• Proceedings of the Korea Society for Industrial Systems Conference
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• 1999.12a
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• pp.121-129
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• 1999

When continuous media are transmitted over the communication networks, asynchrony which can not maintain temporal relationships among packets may occur due to a random transit delay. There exist two types of synchronization schemes ; for guaranteed or non-guaranteed resource networks. The former which applies a resource reservation technique maintains delay characteristics, however, the latter supply a best-effort service. In this paper, I propose a intra-media synchronization scheme to transmit continuous media on general networks not guaranteeing a bounded delay tome. The scheme controls transmission times of the packets by estimating next delay time with the delay distribution. So, the arriving packets may be maintained within a limited delay boundary, and playout will be performed after buffering to smoothen small delay variations. The continually increasing delay due to network overload causes buffer underflow at the receiver. To solve it, the transmitter is required to speed up instantaneously. Too much increase of transmission-rate may cause network congestion. At that time, the transmitter drops the current packet when informed excessive delay from the receiver.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.1
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• pp.23-34
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• 2004

This paper presents architecture and control method of packet scheduler to guarantee QoS of high quality streaming services in high-speed packet-switched networks. Since streaming services need far more stringent QoS requirements than the typical sort of burst data applications, they should be guaranteed minimum bandwidth and end-to-end delay bound to each flow, regardless of the behavior of other flows. To meet these requirements, a packet scheduler isolate a flow from the undesirable effects of other flows and provides end-to-end delay guarantees for individual flow and divides stringently the available link bandwidth among flows sharing the link. Until now, many vendors are developing traffic management chips running at 10Gbps, but most of chips have drawbacks to support high quality streaming services. In this paper, we investigate the drawbacks of commercial TM chips and traffic characteristic of streaming services and present implementation frameworks of the proposed packet scheduler. Finally, we analyze the simulation results of the proposed scheduler.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.3 s.345
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• pp.118-125
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• 2006

Mobile nodes in FHMIPv6 has both advantages of HMIPv6 protocol which reduces signaling delay time and resource consumption during a handover and fast handover algorithm which reduces packet loss. Fast handover algorithm can reduce packet loss by 'tunneling' method ; that transmits a packet from old access router to new access router in case of handover. However, the fast handover algorithm can cause a reordering problem in a receiver between packets tunneled from the previous access router and packets transmitted directly to the new access router, which could degrade the TCP performance due to congestion control. In this paper, we propose two algorithms to solve the reordering problem in fast handover. The first one uses a holding timer for tunneling, the other adds a new algorithm to routers that adopt snoop protocol. We compare the performance of the proposed reseuquencing algorithms with that of the existing FHMIPv6 protocol by simulation. The simulation results show that the proposed algorithms solve the reordering problems and enhance TCP performance by preventing TCP sender entering congestion control.

• 전자공학회논문지 IE
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• v.43 no.3
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• pp.41-48
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• 2006

In this paper, we propose a new active queue management (AQM) scheme by utilizing the predictability of the Internet traffic. The proposed scheme predicts future traffic input rate by using the auto-regressive (AR) time series model and determines the future congestion level by comparing the predicted input rate with the service rate. If the congestion is expected, the packet drop probability is dynamically adjusted to avoid the anticipated congestion level. Unlike the previous AQM schemes which use the queue length variation as the congestion measure, the proposed scheme uses the variation of the traffic input rate as the congestion measure. By predicting the network congestion level, the proposed scheme can adapt more rapidly to the changing network condition and stabilize the average queue length and its variation even if the traffic input level varies widely. Through ns-2 simulation study in varying network environments, we compare the performance among RED, Adaptive RED (ARED), REM, Predicted AQM (PAQM) and the proposed scheme in terms of average queue length and packet drop rate, and show that the proposed scheme is more adaptive to the varying network conditions and has shorter response time.

• Journal of Advanced Navigation Technology
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• v.18 no.5
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• pp.461-467
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• 2014

Active queue management (AQM) algorithms run on routers and detect incipient congestion by typically monitoring the instantaneous or average queue size. When the average queue size exceeds a certain threshold, AQM algorithms infer congestion on the link and notify the end systems to back off by proactively dropping some of the packets arriving at a router or marking the packets to reduce transmission rate at the sender. Among the existing AQM algorithms, random early detection (RED) is well known as the representative queue-based management scheme by randomizing packet dropping. To reduce the number of timeouts in TCP and queuing delay, maintain high link utilization, and remove bursty traffic biases, the RED considers an average queue size as a degree of congestions. However, RED do not well in the specified networks conditions due to the fixed parameters($P_{max}$ and $TH_{min}$) of RED. This paper addresses a extended RED to be adapted in various networks conditions. By sensing network state, $P_{max}$ and $TH_{min}$ can be automatically changed to proper value and then RED do well in various networks conditions.

• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• 2004.04a
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• pp.43-48
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• 2004

RTP that is proposed supplement of real-time services on internet environment, as Real-time Transport Protocol, is the protocol that for the purpose of sending data of stream type. RTP and RTCP(Real-time Transport Control Protocol) basically work at the same time, RTCP serves with state information of network at present. RTP has important properties of a transport protocol that runs on end-to-end systems and provides demultiplexing. It also offer reliability and protocol-defined flow／congestion control that transport protocol like TCP can not provides. In this paper, we look around concept and construction of Differentiated sen1ice tint run on RTP and by setting parameters of packet transfer method be used CBQ(Class-Based Queuing) for packet transfer on Differentiated service, each service queue controls properly through packet scheduling method, such as WRR(Weighted Round Robin) and PRR(Packet-by-packet Round Robin) all service classes do not experience the starvation and confirm the performance through computer simulation to achieve fairly scheduling.

• Journal of the Korean Professional Engineers Association
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• v.28 no.3
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• pp.121-134
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• 1995

To operate the automatic devices of manufacturing process more effectively and to solve the needs of the resource sharing, network technology is applied to the unit con-trol devices located In common manufacturing zone and operated by connecting them. The network management system, which controls and supervises the network resources in the communication network, is the function that is necessarily required in this net-work. For the stability in this network, this paper measures the variation of data packet length and node number and analyzes the variated value of the token rotation time and waiting time for the network operation. On the evalution of the performance, the data packet length varies from 10 bytes to 100 bytes in the operation of the group manage-ment network, the variation of the token rotation time and waiting time is less than 10 msec. When the variation of the data packet lenhth is less than 50 bytes, differences in token rotation time or waiting time is unnoticable. Since the waiting time in this case is less than 10 msec, response time Is fast enough.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.1B
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• pp.1-12
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• 2002

In the multimedia CDMA mobile communication service, it is required to support various applications, such as voice, video, file transfer, e-mail, and Internet access, with guaranteed QoS. In the mixed traffic environment ,which consists of voice, stream data, and packet data, we analyze the network where preemptive priority is granted to delay-intolerant voice service and a buffer is offered to delay-tolerant stream data service. And, for best-effort packet data service, the access control by transmission permission probability is applied to obtain prominent throughput. To analyze the multimedia CDMA mobile communication system, we build a 2-dimensional markov chain model about prioritized-voice and stream data services and accomplish numerical analysis in combination with packet data traffic based on residual capacity equation.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.4
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• pp.10-21
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• 2003

Recently, the optical subscriber interface module uses the high performance network processor to quickly develop new application services such as MPLS, VPN, RPR and EPON with a short time-to-market. Although a number of vendors are developing the network processor at 2.5Gbps, only the IBM NP4GS3 can provide packet processing with wire-speed at 2.5Gbps. IBM NP4GS3, however, uses its unique speed DASL interface instead of CSIX-Ll interface, which has standardized by M: Forum currently Therefore, we implement an interconnection mechanism to use the switch fabric with CSIX-Ll interface. In this paper, we suggest the architecture and a packet control mechanism supporting interconnection between IBM NP4GS3 DASL and CSIX-Ll switch interface using the common IBM UDASL ASIC and XILINX FPGA.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.5B
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• pp.413-420
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• 2006

HMIPv6 is designed to reduce the signaling load to external network and improve handover speed of MN by including Mobility Anchor Point(MAP) in local handover. However in this case of macro handover, it's just used pervious MIPv6 handover algorithm. So, it occurs packet loss and transmission delay problem. In this paper, we propose the mechanism applying the HMIPv6 for Fast Handover to choose suitable to the condition buffering handover. The condition for the selection is result distance measurement between MN and CN, between MN and NAR. Furthermore, using F-SNOOP protocol, it is possible to improve wireless network performance. Wireless network has high Bit Error Rate(BER) characteristic because of path loss, fading, noise and interference. TCP regards such errors as congestion and starts congestion control. This congestion control makes packet transmission rate low. However, F-SNOOP improves TCP performance based on SNOOP and Freeze TCP that use Zero Window Advertisement(ZWA) message when handoff occurs in wireless network.