The application of forward error correction techniques in wireless ATM

Bibliography: pages 116-121.The possibility of providing wireless access to an ATM network promises nomadic users a communication tool of unparalleled power and flexibility. Unfortunately, the physical realization of a wireless A TM system is fraught with technical difficulties, not the least of which is the problem of supporting a traditional ATM protocol over a non-benign wireless link. The objective of this thesis, titled "The Application of Forward Error Correction Techniques in Wireless ATM' is to examine the feasibility of using forward error correction techniques to improve the perceived channel characteristics to the extent that the channel becomes transparent to the higher layers and allows the use of an unmodified A TM protocol over the channel. In the course of the investigation that this dissertation describes, three possible error control strategies were suggested for implementation in a generic wireless channel. These schemes used a combination of forward error correction coding schemes, automatic repeat request schemes and interleavers to combat the impact of bit errors on the performance of the link. The following error control strategies were considered : 1. A stand alone fixed rate Reed-Solomon encoder/decoder with automatic repeat request. 2. A concatenated Reed-Solomon, convolution encoder/decoder with automatic request and convolution interleaving for the convolution codec. 3. A dynamic rate encoder/decoder using either a concatenated Reed-Solomon, convolution scheme or a Reed-Solomon only scheme with variable length Reed-Solomon words

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.

Statistical characterisation and stochastic modelling of 1-layer variable bit rate H.261 video codec traffic

The Integrated Services Digital Network(ISDN) is under re-design to provide flexibility which will ensure efficient network utilisation in the provision of broadband services. The main broadband services envisaged for provision on the Broadband ISDN(B-ISDN) are : Videophone; Videoconferencing; Television and High Definition TV. The B-ISDN will be a packet switched network where the packets(cells) will be transferred by the Asynchronous Transfer Mode(ATM) concept. Unlike voice and data services, the impact video services will have on the BISDN is unknown and hence loss of information is difficult to predict. Present videophone terminals are based on the CCITT H.261 Video Coding standard hence the picture quality is variable because video codec traffic is transmitted at a constant rate. To maintain a constant quality picture the codec output data must be transmitted at a variable rate or alternatively, for constant rate video codecs extra information must be made available to achieve constant picture quality. This latter technique is 2- Layer video coding where the first layer transmits at a constant rate and the second layer at a variable rate. The ATM B-ISDN promises constant picture quality video services, therefore to achieve this aim the impact variable rate video sources will have on the network must be determined by network simulation, thus variable rate video source models must be derived. To statistically characterise and stochastically model 1-Layer VBR(Variable Bit Rate) H.261 Video Codec traffic, here a videophone sequence is analysed by two alternative strategies : Talk-Listen and Motion Level. This analysis also found that 2-Layer H.261 Video Codec traffic can be stochastically modelled via a 1-Layer VBR H.261 Video Codec traffic model. Numerous hierarchical stochastic models with the ability to capture the statistical characteristics of long video sequences, in particular the short-term and long-term autocorrelations, are presented. One such model was simulated and the resulting simulated traffic was analysed to confirm the advantage hierarchical stochastic models have over non-hierarchical stochastic models in modelling video source traffic

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

Layer-based coding, smoothing, and scheduling of low-bit-rate video for teleconferencing over tactical ATM networks

This work investigates issues related to distribution of low bit rate video within the context of a teleconferencing application deployed over a tactical ATM network. The main objective is to develop mechanisms that support transmission of low bit rate video streams as a series of scalable layers that progressively improve quality. The hierarchical nature of the layered video stream is actively exploited along the transmission path from the sender to the recipients to facilitate transmission. A new layered coder design tailored to video teleconferencing in the tactical environment is proposed. Macroblocks selected due to scene motion are layered via subband decomposition using the fast Haar transform. A generalized layering scheme groups the subbands to form an arbitrary number of layers. As a layering scheme suitable for low motion video is unsuitable for static slides, the coder adapts the layering scheme to the video content. A suboptimal rate control mechanism that reduces the kappa dimensional rate distortion problem resulting from the use of multiple quantizers tailored to each layer to a 1 dimensional problem by creating a single rate distortion curve for the coder in terms of a suboptimal set of kappa dimensional quantizer vectors is investigated. Rate control is thus simplified into a table lookup of a codebook containing the suboptimal quantizer vectors. The rate controller is ideal for real time video and limits fluctuations in the bit stream with no corresponding visible fluctuations in perceptual quality. A traffic smoother prior to network entry is developed to increase queuing and scheduler efficiency. Three levels of smoothing are studied: frame, layer, and cell interarrival. Frame level smoothing occurs via rate control at the application. Interleaving and cell interarrival smoothing are accomplished using a leaky bucket mechanism inserted prior to the adaptation layer or within the adaptation layerhttp://www.archive.org/details/layerbasedcoding00parkLieutenant Commander, United States NavyApproved for public release; distribution is unlimited

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

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

Bandwidth Allocation By Pricing In ATM Networks

Admission control and bandwidth allocation are important issues in telecommunications networks, especially when there are random fluctuating demands for service and variations in the service rates. In the emerging broadband communications environment these services are likely to be offered via an ATM network. In order to make ATM future safe, methods for controlling the network should not be based on the characteristics of present services. We propose one bandwidth allocation method which has this property . Our proposed approach is based on pricing bandwidth to reflect network utilization, with users competing for resources according to their individual bandwidth valuations. The prices may be components of an actual tariff or they may be used as control signals, as in a private network. Simulation results show the improvement possible with our scheme versus a leaky bucket method in terms of cell loss probability, and confirm that a small queue with pricing can be efficient to multiplex heterogeneous sources