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Efficient Audio Communication over Heterogeneous Packet Networks with Wireless Access

Author:
Frank Mertz
Editor:
Peter Vary
Type:
Dissertation
Series:
Aachener Beiträge zu Digitalen Nachrichtensystemen (ABDN)
Number:
28
School:
IND, RWTH Aachen
Publisher:
Verlag Mainz
Location:
Aachen, Germany
Date:
May 2011
ISBN:
3-861-30654-9
URL:
http://darwin.bth.rwth-aachen.de/opus3/volltexte/2...
Language:
English

Abstract

Communication networks are transforming towards all-IP networks with different fixed-lineand wireless access technologies (e.g., DSL, UMTS, WLAN). One of the biggest challengesin such heterogeneous packet networks for the realization of real-time services (e.g., Voiceover IP, music and video streaming) is dealing with transmission impairments which includevariable packet transmission delays and packet losses. Current systems utilize differenttechniques to combat these impairments, such as a Jitter Buffer to compensate the variancein transmission delay and reduce jitter based losses, Forward Error Correction (FEC) torecover lost frames, and Packet Loss Concealment (PLC) to estimate unrecoverable framelosses. An issue which has not yet been sufficiently addressed is how to choose the bestsuitable methods and parameterize them optimally for a given scenario of application andtransmission channel.

This dissertation develops a new methodology to analyze and determine the most suitableFEC method for a given transmission scenario. This approach aims at the best possibletransmission quality by considering both signal quality and end-to-end delay. The mainresults include:

- A new channel model for packet losses which is based on the Gilbert-Elliott modeland can be adapted for different packet sizes and transmission intervals;

- Analytical determination of residual frame loss probabilities after erasure correctionfor different FEC methods based upon the new channel model;

- Fair comparison of methods with different packet sizes and transmission intervalsgiven the adaptability of the new channel model;

- Derivation of optimal system parameters by applying the newly developedmethodologies to real-life scenarios of speech and music transmission in UMTS,WLAN, and heterogeneous packet networks.

Furthermore, new PLC techniques for state-of-art speech codecs are developed, which areparticularly suitable for wireless transmission channels. In particular, the estimation of lostcodec parameters is improved by transmission of low-rate side information in followingpackets. By utilizing steganographic methods, this side information can be transmitted ashidden bit stream in the encoded bits of the following frame, thereby requiring no additionaldata rate.

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