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packet-loss
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Performance analysis of packet loss recovery policies in P2P video streaming
, Article International Journal of Internet Protocol Technology ; Vol. 8, issue. 1 , 2014 , p. 44-53 ; Rabiee, H. R ; Ghanbari, M ; Sharif University of Technology
Abstract
Packet loss recovery is an important part of P2P video streaming networks due to inevitable packet loss in today's internet and interdependency of data units in compressed video streams. In addition, the architecture of P2P streaming networks, in which the data delivered to the receivers through chain of peers, can intensify the impact of the internet packet loss on the quality of perceived video at the receivers. FEC and ARQ are the two most important techniques that can be used to overcome the side effect of the internet packet loss in P2P video streaming networks. Based on these two techniques, different packet loss recovery strategies can be applied in different overlay hops of a given...
Packet loss recovery schemes for peer-to-peer video streaming
, Article 3rd International Conference on Networking and Services, ICNS 2007, Athens, 19 June 2007 through 25 June 2007 ; January , 2007 ; Rabiee, H. R ; Ghanbari, M ; Sharif University of Technology
2007
Abstract
One of the man concerns of peer-to-peer video streaming over the internet is the cumulative impact of the Internet packet loss due to the decoding dependency of the compressed video frames. In this paper we study the efficiency of various packet loss recovery policies that can be used in peer-to-peer streaming. Our analytical and simulation results show how different packet loss recovery policies can be effective for peer-to-peer video streaming. © 2007 IEEE
EECA: Energy efficient congestion avoidance in wireless multimedia sensor network
, Article 2012 6th International Symposium on Telecommunications, IST 2012, Tehran, 6 November 2012 through 8 November 2012 ; 2012 , Pages 656-661 ; 9781467320733 (ISBN) ; Heidari, V ; Khansari, M ; Sharif University of Technology
Abstract
The quality of service in Wireless Multimedia Sensor Networks (WMSN) is related to congestion in network. Recently different studies have been done on developing efficient protocols in the transport layer for controlling packet loss in WMSN. However, all of these protocols are independent of the characteristics of multimedia content. In this paper, a novel transport layer protocol, called Energy Efficient Congestion Avoidance (EECA), is proposed to minimize the packet loss ratio in WMSN by predicting congestion in network, and adjusting the source nodes output rate before congestion happening. EECA adjusts the output video quality in source nodes to adjust the source nodes video quality. The...
Packet loss in peer-to-peer video streaming over the Internet
, Article Multimedia Systems ; Volume 13, Issue 5-6 , 2008 , Pages 345-361 ; 09424962 (ISSN) ; Rabiee, H. R ; Ghanbari, M ; Sharif University of Technology
2008
Abstract
Peer-to-peer streaming has recently gained attention as an effective solution to support large scale media streaming applications over the Internet. One of the main challenges of peer-to-peer video streaming is the cumulative impact of the Internet packet loss due to the decoding dependency of the compressed video frames. In this paper we study the impact of the Internet packet loss on the performance of peer-to-peer video streaming systems, and analyze the efficiency of various packet loss recovery policies in such systems. Our analytical and simulation results show how the Internet packet loss can affect the performance of peer- to-peer video streaming systems and how different packet loss...
LPRE: Lost speech packet recovery withenhancement
, Article 2007 IEEE International Conference on Communications, ICC'07, Glasgow, Scotland, 24 June 2007 through 28 June 2007 ; August , 2007 , Pages 1778-1783 ; 05361486 (ISSN); 1424403537 (ISBN); 9781424403530 (ISBN) ; Manzuri Shalmani, M. T ; Sharif University of Technology
2007
Abstract
In the internet telephony, loss of IP packets causes instantaneous discontinuities in the received speech. In this paper, we have focused on finding an error resilient method for this problem. Our proposed method creates artificial correlation between speech samples that pre-distorts the speech signal. The receiver uses this correlation to reconstruct the lost speech packets. An appropriate speech enhancement technique is designed for the reduction of the processing error in the recovered speech caused by the speech codecs. The SegSNR results show the superiority of our proposed speech enhancement method over a recently proposed one. © 2007 IEEE
UDDP: A user datagram dispatcher protocol for wireless multimedia sensor networks
, Article 2012 IEEE Consumer Communications and Networking Conference, CCNC'2012 ; 2012 , Pages 765-770 ; 9781457720710 (ISBN) ; Khansari, M ; Rabiee, H. R ; Salehi, M ; Sharif University of Technology
2012
Abstract
The quality of service inWireless Multimedia Sensor Networks (WMSN) is related to packet loss rate. Recently different studies have been done on developing efficient protocols in the transport layer for controlling packet loss in WMSN. However, all of these protocols are independent of the characteristics of multimedia content. In this paper, a novel transport layer protocol, called User Datagram Dispatcher Protocol (UDDP), is proposed to minimize the packet loss ratio in WMSN by considering traffic characteristics, the inter-arrival pattern of packets and packet priority. UDDP is a new cross-layer transport layer that uses information of MAC and application layers to distribute packet...
