• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.15 no.5
• /
• pp.147-154
• /
• 2015

Satellite broadcasting networks, like if you have if you have just received information that everyone must bring the required security attributes this earth should be done as encryption. In this paper, a satellite communication network AH additional security header in transport mode IPsec VPN by applying the SHA-256 and MD-5 authentication algorithm to authenticate the data portion Error rate and analyze the BER and Throughput. First, to generate a normal IP packet added to IPsec transport mode security header AH were constructed internal authentication data by applying the SHA-256 and MD-5 algorithm. Channel coder was applied to the Rate Compatible Punctured Turbo Codes, packet retransmission scheme Hybrid-ARQ Type-II and Type-III were used. Modulation method was applied to the BPSK, the wireless channel Markov channel (Rician 80%, Rayleigh 20% and Rician 90%, Rayleigh 10%) as an authentication algorithm according to the satellite channel state analyzed how they affect the error rate and Throughput.

• Journal of KIISE:Information Networking
• /
• v.36 no.1
• /
• pp.34-42
• /
• 2009

In this paper, we propose a practical hybrid transport error control scheme to provide SVC video streaming service over error-prone IP networks. Many error control mechanisms for various video coding standards have been proposed in the literature. However, there is little research result which can be practically applicable to the multilayered coding structure of SVC(the scalable extension of H.264/AVC). We present a new hybrid transport error control scheme that efficiently combines layered Forward Error Correction(FEC) and Automatic Repeat Request(ARQ) for better packet-loss resilience. In the proposed hybrid error control, we adopt ACK-based ARQ instead of NACK-based ARQ to maximize throughput which is the amount of effective data packets delivered over a physical link per time unit. In order to prove the effectiveness of the proposed hybrid error control scheme, we adopt NIST-Net network emulator which is a general-purpose tool for emulating performance dynamics in IP networks. It is shown by simulations over the NIST-Net that the proposed hybrid error control scheme shows improved packet-loss resilience even with much less number of overhead packets compared to various conventional error control schemes.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.13 no.7
• /
• pp.3369-3385
• /
• 2019

There are various research challenges in vehicular ad hoc networks (VANETs) that need to be focused until an extensive deployment of it becomes conceivable. Design and development of a scalable routing algorithm for VANETs is one of the critical issue due to frequent path disruptions caused by the vehicle's mobility. This study aims to provide a novel road-aware routing protocol for vehicular networks named as Two-level hierarchical Hybrid Road-Aware (THERA) routing for vehicular ad hoc networks. The proposed protocol is designed explicitly for inter-vehicle communication. In THERA, roads are distributed into non-overlapping road segments to reduce the routing overhead. Unlike other protocols, discovery process does not flood the network with packet broadcasts. Instead, THERA uses the concept of Gateway Vehicles (GV) for the discovery process. In addition, a route between source and destination is flexible to changing topology, as THERA only requires road segment ID and destination ID for the communication. Furthermore, Road-Aware routing reduces the traffic congestion, bypasses the single point of failure, and facilitates the network management. Finally yet importantly, this paper also proposes a probabilistical model to estimate a path duration for each road segment using the highway mobility model. The flexibility of the proposed protocol is evaluated by performing extensive simulations in NS3. We have used SUMO simulator to generate real time vehicular traffic on the roads of Gangnam, South Korea. Comparative analysis of the results confirm that routing overhead for maintaining the network topology is smaller than few previously proposed routing algorithms.

• The KIPS Transactions:PartC
• /
• v.15C no.5
• /
• pp.429-438
• /
• 2008

NGN is a packet-based converged network to support session and non-session services in QoS-enabled broadband transport network. QoS based resource control must be defined to support differentiated services for various network users in NGN. We designed and implemented DIAMETER protocol as the Rs interface, and also defined mapping rules between DIAMETER information elements and SDP(Session Description Protocol) attributes for QoS based resource control in NGN. We selected and implemented DIAMETER protocol among alternate resource control protocols in ITU-T as the Rw interface because of simple interworking method with Rs interface and adequate AAA functionality. We defined mapping rules of messages and information elements between Rs and Rw interfaces for resource control from a service layer to a transport layer. Based on the mapping rule of QoS information elements and the interworking method between Rs and Rw interfaces, we built up a test-bed that support differentiated delivery services.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2008.10a
• /
• pp.685-688
• /
• 2008

A Next Generation Network (NGN) is a packet-based network able to provide services including Telecommunication Services and able to make use of multiple broadband, QoS-enabled transport technologies and in which service-related functions are independent from underlying transport-related technologies. It offers unrestricted access by users to different service providers. It supports generalized mobility which will allow consistent and ubiquitous provision of services to users in this paper. we describe the trend of standardization activity related with open services environment capabilities which had been finally approved at September 2008 in ITU-T. ITU-T SG13 is responsible for studies relating to the architecture, evolution and convergence of next generation networks including frameworks and functional architectures, signalling requirements for NGN.

