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

Quality aspects of Internet telephony

Internet telephony has had a tremendous impact on how people communicate. Many now maintain contact using some form of Internet telephony. Therefore the motivation for this work has been to address the quality aspects of real-world Internet telephony for both fixed and wireless telecommunication. The focus has been on the quality aspects of voice communication, since poor quality leads often to user dissatisfaction. The scope of the work has been broad in order to address the main factors within IP-based voice communication. The first four chapters of this dissertation constitute the background material. The first chapter outlines where Internet telephony is deployed today. It also motivates the topics and techniques used in this research. The second chapter provides the background on Internet telephony including signalling, speech coding and voice Internetworking. The third chapter focuses solely on quality measures for packetised voice systems and finally the fourth chapter is devoted to the history of voice research. The appendix of this dissertation constitutes the research contributions. It includes an examination of the access network, focusing on how calls are multiplexed in wired and wireless systems. Subsequently in the wireless case, we consider how to handover calls from 802.11 networks to the cellular infrastructure. We then consider the Internet backbone where most of our work is devoted to measurements specifically for Internet telephony. The applications of these measurements have been estimating telephony arrival processes, measuring call quality, and quantifying the trend in Internet telephony quality over several years. We also consider the end systems, since they are responsible for reconstructing a voice stream given loss and delay constraints. Finally we estimate voice quality using the ITU proposal PESQ and the packet loss process. The main contribution of this work is a systematic examination of Internet telephony. We describe several methods to enable adaptable solutions for maintaining consistent voice quality. We have also found that relatively small technical changes can lead to substantial user quality improvements. A second contribution of this work is a suite of software tools designed to ascertain voice quality in IP networks. Some of these tools are in use within commercial systems today

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

A cross-layer quality-oriented energy-efficient scheme for multimedia delivery in wireless local area networks

Wireless communication technologies, although emerged only a few decades ago, have grown fast in both popularity and technical maturity. As a result, mobile devices such as Personal Digital Assistants (PDA) or smart phones equipped with embedded wireless cards have seen remarkable growth in popularity and are quickly becoming one of the most widely used communication tools. This is mainly determined by the flexibility, convenience and relatively low costs associated with these devices and wireless communications. Multimedia applications have become by far one of the most popular applications among mobile users. However this type of application has very high bandwidth requirements, seriously restricting the usage of portable devices. Moreover, the wireless technology involves increased energy consumption and consequently puts huge pressure on the limited battery capacity which presents many design challenges in the context of battery powered devices. As a consequence, power management has raised awareness in both research and industrial communities and huge efforts have been invested into energy conservation techniques and strategies deployed within different components of the mobile devices. Our research presented in this thesis focuses on energy efficient data transmission in wireless local networks, and mainly contributes in the following aspects: 1. Static STELA, which is a Medium Access Control (MAC) layer solution that adapts the sleep/wakeup state schedule of the radio transceiver according to the bursty nature of data traffic and real time observation of data packets in terms of arrival time. The algorithm involves three phases– slow start phase, exponential increase phase, and linear increase phase. The initiation and termination of each phase is self-adapted to real time traffic and user configuration. It is designed to provide either maximum energy efficiency or best Quality of Service (QoS) according to user preference. 2. Dynamic STELA, which is a MAC layer solution deployed on the mobile devices and provides balanced performance between energy efficiency and QoS. Dynamic STELA consists of the three phase algorithm used in static STELA, and additionally employs a traffic modeling algorithm to analyze historical traffic data and estimate the arrival time of the next burst. Dynamic STELA achieves energy saving through intelligent and adaptive increase of Wireless Network Interface Card (WNIC) sleeping interval in the second and the third phase and at the same time guarantees delivery performance through optimal WNIC waking timing before the estimated arrival of new data burst. 3. Q-PASTE, which is a quality-oriented cross-layer solution with two components employed at different network layers, designed for multimedia content delivery. First component, the Packet/ApplicaTion manager (PAT) is deployed at the application layer of both service gateway and client host. The gateway level PAT utilizes fast start, as a widely supported technique for multimedia content delivery, to achieve high QoS and shapes traffic into bursts to reduce the wireless transceiver’s duty cycle. Additionally, gateway-side PAT informs client host the starting and ending time of fast start to assist parameter tuning. The client-side PAT monitors each active session and informs the MAC layer about their traffic-related behavior. The second component, dynamic STELA, deployed at MAC layer, adaptively adjusts the sleep/wake-up behavior of mobile device wireless interfaces in order to reduce energy consumption while also maintaining high Quality of Service (QoS) levels. 4. A comprehensive survey on energy efficient standards and some of the most important state-of-the-art energy saving technologies is also provided as part of the work

