• International Journal of Highway Engineering
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• v.15 no.5
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• pp.227-234
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• 2013

PURPOSES : In order to do an improving countermeasures for congestion on the highway with a limited budget, it is very important to select a habitual congestion section effectively. This study is develop CSI(Consitency Service Index) which contained the service for drivers on the highway to select a habitual congestion section. METHODS : By applying the concept of service for the users paying a fee, proposed CSI(Consistency Service Index) to determine habitual delay. CSI is mean that users using the highway road must be provided an environment which can driving more than 80kph, anytime, anywhere. RESULTS : The result applying developed method in this study included most of congestion sections selected by conventional method. but, in some section of existing non-congestion section were included by CSI. The annual average speed and CSI correlation analysis result was high correlation. This result proved that CSI was reflecting road traffic condition well. CONCLUSIONS : It was verified practicality from the delay section of gyeonggi-do area highway. we can judge whether or not to be a habitual congestion in the specific highway and do the traffic improving countermeasures accordingly.

• Journal of applied mathematics & informatics
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• v.28 no.1_2
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• pp.275-282
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• 2010

This paper analyzes a discrete-time bulk-service queue with probabilistic bulk size, where the service process is interrupted by a Markov chain. We study the joint probability generating function of system occupancy and the state of the Markov chain. We derive several performance measures of interest, including average system occupancy and delay distribution.

• The Transactions of the Korean Institute of Electrical Engineers P
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• v.63 no.2
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• pp.81-88
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• 2014

In this paper, an implementation of concurrent backoff delay process on a single chip with IEEE 802.15.4 hardware and 8051 processor core that can be used for analyzing the interference on IEEE 802.15.4 channels due to WiFi traffics is studied. The backoff delay process of IEEE 802.15.4 CSMA-CA algorithm is explained. The characteristics of random number generator, timer, and CCA register included in the single chip are described with their control procedure in order to implement the process. A concurrent backoff delay process to evaluate multiple IEEE 802.15.4 channels is proposed, and a method to service the associated tasks at sequentially ordered backoff delay events occurring on the channels is explained. For the implementation of the concurrent backoff delay process on a single chip IEEE 802.15.4 hardware, the elements for the single channel backoff delay process and their control procedure are used to be extended to multiple channels with little modification. The medium access delay on each channel, which is available after execution of the concurrent backoff delay process, is displayed on the LCD of an IEEE 802.15.4 channel analyzer. The experimental results show that we can easily identify the interference on IEEE 802.15.4 channels caused by WiFi traffics in comparison with the way displaying measured channel powers.

• Journal of Internet Computing and Services
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• v.2 no.2
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• pp.23-36
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• 2001

To support real-time multicast, the existing DS network may use the EF PHB for transmission, However, because of the dynamic characteristics of the multicast flow and the features of the EF PHB. it is very difficult to make a provision of resources or to manage the network when EF multicast flows increase, Also, the existing LBE PHB can not guarantee the transmission delay bounds required for the real-time multicast traffic, In this paper. a new concept of PHB. namely, low-delay multicast PHB, that may guarantee approximately the per-hop delay is proposed, For the low-delay multicast PHB, router architectures are devised, The queue management algorithm on a low-delay multicast queue and a scheduling method are also devised. This new architecture of the LDM AF class is simple to be implemented.

• Journal of the Korea Society of Computer and Information
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• v.13 no.1
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• pp.175-183
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• 2008

QoS-support technology in networks is based on measuring QoS metrics which reflect a magnitude of stability and performance. The one-way delay measurement of the QoS metrics especially requires a guarantee of clock synchronization between end-to-end hosts. However, the hosts in networks have a relative or absolute difference in clock time by reason of clock offsets. flock skews and clock adjustments. In this paper, we present a theorem, methods and simulation results of one-way delay and clock offset estimations between end-to-end hosts. The proposed theorem is a relationship between one-way delay, one-way delay variation and round-trip time And we show that the estimation error is mathematically smaller than a quarter of round-trip time.

• ETRI Journal
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• v.37 no.2
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• pp.380-386
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• 2015

Guaranteeing quality of service over a multihop wireless network is difficult because end-to-end (ETE) delay is accumulated at each hop in a multihop flow. Recently, research has been conducted on network coding (NC) schemes as an alternative mechanism to significantly increase the utilization of valuable resources in multihop wireless networks. This paper proposes a new section-based joint NC and scheduling scheme that can reduce ETE delay and enhance resource efficiency in a multihop wireless network. Next, this paper derives the average ETE delay of the proposed scheme and simulates a TDMA network where the proposed scheme is deployed. Finally, this paper compares the performance of the proposed scheme with that of the conventional sequential scheduling scheme. From the performance analysis and simulation results, the proposed scheme gives more delay-and energy-efficient slot assignments even if the NC operation is applied, resulting in a use of fewer network resources and a reduction in ETE delay.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.10B
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• pp.1852-1860
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• 1999

This paper studied performance of the AAL2 for variable rate real time services in ATM network with discrete-time simulation model. In this simulation, input parameters are packet fill delay for AAL2 PDU generation, guard time for ATM cell generation, burstness and number of channels. Though variation of the above mentioned parameters, we obtained end-to end delay variations and throughput, analyzed performance effect of the each parameter for voice packet service.

• Proceedings of the IEEK Conference
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• 1999.11a
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• pp.121-124
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• 1999

In a packet-switched wireless cellular network, a packet destined to a mobile station is queued at a base station and then broadcast over the base station's cell. When an active mobile station leaves a cell, there re-main packets which are destined to the mobile and not yet delivered to it at the cell's base station. For application which are sensitive to packet losses, such back- logged packets must be forwarded to the new base station. Otherwise, an end-to-end retransmission may be required. However, an increase in packet delay is incurred by employing the packet forward scheme, since a packet may be forwarded many times before it is delivered to the destined mobile station. For an enhanced quality-of-service level, it is preferred to reduce tile packet delay time. In this paper, we develop an analytical approximation method for deriving mean packet delay times. Using the approximation and simulation methods, we investigate the effect of network parameters on the packet delay time.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.6 no.4
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• pp.1006-1025
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• 2012

Improving the quality of service in IP networks is a major challenge for real-time voice communications. In particular, packet arrival-delay variation, so-called "jitter," is one of the main factors that degrade the quality of voice in mobile devices with the voice-over Internet protocol (VoIP). To resolve this issue, a receiver-based enhanced timing recovery algorithm combined with active jitter estimation is proposed. The proposed algorithm copes with the effect of transmission jitter by expanding or compressing each packet according to the predicted network delay and variations. Additionally, the active network jitter estimation incorporates rapid detection of delay spikes and reacts to changes in network conditions. Extensive simulations have shown that the proposed algorithm delivers high voice quality by pursuing an optimal trade-off between average buffering delay and packet loss rate.

• The Transactions of the Korea Information Processing Society
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• v.6 no.11
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• pp.3060-3069
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• 1999

The existed transmissions have PDH and SDH transmission system, which are based on PSTN network. But as the future ATM network is going to be constructed, it is very necessary for ATM network to interwork with PSTN for telephony service. The international standardization organizations like ITU-T, ATM Forum offered CES, VTOA1, VTOA2 and so on for interworking with PSTN service, which spend many network resources and considerable transmission delay. So in this paper, new interworking PSTN for transmission will be proposed to lessen the transmission delay and the network resources.