• Management & Information Systems Review
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• v.28 no.3
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• pp.191-210
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• 2009

Internet was first designed only for sending data, but as the time passed, internet started to evolve into a broadband multi-media web that is capable of transmitting sound, video, high-capacity data and more due to the demands of internet users and the rapid changing internet-communication technology. Domestically, in January, 2000 Saerom C&T, launched a free VoIP, but due to limited ways of conversation(PC to PC) and absence of a revenue model, and bad speech quality, it had hit it's growth limit. This research studied VoIP based on technological enhancement in super-speed internet. According to IDC, domestic internet market's size was 80,800 million in 2008, and it formed a percentage of 12.5% out of the whole sound-communication market. in case of VoIP, it is able to maximize it's profit by connecting cable and wireless network, also it has a chance of becoming firm-concentrated monopoly market by fusing with IPTV. Considering the fact that our country is insignificant in MVNO revitalization, regulating organizations will play a significant roll on regulating profit between large and small businesses. Further research should be done to give VoIP a secure footing to prosper and become popularized.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.1
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• pp.145-152
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• 2009

IEEE 802.16e standard, a kind of WMAN standard, was established to support data services with cheaper cost to mobile users than traditional mobile communications system and wireless LAN system can do. In this paper, we propose an uplink QoS packet-scheduler for VoIP service which can be installed in IEEE 802.16 system and evaluate its performance with NS-2 network simulator. The proposed QoS packet-scheduler consists of three procedures: determining scheduler interval, determining the amount of resource assignment, and deciding which mobile station the base station should serve first among multiple mobile stations. According to numerical results, the proposed QoS packet-scheduler can provide more increased system capacity by 220% than UGS service scheme does and by 25 % than ertPS service scheme does.

• Journal of Korean Institute of Industrial Engineers
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• v.32 no.1
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• pp.1-8
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• 2006

A modified PCF(Point Coordination Function) protocol with the optimal polling sequence is defined in detail and shown to improve the efficiency of the conventional PCF protocol in IEEE 802.11 wireless LAN standard. The problem for the optimal polling sequence is formulated as TSP(Travelling Salesman Problem) with the distance values of 1's or 0's. Numerical examples show that the optimal polling sequence increases the capacity of the real-time service such as VoIP(Voice over Internet Protocol).

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38B no.5
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• pp.355-367
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• 2013

In the Policy and Charging Control(PCC) architecture of the 3GPP and WiMAX forum, the Policy and Charging Rule Function(PCRF) controls and configures the policies of quality of services(QoS) and accounting to a Policy and Charging Rule Enforcement Function(PCEF). In this paper, we categorize the errors into three types that include a disconnection with peer entity, a delayed or no response and a response with failure result, and propose a scheme to automatically synchronize the data between the PCRF and the PCEF for each type of error. Also, we conduct a quantitative analysis to see how much it can help to maintain the cell capacity of the mobile VoIP service in a WiBro network, which is implemented in Korea according to the standards of WiMAX. We can see that our proposed scheme improves the performances of PCC by preventing the waste of radio resources, the overload of systems, the memory leakage of servers, the system shutdown, and so on. We have validated the scheme by implementation, and reflected it in WiMAX standard documents.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.9
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• pp.3159-3176
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• 2014

This work presents an analysis of the buffer features of an access router, especially the size, the impact on delay and the packet loss rate. In particular, we study how these features can affect the Quality of Service (QoS) of multimedia applications when generating traffic bursts in local networks. First, we show how in a typical SME (Small and Medium Enterprise) network in which several multimedia flows (VoIP, videoconferencing and video surveillance) share access, the upgrade of the bandwidth of the internal network may cause the appearance of a significant amount of packet loss caused by buffer overflow. Secondly, the study shows that the bursty nature of the traffic in some applications traffic (video surveillance) may impair their QoS and that of other services (VoIP and videoconferencing), especially when a certain number of bursts overlap. Various tests have been developed with the aim of characterizing the problems that may appear when network capacity is increased in these scenarios. In some cases, especially when applications generating bursty traffic are present, increasing the network speed may lead to a deterioration in the quality. It has been found that the cause of this quality degradation is buffer overflow, which depends on the bandwidth relationship between the access and the internal networks. Besides, it has been necessary to describe the packet loss distribution by means of a histogram since, although most of the communications present good QoS results, a few of them have worse outcomes. Finally, in order to complete the study we present the MOS results for VoIP calculated from the delay and packet loss rate.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.2
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• pp.254-259
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• 2008

