Towards scalable end-to-end QoS provision for VoIP applications

The growth of the Internet and the development of its new applications have increased the demand for providing a certain level of resource assurance and service support. The concept of ensuring quality of service (QoS) has been introduced in order to provide the support and assurance for these services. Different QoS mechanisms, such as integrated services (IntServ) and differentiated services (DiffServ), have been developed and introduced to provide different levels of QoS provision. However, IntServ can suffer from scalability issues that make it infeasible for large-scale network implementations. On the other hand, the aggregated-based per-flow technique of DiffServ does not provide such an end-to-end QoS guarantee. Recently, the IETF have proposed a new QoS architecture that implements IntServ over DiffServ in order to provide an end-to-end QoS for scalable networks. Hence, it became possible to provide and support a certain level of QoS for some delay sensitive and bandwidth-demanding applications such as voice over Internet Protocol (VoIP). With regard to VoIP applications, delay, jitter and packet loss are crucial issues that have to be taken into consideration for any VoIP system design and such parameters need a distinct level of QoS support

Service oriented networking for multimedia applications in broadband wireless networks

Extensive efforts have been focused on deploying broadband wireless networks. Providing mobile users with high speed network connectivity will let them run various multimedia applications on their wireless devices. In order to successfully deploy and operate broadband wireless networks, it is crucial to design efficient methods for supporting various services and applications in broadband wireless networks. Moreover, the existing access-oriented networking solutions are not able to fully address all the issues of supporting various applications with different quality of service requirements. Thus, service-oriented networking has been recently proposed and has gained much attention. This dissertation discusses the challenges and possible solutions for supporting multimedia applications in broadband wireless networks. The service requirements of different multimedia applications such as video streaming and Voice over IP (VoIP) are studied and some novel service-oriented networking solutions for supporting these applications in broadband wireless networks are proposed. The performance of these solutions is examined in WiMAX networks which are the promising technology for broadband wireless access in the near future. WiMAX networks are based on the IEEE 802.16 standards which have defined different Quality of Service (QoS) classes to support a broad range of applications with varying service requirements to mobile and stationary users. The growth of multimedia traffic that requires special quality of service from the network will impose new constraints on network designers who should wisely allocate the limited resources to users based on their required quality of service. An efficient resource management and network design depends upon gaining accurate information about the traffic profile of user applications. In this dissertation, the access level traffic profile of VoIP applications are studied first, and then a realistic distribution model for VoIP traffic is proposed. Based on this model, an algorithm to allocate resources for VoIP applications in WiMAX networks is investigated. Later, the challenges and possible solutions for transmitting MPEG video streams in wireless networks are discussed. The MPEG traffic model adopted by the WiMAX Forum is introduced and different application-oriented solutions for enhancing the performance of wireless networks with respect to MPEG video streaming applications are explained. An analytical framework to verify the performance of the proposed solutions is discoursed, and it is shown that the proposed solutions will improve the efficiency of VoIP applications and the quality of streaming applications over wireless networks. Finally, conclusions are drawn and future works are discussed

An optimized QoS scheme for IMS-NEMO in heterogeneous networks

The network mobility (NEMO) is proposed to support the mobility management when users move as a whole. In IP Multimedia Subsystem (IMS), the individual Quality of Service (QoS) control for NEMO results in excessive signaling cost. On the other hand, current QoS schemes have two drawbacks: unawareness of the heterogeneous wireless environment and inefficient utilization of the reserved bandwidth. To solve these problems, we present a novel heterogeneous bandwidth sharing (HBS) scheme for QoS provision under IMS-based NEMO (IMS-NEMO). The HBS scheme selects the most suitable access network for each session and enables the new coming non-real-time sessions to share bandwidth with the Variable Bit Rate (VBR) coded media flows. The modeling and simulation results demonstrate that the HBS can satisfy users' QoS requirement and obtain a more efficient use of the scarce wireless bandwidth

