• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• v.2
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• pp.317-320
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• 2006

Ships and their cargos have been managed safely by positioning report system. However, little attention has been paid to safety of crew's works with danger. The attempt that used PHS inboard was before by the present authors. However, the functions were just voice call and mail exchange. The data acquisition from the terminal by proper control was not possible. Thus the position of the terminal was not available. As for the cell phone of next generation, GPS receiver and wireless LAN are installed by manufacturers. Therefore, we propose a system which uses a cell-phone with GPS receiver on a ship in order to promote the safety of ship's crew. We checked the availability of cell-phone GPS receiver at thirty different points inboard. The positioning was not possible in the areas further than 4m from the window. Then, we proposed the system which follows the positions of the crews and confirms their safety inboard by using the VoIP (Voice over Internet Protocol) function by wireless LAN.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.9 no.4
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• pp.313-322
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• 2016

This paper presents a method to improve the performance of Scalable Multicast Protocol deployed in wired and wireless network by adding retransmission function on base stations. When using Scalable Multicast Protocol over wireless and wired networks, packet drops on the wireless link produce the initiation of retransmission request packets and the implosion of retransmission packets, which deteriorate the multicast session performance. The efficient reliable multicast mechanism in wireless networks utilizing the retransmission function on the base station is addressed in this paper. We explain the design of a retransmission function which improves the performance of Scalable Multicast Protocol sessions in wireless and wirednetwork. The main idea is to cache Scalable Multicast Protocol packets at the base station and perform local retransmissions across the wireless link. ARENA has been used to simulate and to get performance for reducing signaling overhead and processing delay through the comparison of the proposed function to the Scalable Multicast Protocol.

• Journal of KIISE:Information Networking
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• v.32 no.1
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• pp.51-59
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• 2005

In this paper, we propose and implement a hybrid-type video conference system using SIP protocol proposed in IEFT. This conference system, all signaling is controled by the server, while the media communication can be managed either by UAs or by the server. as a result, the complexity of its routing and user state management is reduced and it is applicable to the mobile environment of the ad-hoc models. The implemented server is scale enough to be used in a large conference because it only takes charge of signaling part. And also we see that UAs are only suitable in the small conference where less than ten people can join.

• Journal of Internet Computing and Services
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• v.3 no.3
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• pp.95-104
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• 2002

The interworking for voice service between next generation mobile network and traditional network can be deployed better flexible and expansible network conditions with providing efficiency and economy of network at the same time. So it is essential to develop the interworking strategies together with a evolved network and traditional network. This paper describes a design and verification of internetworking protocol for CC of next generation mobile network and SIP of IP network for applying interworking technology to next generation mobile network, which can harmoniously expropraise voice service from traditional network.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.1
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• pp.32-39
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• 2015

The voice communications with wireless backbone based networks are evolving into a packet switching VoIP systems. In those networks, a call processing scheme is required for management of subscribers and connection between them. A VoIP service scheme for those systems requires reliable subscriber management and connection establishment schemes, but the conventional call processing schemes based on the centralized server has lack of reliability. Thus, the mutual-backup architecture of SIP-servers is required to ensure efficient subscriber management and reliable VoIP call processing capability, and the synchronization and call processing schemes should be changed as the architecture is changed. In this paper, a mutual-backup architecture of SIP-servers is proposed for wireless backbone based networks. A message format for synchronization and information exchange between SIP servers is also proposed in the paper. This paper also proposes a FSM scheme for the fast call processing in unreliable networks to detect multiple servers at a time. The performance analysis results show that the mutual backup server architecture increases the call processing success rates than conventional centralized server architecture. Also, the FSM scheme provides the smaller call processing times than conventional SIP, and the time is not increased although the number of SIP servers in the networks is increased.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.2
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• pp.331-337
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• 2004

Multiparty conference service based on SIP supported by VoIP network is gradually increased in use and the continuous development and standardization works on SIP are in the process of advancing. But SIP used in currently does not support identity discovery and distribution of each participant for multipath conference. In this paper, a SIP extension method for guaranteeing security from the multiparty conference based on SIP is proposed. We design a new SIP header and method for discovering and distributing a participant's identity in closed multiparty conference when the call initiation is established. And it can ensure that each participant is notified before a new participant joins.

• Journal of Internet Computing and Services
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• v.6 no.3
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• pp.85-95
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• 2005

Whereas the traditional multicast schemes based on Mobile IP can support a limited number of very large multicast groups, the Xcast protocol can support a very large number of small multicast groups, In the Xcast, the source node encodes the list of destinations in the Xcast header. Therefore, the maximum packet size in the network limits the number of destinations that a Xcast packet may have. In this paper, an enhanced Xcast protocol is proposed to solve the multicast group size limitation of the existing Xcast protocol in providing multicast service for mobile nodes. Moreover, the SIP (a very flexible control plane protocol) is integrated with the proposed multicast scheme to provide mobility awareness on the application layer. The simulation results verify that the proposed multicast scheme not only increases the packet delivery ratio and the data packet forwarding efficiency but also achieves low latency of packets in the network.

• Journal of KIISE:Information Networking
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• v.30 no.5
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• pp.621-630
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• 2003

In this paper, we implemented SIP protocol that supports IPv6 and differentiated call processing scheme for NGN(Next Regeneration Network). In NGN, SIP processes call signaling among various application services. A softswitch and SIP server must give priority to sensitive services such as Fax, network management and home networking that require a fast call setup time. Also, the support of IPv6 is needed under consideration of All-IP. We proposed differentiated call processing scheme. The differentiated call processing scheme supports differentiated call processing as priority of service class on call processing in SW server We defined three service classes and use the Flow Label field of the IPv6 header for setting service class. Through the performance analysis, we proved that it improves throughput for call message with the high priority. The result of performance analysis demonstrates that differentiated call processing scheme gives better performance for the service requiring a fast session establishment in NGN.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.12 no.3
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• pp.224-230
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• 2019

In this paper, a new multiple servers conference service system using media control channel/distributed conference manipulation architecture has been presented. This conference service system can handle loads effectively from many conference participants. In this suggested architecture, media control channels are established between conference server, and distributed conference manipulation messages for distributing system loads are exchanged through this channels. These messages are transported between servers using media control channel created after stable transport sessions, and can be used to process server loads according to participants effectively. So this method can be used to implement large scale conference service system. For these purposes, formats of distributed conference manipulation messages which transmitted through media control channels are designed. and messages exchange procedures between conference servers are also presented. The performance of the proposed conference service system has been analysed by experiments, and the results show that the performances are improved according to participants.

• Proceedings of the Korean Information Science Society Conference
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• 2005.11a
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• pp.622-624
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• 2005

본 논문은 manet 환경에서 SIP(Session Initiation Protocol)를 사용하여 효과적인 인증 기법을 제안한다. 각 노드는 플루딩을 통하여 자신의 범위에서 타 노드와 통신을 할 수 있다. 반경(Radius)내의 네트워크에서 한개의 노드는 클러스터 헤더로 지정되며. 지정된 노드는 자신이 포함하고 있는 반경내의 모든 노드로부터 비콘(Beacon)신호를 통하여 노드들의 정보를 가지고 있다. 또한 타 네트워크와의 통신을 위하여 자신의 정보를 갱신한다. 만약 타 네트워크에서 정보를 요청하면 클러스터 헤드는 SIP를 통하여 상대 클러스터 헤더를 인증하고, SIP 세션 과정을 수행한다. SIP 과정을 수행한 클러스터 헤드는 자신의 반경내의 모든 노드들의 인증을 수행한다.