• Journal of Internet Computing and Services
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• v.9 no.4
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• pp.1-10
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• 2008

In this paper, a packet scheduling algorithm, called the modified PLFS, is proposed for real-time traffic in IEEE 802.22 WRAN systems. The modified PLFS(Packet Loss Fair Scheduling) algorithm utilizes not only the delay of the Head of Line(HOL) packets in buffer of each user but also the amount of expected loss packets in the next-next frame when a service will not be given in the next frame. The performances of the modified PLFS are compared with those of PLFS and M-LWDF in terms of the average packet loss rate and throughput. The simulation results show that the proposed scheduling algorithm performs much better than the PLFS and M-LWDF algorithms.

• Journal of Digital Contents Society
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• v.10 no.2
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• pp.279-289
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• 2009

As the TCP is the protocol designed for the wired network that packet loss probability is very low, because TCP transmitter takes it for granted that the packet loss by the wireless network characteristics is occurred by the network congestion and lowers the transmitter's transmission rate, the performance is degraded. In this article, we suggest the newly improved algorithm using two parameters, the local retransmission time value and the local retransmission critical value to the BS based on the Snoop. This technique adjusts the base stations local retransmission timer effectively according to the wireless link status to recover the wireless packet loss rapidly. We checked that as a result of the suggested algorithm through various simulations, A-Snoop protocol improve more the wireless TCP transmission rate by recovering the packet loss effectively in the wireless link that the continuous packet loss occur than the Snoop protocol.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.7
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• pp.1-5
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• 1998

In this paper, recovery method with high-bitrate and low-bitrate coder was implemented in order to recover consecutive packet loss over the Internet. LPC was used as redundant audio data for recover of lost packets and RTP parcket format was modified for accommodation of redundant data. In measuring results using random packet loss rate with three redundant datra in every packet, it has shown that recovery rate was 80% in los rate of 50%. Since the processing delay for recovery of the lost packet was 200ms, this recovery method can be applied to real-time Internet sevice such as Internet phone.

• Proceedings of the KIEE Conference
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• 2006.04a
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• pp.105-107
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• 2006

As internet streaming data increase, transport protocol such as TCP, TGP-Friendly is important to study control transmission rate and share of Internet bandwidth. In this paper, we propose a TCP-Friendly protocol using Neural Network for media delivery over wired Internet which has various traffic size(PTFRC). PTFRC can effectively send streaming data when occur congestion and predict one-step ahead round trip time and packet loss rate. A multi-layer perceptron structure is used as the prediction model, and the Levenberg-Marquardt algorithm is used as a traning algorithm. The performance of the PTFRC was evaluated by the share of Bandwidth and packet loss rate with various protocols.

• Proceedings of the Korea Multimedia Society Conference
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• 2012.05a
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• pp.166-167
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• 2012

High packet loss rate and impatience of messages passing through optical switches are essential characteristics in Optical Burst Switching system equipped with Optic Delay Lines buffer, which have not been solved efficiently yet by current existing models. In order to capture both effects, this paper introduces an analytical model from the viewpoint of classical queuing theory with impatient customers. We then apply it to evaluate and compare two wavelength-sharing cases, (1) all delay lines share a common wavelength resource and (2) each wavelength is associated with a number of delay lines. Our numerical results suggest to implement the first case because of lower packet loss rate for a fairly broad range of traffic load.

• 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 Communications and Networks
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• v.16 no.3
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• pp.322-334
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• 2014

Multimedia content is very sensitive to packet loss and therefore multimedia streams are typically protected against packet loss, either by supporting retransmission requests or by adding redundant forward error correction (FEC) data. However, the redundant FEC information introduces significant additional bandwidth requirements, as compared to the bitrate of the original video stream. Especially on wireless and mobile networks, bandwidth availability is limited and variable. In this article, an adaptive FEC (A-FEC) system is presented whereby the redundancy rate is dynamically adjusted to the packet loss, based on feedback messages from the client. We present a statistical model of our A-FEC system and validate the proposed system under different packet loss conditions and loss probabilities. The experimental results show that 57-95%bandwidth gain can be achieved compared with a static FEC approach.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.7
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• pp.3150-3171
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• 2018

In this paper, a heuristic method that leverages packet traces captured at the entire boarder of an ISP to distinguish and estimate the overall packet loss within an ISP's management domain (Intra_Path_Loss) and that in the outside Internet (Inter_Path_Loss) is proposed. Our method is inspired by that packet losses happened at different locations will cause different TCP sequence number patterns at the border of an ISP. Thereby, we leverage these TCP sequence number patterns to build a series of heuristic rules to estimate Intra_Path_Loss and Inter_Path_Loss, respectively. We do this work with an eye towards showing that the overall packet losses defined and estimated in this paper can provide the operators with some valuable information to help them precisely grasp the overall performance of network paths and narrow down the range of network anomalies. The proposed method is rigorously validated with simulations, and finally the results from a regional academic network JSERNET verify its effectiveness and practicability.

• 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.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.11
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• pp.63-71
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• 2003

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. Adaptive Multi Rate Wideband(AMR-WB) codec which is recently selected by the Third Generation Partnership Project(3GPP) for GSM and the third generation mobile communication WCDMA system and has also been standardized in ITU-T for providing wideband speech services is used. The major cause for speech qualitly degradation in IP-networks is packet loss. So, We recovered single lossy packet by using FEC method and concealed continued errors. The proposed scheme if evaluated in the Gilbert Internet channel model. The high quality of audio maintained up to 30% packet loss.