• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.3
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• pp.654-660
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• 2004

In this paper, we define that pervasive home network, which provides necessary services for user properties and removes distractions to improve the quality of human life. So, user can enjoy home network technology including devices and softwares at any place with no knowledge of networked home, devices, and softwares. In this home network, a mobile agent, called LAFA, can migrate to unfamiliar home network and control the necessary devices. For this environment, we design security management module for authenticating user and home server that access some other home networks, and for protecting text, multimedia data, and mobile agent that are transferred between home networks. The security management module is composed of a key exchange management module and an access control management module, for key exchange management module, we propose a key exchange protocol, which provides multimode of authentication mode and key exchange mode. One of these two modes is selected according to the data type.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2015.10a
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• pp.791-793
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• 2015

Recently, UCC (User Created Contents) and VoD (Video on Demand), and multimedia content are growing, IP-TV, Smart TV, OHTV (Open Hybrid TV) various services such as multi platform (Multi-platform) environment, services and QoS issues. To solve this problem, the network efficiently, and improve the quality of content is necessary for the system. In this paper, the network of channels State and transmission of multimedia data based on dynamic resource usage, TCP and UDP, Adaptive dual-streaming system used for design and analysis. In addition, the existing TCP and UDP streaming system using a single protocol for analysis and verification of the effectiveness of the difference between and. This is a disaster, and medical/first aid system will be utilized in the field of feed, are ubiquitous.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39C no.11
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• pp.1167-1176
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• 2014

In this paper, a relay communication scheme for enhancing the WiMedia network performance by device's mobility is proposed. WiMedia protocol is suitable for the application that supports the real-time multimedia service in Ship Area Network since it supports high speed data transfer. However, the device's mobility is caused the dramatic change of link state and network topology, and is occurred the degradation of network performance. Therefore, a relay communication scheme for WiMedia network is proposed in this paper. The proposed technique can intelligently treat the change of link state, and solve the degradation of network performance.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.12
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• pp.2301-2306
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• 2008

SIP(Session Initiation Protocol) has become an universal standard for multimedia communications for both wired and wireless networks since it has been adopted as a standard protocol for IMS platform in 3GPP standardization organization at November 2000. In this paper, we design and implement a SIP-based softphone client program which provides telephony service between internet users and a call center equipped with VoIP gateway. A softphone client based on PC-to-phone connection should guarantee to provide interoperability with various VoIP gateways and higher portability to be able to operate on different PC environments. The softphone client program in this paper has been developed with SIP 2.0 standard protocol to support interoperability and with JAIN SIP and JMF package to achieve higher portability.

• Journal of Korea Multimedia Society
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• v.15 no.5
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• pp.587-594
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• 2012

IEEE 802.15.6 standard technology is proposed for low-power wireless communication in, on and around body, where vital signs such as pulse, blood pressure, ECG, and EEG signals are transmitted as a type of data packet. Especially, these vital signs should be delivered in real time, so that the latency from slave node to hub node can be one of the pivotal performance requirements. However, in the case of IEEE 802.15.6 technology data retransmission caused by transmission failure can be done in the next superframe. In order to overcome this limitation, we propose an adaptive polling algorithm for IEEE 802.15.6 technology. The proposing algorithm makes the hub to look for an appropriate time period in order to make data retransmission within the superframe. Through the performance evaluation, the proposing algorithm achieves a 61% and a 73% latency reduction compared to those of IEEE 802.15.6 technology in the environment of 70% traffic offered load with 10ms and 100ms superframe period. In addition, the proposing algorithm prevents bursty traffic transmission condition caused by mixing retransmission traffic with the traffic reserved for transmission. Through the proposing adaptive polling algorithm, it will be possible to transmit time-sensitive vital signs without severe traffic delay.

