• Proceedings of the Korea Contents Association Conference
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• 2006.11a
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• pp.75-79
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• 2006

In a VoIP(Voice over IP) system which support the realtime speech service, speech quality is deteriorated by the delay, the jitter, the loss, and the reversed packet order. In this thesis, APBC for receiver site is proposed, which compensate the jitter by the adaptive playout algorithm and conceal the packet loss, and align the packet order. Also, a VoIP application system is implemented, and the performance of APBC is verified on the implemented system by measuring the processing speed and the speech quality. From the result, processing speed is 257$\mu$sec, which is fast enough to deal with packet being received in realtime. Also, the speech quality by MOS(Mean Opinion Score) is improved as 18 percent compared with algorithm of fixed playout delay.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.6
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• pp.1313-1319
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• 2011

VoIP service by taking advantage of the current PSTN network and internet over the existing telephone network at an affordable price allows you to make voice calls to the service is being expanded. However, the security of public must be maintained for security vulnerabilities in Smart Phone VoIP case problems arise, and is likely to be attacked by hackers. In this paper, the Internet, using wired and Smart Phone VoIP services may occur during analysis of the type of incident and vulnerability analysis, the eavesdropping should conduct an attack. Smart Phone VoIP with institutional administration to analyze the vulnerability OmniPeek, AirPcap the equipment is installed in a lab environment to conduct eavesdropping attack. Packet according to the analysis and eavesdropping attacks, IP confirmed that the incident as an attack by the eavesdropping as to become the test proves. In this paper, as well as Smart Phone VoIP users, the current administration and the introduction of Smart Phone service and VoIP service as a basis for enhanced security will be provided.

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

• 한국정보통신설비학회:학술대회논문집
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• 2007.08a
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• pp.331-335
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• 2007

방송과 초고속인터넷 서비스를 동시에 제공할 수 있는 HFC(Hybrid Fiber Coaxial) 망은 상/하향이 비대칭 구조이며, 하향속도에 비해 상향속도가 1/10 수준이어서 상향 트래픽이 과다하게 생성될 경우 인터넷속도 지연이 발생한다. 지연에 민감한 VoIP 서비스의 품질보장 방안으로는, DOCSIS(Data Over Cable System Interface Specification) 1.1 기반의 상향 스케쥴링 기능을 이 용한 VoCM(Voice Over Cable Modem)이 있다. 그러나 별도의 VoCM을 사용해야 하며 아날로그 전화기를 사용해 IP 기반의 VoIP 단말을 사용할 수 없다는 단점이 있다. 일반 CM(Cable Modem)에 DOCSIS 1.1 Config File을 이용하여 VoIP 품질을 보장할 경우 별도의 트래픽 대역을 항상 점유해야 하는 단점이 있다. 이에, 본 논문에서는 효율적 대역폭 이용과 단말장비에 종속적이지 않은 방안을 제안하고 일반 CM을 통한 유무선 환경하에서 Dynamic QoS(Quality Of Service)를 제공할 수 있는 PCMM(Packet Cable MultiMedia) 적용 방안 및 시험결과에 대해 고찰하고자 한다.

• Journal of The Institute of Information and Telecommunication Facilities Engineering
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• v.2 no.3
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• pp.9-16
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• 2003

In this paper, we propose the burst error reduction (BER) algorithm for VoIP service in the wireless LAN network. In end point device, this BER algorithm can be achieved packet loss bounded QoS provisioning using interleaving in buffering and FEC (Forward Error Correction) through transmitting voice packet. BER algorithm can reduced the voice packet loss rate 5.5%-60% in VoIP network using wireless LAN.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2002.11a
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• pp.844-848
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• 2002

In this paper, we present a VoIP service provisioning architecture based on MEGACO/H.248 which is one of the key protocols for VoIP services. MEGACO/H.248 is a media gateway control protocol standardized by both ITU-T and IETF, and many ITSPs, carriers, and vendors currently have a lot of interest in the protocol. MEGACO/H.248 is used by a softswitch a key component of the next generation VoIP network, in order to control various media gateways and provide seamless interworking between PSTN and Yon networks.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.6 no.4
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• pp.1006-1025
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• 2012

Improving the quality of service in IP networks is a major challenge for real-time voice communications. In particular, packet arrival-delay variation, so-called "jitter," is one of the main factors that degrade the quality of voice in mobile devices with the voice-over Internet protocol (VoIP). To resolve this issue, a receiver-based enhanced timing recovery algorithm combined with active jitter estimation is proposed. The proposed algorithm copes with the effect of transmission jitter by expanding or compressing each packet according to the predicted network delay and variations. Additionally, the active network jitter estimation incorporates rapid detection of delay spikes and reacts to changes in network conditions. Extensive simulations have shown that the proposed algorithm delivers high voice quality by pursuing an optimal trade-off between average buffering delay and packet loss rate.

• 한국IT서비스학회:학술대회논문집
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• 2008.11a
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• pp.139-142
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• 2008

VoIP(Voice over IP)는 패킷교환망을 통해 음성통신을 제공하는 기술이다. 초기의 VoIP에서는 사용상 제약조건과 불편함 때문에 사업모델이 성립하지 못하였다. 무선 광대역통신이 가능해지고, 멀티미디어 서비스에 적합하게 진화함에 따라 VoIP에서 무선접속기술을 이용한 Mobile VoIP서비스를 제공하고 있다. Mobile VoIP의 편리성으로 인하여 VoIP에서 새로운 사업모델이 창출되었고, 웹 2.0과 Mobile 인터넷전화의 결합에 의해 SNS(Social Network Service)으로 서비스가 확장되었다. 이로 인해 많은 서비스 제공자가 발생하였고, 이 서비스 제공 사업자는 크게 이동통신사업자, 소프트웨어 서비스 사업자, 가상이동통신사업자, 그리고 별정 사업자로 분류된다. 4가지의 서비스 제공 사업자는 각각 사업모델이 차별화 되어 있다. 본 논문에서는 Mobile 인터넷전화 기술동향을 살펴보고 서비스 동향을 분석한다. 그리고 Mobile 인터넷전화의 서비스 제공사업자의 차별화된 사업모델을 분석한다.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.8B
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• pp.683-690
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• 2003

In this paper we present a theoretical framework for the network design with delay QoS guarantee to a voice at the packet level. Especially, we propose a method for estimating the bandwidth at the ingress edge routers accommodating the voice connections and data sessions in the next-generation If network. First, we describe network architecture for VoIP (Voice over IP) services in the NGN (Next Generation Network). After that, we propose a procedure for dimensioning the bandwidth at the output port of a router that accommodates voice and data traffic using the non-preemptive queuing system with strict priority service scheme. Via numerical experiments we illustrate the implication of the proposition.