Video traffic modeling and delivery

Video is becoming a major component of the network traffic, and thus there has been a great interest to model video traffic. It is known that video traffic possesses short range dependence (SRD) and long range dependence (LRD) properties, which can drastically affect network performance. By decomposing a video sequence into three parts, according to its motion activity, Markov-modulated self-similar process model is first proposed to capture autocorrelation function (ACF) characteristics of MPEG video traffic. Furthermore, generalized Beta distribution is proposed to model the probability density functions (PDFs) of MPEG video traffic. It is observed that the ACF of MPEG video traffic fluctuates around three envelopes, reflecting the fact that different coding methods reduce the data dependency by different amount. This observation has led to a more accurate model, structurally modulated self-similar process model, which captures the ACF of the traffic, both SRD and LRD, by exploiting the MPEG structure. This model is subsequently simplified by simply modulating three self-similar processes, resulting in a much simpler model having the same accuracy as the structurally modulated self-similar process model. To justify the validity of the proposed models for video transmission, the cell loss ratios (CLRs) of a server with a limited buffer size driven by the empirical trace are compared to those driven by the proposed models. The differences are within one order, which are hardly achievable by other models, even for the case of JPEG video traffic. In the second part of this dissertation, two dynamic bandwidth allocation algorithms are proposed for pre-recorded and real-time video delivery, respectively. One is based on scene change identification, and the other is based on frame differences. The proposed algorithms can increase the bandwidth utilization by a factor of two to five, as compared to the constant bit rate (CBR) service using peak rate assignment

Performance analysis of an ATM network with multimedia traffic: a simulation study

Traffic and congestion control are important in enabling ATM networks to maintain the Quality of Service (QoS) required by end users. A Call Admission Control (CAC) strategy ensures that the network has sufficient resources available at the start of each call, but this does not prevent a traffic source from violating the negotiated contract. A policing strategy (User Parameter Control (UPC)) is also required to enforce the negotiated rates for a particular connection and to protect conforming users from network overload. The aim of this work is to investigate traffic policing and bandwidth management at the User to Network Interface (UNI). A policing function is proposed which is based on the leaky bucket (LB) which offers improved performance for both real time (RT) traffic such as speech and video and non-real time (non-RT) traffic, mainly data by taking into account the QoS requirements. A video cell in violation of the negotiated bit rate causes the remainder of the slice to be discarded. This 'tail clipping' provides protection for the decoder from damaged video slices. Speech cells are coded using a frequency domain coder, which places the most significant bits of a double speech sample into a high priority cell and the least significant bits into a high priority cell. In the case of congestion, the low priority cell can be discarded with little impact on the intelligibility of the received speech. However, data cells require loss-free delivery and are buffered rather than being discarded or tagged for subsequent deletion. This triple strategy is termed the super leaky bucket (SLB). Separate queues for RT and non-RT traffic, are also proposed at the multiplexer, with non pre-emptive priority service for RT traffic if the queue exceeds a predetermined threshold. If the RT queue continues to grow beyond a second threshold, then all low priority cells (mainly speech) are discarded. This scheme protects non-RT traffic from being tagged and subsequently discarded, by queueing the cells and also by throttling back non-RT sources during periods of congestion. It also prevents the RT cells from being delayed excessively in the multiplexer queue. A simulation model has been designed and implemented to test the proposal. Realistic sources have been incorporated into the model to simulate the types of traffic which could be expected on an ATM network. The results show that the S-LB outperforms the standard LB for video cells. The number of cells discarded and the resulting number of damaged video slices are significantly reduced. Dual queues with cyclic service at the multiplexer also reduce the delays experienced by RT cells. The QoS for all categories of traffic is preserved

Dynamic bandwidth allocation in ATM networks

Includes bibliographical references.This thesis investigates bandwidth allocation methodologies to transport new emerging bursty traffic types in ATM networks. However, existing ATM traffic management solutions are not readily able to handle the inevitable problem of congestion as result of the bursty traffic from the new emerging services. This research basically addresses bandwidth allocation issues for bursty traffic by proposing and exploring the concept of dynamic bandwidth allocation and comparing it to the traditional static bandwidth allocation schemes

Transmission of variable bit rate video over an Orwell ring

Asynchronous Transfer Mode (ATM) is fast emerging as the preferred information transfer technique for future Broadband Integrated Services Digital Networks (BISON), offering the advantages of both the simplicity of time division circuit switched techniques and the flexibility of packet switched techniques. ATM networks with their inherent rate flexibility offer new opportunities for the efficient transmission of real time Variable Bit Rate (VBR) services over such networks. Since most services are VBR in nature when efficiently coded, this could in turn lead to a more efficient utilisation of network resources through statistical multiplexing. Video communication is typical of such a service and could benefit significantly if supported with VBR video over ATM networks. [Continues.