Excitação multi-taxa usando quantização vetorial estruturada em árvore para o codificador CS-ACELP com aplicação em VoIP

Dissertação (mestrado) - Universidade Federal de Santa Catarina, Centro Tecnológico. Programa de Pós-Graduação em Engenharia Elétrica.Este trabalho apresenta um estudo sobre codificação multi-taxa estruturada sobre o algoritmo CS-ACELP (Conjugate-Structure Algebraic-Code-Excited Linear-Prediction) e a especificação G.729, cujo objetivo é propor um codificador com taxa variável, através da busca da melhor excitação fixa usando codebook estruturado em árvore, para aplicações VoIP (Voice-over-IP). A mudança progressiva do transporte de voz das redes de circuito para as redes IP (Internet Protocol), apesar dos diversos aspectos positivos, tem exposto algumas deficiências intrínsecas destas, mais apropriadas ao tráfego de #melhor esforço# do que ao tráfego com requisitos de tempo. Esta proposta está inserida no conjunto das iniciativas, no âmbito do transmissor, que procuram minimizar os efeitos danosos da rede sobre a qualidade da voz reconstruída. O codebook proposto tem estrutura em árvore binária, concebida a partir de uma heurística onde os vetores CS-ACELP são ordenados por valor de forma decrescente. Uma estratégia particular de armazenamento dos nós, envolvendo simplificação nos centróides, codificação diferencial e geração automática dos dois últimos níveis da árvore, permite reduzir o espaço de armazenamento de 640 para apenas 7 kwords. Através deste modelo chega-se a 13 taxas de codificação, de 5,6 a 8,0 kbit/s, com passo de 0,2 kbit/s. A relação sinal ruído fica em 1,5 dB abaixo da mesma medida na especificação G.729 para a taxa de 5,6 kbit/s, e apenas 0,6 dB abaixo quando na taxa 8,0 kbit/s. Testes subjetivos mostraram uma qualidade bastante aceitável para a taxa mínima e praticamente indistinguível do codec original na taxa máxima. Além disso, a busca da melhor excitação é 2,4 vezes mais rápida em comparação ao codec G.729 e pode ser totalmente compatível com este se a taxa for fixa em 8,0 kbit/s. This work presents a study about multi-rate coding structured over CS-ACELP (Conjugate-Structure Algebraic-Code-Excited Linear-Prediction) algorithm and G.729 standard, whose purpose is to come up with a variable rate codec by means of best fixed excitation search using a tree structured codebook, for VoIP (Voice-over-IP) applications. The progressive change of voice transmission from circuit switched to IP (Internet orks, besides its many positive aspects, has exposed some natural deficiencies of the latter, better suited to best effort traffics than traffics with time requirements. This proposition can be inserted in the bunch of efforts, related to the sender, that seek to reduce the network impairments over the quality of reconstructed voice. The suggested codebook has a binary tree structure heuristically conceived where algebraic CSACELP vectors are disposed by value in a decreasing order. Additionally, a particular approach to store the tree nodes are considered, which involves centroid implification, differential coding and automatic generation of the last two layers of the tree, squeezing the storing space from 640 down to 7 kwords. Through this model we reach 13 coding rates, ranging from 5.6 to 8.0 kbit/s, with 0.2 kbit/s step. The signal-to-noise ratio is 1.5 dB below the same measure for G.729 standard at the rate 5.6 kbit/s, and just 0.6 dB lower at 8.0 kbit/s. Subjective tests pointed to an acceptable quality at minimum rate and virtually indistinguishable quality from the original codec at the maximum one. Also, searching for the best fixed excitation is 2.4 times faster than G.729 and can be truly compatible with it if the rate is fixed in 8 kbit/s