• Journal of the Korea Society of Computer and Information
• /
• v.12 no.1 s.45
• /
• pp.137-144
• /
• 2007

Advances in networking technology digital media, and codecs have made it possible for the Internet evolves into a Broadband convergence Network (BcN) and provides various services including Voice over Internet Protocol (VoIP) and IPTV over their high-speed IP networks. In order for the Internet to make a profit as traditional Public Switched Telephone Network (PSTN), it must provide high qualify VoIP services. Therefore, real time qualify measurement framework is the most important requisite to provide VoIP service. For this, IETF (Internet Engineering Task Force) defined RTCP-Extended Reports (RTCP-XR) that extend RTCP (Real-Time Transport Protocol Control Protocol). However, procedure and method tot actually VoIP qualify measurement did not recommended nothing but defined item to measure voice quality. Our objective in this paper is to describes a practical measuring framework for end-to-end QoS of switched voice packet in an IP environment. It includes concepts as well as step-by-step procedures for measuring packetized voice streams. It also proposes new formats that extend RTCP-XR's concept.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2006.05a
• /
• pp.978-980
• /
• 2006

A Next Generation Network (NGN) is a packet-based network able to provide services including Telecommunication Services and able to make use of multiple broadband, QoS-enabled transport technologies and in which service-related functions are independent from underlying transport-related technologies. It offers unrestricted access by users to different service providers. It supports generalized mobility which will allow consistent and ubiquitous provision of services to users In this paper, we explain the trend of standardization activity related with open services in ITU-T SG13(Study Group 13). ITU-T SG13 is responsible for studies .elating to the architecture, evolution and convergence of next generation networks including frameworks and functional architectures, signalling requirements for NGN.

• Journal of KIISE:Information Networking
• /
• v.34 no.3
• /
• pp.226-238
• /
• 2007

In this paper, we present a novel fully distributed topology control protocol that can construct the multiple-ring topology of Minimal Connected Dominating Set (MCDS) as the transport backbone for mobile ad hoc networks. It makes a topology from the minimal nodes that are chosen from all the nodes, and the constructed topology is comprised of the minimal physical links while preserving connectivity. This topology reduces the interference. The all nodes work as the nodes of the distributed parallel Boltzmann machine, of which the objective function is consisted of two Boltzmann factors: the link degree and the connection domination degree. To define these Boltzmann factors, we extend the Connected Dominating Set into a fuzzy set, and also define the fuzzy set of nodes by which the multiple-ring topology can be constructed. To construct the transport backbone of the mobile ad hoc network, the proposed protocol chooses the nodes that are the strong members of these two fuzzy sets as the clusterheads. We also ran simulations to provide the quantitative comparison against the related works in terms of the packet loss rate and the energy consumption rate. As a result, we show that the network that is constructed by the proposed protocol has far better than the other ones with respect to the packet loss rate and the energy consumption rate.

• The Journal of The Korea Institute of Intelligent Transport Systems
• /
• v.11 no.3
• /
• pp.48-54
• /
• 2012

The quality of real-time voice communication over Internet Protocol networks based on telematics is affected by network impairments such as delays, jitters, and packet loss. To resolve this issue, this paper proposes a receiver-based enhancing method of VoIP speech quality. The proposed method enables users to deliver high-quality voice using playout control and signal reconstruction, which consists of concealment of lost packets, adaptive playout-buffer scheduling using active jitter estimation, and smooth interpolation between two signals in a transition region. The proposed algorithm achieves higher Perceptual Evaluation of Speech Quality (PESQ) values and low buffering delay than the reference algorithm.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2003.10a
• /
• pp.916-919
• /
• 2003

Reliable multicast data transmission in a 1:N environment needs more sophisticated error control mechanism than that of in 1:1 environment due to ACK implosion and duplicated retransmission. Although there have been many related research on error control in reliable multicast, real implemented protocols are rare. As one of the reliable multicast transport protocols, ECTP is selected as an international standard reliable multicast protocol by ITU-T and ISO and implemented on RedHat 7.2 machine by us. In this paper, we evaluate the performance of the error control mechanism in the respect of throughput and generated control packet numbers with a real implementation code. From the results, it is concluded that the suitable values of error control parameters can be obtained from the local group size and network environments.