Detecting malicious packet drops and misroutings using header space analysis
, Article 8th International Symposium on Telecommunications, IST 2016, 27 September 2016 through 29 September 2016 ; 2017 , Pages 521-526 ; 9781509034345 (ISBN) ; Kazemian, P ; Pakravan, M. R ; Sharif University of Technology
Institute of Electrical and Electronics Engineers Inc
2017
Abstract
Software Defined Networking (SDN) provides a logically centralized view of the state of the network, and as a result opens up new ways to manage and monitor networks. In this paper we introduce a novel approach to network intrusion detection in SDNs that takes advantage of these attributes. Our approach can detect compromised routers that produce faulty messages, copy or steal traffic or maliciously drop certain types of packets. To identify these attacks and the affected switches, we correlate the forwarding state of network - i.e. installed forwarding rules - with the forwarding status of packets - i.e. the actual route packets take in the network and detect anomaly in routes. Thus, our...
AMPCS: Adaptive model predictive control scheduler for guaranteed delay in DiffServ architecture
, Article International Journal of Communication Systems ; Volume 21, Issue 3 , 2008 , Pages 233-249 ; 10745351 (ISSN) ; Taheri, H ; Haeri, M ; Sharif University of Technology
2008
Abstract
An increasing number of different applications face the challenge of providing end-to-end quality of service (QoS) support such as bandwidth, delay, jitter, and packet loss. In this paper, we have focused on DiffServ architecture to improve its accuracy. We proposed a new algorithm, called Adaptive Model Predictive Control Scheduler (AMPCS), to schedule differentiated buffers in routers, using Adaptive Model Predictive Control as the controller. AMPCS regulates the service rates of aggregated traffic classes dynamically in a way that some constraints on proportional delay or absolute delay can be guaranteed. Our contribution is to apply a model predictive controller to the scheduling problem...
A mixed layer multiple description video coding scheme
, Article IEEE Transactions on Circuits and Systems for Video Technology ; Volume 22, Issue 2 , February , 2012 , Pages 202-215 ; 10518215 (ISSN) ; Sadeghi, K. H ; Shirmohammadi, S ; Sharif University of Technology
Abstract
Multiple description coding (MDC) is a technique where multiple streams from a source video are generated, each individually decodable and mutually refinable. MDC is a promising solution to overcome packet loss in video transmission over noisy channels, particularly for real-time applications in which retransmission of lost information is not practical. A problem with conventional MDC is that the achieved side distortion quality is considerably lower than single description coding (SDC) quality except at high redundancies which in turn leads to central quality degradation. In this paper, a new mixed layer MDC scheme is presented with no degradation in central quality, and providing better...
A Study of the Effects of H.264 Encoding Parameters on Video Streaming Quality Over Lossy Networks
, M.Sc. Thesis Sharif University of Technology ; Pakravan, Mohammad Reza (Supervisor) ; Shirmohammadi, Shervin (Co-Advisor)
Abstract
Due to the expansion of Data Networks and the ubiquity of high speed Internet, new services especially multimedia services have become more popular. Video streaming is one such service, but video data are quite bulky and need to be compressed for transmission since the network has limited bandwidth. This compression is achieved through a video encoder, such as H.264, which is currently the most advanced video encoder. In addition, the network introduces lag in the form of packet loss, delay and jitter which adversely affect real-time video applications.
In this thesis, we try to answer the following question in the context of a video streaming application: what encoding parameters should...
In this thesis, we try to answer the following question in the context of a video streaming application: what encoding parameters should...
