• Journal of the Korea Computer Industry Society
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• v.3 no.5
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• pp.631-640
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• 2002

The internet phones that are currently available in use adopt packet exchange system, transferring through various routes and lacking sufficient band width with a result that there is an accompanied delay for packet transmission since the traffic is increased, accordingly affecting a lot in sound quality and speed. Two solutions for such troubles are suggested in this study to improve sound quality of internet phones. Firstly, we minimize the delay and damage regarding packet size based on traffic size by using the data algorithm from variable packets in order to supplement decreased sound quality due to the delay and damage of sound data. The second suggestion is to employ a method of Jitter compensation by giving an appropriate initial delay time with regenerating buffers to bypass troubles from Jitter, From employing the Jitter compensation method, we found that there is a sound quality improvement due to the less stoppage phenomenon.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.9A
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• pp.2305-2312
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• 1998

This paper analyzed the delay characteristics of speech packets in virtual cellular network(VCN). The probability distribution of packet delay is obtained using the markov chain model when periodic speech packets are transmitted by slotted-ALOHA protocol. The effects of probility of capture and retransmission policy on the performance were also analyzed. At first, the probability cumulative function of packet delay is calculated from the probability of capture as a function of location of mobile terminal. In order to investigate the effects of backoff delay, we defined a parameter NPr, where N is the period (frame size) of the speech packets and Pr is the retransmission probability for each speech packet. We also obtained the 1% outage delay for various frame size N.

• The KIPS Transactions:PartC
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• v.10C no.7
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• pp.937-942
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• 2003

In this paper, we measured and examined RTT delays and packet losses according to the changes of stationary loads for two typical stream-type traffics, a DV and a MPGE2 on the R＆D Gigabit Network testbed, JGN. As the result of our actual measurements, we realized that the packet size of stationary load have no effects on a DV and a MPGE2 stream on the very high-speed network(50Mbps, IP over ATM). When its bandwidth and stationary load exceeds 95％ of network bandwidth, packet losses appeared and RTT delay increased rapidly. Also we realized that the number and size of Receive ＆ Transmit buffer on the end systems have no effects on packet losses and RTT delays.

• Journal of Korea Society of Industrial Information Systems
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• v.4 no.4
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• pp.1-6
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• 1999

When continuous media are transmitted over the communication networks, asynchrony which can not maintain temporal relationships among packets my occur due to a random transit delay. There exist two types of synchronization schemes ; for guaranteed or non-guaranteed resource networks. The former which applies a resource reservation technique maintains delay characteristics however, the latter supply a best-effort service. In this paper, I propose a intra-media synchronization scheme to transmit continuous media on general networks not guaranteeing a bounded delay time. The scheme controls transmission times of the packets by estimating next delay time with the delay distribution So, the arriving packets my be maintained within a limited delay boundary, and playout will be performed after buffering to smoothen small delay variations. To prevent network congestion and maintain minimum quality of service the transmitter performs media scaling-down by dropping the current packet when informed excessive delay from the receiver.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.9A
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• pp.684-693
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• 2003

This paper analyzes transmissiondelays of a message and a packet in a time-varying and packet-based radio link. The paper assumes that thearrivals of messages have a Bernoulli process and the lengths of the messages a exponential distribution. To reflect the feature of the time-varying radio link, we use a two-state Markov model. From the model the mean transmission delay of and the mean queue length of the packet are analyzed in terms of the packet distribution function, the packet transmission service time, and the PER of the radio link. And the mean message transmission delay time and the mean queue length are derived using the performance indices of the packet. Numerical results show that the message arrival rate and the message length have some bounds to keep the transmission of the message steady and to improve the performance indices of the message. It can be known that the PER of the state influences on the performance indices more than the sojourn time of the state.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.9
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• pp.373-383
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• 2003

