• The Journal of the Korea institute of electronic communication sciences
• /
• v.7 no.4
• /
• pp.747-752
• /
• 2012

In this paper we evaluated by simulation the effect of VoIP roaming and data traffic roaming on other VoIP calls. Regardless of MIPv4 or MIPv6, the quality of voice of all VoIP calls falls down quickly to the bottom level after the start of roaming by one voice terminal. That was caused by the excessive retransmission on downlink. Thus it seems that we need a kind of call admission control when we adopt the roaming service on VoIP calls. Data traffic degrades also the voice quality especially at the foreign agent side.

• Journal of the Korea Society of Computer and Information
• /
• v.11 no.4 s.42
• /
• pp.155-164
• /
• 2006

VoIP technology is Eight New Services among Ubiquitous-IT839 strategies. This paper tested wiretapping or VoIP service in connected a soft phone and LAN and WAN sections, Internet telephones and a device. IP PBX, a banner operator network to have been connected to VoIP Internet network. As a result of having experimented on wiretapping of VoIP networks, Vulnerability was found. and a wiretapping by attacks of a hacker was succeeded in a terminal and proxy and attachment points of a VoIP network like a hub to follow a CVE list. Currently applied a security plan of an each wiretapping section in viewpoints of 6 security function of Access Control. Confidentiality, Authentication. Availability, Integrity. Non-repudiation in VoIP networks named to 070. Prevented wiretapping of contents by the results, the AES encryption that executed wiretapping experiment about a packet after application of a security plan. Prevented wiretapping, and kept security and audit log. and were able to accomplish VoIP information protection to network monitoring and audit log by an access interception and qualification and message hash functions and use of an incoming refusal.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.15 no.12
• /
• pp.2555-2562
• /
• 2011

Binary CDMA is a new standard technology for wireless communication developed by our country that makes high speed communications and good quality of services. In this paper we researched the development methods and implementation of a wireless VoIP System based on the Binary-CDMA technology that makes it freely installed in any place without phone cables and laying works. We designed the structure of the Binary-CDMA Wireless VoIP System and defined messages and data structures for call control. And then we designed and implemented call control message flow between VoIP terminals for user's communication.

• Journal of Korea Society of Digital Industry and Information Management
• /
• v.7 no.1
• /
• pp.31-39
• /
• 2011

VoIP service uses packet network of ip-based because that has eavesdropping, interception, illegal user as vulnerable elements. In addition, PSTN of existing telephone network is subordinate line but VoIP service using the ip packet provide mobility. so The user authentication and VoIP user's account service using VoIP has emerged as a problem. To solve the vulnerability of SIP, when you use VoIP services with SIP, this paper has made it possible to authenticate user's terminal by using proxy server and proxy server by using authentication server. In conclusion, sender and receiver are mutually authenticated. In the mutual authentication process, the new session key is distributed after exchanging for the key between sender and receiver. It is proposed to minimize of service delay while the additional authentication. The new session key is able to authenticate about abnormal messages on the phone. This paper has made it possible to solve the vulnerability of existing SIP authentication by using mutual authentication between user and proxy server and suggest efficient VoIP service which simplify authentication procedures through key distribution after authentication.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.29 no.7C
• /
• pp.1007-1014
• /
• 2004

In this paper, H.235 based security mechanism for H.323 multimedia applications was implemented in embedded environment. H.235 covers authentication using HMAC-SHAI -96, authenticated Diffie-Hellman key exchange, security capability exchange, session key management for voice encryption, and encryption functions such as DES, 3DES, RC2. H.235-based mechanisms were also analyzed in terms of its security and possible attacks.

• Proceedings of the Korean Information Science Society Conference
• /
• 2011.06d
• /
• pp.215-218
• /
• 2011

