• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2011.04a
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• pp.808-813
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• 2011

There are hundreds of thousands call center workers wearing acoustic device. However, researches and noise exposure measurements on the noise transmitted from acoustic devices have seldom been performed due to the difficulty of measurement and to the absence of the measuring method in Korea. The aim of this study is to set up management measures to protect hearing loss on the call operator by acquiring measurement data of noise transmitted from the headset Noise exposure measurements of 17 operators were performed in 7 call centers and Head and Torso Simulator method in compliance with the ISO Standard 11904-2 was used for the measurement of noise transmitted from the headset Sound pressure levels(SPL) transmitted from the headset were 73.2~86 dB(A). The operator exposed to the highest SPL set up his volume control at 9 which was the highest volume level. The volume control level, adjustable from 1 to 9, could be identified 12 out of 17 operators and the range of volume levels was 4.5~9. As a result of Pearson Correlation Analysis, the correlation between volume level and SPL transmitted from the headset showed high relation as significance at the 0.672 level(p<0.05). To protect hearing loss of call center operators, it is more practical and effective measure to limit the volume level below the noise exposure level, i.e. 85 dB(A), rather than to carry out noise monitoring considering cost-effective aspect.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.21 no.7
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• pp.615-621
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• 2011

There are hundreds of thousands call center workers wearing acoustic device. However, researches and noise exposure measurements on the noise transmitted from acoustic devices have seldom been performed due to the difficulty of measurement and to the absence of the measuring method in Korea. The aim of this study is to set up management measures to protect hearing loss on the call operator by acquiring measurement data of noise transmitted from the headset. Noise exposure measurements of 17 operators were performed in 7 call centers and head and Torso simulator method in compliance with the ISO standard 11904-2 was used for the measurement of noise transmitted from the headset. Sound pressure levels(SPL) transmitted from the headset were 73.2~86 dB(A). The operator exposed to the highest SPL set up his volume control at 9 which was the highest volume level. The volume control level, adjustable from 1 to 9, could be identified 12 out of 17 operators and the range of volume levels was 4.5~9. As a result of pearson correlation analysis, the correlation between volume level and SPL transmitted from the headset showed high relation as significance at the 0.672 level(p<0.05). To protect hearing loss of call center operators, it is more practical and effective measure to limit the volume level below the noise exposure level, i.e. 85 dB(A), rather than to carry out noise monitoring considering cost-effective aspect.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.6
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• pp.1480-1489
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• 1996

In ATM networks there are two methods in traffic control as schemes advancing the quality of service. One is reactive control after congestion and the other which is generally recommended, is preventive control before congestion, including connection admission control on call leel and usage parameter control, network parameter control, priority control and congestion control on cell level. In particular, usage parameter control is required for restricting the peak cell rate of bursy tracffic to the parameter negotiated at call set-up phase since the peak cell rate significantly influences the network quality of service. The scheme for progressing quality of service by usage parameter control is themethod using VSA(Virtual Scheduling Algorlithm) recommended ITU-T. The method using VSSA(Virtual Scheduling Suggested Algorlithm) in this paper is suggested by considering cell delay variation and token rate of leaky bucket, compared VSA and VSANT(Virtual Scheduling Algolithm with No Tolerance) with VSSA which polices violated cell probability of conformed peak cell rate and intentionally excessive peak cell rate. VSSA method using IPP(Interruped Poisson Process) model of input traffic source showed more quality of service than VSA and VSANT methods as usage parameter control because the suggested method reduced the violated cell probability of contformed peak cell rate and intentionally excessive peak cell rate.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.4
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• pp.957-966
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• 1998

In this paper an imrpoved Signal-to-Interference ratio(SIR)-based call admission control(CAC) algorithm for DS-CDMA cellular system is proposed and its performance is analyzed. This algorithm uses Residual-Capacity defined asthe additional number of initial calls that a base station can accept such that system-wide outage probability will guaranteed to remain below a certain level. the residual capcity at each cell is calculated according to the reverse-link SIR measured not only at the home cell but also the adjacent cells. Then the adjacent cell interference-coupling coefficient .betha. is used. In this work we propose an improved algorithm that .betha. varies according to the traffic load of the home cell. The influence of traffic condition on system performance, namely blocking probability and outage probability, is then examined via simulation. The performance of the improved algorithm is evaluated both under homogeneous and hot spot traffic loads. The results show that the improved algorithm outperforms conventional algorithms under all load values. Under over-load situation, especially, the improved algorithm gives almost constant outage performance the QoS(quality of service) can be guranted.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.7A
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• pp.987-995
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• 1999

The handoff is a distinctive characteristic of the mobile networks. In the CDMA systems, if base stations support multiple Frequency Assignment (FA), they provide both soft handoff and hard handoff. Under the CDMA environments, the soft handoff guarantees the favorable service quality and the continuity of call connection without interruption, and increases the service capacity of the base stations. This paper proposes call connection control schemes with handoff queue for supporting efficient handoff processing. The proposed schemes are divided into two categories: single handoff queue scheme and multiple handoff queue scheme. We analyze the performance of the proposed call connection control schemes using numerical analysis. From the analysis results, we can say that it is more desirable to avoid hard handoff as long as handoff queues are used. When a single handoff queue used, adaptive scheme that properly mixes avoidable and avoidable hard handoff method under the given traffic condition is more desirable. In case that multiple handoff queues are used, the suitable trade-off needs to be developed between handoff blocking probability and hard handoff probability to guarantee a given blocking probability threshold.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.12
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• pp.106-115
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• 2003

