• The Journal of the Korea Contents Association
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• v.10 no.3
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• pp.80-91
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• 2010

Due to the convergence of broadcasting and communications, IPTV services are spotlighted as the that next-generation multimedia services. IPTV services should have functionality such as unlimited channel capacity, extension of media, QoS awareness and are required increasing traffic and quality control technology to adapt the attributes of IPTV service. Consequently, flow based quality control techniques are needed. Therefore, many studies for providing Internet QoS are performed at IETF (Internet Engineering Task Force). As the buffer management mechanism among IP QoS methods, active queue management method such as RED(Random Early Detection) and modified RED algorithms have proposed. However, these algorithms have difficulties to satisfy the requirements of various Internet user QoS. Therefore, in this paper we propose the Flow based AQM(Active Queue Management) algorithm for the multimedia services that request various QoS requirements. The proposed algorithm can converge the packet loss ratio to the target packet loss ratio of required QoS requirements. And we present a performance evaluation by the simulations using the ns-2.

• Journal of Korea Multimedia Society
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• v.7 no.1
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• pp.54-63
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• 2004

This paper describes a hybrid QoS guarantee scheme for high quality audio streaming services on the Internet. The continuous playback of the audio data requires the isochronous transmission of the audio data packet through the Internet. In order to retain the QoS at the ultimate destination (client) as the same as servers provide, the transmission protocols should consider the error conditions such as packet loss, and out of order delivery. Generally, the protocols supporting the transmission of continuous media data do not try to recover the errors. The protocols are working somehow for the toll quality multimedia streaming services, but rot for the high quality streaming services, such as the DVD sound/music payback. The hybrid QoS guarantee scheme includes the three mechanisms to overcome the problem. The selective retransmission for the lost packet, the adaptive buffering at client-side, and the adaptive transmission rate at server-side are totally adopted to recover the packet loss with the minimal overhead, to prevent from the buffer starvation during the retransmission, and to maintain the isochronous transmission even after the retransmission. The experiments have shown good results for the high Quality audio streaming services on the Internet.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.5
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• pp.941-950
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• 2001

In this paper, we present the integrated network architecture for supporting mobile IP in third generation mobile communication networks, and propose the end-to-end QoS control mechanisms of DiffServ and QoS control functions of each network element in the proposed network. First, the QoS supporting schemes of IMT-2000 are described. Second, the necessities of integrating the networks are discussed and the integrated architecture are proposed. Third, the mapping between wireless channels and DiffServ classes are presented. Finally, the end-to-end QoS control mechanisms are proposed. We also analyze the link level protocols with QoS provisioning for mobile multimedia assuming that the system support voice and data traffic simultaneously. In case of data traffic, the delay and throughput of SREJ ARQ and Type-1 Hybrid ARQ scheme are compared, and In case of voice traffic, the packet loss rate of BCH coding is analyzed according to the varying data traffic loads. The results indicate that the adaptive link level protocols are efficient to meet the QoS requirements while the complexities are increased.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.14 no.4
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• pp.1935-1941
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• 2013

We can use NAT(Network Address Translation) technology to solve the lack of IP address. The problem of NAT traversal is happened when the filtering characteristics of NAT remove the packet that has no binding in the address translation table of NAT. There were many kinds of way to solve these problems by using additional device. Lately, network market is changed into integrating wired and wireless network by the IMS(IP Multimedia Subsystem). The IMS integrates to control network of wired and wireless network, has emerged to control convergence network effectively. Lately, the additional devices like IBCF(Interconnection Border Control Function) and IBGF(Interconnection Border Gateway Function) are used to solve the NAT traversal problem in the IMS. In this paper, we propose the solution of NAT traversal using P-CSCF without any additional equipment in the IMS.

• Journal of KIISE:Information Networking
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• v.32 no.6
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• pp.676-686
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• 2005

The Internet users as well as network providers are eager to have different qualities of service among users beyond the best-effort. In this paper, we propose a scheduling algorithm that provides a differentiated service in the granularity of user sessions. The proposed algorithm is a Hierarchical Deficit Round-Robin (H-DRR) algorithm that is an extension of an existing DRR algorithm. A main advantage is that H-DRR provides service differentiation for throughput-intensive applications such as FTP as well as delay-sensitive applications such as telnet or VoIP without distinguishing the types of applications. The most importance in providing a service differentiation in term of network providers is to have controllability and predictability. We show that H-DRR is superior to DRR in terms of controllability and predictability through both mathematical analysis and simulation experiments. Nevertheless, H-DRR requires O(1) complexity for implementation.

