• 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 Korea Contents Association Conference
• /
• 2004.05a
• /
• pp.342-347
• /
• 2004

The VoIP(Voice over Internet Protocol) technology which is able to use a voice service through internet is more cheaper then existing telephone charges, and is easily accept the various of multimedia services from internet. Previous connection method 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 SW 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.

• Proceedings of the Korean Information Science Society Conference
• /
• 2002.10e
• /
• pp.124-126
• /
• 2002

Voice over Internet Protocol(VoIP) is expected to be a major application on the Internet in the future This paper propose an approach to VoIP that uses Differentiated Service and Multi-protocol Label Switching(MPLS) to provide quantitative QoS guarantees over an IP network. An algorithm that determines QoS-constrained routes is proposed and a framework that uses such an algorithm for traffic engineering is outlined. the key component of this framework is a Centralize Resource Manager(CRM) responsible for monitoring and managing resources within the network and making all decisions to route／reroute traffic according to QoS requirement

• MALSORI
• /
• no.64
• /
• pp.105-119
• /
• 2007

Noise reduction technologies are indispensable to achieve acceptable speech quality in VoIP systems. This paper proposes a Wiener filter optimized to the estimated SNR of noisy speech for the noise reduction in VoIP environments. The proposed noise canceller is applied as a pre-processor before speech encoding. The performance of the proposed method is evaluated by the PESQ in various noisy conditions. In this paper, the proposed algorithm is applied to G.711, G.723.1, and G.729A which are all VoIP speech codecs. The PESQ results show that the performance of our proposed noise reduction scheme outperforms those of the noise suppression in the IS-127 EVRC and the ETSI standard for the advanced distributed speech recognition front-end.

• 제어로봇시스템학회:학술대회논문집
• /
• 2005.06a
• /
• pp.1391-1394
• /
• 2005

The VoIP (Voice over IP) is being used world-widely and already put to practical use in many fields. However, it is needed to ensure the security of VoIP call in special situations. It is relatively difficult to eavesdrop commonly used PSTN network in that a 1:1 circuit connects it. However, it is difficult to ensure the security of a call on Internet because many users are connected to the Internet concurrently. This paper suggests a new model for Internet telephony to prevent eavesdrops, using VoIP (using SIP protocol) with the use the VPN protocol and establish the feasibility of its practical use comparing it with the conventional Internet telephony.

• Journal of information and communication convergence engineering
• /
• v.6 no.1
• /
• pp.68-72
• /
• 2008

The VoIP(Voice over IP) has been worldwide used and already put to practical use in many fields. However, it is needed to ensure secret of VoIP call in a special situation. It is relatively difficult to eaves-drop the commonly used PSTN in that it is connected with 1:1 circuit. However, it is difficult to ensure the secret of call on Internet because many users can connect to the Internet at the same time. Therefore, this paper suggests a new model of Internet telephone for eavesdrop prevention enabling VoIP(using SIP protocol) to use the VPN protocol and establish the probability of practical use comparing it with Internet telephone.

• Proceedings of the Korea Information Processing Society Conference
• /
• 2014.11a
• /
• pp.218-220
• /
• 2014

VoIP(Voice over Internet Protocol)는 음성 및 화상과 같은 멀티미디어 세션을 인터넷과 같은 IP 기반 네트워크를 통해 통신하는 기술이다. 최근에는 기존의 PC 시스템 이외에 이동통신기기와 다양한 무선네트워크 기반 휴대용 기기들의 보급으로 VoIP 의 사용량은 크게 증가하고 있다. 하지만, 무선네트워크는 그 특성과 환경적 요인으로 NAT 에서의 차단, 지연, 유실등과 같이 통신의 연속성을 보장해 주지 못하는 문제가 발생할 수 있다. 본 논문에서는 무선네트워크에서 통신할 때 발생할 수 있는 이런 문제들에 대응하는 해결 방안을 제시하고 RTP 미디어 재생의 연속성을 보장하는 ICE 기반 다자간 VoIP 시스템 설계와 구현모델에 대해서 기술하고자 한다.

• Proceedings of the Korea Information Processing Society Conference
• /
• 2013.11a
• /
• pp.257-258
• /
• 2013

VoIP(Voice over Internet Protocol)는 인터넷과 같은 IP 망에서 음성과 영상을 전송하기 위한 기술이며, 차세대 망에서의 음성, 테이터 통신을 위한 기술로 부상되고 있다. 따라서 VoIP 응용은 인터넷 망을 이용하는 다양한 단말기들 사이의 음성 및 영상 통신을 위하여 더욱더 많이 사용되어 질 것으로 예상된다. 본 논문에서는 VoIP 기술을 여러 분야에 적용할 수 있는 응용성과 실제 다자간 음성통신의 구현 방법에 대해서 기술하겠다.

• Journal of KIISE:Computing Practices and Letters
• /
• v.9 no.1
• /
• pp.47-59
• /
• 2003

The Internet telephony service is one of the commercially successful Internet application services. VoIP technology makes the service come true. VoIP deploys H.323 or SIP as the standard protocol for the distributed multimedia services over the Internet in which QoS is not guaranteed. VoIP carries the packetized voice over the RTP/UDP/IP protocol stack. The data transmission trouble is caused by UDP when the service is provided in private networks and some ISP-provided Internet access networks in the private address space. The Internet telephony users in such networks cannot listen the voices of the other parties in the public Internet or PSTN. Making the problem more difficult, the Internet telephony service considered in this paper gets the incoming voice packets of every session through only one UDP port number. In this paper, three schemes including the terminal proxy, the gateway proxy, and the protocol translation are suggested to solve the problems. The design and implementation of the NAT proxy server based on gateway proxy scheme are described in detail.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.12 no.6
• /
• pp.1022-1025
• /
• 2008

Session Initiation Protocol (SIP) has become the call control protocol of choice for Voice over IP (VoIP) networks because of its open and extensible nature. However, the integrity of call signaling between sites is of utmost importance, and SIP is vulnerable to attackers when left unprotected. Currently a herby-hop security model is prevalent, wherein intermediaries forward a request towards the destination user agent sewer (UAS) without a user agent client (UAC) knowing whether or not the intermediary behaved in a trusted manner. This paper presents an integrated security model for SIP-based VoIP network by combining hop-by-hop security and end-to-end security.