• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.1A
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• pp.33-43
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• 2010

Recently, users want to use real-time multimedia services, such as internet, VoIP, etc., using their IEEE 802.11 wireless lan mobile stations. In order to provide such services, a handoff among access points is essential to support the mobility of a node, in such an wide area. However, the legacy handoff methods of IEEE 802.11 technology are easy to lose connections. Also, the recognition of a disconnection and channel re-searching time make the major delay of the next AP to connect. In addition, because IEEE 802.11 decides the selection of an AP depending only on received signal strength, regardless of a node direction, position, etc., it cannot guarantee a stable bandwidth for communication. Therefore, in order to provide a real-time multimedia service, a node must reduce the disconnection time and needs an appropriate algorithm to support a sufficient communication bandwidth. In this paper, we suggest an algorithm which predicts a handoff point of a moving node by using GPS location information, and guarantees a high transmission bandwidth according to the signal strength and the distance. We implemented the suggested algorithm, and confirmed the superiority of our algorithm by reducing around 3.7ms of the layer-2 disconnection time, and guaranteed 24.8% of the communication bandwidth.

• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.834-845
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• 2011

Massive broadcasting contents such as UHD(Ultra High Definition) TV which requires multi-channel capacity for transmission has been introduced in recent years. A transmission scheme with channel bonding has been considered for transmission of massive broadcasting contents. In HFC(Hybrid Fiber Coaxial) networks, DOCSIS 3.0(Data Over Cable Service Interface Specification 3.0) has already applied channel bonding schemes for up/downstream of data service. A method unlike DOCSIS 3.0 is required to introduce a channel bonding scheme in the broadcasting service having unidirectional transmission with a downstream. Since a massive broadcasting content requires several channels for transmission, VBR(Variable Bit Rate) transmission has been emerging for the bandwidth efficiency. In addition, research on channel allocation and resource scheduling is required to guarantee QoS(Quality of Service) for the broadcasting service based on VBR. In this paper, we propose a transmission method for mass broadcasting service in HFC network and show the UHD transmission simulator developed to evaluate the performance. In order to evaluate the performance, we define various scenarios. Using the simulator, we assess the possibility of channel bonding and VBR transmission for UHD broadcasting system to provide mass broadcasting service efficiently. The developed simulator is expected to contribute to the efficient transmission system development of mass broadcasting service.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.19 no.8
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• pp.19-24
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• 2018

The IP Multimedia Subsystem(IMS) is an architectural framework for delivering IP multimedia services to mobile users. In order to guarantee the reliability and Quality of Service(QoS) of a variety of multimedia services, we need a new evolutionary approach that maintains the IMS based signaling platform which can perform the processing of flow through distributed controllers. Software Defined Network(SDN) is an architecture purporting to be distributed, dynamic, cost-effectives as well as adapting and seeking to be suitable for the high-bandwidth, dynamic nature of today's applications. It requires some methods for the control plane to communication with the data plane. One of such mechanisms is OpenFlow which is a prominent standard protocol and interface that is responsible for managing the network resources by using the remote SDN controller. In this paper, we propose a straightforward approach for integrating SDN technology together with the IMS architecture. Therefore we propose and construct a combined architecture model that performs flow processing using OpenFlow via the IMS based signaling platform, which maintains the existing telecom call service. Additionally, we describe some relevant experimentation results from the proposed architecture.

• Journal of Korea Multimedia Society
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• v.5 no.6
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• pp.697-703
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• 2002

In this paper, we propose the adaptive multi stream transmission technique based on SPIHT video signal for the highest quality service over the current Internet that does not guarantee QoS. In addition to the reliable transmission of the video stream over the asynchronous packet network, the proposed approach provides the transmission using the adaptive frame pattern control and multi steam over the TCP for continuous replay. The adaptive frame pattern control makes the transmission date scalable in accordance with the client's buffer status. Apart from this, the multi stream transmission improves the efficiency of video stream, and is robust to the network jitter problem, and maximally utilizes the bandwidth of the client's. As a result of the experiment, the DR(delay ratio) in the proposed adaptive multi-stream transmission is more close to zero than in the existing signal stream transmission, which enables the best-efforts service to be implemented.

• Journal of the Korea Institute of Military Science and Technology
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• v.21 no.4
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• pp.497-507
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• 2018

Tactical networks are evolving into an All-IP based network for network centric warfare(NCW). Owing to the flexibility of IP based network, various military data applications including real-time and multi-media services are being integrated in tactical networks. Because each application has diverse Quality-of-service(QoS) requirements, it is crucial to develop a QoS provisioning method for guaranteeing QoS requirements efficiently. Conventionally, differentiated services(DiffServ) have been used to provide a different level of QoS for traffic flows. However, DiffServ is not designed to guarantee a specific requirement of QoS such as delay, loss, and bandwidth. Therefore, it is not suitable for military applications with a tight bound of QoS requirements. In this paper, we propose a multi-layer network virtualization scheme that allocates traffic flows having different QoS requirements to multiple virtual networks, which are constructed to support different QoS policies such as virtual network functions(VNFs), routing, queueing/active queue management(AQM), and physical layer policy. The experiment results indicate that the proposed scheme achieves lower delays and losses through multiple virtual networks having differentiated QoS policies in comparison with conventional networks.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.1
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• pp.73-82
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• 2004