• The Transactions of the Korea Information Processing Society
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• v.5 no.12
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• pp.3047-3062
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• 1998

This paper deseribes the desihn and the implementation of the Realtime Transport Protocol(RTP)/ Rdaltime Control Protocol(RTCP) (RFC 1889,1890) that is used to transmit the audio/video data to any destination and to feedback the Quality of Service (QoS) information of the received media data to the sender, in the teleconferencing systems proposed by ITU-T. These protocols are implemented with multi thead technique and run on top of UDP/IP-Multicast through the socket interface as the underlying protocol. The upper layer is impelmented such that in can be accessed by the H245 comference control protocol. The RTP packetizes the digitized audio/video data from the encoder info a fixed format, and multieast to the participants. The RTCP monitors RTP packets and extracts the QoS values from it such as round-trip delay, jiter and packet loss to form RTCP packets and non periokically sends them to the sender site. In this Paper, we also descritx the study of measurement and analysis for QoS factors that observed on performing teleconferencing system over Internet. The results from this experiment is indicate that RTT and Jitter value are acceptable even entwork load is high. However, it appears that packet loss rate is high in daytime and most losses periods have length one or two.

• Journal of KIISE:Computing Practices and Letters
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• v.13 no.3
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• pp.153-164
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• 2007

The Access Grid (AG) provides collaboration environments over the IP multicast networks by enabling efficient exchange of multimedia contents among remote users; however, since lots of current networks are still multicast-disabled, it is not easy to deploy this multicast-based multi-party AG. For this problem, the AG provides multicast bridges as a solution by putting a relay server into the multicast networks. Multicast-disabled clients make UDP connections with this relay server and receive forwarded multicast traffics in unicast UDP packets. This solution is facing several limitations since it requires duplicate forwarding of the same packet for each unicast peer. Thus, in this paper, we propose an alternate solution for the multicast connectivity problem of the AG based on the UMTP (UDP multicast tunneling protocol). By taking advantage of flexibilities of UMTP, the proposed solution is designed to improve the efficiency of network and system utilization, to allow reuse of multicast-based AG applications without modification, and to partially address the NAT/firewall traversal issues. To verify the feasibility of proposed solution, we have implemented a prototype AG connectivity tool based on the UMTP, named as the AG Connector.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.12B
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• pp.1259-1266
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• 2010

M2M communication is considered as a key enabling technology to monitor the status of objects, vehicles, humans through auto-configuring wireless networks. In M2M network, there are active research activities to enhance the reliability on data while being collected from wireless sensor network. The reliability issue becomes more important as wireless sensor networks carry multimedia data, which is delay sensitive. The interference caused by the adjacent neighbor sensor nodes is a major factor in network performance degradation, which becomes more severe in multi-hop routing environment. In this paper, we propose inerfernce-free multipath routing protocol for M2M wireless network for enhancement of packet delay performance. The simulation results show that the proposed routing algorithm outperforms the existing routing protocols in terms of packet delay and throughput.

• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.738-749
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• 2011

In this paper, we study QoS(Quality of Service) control protocols and the effect using MAC/PHY parameters of client device in mobile network. We proposes the way of controling the bit-rate by estimating the channel condition of the client with measured MAC/PHY parameters which is sent from the client. With the proposed method, more accurate available bit-rate can be estimated compared to conventional protocol, RTCP(Real-time Transport Control Protocol). The accurate bit-rate estimation can decrease wasted bit-rate and transport delay. In the result of the advantages, the transported video quality can be enhanced. In this paper, we show the effects of enhancement using client's the field data measured in WiMAX.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.4C
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• pp.273-281
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• 2002

Recently, demand for real-time data and multimedia data is rapidly increasing, and it is impossible for mobile Ad Hoc networks with only best effort service to efficiently transfer such data. So, we absolutely need the QoS service which reserves the communication resources in advance. The existing routing protocols, which assume that the links between nodes are bidirectional, provide the convenience for the route discovery and maintenance, but can not support the routing function for the unidirectional links due to the wireless link property easily changing with time under the real wireless environment. In order to solve such problems, in this dissertation we suggested a unidirectional QoS routing a waste of communication resources. The waste of protocol that supports unidirectional links and reduces communication resources is reduced by establishing the shortest route suitable to QoS support, considering in advance the usable communication resources at each node.