• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.13 no.8
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• pp.764-768
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• 2002

In this paper, a double balanced gilbert cell MMIC mixer was realized in Tachyonics SiGe HBT technology. The fabricated mixer has 17 dB conversion gain, 9.8 dB noise figure, -4.2 dBm output 1 dB compression point, -27 dBc RF to IF isolation, and the good input, output matching characteristics. It draws 10 mA from a 3 V supply. The simulation and the measured results are closer to each other, which confirms accuracy of the model library and reliability of the process.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.1B
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• pp.52-63
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• 2004

Packet loss process of cdma2000 mobile Internet channel deployed in Korea is modeled as a two state Markov process known as Gilbert model. This paper proposes the procedures to derive four parameters of the our modified Gilbert model from packet loss trace taken from two major cdma2000 networks in Korea. These four parameters are derived in various situations, that is, with fixed and moving terminals, in open field and urban areas. They can be used to produce synthetic packet loss patterns for study of the channel. Moreover, if they are calculated on-line during multimedia service, they can be used to make loss protection controls adaptive to network condition.

• International Journal of Internet, Broadcasting and Communication
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• v.15 no.3
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• pp.43-51
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• 2023

Slotted ALOHA (S-ALOHA) is a classical medium access control protocol widely used in multiple access communication networks, supporting distributed random access without the need for a central controller. Although stability and delay have been extensively studied in existing works, most of these studies have assumed ideal channel conditions or independent fading, and the impact of time-correlated wireless channels has been less addressed. In this paper, we investigate the queueing delay performance in S-ALOHA networks under time-correlated channel conditions by utilizing a Gilbert-Elliott model. Through simulation studies, we demonstrate how temporal correlation in the wireless channel affects the queueing delay performance. We find that stronger temporal correlation leads to increased variability in queue length, a larger probability of having queue overflows, and higher congestion levels in the S-ALOHA network. Consequently, there is an increase in the average queueing delay, even under a light traffic load. With these findings, we provide valuable insights into the queueing delay performance of S-ALOHA networks, supplementing the existing understanding of delay in S-ALOHA networks.

• 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.

• ETRI Journal
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• v.25 no.5
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• pp.356-368
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• 2003

This paper analyzes the characteristics of packet losses in mobile computing environments based on the Gilbert model and then describes a mechanism that can recover the lost audio packets using redundant data. Using information periodically reported by a receiver, the sender dynamically adjusts the amount and offset values of redundant data with the constraint of minimizing the bandwidth consumption of wireless links. Since mobile computing environments can be often characterized by frequent and consecutive packet losses, loss recovery mechanism need to deal efficiently with both random and consecutive packet losses. To achieve this, the suggested mechanism uses relatively large, discontinuous exponential offset values. That gives the same effect as using both the sequential and interleaving redundant information. To verify the effectiveness of the mechanism, we extended and implemented RTP/RTCP and applications. The experimental results show that our mechanism, with an exponential offset, achieves a remarkably low complete packet loss rate and adapts dynamically to the fluctuation of the packet loss pattern in mobile computing environments.

• The Journal of the Acoustical Society of Korea
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• v.28 no.2E
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• pp.35-42
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• 2009

In this paper, we present criteria of objective measurement of speech quality to provide the mobile-VoIP services efficiently over wireless mobile internet. The mobile-VoIP service, which is based on mobility and is error-prone compared to conventional VoIP over wired network, is about to be launched, but there have not been adequate quality indexes and the Quality of Service (QoS) standards for evaluating speech quality of Mobile-VoIP. In addition, there are many factors influencing on the speech quality in packet network of which packet loss contribute directly to the overall voice communication quality. For this reason, we adopt the Gilbert-Elliot Channel Model for modeling packet network based on IP and assess the voice quality through the objective speech method of ITU-T P. 862 PESQ and ITU-T P. 862.1 MOS-LQO under various packet loss rates in the transmission channel environments. Our simulation results address the specific criteria and QoS for the mobile-VoIP services in terms of the various packet loss environments.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.10B
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• pp.1824-1831
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• 1999

In this paper, the data recovery performance of systematic erasure codes in burst loss environments is analyzed and the estimation method of redundant data according to loss characteristics is suggested. The burstness of packet loss is modeled by Gilbert model, and the performance of proposed packet loss recovery method in the case of using systematic erasure code is analyzed based on previous study on the loss recovery in the case of using erasure code. The required redundancy data fitting method for systematic erasure code in the condition of given loss property is suggested in the consideration of packet loss characteristics such as average packet loss rate and average loss length.

• The Journal of the Acoustical Society of Korea
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• v.29 no.5
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• pp.347-353
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• 2010

In VoIP system, the speech quality is seriously degraded due to packet loss, and the degree of degradation by each packet loss depends on the characteristics of the corresponding packet. Therefore, it is possible to improve the speech quality of VoIP by selectively controlling the packet to be lost during transmission based on the expected degradation by the loss of each packet. In this paper, a new scheme to improve speech quality of DiffServ-based VoIP by assigning priority to each packet is proposed, and a method to determine the priority of each packet is developed. The performance of proposed method was measured in packet loss environment based on Gilbert model, and it was verified both objectively and subjectively that the speech quality is improved by the proposed method.

• Computers and Concrete
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• v.1 no.2
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• pp.151-168
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• 2004

A two-dimensional nonlinear finite element model is developed to simulate time-dependent cracking of reinforced concrete members under service loads. To predict localized cracking, the crack band model is employed to model individual crack opening. In conjunction with the crack band model, a bond-interface element is used to model the slip between concrete and reinforcing steel permitting large slip displacements between the concrete element nodes and the steel truss element nodes at crack openings. The time-dependent effects of concrete creep and shrinkage are incorporated into the smeared crack model as inelastic pre-strains in an iterative solution procedure. Two test examples are shown to verify the finite element model with good agreement between the model and the observed test results.

• The Journal of the Korea Contents Association
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• v.5 no.4
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• pp.254-264
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• 2005

In this paper, an advanced E-Model was proposed in order to overcome disadvantages of conventional method. A new model makes the accurate VoIP call quality assessment possible by applying the burst packet loss and recency effect. In order to assess the performance of this advanced E-Model, we gained the estimated MOS value from NR(Network R) value and UR(User R) value resulted from the burst packet loss values by Gilbert Model. Through simulations and comparisons with conventional models such as MOS, PESQ, and I-Model, we reach a conclusion that advanced E-Model is more accurate and reliable method than conventional models.