• 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.

• Journal of the Optical Society of Korea
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• v.13 no.3
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• pp.373-378
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• 2009

An Ethernet passive optical network (EPON) is a low-cost, high-speed solution to the bottleneck problem of a broadband access network. This paper analyzes the delay and the jitter performance of dynamic bandwidth allocation (DBA) schemes for differentiated services in EPONs. Especially, the average packet delay and the delay jitter of the expedited forwarding (EF) traffic class are compared, with consideration as to whether a cyclic or an interleaved polling scheme is superior. This performance evaluation reveals that the cyclic polling based DBA scheme provides constant and predictable average packet delay and improved jitter performance for the EF traffic class without the influence of load variations.

• The Journal of the Acoustical Society of Korea
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• v.33 no.2
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• pp.133-138
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• 2014

Packet arrival-delay variation, so-called 'jitter' is one of the main factors that degrade the quality of voice in mobile devices at the Voice over Internet Protocol (VoIP). To resolve this issue, a playout scheduling based on adaptive jitter estimation for enhancing VoIP speech quality 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. The experimental results have shown that the proposed algorithm delivers high voice quality in unstable network environment.

• Proceedings of the Optical Society of Korea Conference
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• 2002.11a
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• pp.58-59
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• 2002

Jitter noise power at the all optical packet header processor, which uses a fiber-optic delay-line matched filter, has been calculated for various optical pulse shapes.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.10A
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• pp.877-885
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• 2005

The Home Phoneline Networking Alliance (HomePNA) 2.0 technology can establish a home network using existing in-home phone lines, which provides a channel rate of 4-32 Mbps. HomePNA 2.0 Medium Access Control(MAC) protocol adopts an IEEE 802.3 Carrier Sense Multiple Access with Collision Detection (CSMA/CD) access method, Quality of Service(QoS) algorithm, and Distributed Fair Priority Queuing(DFPQ) collision resolution algorithm. In this paper, we propose some mathematical models about the important elements of HomePNA 2.0 MAC protocol performance, which are Saturation Throughput, Packet Delay and Packet Jitter. Then, we present an overall performance analysis of HomePNA 2.0 MAC protocol along with simulations.

• 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.

• Proceedings of the IEEK Conference
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• 2001.06c
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• pp.93-96
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• 2001

In this paper, we propose a delay-constrained retransmission method to control packet error or loss in common internet. The Delay Regulator(or Jitter Buffer) which is used to control errors caused by unreliable UDP connection, stores received data packets fDr a small amount of time to prevent network jitter from affecting display quality, which causes constant delay. In this paper, we propose a retransmission method to increase efficiency of ARQ(Automatic Repeat reQuest) by using characteristic of delay regulator.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.6
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• pp.1168-1176
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• 2003

Internet telephony service is very cheap and very easy to introduce the value-added service than the POTS, but is difficult to the QoS of telephone service. The existing internet typically offers "best effort" services only, which do not make any commitment about delay, packet loss and jitter. This paper compensates the low quality of the speech for packet loss or delay using FEAPT algorithm in Internet phone. In the FEAPT algorithm, the receiver expands the received packet under resonable threshold, and hence compensates the QoS of speech.of speech.

• 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.