• The Journal of the Korea Contents Association
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• v.5 no.4
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• pp.227-236
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• 2005

We study, in terms of VoIP user assessment of voice quality, the synchronization of measurement system is important. Commonly the synchronization system uses NTP(Network Time Protocol) or GPS(Global Positioning System), these synchronization method has time error of distance, system overhead of data processing, and system specialized clock error. we propose and implement the synchronization method to correct time error between two measurement system in the internet. So the time synchronization of systems can get time error, then user assessment of voice quality become reliable.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.4 s.346
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• pp.1-12
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• 2006

In this paper we propose an efficient mobile multimedia communication protocol, mobile terminal software platform and mobile VoIP application for IP-based convergence networks. The Proposed mobile multimedia communication protocol is called as ST-MRSVP (Split tunnel based Mobile Resource reServation Protocol) which integrates split tunnel based Mobile IP and RSVP in order to support hish speed mobility. Since mobile terminal platform supports QoS (Qualify of Service) with keeping seamless mobility, mobile QoS supporting modules are developed and interworked together by means of shared memory mechanism. Testbed is composed of a core-network embedding the proposed protocols and wireless LAN-based access networks. We verify functionality and performance of the proposed techniques by using various mobility test over the testbed. As a result, the proposed architecture can reduce the handover delay time with QoS support under 30% comparing with the standard mechanisms and support voice quality as good as CDMA phone.

• Annual Conference of KIPS
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• 2010.04a
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• pp.729-732
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• 2010

통신망의 발달로 다양한 인터넷 기반 기술들이 등장함에 따라 현재는 데이터뿐만 아닌 음성에 대한 부분도 IP 네트워크를 통해 전송하려는 움직임이 발판이 되어 VoIP(Voice Over Internet Protocol)라는 기술이 등장하였다. SIP(Session Initiation Protocol) 프로토콜 기반 VoIP 서비스는 통신 절감 효과가 큰 장점과 동시에 다양한 부가서비스를 제공하여 사용자 수가 급증하고 있다. VoIP 서비스는 호(Call)를 제어하기 위해 SIP 기반으로 구성이 되며, SIP 프로토콜은 IP 망을 이용하여 다양한 음성과 멀티미디어 서비스를 제공하게 되는데 IP 프로토콜에서 발생하는 인터넷 보안 취약점을 그대로 동반하기 때문에 DoS(Denial of Service) 및 DDoS(Distribute Denial of Service)에 취약한 성향을 가지고 있다. DDoS 공격은 단시간 내에 대량의 패킷을 타깃 호스트 또는 네트워크에 전송하여 네트워크 접속 및 서비스 기능을 정상적으로 작동하지 못하게 하거나 시스템의 고장을 유도하게 된다. 인터넷 기반 생활이 일상화 되어 있는 현 시점에서 안전한 네트워크 환경을 만들기 위해 DDoS 공격에 대한 대응 방안이 시급한 시점이다. DDoS 공격에 대한 탐지는 매우 어렵기 때문에 근본적인 대책 마련에 대한 연구가 필요하며, 정상적인 트래픽 및 악의적인 트래픽에 대한 탐지 시스템 개발이 절실히 요구되는 사항이다. 본 논문에서는 SIP 프로토콜 및 공격기법에 대해 조사하고, DoS와 DDoS 공격에 대한 특성 및 종류에 대해 조사하였으며, SIP를 이용한 VoIP 서비스에서 IP 분류와 메시지 중복 검열을 통한 DDoS 공격 탐지기법을 제안한다.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.6
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• pp.687-695
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• 2002

Media-specific FEC techniques, suggested to confront with VoIP speech packet loss, improve speech quality at the expense of generating additional one-frame delay. In this paper, we suggest new media-specific FEC, i.e, LSF FEC technique which is able to improve speech quality with much shortened additional delay. In the proposed technique, the LSF parameters of the future frame are utilized to recover a lost packet. To evaluate performance of the proposed technique, we use ITU-T G.723.1 and G.729 Codec and apply Gilbert packet loss model and estimate MOS per every packet loss rate using PESQ speech quality estimation algorithm. The proposed technique has effect of shortening delay over from 6.5ms to 27ms compared with existing media-specific FEC techniques. Simulation results for comparison of reconstructed speech quality show this novel technique improves the MOS over 0.1 in practical lossy environment of 5 % packet loss rate.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2010.10a
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• pp.153-155
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• 2010