Diverse multimedia services will be deployed at hand on 3G-and-beyond multi-service CDMA systems in order to satisfy different quality of service (QoS) according to traffic types. In order to use appropriate resources efficiently the call admission control (CAC) as a major resource control mechanism needs to be used to take care of efficient utilization of limited resources. In this paper, we propose a QoS-aware CAC (QCAC) that is enabled to provide service fairness and service differentiation in accordance with priority order and that applies the different thresholds in received power considering different QoS requirements such as different bit error rates (BER) when adopting total received power as the ceil load estimation. The proposed QCAC calculates the different thresholds of the different traffic types based on different required BER applies it for admission policy, and can get service fairness and differentiation in terms of call dropping probability as a main performance metric. The QCAC is aware of the QoS requirement per traffic type and allows admission discrimination according to traffic types in order to minimize the probability of QoS violation. Also the CAC needs to consider the resource allocation schemes such as complete sharing (CS), complete partitioning (CP), and priority sharing(PS) in order to provide fairness and service differentiation among traffic types. Among them, PS is closely related with the proposed QCAC having differently calculated threshold per each traffic type according to traffic priority orders.

• Journal of The Korean Society of Clinical Toxicology
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• v.12 no.1
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• pp.1-7
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• 2014

Purpose: The purpose of this study is to find differences in the demographics of toxic exposed patients and substance between call based poison information data and hospital based poison information data. Methods: Seoul 1339 call-response data were used as call based poison data and toxic related injury surveillance data of the Korean center for disease control and prevention (KCDC) were used as hospital based poison data. Age, sex, the kind of exposed substance, reasons for exposure, and exposure routes were compared between two data sets. We analyzed the presence or not of documentation on the name and amount of exposed substance, symptoms after exposure in call based poison data. Results: Seoul1339 poison data included a total of 2260 information related to toxic exposure and KCDC poison data included 5650 poison cases. There was no difference in sexual distribution. Pediatric exposure and accidental exposure were more common in call based poison data. The most common exposed substances were household products in call based poison data and medicines in hospital based poison data, respectively. Documents regarding amount and time of toxic exposure and symptoms after toxic exposure were not recorded exactly in call based poison data. Conclusion: There were significant differences in age, reasons for toxic exposure, and the kinds of exposed substances. Poison information data from both pre-hospital and hospital must be considered.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.1B
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• pp.101-108
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• 2004

SIP(Session Initiation Protocol) is a suitable protocol for supporting Internet telephony services and currently market requirements fur developing supplementary telephony services such as unconditional call forwarding, call forwarding on busy or no answer, call filtering services have recently grown. CPL(Call Processing Language) is a standard technology that can be used to describe and control internet telephony services. In this paper, we describe the CPL system for supplementary Internet telephony services using SIP as an application level call signaling protocol. Those supplementary services are composed of CPL client which is a SIP UA, SIP Proxy server, Registrar and CPL server In this paper, we describe the design and implementation of the CPL server system in detail which is developed in Linux 7.2 using C and C++ programming languages.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.10
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• pp.2219-2235
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• 1997

The endpoints and exchanges involved in a call can establish heterogeneous connection each other according to required bandwidth and bearer availability in multimedia communication. and as participating users are increased, call setup delay must not be exceedingly increased. In this paepr, we propose DMMSP(distributed multiparty multiconnection signaling protocol) which can support heterogeneous connections and multimedia communications. DMMSP gets over a limitation of existing B-ISDN protocol and provides signaling capabilities to support various kinds of conncetions, and each endpint can setup individual connection with bearer availability. Moreover since DMMSP separatively performs call processing in distributed scheme, call setup delay can be minimized. We present call control procedures which can be applied to existing B-ISDN protocol and DMMSP respectively. We take multiparty multimedia conference call as an example thta is applied to the existing B-ISDN protocol and DMMSP, and compare and quantitatively analyze each procedures.

• The Journal of the Korea Contents Association
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• v.7 no.3
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• pp.34-41
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• 2007

The presence service which provides user's presence information by subscription and notification is one of SIP(session initiation protocol) extension services, and it is used importantly in VoIP(Voice over IP) and Instant Messaging service. In this paper, we propose a new method in which users can combine and control presence service and call processing services in various ways by extending call processing language, and only changed parts of the presence information are published instead of full presence information document. Each user registers full presence information document with his own call processing script during the first publication to a presence server. The presence server executes these call processing scripts, so it can provide various services with combination of presence service and call processing services during the presence subscriptions and notifications. Afterwards, users publish only changed parts of the presence information and the presence server notify only these changed parts to watchers. Therefore the efficiency of the overall system can be improved. The performance of our proposed model is evaluated by experiments.