ADAPTIVE SPEECH QUALITY IN VOICE-OVER-IP COMMUNICATIONS

The quality of VoIP communication relies significantly on the network that transports the voice packets because this network does not usually guarantee the available bandwidth, delay, and loss that are critical for real-time voice traffic. The solution proposed here is to manage the voice-over-IP stream dynamically, changing parameters as needed to assure quality. The main objective of this dissertation is to develop an adaptive speech encoding system that can be applied to conventional (telephony-grade) and wideband voice communications. This comprehensive study includes the investigation and development of three key components of the system. First, to manage VoIP quality dynamically, a tool is needed to measure real-time changes in quality. The E-model, which exists for narrowband communication, is extended to a single computational technique that measures speech quality for narrowband and wideband VoIP codecs. This part of the dissertation also develops important theoretical work in the area of wideband telephony. The second system component is a variable speech-encoding algorithm. Although VoIP performance is affected by multiple codecs and network-based factors, only three factors can be managed dynamically: voice payload size, speech compression and jitter buffer management. Using an existing adaptive jitter-buffer algorithm, voice packet-size and compression variation are studied as they affect speech quality under different network conditions. This study explains the relationships among multiple parameters as they affect speech transmission and its resulting quality. Then, based on these two components, the third system component is a novel adaptive-rate control algorithm that establishes the interaction between a VoIP sender and receiver, and manages voice quality in real-time. Simulations demonstrate that the system provides better average voice quality than traditional VoIP

Secure VoIP Performance Measurement

This project presents a mechanism for instrumentation of secure VoIP calls. The experiments were run under different network conditions and security systems. VoIP services such as Google Talk, Express Talk and Skype were under test. The project allowed analysis of the voice quality of the VoIP services based on the Mean Opinion Score (MOS) values generated by Perceptual valuation of Speech Quality (PESQ). The quality of the audio streams produced were subjected to end-to-end delay, jitter, packet loss and extra processing in the networking hardware and end devices due to Internetworking Layer security or Transport Layer security implementations. The MOS values were mapped to Perceptual Evaluation of Speech Quality for wideband (PESQ-WB) scores. From these PESQ-WB scores, the graphs of the mean of 10 runs and box and whisker plots for each parameter were drawn. Analysis on the graphs was performed in order to deduce the quality of each VoIP service. The E-model was used to predict the network readiness and Common vulnerability Scoring System (CVSS) was used to predict the network vulnerabilities. The project also provided the mechanism to measure the throughput for each test case. The overall performance of each VoIP service was determined by PESQ-WB scores, CVSS scores and the throughput. The experiment demonstrated the relationship among VoIP performance, VoIP security and VoIP service type. The experiment also suggested that, when compared to an unsecure IPIP tunnel, Internetworking Layer security like IPSec ESP or Transport Layer security like OpenVPN TLS would improve a VoIP security by reducing the vulnerabilities of the media part of the VoIP signal. Morever, adding a security layer has little impact on the VoIP voice quality

Measuring and Monitoring Speech Quality for Voice over IP with POLQA, ViSQOL and P.563

There are many types of degradation which can occur in Voice over IP (VoIP) calls. Of interest in this work are degradations which occur independently of the codec, hardware or network in use. Specifically, their effect on the subjective and objec- tive quality of the speech is examined. Since no dataset suit- able for this purpose exists, a new dataset (TCD-VoIP) has been created and has been made publicly available. The dataset con- tains speech clips suffering from a range of common call qual- ity degradations, as well as a set of subjective opinion scores on the clips from 24 listeners. The performances of three ob- jective quality metrics: POLQA, ViSQOL and P.563, have been evaluated using the dataset. The results show that full reference metrics are capable of accurately predicting a variety of com- mon VoIP degradations. They also highlight the outstanding need for a wideband, single-ended, no-reference metric to mon- itor accurately speech quality for degradations common in VoIP scenarios

Impact of Different Active-Speech-Ratios on PESQ’s Predictions in Case of Independent and Dependent Losses (in Presence of Receiver-Side Comfort-Noise)

