• Proceedings of the KIEE Conference
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• 2004.11c
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• pp.237-239
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• 2004

Wireless LAN Voice over IP(VoIP) equipment needs Quality-of-Service(QoS) with priority for processing real-time traffic. This paper shows transmit function implementation of wireless LAN(WLANs) media access control(MAC) support VoIP, and it has an advantage of guarantee of QoS and is adaptable to VoIP or mobile wireless equipment. The IEEE 802.11e standard in progress has four queues according to four access categories(AC) for transmit and the MAC transmits the data based on EDCA. The value of AC is from AC0 to AC3 and AC3 has the highest priority. The transmit method implemented at this paper ensure QoS using one transmit FIFO in hardware since real-time traffic data and non real-time traffic data has the different priority. The device driver classifies real-time data and non real-time data and transmit data to hardware with information about data type. The hardware conducts shorter backoff and selects faster AIFS slot for real-time data than it for non real-time data. Therefor It make give the real-time traffic data faster channel access chance than non real-time data and enhances QoS.

• Proceedings of the Korea Contents Association Conference
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• 2009.05a
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• pp.884-888
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• 2009

Vehicular Adhoc Networks (VANET), a Vehicle-to-Infrastucture or Vehicle-to-Vehicle communication technology, is an area that makes more specific use of Mobile Adhoc Networks(MANET). VANET's Quality of Service(QoS) focuses on preventing possible emergencies like car crash from happening by immediately transmitting information to the cars around, while MANET's QoS is being studied for the quality of multimedia data such as Video on Demand(VoD), Video streaming, Voice over IP(VoIP), etc. In this paper, I structure the actual network configuration using Link State Routing(LSR), implement QoS Provisioning Model using Common Open Policy Service(COPS), and suggest more effective k-hop Cluster and inter-domain policy negotiation which fit better to the characteristics of VANET.

• The KIPS Transactions:PartC
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• v.17C no.3
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• pp.299-306
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• 2010

IPTV and VoIP services are considered as killer applications over Mobile WiMAX network, which provideshigh mobility and data rate. Among those which affect the quality of voice in those services, the jitter buffer or playout buffer can compensate the poor voice quality caused by the packet drop due to frequent route change or differences among routes between service endpoints. In this paper, we analyze the correlation between the sleep interval length and jitter buffer size in order to guarantee a predefined level of voice quality. For this purpose, we present an end-to-end delay model considering additional delay incurred by the WiMAX PSC-II sleep mode and a VoIP service quality requirement based on the delay constraints. Through extensive simulation experiments, we also show that the increase of jitter buffer size may degrade the voice quality since it can introduce additional packet drop in the jitter buffer under WiMAX power saving mode.

• Journal of the Korea Society of Computer and Information
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• v.10 no.6 s.38
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• pp.261-268
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• 2005

Recently, Many People have studied in Mobile Ipv6 to provide seamless service. FMIPv6 is designed to reduce the procedure of handover and will be used SIP, VoIP and Mobile Internet moving picture service. But, If mobile node moves fast, it will happen to packet loss and interruption problem in handover procedure. So, In this paper, designed and evaluated traffic control mechanism using WFQ scheduling to reduce the packet loss that occurred to handover procedure.

• Journal of Information Technology Applications and Management
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• v.14 no.4
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• pp.221-233
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• 2007

Which factors should influence consumer consideration to subscribe to Voice over Internet Protocol (VoIP)? Policy issues, managerial concerns, and demographic variables are possible factors. This paper discusses policy and managerial issues regarding VoIP adoption. A model that explains VoIP adoption is proposed and tested. This study analyzes a survey of 750 prospective VoIP users in Korea. The testing is accompanied by logistic regression and discriminant analysis. The results show that trust in VoIP, relative comparison of Quality to fixed service, numbering plan, satisfactions of call Quality and customer services on both fixed and mobile services have impacts on the adoption of VoIP. Implications for VoIP providers and policy makers are presented.

• Journal of Advanced Navigation Technology
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• v.21 no.1
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• pp.113-118
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• 2017

As interest in mobile devices and networks has increased recently, voice over internet protocol (VoIP) service, which is a technology for transmitting voice data using an existing internet protocol (IP) network, has rapidly spread, Cheap voice call service has become possible. As the digital broadcasting service becomes popular, hybrid fiber coaxial (HFC) network technology, which uses broadband cable network through fusion of broadcasting and communication, utilizes existing communication system and network equipment to provide various new services such as interactive broadcasting service. Therefore, if UGS-AD is applied to VoCM and RTPS is applied to MTA in order to guarantee the quality of voice data in actual HFC Internet service network, it is possible to smoothly perform voice data transmission in narrow upstream band which is a problem in actual commercial HFC network We also proposed a method to improve VoIP service by improving QoS of voice data in HFC Internet service network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.6A
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• pp.558-564
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• 2007

3G Long Term Evolution, which aims for various mobile multimedia service provision by enhanced wireless interface, proposes VoIP-based voice service through a Packet Switching (PS) domain. As delay and loss-sensitive VoIP traffic flows through the PS domain, more challenging technical difficulties are expected than in Circuit Switching (CS) domain based VoIP services. Moreover, since 3G LTE, which adopts the OFDM as its physical layer, introduces Physical Resource Block (PRB) as a unit for transmission resources, new types of resource management schemes are needed. This paper proposes a PRB scheduling algorithm of MAC layer for VoIP service in 3G LTE and shows the simulation results. The proposed algorithm has two key parts; dynamic activation of VoIP priority mode to satisfy VoIP QoS requirements and adaptive adjustment of the priority mode duration in order to minimize the degradation of resource utilization.

• Proceedings of the Korea Contents Association Conference
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• 2003.05a
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• pp.383-389
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• 2003

In this paper, we propose a solution, called M_DQDMR, for generating delay-constrained boundwidth-appropriated multicast routing trees to reduce the delay and conserved boundwidth resources of nodes in mobile computing environment. At the current router installation, we according the routing table and the information of link which neighboring rude to guarantee QoS(Quality of Service). When we construct multicast tree, M_DQDMR algorithm dynamically adjusts its appropriate tree construction policy based on how far the destination node from the delay bound and boundwidth our QoS requirement. Through simulations and comparing to another multicast algorithm, we reach a conclusion is that M_DQDMR can simply and dynamically adjusts the construction of multicast tree in hight-speed and conserve boundwidth resources.

• Journal of Communications and Networks
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• v.11 no.1
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• pp.36-41
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• 2009

The IEEE 802.16 wireless metropolitan area network (WMAN) standard is designed to correct expensive communication costs in CDMA-based mobile communication systems and limited coverage problems in wireless LAN systems. Thus, the IEEE 802.16e standard can provide mobile high-speed packet access between mobile stations and the Internet service provider through the base station with cheap communication fees. To efficiently accommodate voice over IP (VoIP) services in IEEE 802.16 systems, an uplink quality of service packet-scheduling scheme is proposed, and its performance is evaluated with an NS-2 network simulator in this paper. Numerical results show that this proposed scheme can increase the system capacity by 100% more than in the unsolicited rand service (UGS) scheme and 30% more than the extended real-time polling service (ertPS) scheme, respectively.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.8
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• pp.1640-1646
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• 2004

In this study, we propose the multiple VoIP service networks for P2P network environments based on portable internet. We found that voiceservice using portable internet is possible for real time services. After we designe the multiple VoIP service networks for P2P network environments based on portable internet, evaluate the performances. from the performance evaluations, we found satisfactory results and so we can expect the possibility of the portable internet services.