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

Quality of Service optimisation framework for Next Generation Networks

Within recent years, the concept of Next Generation Networks (NGN) has become widely accepted within the telecommunication area, in parallel with the migration of telecommunication networks from traditional circuit-switched technologies such as ISDN (Integrated Services Digital Network) towards packet-switched NGN. In this context, SIP (Session Initiation Protocol), originally developed for Internet use only, has emerged as the major signalling protocol for multimedia sessions in IP (Internet Protocol) based NGN. One of the traditional limitations of IP when faced with the challenges of real-time communications is the lack of quality support at the network layer. In line with NGN specification work, international standardisation bodies have defined a sophisticated QoS (Quality of Service) architecture for NGN, controlling IP transport resources and conventional IP QoS mechanisms through centralised higher layer network elements via cross-layer signalling. Being able to centrally control QoS conditions for any media session in NGN without the imperative of a cross-layer approach would result in a feasible and less complex NGN architecture. Especially the demand for additional network elements would be decreased, resulting in the reduction of system and operational costs in both, service and transport infrastructure. This thesis proposes a novel framework for QoS optimisation for media sessions in SIP-based NGN without the need for cross-layer signalling. One key contribution of the framework is the approach to identify and logically group media sessions that encounter similar QoS conditions, which is performed by applying pattern recognition and clustering techniques. Based on this novel methodology, the framework provides functions and mechanisms for comprehensive resource-saving QoS estimation, adaptation of QoS conditions, and support of Call Admission Control. The framework can be integrated with any arbitrary SIP-IP-based real-time communication infrastructure, since it does not require access to any particular QoS control or monitoring functionalities provided within the IP transport network. The proposed framework concept has been deployed and validated in a prototypical simulation environment. Simulation results show MOS (Mean Opinion Score) improvement rates between 53 and 66 percent without any active control of transport network resources. Overall, the proposed framework comes as an effective concept for central controlled QoS optimisation in NGN without the need for cross-layer signalling. As such, by either being run stand-alone or combined with conventional QoS control mechanisms, the framework provides a comprehensive basis for both the reduction of complexity and mitigation of issues coming along with QoS provision in NGN

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

Technology Directions for the 21st Century

New technologies will unleash the huge capacity of fiber-optic cable to meet growing demands for bandwidth. Companies will continue to replace private networks with public network bandwidth-on-demand. Although asynchronous transfer mode (ATM) is the transmission technology favored by many, its penetration will be slower than anticipated. Hybrid networks - e.g., a mix of ATM, frame relay, and fast Ethernet - may predominate, both as interim and long-term solutions, based on factors such as availability, interoperability, and cost. Telecommunications equipment and services prices will decrease further due to increased supply and more competition. Explosive Internet growth will continue, requiring additional backbone transmission capacity and enhanced protocols, but it is not clear who will fund the upgrade. Within ten years, space-based constellations of satellites in Low Earth orbit (LEO) will serve mobile users employing small, low-power terminals. 'Little LEO's' will provide packet transmission services and geo-position determination. 'Big LEO's' will function as global cellular telephone networks, with some planning to offer video and interactive multimedia services. Geosynchronous satellites also are proposed for mobile voice grade links and high-bandwidth services. NASA may benefit from resulting cost reductions in components, space hardware, launch services, and telecommunications services

VOIP WITH ADAPTIVE RATE IN MULTI- TRANSMISSION RATE WIRELESS LANS

“Voice over Internet Protocol (VoIP)” is a popular communication technology that plays a vital role in term of cost reduction and flexibility. However, like any emerging technology, there are still some issues with VoIP, namely providing good Quality of Service (QoS), capacity consideration and providing security. This study focuses on the QoS issue of VoIP, specifically in “Wireless Local Area Networks (WLAN)”. IEEE 802.11 is the most popular standard of wireless LANs and it offers different transmission rates for wireless channels. Different transmission rates are associated with varying available bandwidth that shall influence the transmission of VoIP traffic

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

A unified data repository for rich communication services

Rich Communication Services (RCS) is a framework that defines a set of IP-based services for the delivery of multimedia communications to mobile network subscribers. The framework unifies a set of pre-existing communication services under a single name, and permits network operators to re-use investments in existing network infrastructure, especially the IP Multimedia Subsystem (IMS), which is a core part of a mobile network and also acts as a docking station for RCS services. RCS generates and utilises disparate subscriber data sets during execution, however, it lacks a harmonised repository for the management of such data sets, thus making it difficult to obtain a unified view of heterogeneous subscriber data. This thesis proposes the creation of a unified data repository for RCS which is based on the User Data Convergence (UDC) standard. The standard was proposed by the 3rd Generation Partnership Project (3GPP), a major telecommunications standardisation group. UDC provides an approach for consolidating subscriber data into a single logical repository without adversely affecting existing network infrastructure, such as the IMS. Thus, this thesis details the design and development of a prototypical implementation of a unified repository, named Converged Subscriber Data Repository (CSDR). It adopts a polyglot persistence model for the underlying data store and exposes heterogeneous data through the Open Data Protocol (OData), which is a candidate implementation of the Ud interface defined in the UDC architecture. With the introduction of polyglot persistence, multiple data stores can be used within the CSDR and disparate network data sources can access heterogeneous data sets using OData as a standard communications protocol. As the CSDR persistence model becomes more complex due to the inclusion of more storage technologies, polyglot persistence ensures a consistent conceptual view of these data sets through OData. Importantly, the CSDR prototype was integrated into a popular open-source implementation of the core part of an IMS network known as the Open IMS Core. The successful integration of the prototype demonstrates its ability to manage and expose a consolidated view of heterogeneous subscriber data, which are generated and used by different RCS services deployed within IMS

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