SZEROKOPASMOWE SATELITARNE SIECI DANYCH W KONTEKŚCIE DOSTĘPNYCH PROTOKOŁÓW I PLATFORM CYFROWYCH

Satellites are the transmission medium for providing connectivity and building global, continental, or regional networks around the world (satellite operators effectively use satellites to support Internet traffic), and point-to-point connections are also possible. In practical use, there are combinations of VSAT networks with terrestrial wireless extensions, allowing end users to increase the capabilities offered via satellite. This paper provides selected information on broadband satellite networks using VSAT technology, including available protocols and transmission platforms. The aim of the article is also to present the chosen technical aspects of satellite networks working with the usage of VSAT technology.Satelity stanowią medium transmisyjne dla zapewnienia łączności i budowy sieci globalnych, kontynentalnych czy regionalnych na całym świecie (operatorzy satelitarni efektywnie wykorzystują satelity do obsługi ruchu internetowego), przy czym możliwe są także połączenia typu punkt-punkt. W praktycznym użyciu są kombinacje sieci VSAT z bezprzewodowymi rozszerzeniami naziemnymi, co pozwala zwiększyć użytkownikom końcowym możliwości oferowane za pośrednictwem satelity. W artykule zamieszczono wybrane informacje na temat szerokopasmowych sieci satelitarnych z wykorzystaniem technologii VSAT, z uwzględnieniem dostępnych protokołów i platform transmisyjnych. Celem artykułu jest także przedstawienie wybranych aspektów technicznych sieci satelitarnych pracujących z wykorzystaniem technologii VSAT

Online Classification of RTC Traffic

Real-time communication (RTC) platforms have become increasingly popular in the last decade, together with the spread of broadband Internet access. They are nowadays a fundamental means for connecting people and supporting the economy, which relies more and more on forms of remote working. In this context, it is particularly important to act at the network level to ensure adequate Quality of Experience (QoE) to users, where proper traffic management policies are essential to prioritize RTC traffic. This, in turn, requires in-network devices to identify RTC streams and the type of content they carry. In this paper, we propose a machine learning-based application to classify, in real-time, the media streams generated by RTC applications encapsulated in Secure Real Time Protocol (SRTP) flows. Using carefully tuned features extracted from packet characteristics, we train a model to classify streams into an ample set of classes, including media type (audio/video), video quality and redundant streams. To validate our approach, we use traffic from more than 88 hours of multi-party meeting calls made using the Cisco Webex Teams application. We reach an overall accuracy of 97% with a light-weight decision tree model, which makes decisions using only 1 second of traffic

Modeling And Dynamic Resource Allocation For High Definition And Mobile Video Streams

Video streaming traffic has been surging in the last few years, which has resulted in an increase of its Internet traffic share on a daily basis. The importance of video streaming management has been emphasized with the advent of High Definition: HD) video streaming, as it requires by its nature more network resources. In this dissertation, we provide a better support for managing HD video traffic over both wireless and wired networks through several contributions. We present a simple, general and accurate video source model: Simplified Seasonal ARIMA Model: SAM). SAM is capable of capturing the statistical characteristics of video traces with less than 5% difference from their calculated optimal models. SAM is shown to be capable of modeling video traces encoded with MPEG-4 Part2, MPEG-4 Part10, and Scalable Video Codec: SVC) standards, using various encoding settings. We also provide a large and publicly-available collection of HD video traces along with their analyses results. These analyses include a full statistical analysis of HD videos, in addition to modeling, factor and cluster analyses. These results show that by using SAM, we can achieve up to 50% improvement in video traffic prediction accuracy. In addition, we developed several video tools, including an HD video traffic generator based on our model. Finally, to improve HD video streaming resource management, we present a SAM-based delay-guaranteed dynamic resource allocation: DRA) scheme that can provide up to 32.4% improvement in bandwidth utilization