In this paper we discuss the effect of the delay limit and the packet size related to the quality of service on a VoIP system using the Internet. We also provide a guideline to determining the optimal packet size of the voice data for a given delay limit. Empirical studies are done with two personal computers connected through the packet switched public IP network. The sender encodes the voice signal from the microphone to get PCM and ADPCM data and sends the data to the receiver using UDP packets. The receiver plays the reconstructed voice from the stream with lost and delayed packets. The quality of the reconstructed voice is evaluated offline by the MNB (Measuring Normal Block) method using the data acquired from the both sides. The result shows that under the delay limit of 100ms for 40Kbps, 32Kbps and l6Kbps of ADPCM data, the minimum packet size should be 300bytes, 400bytes and 600bytes respectively and the maximum packet size should be l200bytes commonly for the best quality of voice.

• Proceedings of the Korean Information Science Society Conference
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• 2000.04a
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• pp.454-456
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• 2000

실시간의 특성을 가지는 데이터의 경우 네트워크상에서 분실된 패킷을 복구시키기 위해서 FEC 방법을 사용한다. FEC는 최소한의 지연만으로 손실 패킷의 복구를 효율적으로 할 수 있는 장점을 가지고 있으나 네트워크상에서의 패킷 손실 특성에 많이 의존되는 경향이 있다. ITU-T의 Study Group 16 에서의 Real-Time Transport Protocol(RTP)를 사용하여 네트워크에서 분실된 패킷을 복원시키는 방법으로 Media-independent error-correction scheme을 정하였다. 이 Scheme에 의해 만들어진 error-correction을 위한 신호화 media bitstream은 UDP 에 의해 encapsulation될 RTP에 실리게 된다. Scheme은 real-time이라는 환경에 유리하도록 bandwidth 와 latency 그리고 cost를 최소화하려고 했으며 이에 따라 네 가지 scheme을 정하였다. 네 가지의 Scheme은 오버헤드와 지연시간이 크기가 차별화를 두어 네트워크 환경의 변화에 적응하도록 하였다. 그러나 네트워크 환경에 보다 더 탄력적이며 효율적으로 적응하기 위해서 또 하나의 scheme을 제안한다. 새로 고안한 이 다섯 번째 scheme은 scheme 3 에 비해 작은 latency를 가지고 장점이 있는 반면 연속적으로 분실된 패킷에 대한 복원확률이 다소 떨어진다. 하지만 scheme 1과 2에 비해서는 연속적인 패킷 분실의 복원확률이 높아 네트워크환경에 따라 scheme 4를 사용하면 네 개의 scheme을 사용하여 분실패킷의 복원을 하는 경우보다 보다 효율적인 전송과 복원이 이루어질 것이다.

• The KIPS Transactions:PartC
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• v.12C no.3 s.99
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• pp.437-442
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• 2005

The measurement of RTT (Round Trip Time) can be used for the analysis of Internet congestion. However, simple measuring of RTT which measures only hun around time of a packet can not infer a packet forward/backward delay variation. In this thesis, we present a new algorithm which can be used for the estimation of forward/backward delay variation of packets. These delay variations are implication of network congestion state. In this algorithm, the reference forward/backward delay can be determined based on the minimum RTT value. The delay variation of each packet can be calculated by comparing reference delay with the packet delay. We verified our proposed algorithm by NS-2 simulation and delay measuring in a real network.

• 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 Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.6A
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• pp.445-454
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• 2005

To measure the performance of call quality, we have built the experiment environment and observed that the delay caused by encapsulation between internet and VoIP telephones is under 5ms at most. The major delay is assumed to be the time required to capsulate the packet for tunnelling of VPN. Because the difference of average delay time is under $4ms{\sim}5ms$, the difference of call quality between VoIP and VoIP telephone adopting VPN is negligible. We have concluded that the capsulation process between PAC and PNS is the major factor influencing the network load by changing the number of fames in a packet during communication Also, we have concluded that the most suitable frame numbers is tow or three by adding the frame numbers in a packet to obtain the suitable frames in a packet and setting up end-to-end delay under 150ms.