• Journal of Communications and Networks
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• v.11 no.3
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• pp.297-305
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• 2009

Multiprotocol label switching (MPLS) networks require dynamic flow admission control to guarantee end-to-end quality of service (QoS) for each Internet protocol (IP) traffic flow. In this paper, we propose to tackle the joint routing and admission control problem for the IP traffic flows in MPLS networks without rerouting already admitted flows. We propose two mathematical programming models for this problem. The first model includes end-to-end delay constraints and the second one, end-to-end packet loss constraints. These end-to-end QoS constraints are imposed not only for the new traffic flow, but also for all already admitted flows in the network. The objective function of both models is to minimize the end-to-end delay for the new flow. Numerical results show that considering end-to-end delay (or packet loss) constraints for all flows has a small impact on the flow blocking rate. Moreover, we reduces significantly the mean end-to-end delay (or the mean packet loss rate) and the proposed approach is able to make its decision within 250 msec.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.6B
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• pp.1113-1122
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• 1999

In this paper, we propose MSS(Multicast Synchronization Server) and QCS(QoS Conversion Server) models that can support IP multicast and QoS(Quality of Service) over the ATM network more efficiently. In the MSS model, it is possible to establish shortcut VCs(Virtual Circuits) among all hosts in the ATM network and to transfer multicast data at high speed. Also the MSS model is more scalable, because the number of inter-cluster VCs needed in the MSS model is less than that of EARTH. In the QCS model, ATM switch is modified to support one QoS service and best effort service through the one point-to-multipoint VC at the same time so required network resources are reduced, and dedicated server is used for QoS conversion to accept heterogeneous receivers more efficiently. In this Paper, the proposed MSS model and QCS model have solved both efficiency problem and scalability problem. It is proved through the comparison of the VCs required in each model.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.4B
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• pp.368-376
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• 2009

Over recent years, the development of multimedia devices has meant that a wider multimedia streaming service can be supported, and there are now many ways in which TV channels can communicate with different terminals. Generally, scalable video streaming is known to provide more efficient channel capacity than simulcast video streaming. Simulcast video streaming requires a large network bandwidth for all resolutions, but scalable video streaming needs only one flow for all resolutions. On the contrary, to preserve the same video quality, SVC(Sealable Video Coding) needs a higher bit-rate than AVC(non-layered Video Coding) due to the coding penalty($10%{\sim}30%$). In previous research, scalable video streaming has been compared with simulcast video streaming for network channel capacity, in two-user simulation environments. The simulation results show that the channel capacity of SVC is $16{\sim}20%$ smaller than AVC, but scalable video streaming is not efficient because of the limit of the present network framework. In this paper, we propose a new network framework with a new router using EDE(Extraction Decision Engine) and SVC Extractor to improve network performance. In addition, we compare the SVC environment in the proposed framework with previous research on the same way subject. The proposed network framework shows a channel capacity 50%(maximum) lower than that found in previous research studies.

• Journal of the Korea Society of Computer and Information
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• v.18 no.4
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• pp.87-96
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• 2013

In this paper, we suggest redesigned RTP protocol that is able to perform encryption of VoIP media information for single private network and between the different private networks. And we conduct a test for performance analysis. Such as SRTP or ZRTP methods have been used for VoIP media encryption. But, the existing encryption techniques have problem that can not perform end-to-end encryption between different private networks. In order to solve this problem, in this paper, we redesign RTP protocol. Redesigned RTP includes all information for encryption of VoIP media. Therefore the encryption is not affected by modification of SIP and SDP information that occurred in gateway. Also, redesigned RTP includes code for whether or not to apply encryption. By using the code, modification of RTP header from gateway prevents. As a result, redesigned RTP maintain the integrity and the RTP is able to perform encryption between the different private networks. Also, we conduct a test for performance analysis of SRTP, ZRTP and redesigned RTP.

• 대한공업교육학회지
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• v.32 no.1
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• pp.134-152
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• 2007

This study deals with the development of computer program which can restrict students to access to the unallowable web sites during the Internet based class. Our suggested program can find the student's access list to the unallowable sites, display it on the teacher's computer screen. Through the limitation of the student's access, teacher can enhance the efficiency of class and fulfill his educational purpose for the class. The use of our results leads to the effective and safe utilization of the Internet as the teaching tools in the class. Meanwhile, the typical method is to turn off the LAN (Local Area Network) power in order to limit the student's access to the unallowable web sites. Our program has been developed on the Linux operating systems in the small network environment. The program includes following five functions: the translation function to change the domain name into the IP(Internet Protocol) address, the search function to find the active students' computers, the packet snoop to capture the ongoing packets and investigate their contents, the comparison function to compare the captured packet contents with the predefined access restriction IP address list, and the restriction function to limit the network access when the destination IP address is equal to the IP address in the access restriction list. Our program can capture all passing packets through the computer laboratory in real time and exactly. In addition, it provides teacher's computer screen with the all relation information of students' access to the unallowable sites. Thus, teacher can limit the student's unallowable access immediately. The proposed program can be applied to the small network of the elementary, junior and senior high school. Our research results make a contribution toward the effective class management and the efficient computer laboratory management. The related researches provides teacher with the packet observation and the access limitation for only one host, but our suggested program provides teacher with those for all active hosts.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.12B
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• pp.1099-1111
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• 2006