Journal of Telecommunications and Information Technology, 2003, nr 4

kwartalni

Scalable Speech Coding for IP Networks

The emergence of Voice over Internet Protocol (VoIP) has posed new challenges to the development of speech codecs. The key issue of transporting real-time voice packet over IP networks is the lack of guarantee for reasonable speech quality due to packet delay or loss. Most of the widely used narrowband codecs depend on the Code Excited Linear Prediction (CELP) coding technique. The CELP technique utilizes the long-term prediction across the frame boundaries and therefore causes error propagation in the case of packet loss and need to transmit redundant information in order to mitigate the problem. The internet Low Bit-rate Codec (iLBC) employs the frame-independent coding and therefore inherently possesses high robustness to packet loss. However, the original iLBC lacks in some of the key features of speech codecs for IP networks: Rate flexibility, Scalability, and Wideband support. This dissertation presents novel scalable narrowband and wideband speech codecs for IP networks using the frame independent coding scheme based on the iLBC. The rate flexibility is added to the iLBC by employing the discrete cosine transform (DCT) and iii the scalable algebraic vector quantization (AVQ) and by allocating different number of bits to the AVQ. The bit-rate scalability is obtained by adding the enhancement layer to the core layer of the multi-rate iLBC. The enhancement layer encodes the weighted iLBC coding error in the modified DCT (MDCT) domain. The proposed wideband codec employs the bandwidth extension technique to extend the capabilities of existing narrowband codecs to provide wideband coding functionality. The wavelet transform is also used to further enhance the performance of the proposed codec. The performance evaluation results show that the proposed codec provides high robustness to packet loss and achieves equivalent or higher speech quality than state-of-the-art codecs under the clean channel condition

QoS in LEO satellite networks with multipacket reception

Dissertação apresentada para obtenção do Grau de Mestre em Engenharia Electrotécnica e de Computadores, pela Universidade Nova de Lisboa, Faculdade de Ciências e TecnologiaLow Earth Orbit (LEO) satellite networks can improve terrestrial wireless networks to allow global broadband services for Mobile Terminals (MT), regardless of the users' location. In this context, hybrid telecommunication systems combining satellites with Long Term Evolution (LTE) networks, like the LightSquared technology, are intended to provide ubiquitous high-speed services. This dissertation analyses the performance of a random access protocol that uses Hybrid Network-assisted Diversity Multiple Access (H-NDMA), for a LEO satellite system network, named by Satellite Random NDMA (SR-NDMA). The protocol also considers a Single Carrier-Frequency Domain Equalization (SC-FDE) scheme for the uplink transmission and a Multipacket Reception (MPR) receiver. In this scenario, the transmission of data packets between MTs and the Base Station (BS) is made through random access and schedule access slots, organized into super-frames with the duration of a Round Trip Time (RTT). A SR-NDMA simulator is implemented to measure the system performance in matters of throughput, energy consumption, system delay and also the protocol capacity to meet Quality of Service (QoS) requirements. A set of simulations tests were made with a random Poisson process tra c generation to validate the analytical model. The capacity to ful l the QoS requirements of a real-time tra c class was also tested.FCT/MEC: MPSat - PTDC/EEA-TEL/099074/2008, OPPORTUNISTIC CR - PTDC/EEA-TEL/115981/2009, Femtocells - PTDC/EEA-TEL/120666/2010 e ADIN - PTDC/EEI-TEL/2990/201

Non-intrusive identification of speech codecs in digital audio signals

Speech compression has become an integral component in all modern telecommunications networks. Numerous codecs have been developed and deployed for efficiently transmitting voice signals while maintaining high perceptual quality. Because of the diversity of speech codecs used by different carriers and networks, the ability to distinguish between different codecs lends itself to a wide variety of practical applications, including determining call provenance, enhancing network diagnostic metrics, and improving automated speaker recognition. However, few research efforts have attempted to provide a methodology for identifying amongst speech codecs in an audio signal. In this research, we demonstrate a novel approach for accurately determining the presence of several contemporary speech codecs in a non-intrusive manner. The methodology developed in this research demonstrates techniques for analyzing an audio signal such that the subtle noise components introduced by the codec processing are accentuated while most of the original speech content is eliminated. Using these techniques, an audio signal may be profiled to gather a set of values that effectively characterize the codec present in the signal. This procedure is first applied to a large data set of audio signals from known codecs to develop a set of trained profiles. Thereafter, signals from unknown codecs may be similarly profiled, and the profiles compared to each of the known training profiles in order to decide which codec is the best match with the unknown signal. Overall, the proposed strategy generates extremely favorable results, with codecs being identified correctly in nearly 95% of all test signals. In addition, the profiling process is shown to require a very short analysis length of less than 4 seconds of audio to achieve these results. Both the identification rate and the small analysis window represent dramatic improvements over previous efforts in speech codec identification