Persistent Link:
http://hdl.handle.net/10150/193858
Title:
QoS Improvement Schemes for Real-Time Wireless VoIP
Author:
Liu, Mingkuan
Issue Date:
2006
Publisher:
The University of Arizona.
Rights:
Copyright © is held by the author. Digital access to this material is made possible by the University Libraries, University of Arizona. Further transmission, reproduction or presentation (such as public display or performance) of protected items is prohibited except with permission of the author.
Abstract:
There is a tremendous demand on real-time multimedia delivery over wireless Internet due to the dramatic increase in wireless communication and the growth of the Internet. However, real-time multimedia over wireless Internet poses many challenges. First of all, the inherent best-effort characteristic of packet-switched networks makes it difficult to provide guaranteed QoS for real-time multimedia delivery. Secondly, wireless channels have much higher packet-loss rate, bit-error rate, and channel instability compared to wired channels due to noise, path loss, multi-path fading and shadowing, which result in fluctuating communication channel statistics. Thirdly, the real-time communication demands strict time limitations on the network end-to-end delay and delay jitter.In this dissertation, an intelligent application architecture and several QoS improvement mechanisms are proposed to timely estimate the current wireless network statistics and dynamically take smart actions to improve the overall performance of a real-time wireless Internet telephony system. An online network traffic modeling method based on time series analysis was used to estimate the dynamic wireless network statistics such as end-to-end packet delay and delay jitters. Using this real-time updated information, the application's sender side can take some adaptive actions such as voice codec selection and forward error-correction schemes for packet-loss concealment to improve the QoS under current available network resources. Also, a novel adaptive playout jitter buffer adjustment algorithm is proposed. The proposed algorithm achieved 11%-15% performance improvement compared to traditional adaptive playout adjustment algorithms using the ITU-E model measurement metric.
Type:
text; Electronic Dissertation
Degree Name:
PhD
Degree Level:
doctoral
Degree Program:
Electrical & Computer Engineering; Graduate College
Degree Grantor:
University of Arizona
Advisor:
Rodriguez, Jeffrey J.; McNeill, Kevin M.
Committee Chair:
Rodriguez, Jeffrey J.; McNeill, Kevin M.

Full metadata record

DC FieldValue Language
dc.language.isoENen_US
dc.titleQoS Improvement Schemes for Real-Time Wireless VoIPen_US
dc.creatorLiu, Mingkuanen_US
dc.contributor.authorLiu, Mingkuanen_US
dc.date.issued2006en_US
dc.publisherThe University of Arizona.en_US
dc.rightsCopyright © is held by the author. Digital access to this material is made possible by the University Libraries, University of Arizona. Further transmission, reproduction or presentation (such as public display or performance) of protected items is prohibited except with permission of the author.en_US
dc.description.abstractThere is a tremendous demand on real-time multimedia delivery over wireless Internet due to the dramatic increase in wireless communication and the growth of the Internet. However, real-time multimedia over wireless Internet poses many challenges. First of all, the inherent best-effort characteristic of packet-switched networks makes it difficult to provide guaranteed QoS for real-time multimedia delivery. Secondly, wireless channels have much higher packet-loss rate, bit-error rate, and channel instability compared to wired channels due to noise, path loss, multi-path fading and shadowing, which result in fluctuating communication channel statistics. Thirdly, the real-time communication demands strict time limitations on the network end-to-end delay and delay jitter.In this dissertation, an intelligent application architecture and several QoS improvement mechanisms are proposed to timely estimate the current wireless network statistics and dynamically take smart actions to improve the overall performance of a real-time wireless Internet telephony system. An online network traffic modeling method based on time series analysis was used to estimate the dynamic wireless network statistics such as end-to-end packet delay and delay jitters. Using this real-time updated information, the application's sender side can take some adaptive actions such as voice codec selection and forward error-correction schemes for packet-loss concealment to improve the QoS under current available network resources. Also, a novel adaptive playout jitter buffer adjustment algorithm is proposed. The proposed algorithm achieved 11%-15% performance improvement compared to traditional adaptive playout adjustment algorithms using the ITU-E model measurement metric.en_US
dc.typetexten_US
dc.typeElectronic Dissertationen_US
thesis.degree.namePhDen_US
thesis.degree.leveldoctoralen_US
thesis.degree.disciplineElectrical & Computer Engineeringen_US
thesis.degree.disciplineGraduate Collegeen_US
thesis.degree.grantorUniversity of Arizonaen_US
dc.contributor.advisorRodriguez, Jeffrey J.en_US
dc.contributor.advisorMcNeill, Kevin M.en_US
dc.contributor.chairRodriguez, Jeffrey J.en_US
dc.contributor.chairMcNeill, Kevin M.en_US
dc.contributor.committeememberHariri, Salimen_US
dc.identifier.proquest1981en_US
dc.identifier.oclc659746553en_US
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