Multiple Description Video Coding Based on Base and Enhancement Layers of SVC and Channel Adaptive Optimization
, Ph.D. Dissertation Sharif University of Technology ; Haj Sadghi, Khosrow (Supervisor) ; Shirmohammadi, Shervin (Supervisor)
Abstract
Multiple distortion coding is a promising solution for video transmission over lossy channels. In MDC multiple descriptions of a source are generated which are independently decodable and mutually refinable. When all descriptions are available, the corresponding quality is called central quality; otherwise it is called side quality. Generally, there exist a trade-off between side and central quality in all MDC schemes. The MDC methods which provide better central-side quality trade-off are of more interest to the designers.In this thesis a new MDC scheme is introduced which has better trade-off between side and central quality. In other words, for the same central quality, it provides higher...
Modeling and evaluating reliable real-time degree in multi-hop wireless sensor networks
, Article 2009 IEEE Sarnoff Symposium, SARNOFF 2009, Princeton, NJ, 30 March 2009 through 1 April 2009 ; 2009 ; 9781424433827 (ISBN) ; Yousefi, H ; Jahangir, A. H ; IEEE ; Sharif University of Technology
2009
Abstract
Wireless Sensor Network (WSN) should be capable of fulfilling its mission, in a timely manner and without loss of important information. In this paper, we propose a new analytical model for calculating RRT (Reliable Real-Time) degree in multihop WSNs, where RRT degree describes the percentage of real-time data that the network can reliably deliver on time from any source to its destination. Also, packet loss probability is modeled as a function of the probability of link failure when the buffer is full and the probability of node failure when node's energy is depleted. Most of network properties are considered as random variables and a queuing-theory based model is derived. In this model,...
Cross-layer optimization of adaptive modulation and coding preserving packet average delay time
, Article 2008 IEEE Global Telecommunications Conference, GLOBECOM 2008, New Orleans, LA, 30 November 2008 through 4 December 2008 ; December , 2008 , Pages 1278-1282 ; 9781424423248 (ISBN) ; Ashtiani, F ; Sharif University of Technology
2008
Abstract
In this paper we introduce a novel cross layer design between MAC and physical layers. Our novel design is based on controlling the adaptive modulation and coding (AMC) transmission mode at the physical layer according to the queue length of finite buffer at the data link layer and each transmission mode of AMC. The aim is optimizing system performance over a wireless link, preserving the important Quality of Service (QoS) parameter, packet average delay time. Analytical expressions such as packet drop probability, channel packet error rate, packet loss rate and packet average delay time are derived both analytically and through simulations according to the system and channel parameters....
A packet-based photonic label switching router for a multirate all-optical CDMA-based GMPLS switch
, Article IEEE Journal on Selected Topics in Quantum Electronics ; Volume 13, Issue 5 , 2007 , Pages 1522-1530 ; 1077260X (ISSN) ; Ibrahirni, M ; Salehi, J. A ; Sharif University of Technology
2007
Abstract
A novel packet-based photonic label switching router for a multirate all-optical switch using generalized multiprotocol label switching is proposed. The idea is based on using optical codedivision multiple access (OCDMA) as multiplexing technique and treating OCDMA codes as labels. The system can coexist with current wavelength division multiplexing systems on the same infrastructure. The concept of switch fabric is introduced. Label processing and label swapping functionalities of the switch are discussed. In-depth analyses are made for spectrally phase-encoded OCDMA (SPE-OCDMA) due to its capabilities of supporting high data rates, large code cardinality, and its secure transmission....
