• 한국정보통신설비학회:학술대회논문집
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• 2007.08a
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• pp.136-141
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• 2007

The frequency accuracy and phase alignment is necessary for ensuring the quality of service (QoS) for applications such as voice, real-time video, wireless hand-off, and data over a converged access medium at the telecom network. As telecom networks evolve from circuit to packet switching, proper synchronization algorithm should be meditated for IP networks to achieve performance quality comparable to that of legacy circuit-switched networks. The Time of Packet (ToP) specified in IEEE 1588 is able to synchronize distributed clocks with an accuracy of less than one microsecond in packet networks. But, The ToP can be affected by impairments of a network such as packet delay variation. This paper proposes the efficient method to minimize the expectable delay variation when ToP synchronizes the distributed clocks. The simulation results are presented to demonstrate the improved performance case when the efficient ToP transmit algorithm is applied.

• Journal of Korea Game Society
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• v.15 no.6
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• pp.97-106
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• 2015

In an Unity engine for multi-player online games, the main thread processing game logic must be separated from the network thread that is responsible for network packet communication. Packet communication between the network threads needs to drop packets that overlap in order to improve the rendering speed. In this paper, the packet discard policy of network thread is proposed for an Unity engine for multi-player online games. The proposed method is the hybrid method of both Partial Packet Discard and Periodic Packet Discard methods to improve the rendering speed by periodically discarding overlapped network packets managed by the queue. The rendering speed of the proposed method is analyzed and its effectiveness is verified by various packet generating simulations of the Unity engine for multi-player online games.

• Journal of the Korea Society of Computer and Information
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• v.10 no.6 s.38
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• pp.227-232
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• 2005

A network of Packet based switch Mode will be required to carry the traffics(CVR,VBR, UBR, ABR) generated by a wide range of services. Packet based Network services the quality-of-Service (QoS) management of traffic sources and bandwidth. Besides efficiency and throughput services are achieved in the multimedia traffic sent in the network. In this paper. the scheduler transmits the safe packet, drop the unsafe packet and evaluate unsafe packet as the requirement of the delay avoiding the network congestion for improving the QoS of the multimedia network In this paper. we Propose the scheduling algorithm which evaluates and drops the packet . The suggested model performance of the switch is estimated and simulated in terms of the delay by computer.

• Journal of the Korea Society of Computer and Information
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• v.10 no.5 s.37
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• pp.245-250
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• 2005

A network of Packet based switch Mode will be required to carry the traffics(CVR,VBR, UBR, ABR) generated by a wide range of services. Packet based Network services the Qualify-of-Service (QoS) management of traffic sources and bandwidth. Besides efficiency and throughput services are achieved in the multimedia traffic sent in the network. In this paper, the scheduler transmits the safe packet, drop the unsafe packet and evaluate unsafe Packet as the requirement of the delay avoiding the network congestion for improving the QoS of the multimedia network. In this paper, we Propose the scheduling algorithm which evaluates and drops the packet The suggested model Performance of the switch is estimated and simulated in terms of the delay by computer.

• The Transactions of the Korea Information Processing Society
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• v.6 no.11S
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• pp.3190-3198
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• 1999

Because of the rapid growth of the mobile communication, the need for the mobile internet access has grown up as well. since the current mobile communication network, however, is optimized for a voice communication system, which exclusively occupies a channel for a given time, it is not suitable for variable rate packet data. In order to support the mobile internet access, it is essential do design a reasonable packet switching network which supports the mobility. Since mobile packet network has longer latency, high speed switching and QoS are required to meet the user's requirements. In this paper, we suggest an resonable way to construct a network and its operation procedures utilizing GPRS(General Packet Radio Service) network and MPLS(Multi Protocol Label Switching) to provide a high speed switching and QoS mobile internet access. GPRS is used as a network which supports the mobility and MPLS guarantees the QoS and high speed IP protocol transmission based on the ATM switching technology.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.3 no.4
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• pp.274-279
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• 2002

