Final report on the evaluation of RRM/CRRM algorithms

Deliverable public del projecte EVERESTThis deliverable provides a definition and a complete evaluation of the RRM/CRRM algorithms selected in D11 and D15, and evolved and refined on an iterative process. The evaluation will be carried out by means of simulations using the simulators provided at D07, and D14.Preprin

LTE Optimization and Resource Management in Wireless Heterogeneous Networks

Mobile communication technology is evolving with a great pace. The development of the Long Term Evolution (LTE) mobile system by 3GPP is one of the milestones in this direction. This work highlights a few areas in the LTE radio access network where the proposed innovative mechanisms can substantially improve overall LTE system performance. In order to further extend the capacity of LTE networks, an integration with the non-3GPP networks (e.g., WLAN, WiMAX etc.) is also proposed in this work. Moreover, it is discussed how bandwidth resources should be managed in such heterogeneous networks. The work has purposed a comprehensive system architecture as an overlay of the 3GPP defined SAE architecture, effective resource management mechanisms as well as a Linear Programming based analytical solution for the optimal network resource allocation problem. In addition, alternative computationally efficient heuristic based algorithms have also been designed to achieve near-optimal performance

Performance of Wi-Fi coordination schemes for VolP in the presence of FTP data.

Evolved 3GPP cellular core networks have made co-existence of heterogeneous Wireless Access networks (HetNets) possible. The evolved core network along with the development of multimode end user devices have led to the realisation of converged Access Networks. Wireless Local Area Networks (WLANs) are assuming a prominent role in the telecommunications ecosystem due to their cost effectiveness, ease of deployment and operation in the free spectrum. Although WLANs are only data centric, there will be greater demand for Voice over Internet Protocol (VoIP) over WLANs as multimode smart-phones become accessible and operators integrate WLANs into their business models. Therefore, it is imperative that WLAN’s ability to support VoIP services is thoroughly understood. Currently, the design of call admission control mechanisms for WLANs that support heterogeneous (data and voice) traffic is a challenging issue. The challenge stems from the difficulty of modelling the behaviour heterogeneous traffic, mixed VoIP and data traffic. IEEE 802.11 WLANs use two types of medium access schemes, the polling based schemes and the contention based schemes. Both types of WLAN coordination schemes have not been thoroughly investigated for their ability to support VoIP over WLANs in the presence of File Transfer Protocol (FTP) data sessions. File Transfer Protocol (FTP) is a Transport Control Protocol(TCP) based file exchange protocol. TCP was optimised for wired networks and as a result it is unsuitable for wireless network. Furthermore, it was not optimised to co-exist with VoIP and as a result of its burstiness it has severe impact on the jitter, packet-loss and delay of VoIP traffic. The purpose of the work presented in this report is to evaluate the performance of Distributed Coordinated Function (DCF), Point Coordination Function (PCF) and Enhanced Distributed Coordinated Function (EDCF) techniques’ ability to manage Voice Over Internet Protocol (VoIP) over WLAN in the presence of contending heavy FTP data. The key question this work seeks to answer is, are the Medium Access Control (MAC) coordination techniques in their present form capable of carrying VoIP data in the presence of other data. In other words, how realistic is the deployment of VoIP services with FTP services in the same network, using the current coordination schemes for WLAN? Can these coordination schemes be improved by using current MAC enhancements such as fragmentation and increasing the Access Point buffer? The study is carried out for IEEE 802.11g as this is still the most widely deployed standard. The performance is evaluated by setting up a network of stations that generate both voice and FTP traffic in OPNET. The two network configurations are 30-Voice stations and 30-FTP stations; 15-Voice stations and 45-FTP stations. Moreover, two codecs G.711 and G.723 are compared to assess the effect of codec selection on performance

Bilateral Waveform Similarity Overlap-and-Add Based Packet Loss Concealment for Voice over IP

This paper invested a bilateral waveform similarity overlap-and-add algorithm for voice packet lost. Since Packet lost will cause the semantic misunderstanding, it has become one of the most essential problems in speech communication. This investment is based on waveform similarity measure using overlap-and-Add algorithm and provides the bilateral information to enhance the speech signal reconstruction. Traditionally, it has been improved that waveform similarity overlap-and-add (WSOLA) technique is an effective algorithm to deal with packet loss concealment (PLC) for real-time time communication. WSOLA algorithm is widely applied to deal with the length adaptation and packet loss concealment of speech signal. Time scale modification of audio signal is one of the most essential research topics in data communication, especially in voice of IP (VoIP). Herein, the proposed the bilateral WSOLA (BWSOLA) that is derived from WSOLA. Instead of only exploitation one direction speech data, the proposed method will reconstruct the lost voice data according to the preceding and cascading data. The related algorithms have been developed to achieve the optimal reconstructing estimation. The experimental results show that the quality of the reconstructed speech signal of the bilateral WSOLA is much better compared to the standard WSOLA and GWSOLA on different packet loss rate and length using the metrics PESQ and MOS. The significant improvement is obtained by bilateral information and proposed method. The proposed bilateral waveform similarity overlap-and-add (BWSOLA) outperforms the traditional approaches especially in the long duration data loss