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

A MODEL FOR PREDICTING THE PERFORMANCE OF IP VIDEOCONFERENCING

With the incorporation of free desktop videoconferencing (DVC) software on the majority of the world's PCs, over the recent years, there has, inevitably, been considerable interest in using DVC over the Internet. The growing popularity of DVC increases the need for multimedia quality assessment. However, the task of predicting the perceived multimedia quality over the Internet Protocol (IP) networks is complicated by the fact that the audio and video streams are susceptible to unique impairments due to the unpredictable nature of IP networks, different types of task scenarios, different levels of complexity, and other related factors. To date, a standard consensus to define the IP media Quality of Service (QoS) has yet to be implemented. The thesis addresses this problem by investigating a new approach to assess the quality of audio, video, and audiovisual overall as perceived in low cost DVC systems. The main aim of the thesis is to investigate current methods used to assess the perceived IP media quality, and then propose a model which will predict the quality of audiovisual experience from prevailing network parameters. This thesis investigates the effects of various traffic conditions, such as, packet loss, jitter, and delay and other factors that may influence end user acceptance, when low cost DVC is used over the Internet. It also investigates the interaction effects between the audio and video media, and the issues involving the lip sychronisation error. The thesis provides the empirical evidence that the subjective mean opinion score (MOS) of the perceived multimedia quality is unaffected by lip synchronisation error in low cost DVC systems. The data-gathering approach that is advocated in this thesis involves both field and laboratory trials to enable the comparisons of results between classroom-based experiments and real-world environments to be made, and to provide actual real-world confirmation of the bench tests. The subjective test method was employed since it has been proven to be more robust and suitable for the research studies, as compared to objective testing techniques. The MOS results, and the number of observations obtained, have enabled a set of criteria to be established that can be used to determine the acceptable QoS for given network conditions and task scenarios. Based upon these comprehensive findings, the final contribution of the thesis is the proposal of a new adaptive architecture method that is intended to enable the performance of IP based DVC of a particular session to be predicted for a given network condition

Bandwidth reservation in mobile ad hoc networks for providing QoS : adaptation for voice support

Le support de qualité de service (QoS) dans les réseaux MANETs (Mobile Ad-Hoc NETworks) a attiré une grande attention ces dernières années. Bien que beaucoup de travaux de recherche ont été consacré pour offrir la QoS dans les réseaux filaires et cellulaires, les solutions de QoS pour le support du trafic temps réel dans les MANET reste l'un des domaines de recherche les plus difficiles et les moins explorés. En fait, les applications temps réel telles que la voix et la vidéo ne pourrait pas fonctionner correctement dans les MANET sans l'utilisation d'un protocole de contrôle d'accès au support (MAC) orienté QoS. En effet, les trafics temps réel demandent des exigences strictes en termes de délai de transmission et de taux de perte de paquets qui peuvent être remplies uniquement si la sous-couche MAC fournit un délai d'accès au canal borné, et un faible taux de collision. Le but de cette thèse est la proposition et l'analyse d'un protocole MAC basé sur la réservation pour garantir la QoS dans les MANETs. Tout d'abord, nous étudions un problème majeur dans la réservation de ressources dans les MANETs qui est la cohérence des réservations. Notre analyse des protocoles de réservation existant pour les MANETs révèle que de nombreux conflits de réservations entre les nœuds voisins se produisent pendant la phase d'établissement de réservation. Ces conflits, qui sont principalement dues à la collision des messages de contrôle de réservation, ont un impact important sur les performances du protocole de réservation, et conduisent à un taux de collision et de perte de paquet importants pendant la durée de vie de la connexion, ce qui n'est pas acceptable pour les trafics temps réels. Nous proposons un nouveau protocole MAC basé sur la réservation qui résout ces conflits. Le principe de notre protocole est d'établir une meilleure coordination entre les nœuds voisins afin d'assurer la cohérence des réservations. Ainsi, avant de considérer qu'une réservation est réussite, le protocole s'assure que chaque message de contrôle envoyé par un nœud pour établir une réservation est bien reçu par tous ses nœuds voisins. Dans la deuxième partie de cette thèse, nous appliquons le protocole de réservation proposé au trafic de type voix. Ainsi, nous étendons ce protocole afin de prendre en compte les caractéristiques du trafic voix, tout en permettant le transport de trafic de données. Nous nous focalisons sur l'utilisation efficace de la bande passante et les mécanismes pour réduire le gaspillage de bande passante. La dernière partie de cette thèse concerne l'extension du protocole proposé en vue de réserver la bande passante pour une connexion temps réel sur un chemin. Ainsi, le protocole MAC de réservation proposé est couplé avec un protocole de routage réactif. En outre, le protocole est étendu avec des mécanismes de gestion de à mobilité afin de faire face à la dégradation des performances due à la mobilité des nœuds. Nous évaluons les performances du protocole proposé dans plusieurs scénarios dans lesquels nous montrons sa supériorité par rapport aux standards existants.QoS provisioning over Mobile Ad-Hoc Networks (MANETs) has attracted a great attention in recent years. While much research effort has been devoted to provide QoS over wired and cellular networks, QoS solutions for the support of real-time traffic over MANETs remains one of the most challenging and least explored areas. In fact, real-time applications such as voice and video could not function properly on MANETs without a QoS oriented medium access control (MAC) scheme. Indeed, real-time traffics claim strict requirements in terms of transmission delay and packet dropping that can be fulfilled only if the MAC sub-layer provides bounded channel access delay, and low collision rate. The purpose of this thesis is the proposal and analysis of an efficient reservation MAC protocol to provide QoS support over MANETs. Firstly, we study one major issue in resource reservation for MANETs which is reservation consistency. Our analysis of existing reservation MAC protocols for MANETs reveals that many reservation conflicts between neighbor nodes occur during the reservation establishment phase. These conflicts which are mainly due to collisions of reservation control messages, have an important impact on the performance of the reservation protocol, and lead to a significant collision and loss of packets during the life-time of the connection, which is not acceptable for real-time traffics. We design a new reservation MAC protocol that resolves these conflicts. The main principle of our protocol is to achieve better coordination between neighbor nodes in order to ensure consistency of reservations. Thus, before considering a reservation as successful, the protocol tries to ensure that each reservation control message transmitted by a node is successfully received by all its neighbors. In the second part of this thesis, we apply the proposed reservation protocol to voice traffic. Thus, we extend this protocol in order to take into account the characteristics of voice traffic, while enabling data traffic. We focus on efficient bandwidth utilization and mechanisms to reduce the waste of bandwidth. The last part of this thesis relates to the extension of the proposed protocol in order to reserve resources for a real-time connection along a path. Thus, the proposed reservation MAC protocol is coupled with a reactive routing protocol. In addition, the protocol is extended with mobility handling mechanisms in order to cope with performance degradation due to mobility of nodes. We evaluate the performance of the proposed scheme in several scenarios where we show its superiority compared to existing standards