ATM packet switching technologies for the purpose of the B-ISDN service are focused on high performance which represents good qualities on throughput, packet loss, and packet delay. ATM switch designs on a class of parallel interconnection network have been researched. But these are based on the self-routing function of it. It leads to conflict with each other, and to lose the packets. Therefore, this paper proposes the method based on Sort-Banyan network should be adopted for optimal routing algorithm. It is difficult to expect good hardware complexity. For good performance, a switch design based on the development of new routing algorithm is required. For the design of switch network, the packet distributor and multiplane are proposed. They prevent each packet from blocking as being transmitted selectively by two step distributed decision algorithm. This switch will be proved to be a good performance switch network that internal blocking caused from self-routing function is removed. Also, it is expected to minimize the packet loss and decrease the packet delay according to packet transmission.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.767-772
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• 2005

The Internet is a packet switched network which offers best-effort service, but current IP network provide enhanced services such Quality of Services, Virtual Private Network (VPN) services, Distribute Firewall and IP Security Gateways. All such services need packet classification for determining the flow. The problem is performing scalable packet classification at wire speeds even as rule databases increase in size. Therefore, this research offer packet classification algorithm that increase classifier performance when working with enlarge rules database by rearrange rule structure into Bitmap Intersection Lookup (BIL) tables. It will use packet's header field for looking up BIL tables and take the result with intersection operation by logical AND. This approach will use simple algorithm and rule structure, it make classifier have high search speed and fast updates.

• Journal of Industrial Technology
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• v.27 no.B
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• pp.235-243
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• 2007

Sensor network is used to obtain sensing data in various area. The interval to sense the events depends on the type of target application and the amounts of data generated by sensor nodes are not constant. Many applications exploit long sensing interval to enhance the life time of network but there are specific applications that requires very short interval to obtain fine-grained, high-precision sensing data. If the number of nodes in the network is increased and the interval to sense data is shortened, the amounts of generated data are greatly increased and this leads to increased amount of packets to transfer to the network. To transfer large amount of packets fast, it is necessary that the delay between successive packet transmissions should be minimized as possible. In Sensor network, since the Operating Systems are worked on the event driven, the Timer Event is used to transfer packets successively. However, since the transferring time of packet completely is varies very much, it is very hard to set appropriate interval. The longer the interval, the higher the delay and the shorter the delay, the larger the fail of transfer request. In this paper, we propose ESTEO which reduces the delay between successive packet transmissions by using SendDone Event which informs that a packet transmission has been completed.In ESTEO, the delay between successive packet transmissions is shortened very much since the transmission of next packet starts at the time when the transmission of previous packet has completed, irrespective of the transmission timee. Therefore ESTEO could provide high packet transmission rate given large amount of packets.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.11B
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• pp.1225-1233
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• 2009

ISPs and Enterprises are equipping their networks with sufficiently high speed facilities and provide large bandwidths members. However the high speed enterprise network does not have satisfying end-to-end network performance within the network in spite of under utilization. The root cause of this performance degradation is a micro-congestion, which is a short-live event of traffic congestion. A micro-congestion causes packet loss, delay and packet reodering, and finally results in end-to-end network performance degradation. In this paper, we propose a micro-congestion detection method and find out the characteristics of performance degradation by analyzing traffic archives which is collected from a network link when a micro-congestion occurs.

• The Journal of the Acoustical Society of Korea
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• v.28 no.3E
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• pp.99-108
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• 2009

In this paper, we suggest criteria for objective measurement of speech quality in mobile VoIP (Voice over Internet Protocol) services over wireless mobile internet such as mobile WiMAX networks. This is the case that voice communication service is available under other networks. When mobile VoIP service users in the mobile internet network based on packet call up PSTN and mobile network users, but there have not been relevant quality indexes and quality standards for evaluating speech quality of mobile VoIP. In addition, there are many factors influencing on the speech quality in packet network. Especially, if the degraded speech with packet loss transfers to the other network users through the handover, voice communication quality is significantly deteriorated by the transformation of speech codecs. In this paper, we eventually adopt the Gilbert-Elliot channel model to characterize packet network and assess the voice quality through the objective speech quality method of ITU-T P. 862. 1 MOS-LQO for the various call scenario from mobile VoIP service user to PSTN and mobile network users under various packet loss rates in the transmission channel environments. Our simulation results show that transformation of speech codecs results in the degraded speech quality for different transmission channel environments when mobile VoIP service users call up PSTN and mobile network users.