• Journal of Korea Multimedia Society
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• v.20 no.7
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• pp.1030-1037
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• 2017

Traffic flows of real-time multimedia services such as Internet phone and IPTV are bounded on the end-to-end delay. Packets violating their delay limits will be dropped at a router because of not useful anymore. Service providers promise the quality of their providing services in terms of SLA(Service Level Agreement), and they, especially, have to guarantee the packet loss rates listed in the SLA. This paper is about a method to guarantee the required packet loss rate of each traffic flow keeping the high network resource utilization as well. In details, it assures the required loss rate by adjusting adaptively the timestamps of packets of the flow according to the difference between the required and measured loss rates in the lossy Weighted Fair Queuing(WFQ) scheduler. The proposed method is expected to be highly applicable because of assuring the packet loss rates regardless of the fluctuations of offered traffic load in terms of quality of services and statistical characteristics.

• Proceedings of the Korea Society of Information Technology Applications Conference
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• 2005.11a
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• pp.61-64
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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.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2000.11b
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• pp.15-19
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• 2000

This paper is about MPEG-4 error resilience tools of video streaming on packet service(ex, Internet). It is need to packetization for MPEG-4 video transport by packet unit on MPEG-4 system, this paper suggest packetization method of minimizing packet error on packet service[1]. FEC(Forward Error Correction) and retransmission is usually used for recovery of packet loss, and this paper suggest applying these method to DMIF(Delivery Multimedia Integration Framework) for minimizing packet loss[2].

• The Journal of the Acoustical Society of Korea
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• v.29 no.5
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• pp.347-353
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• 2010

In VoIP system, the speech quality is seriously degraded due to packet loss, and the degree of degradation by each packet loss depends on the characteristics of the corresponding packet. Therefore, it is possible to improve the speech quality of VoIP by selectively controlling the packet to be lost during transmission based on the expected degradation by the loss of each packet. In this paper, a new scheme to improve speech quality of DiffServ-based VoIP by assigning priority to each packet is proposed, and a method to determine the priority of each packet is developed. The performance of proposed method was measured in packet loss environment based on Gilbert model, and it was verified both objectively and subjectively that the speech quality is improved by the proposed method.

• The Journal of the Acoustical Society of Korea
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• v.33 no.1
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• pp.75-80
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• 2014

The quality of real-time Voice over Internet Protocol (VoIP) network is affected by network impariments such as delays, jitters, and packet loss. This paper proposes a packet loss concealment algorithm based on voice classification for enhancing VoIP speech quality. In the proposed method, arriving packets are classified by an adaptive thresholding approach based on the analysis of multiple features of short signal segments. The excellent classification results are used in the packet loss concealment. Additionally, linear prediction-based packet loss concealment delivers high voice quality by alleviating the metallic artifacts due to concealing consecutive packet loss or recovering lost packet.

• Journal of Korea Multimedia Society
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• v.17 no.12
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• pp.1453-1460
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• 2014

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 the condition of no packet loss, and the WFQ scheduler guarantees those QoS requirements with the allocated resource. In practice, however, most QoS-guaranteed services allow a degree of packet loss, especially from 0.1% to 3% for Voice over IP. This paper discovers that the packet loss rate of each traffic flow is determined by only its time-stamp adjustment value, and then enhances the WFQ to provide a differentiated packet loss guarantee under general traffic conditions in terms of both traffic characteristics and QoS requirements. The performance evaluation showed that the proposed WFQ could increase the utilization of bandwidth by 8~11%.

• Journal of Communications and Networks
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• v.6 no.3
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• pp.235-244
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• 2004

There have been a lot of approaches to evaluate and predict transmission control protocol (TCP) performance in a numerical way. Especially, under the recent advance in wireless transmission technology, the issue of TCP performance over wireless links has come to surface. It is because TCP responds to all packet losses by invoking congestion control and avoidance algorithms, resulting in degraded end-to-end performance in wireless and lossy systems. By several previous works, although it has been already proved that overall TCP performance is largely dependent on its loss recovery performance, there have been few works to try to analyze TCP loss recovery performance with thoroughness. In this paper, therefore, we focus on analyzing TCP's loss recovery performance and have developed a simple model that facilitates to capture the TCP sender's behaviors during loss recovery period. Based on the developed model, we can derive the conditions that packet losses may be recovered without retransmission timeout (RTO). Especially, we have found that TCP Reno can retransmit three packet losses by fast retransmits in a specific situation. In addition, we have proved that successive three packet losses and more than four packet losses in a window always invoke RTO easily, which is not considered or approximated in the previous works. Through probabilistic works with the conditions derived, the loss recovery performance of TCP Reno can be quantified in terms of the number of packet losses in a window.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.10B
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• pp.1824-1831
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• 1999

In this paper, the data recovery performance of systematic erasure codes in burst loss environments is analyzed and the estimation method of redundant data according to loss characteristics is suggested. The burstness of packet loss is modeled by Gilbert model, and the performance of proposed packet loss recovery method in the case of using systematic erasure code is analyzed based on previous study on the loss recovery in the case of using erasure code. The required redundancy data fitting method for systematic erasure code in the condition of given loss property is suggested in the consideration of packet loss characteristics such as average packet loss rate and average loss length.

• ETRI Journal
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• v.26 no.5
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• pp.391-396
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• 2004

In this paper, we propose a packet scheduling discipline called packet loss fair scheduling, in which the packet loss of each user from different real-time traffic is fairly distributed according to the quality of service requirements. We consider an orthogonal frequency division multiple access (OFDMA) system. The basic frame structure of the system is for the downlink in a cellular packet network, where the time axis is divided into a finite number of slots within a frame, and the frequency axis is segmented into subchannels that consist of multiple subcarriers. In addition, to compensate for fast and slow channel variation, we employ a link adaptation technique such as adaptive modulation and coding. From the simulation results, our proposed packet scheduling scheme can support QoS differentiations while guaranteeing short-term fairness as well as long-term fairness for various real-time traffic.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.353-354
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• 2006

This paper suggests the packet loss recovery method to communicate in real-time in the Internet. To reduce the effects of packet loss, Forward Error Correction(FEC) that adds redundant information to voice packets can be used. The major cause for speech quality degradation in IP-networks is packet loss. So, We recovered single lossy packet by using FEC method and concealed continued errors. The proposed scheme is evaluated in the Gilbert Internet channel model. The high quality of audio maintained up to 30% packet loss.