• Journal of Information Technology and Architecture
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• v.9 no.2
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• pp.177-186
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• 2012

Navy establishes the Naval Task Forces (TF) for many kinds of maritime operations. Then the TF in the maritime environment performs simultaneous component operations such as ASUW (Anti-Surface Warfare), ASW (Anti-Submarine Warfare), AAW (Anti-Aircraft Warfare), and assault operations. The TF consists of many tactical systems for the completion of missions C4I, VOIP (Voice Over Internet Protocol), DMHS (Digital Massage Handling System), and TDLs (Tactical Data Links) such as LINK-11, 16, ISDL (Inter Site Data Link). When the TF executes naval operations to complete a mission, we are interested in the kill chain for the maritime operations in the TF. The kill chain is a standard procedure for the naval operations to crush enemy defenses. Although each ship has a procedure about a manual for 'how to fight', it leave something to be desired for the TF detailed kill chain currently. Therefore, in this paper, we propose the naval TF's kill chain to perform the naval operations. Then, the operational effectiveness of the TF in the kill chain environment is determined through operation scenarios of TDL system implementation. It is to see the operational information sharing effect to a data link model based on MND-AF OV 6c (statement of tracking operational status) in the maritime operations applied to TDL and is to identify improvements in information dissemination process. We made the kill chain of maritime TF for the effective naval operations.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.6 no.7
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• pp.1854-1873
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• 2012

Owing to its simplicity and flexibility, the session initiation protocol (SIP) has been widely adopted as a major session-management protocol for Internet telephony or Voice-over IP (VoIP) services. However, SIP has faced various types of security threats. Call-disruption attacks are some of the most severe threats they face, and can greatly inconvenience consumers. In this paper, we analyze such SIP call-disruption attacks, and propose a method for detecting and counteracting them by extending the SIP INFO method with authentication. Using the proposed method, both the target user and the SIP server can detect the existence of a call-disruption attack on a user and counteract the attack. We demonstrate the effectiveness of the proposed method from the viewpoint of computational complexity by configuring a test-bed with an Asterisk SIP proxy server and an SIP performance (SIPp) emulator.

• Journal of the Korea Society of Computer and Information
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• v.10 no.5 s.37
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• pp.237-244
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• 2005

IP-Internet Telephony Service has not vet been achieved that of operating an IP-PBX service and a consumer Internet telephone services using VoIP technologies. In this paper, i suggest that the technologies of the VoIP Secure Gateway have connecting and securing for IP-Internet Telephony Service which makes If telephony protocols, firewall VPN tunneling, using Application Level Gateway, connection of the VoIP Secure Gateway. I suggest of telecommunication technologies that are enables an enterprise If-PBX service to interoperate with a consumer IP telephony service through a firewall. Also, I have proposed the solutions of security problems which was the security for VoIP Secure Gateway.

• 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.

• Journal of Digital Contents Society
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• v.9 no.1
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• pp.101-107
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• 2008

The level of transference of NGN is beginning as the operation of Access Gateway in korea at present, but NGN will keep developing continuously to the NGN integrated network until 2010. This research is finding the meaning and assignment of NGN QoS to deduct how to manage the control system and presenting the QoS control process and trial framework. The trial framework is the modeling of the QoS measurement metrics, the measurement time schedule, the section, hierarchy, instrument, equipment and method of measurement and the series of cycle & the methodology about analysis of the result of measurement. The objective standard of quality in communication service is guaranteed not by itself but by controlling and measuring continuously. Especially it's very important time to maintain the research about NGN QoS measurement and control because the big conversion of new network technology paradigm is now spreading.

• KSCI Review
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• v.14 no.2
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• pp.235-244
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• 2006

This paper attacked the unknown DoS which mixed a DoS attack, Worm and the Trojan horse which used IP Source Address Spoofing and Smurf through the SYN Flooding way that UDP, ICMP, Echo, TCP Syn packet operated. the applications that used TCP/UDP in VoIP service networks. Define necessity of a Dynamic Update Engine for a prevention, and measure Miss traffic at RT statistics of inbound and outbound parts in case of designs of an engine at IPS regarding an Self-learning module and a statistical attack spread. and design a logic engine module. Three engines judge attack grades (Attack Suspicious, Normal), and keep the most suitable filtering engine state through AND or OR algorithms at Footprint Lookup modules. A Real-Time Dynamic Engine and Filter updated protected VoIP service from DoS attacks, and strengthened Ubiquitous Security anger, and were turned out to be.

• The KIPS Transactions:PartC
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• v.9C no.3
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• pp.375-384
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• 2002

In the current Internet, the QoS of interactive applications is hardly guaranteed because of variable bandwidth, packet loss and delay. Moreover, VoIP which is becoming an important part of the information infra-structure in these days, is susceptible to network packet loss and end-to-end delay. Therefore, it needs error control mechanisms in network level or application level. The FEC-based error control mechanisms are used for interactive audio application such as VoIP. The FEC sends a main information along with redundant information to recover the lost packets and adjusts redundant information depending on network conditions to reduce the bandwidth overhead. However, because most of the error control mechanisms do not consider end-to-end delay but packet loss rate, their performances are poor. In this paper, we propose a new error control algorithm, SCCRP, considering packet loss rate as well as end-to-end delay. Through experiments, we confirm that the SCCRP has a lower packet loss rate and a lower end-to-end delay after reconstruction.

• The Transactions of the Korea Information Processing Society
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• v.7 no.8
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• pp.2463-2472
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• 2000

In this paper, we consider pits and falls of V oIP service scheme over the air link environment. It results that V oIP over packet switching is a more attractive approach in several points. We point out the ffilIior requirements for successful V oIP service over the air. Also, we propose V oIP CP concept for efficient wireless channel utilization. Additionally, we analyze and evaluate the performance. According to the results, It shows that the long cycle VolP vocoder CODEC such as ITU-T G.723 is better than short cycle V oIP vocoder CODEC. In this case, the increase of the simultaneous user of system is almost 60% larger than conventional circuit switching.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.129-132
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• 2003

In this paper, which sees the Strong-ARM SA1110 it used the main CPU and RTP in VoIP system. It will be able to apply the information communication field it embodied. It used the Tynux_box2 with the hardware side and it composed a VOIP system. And it used the RTP which is a real-time protocol in software control portion. The development environment of the paper that used the Target board and a Linux PC for connection used the RS-232C, USB connection, Ethernet LAN. The VoIP the environment for a communication used the wave file in the substitution which changes analog signal with the digital signal. And For the communication of the both sides it used the socket. This paper explained the fact that against a general technique from the operation of VoIP system. Using the Embedded linux development board which explained an operational process of the RTP protocol.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.10B
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• pp.928-938
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• 2008

Though services requiring Quality-of-Service (QoS) guarantees such as Voice over Internet Protocol (VoIP) have been widely deployed on the internet, most of internet backbone networks, unfortunately, do not distinguish them from the best-effort services. Thus estimating the effective capacity meaning the traffic volume that the backbone networks maximally accommodate with keeping QoS guarantees for the services is very important for Internet Service Providers. This paper proposes a test-bed based on ns-2 to evaluate the effective capacity of backbone networks and then estimates the effective capacity of an experimental backbone network using the test-bed in terms of the service standard of the VoIP service. The result showed that the effective capacity of the network is estimated as between 12% and 55% of its physical capacity, which is depending on the maximum delay guarantee probability, and strongly affected by not only the type of offered workload but also the quality standard. Especially, it demonstrated that in order to improve the effective capacity the maximum end-to-end delay requirement of the VoIP service needs to be loosened in terms of the probability to guarantee.