As a new service category to better support TCP traffic in ATM networks, the Guaranteed Frame Rate (GFR) service category aims to support minimum cell rate guarantee, fairly distribute available bandwidth while keeping the simplicity of Unspecified Bit Rate (UBR). In this paper, we proposed a buffer management scheme which uses the per-VC accounting of single FWO queue and capable of supporting both GFR.1 and GFR.2 conformance definition. The proposed buffer management deal with GFR.1 and GFR.2 conformance definitions differentially by controlling the number of CLP=0 cell and CLP=1 cell which are occupying buffer space. The simulation results show that our proposed scheme satisfies the requirements of GFR services as well as improves total fairness index and each conformance definition fairness index.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.9A
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• pp.1067-1074
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• 2004

In the wireless mobile networks, it IS important to provide the quality-of-service (QoS) guarantees as they are increasingly expected to support the multimedia applications Although the QoS provisioning problem arises in the wire-line networks as well, the mobility of hosts and the scarcity of bandwidth make QoS provisioning a challenging task in wireless mobile networks. The resource allocation to multimedia applications of varying QoS reqUlrement 15 a complex issue. In this paper, we propose a new adaptive resource allocation scheme based on the concept of the resource reservation and the renegotiation in order to guarantee the QoS of the real-tune traffic. The proposed scheme is aimed at improving the perfonnance in terms of the new call blocking rate, the bandoff dropping rate, and the bandwIdth utilization.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.5 s.347
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• pp.84-93
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• 2006

Ethernet-PON is an emerging access network technology that provides a low-cost method of deploying optical access lines between OLT and ONUs. It has a point-to-multipoint and multipoint-to-point architecture in downstream and upstream direction, respectively. Therefore, downstream packets are broadcast from an OLT toward all ONUs sithout collision. On the other hand, since alt ONUs share a common channel, the collision may be occurred for the upstream transmission. Therefore, earlier efforts on Ethernet-PON have been concentrated on an upstream MAC protocol to avoid collision. But it is needed to control downstream traffic in practical access network, where the network provider limits available bandwidth according to the number of users. In this paper, we propose a traffic control scheme for supporting the fairness of the downstream bandwidth. The objective of this algorithm is to guarantee the fairness of ONUs while maintaining good performance. In order to do this, we define the service probability that considers the past traffic information for downstream, the number of tokens and the relative size of negotiated bandwidth. We develop the simulation model for Ethernet-PON to evaluate the rate-limiting algorithm by using AWESIM. Some results are evaluated and analyzed in terms of defined fairness factor, delay and dropping rate under various scenario.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.9-16
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• 2005

A new online routing algerian is proposed in this paper, which use the interference-prediction to solve the network congestion originated from extension of Internet scope and increasing amount of traffic. The end-to-end QoS has to be guaranteed in order to satisfy service level agreements (SLAs) in the integrated networks of next generation. For this purpose, bandwidth is allocated dynamically and effectively, moreover the path selection algorithm is required while considering the network performance. The proposed algorithm predicts the level of how much the amount of current demand interferes the future potential traffic, and then minimizes it. The proposed algorithm considers the bandwidth on demand, link state, and the information about ingress-egress pairs to maximize the network performance and to prevent the waste of the limited resources. In addition, the interference-prediction supports the bandwidth guarantee in dynamic network to accept more requests. In the result, the proposed algorithm performs the effective admission control and QoS routing. In this paper, we analyze the required conditions of routing algorithms, the aspect of recent research, and the representative algorithms to propose the optimized path selection algorithm adequate to Internet franc engineering. Based on these results, we analyze the problems of existing algorithms and propose our algorithm. The simulation shows improved performance by comparing with other algorithms and analyzing them.

• Journal of KIISE:Information Networking
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• v.30 no.4
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• pp.461-473
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• 2003

Multimedia applications are much more sensitive to QoS(Quality of Service) than text based ones due to their data continuity. In order to provide a fast moving MH(Mobil Host) using multimedia application with a consistent QoS,an efficient call admission mechanism is in need. This paper proposes the 2SCA(2-Step Call Admission) scheme based on cal admission scheme using pripority to guarantee the consistent QoS for mobile multimedia applications. A calls of MH are classified new calls, hand-off calls, and QoS upgrading calls. The 2SCA is composed of the basic call admission and advanced call admission; the former determines the call admission based on bandwidth available in each cell and the latter determines the call admission by applying DTT(Delay Tolerance Time), PQeueu(Priority Queue), and UpQueue(Upgrade Queue) algorithm according to the type of each call blocked at the basic call admission stage. In order to evaluate the performance of our mechanism, we measure the metrics such as the dropping probability of new calls, dropping probability of hand-off calls, and bandwidth utilization. The result shows that the performance of our mechanism is superior to that of existing mechanisms such as CSP(Complete Sharing Policy), GCP(Guard Channel Policy) and AGCP(Adaptive Guard Channel Policy).