A new cross layer design of adaptive modulation and coding in finite buffer wireless links
, Article 2007 International Conference on Future Generation Communication and Networking, FGCN 2007, Jeju Island, 6 December 2007 through 8 December 2007 ; Volume 1 , 2007 , Pages 499-504 ; 0769530486 (ISBN); 9780769530482 (ISBN) ; Ashtiani, F ; Sharif University of Technology
Institute of Electrical and Electronics Engineers Inc
2007
Abstract
In this paper, a cross-layer approach introducing new design parameters is developed for multi-rate wireless links. The novel design jointly exploits the finite-length queuing at the data link layer and the adaptation capability of the adaptive modulation and coding (AMC) scheme at the physical layer to optimize system performance for a wireless link. The analytical framework is based on discrete time Markov Chain. The performance metrics such as packet drop probability, channel packet error rate and packet loss rate are derived. Using these metrics, a constrained optimization problem is solved numerically to maximize the overall system throughput
Adaptive online prediction of operator position in teleoperation with unknown time-varying delay: simulation and experiments
, Article Neural Computing and Applications ; Volume 33, Issue 13 , 2021 , Pages 7575-7592 ; 09410643 (ISSN) ; Yazdankhoo, B ; Beigzadeh, B ; Meghdari, A ; Sharif University of Technology
Springer Science and Business Media Deutschland GmbH
2021
Abstract
One of the most important problems in teleoperation systems is time delay and packet loss in the communication channel, which can affect transparency and stability. One way to overcome the time delay effects in a teleoperation system is to predict the master-side motion. In this way, when data is received in the slave side, it will be considered as the current position of the master robot and, thus, complete transparency could be achieved. The majority of the previous works regarding operator position prediction have considered known and constant time delay in the system; however, in the real applications, time delay is unknown and variable. In this paper, an adaptive online prediction...
Packet Loss Replacement in VOIP Using Linear Prediction Method
, M.Sc. Thesis Sharif University of Technology ; Ghorshi, Mohammad Ali (Supervisor) ; Mortazavi, Mohammad (Supervisor)
Abstract
In real-time packet-based communication systems one major problem is misrouted or delayed packets which result in degraded perceived voice quality. If some speech packets are not available on time, the packet is known as lost packet. The easiest task of a network terminal receiver is to replace silence for the duration of lost speech segments. In a high quality communication system in order to avoid quality reduction due to packet loss a suitable method and/or algorithm is needed to replace the missing segments of speech.There are several methods which have been proposed to reduce the effect of packet loss such as Waveform Substitution, High Order Autoregressive, Linear Prediction (LP),...
Kalman filter based packet loss replacement in presence of additive noise
, Article 2012 25th IEEE Canadian Conference on Electrical and Computer Engineering: Vision for a Greener Future, CCECE 2012 ; 2012 ; 9781467314336 (ISBN) ; Ghorshi, S ; Tahaei, A
2012
Abstract
A major problem in real-time packet-based communication systems, is misrouted or delayed packets which results in degraded perceived voice quality. If packets are not available on time, the packets are considered as lost. The easiest solution in a network terminal receiver is to replace silence for the duration of lost speech segments. In a high quality communication system, to avoid degradation in speech quality due to packet loss, a suitable method or algorithm is needed to replace the missing segments of speech. In this paper, we introduce an adaptive filter for replacement of lost speech segment. In this method Kalman filter as a state-space based method will be used to predict the...
A Kalman filter approach to packet loss replacement in presence of additive noise
, Article 2012 11th International Conference on Information Science, Signal Processing and their Applications, ISSPA 2012 ; 2012 , Pages 310-314 ; 9781467303828 (ISBN) ; Ghorshi, S ; Tahaei, A ; Sharif University of Technology
2012
Abstract
A major problem in real-time packet-based communication systems, is misrouted or delayed packet which results in degraded perceived voice quality. If packets are not available on time, the packets are considered lost. The easiest solution in a network terminal receiver is to replace silence for the duration of lost speech segments. In a high quality communication system, to avoid degradation in speech quality due to packet loss, a suitable method or algorithm is needed to replace the missing segments of speech. In this paper, we introduce an adaptive filter for replacement of lost speech segment. In this method Kalman filter as a state-space based method will be used to predict the clean...
Kalman filter method for packet loss replacement in presence of background noise
, Article International Multi-Conference on Systems, Signals and Devices, SSD 2012 - Summary Proceedings, 20 March 2012 through 23 March 2012 ; March , 2012 ; 9781467315906 (ISBN) ; Ghorshi, S ; Tahaei, A ; Rahimi, A ; Sharif University of Technology
2012
Abstract
A major problem in real-time packet-based communication systems, is misrouted or delayed packets which results in degraded perceived voice quality. If packets are not available on time, the packets are considered as lost. The easiest solution in a network terminal receiver is to replace silence for the duration of lost speech segments. In a high quality communication system, to avoid degradation in speech quality due to packet loss, a suitable method or algorithm is needed to replace the missing segments of speech. In this paper, we introduce an adaptive filter for replacement of lost speech segment. In this method Kalman filter as a state-space based method will be used to predict the clean...