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

Telemedicine system in the South Atlantic. Phase VII (I)

The information technology and communications (also known as ICT) are an essential part of the evolution in the economy and society. Access to information has become a vital tool for the development of a community. Consequently, there has been a phenomenon known as "digital divide", which refers to differences between countries that have access to ICT and those without. This project aims to provide a small gesture to reduce the gap between what is often called "differences between North and South", providing an improvement in the telemedicine system that the NGO Telecos Sense Fronteres is developing to the Region of South Atlantic of Nicaragua since 2006. This seventh stage of the project has tried, first of all, to make a careful maintenance of the infrastructure and equipment which currently has the Hospital Ernesto Sequeira Blanco of Bluefields; to improve their performance by incorporating a system of virtual desktops. In addition, the VoIP telephony system has been expanded to other departments of the hospital and other health centres of Bluefields, plus designing a virtual private network for their internal use. Secondly, there has been worked in a radio-link to bring telemedicine to Monkey Point, a rural community, isolated and without access to telecommunications. Finally, it has been emphasized the fact to bring the project to the community of Bluefields in order that they appropriate it and not see it as mere spectators. Related to this, some collaboration agreements have been closed with various local institutions that can contribute to self-management of the project, such as Bluefields Indian & Caribbean University (with engineering students) or the Sistema Local de Atención Integral en Salud (both with the economic and health sides).Català: Les tecnologies de la informació i les comunicacions (també anomenades TIC) són una part essencial dels canvis en l'economia i la societat actual. L'accés a la informació s'ha convertit en una eina vital pel desenvolupament d'una comunitat. Arran d'això, ha sorgit un fenomen conegut com a “bretxa digital”, el qual fa referència a les diferències entre països que tenen accés a les TIC, i aquells que no. Aquest projecte pretén aportar un petit gest per tal de disminuir aquesta bretxa entre el que freqüentment s'anomena “diferències entre Nord i Sud”, proporcionant una millora en el Sistema de Telemedicina que la ONG Telecos Sense Fronteres està desenvolupant a la Regió de l'Atlàntic Sur de Nicaragua des del 2006. En aquesta setena fase del projecte, s'ha pretés, en primer lloc, fer un manteniment acurat de la infraestructura i els equips dels quals disposa l'Hospital Ernesto Sequeira Blanco de Bluefields, millorat-ne les prestacions amb la incorporació d'un sistema d'escriptoris virtuals. A més, s'ha ampliat la xarxa de telefonia VoIP per tal de fer-la extensiva a altres departaments de l'hospital i a altres centres de salut de la població, concretant una xarxa privada virtual pel seu ús intern. En segon terme, s'ha treballat en un radioenllaç per fer arribar la telemedicina a Monkey Point, una comunitat rural, aïllada i sense accés a les telecomunicacions. Finalment, s'ha fet èmfasi en apropar el projecte a la pròpia comunitat de Bluefields amb la finalitat que se l'apropiïn i no el vegin com a mers espectadors. En aquesta direcció, s'han establert convenis de col·laboració amb diferents entitats que poden contribuir a l'autogestió del projecte, com són la Bluefields Indian & Caribbean University (amb estudiants d'enginyeria) o el Sistema Local de Atención Integral en Salud (amb la part econòmica i relacionada amb la salut)

An intelligent radio access network selection and optimisation system in heterogeneous communication environments

PhDThe overlapping of the different wireless network technologies creates heterogeneous communication environments. Future mobile communication system considers the technological and operational services of heterogeneous communication environments. Based on its packet switched core, the access to future mobile communication system will not be restricted to the mobile cellular networks but may be via other wireless or even wired technologies. Such universal access can enable service convergence, joint resource management, and adaptive quality of service. However, in order to realise the universal access, there are still many pending challenges to solve. One of them is the selection of the most appropriate radio access network. Previous work on the network selection has concentrated on serving the requesting user, but the existing users and the consumption of the network resources were not the main focus. Such network selection decision might only be able to benefit a limited number of users while the satisfaction levels of some users are compromised, and the network resources might be consumed in an ineffective way. Solutions are needed to handle the radio access network selection in a manner that both of the satisfaction levels of all users and the network resource consumption are considered. This thesis proposes an intelligent radio access network selection and optimisation system. The work in this thesis includes the proposal of an architecture for the radio access network selection and optimisation system and the creation of novel adaptive algorithms that are employed by the network selection system. The proposed algorithms solve the limitations of previous work and adaptively optimise network resource consumption and implement different policies to cope with different scenarios, network conditions, and aims of operators. Furthermore, this thesis also presents novel network resource availability evaluation models. The proposed models study the physical principles of the considered radio access network and avoid employing assumptions which are too stringent abstractions of real network scenarios. They enable the implementation of call level simulations for the comparison and evaluation of the performance of the network selection and optimisation algorithms

Subjective Audio Quality over a Secure IEEE 802.11n Draft 2.0 Wireless Local Area Network

This thesis investigates the quality of audio generated by a G.711 codec and transmission over an IEEE 802.11n draft 2.0 wireless local area network (WLAN). Decline in audio quality due to additional calls or by securing the WLAN with transport mode Internet Protocol Security (IPsec) is quantified. Audio quality over an IEEE 802.11n draft 2.0 WLAN is also compared to that of IEEE 802.11b and IEEE 802.11g WLANs under the same conditions. Audio quality is evaluated by following International Telecommunication Union Telecommunication Standardization Sector (ITU-T) Recommendation P.800, where human subjects rate audio clips recorded during various WLAN configurations. The Mean Opinion Score (MOS) is calculated as the average audio quality score given for each WLAN configuration. An 85% confidence interval is calculated for each MOS. Results suggest that audio quality over an IEEE 802.11n draft 2.0 WLAN is not higher than over an IEEE 802.11b WLAN when up to 10 simultaneous G.711 calls occur. A linear regression of the subjective scores also suggest that an IEEE 802.11n draft 2.0 WLAN can sustain an MOS greater than 3.0 (fair quality) for up to 75 simultaneous G.711 calls secured with WPA2, or up to 40 calls secured with both WPA2 and transport mode IPsec. The data strongly suggest that toll quality audio (MOS ≥ 4.0) is not currently practical over IEEE 802.11 WLANs secured with WPA2, even with the G.711 codec

An Investigation into the Effect of Security on Performance in a VoIP Network

Voice over Internet Protocol (VoIP) is a communications technology that transmits voice over packet switched networks such as the Internet. VoIP has been widely adopted by home and business customers. When adding security to a VoIP system, the quality of service and performance of the system are at risk. This study has two main objectives, firstly it illustrates suitable methods to secure the signalling and voice traffic within a VoIP system, secondly it evaluates the performance of a VoIP system after implementing different security methods. This study is carried out on a pilot system using an asterisk based SIP (Session initiation Protocol) server (Asterisk, 2009). Since VoIP is intended for use over the Internet, VPNs (Virtual Private Networks) have been used in a tunnel configuration to provide the service. Additionally the performance of networks level IPSec (Internet Protocol Security) and application level ZRTP (Zimmerman Real Time Transport Protocol) security have been compared with no security. Registration, call setup and voice transmission packets have been captured and analysed. The results have then been extrapolated to the Internet

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