Análisis de tráfico, modelación y simulación del tráfico de video en redes Wi-Fi 802.11e

Las WLAN (Wireless LAN) basadas en IEEE 802.11 se han vuelto las redes más populares en el acceso a los servicios de red y de banda ancha móvil/wireless a Internet. Y se espera que en los próximos años los servicios y aplicaciones de streaming o interactivos de video, alcancen hasta un 70% del tráfico total sobre dichas redes. En orden a satisfacer los requerimientos de QoS (Quality of Service - Calidad de Servicio) se introdujo la tecnología Wi-Fi EDCA (Enhanced Distributed Channel Access) 802.11e. Diferentes líneas de investigación estudian el comportamiento de la norma 802.11e, para mejorar sus mecanismos de diferenciación de tráfico a nivel MAC, en orden a gestionar y priorizar aún más eficientemente los diferentes perfiles de tráfico. En este trabajo de investigación se pretende efectuar capturas y mediciones de tráfico de video en redes Wi- Fi 802.11e, obtener su modelación y simulación usando Redes de Petri con el simulador Möbius (Universidad de Illinois). El objetivo es determinar por simulación, en escenarios previstos al efecto, el impacto cuantitativo y cualitativo que tendrá en la QoS de las redes Wi-Fi 802.11e, la mezcla de tráficos diversos con alta presencia de tráfico de streaming de video o tráfico de videoconferencias.Eje: Arquitectura, Redes y Sistemas OperativosRed de Universidades con Carreras en Informática (RedUNCI

Análisis de tráfico, modelación y simulación del tráfico de video en redes Wi-Fi 802.11e

Las WLAN (Wireless LAN) basadas en IEEE 802.11 se han vuelto las redes más populares en el acceso a los servicios de red y de banda ancha móvil/wireless a Internet. Y se espera que en los próximos años los servicios y aplicaciones de streaming o interactivos de video, alcancen hasta un 70% del tráfico total sobre dichas redes. En orden a satisfacer los requerimientos de QoS (Quality of Service - Calidad de Servicio) se introdujo la tecnología Wi-Fi EDCA (Enhanced Distributed Channel Access) 802.11e. Diferentes líneas de investigación estudian el comportamiento de la norma 802.11e, para mejorar sus mecanismos de diferenciación de tráfico a nivel MAC, en orden a gestionar y priorizar aún más eficientemente los diferentes perfiles de tráfico. En este trabajo de investigación se pretende efectuar capturas y mediciones de tráfico de video en redes Wi- Fi 802.11e, obtener su modelación y simulación usando Redes de Petri con el simulador Möbius (Universidad de Illinois). El objetivo es determinar por simulación, en escenarios previstos al efecto, el impacto cuantitativo y cualitativo que tendrá en la QoS de las redes Wi-Fi 802.11e, la mezcla de tráficos diversos con alta presencia de tráfico de streaming de video o tráfico de videoconferencias.Eje: Arquitectura, Redes y Sistemas OperativosRed de Universidades con Carreras en Informática (RedUNCI

Mixture block coding with progressive transmission in packet video. Appendix 1: Item 2

Video transmission will become an important part of future multimedia communication because of dramatically increasing user demand for video, and rapid evolution of coding algorithm and VLSI technology. Video transmission will be part of the broadband-integrated services digital network (B-ISDN). Asynchronous transfer mode (ATM) is a viable candidate for implementation of B-ISDN due to its inherent flexibility, service independency, and high performance. According to the characteristics of ATM, the information has to be coded into discrete cells which travel independently in the packet switching network. A practical realization of an ATM video codec called Mixture Block Coding with Progressive Transmission (MBCPT) is presented. This variable bit rate coding algorithm shows how a constant quality performance can be obtained according to user demand. Interactions between codec and network are emphasized including packetization, service synchronization, flow control, and error recovery. Finally, some simulation results based on MBCPT coding with error recovery are presented

Designing new network adaptation and ATM adaptation layers for interactive multimedia applications

