PERFORMANCE ANALYSIS AND PLAYOUT TIME ESTIMATION FOR MULTIMEDIA OVER INTERNET PROTOCOL (MOIP)

This thesis presents the algorithms to estimate the minimum buffering delay and playout delay. The possibility to use the buffering delay estimation in Multimedia Application at the receiver site will reduce the effect of jitter and will also optimize the packet loss. The increasing of Traffic of loaded generator will influence the network behaviour and affect to transmission data using video conferencing netmeeting application over the network connectio

A Statistical Approach to Adaptive Playout Scheduling in Voice Over Internet Protocol Communication

Factors like network delay, latency and bandwidth significantly affect the quality of communication using Voice over Internet Protocol. The use of jitter buffer at the receiving end compensates the effect of varying network delay up to some extent. But the extra buffer delay given for each packet plays a major role in playing late packets and thereby improving voice quality. As the buffer delay increases packet loss rate decreases, which in general is a very good sign. However, an increase of buffer delay beyond a certain limit affects the interactive quality of voice communication. In this paper, we propose a statistical framework for adaptive playout scheduling of voice packets based on network statistics, packet loss rate and availability of packets in the buffer. Experimental results show that the proposed model allocates optimal buffer delay with the lowest packet loss rate when compared with other algorithms

Detailed comparative analysis of PESQ and VISQOL behaviour in the context of playout delay adjustments introduced by VOIP jitter buffer algorithms

The default best-effort Internet presents significant challenges for delay-sensitive applications such as VoIP. To cope with non determinism, receiver playout strategies are utilised in VoIP applications that adapt to network condition. Such strategies can be divided into two different groups, namely per-talkspurt and per-packet. The former make use of silence periods within natural speech and adapt such silences to track network conditions, thus preserving the integrity of active speech talkspurts. Examples of this approach are described in [1, 2]. Per packet strategies are different in that adjustments are made both during silence periods and during talkspurts by time-scaling of packets, a technique also known in the literature as time-warping. This approach is more effective in coping with short network delay changes because the per talkspurt approach can only adapt during recognized silences even though the duration of many delay spikes may be less than that of a talkspurt. This approach however introduces potential degradation caused by the scaling of speech packets. Examples of this approach are described in [3, 4] and such techniques are frequently deployed in popular VoIP applications such as GoogleTalk and Skype. In this research, we focus on applications that deploy per talkspurt strategies, which are commonly found in current telecommunication networks

Implementation of Playout Algorithm to Streaming Media File for Reducing Packet Loss Over Internet Protocol

PlayOut Algorithm has considerable contributed to the interactive communication over Internet like voice communication and VoIP. Some papers have, including playout adaptive Algorithm, investigated how to reduce and minimize the effect of Jitter which are much constitute to the packet loss. In streaming media file using H.323 format for video has been investigated in this paper. By using Network Analyzer for various network condition has represented network behaviour over the network VLAN. The use of playout Algorithm to estimate the buffering delay time for scheduling each packet arrived in receiver has remained the reduction of packet loss under 5% for traffic load utilization generator up to 21% using Network Analyzer

Detailed comparative analysis of PESQ and VISQOL behaviour in the context of playout delay adjustments introduced by VOIP jitter buffer algorithms

The default best-effort Internet presents significant challenges for delay-sensitive applications such as VoIP. To cope with non determinism, receiver playout strategies are utilised in VoIP applications that adapt to network condition. Such strategies can be divided into two different groups, namely per-talkspurt and per-packet. The former make use of silence periods within natural speech and adapt such silences to track network conditions, thus preserving the integrity of active speech talkspurts. Examples of this approach are described in [1, 2]. Per packet strategies are different in that adjustments are made both during silence periods and during talkspurts by time-scaling of packets, a technique also known in the literature as time-warping. This approach is more effective in coping with short network delay changes because the per talkspurt approach can only adapt during recognized silences even though the duration of many delay spikes may be less than that of a talkspurt. This approach however introduces potential degradation caused by the scaling of speech packets. Examples of this approach are described in [3, 4] and such techniques are frequently deployed in popular VoIP applications such as GoogleTalk and Skype. In this research, we focus on applications that deploy per talkspurt strategies, which are commonly found in current telecommunication networks

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