Internet phone (VoIP) services in Korea have achieved noticeable growth year after year since the service launching, and the growth still coninues. The market of mobile internet phone also expands sharply. Therefore, it is crucial to deploy networks which can support mobile internet phone services with excellent quality. For mobile internet phone services, it will be necessary to build and use networks with good mobility and high transmission rate. Current wireless networks for Internet services include 3G, Wi-Fi, and mobile WiMAX networks. 3G provides good mobility but lower transmission rate, whereas Wi-Fi exhibits excellent transmission rate but less mobility. Mobile WiMAX networks taking the merits of both, high mobility and transmission rate, are being deployed widely in recent years. This article examines the recent standardization efforts of WiMAX Forum for VoIP service in WiMAX networks.

• Journal of the Korea Society of Computer and Information
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• v.14 no.12
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• pp.147-156
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• 2009

This paper proposes a SRTP key exchange scheme using split transfer of divided RSA public key in SIP-based VoIP environment without PKI. The existing schemes are hard to apply to real VoIP environment, because they require a PKI and certificates in the end devices. But in case of ZRTP. which is one of existing schemes, it's able to exchange SRTP Key securely without PKI, but it is inconvenient since it needs user's involvement. To solve these problems, the proposed scheme will split RSA public key and transmit them to SIP signaling secession and media secession respectively. It can defend effectively possible Man-in-The-Middle attacks, and it is also able to exchange the SRTP key without the user's involvement. Besides, it meets the requirements for security of SRTP key exchange. Therefore, it's easy to apply to real VoIP environment that is not available to construct PKL.

• Proceedings of the KIEE Conference
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• 2004.11c
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• pp.237-239
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• 2004

Wireless LAN Voice over IP(VoIP) equipment needs Quality-of-Service(QoS) with priority for processing real-time traffic. This paper shows transmit function implementation of wireless LAN(WLANs) media access control(MAC) support VoIP, and it has an advantage of guarantee of QoS and is adaptable to VoIP or mobile wireless equipment. The IEEE 802.11e standard in progress has four queues according to four access categories(AC) for transmit and the MAC transmits the data based on EDCA. The value of AC is from AC0 to AC3 and AC3 has the highest priority. The transmit method implemented at this paper ensure QoS using one transmit FIFO in hardware since real-time traffic data and non real-time traffic data has the different priority. The device driver classifies real-time data and non real-time data and transmit data to hardware with information about data type. The hardware conducts shorter backoff and selects faster AIFS slot for real-time data than it for non real-time data. Therefor It make give the real-time traffic data faster channel access chance than non real-time data and enhances QoS.

• Journal of Internet Computing and Services
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• v.8 no.5
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• pp.141-150
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• 2007

This paper proposes new Steganographic model for VoIP that has very effective method using low bit encoding. Most of Steganographic models using Low bit Encoding have two disadvantages; one is that the existence of hidden secret message can be easily detected by auditory, the other is that the capacity of stego data is low. To solve these problems, this method embed more than one bit in inaudible range, so this method can improve the capacity of the hidden message in cover data. The embedding bit position is determined by using a pseudo random number generator which has seed with remaining message length, so it is hard to detect the stego data produced by the proposed method. This proposed model is able to use not only to communicate wave file with hidden message in VoIP environment but also to hide vary information which is user basic information, authentication system, etc.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.23 no.5
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• pp.867-871
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• 2013

In this paper, a major factor in determining voice quality that corresponds to the jitter and jitter buffer algorithm for removing jitter will be described. We analyze various jitter buffer algorithms and suggest ways to improve performance of jitter buffer algorithm.

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.113-118
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• 2005

Softswitch is the core BcN equipment which voice and multimedia switching based on the IP Technologies. It is designed to replace the Class 5(local Exchange) and Class 4(Toll Exchange) switch based on the circuit wired and wireless switching network technologies. Softswitch gets its name because typically it is a software based solution implemented on general purpose computers/servers. While the traditional PSTN switches are rely on dedicated facilities for T and S inter-connection and are designed primarily for voice communications. Packet based Softswitch is divided the control of call and bearer, very different from Public telephone network. Sometimes Call Agent or Media Gateway Controller, a key component in the VoIP solution, is also called Softswitch. This paper will suggest the software architecture of softswitch for performance in call processing part, also suggest the session management model to cover call setup latency.