Multimedia services, audiovisual applications composed of a combination of discrete and continuous data streams, will be a major part of the traffic flowing in the next generation of high speed networks. The cornerstones for multimedia are Asynchronous Transfer Mode (ATM) foreseen as the technology for the future Broadband Integrated Services Digital Network (B-ISDN) and audio and video compression algorithms such as MPEG-2 that reduce applications bandwidth requirements. Powerful desktop computers available today can integrate seamlessly the network access and the applications and thus bring the new multimedia services to home and business users. Among these services, those based on multipoint capabilities are expected to play a major role.    Interactive multimedia applications unlike traditional data transfer applications have stringent simultaneous requirements in terms of loss and delay jitter due to the nature of audiovisual information. In addition, such stream-based applications deliver data at a variable rate, in particular if a constant quality is required.    ATM, is able to integrate traffic of different nature within a single network creating interactions of different types that translate into delay jitter and loss. Traditional protocol layers do not have the appropriate mechanisms to provide the required network quality of service (QoS) for such interactive variable bit rate (VBR) multimedia multipoint applications. This lack of functionalities calls for the design of protocol layers with the appropriate functions to handle the stringent requirements of multimedia.    This thesis contributes to the solution of this problem by proposing new Network Adaptation and ATM Adaptation Layers for interactive VBR multimedia multipoint services.    The foundations to build these new multimedia protocol layers are twofold; the requirements of real-time multimedia applications and the nature of compressed audiovisual data.    On this basis, we present a set of design principles we consider as mandatory for a generic Multimedia AAL capable of handling interactive VBR multimedia applications in point-to-point as well as multicast environments. These design principles are then used as a foundation to derive a first set of functions for the MAAL, namely; cell loss detection via sequence numbering, packet delineation, dummy cell insertion and cell loss correction via RSE FEC techniques.    The proposed functions, partly based on some theoretical studies, are implemented and evaluated in a simulated environment. Performances are evaluated from the network point of view using classic metrics such as cell and packet loss. We also study the behavior of the cell loss process in order to evaluate the efficiency to be expected from the proposed cell loss correction method. We also discuss the difficulties to map network QoS parameters to user QoS parameters for multimedia applications and especially for video information. In order to present a complete performance evaluation that is also meaningful to the end-user, we make use of the MPQM metric to map the obtained network performance results to a user level. We evaluate the impact that cell loss has onto video and also the improvements achieved with the MAAL.    All performance results are compared to an equivalent implementation based on AAL5, as specified by the current ITU-T and ATM Forum standards.    An AAL has to be by definition generic. But to fully exploit the functionalities of the AAL layer, it is necessary to have a protocol layer that will efficiently interface the network and the applications. This role is devoted to the Network Adaptation Layer.    The network adaptation layer (NAL) we propose, aims at efficiently interface the applications to the underlying network to achieve a reliable but low overhead transmission of video streams. Since this requires an a priori knowledge of the information structure to be transmitted, we propose the NAL to be codec specific.    The NAL targets interactive multimedia applications. These applications share a set of common requirements independent of the encoding scheme used. This calls for the definition of a set of design principles that should be shared by any NAL even if the implementation of the functions themselves is codec specific. On the basis of the design principles, we derive the common functions that NALs have to perform which are mainly two; the segmentation and reassembly of data packets and the selective data protection.    On this basis, we develop an MPEG-2 specific NAL. It provides a perceptual syntactic information protection, the PSIP, which results in an intelligent and minimum overhead protection of video information. The PSIP takes advantage of the hierarchical organization of the compressed video data, common to the majority of the compression algorithms, to perform a selective data protection based on the perceptual relevance of the syntactic information.    The transmission over the combined NAL-MAAL layers shows significant improvement in terms of CLR and perceptual quality compared to equivalent transmissions over AAL5 with the same overhead.    The usage of the MPQM as a performance metric, which is one of the main contributions of this thesis, leads to a very interesting observation. The experimental results show that for unexpectedly high CLRs, the average perceptual quality remains close to the original value. The economical potential of such an observation is very important. Given that the data flows are VBR, it is possible to improve network utilization by means of statistical multiplexing. It is therefore possible to reduce the cost per communication by increasing the number of connections with a minimal loss in quality.    This conclusion could not have been derived without the combined usage of perceptual and network QoS metrics, which have been able to unveil the economic potential of perceptually protected streams.    The proposed concepts are finally tested in a real environment where a proof-of-concept implementation of the MAAL has shown a behavior close to the simulated results therefore validating the proposed multimedia protocol layers

Study and simulation of low rate video coding schemes

The semiannual report is included. Topics covered include communication, information science, data compression, remote sensing, color mapped images, robust coding scheme for packet video, recursively indexed differential pulse code modulation, image compression technique for use on token ring networks, and joint source/channel coder design

Concealment algorithms for networked video transmission systems

This thesis addresses the problem of cell loss when transmitting video data over an ATM network. Cell loss causes sections of an image to be lost or discarded in the interconnecting nodes between the transmitting and receiving locations. The method used to combat this problem is to use a technique called Error Concealment, where the lost sections of an image are replaced with approximations derived from the information in the surrounding areas to the error. This technique does not require any additional encoding, as used by Error Correction. Conventional techniques conceal from within the pixel domain, but require a large amount of processing (2N2 up to 20N2) where N is the dimension of an N×N square block. Also, previous work at Loughborough used Linear Interpolation in the transform domain, which required much less processing, to conceal the error. [Continues.