Mesh WLANs, which consist of wireless mesh routers connecting each other in a mesh topology and self-operate after their autoconfiguration, have several advantages in convenience, swiftness and flexibility of deployment and operation over existing WLANs the expansions of which are done by connecting the APs with wires. However, many technical issues still remain to be solved. Among them, network performance degradations due to the interference between the adjacent hops in multi-hop mesh WLANs, and the reusability of the existing wireless network protocols are critical problems to be answered. This work evaluates the VoIP support performance of IEEE 802.11a/g-based mesh WLANs with multiple wireless interfaces with simulations. The results show that there exit an unfairness in VoIP packet delay performances among mobile routers located at different hops, and that although the capacity of the admitted calls can be increased by increasing the size of voice packet payload it is far less than the expected one. This suggests that the existing 802.11 MAC protocols have their limitation when applied in mesh networks and their enhancement or even a newer one nay be required.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.1
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• pp.92-100
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• 2016

It is informed that the Voice over LTE(VoLTE) which serves voice and message on IP networks is better in terms of economies of scale than the legacy voice service on 2G/3G circuit-switched networks because of its technological and cost efficiency. In addition, services of voice and data are running on a single LTE network and as a result VoLTE has the more economies of scope. But, there is no study about how much technology-efficiency VoLTE has compared to circuit-based voice service and how much voice charge can be reduced as VoLTE grows up. This paper analyzes empirically cost-efficiency of VoLTE against circuit-based voice service and quantifies the reduction of voice charge as 2G/3G voice traffic shifts to VoLTE. The results describe the first is that the average cost of the total voice traffic rises shortly just after the investment of LTE network for providing VoLTE but it will soon have a capacity available to reduce the charge due to VoLTE's outstanding cost efficiency on the assumption that voice traffic is fixed, and the second is that the charge can be cut to 60% of the current rate in case of all the voice traffic moves to VoLTE. The latter proves partially the validation of data-focusing pricing plan. Our results are expected to become basic data for network operators' establishing pricing strategies and for policy makers' inducing price cutting.

• The Journal of the Korea Contents Association
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• v.4 no.4
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• pp.123-131
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• 2004

Internet is absolutely contributed to information technology revolution nowadays. Internet services such as voice and data, etc. are provided home or small office via home gateway. The development of communication equipment via home gateway is implemented rapidly, and its product various. This paper presents the design and implementation of the VoIP equipment including the telephone function based on the embedded environment and being connected to the home gateway and the PC because of taking 2-ethernet LAN ports. As developing environment, the STLC1502 developed at ST Microelectronics as single chip solution, VxWorks as RTOS, and C language as coding mechanism are used. The verification of the developed systems for the voice test is carried out for the gatekeeper via Internet. The performance parameter is considered as the call processing capacity measuring the time of the call setup and clearing, and the data processing capacity for the file transfer. As a call setup and clearing is about 95ms, the call processing capacity is about 10 calls per second. The data processing capacity is 5.7Mbps in case of file transfer of server and client environment. Therefore the performance result is satisfied in the aspect of the call processing time and the data transfer time in Internet.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.7
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• pp.1359-1366
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• 2007

The recent, a large capacity of communication is required in order to it is in proportion to capacity of information communication increase and to satisfy a demand because of the demand about Internet, a multimedia service of Video of internet protocol(VoIP), Audio/Video steaming. DWDM(Dense Wavelength Division Multiplexing) technique has been emerged as the method of solving it without additional optical fiber network building and hish-speed equipment. Therefore this thesis proposed the optical filter of fiber/multilayer slab coupled structure combining it to multilayer slab waveguide by polishing the cladding of one side of fiber to design the optical filter having these functions. The optical filter proposed as the simulation result satisfies DWDM filter characteristic; 1) when the separation distance between fiber and slab is $4.15{\mu}m$ at the communication window of $1.5{\mu}m$, the polarization independence is 65nm, 2) when the center wavelengths about TM mode and TE mode are each ${\lambda}_0=1.54946\;{\mu}m$ and ${\lambda}_0=1.6144\;{\mu}m$ and, FWHM(Full at Half Maximun) is 0.1nm.