Device characteristics-based differentiated energy-efficient adaptive solution for multimedia delivery over heterogeneous wireless networks

Energy efficiency is a key issue of highest importance to mobile wireless device users, as those devices are powered by batteries with limited power capacity. It is of very high interest to provide device differentiated user centric energy efficient multimedia content delivery based on current application type, energy-oriented device features and user preferences. This thesis presents the following research contributions in the area of energy efficient multimedia delivery over heterogeneous wireless networks: 1. ASP: Energy-oriented Application-based System profiling for mobile devices: This profiling provides services to other contributions in this thesis. By monitoring the running applications and the corresponding power demand on the smart mobile device, a device energy model is obtained. The model is used in conjunction with applications’ power signature to provide device energy constraints posed by running applications. 2. AWERA 3. DEAS: A Device characteristics-based differentiated Energy-efficient Adaptive Solution for video delivery over heterogeneous wireless networks. Based on the energy constraint, DEAS performs energy efficient content delivery adaptation for the current application. Unlike the existing solutions, DEAS takes all the applications running on the system into account and better balances QoS and energy efficiency. 4. EDCAM 5. A comprehensive survey on state-of-the-art energy-efficient network protocols and energy-saving network technologies

Control of real-time multimedia applications in best-effort networks

The increasing demand for real-time multimedia applications and the lack of quality of service (QoS) support in public best-effort or Internet Protocol (IP) networks has prompted many researchers to propose improvements on the QoS of such networks. This research aims to improve the QoS of real-time multimedia applications in public best-effort networks, without modifying the core network infrastructure or the existing codecs of the original media applications. A source buffering control is studied based on a fluid model developed for a single flow transported over a best-effort network while allowing for flow reversal. It is shown that this control is effective for QoS improvement only when there is sufficient flow reversal or packet reordering in the network. An alternate control strategy based on predictive multi-path switching is studied where only two paths are considered as alternate options. Initially, an emulation study is performed, exploring the impact of path loss rate and traffic delay signal frequency content on the proposed control. The study reveals that this control strategy provides the best QoS improvement when the average comprehensive loss rates of the two paths involved are between 5% and 15%, and when the delay signal frequency content is around 0.5 Hz. Linear and nonlinear predictors are developed using actual network data for use in predictive multi-path switching control. The control results show that predictive path switching is better than no path switching, yet no one predictor developed is best for all cases studied. A voting based control strategy is proposed to overcome this problem. The results show that the voting based control strategy results in better performance for all cases studied. An actual voice quality test is performed, proving that predictive path switching is better than no path switching. Despite the improvements obtained, predictive path switching control has some scalability problems and other shortcomings that require further investigation. If there are more paths available to choose from, the increasing overhead in probing traffic might become unacceptable. Further, if most of the VoIP flows on the Internet use this control strategy, then the conclusions of this research might be different, requiring modifications to the proposed approach. Further studies on these problems are needed