Improving the robustness of CELP-like speech decoders using late-arrival packets information : application to G.729 standard in VoIP

L'utilisation de la voix sur Internet est une nouvelle tendance dans Ie secteur des télécommunications et de la réseautique. La paquetisation des données et de la voix est réalisée en utilisant Ie protocole Internet (IP). Plusieurs codecs existent pour convertir la voix codée en paquets. La voix codée est paquetisée et transmise sur Internet. À la réception, certains paquets sont soit perdus, endommages ou arrivent en retard. Ceci est cause par des contraintes telles que Ie délai («jitter»), la congestion et les erreurs de réseau. Ces contraintes dégradent la qualité de la voix. Puisque la transmission de la voix est en temps réel, Ie récepteur ne peut pas demander la retransmission de paquets perdus ou endommages car ceci va causer plus de délai. Au lieu de cela, des méthodes de récupération des paquets perdus (« concealment ») s'appliquent soit à l'émetteur soit au récepteur pour remplacer les paquets perdus ou endommages. Ce projet vise à implémenter une méthode innovatrice pour améliorer Ie temps de convergence suite a la perte de paquets au récepteur d'une application de Voix sur IP. La méthode a déjà été intégrée dans un codeur large-bande (AMR-WB) et a significativement amélioré la qualité de la voix en présence de <<jitter » dans Ie temps d'arrivée des trames au décodeur. Dans ce projet, la même méthode sera intégrée dans un codeur a bande étroite (ITU-T G.729) qui est largement utilise dans les applications de voix sur IP. Le codeur ITU-T G.729 défini des standards pour coder et décoder la voix a 8 kb/s en utilisant 1'algorithme CS-CELP (Conjugate Stmcture Algebraic Code-Excited Linear Prediction).Abstract: Voice over Internet applications is the new trend in telecommunications and networking industry today. Packetizing data/voice is done using the Internet protocol (IP). Various codecs exist to convert the raw voice data into packets. The coded and packetized speech is transmitted over the Internet. At the receiving end some packets are either lost, damaged or arrive late. This is due to constraints such as network delay (fitter), network congestion and network errors. These constraints degrade the quality of speech. Since voice transmission is in real-time, the receiver can not request the retransmission of lost or damaged packets as this will cause more delay. Instead, concealment methods are applied either at the transmitter side (coder-based) or at the receiver side (decoder-based) to replace these lost or late-arrival packets. This work attempts to implement a novel method for improving the recovery time of concealed speech The method has already been integrated in a wideband speech coder (AMR-WB) and significantly improved the quality of speech in the presence of jitter in the arrival time of speech frames at the decoder. In this work, the same method will be integrated in a narrowband speech coder (ITU-T G.729) that is widely used in VoIP applications. The ITUT G.729 coder defines the standards for coding and decoding speech at 8 kb/s using Conjugate Structure Algebraic Code-Excited Linear Prediction (CS-CELP) Algorithm

A-Interface Over Internet Protocol For User-Plane Connection Optimization In GSM/EDGE Radio Access Network

This thesis will cover a detailed study about the main motivations and benefits from using IP as a transport protocol for specifically A-interface in GERAN for Circuit Switched User-Plane (CS-UP) connection, in addition to the required protocols. The main study in this document will be around Real Time Protocol (RTP), Real Time Control Protocol (RTCP) negotiation for RTP packets multiplexing, for both cases, with and without RTP header compression. The focus will be about the communication between the Base Station Controller (BSC) and the Media GateWay (MGW), the bandwidth gain in accordance to the multiplexing delay for processing and buffering, the voice Quality of Service (QoS) and some other parameters

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

Tret personaliti dan faktor pemilihan kategori pekerjaan dalam kalangan pelajar kolej vokasional

