• The Journal of the Acoustical Society of Korea
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• v.37 no.6
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• pp.512-517
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• 2018

Improving voice quality is a major concern in telecommunications. In this paper, we propose a robust speech quality enhancement against background noise and packet loss at VoIP (Voice-over-IP) receiver. The proposed method combines network jitter estimation based on hybrid Markov chain, adaptive playout scheduling using the estimated jitter, and speech enhancement based on restoration of amplitude and phase to enhance the quality of the speech signal arriving at the VoIP receiver over IP network. The experimental results show that the proposed method removes the background noise added to the speech signal before encoding at the sender side and provides the enhanced speech quality in an unstable network environment.

• Proceedings of the IEEK Conference
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• 2003.07a
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• pp.430-433
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• 2003

In this paper, we propose the method of voice packet transformation for the improvement of voice quality in embedded VoIP system terminals. For this purpose, it was analyzed about the RTP header in the voice packet receiving side and designed about the handling method of voice packets by using jitter buffer. Through this analyzed and designed results, by implementing the voice packet transformation method and testing on the self-product VoIP system, we reached that a conclusion is satisfied with performance of VoIP services.

• 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.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4
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• pp.349-360
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• 2002

In ITU H.323 teleconference system, the RTP/RTCP protocol is offered to transfer real-time multimedia stream. Both sender and receiver hate experience in packet loss and jitter which result from network congestion over Internet. Audio quality over Internet depends on the number of lost packets and on jitter between successive packets. The goal of our study is to improve the speech quality over Internet by checking the packet loss characteristics of the network and adopting the but for control management mechanism at the receiver. We suggest a dynamic redundant audio transmission mechanism which examines the packet loss rate and uses the feedback information through RTCP.

• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.218-223
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• 2008

In this paper, we implement several methods to improve an audio collaboration system over IP networks, and then evaluate the performance of the implemented methods. In general, speech and audio quality degrades depending on the characteristics of IP networks such as jitter and packet loss. In order to reduce this quality degradation, we propose a lower bit rate audio delivery scheme using the MPEG-2 AAC (Advanced Audio Coding) audio codec in a viewpoint that a packet loss rate could be reduced by a smaller packet size. In addition, iLBC (Internet Low-Bitrate Codec) and the G.711 packet loss concealment algorithm defined by IEFT and ITU-T, respectively, are applied to a audio collaboration system. RAT (Robust-Audio Tool)[7] is used as a baseline platform for the implementation of the proposed methods. It is shown from the implementation that the implemented MPEG-2 AAC audio codec with a bitrate of 256 kbit/s performs as similar as the uncompressed audio quality with a bitrate of 512 kbit/s, and that iLBC and the G.711 packet loss concealment algorithm can improve speech quality when a packet loss rate is 2~10%.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.41 no.11
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• pp.79-85
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• 2004

A digitally-controlled analog-block inevitably undergoes the bang-bang oscillations which may cause a big amplitudes of the glitches if the oscillation occurs at the MSB transition points of a binary counter. The glitch results into the jitter noise for the case of the DLL. In this paper, we devise a new counter code that has the hysteresis in the bit transitions in order to prevent the transitions of the significant counter-bits at the locking state. The maximum clock jitter is simulated to considerably reduce over the voltage-temperature range guaranteed by specifications. The counter is employed to implement the high speed packet-base DRAM and contributes to the maximized valid data-window.

• ETRI Journal
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• v.33 no.3
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• pp.326-334
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• 2011

This paper outlines the time jitter effect of a sampling clock on a software-defined radio technology-based digital intermediate frequency (IF) transceiver for a mobile communication base station. The implemented digital IF transceiver is reconfigurable to high-speed data packet access (HSDPA) and three bandwidth profiles: 1.75 MHz, 3.5 MHz, and 7 MHz, each incorporating the IEEE 802.16d worldwide interoperability for microwave access (WiMAX) standard. This paper examines the relationship between the signal-to-noise ratio (SNR) characteristics of a digital IF transceiver with an under-sampling scheme and the sampling jitter effect on a multichannel orthogonal frequency-division multiplexing (OFDM) signal. The simulation and experimental results show that the SNR of the OFDM system with narrower band profiles is more susceptible to sampling clock jitter than systems with relatively wider band profiles. Further, for systems with a comparable bandwidth, HSDPA outperforms WiMAX, for example, a 5 dB error vector magnitude improvement at 15 picoseconds time jitter for a bandwidth of WiMAX 3.5 MHz profile.

• The Journal of the Korea Contents Association
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• v.6 no.2
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• pp.9-16
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• 2006

This paper proposes a scheduling algorithm for efficiently processing multimedia pakcet services in OFDMA physical system of the future broadband wireless access networks. The scheduling algorithm uses wireless channel state estimation, and allocates transmission rates after deciding transmission ordering based on class and priority policy. As the result, the proposed scheduling algorithm offers maximum throughput and minimum jitter for realtime services, and fairness for non-realtime services. In simulation study, the proposed algorithm proves superior performances than traditional round robin method.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.11 no.9
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• pp.3502-3509
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• 2010

Streaming service in the wireless mobile network environment has been a very challenging issue due to the dynamic uncertain nature of the channels. Overhead such as congestion, latency, and jitter lead to the problem of performance degradation of an adaptive multi-streaming service. This paper proposes a AMSS (Adaptive Multi-level Streaming Service) mechanism to reduce the performance degradation due to overhead such as variable network bandwidth, mobility and limited resources of the wireless mobile network. The proposed AMSS optimizes streaming services by: 1) use of fuzzy similarity metric, 2) minimization of packet loss due to buffer overflow and resource waste, and 3) minimization of packet loss due to congestion and delay. The simulation result shows that the proposed method has better performance in congestion control and packet loss ratio than the other existing methods of TCP-based method, UDP-based method and VBM-based method. The proposed method showed improvement of 10% in congestion control ratio and 8% in packet loss ratio compared with VBM-based method which is one of the best method.