• Journal of Broadcast Engineering
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• v.13 no.5
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• pp.696-706
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• 2008

In this paper we show the performance of the network-adaptive high-definition scalable video streaming using QWLAN board with IEEE 802.11e MAC monitoring and control. Realtime collected MAC parameters are used to determine which video data is extracted for the predicted available bandwidth. To achieve performance, extraction through R-D is proposed instead of the standard video packet extraction. It is shown through experiments that streaming video quality can be enhanced by fast adaptation to network conditions by using the proposed method.

• The Transactions of the Korea Information Processing Society
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• v.7 no.11
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• pp.3479-3489
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• 2000

Most of the study on fairness control method for Distributed-Queue Dual-Bus(DQDB) have been performed under specific load types such as equal probability load types or symmetric load types. On Web-based internet enviroments client-server load types are more practical traffic patlerns than specrfic load types. In this paper, an effiective fairness control method to distribute DQDR network bandwidth fairly to all stations under a client-server load is proposed. In order to implement a dynamic bandwidth timing capabihty needed to distribute the bandwidth fairty at heavy loads, the proposed method uses two pararnetexs, one is an access hrnit to legulate each station's packet transmission and the other is the number of extra emply slots that are yielded to downstream stations. In point of implementation this mechanism is simpler and easier than Bandwidth Tuning Mechanism(BTM) that uses an intermediate pattern and an adptation function. Simulation results show that it outperforms othen mecharusms.

• The Journal of the Korea Contents Association
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• v.10 no.1
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• pp.27-36
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• 2010

In wireless networks environment, there are attempts for constructing systems through which we can monitor and control various electronic devices used in large buildings at remoteness by using Zigbee protocol. Since address assignment method of DAA in Zigbee has a depth restriction, we cannot construct a network in large buildings. And also communication failures are frequent in a large network since broadcast storm can be occurred due to frequent broadcasts among a lot of nodes at AODV routing used in Zigbee. In order to solve these problems, in this paper, we propose a novel protocol which has improved the restriction of depth by using a fixed address assignment method, and has enhanced the broadcast occurrence by devising a static routing method to a packet movement between floors. Regardless of entire network, additionally, this was to enable internal communication reliably by composing an independent PAN. Specially, in this paper, we implemented the proposed wireless protocol and proved stability and practicality through experiment appliances of control of devices established in a large building.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.10
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• pp.52-63
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• 1998

In this paper, several improvements to a copy network proposed previously for multicast packet switching are described. The improvements provide a solution to some problems inherent in multicasting. The input fairness problem caused by overf low is solved by a dynamic starting point decider(DSD), which can calculate running sums of copy requests starting from any input port. The starting point is changed adaptively in every time slot based on both the fill level of the input buffers in current time slot and the overflow situations of the previous time slot. Using the fill level of the conventional network. The DSD also provides the function of regulating overall copy requests according to the amount of input traffics. This is an essential function in improving overall throughputs of the copy networks. The throughput of a multicast switch can be improved substantially if partial service of copy request is implemented when overflow occurs. Call-splitting can also be implemented by the DSD in a straightforward manner. The hardware for the DSD is derived with the objective of simple architectures for the high speed operation. Simulation study of the copy network under various traffic conditions is presented to evaluate its performance.

• Journal of Digital Contents Society
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• v.18 no.2
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• pp.365-371
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• 2017

IEEE 802.11 Wireless Local Area Network (WLAN) adopts Power Saving Mode(PSM) to save the power. Unlike existing PSM, this paper proposes a new scheme for the power saving of the Mobile Host(MH) when a MH performs the data transmission after the competition-based DCF(Distributed Coordination Function) and competition free-based PCF(Point Coordination Function). In this paper, The proposed scheme estimates the distance between the MH with the authority of data transmission and the Access Point(AP) and then adaptively controls the power of the MH considering the distance. Through the simulation result, we find that the proposed scheme consumes the smaller transmission power and has the similar success rate of packet transmission when it is compared to the existing scheme which uses the same power without the consideration of the distance.