As W-CDMA R5(HSDPA) service became recently commercialized in Korea, there is increasing demand for new pricing methodology for the 3G mobile interconnection at the earliest possible time. However, it is hard to find any case of systematic study on it. In this paper, a synthetic investigation into the three aspects of new mobile telecommunication service - technology, market, and regulation - is conducted to verify the validity of new interconnection pricing methodology. As the findings of this research indicate, given the several technological limits of 3G, speed of mobile telecommunication industry evolution, the problem of verifying ex-ante regulations' validity, and the cases of some foreign countries, it is suggested that immediate revision of interconnection policy is not sensible and that the revision is more desirable to be postponed until the maturity of the 3G market or the adoption of All-IP network.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.8
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• pp.1780-1784
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• 2015

Traditional IP routing, see only the Destination Address When Forwarding Layer 3 routing and exchange information and Destination-Based Routing Lookup is required for all Hop. Thus, all routers Full Internet routing information, the route information of more than about 120,000 may require. Therefore, the router configuration, which can be dispersed in the environment, the traffic load is required in accordance with this congestion. In this study, a unique characteristic of the Internet in the environment of an existing network Best Effect for QoS guarantee and hardware high speed switching of large multimedia data transmitted using a Labeling for forwarding a packet environment configuration is required. Video Stream Broadcast Transport Labeling rather than in much of the higher performance of the multi-step policy to most of the Video Stream Packet deulim was fixed to Labeling Header Format proposes a method of applying an effective QoS policy to a more simplified policy.

• Journal of Korean Society for Quality Management
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• v.35 no.3
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• pp.37-46
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• 2007

Our society is increasingly dependent on the Internet, and this trend will be accelerating in the ear of All-IP convergence. As the core infrastructure of the All-IP convergence, the next generation Internet should enable end-to-end Quality of Service(e2e QoS). For that purpose, major international standardization institutes such as ITU(International Telecommunication Union) and IETF(Internet Engineering Task Force) are building the concept and architecture of the Next Generation Network(NGN). However, these institutes focus only on technical issues, and leave many business/policy-oriented challenges unresolved. For example, the standards raised by ITU assume that the entire service delivery process for e2e QoS services is perfectly working. But required is a standardized business interface and process for seamless inter-operations across many stakeholders including ISPs(Internet Service Providers), CPs(Content Providers), so on. On the other hand, TMF(TeleManagement Forum) and QuEST(Quality Excellence for Suppliers of Telecommunication) Forum, global consortiums of telecom operators and vendors, present sets of interface rules and process which playa crucial role as de facto standards: for example, NGOSS/eTOM and TL9000 authorization system. However, these standards focus on telephone services, and provide little principle for the next generation Internet. Our study seeks a way to combine these two strains for a successful implementation of NGN. In particular, we find a missing link in the NGN architecture and the elements that could be complemented with the help of NGOSS/eTOM and TL9000. Finally, presented is a strategic direction that our standardization policy should purse in order to reinforce our global leadership for the next generation Internet.

• The KIPS Transactions:PartC
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• v.11C no.3
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• pp.361-370
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• 2004

All active nudes which have no physically direct connection with each other in If network must be able to compose and manage network topology Informations. Besides one active program can be performed by the active nodes when every active packet for this program is transmitted without any loss of packets. Also the active packets should be transmitted effectively to minimize the transmission delay and securely from threatens. In this thesis, the discovery scheme of active nodes is adapted for active nodes in IP networks to compose and manage the topology information. The scheme for the efficient, reliable and secure transmission of active packets is also proposed. The sequence number is assigned to every active packet. If a receiver detects the loss of active packet checking the sequence number, the receiver requests the retransmission of the lost packet to the previous active node. kiter receiving an active packet and adapting security and reliability schemes, intermediate active nodes not only copy and send the Packet Instantly but also apply some suity mechanisms to it. And the active packet transmission engine is proposed to provide these transmission schemes The simulation of the adapted active node discovery scheme and the proposed active packet transmission engine is performed. The simulation results show that the adapted active node discovery scheme is efficient and the proposed active engine has the low latency and the high performance.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2014.06a
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• pp.254-257
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• 2014

Recently, various applications using real-time face detection have been developed as face recognition technology and hardware grows. While network service is developing and video instruments costs lower, it is needed that smart surveillance camera and service using network camera based on IP and face detection technology. However, videos should be compressed for reducing network bandwidth and storage capacity in surveillance system. As it requires high-level improvement of system performance when all the compressed frames are processed in a face detection program, fast face detection method is needed. In this paper, not only a fast way of algorithm using Haar like features and adaboost learning and motion information but also an application on broadcast system is suggested.