• 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 Korean Society of Broadcast Engineers Conference
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• 1996.06a
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• pp.153-158
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• 1996

Delay and jitter are critical factors in the real-time video services over ATM network. Mostly, delay and jitter problem are generated in input buffer when video are multiplexed. In this paper, we analyze delay and jitter of input buffer, and consider efficient control and flexible bandwidth allocation of video traffic. Also, we analyze decision of buffer size related maximum allowable delay.

• The KIPS Transactions:PartC
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• v.17C no.3
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• pp.299-306
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• 2010

IPTV and VoIP services are considered as killer applications over Mobile WiMAX network, which provideshigh mobility and data rate. Among those which affect the quality of voice in those services, the jitter buffer or playout buffer can compensate the poor voice quality caused by the packet drop due to frequent route change or differences among routes between service endpoints. In this paper, we analyze the correlation between the sleep interval length and jitter buffer size in order to guarantee a predefined level of voice quality. For this purpose, we present an end-to-end delay model considering additional delay incurred by the WiMAX PSC-II sleep mode and a VoIP service quality requirement based on the delay constraints. Through extensive simulation experiments, we also show that the increase of jitter buffer size may degrade the voice quality since it can introduce additional packet drop in the jitter buffer under WiMAX power saving mode.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.6 no.3
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• pp.386-392
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• 2002

Internet telephony is an application that transmits voice data for conversation. Therefore it must provide high sound quality. However while audio packets are transferred through the network, they are affected by delay variations and jitters, which could result in poor sound quality of the receiving end does not have an appropriate jitter buffer to overcome network factors. This thesis introduces a buffer management algorithm that could be used to provide better sound quality for Internet phone terminals. This algorithm actively responds to both the compression algorithms that are used by the terminals, as well as to the received data to provide an improvement in sound quality. In order to verify the effectiveness of the proposed algorithm, we experimented in variance network settings. The results show that the proposed algorithm improves on the performance of the conventional buffer management algorithm.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2002.05a
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• pp.384-387
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• 2002

Internet telephony is an application that transmits voice data for conversation. Therefore it must provide high sound quality. However while audio packets are transferred through the network, they are affected by delay variations and jitters, which could result in poor sound quality if the receiving end does not have an appropriate jitter buffer to overcome network factors. This thesis introduces a buffer management algorithm that could be used to provide better sound quality for Internet phone terminals. This algorithm actively responds to both the compression algorithms that are used by the terminals, as well as to the received data to provide an improvement in sound quality. In order to confirm the validity of the suggested algorithm, comparisons of the performance have been made between the existing buffer management algorithms and this new algorithm in various network settings.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12C
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• pp.199-207
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• 2001

We propose an adaptive shaping controller equipped with the technologies of shaping and buffering VoIP packets arriving at the receiving end by the CAM-type controller. In order to conceal interarrival packet delay variation, the conventional jitter buffers force them to be too large, thereby causing the audio quality to suffer excessive delay. However, by using our proposed method, the delay caused by shaping operation dynamically increases or decreases on the level of jitter that exists with in the IP network. This makes the delay accommodates adaptively the network jitter condition. The less jitter network has the fewer delay the shaping controller requires for jitter elimination. And the CAM-type method generally makes the shaping operation faster and leads to processing packets in as little time as can. We analyse the packet loss and delay performance dependency on the average talk ratio and the number of jitter buffer entries in shaping controller. Surprising, we show that the average delay using our shaping controller is about 70msec. This performance is much better than with the delay equalization method which forces the receiving end to delay about 60msec.

• IEIE Transactions on Smart Processing and Computing
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• v.5 no.3
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• pp.207-214
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• 2016

Optical communication systems are rapidly spread following increases in data traffic. In this work, a 32-Gb/s inductorless output buffer circuit with adjustable pre-emphasis is proposed. The proposed circuit consists of an output buffer circuit and an emphasis circuit. The emphasis circuit emphasizes the high frequency components and adds the characteristics of the output buffer circuit. We proposed a design method using a small-signal equivalent-circuit model and designed the compensation characteristics with a 65-nm CMOS process in detail using HSPICE simulation. We also realized adjustable emphasis characteristics by controlling the voltage. To confirm the advantages of the proposed circuit and the design method, we fabricated an output buffer IC with adjustable pre-emphasis. We measured the jitter and eye height with a 32-Gb/s input using the IC. Measurement results of double-emphasis showed that the jitter was 14% lower, and the eye height was 59% larger than single-emphasis, indicating that our proposed configuration can be applied to the design of an output buffer circuit for higher operation speed.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.10 no.4
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• pp.93-99
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• 2011

This paper proposes a novel network jitter estimation algorithm for robust VoIP communication system. The proposed method computes the current network environment mode using the differences of arrival time and generation time from sequential received packets. According to the current network environment mode, the jitter variance weights is adjusted to minimize the error for estimating the network jitter. The jitter average and variance are calculated by the autoregressive estimated algorithm, and then the network jitter is estimated by applying the jitter variance weights.

• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.722-737
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• 2011

In this paper, we propose an estimation of network jitter that occurs when transmitting TCP packets containing MPEG-2 TS in progressive streaming service over wired or wireless Internet networks. Based on the estimated network jitter size, we can calculate required de-jitter buffering time to absorb the network jitter at the receiver side. For this purpose, by exploiting the PCR timestamp existing in the TS packet header, we create a new timestamp information that is marked in the optional field of TCP packet header to estimate the network jitter. By using the proposed de-jitter buffering scheme, it is possible to employ the conventional T-STD buffer model without any modification in the progressive streaming service over IP networks. The proposed method can be applicable to the recently developed international standard, MPEG DASH (dynamic adaptive streaming over HTTP) technology.

• The Transactions of the Korea Information Processing Society
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• v.7 no.2S
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• pp.721-732
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• 2000

Multimedia communication systems require not only high-performance computer hardwares and high-speed networks, but also a buffer management mechanism to process many data efficiently. Two buffer handling methods, Push and Pull, are commonly used. In the Push method, a server controls the flow of dat to a client, while in the Pull method, a client controls the flow of data from a server. Those buffering schemes can be applied to the data transfer between the packet receiving buffer, which receives media data from a network server, and media playout devices, which play the recived media data. However, the buffer management mechanism in client-sides mainly support either one of the Push or the Pull method. Consequently, they have some limitations to support various media playout devices. Futhermore, even though some of them support both methods, it is difficult to use since they can't provide a unified structure. To resolved these problems, in this paper, we propose an efficient and flexible Push/Pull buffer management mechanism at client-side. The proposed buffer management scheme supports both Push and Pull method to provide various media playout devices and to support buffering function to absorb network jitter. The proposed scheme can support the various media playback devices using a single buffer space which in consequence, saves memory space compared to the case that a client keeps tow types of buffers. Moreover, it facilitates the single buffer as a mechanism for the absorbing network jitter effectively and efficiently. The proposed scheme has been implemented in an existing multimedia communication system, so called ISSA (Integrated Streaming Service Architecture), and it shows a good performance result compared to the conventional buffering methods in multimedia communication environments.