• Journal of the Korea Academia-Industrial cooperation Society
• /
• v.14 no.10
• /
• pp.5140-5147
• /
• 2013

This paper presents a new idea of structural design of the optical access network for IPOW(IP over WDM) services. More efficient network can be constructed, because the IP packets are transmitted directly to the WDM without going through an intermediate layer of networks. The wavelength Routing is based on a label switching technology. The ability to transmission of high volume traffics and QoS capability of the optical label switching directly to the end user of the IPOW optical internet networks is provided. As in AON(Active Optical Network) flexible bandwidth on demand subscribers is allocated. By the Simulation of the proposed optical access networks to measure the BER(Bits Error Ratio) at the end of the nodes the network characteristics are analyzed. These results are based on the design of efficient optical network.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.23 no.8
• /
• pp.2096-2105
• /
• 1998

As there can be coincident bursts which may result in congetsion in a node of ATM network, reactive flow control schemes are required to guarantee user's Quality of Service. But, the high speed characteristics of ATM networks make it difficult to control source transmission rate in reacting to congestions in intermediate nodes. Therefore, flow control in Customer Premise Network may be more efficient than end-to-end flow control. In this paper, we propose a management model for flow ontrol in CPN and new Network Access Flow/Congestsion control scheme to utilize efficiently Virtual Path Connection.

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

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.31 no.9B
• /
• pp.777-784
• /
• 2006

In this paper, we propose the PON-based access network based on conventional TDM-PON architecture, which utilizes WDM wavelength channel and bandwidth dynamically. It is also described a dynamic MAC protocol in order to increase the number of subscribers and efficiency of resource utilization. Of particular importance in the proposed approach for MAC protocol is that the wavelength channel and time slot for up/downlink is dynamically allocated according to the required QoS level and the amount of data in data transmission, through the dedicated control channel between OLT and ONU. We evaluate the performance of average packet end-to-end delay in a statistical analysis and numerical analysis. In addition, through simulations with various traffic models, we verified the superior performance of the proposed approach by comparing with the results of other E-PONs.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.13 no.7
• /
• pp.1411-1421
• /
• 2009

As the wireless internet grows exponentially, the recent trend has an increasing demand for wireless network and multimedia services. RTP is used to support the multimedia communication over the Internet and it supports the flexibility and adaptability over a wide range. However, RTP has a limitation that it cannot support end-to-end QoS guarantee in a wireless home network which has low throughput and high delay. In this paper, we propose the architecture of a real-time multimedia communication and design and implement the hybrid flow control in the architecture. The hybrid flow control mechanism is based on modified AIMD using metrics such as the network state information and the user properties. We implement the porposed flow control using JMF to evaluate the performance of the proposed flow control. The experimental results show that the proposed flow control has better performance than the AIMD.

• Journal of KIISE:Computing Practices and Letters
• /
• v.6 no.2
• /
• pp.234-243
• /
• 2000

We design a modified network architecture with condsidering current Internet network model and traffic characteristics, and propose EDCBT(Enhanced Dispersed Core-based Tree) multicast routing protocol, which enhances scalabity, reliability, end-to-end delay and resource utilization EDBCT adopts bidirectional dispersed shared trees and manages both of intradomain and interdomain multicast trees for a multicast group. Each regional multicast tree is estabilshed from its core router and they are interconnected by the operation between border routers on edges of each regional network. As a result, interdomain multicast tree can be easily established. We introduce a new concept named RBTS(Region-based Tree Switching), which dramatically enhances QoS and network utilization. Finally, protocol performance and the effect of core router location are evaluated with MIL3 OPNet network simulator, in terms of end-to-end delay, packet loss and throughput.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.5 no.2
• /
• pp.279-287
• /
• 2001

In this paper, we discuss the scale of subjective speech quality measurement over VoIP(Voice over IP) network which is a component of broadband networks. Objective parameters of multimedia services like PSNR or jitter can easily measured and defined, but these factors are not easily meet the user's perceptual recognition. We suggest the speech quality measurement scale through the subjective measurement for end-to-end speech quality composed of sender-side quality, transmission quality, receiver-side quality, which provide the degree of correctness of representation of speaker, the degree of impairment caused by various factors, the degree of recognition of processed speech, respectively. Also, we examined the proposed method and verify it's availability.

• Journal of Internet Computing and Services
• /
• v.11 no.2
• /
• pp.17-30
• /
• 2010

Many applications in Wireless Sensor Networks require collecting all data without loss from nodes. End-to-End data retransmission, which is used in the Internet for reliable transport, becomes very inefficient in Wireless Sensor Networks, since wireless communication, and constrained resources pose new challenges. We look at factors affecting reliability, and search for efficient combinations of the possible options. This paper proposes an efficient Overhearing based reliable transfer protocol in Wireless Sensor Networks by introducing Selective and implicit Acknowledgement. Finally, it is clarified that the proposed scheme is efficient for reliable data transfer in WSN.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.14 no.7
• /
• pp.3057-3075
• /
• 2020

End-to-end network delay plays an vital role in distributed services. This delay is used to measure QoS (Quality-of-Service). It would be beneficial to know all node-pair delay information, but unfortunately it is not feasible in practice because the use of active probing will cause a quadratic growth in overhead. Alternatively, using the measured network delay to estimate the unknown network delay is an economical method. In this paper, we adopt the state-of-the-art matrix completion technology to better estimate the network delay from limited measurements. Although the number of measurements required for an exact matrix completion is theoretically bounded, it is practically less helpful. Therefore, we propose an online adaptive sampling algorithm to measure network delay in which statistical leverage scores are used to select potential matrix elements. The basic principle behind is to sample the elements with larger leverage scores to keep the traits of important rows or columns in the matrix. The amount of samples is adaptively decided by a proposed stopping condition. Simulation results based on real delay matrix show that compared with the traditional sampling algorithm, our proposed sampling algorithm can provide better performance (smaller estimation error and less convergence pressure) at a lower cost (fewer samples and shorter processing time).

• Journal of the Korea Academia-Industrial cooperation Society
• /
• v.20 no.1
• /
• pp.35-40
• /
• 2019

It is important to constantly provide service in real-time multimedia applications using multicast. Transmission path reconstruction occurs in hybrid multicast using Internet Protocol (IP) multicast and ALM in order to adapt the network status to things like congestion. So, there is a problem in which real-time QoS is reduced, caused by an increase in end-to-end delay. In this paper, we want to solve this problem through multi-layer transmission path construction. In the proposed method, we deploy the control server and application layer overlay host (ALOH) in each multicast domain (MD) for hybrid multicast construction. After the control server receives the control information from an ALOH that joins the MD, it makes a group based on the hop count and sends it to the ALOH in each MD. The ALOH in the MD performs the role of sending the packet to another ALOH and constructs the multi-layered transmission path in order of priority by using control information that is received from the control server and based on the delay between neighboring ALOHs. When congestion occurs in, or is absent from, the ALOH in the upper MD, the ALOH selects the path with the highest priority in order to reduce end-to-end delay. Simulation results show that the proposed method could reduce the end-to-end delay to less than 289 ms, on average, under congestion status.