VoIP(Voice over Internet Protocol) 서비스는 기존의 음성전화 서비스(Public Switched Telephone Network, PSTN)와 달리 IP 프로토콜을 이용한 저렴한 통신비용 등의 장점이 있는 음성통신 기술로써, 기존의 아날로그 음성전화 서비스를 대신하는 서비스이며, 새로운 인터넷 융합서비스로 많은 사용자가 이용하고 있다. 하지만 VoIP 서비스가 인터넷망을 이용함으로 IP Spoofing, DoS (Denial of Server) / DDoS(Distributed Denial of Service), 등의 여러 가지 보안의 문제점을 가지고 있다. VoIP 서비스에서 DDoS 공격은 Proxy 서버 등에 대량의 공격 메시지를 보냄으로써 서버의 자원을 고갈시켜 정상적인 서비스를 하지 못하게 한다. DoS, DDoS 공격 중 Invite Flooding 공격은 1분에 수천 개의 Invite 메시지를 보내 회선의 자원을 고갈시키는 공격이다. 특히 IP/Port 위조하여 공격 경우 공격 패킷 탐지하기 어려우므로 차단할 수 없다. 따라서 본 논문에서는 VoIP의 DoS/DDoS 중 하나인 Invite Flooding 공격 시 SIP Proxy Server에서 메시지 분산시키는 방법과 MAC Address와 사용자 번호 등 IP 이외의 고정적인 사용자 정보를 확인하여 공격을 탐지하고, 공격 Agent에 감염된 Phone을 공격차단서비스로 보내 복구시키는 방법을 제안한다.

• 한국IT서비스학회:학술대회논문집
• /
• 2010.05a
• /
• pp.147-150
• /
• 2010

본 논문은 PSTN과 인터넷전화기능을 모두 갖춘 단말(일명 듀얼폰)에서 데이터서비스를 제공하는 방안을 소개한다. 듀얼폰은 음성단말로써 PSTN 또는 VoIP 통화를 모두 수용하고, 발신시에는 발신모드를 선택하여 제공함으로써 사용자가 저렴한 통신요금을 활용할 수 있는 장점이 있다. 그런데 본 단말은 인터넷브라우저를 탑재하지 않으므로 인터넷이 연결되어 있음에도 데이터서비스를 위한 웹 서버와 다이렉트 메시지교환이 용이하지 않다. 본 논문은 웹이용이 용이하지 않은 단말에서 데이터 서비스를 제공하는 방안을 소개한다.

• The Journal of The Korea Institute of Intelligent Transport Systems
• /
• v.10 no.4
• /
• pp.71-77
• /
• 2011

In this paper, we propose a conversational quality measurement (CQM) system for providing the objective QoS of high quality mobile VoIP voice telecommunication. For measuring the conversational quality, the VoIP telecommunication system is implemented in two smart phones connected with VoIP. The VoIP telecommunication system consists of echo cancellation, noise reduction, speech encoding/decoding, packet generation with RTP (Real-Time Protocol), jitter buffer control and POS (Play-out Schedule) with LC (loss Concealment). The CQM system is connected to a microphone and a speaker of each smart phone. The voice signal of each speaker is recorded and used to measure CE (Conversational Efficiency), CS (Conversational Symmetry), PESQ (Perceptual Evaluation of Speech Quality) and CE-CS-PESQ correlation. We prove the CQM system by measuring CE, CS and PESQ under various SNR, delay and loss due to IP network environment.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.15 no.6
• /
• pp.89-94
• /
• 2015

FMS which has a concept that wireless terminal can replace wire terminal services is a technologies that is can provide service costs same as wire terminal in the special zone. Enterprise type of FMC that is developed making up for the weak point is must have to improve voice quality and FMC performance in the soft phone. This paper measure voice quality based on the one way of the total estimated delay time of FMC to carry out IMS services between IP-PBX and FMC soft-phone to operate it's controller optimally and put forward evidence to be in 120ms and 150ms in the VoIP FMC voice quality. To measure FMC performances in four categories evaluated trials and prove its performances.

• Annual Conference on Human and Language Technology
• /
• 2010.10a
• /
• pp.141-142
• /
• 2010

최근 컴퓨터와 통신 기술의 발달로 인해 세계 구석구석을 연결하는 인터넷(Internet)이 대중화되었고, 실시간 멀티미디어를 이용한 다양한 응용 서비스들이 출현하고 있다. 응용 서비스들 중의 하나인 VoIP(Voice over Internet Protocol)기반의 인터넷 전화 서비스는, 일반 사용자에게 인터넷 전화에 대한 인식을 확산시켰으며 기존의 음성 통신 시장에 새로운 변화를 가져왔다. 본 연구에서는 음성 데이터를 인터넷 프로토콜 데이터 패킷으로 변화하여 인터넷망에서 통화를 하는 통신 서비스 기술인 VoIP방식의 인터넷폰의 시험환경을 구축하여 향후 실질적인 활용을 통해 고품질 S/W의 개발을 촉진함으로써 높은 부가가치를 창출하고 경쟁력을 갖춘 제품의 개발 지원이 가능하다고 본다.