• Journal of Communications and Networks
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• v.4 no.4
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• pp.266-281
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• 2002

The General Packet Radio Service (GPRS) is being deployed by GSM network operators world-wide, and promises to provide users with “always-on” data access at bandwidths comparable to that of conventional fixed-wire telephone modems. However, many users have found the reality to be rather different, experiencing very disappointing performance when, for example, browsing the web over GPRS. In this paper, we examine the causes, and show how unfortunate interactions between the GPRS link characteristics and TCP/IP protocols lead to poor performance. A performance characterization of the GPRS link-layer is presented, determined through extensive measurements taken over production networks. We present measurements of packet loss rates, bandwidth availability, link stability, and round-trip time. The effect these characteristics have on TCP behavior are examined, demonstrating how they can result in poor link utilization, excessive packet queueing, and slow recovery from packet losses. Further, we show that the HTTP protocol can compound these issues, leading to dire WWW performance. We go on to show how the use of a transparent proxy interposed near the wired-wireless border can be used to alleviate many of these performance issues without requiring changes to either client or server end systems.

• Proceedings of the IEEK Conference
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• 2004.08c
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• pp.549-551
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• 2004

This paper used OPNET to simulate video streaming service a test IP network and MPLS network with the traffic shaping that have with CQ_ LLQ algorithm, LSP of fixed bandwidth, policy of limitation users and measures parameters such as delay, throughput, packet loss. To verify the performance of video streaming service in IP network and MPLS network, two scenario that have same topology and traffic source. One is the simulation for best-effort service in pure IP network. The other is the simulation for QoS-enabled service in MPLS Network. Based on simulation result, the MPLS network with CQ_ LLQ algorithm and fixed LSP show advantage of the video streaming service QoS, specially delay and packet loss

• 제어로봇시스템학회:학술대회논문집
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• 2000.10a
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• pp.133-133
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• 2000

This paper implements Message Service of TCN or IEC 61375-1. TCN is divided into two services, Variable and Message Service. Variable Service uses the broadcasting method with Source Address, but Message Service uses peer-to-peer method with Destination Address and has OSI 7 Layer. In TCN, interface between Transport and Network Layer has not been defined and Meaning of Packet Pool has not been defined exactly. Therefore, this paper proposes the Implementation method for both the interface between Transport and Network Layer and the packet pool for Message Service of TCN.

• 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 Multimedia Society
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• v.13 no.11
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• pp.1667-1676
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• 2010

In this paper, we propose a packet scheduling algorithm that assigns different number of HS-PDSCH(High Speed Primary Downlink Shared Channel) to the service user according to the received signal to interference ratio of CPICH(Common Pilot Channel) and to the traffic characteristics. Assigned channel number is determined by the signal to interference ratio level from CPICH. The highest signal to interference ratio user gets the number of channels based on the signal to interference ratio table and the remained channels are assigned to the other level users. Therefore the proposed scheme can provide the similar maximum service throughput and higher fairness than existing scheduling algorithm. Simulation results show that our algorithm can provide the similar maximum service throughput and higher fairness than MAX C/I algorithm and can also support the higher service throughput than proportional fairness scheme.

• The Journal of the Korea institute of electronic communication sciences
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• v.4 no.2
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• pp.93-100
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• 2009

As the Internet Protocol be widely/rapidly used in packet communication, common carriers are providing the multimedia service(Both direction real-time voice, video conference, remote educational etc.)on the Internet. Also the 070 VoIP (Voice over IP) service is provided by the carriers on the packer network. In order to offer VoIP service in Korea, common carrier has to acquire the optimum level for QoS(quality of service). In this paper, we study on CODEC quality to get a higher QoS for VoIP.

• Proceedings of the IEEK Conference
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• 2002.06a
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• pp.267-270
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• 2002

Downlink Shared Channel (OSCH) is used for high-speed packet service transmission in WCOMA. The power-controlled performances of the OSCH are studied through simulation works. Under various channel conditions and mobile speeds, the simulation results show that current power control scheme in WCDMA, especially for the OSCH power control, has some problems. In this paper, cause of problems is revealed. Also, the quantitative result related to the OSCH power control problem is analyzed.

• Journal of Korean Institute of Industrial Engineers
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• v.28 no.2
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• pp.139-146
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• 2002

The third generation mobile communication needs to provide multimedia service with increased data rates. Thus an efficient allocation of radio and network resources is very important. This paper models the 'burst switching' as an efficient radio resource allocation scheme and the performance is compared to the circuit and packet switching. In burst switching, radio resource is allocated to a call for the duration of data bursts rather than an entire session or a single packet as in the case of circuit and packet switching. After a stream of data burst, if a packet does not arrive during timer2 value ($\tau_{2}$), the channel of physical layer is released and the call stays in suspended state. Again if a packet does not arrive for timerl value ($\tau_{1}$) in the suspended state, the upper layer is also released. Thus the two timer values to minimize the sum of access delay and queuing delay need to be determined. In this paper, we focus on the decision of $\tau_{2}$ which minimizes the access and queueing delay with the assumption that traffic arrivals follow Poison process. The simulation, however, is performed with Pareto distribution which well describes the bursty traffic. The computational results show that the delay and the packet loss probability by the burst switching is dramatically reduced compared to the packet switching.

• Journal of Information Processing Systems
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• v.16 no.5
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• pp.1095-1104
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• 2020

This paper introduces a design scheme of intelligent gas valve management and service system based on Internet. This scheme adds sensor and general packet radio service (GPRS) modules to the traditional gas valve and establishes communication connection between gas valve and the server through wireless packet communication technology, which makes the traditional gas valve have the networking ability. Compared with the traditional gas valve management and service business, the method proposed in this paper is more convenient and efficient.

• Journal of KIISE:Information Networking
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• v.32 no.4
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• pp.495-507
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• 2005

Measuring the end-to-end available bandwidth in the Internet is a useful tool for distributedapplication services or QoS (Quality-of-Service) guarantee. To measure the end-to-end available bandwidth, Single-hop Gap model-based packet train measurement techniques are well-known. However, the error of packet train output gap can happen by network topologies. This error of the output gap causes the inaccuracy of the available bandwidth measurement. In this paper, we propose a new end-to-end available bandwidth measurement technique with accurate capacity measurement and fast convergence methods. To solve the erroneous capacity measurement problem of the back-to-back packet train transmission, we propose a new available bandwidth measurement method by decoupling the capacity measurement with the initial gap of the packet train. Also, we propose a new technique to predict the proper initial gap of the packet train for faster convergence. We evaluate our proposed method by the simulation in various topologies comparing with previous methods.