Fairness for ABR multipoint-to-point connections

In multipoint-to-point connections, the traffic at the root (destination) is the combination of all traffic originating at the leaves. A crucial concern in the case of multiple senders is how to define fairness within a multicast group and among groups and point-to-point connections. Fairness definition can be complicated since the multipoint connection can have the same identifier (VPI/VCI) on each link, and senders might not be distinguishable in this case. Many rate allocation algorithms implicitly assume that there is only one sender in each VC, which does not hold for multipoint-to-point cases. We give various possibilities for defining fairness for multipoint connections, and show the tradeoffs involved. In addition, we show that ATM bandwidth allocation algorithms need to be adapted to give fair allocations for multipoint-to-point connections.Comment: Proceedings of SPIE 98, November 199

IP and ATM - a position paper

This paper gives a technical overview of different networking technologies, such as the Internet, ATM. It describes different approaches of how to run IP on top of an ATM network, and assesses their potential to be used as an integrated services network

IP and ATM integration: A New paradigm in multi-service internetworking

ATM is a widespread technology adopted by many to support advanced data communication, in particular efficient Internet services provision. The expected challenges of multimedia communication together with the increasing massive utilization of IP-based applications urgently require redesign of networking solutions in terms of both new functionalities and enhanced performance. However, the networking context is affected by so many changes, and to some extent chaotic growth, that any approach based on a structured and complex top-down architecture is unlikely to be applicable. Instead, an approach based on finding out the best match between realistic service requirements and the pragmatic, intelligent use of technical opportunities made available by the product market seems more appropriate. By following this approach, innovations and improvements can be introduced at different times, not necessarily complying with each other according to a coherent overall design. With the aim of pursuing feasible innovations in the different networking aspects, we look at both IP and ATM internetworking in order to investigating a few of the most crucial topics/ issues related to the IP and ATM integration perspective. This research would also address various means of internetworking the Internet Protocol (IP) and Asynchronous Transfer Mode (ATM) with an objective of identifying the best possible means of delivering Quality of Service (QoS) requirements for multi-service applications, exploiting the meritorious features that IP and ATM have to offer. Although IP and ATM often have been viewed as competitors, their complementary strengths and limitations from a natural alliance that combines the best aspects of both the technologies. For instance, one limitation of ATM networks has been the relatively large gap between the speed of the network paths and the control operations needed to configure those data paths to meet changing user needs. IP\u27s greatest strength, on the other hand, is the inherent flexibility and its capacity to adapt rapidly to changing conditions. These complementary strengths and limitations make it natural to combine IP with ATM to obtain the best that each has to offer. Over time many models and architectures have evolved for IP/ATM internetworking and they have impacted the fundamental thinking in internetworking IP and ATM. These technologies, architectures, models and implementations will be reviewed in greater detail in addressing possible issues in integrating these architectures s in a multi-service, enterprise network. The objective being to make recommendations as to the best means of interworking the two in exploiting the salient features of one another to provide a faster, reliable, scalable, robust, QoS aware network in the most economical manner. How IP will be carried over ATM when a commercial worldwide ATM network is deployed is not addressed and the details of such a network still remain in a state of flux to specify anything concrete. Our research findings culminated with a strong recommendation that the best model to adopt, in light of the impending integrated service requirements of future multi-service environments, is an ATM core with IP at the edges to realize the best of both technologies in delivering QoS guarantees in a seamless manner to any node in the enterprise

IP and ATM - current evolution for integrated services

Current and future applications make use of different technologies as voice, data, and video. Consequently network technologies need to support them. For many years, the ATM based Broadband-ISDN has generally been regarded as the ultimate networking technology, which can integrate voice, data, and video services. With the recent tremendous growth of the Internet and the reluctant deployment of public ATM networks, the future development of ATM seems to be less clear than it used to be. In the past IP provided (and was though to provide) only best effort services, thus, despite its world wide diffution, was not considered as a network solution for multimedia application. Currently many of the IETF working groups work on areas related to integrated services, and IP is also proposing itself as networking technology for supporting voice, data, and video services. This paper give a technical overview on the competing integrated services network solutions, such as IP, ATM and the different available and emerging technologies on how to run IP over ATM, and tries to identify their potential and shortcomings