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

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

Context-awareness for ubiquitous media service delivery in next generation networks

Les récentes avancées technologiques permettent désormais la fabrication de terminaux mobiles de plus en plus compacts et dotés de plusieurs interfaces réseaux. Le nouveau modèle de consommation de médias se résume par le concept "Anytime, Anywhere, Any Device" et impose donc de nouvelles exigences en termes de déploiement de services ubiquitaires. Cependant la conception et le developpement de réseaux ubiquitaires et convergents de nouvelles générations soulèvent un certain nombre de défis techniques. Les standards actuels ainsi que les solutions commerciales pourraient être affectés par le manque de considération du contexte utilisateur. Le ressenti de l'utilisateur concernant certains services multimédia tels que la VoIP et l'IPTV dépend fortement des capacités du terminal et des conditions du réseau d'accès. Cela incite les réseaux de nouvelles générations à fournir des services ubiquitaires adaptés à l'environnement de l'utilisateur optimisant par la même occasion ses resources. L'IP Multimedia Subsystem (IMS) est une architecture de nouvelle génération qui centralise l'accès aux services et permet la convergence des réseaux fixe/mobile. Néanmoins, l'évolution de l'IMS est nécessaire sur les points suivants :- l'introduction de la sensibilité au contexte utilisateur et de la PQoS (Perceived QoS) : L'architecture IMS ne prend pas en compte l'environnement de l'utilisateur, ses préférences et ne dispose pas d'un méchanisme de gestion de PQOS. Pour s'assurer de la qualité fournit à l'utilisateur final, des informations sur l'environnement de l'utilisateur ainsi que ses préférences doivent transiter en cœur de réseau afin d'y être analysés. Ce traitement aboutit au lancement du service qui sera adapté et optimisé aux conditions observées. De plus pour le service d'IPTV, les caractéristiques spatio-temporelles de la vidéo influent de manière importante sur la PQoS observée côté utilisateur. L'adaptation des services multimédias en fonction de l'évolution du contexte utilisateur et de la nature de la vidéo diffusée assure une qualité d'expérience à l'utilisateur et optimise par la même occasion l'utilisation des ressources en cœur de réseau.- une solution de mobilité efficace pour les services conversationnels tels que la VoIP : Les dernières publications 3GPP fournissent deux solutions de mobilité: le LTE proposeMIP comme solution de mobilité alors que l'IMS définit une mobilité basée sur le protocoleapplicatif SIP. Ces standards définissent le système de signalisation mais ne s'avancent pas sur la gestion du flux média lors du changement d'interface réseau. La deuxième section introduit une étude comparative détaillée des solutions de mobilité dans les NGNs.Notre première contribution est la spécification de l'architecture globale de notre plateforme IMS sensible au contexte utilisateur réalisée au sein du projet Européen ADAMANTIUM. Nous détaillons tout d'abord le serveur MCMS intelligent placé dans la couche application de l'IMS. Cet élément récolte les informations de qualité de services à différents équipements réseaux et prend la décision d'une action sur l'un de ces équipements. Ensuite nous définissons un profil utilisateur permettant de décrire son environnement et de le diffuser en coeur de réseau. Une étude sur la prédiction de satisfaction utilisateur en fonction des paramètres spatio-temporels de la vidéo a été réalisée afin de connaître le débit idéal pour une PQoS désirée.Notre deuxième contribution est l'introduction d'une solution de mobilité adaptée aux services conversationnels (VoIP) tenant compte du contexte utilisateur. Notre solution s'intègre à l'architecture IMS existante de façon transparente et permet de réduire le temps de latence du handover. Notre solution duplique les paquets de VoIP sur les deux interfaces actives pendant le temps de la transition. Parallèlement, un nouvel algorithme de gestion de mémoire tampon améliore la qualité d'expérience pour le service de VoIP.The latest advances in technology have already defied Moore s law. Thanks to research and industry, hand-held devices are composed of high processing embedded systems enabling the consumption of high quality services. Furthermore, recent trends in communication drive users to consume media Anytime, Anywhere on Any Device via multiple wired and wireless network interfaces. This creates new demands for ubiquitous and high quality service provision management. However, defining and developing the next generation of ubiquitous and converged networks raise a number of challenges. Currently, telecommunication standards do not consider context-awareness aspects for network management and service provisioning. The experience felt by the end-user consuming for instance Voice over IP (VoIP) or Internet Protocol TeleVision (IPTV) services varies depending mainly on user preferences, device context and network resources. It is commonly held that Next Generation Network (NGN) should deliver personalized and effective ubiquitous services to the end user s Mobile Node (MN) while optimizing the network resources at the network operator side. IP Multimedia Subsystem (IMS) is a standardized NGN framework that unifies service access and allows fixed/mobile network convergence. Nevertheless IMS technology still suffers from a number of confining factors that are addressed in this thesis; amongst them are two main issues :The lack of context-awareness and Perceived-QoS (PQoS):-The existing IMS infrastructure does not take into account the environment of the user ,his preferences , and does not provide any PQoS aware management mechanism within its service provisioning control system. In order to ensure that the service satisfies the consumer, this information need to be sent to the core network for analysis. In order to maximize the end-user satisfaction while optimizing network resources, the combination of a user-centric network management and adaptive services according to the user s environment and network conditions are considered. Moreover, video content dynamics are also considered as they significantly impact on the deduced perceptual quality of IPTV services. -The lack of efficient mobility mechanism for conversational services like VoIP :The latest releases of Third Generation Partnership Project (3GPP) provide two types of mobility solutions. Long-Term Evolution (LTE) uses Mobile IP (MIP) and IMS uses Session Initiation Protocol (SIP) mobility. These standards are focusing on signaling but none of them define how the media should be scheduled in multi-homed devices. The second section introduces a detailed study of existing mobility solutions in NGNs. Our first contribution is the specification of the global context-aware IMS architecture proposed within the European project ADAptative Management of mediA distributioN based on saTisfaction orIented User Modeling (ADAMANTIUM). We introduce the innovative Multimedia Content Management System (MCMS) located in the application layer of IMS. This server combines the collected monitoring information from different network equipments with the data of the user profile and takes adaptation actions if necessary. Then, we introduce the User Profile (UP) management within the User Equipment (UE) describing the end-user s context and facilitating the diffusion of the end-user environment towards the IMS core network. In order to optimize the network usage, a PQoS prediction mechanism gives the optimal video bit-rate according to the video content dynamics. Our second contribution in this thesis is an efficient mobility solution for VoIP service within IMS using and taking advantage of user context. Our solution uses packet duplication on both active interfaces during handover process. In order to leverage this mechanism, a new jitter buffer algorithm is proposed at MN side to improve the user s quality of experience. Furthermore, our mobility solution integrates easily to the existing IMS platform.BORDEAUX1-Bib.electronique (335229901) / SudocSudocFranceF