This paper deals with the investigation of PESQ’s behavior under independent and dependent loss conditions from an Active-Speech-Ratio perspective in presence of receiver-side comfort-noise. This reference signal characteristic is defined very broadly by ITU-T Recommendation P.862.3. That is the reason to investigate an impact of this characteristic on speech quality prediction more in-depth. We assess the variability of PESQ’s predictions with respect to Active-Speech-Ratios and loss conditions, as well as their accuracy, by comparing the predictions with subjective assessments. Our results show that an increase in amount of speech in the reference signal (expressed by the Active-Speech-Ratio characteristic) may result in an increase of the reference signal sensitivity to packet loss change. Interestingly, we have found two additional effects in this investigated case. The use of higher Active-Speech-Ratios may lead to negative shifting effect in MOS domain and also PESQ’s predictions accuracy declining. Predictions accuracy could be improved by higher packet losses

Feasibility study of VoIP in 3GPP UMTS release 5 interworking with fixed networks

Masteroppgave i informasjons- og kommunikasjonsteknologi 2003 - Høgskolen i Agder, GrimstadThe Universal Mobile Telecommunications System (UMTS) is denoted as a 3rd generation cellular system and has been designed with the objective to be a system with global coverage. With improvement of bandwidth capabilities, the UMTS system has the ability to support real time multimedia services. The focus in this thesis is Voice over IP (VoIP) which enables a user to make phone calls in the packet switched network in UMTS. This thesis starts with a presentation of VoIP with the quality requirements related to a voice session. A voice conversation needs a guaranteed quality to satisfy the participants. This thesis focuses on three main aspects; Quality of Service mechanisms (Best Effort, IntServ and DiffServ), VoIP in UMTS with a certain quality and last but not least implementation of Quality of Service (QoS) in a voice call interworking with external networks. Best Effort cannot be used when dealing with real time traffic such as VoIP. IntServ reserves resources from the application itself, and gives opportunity for each application in the terminal to request a certain quality. DiffServ works on a higher level and classifies traffic based on type of traffic, not for a particular request. For UMTS interworking with IP networks, the theoretical results suggest that IntServ over DiffServ should be used in the UMTS gateway node. An evaluation of the UMTS network is done by checking the voice quality attained by the network during a VoIP session in comparison of a traditional circuit switched call setup. Moreover, tests from the Norwegian UMTS network operator NetCom became useful when evaluating how well the VoIP could work when implementing UMTS release 5. The tests were set up with the focus on delay and voice quality in the network, and were meant for disclosing the differences with and without quality parameters during a transmission. Due to network restrictions the test results are limited

Power control for WCDMA

This project tries to introduce itself in the physical implementations that make possible the denominated third generation mobile technology. As well as to know the technology kind that makes possible, for example, a video-call in real time. During this project, the different phases passed from the election of WCDMA like the access method for UMTS will appear. Its coexistence with previous network GSM will be analyzed, where the compatibility between systems has been one of the most important aspects in the development of WCDMA, the involved standardization organisms in the process, as well as the different protocols that make the mobile communications within a network UTRAN possible. Special emphasis during the study of the great contribution that has offered WCDMA with respect to the control of power of the existing signals will be made. The future lines that are considered in the present, and other comment that already are in their last phase of development in the field of the mobile technology. UMTS through WCDMA can be summarized like a revolution of the air interface accompanied by a revolution in the network of their architecture

Wireless communications in the new millennium and third generation wireless networks

At the end of the 20 century, and at the beginning of this one, wireless communications are making large advances. The new technologies are on the way to provide a high-speed, high-quality information exchange between handheld terminals, and information repositories. The so called 2,5 generation networks, using the techniques like the HSCSD1, GPRS2, EDGE3, and the 3r generation wireless systems will help the wireless world to reach those goals. In this thesis I will start from the first and second-generation wireless networks, and then look into the 2,5 generation and 3rd generation wireless communications more in detail. The latest advances in the wireless world are the main focus of this paper although a short history of wireless communications is also given. The various aspects related to 3rd generation systems will be explored in this thesis, for example the air interface discussions, its time scale, its elements like the mobile equipment, software and security, USLM4, services that will be offered, etc. In addition, the technical factors and key technologies that are likely to shape the wireless network environment of the future will be explored. This part is expected to help us to see beyond the 3rd generation