1,287 research outputs found
Speech quality prediction for voice over Internet protocol networks
Merged with duplicate record 10026.1/878 on 03.01.2017 by CS (TIS). Merged with duplicate record 10026.1/1657 on 15.03.2017 by CS (TIS)This is a digitised version of a thesis that was deposited in the University Library. If you are the author please contact PEARL Admin ([email protected]) to discuss options.IP networks are on a steep slope of innovation that will make them the long-term carrier
of all types of traffic, including voice. However, such networks are not designed to support
real-time voice communication because their variable characteristics (e.g. due to delay, delay
variation and packet loss) lead to a deterioration in voice quality. A major challenge in such networks
is how to measure or predict voice quality accurately and efficiently for QoS monitoring
and/or control purposes to ensure that technical and commercial requirements are met.
Voice quality can be measured using either subjective or objective methods. Subjective
measurement (e.g. MOS) is the benchmark for objective methods, but it is slow, time consuming
and expensive. Objective measurement can be intrusive or non-intrusive. Intrusive methods
(e.g. ITU PESQ) are more accurate, but normally are unsuitable for monitoring live traffic
because of the need for a reference data and to utilise the network. This makes non-intrusive
methods(e.g. ITU E-model) more attractive for monitoring voice quality from IP network impairments.
However, current non-intrusive methods rely on subjective tests to derive model
parameters and as a result are limited and do not meet new and emerging applications.
The main goal of the project is to develop novel and efficient models for non-intrusive
speech quality prediction to overcome the disadvantages of current subjective-based methods
and to demonstrate their usefulness in new and emerging VoIP applications. The main contributions
of the thesis are fourfold:
(1) a detailed understanding of the relationships between voice quality, IP network impairments
(e.g. packet loss, jitter and delay) and relevant parameters associated with speech (e.g.
codec type, gender and language) is provided. An understanding of the perceptual effects of
these key parameters on voice quality is important as it provides a basis for the development
of non-intrusive voice quality prediction models. A fundamental investigation of the impact of
the parameters on perceived voice quality was carried out using the latest ITU algorithm for
perceptual evaluation of speech quality, PESQ, and by exploiting the ITU E-model to obtain an
objective measure of voice quality.
(2) a new methodology to predict voice quality non-intrusively was developed. The method
exploits the intrusive algorithm, PESQ, and a combined PESQ/E-model structure to provide a
perceptually accurate prediction of both listening and conversational voice quality non-intrusively.
This avoids time-consuming subjective tests and so removes one of the major obstacles in the
development of models for voice quality prediction. The method is generic and as such has
wide applicability in multimedia applications. Efficient regression-based models and robust
artificial neural network-based learning models were developed for predicting voice quality
non-intrusively for VoIP applications.
(3) three applications of the new models were investigated: voice quality monitoring/prediction
for real Internet VoIP traces, perceived quality driven playout buffer optimization and
perceived quality driven QoS control. The neural network and regression models were both
used to predict voice quality for real Internet VoIP traces based on international links. A new
adaptive playout buffer and a perceptual optimization playout buffer algorithms are presented.
A QoS control scheme that combines the strengths of rate-adaptive and priority marking control
schemes to provide a superior QoS control in terms of measured perceived voice quality is
also provided.
(4) a new methodology for Internet-based subjective speech quality measurement which
allows rapid assessment of voice quality for VoIP applications is proposed and assessed using
both objective and traditional MOS test methods
Quality of media traffic over Lossy internet protocol networks: Measurement and improvement.
Voice over Internet Protocol (VoIP) is an active area of research in the world of
communication. The high revenue made by the telecommunication companies is a
motivation to develop solutions that transmit voice over other media rather than
the traditional, circuit switching network.
However, while IP networks can carry data traffic very well due to their besteffort
nature, they are not designed to carry real-time applications such as voice.
As such several degradations can happen to the speech signal before it reaches its
destination. Therefore, it is important for legal, commercial, and technical reasons
to measure the quality of VoIP applications accurately and non-intrusively.
Several methods were proposed to measure the speech quality: some of these
methods are subjective, others are intrusive-based while others are non-intrusive.
One of the non-intrusive methods for measuring the speech quality is the E-model
standardised by the International Telecommunication Union-Telecommunication Standardisation
Sector (ITU-T).
Although the E-model is a non-intrusive method for measuring the speech quality,
but it depends on the time-consuming, expensive and hard to conduct subjective
tests to calibrate its parameters, consequently it is applicable to a limited number
of conditions and speech coders. Also, it is less accurate than the intrusive methods
such as Perceptual Evaluation of Speech Quality (PESQ) because it does not consider
the contents of the received signal.
In this thesis an approach to extend the E-model based on PESQ is proposed.
Using this method the E-model can be extended to new network conditions and
applied to new speech coders without the need for the subjective tests. The modified
E-model calibrated using PESQ is compared with the E-model calibrated using
i
ii
subjective tests to prove its effectiveness.
During the above extension the relation between quality estimation using the
E-model and PESQ is investigated and a correction formula is proposed to correct
the deviation in speech quality estimation.