Kajian ini mengkaji tret personaliti dan faktor pemilihan kategori pekerjaan dalam kalangan pelajar tahun dua Diploma Vokasional Malaysia di Kolej Vokasional dari negeri Johor. Kajian ini menggunakan Teori Tipologi Kerjaya Holland (1973) bagi mengenalpasti tret personaliti dan faktor utama yang mendorong pemilihan kerjaya para pelajar di kolej vokasional. Seramai 104 orang telah menurut serta di dalam kajian ini yang terdiri daripada pelajar tahun 2 Diploma Vokasional Malaysia yang menggunakan persampelan secara rawak. Alat kajian Self�Directed Search (SDS)-Form Eassy digunakan bagi meninjau tret personaliti responden, manakala item bagi faktor yang mendorong pemilihan kategori pekerjaan pelajar dibangunkan sendiri orang pengkaji. Analisis data deskriptif yang diperolehi menunjukkan bahawa faktor utama dalam pemilihan kerjaya responden adalah faktor minat. Dapatan kajian juga menunjukkan profil personaliti keseluruhan responden adalah S(sosial), I(investigatif) dan R(realistik). Ciri-ciri bagi kategori kerjaya ini adalah minat terhadap aktiviti-aktiviti yang berkaitan dengan memanipulasi orang lain seperti memberitahu, melatih, mendidik, menghindari aktiviti yang eksplisit, berstruktur serta aktiviti yang melibatkan bahan, peralatan dan mesin. Analisis Independent Sample T-test digunakan bagi melihat perbezaan yang signifikan di antara faktor yang mempengaruhi pemilihan kerjaya responden. Dapatan kajian menunjukkan terdapat perbezaan yang signifikan di antara faktor minat yang mempengaruhi pemilihan kerjaya responden. Kesimpulan dari kajian ini adalah pemilihan kategori pekerjaan pelajar-pelajar di kolej vokasional adalah SIR iaitu berupaya bekerja dengan orang lain dengan keupayaan kemahiran berfikir yang boleh diaplikasikan dalam kerja-kerja kemahiran bersesuaian bidang pengajian teknologi yang dipilih. Selain daripada itu minat merupakan faktor utama dalam pemilihan bidang dan pelajar-pelajar ini telah bersedia untuk menceburi bidang kerjaya yang selari dengan bidang pengajian. Kepentingan kajian ini boleh menjadi sebahagian dari rujukan bagi pemilihan bidang pengajian kepada pelajar-pelajar yang memilih TVET sebagai laluan kerjaya

Analyzing Voice And Video Call Service Performance Over A Local Area Network

Tez (Yüksek Lisans) -- İstanbul Teknik Üniversitesi, Fen Bilimleri Enstitüsü, 2010Thesis (M.Sc.) -- İstanbul Technical University, Institute of Science and Technology, 2010Bu çalışmada, VOIP teknolojisinden ve bu teknolojiyi kablolu ve kablosuz ortamda gerçeklemenin en önemli darboğazları anlatılacaktır. Ayrıca H.323, SIP (Session Initiation Protocol), Megaco ve MGCP gibi yaygın olarak kullanılan ses iletim protokolleri ve H.261, H.263 ve H.264 gibi görüntü iletim protokollerinden bahsedilmiştir. Ses kodek seçimi ve VOIP servis kalitesine etki eden faktörleri anlatılmaktadır. Bu tezde, ses, görüntü ve veri iletişimini aynı anda bünyesinde barındıran gerçek şebekeler simüle edilecektir. Kullanıcılara rastlantısal olarak ses, görüntü ve FTP gibi birtakım uygulamalar atanmıştır. Ayrıca önerilen kablolu şebekeye, kablosuz bir şebeke ilave edilerek sonuçlar incelenecektir. Optimal servis kalitesini sağlamak için seçilen uygun kuyruklama mekanizmaları ve kodek seçimlerini içeren senaryolar incelenecek ve OPNET ile elde edilmiş simülasyon sonuçları tartışılacaktır.In this study, we present a detailed description of the VoIP and also the most common challenges of implementing voice communication into wireline or wireless networks are discussed. Common voice protocols, such as H.323, Session Initiation Protocol (SIP), Megaco, MGCP and video protocols such as H.261, H.263, H.264 are described as well. CODEC selection and factors affecting VoIP Quality of Service are analyzed. We simulate a real network which includes both voice, video and data communication simultaneously. Workstations are randomly assigned to different applications, such as voice, video and FTP. We will also implement a wireless network to our proposed system. The scenarios including selecting appropriate queuing scheme and codec selection are presented and the simulation results with OPNET are drawn.Yüksek LisansM.Sc