Quality of service differentiation for multimedia delivery in wireless LANs

Delivering multimedia content to heterogeneous devices over a variable networking environment while maintaining high quality levels involves many technical challenges. The research reported in this thesis presents a solution for Quality of Service (QoS)-based service differentiation when delivering multimedia content over the wireless LANs. This thesis has three major contributions outlined below: 1. A Model-based Bandwidth Estimation algorithm (MBE), which estimates the available bandwidth based on novel TCP and UDP throughput models over IEEE 802.11 WLANs. MBE has been modelled, implemented, and tested through simulations and real life testing. In comparison with other bandwidth estimation techniques, MBE shows better performance in terms of error rate, overhead, and loss. 2. An intelligent Prioritized Adaptive Scheme (iPAS), which provides QoS service differentiation for multimedia delivery in wireless networks. iPAS assigns dynamic priorities to various streams and determines their bandwidth share by employing a probabilistic approach-which makes use of stereotypes. The total bandwidth to be allocated is estimated using MBE. The priority level of individual stream is variable and dependent on stream-related characteristics and delivery QoS parameters. iPAS can be deployed seamlessly over the original IEEE 802.11 protocols and can be included in the IEEE 802.21 framework in order to optimize the control signal communication. iPAS has been modelled, implemented, and evaluated via simulations. The results demonstrate that iPAS achieves better performance than the equal channel access mechanism over IEEE 802.11 DCF and a service differentiation scheme on top of IEEE 802.11e EDCA, in terms of fairness, throughput, delay, loss, and estimated PSNR. Additionally, both objective and subjective video quality assessment have been performed using a prototype system. 3. A QoS-based Downlink/Uplink Fairness Scheme, which uses the stereotypes-based structure to balance the QoS parameters (i.e. throughput, delay, and loss) between downlink and uplink VoIP traffic. The proposed scheme has been modelled and tested through simulations. The results show that, in comparison with other downlink/uplink fairness-oriented solutions, the proposed scheme performs better in terms of VoIP capacity and fairness level between downlink and uplink traffic