Another extension to the E-model to improve its accuracy in comparison with
the PESQ looks into the content of the degraded signal and classifies packet loss
into either Voiced or Unvoiced based on the received surrounding packets. The accuracy
of the proposed method is evaluated by comparing the estimation of the new
method that takes packet class into consideration with the measurement provided
by PESQ as a more accurate, intrusive method for measuring the speech quality.
The above two extensions for quality estimation of the E-model are combined
to offer a method for estimating the quality of VoIP applications accurately, nonintrusively
without the need for the time-consuming, expensive, and hard to conduct
subjective tests.
Finally, the applicability of the E-model or the modified E-model in measuring
the quality of services in Service Oriented Computing (SOC) is illustrated
Intrusion detection mechanisms for VoIP applications
VoIP applications are emerging today as an important component in business
and communication industry. In this paper, we address the intrusion detection
and prevention in VoIP networks and describe how a conceptual solution based on
the Bayes inference approach can be used to reinforce the existent security
mechanisms. Our approach is based on network monitoring and analyzing of the
VoIP-specific traffic. We give a detailed example on attack detection using the
SIP signaling protocol
Systemization of Pluggable Transports for Censorship Resistance
An increasing number of countries implement Internet censorship at different
scales and for a variety of reasons. In particular, the link between the
censored client and entry point to the uncensored network is a frequent target
of censorship due to the ease with which a nation-state censor can control it.
A number of censorship resistance systems have been developed thus far to help
circumvent blocking on this link, which we refer to as link circumvention
systems (LCs). The variety and profusion of attack vectors available to a
censor has led to an arms race, leading to a dramatic speed of evolution of
LCs. Despite their inherent complexity and the breadth of work in this area,
there is no systematic way to evaluate link circumvention systems and compare
them against each other. In this paper, we (i) sketch an attack model to
comprehensively explore a censor's capabilities, (ii) present an abstract model
of a LC, a system that helps a censored client communicate with a server over
the Internet while resisting censorship, (iii) describe an evaluation stack
that underscores a layered approach to evaluate LCs, and (iv) systemize and
evaluate existing censorship resistance systems that provide link
circumvention. We highlight open challenges in the evaluation and development
of LCs and discuss possible mitigations.Comment: Content from this paper was published in Proceedings on Privacy
Enhancing Technologies (PoPETS), Volume 2016, Issue 4 (July 2016) as "SoK:
Making Sense of Censorship Resistance Systems" by Sheharbano Khattak, Tariq
Elahi, Laurent Simon, Colleen M. Swanson, Steven J. Murdoch and Ian Goldberg
(DOI 10.1515/popets-2016-0028
Enhancement of perceived quality of service for voice over internet protocol systems
Voice over Internet Protocol (WIP) applications are becoming more and more popular in
the telecommunication market. Packet switched V61P systems have many technical advantages
over conventional Public Switched Telephone Network (PSTN), including its efficient and flexible
use of the bandwidth, lower cost and enhanced security.
However, due to the IP network's "Best Effort" nature, voice quality are not naturally guaranteed
in the VoIP services. In fact, most current Vol]P services can not provide as good a voice
quality as PSTN. IP Network impairments such as packet loss, delay and jitter affect perceived
speech quality as do application layer impairment factors, such as codec rate and audio features.
Current perceived Quality of Service (QoS) methods are mainly designed to be used
in a PSTN/TDM environment and their performance in V6IP environment is unknown. It is a
challenge to measure perceived speech quality correctly in V61P system and to enhance user
perceived speech quality for VoIP system.
The main goal of this project is to evaluate the accuracy of the existing ITU-T speech quality
measurement method (Perceptual Evaluation of Speech Quality - PESQ) in mobile wireless
systems in the context of V61P, and to develop novel and efficient methods to enhance the user
perceived speech quality for emerging V61P services especially in mobile V61P environment.
The main contributions of the thesis are threefold:
(1) A new discovery of PESQ errors in mobile VoIP environment. A detailed investigation
of PESQ performance in mobile VoIP environment was undertaken and included setting up a
PESQ performance evaluation platform and testing over 1800 mobile-to-mobile and mobileto-
PSTN calls over a period of three months. The accuracy issues of PESQ algorithm was
investigated and main problems causing inaccurate PESQ score (improper time-alignment in
the PESQ algorithm) were discovered
.
Calibration issues for a safe and proper PESQ testing
in mobile environment were also discussed in the thesis.
(2) A new, simple-to-use, V611Pjit ter buffer algorithm. This was developed and implemented
in a commercial mobile handset. The algorithm, called "Play Late Algorithm", adaptively alters
the playout delay inside a speech talkspurt without introducing unnecessary extra end-to-end
delay. It can be used as a front-end to conventional static or adaptive jitter buffer algorithms
to provide improved performance. Results show that the proposed algorithm can increase user
perceived quality without consuming too much processing power when tested in live wireless
VbIP networks.
(3) A new QoS enhancement scheme. The new scheme combines the strengths of adaptive
codec bit rate (i. e. AMR 8-modes bit rate) and speech priority marking (i. e. giving high priority
for the beginning of a voiced segment). The results gathered on a simulation and emulation test
platform shows that the combined method provides a better user perceived speech quality than
separate adaptive sender bit rate or packet priority marking methods
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