Analysing the characteristics of VoIP traffic

In this study, the characteristics of VoIP traffic in a deployed Cisco VoIP phone system and a SIP based soft phone system are analysed. Traffic was captured in a soft phone system, through which elementary understanding about a VoIP system was obtained and experimental setup was validated. An advanced experiment was performed in a deployed Cisco VoIP system in the department of Computer Science at the University of Saskatchewan. Three months of traffic trace was collected beginning October 2006, recording address and protocol information for every packet sent and received on the Cisco VoIP network. The trace was analysed to find out the features of Cisco VoIP system and the findings were presented.This work appears to be one of the first real deployment studies of VoIP that does not rely on artificial traffic. The experimental data provided in this study is useful for design and modeling of such systems, from which more useful predictive models can be generated. The analysis method used in this research can be used for developing synthetic workload models. A clear understanding of usage patterns in a real VoIP network is important for network deployment and potential network activities such as integration, optimizations or expansion. The major factors affecting VoIP quality such as delay, jitter and loss were also measured and simulated in this study, which will be helpful in an advanced VoIP quality study. A traffic generator was developed to generate various simulated VoIP traffic. The data used to provide the traffic model parameters was chosen from peak traffic periods in the captured data from University of Saskatchewan deployment. By utilizing the Traffic Trace function in ns2, the simulated VoIP traffic was fed into ns2, and delay, jitter and packet loss were calculated for different scenarios. Two simulation experiments were performed. The first experiment simulated the traffic of multiple calls running on a backbone link. The second experiment simulated a real network environment with different traffic load patterns. It is significant for network expansion and integration

Performance evaluation of user mobility on QoS classes in a 3G network

The popularity of IP services is increasing and the demand for managing traffic with different QoS classes has become more challenging. The stability of the system is affected by the rate of voice traffic. Mobility allows users to be connected at all time where handover may occur as it is not always possible to be connected to the same base station. Mobility and handover cause severe interference, which affects overall throughput and capacity of the system. The system requires greater capacity with more coverage area. This study deals with the impact of user mobility on voice quality in IP based application in a 3G Network. The aim is to improve the system performance in mixed traffic environment. A mathematical model is used to analyse the impact of using different type of coder on packet end-to-end delay and packet loss. The simulation results indicate that types of coder affect the system performance. Application of scheduling based on weight and load balancing technique can improve the system performance. The deployment of scheduling based on weight and a load balancing technique have been investigated to reduce the end-to-end delay and to improve overall performance in mixed traffic environment. The results under different conditions are analysed and it is found that by applying scheduling scheme, the quality of voice communication can be improved. In addition, load balancing technique can be used to improve the performance of the system. Apart from the decrease in delay, the technique can increase the capacity of the system and the overall stability of the system can be further improved. Finally, network security is another important aspect of network administration. Security policies have to be defined and implemented so that critical sections of the network are protected against unwarranted traffic or unauthorized personnel. The impact of implementing IPSec has been tested for voice communication over IP in a 3G network. Implementing the security protocol does not significantly degrade the performance of the system.EThOS - Electronic Theses Online ServiceGBUnited Kingdo

Secure covert communications over streaming media using dynamic steganography

Streaming technologies such as VoIP are widely embedded into commercial and industrial applications, so it is imperative to address data security issues before the problems get really serious. This thesis describes a theoretical and experimental investigation of secure covert communications over streaming media using dynamic steganography. A covert VoIP communications system was developed in C++ to enable the implementation of the work being carried out. A new information theoretical model of secure covert communications over streaming media was constructed to depict the security scenarios in streaming media-based steganographic systems with passive attacks. The model involves a stochastic process that models an information source for covert VoIP communications and the theory of hypothesis testing that analyses the adversary‘s detection performance. The potential of hardware-based true random key generation and chaotic interval selection for innovative applications in covert VoIP communications was explored. Using the read time stamp counter of CPU as an entropy source was designed to generate true random numbers as secret keys for streaming media steganography. A novel interval selection algorithm was devised to choose randomly data embedding locations in VoIP streams using random sequences generated from achaotic process. A dynamic key updating and transmission based steganographic algorithm that includes a one-way cryptographical accumulator integrated into dynamic key exchange for covert VoIP communications, was devised to provide secure key exchange for covert communications over streaming media. The discrete logarithm problem in mathematics and steganalysis using t-test revealed the algorithm has the advantage of being the most solid method of key distribution over a public channel. The effectiveness of the new steganographic algorithm for covert communications over streaming media was examined by means of security analysis, steganalysis using non parameter Mann-Whitney-Wilcoxon statistical testing, and performance and robustness measurements. The algorithm achieved the average data embedding rate of 800 bps, comparable to other related algorithms. The results indicated that the algorithm has no or little impact on real-time VoIP communications in terms of speech quality (< 5% change in PESQ with hidden data), signal distortion (6% change in SNR after steganography) and imperceptibility, and it is more secure and effective in addressing the security problems than other related algorithms

Media gateway utilizando um GPU

Mestrado em Engenharia de Computadores e Telemátic

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