer high loss rate as well as high delay. In order to enhance quality-of-service (QoS) of video multicasting, we need to raise transmission reliability and reduce end-to-end delay. This paper proposes a routing mechanism for a OpenFlow-based small-scale network in order to multicast video reliably with low delay. In our method, multipath routing will be applied to Multiple Description (MD) Coded video to exploit its multi-description property. Through performance evaluation, our method shows improvement on loss rate, delay and video distortion.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.8B
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• pp.1150-1158
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• 2010

At present multimedia service composed of voice, data, and video service is prevalent in the Internet. As such, wide-scale penetration of Internet service imposes tremendous pressure to the network infrastructure such as the media servers and links as well as the nodes. In addition, users from a large-scale population require broad bandwidth and high level of QoS. This requires a network with reliable and scalable services to customers, which also necessitates a realistic method for the design of the a media server. In this work, we explore the influence of user's behavior about delay on the performance of media server that takes into account the system and user attributes in a realistic manner. By incorporating user's behavior about the delay-sensitivity, we present an analytic framework for the evaluation of the performance of the media server, via which we illustrate a meaningful intuition in the provision of Internet multimedia service.

• Annual Conference of KIPS
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• 2012.11a
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• pp.42-45
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• 2012

A network processor (NP) is an application-specific instruction-set processor for fast and efficient packet processing. There are many issues in compiler's code generation and optimization due to NP's hardware constraints and special hardware support. In this paper, we describe in detail how to resolve the issues. Our compiler was developed on LLVM 3.0 and the NP target was our in-house network processor which consists of 32 64-bit RISC processors and supports multi-context with special hardware structures. Our compiler incurs only 9.36% code size overhead over hand-written code while satisfying QoS, and the generated code was tested on a real packet processing hardware, called S20 for code verification and performance evaluation.

• Journal of the Korea Society of Computer and Information
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• v.16 no.11
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• pp.45-56
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• 2011

WBAN is short range wireless communication technology which is consists of several small devices close to, attached to or implanted into the human body. WBAN is classified into between medical and non-medical by applications based on technology and medical data with periodic characteristics is used the GTS method for transmitting data to guarantee the QoS. In this paper we proposed algorithm that resolve lack of GTSs while data transmit GTS method in superframe structure of WBAN. Coordinator dynamically allocates GTSs according to the data rate of devices and make devices share GTSs when lack of GTSs. We compared delay bounds, throughput for performance evaluation of the proposed algorithm. In other words, we proposed algorithm adaptive WFQ scheduling that GTS allocation support differential data rate in environments of WBAN. The experiment results show the throughput increased and the maximum delay decreased compared with Round Robin scheduling.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.5B
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• pp.489-497
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• 2009

In the 3GPP LTE and IEEE 802.16m standardizations for IMT-Advanced, the efficient transmission of multicast and broadcast data is one of the important issues to improve system performance. Especially, to guarantee each user's QoS in the multicast group, H-ARQ or adaptive modulation are considered to be adoption, however, reduction of feedback overhead is one of the most important issues. In this paper, performance of the previously proposed OOK (On-Off Keying) based common ACK/NACK feedback channel is analyzed through theoretical analysis and simulation. Based on the results, system parameter optimization methods for common feedback channel are proposed. In addition, through computer simulation, it is shown that the common feedback method according to the proposed feedback parameter optimization can feedback ACK/NACK information for multicast H-ARQ transmission, more reliably than the conventional user dedicated feedback method in 3GPP LTE systems, and can reduce the feedback overhead as well.

• The Transactions of the Korea Information Processing Society
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• v.6 no.11S
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• pp.3390-3398
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• 1999

To achieve high-speed communication and to improve QoS characteristics, ATM is now being used as underlying transfer mechanism of Internet protocols, or TCP/IP. When TCP uses the ABR service in ATM networks, the ABR service control is important to improve the performance of TCP/IP traffic as long as there is no interaction between ABR rate control and TCP flow control. In such environment, the interoperability of EFCI and ER switches becomes unavoidable in order to more effectively control ABR traffic in ATM network. In this paper, we use the model of mixed EFCI-ER environment and discuss the performance issues, e.g., the fairness of bandwidth allocation, drop rate, throughput by using various ER switch algorithms. Among various ERSwitch algorithms, we present one algorithm that have a less network topology and allocate the fair share of VC bandwidth by simulation results.

• Journal of Digital Convergence
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• v.13 no.11
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• pp.179-185
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• 2015

In this paper, we research the IPTV that is the convergence technique of TV and network technique and performed the performance evaluation of picture quality of IPTV in the situation of increasing the network traffic in the Internet environment. To do this, we constructed the mock Internet network similar to the real Internet environment and measured the quality of received video using V-Factor model according to the increase of network traffic, and analyzed the result of the experiment. Making use of the result of this paper for the threshold value of V-Factor, the measured factor of network performance, the measured factor of video performance in the watchable IPTV video quality.