• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.118-123
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• 2009

In this paper, we propose packet selection and significance based interval allocation algorithm for real-time streaming service. In real-time streaming of inter-frame (and layer) coded video, minimizing packet loss does not imply maximizing QoS. It is true that packet loss adversely affects the QoS but one single packet can have more impact than several other packets. We exploit the fact that the significance of each packet loss is different from the frame type it belongs to and its position within GoP. Using packet dependency and PSNR degradation value imposed on the video from the corresponding packet loss, we find each packet's significance value. Based on the packet significance, the proposed algorithm determines which packets to send and when to send them. The proposed algorithm is tested using publicly available MPEG-4 video traces. Our scheduling algorithm brings significant improvement on user perceivable QoS. We foresee that the proposed algorithm manifests itself in last mile connection of the network where intervals between successive packets from the source and to the destination are well preserved.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.10C
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• pp.950-958
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• 2007

This paper presents an efficient packet loss resilient scheme for real-time video transmission over the Internet. By analyzing the temporal and spatial dependencies in inter- and intra-frames, we assign forward error correction codes (FEC) across video packets for minimizing the effect of error concealment and error propagation from packet loss. To achieve optimal allocation of FEC codes, we formulate the effect of packet loss on video quality degradation as packet distortion model. Then we propose an unequal FEC assignment scheme with low complexity based on packet correction rate, which uses the packet distortion model and includes channel status information. Simulation results show that the proposed FEC assignment scheme gives substantial improvement for the received video quality in packet lossy networks. Furthermore the proposed scheme achieves relatively smaller degradation of video quality with higher packet loss rates.

• Journal of the Korea Society for Simulation
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• v.11 no.1
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• pp.1-10
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• 2002

IETF proposed Fast Handover mechanism tn reduce the latency during which the mobile node is effectively dusconnected from the Internet. In this paper, we propose three buffering strategies and a bicasting strategy to resolve packet loss and packet disordering problems in Fast Hangover mechanism. Through Java-based simulation we show thats, packet loss and disordering problems have been resolved by our proposal. As the result the buffering and bicasting strategies can reduce the handover delay caused by packet loss and packet disordering, in addition.

• MALSORI
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• no.43
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• pp.101-112
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• 2002

In real time packetized voice application, missing packets are major source of voice quality degradation. Thus packet loss concealment (PLC) algorithms are needed to guarantee QoS of VoIP. In this paper, we describe packet loss concealment scheme utilizing the next good frame which follows loss packets. When this scheme is combined with other PLC algorithms, such as G.711 pitch waveform replication recommended by ITU-T LP based PLC algorithm, additional voice quality improvement is obtained for consecutive packet loss larger than 60 msec.

• The Journal of the Acoustical Society of Korea
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• v.32 no.2
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• pp.157-166
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• 2013

This paper proposes a non-intrusive speech quality estimation method considering the effects of packet loss to perceptual quality. Packet loss is a major reason of quality degradation in a packet based speech communications network, whose effects are different according to the input speech characteristics or the performance of the embedded packet loss concealment (PLC) algorithm. For the quality estimation system that involves packet loss effects, we first observe the packet loss of G.729 codec which is one of narrowband codec in VoIP system. In order to quantify the lost packet affects, we design a classification algorithm only using speech parameters of G.729 decoder. Then, the degradation values of each class are iteratively selected that maximizes the correlation with the degradation PESQ-LQ scores, and total quality degradation is modeled by the weighted sum. From analyzing the correlation measures, we obtained correlation values of 0.8950 for the intrusive model and 0.8911 for the non-intrusive method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.9B
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• pp.1272-1280
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• 2010

WFQ (Weighted Fair Queuing) provides not only fairness among traffic flows in using bandwidth but also guarantees the Quality of Service (QoS) that individual flow requires, which is why it has been applied to the resource reservation protocol (RSVP)-capable router. The RSVP allocates an enough resource to satisfy both the rate and end-to-end delay requirements of the flow in condition of no packet loss, and the WFQ guarantees those QoS requirements with the allocated resource. In a practice, however, most QoS-guaranteed services, specially the Voice of IP, allow a few percent of packet loss, so it is strongly desired that the RSVP and WFQ make the best use of this allowable packet loss. This paper enhances the WFQ to allow packet loss and investigates its performance. The performance evaluation showed that allowing the packet loss of 0.4% can improve the flow admission capability by around 40 percent.

• Proceedings of the KSPS conference
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• 2003.10a
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• pp.109-112
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• 2003

In real time packetized voice applications, missing frames is a major source of voice quality degradation. Thus packet loss concealment(PLC) algorithms are needed to guarantee the QoS of the VoIP. Still current speech codecs for VoIP work poor when consecutive packet losses are issued. In this paper, we proposed a new PLC algorithm for the G.729 codec. Our algorithm works better especially when the consecutive packet loss occurs mainly because it adopts an adaptive gain controller utilizing the number of missing packet information combined with a fixed codebook vector estimation algorithm and LPC bandwidth expansion.

• Journal of Internet Computing and Services
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• v.6 no.4
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• pp.115-124
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• 2005

The packet loss problem that occurs in the mobile multicast (MoM) protocol due to designated multicast service provider(DMSP) handoff is investigated through simulation experiments for several DMSP selection policies. Then, two enhanced DMSP schemes are proposed to minimize the packet loss of the MoM protocol with single DMSP. The first scheme uses a backup DMSP and greatly reduces the packet loss rate at the expense of the increased network traffic. The second scheme utilizes the extended DMSP operation and shows many desirable features such as the almost zero packet loss rate and relatively low network traffic.

• Journal of Institute of Control, Robotics and Systems
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• v.18 no.8
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• pp.768-773
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• 2012

Packet loss compensation techniques are increasingly important to stable remote control over wireless communication in WNCS (Wireless Networked Control Systems). Its time varying channels, limited bandwidth, interference, and poor signal not only leads to packet loss or latency, but also can negatively affect performance and system stability. This paper presents a compensation technique exploiting an EWMA (Exponentially Weighed Moving Average)-based value estimator to clarify the influence of packet loss on the overall WNCS behavior. As an example of actuator to be remotely controlled, a rotary-type inverted pendulum has been considered, and modeled. Performance evaluation results through Matlab/Simulink and Truetime co-simulation confirm the superiority of the proposed value estimation method over previous approaches.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38B no.7
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• pp.553-563
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• 2013

When the IEEE 802.11 wireless LAN-based broadcasting services, we use broadcast packets to broadcast multimedia contents to a large number of users using limited wireless resources. However, broadcast transmission is difficult to recover the loss packets compared with unicast transmission. Therefore, analysis of packet loss characteristics is required to perform efficient packet recovery. The packet loss in wireless transmissions is often bursty with high loss data rate. Even if loss patterns have the same average packet loss, they are different in the recovery rate of random loss and burst loss depending on the nature. Therefore, the analysis and research of the nature of the loss are needed to recover loss packets considering bursty characteristics. In this paper, we experimented Wi-Fi broadcast transmission according to transmission rate and analyzed bursty characteristics of loss patterns using 4-state markov model.