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

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

Support For IP mobility and diversity in a broadband wireless access network

Broadband wireless access (BWA) network working at millimetre bands possesses the advantages of quick deployment, more flexibility, wide service coverage and cost efficiency. The range of services to be provided via the system includes broadband digital television, Internet data, telephony and videoconference. Apart from broadcast digital television, all traffic is carried in Internetworking Protocol (IP) format. Unfortunately the services of such a system are susceptible to impairment by buildings, vegetation, terrain and attenuation caused by rain, snow and sleet, etc. Accordingly the service availability and system performance can drop dramatically. In the worst case, the system will experience heavy packet loss and the services might be completely unavailable. An extended multiprotocol label switching (MPLS) network architecture is proposed in this thesis, which allows fast mobile IP access and diversity routing for traffic under fade condition. This supports nomadic access, reduced packet loss and improved service availability in BWA network during system outage. Also developed herein is a Diversity and Shadow Flow Merging Mechanism, which, besides sending a packet on its normal path, also duplicates the packet and sends it on a separate, diverted labelled path. The shadow flow merging mechanism is responsible for merging the normal flow and shadow flow together and delivering the merged packet to its destination. It is anticipated that the packet can be successfully delivered to the destination even if one path fails completely during the system outage. The protocol is tested on a general BWA network that is configured with Digital Video Broadcast (DVB) downlink and Multi-Frequency Time Division Multiplex Access (MF-TDMA) uplink equipments. The protocol’s ability of reducing packet loss and improving service availability, during the period of link failure, is verified. It is concluded that the protocol is effective in improving the service availability of BWA network.EThOS - Electronic Theses Online ServiceGBUnited Kingdo

Renegotiable VBR service

In this work we address the problem of supporting the QoS requirements for applications while efficiently allocating the network resources. We analyse this problem at the source node where the traffic profile is negotiated with the network and the traffic is shaped according to the contract. We advocate VBR renegotiation as an efficient mechanism to accommodate traffic fluctuations over the burst time-scale. This is in line with the Integrated Service of the IETF with the Resource reSerVation Protocol (RSVP), where the negotiated contract may be modified periodically. In this thesis, we analyse the fundamental elements needed for solving the VBR renegotiation. A source periodically estimates the needs based on: (1) its future traffic, (2) cost objective, (3) information from the past. The issues of this estimation are twofold: future traffic prediction given a prediction, the optimal change. In the case of a CBR specification the optimisation problem is trivial. But with a VBR specification this problem is complex because of the multidimensionality of the VBR traffic descriptor and the non zero condition of the system at the times where the parameter set is changed. We, therefore, focus on the problem of finding the optimal change for sources with pre-recorded or classified traffic. The prediction of the future traffic is out of the scope of this thesis. Traditional existing models are not suitable for modelling this dynamic situation because they do not take into account the non-zero conditions at the transient moments. To address the shortfalls of the traditional approaches, a new class of shapers, the time varying leaky bucket shaper class, has been introduced and characterised by network calculus. To our knowledge, this is the first model that takes into account non-zero conditions at the transient time. This innovative result forms the basis of Renegotiable VBR Service (RVBR). The application of our RVBR mathematical model to the initial problem of supporting the applications' QoS requirements while efficiently allocating the network resources results in simple, efficient algorithms. Through simulation, we first compare RVBR service versus VBR service and versus renegotiable CBR service. We show that RVBR service provides significant advantages in terms of resource costs and resource utilisation. Then, we illustrate that when the service assumes zero conditions at the transient time, the source could potentially experience losses in the case of policing because of the mismatch between the assumed bucket and buffer level and the policed bucket and buffer level. As an example of RVBR service usage, we describe the simulation of RVBR service in a scenario where a sender transmits a MPEG2 video over a network using RSVP reservation protocol with Controlled-Load service. We also describe the implementation design of a Video on Demand application, which is the first example of an RVBR-enabled application. The simulation and experimentation results lead us to believe that RVBR service provides an adequate service (in terms of QoS guaranteed and of efficient resource allocation) to sources with pre-recorded or classified traffic

ATM PNNI Interfacing Issues with MPLS Networking

School of Electrical and Computer Engineerin