• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.990-994
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• 2003

This paper describes the VoIP(Voice over Internet Protocol) SoC(System on Chip) Design and the test board development for the function verification to support voice communication services using Internet. To implement the simple system of configuration, we designed the VoIP SoC which have ARM922T of 32bit microprocessor, IP network interface, voice signal interface, various user interface function. Also we developed test program and communication protocol to verify the function of this chip. We used several tools of design and simulation, developed and tested a test board with Excalibur which includes ARM922T microprocessor and FPGA.

• International Journal of Computer Science & Network Security
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• v.22 no.2
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• pp.175-184
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• 2022

Voice over Internet Protocol (VoIP) has grown in popularity as a low-cost, flexible alternative to the classic public switched telephone network (PSTN) that offers advanced digital features. However, additional security vulnerabilities are introduced by the VoIP system's flexibility and the convergence of voice and data networks. These additional challenges add to the normal security challenges that a VoIP system's underlying IP data network infrastructure confront. As a result, the VoIP network adds to the complexity of the security assurance task faced by businesses that use this technology. It's time to start documenting the many security risks that a VoIP infrastructure can face, as well as analyzing the difficulties and solutions that could help guide future efforts in research & development. We discuss and investigate the challenges and requirements of VoIP security in this research. Following a thorough examination of security challenges, we concentrate on VoIP system threats, which are critical for present and future VoIP deployments. Then, towards the end of this paper, some future study directions are suggested. This article intends to guide future scholars and provide them with useful guidance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.10B
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• pp.909-918
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• 2006

Recently, the qualities of the call and QoS of data service in mobile wireless IP networks inclusive of internet wired networks are very challenging issues. IntServ, DiffServ and MPLS model for supporting QoS in wired networks are representative models which are investigated generally up to now. While, MIPv4(Mobile IPv4) and MIPv6 are proposed to support mobility management for mobile wireless internet networks. But, MIPv4 and MIPv6 based UDP and TCP/IP protocol have many issues to be solved like efficient QoS and frequent handover management. In this paper, we propose a efficient management platform to support seamless data services and end-to-end QoS management in IP-based wireless mobility environments using transport layer protocol, SCTP which support multi-homing and multi-streaming. We used SCTP protocol and IntServ technology for end-to-end communications to support multi-homing and seamless QoS management in the IP-based mobile wireless networks.

• Journal of the Korea Computer Industry Society
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• v.5 no.9
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• pp.1017-1024
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• 2004

In this paper, we propose the progressed interworking method between H.323 and SlP, then explain the improved property. The VolP(Voice over Internet Protocol) technology which is able to use a voice service through internet is more cheaper then existing telephone charges, and is easil)· accept the various of multimedia services from internet. Previous connectionmethod of VoIP used H.323 protocol, but it is very complex to connection establishment. so, the SIP(Session Initiation Protocol) protocol that propose in SIP-Working Group is in use recently. Therefore, we need new interworking methodology between H.323 and SIP Products. In this thesis, the progress interworking method between H.323 and SIP are Propose, then interpret unnecessary packet delay for call setup and improved feature of message exchange.

• TTA Journal
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• s.71
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• pp.93-104
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• 2000

• The Journal of Engineering Research
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• v.5 no.1
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• pp.45-55
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• 2004

NAT (Network Address Translation) is an IP address modification protocol that translates private IP address into authentic Internet address. The main features of NAT are to improve network security and to save IP address. Generally speaking, in order to perform its functionality, NAT uses the address information in the packet header. Certain application protocols, however, use the information in the packet data as well as the information in the packet header to perform end-to-end communication. Therefore, to support these types of application protocols, NAT should be able to perform appropriate translation of protocol information in the packet data. In this thesis, we design and implement a method which translates virtual IP information in the packet date into real IP information by using port proxy server.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.23 no.3
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• pp.59-64
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• 2023

Fire and disaster broadcasting systems are divided into analog, digital, and network-based digital public address systems, and important specifications in network-based digital public address systems are low-latency audio, high sampling rate, and multi-channel input and output. In the past, it has been widely used to the AoE method for distinguishing based on the MAC address of the data link layer. However, this method has a problem of increasing complexity and cost. This proposal is an AoIP/UDP method, which allows communication to be easily distinguished by IP address without the need for a separate redundant network, so that the network can be freely used and configured, and cost can be reduced by reducing complexity. After implementing the AoIP/UDP method, the experimental results showed that the cost was improved with the equivalent performance with 2.66ms latency.

• Journal of Communications and Networks
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• v.14 no.5
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• pp.510-517
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• 2012

This paper analyzes the performance of voice-over-Internet protocol (VoIP) services in terms of the system throughput, the packet delay, and the signaling overhead in a mobile WiMAX system with a hybrid automatic repeat request (HARQ) mechanism. Furthermore, a queueing analytical model is developed with due consideration of adaptive modulation and coding, the signaling overhead, and the retransmissions of erroneous packets. The arrival process is modeled as the sum of the arrival rate at the initial transmission queue and the retransmission queue, respectively. The service rate is calculated by taking the HARQ retransmissions into consideration. This paper also evaluates the performance of VoIP services in a mobile WiMAX system with and without persistent allocation; persistent allocation is a technique used to reduce the signaling overhead for connections with a periodic traffic pattern and a relatively fixed payload. As shown in the simulation results, the HARQ mechanism increases the system throughput as well as the signaling overhead and the packet delay.

• Journal of Internet of Things and Convergence
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• v.2 no.4
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• pp.41-47
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• 2016

The IP address used in the Internet has the role of both identifier and locator to bind a host and the application, however, this binding restricts some functions such as mobility and multi-homing. As a result, we suggested a host ID-LOC separation protocol using DHT with SDN features. The proposed scheme is a network-based scheme, and uses IPv6 addresses. The underlying network is partitioned into Host Identity domain and IP domain for identifiers and locators. In this paper, we present a simple cost model for analyzing both the proposed scheme and one of the previous works, the MOFI. The result of cost analysis shows better performance of the proposed scheme.

• Journal of Communications and Networks
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• v.18 no.6
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• pp.948-961
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• 2016

Internet protocol (IP) spoofing is a serious problem on the Internet. It is an attractive technique for adversaries who wish to amplify their network attacks and retain anonymity. Many approaches have been proposed to prevent IP spoofing attacks; however, they do not address a significant deployment issue, i.e., filtering inefficiency caused by a lack of deployment incentives for adopters. To defeat attacks effectively, one mechanism must be widely deployed on the network; however, the majority of the anti-spoofing mechanisms are unsuitable to solve the deployment issue by themselves. Each mechanism can work separately; however, their defensive power is considerably weak when insufficiently deployed. If we coordinate partially deployed mechanisms such that they work together, they demonstrate considerably superior performance by creating a synergy effect that overcomes their limited deployment. Therefore, we propose a universal anti-spoofing (UAS) mechanism that incorporates existing mechanisms to thwart IP spoofing attacks. In the proposed mechanism, intermediate routers utilize any existing anti-spoofing mechanism that can ascertain if a packet is spoofed and records this decision in the packet header. The edge routers of a victim network can estimate the forgery of a packet based on this information sent by the upstream routers. The results of experiments conducted with real Internet topologies indicate that UAS reduces false alarms up to 84.5% compared to the case where each mechanism operates individually.