Grid technologies enable large-scale sharing of resources within formal or informal consortia of individuals and/or institutions: what are sometimes called virtual organizations. In these settings, the discovery, characterization, and monitoring of resources, services, and computations are challenging problems due to the considerable diversity, large numbers, dynamic behavior, and geographical distribution of the entities in which a user might be interested. Consequently, information services are a vital part of any Grid software infrastructure, providing fundamental mechanisms for discovery and monitoring, and hence for planning and adapting application behavior. We present here an information services architecture that addresses performance, security, scalability, and robustness requirements. Our architecture defines simple low-level enquiry and registration protocols that make it easy to incorporate individual entities into various information structures, such as aggregate directories that support a variety of different query languages and discovery strategies. These protocols can also be combined with other Grid protocols to construct additional higher-level services and capabilities such as brokering, monitoring, fault detection, and troubleshooting. Our architecture has been implemented as MDS-2, which forms part of the Globus Grid toolkit and has been widely deployed and applied.

Existing multicast routing mechanisms were intended for use within regions where a group is widely represented or bandwidth is universally plentiful. When group members, and senders to those group members, are distributed sparsely across a wide area, these schemes are not efficient; data packets or membership report information are occasionally sent over many links that do not lead to receivers or senders, respectively. Wehave developed a multicast routing architecture that efficiently establishes distribution trees across wide area internets, where many groups will be sparsely represented. Efficiency is measured in terms of the state, control message processing, and data packet processing, required across the entire network in order to deliver data packets to the members of the group. Our Protocol Independent Multicast (PIM) architecture: (a) maintains the traditional IP multicast service model of receiver-initiated membership; (b) can be configured to adapt to different multicast group and network characteristics; (c) is not dependent on a specific unicast routing protocol; and (d) uses soft-state mechanisms to adapt to underlying network conditions and group dynamics. The robustness, flexibility, and scaling properties of this architecture make it well suited to large heterogeneous inter-networks.

The deployment of IP Multicast has fostered the development of a suite of applications, collectively known as the MBone tools, for real-time multimedia conferencingover the Internet. Two of these tools --- nv from Xerox PARC and ivs from INRIA --- provide video transmission using softwarebased codecs. We describe a new video tool, vic, that extends the groundbreaking work of nv and ivs with a more flexible system architecture. This flexibility is characterized by network layer independence, support for hardware-based codecs, a conference coordination model, an extensible user interface, and support for diverse compression algorithms. We also propose a novel compression scheme called "IntraH. 261". Created as a hybrid of the nv and ivs codecs, IntraH. 261 provides a factor of 2-3 improvement in compression gain over the nv encoder (6 dB of PSNR) as well as a substantial improvement in run-time performance over the ivs H.261 coder. KEYWORDS Conferencing protocols; digital video; image ...

Routers classify packets to determine which flow they belong to, and to decide what service they should receive. Classification may, in general, be based on an arbitrary number of fields in the packet header. Performing classification quickly on an arbitrary number of fields is known to be difficult, and has poor worst-case performance. In this paper, we consider a number of classifiers taken from real networks. We find that the classifiers contain considerable structure and redundancy that can be exploited by the classification algorithm. In particular, we find that a simple multi-stage classification algorithm, called RFC (recursive flow classification), can classify 30 million packets per second in pipelined hardware, or one million packets per second in software.

Retransmission based error control for large scale multicast applications is difficult because of two main problems: request implosion and lack of local recovery. Existing schemes (SRM, RMTP, TMTP, LBRRM) have good solutions to request implosion, but only approximate solutions (e.g., based on scoped multicast) for the local recovery problem. Our scheme achieves finer grain fault recovery by exploiting new forwarding services that allow us to create a dynamic hierarchy of receivers. We use a new paradigm, where routers provide a more refined form of multicasting (that may be useful to other applications), that enables local recovery. The new services, however, are simple to implement and do not require routers to examine or store application packets; hence, they do not violate layering. Besides providing good local recovery, our scheme integrates well with the current IP model, has small recovery latencies (it requires no back-off delays), produces fewer duplicates than other schemes, a...

In recent years, the "Internet Multicast Backbone," or MBone, has risen from a small, research curiosity to a largescale and widely used communications infrastructure. A driving force behind this growth was the development of multipoint audio, video, and shared whiteboard conferencing applications. Because these real-time media are transmitted at a uniform rate to all of the receivers in the network, a source must either run at the bottleneck rate or overload portions of its multicast distribution tree. We overcome this limitation by moving the burden of rate adaptation from the source to the receivers with a scheme we call receiver-driven layered multicast, or RLM. In RLM, a source distributes a hierarchical signal by striping the different layers across multiple multicast groups, and receivers adjust their reception rate by simply joining and leaving multicast groups. In this paper, we describe a layered video compression algorithm which, when combined with RLM, provides a comprehensive solution for scalable multicast video transmission in heterogeneous networks. In addition to a layered representation, our coder has low complexity (admitting an efficient software implementation) and high loss resilience (admitting robust operation in loosely controlled environments like the Internet) . Even with these constraints, our hybrid DCT/wavelet-based coder exhibits good compression performance. It outperforms all publicly available Internet video codecs while maintaining comparable run-time performance. We have implemented our coder in a "real" application---the UCB/LBL videoconferencing tool vic. Unlike previous work on layered video compression and transmission, we have built a fully operational system that is currently being deployed on a very large scale over the MBone.

This memo describes extensions to the OSPF protocol to support QoS routes. The focus of the document is on the algorithms used to compute QoS routes and on the necessary modifications to OSPF to support this function, e.g., the information needed, its format, how it is distributed, and how it is used by the QoS path selection process. Aspects related to how QoS routes are established and managed are also briefly discussed, but the development of detailed specifications is left for further study. The goal of this document is to identify a framework and possible approaches to allow deployment of QoS routing capabilities with the minimum possible impact to the existing routing infrastructure. Guerin,Orda,Williams Expires 10 May 1997 [Page i] Internet Draft QoS Routing Mechanisms 5 November 1996 Contents Status of This Memo i Abstract i 1. Introduction 1 1.1. Overall Framework . . . . . . . . . . . . . . . . . . . . 1 1.2. Simplifying Assumptions . . . . . . . . . . . . . . . . . 2 2. Pa...

Recent studies provide evidence that Quality of Service (QoS) routing can provide increased network utilization compared to routing that is not sensitive to QoS requirements of traffic. However, there are still strong concerns about the increased cost of &OS routing, both in terms of more complex and frequent computations and increased routing protocol over-head. The main goals of this paper are to study these two cost components, and propose solutions that achieve good routing performance with reduced processing cost. First, we identify the parameters that determine the protocol traffic overhead, namely (a) policy for triggering updates, (b) sen-sitivity of this policy, and (c) clamp down timers that limit the rate of updates. Using simulation, we study the relative significance of these factors and investigate the relationship between routing performance and the amount of update traf-fic. In addition, we explore a range of design options to reduce the processing cost of QoS routing algorithms, and study their effect on routing performance. Based on the con-clusions of these studies, we develop extensions to the basic QoS routing, that can achieve good routing performance with limited update generation rates. The paper also ad-dresses the impact on the results of a number of secondary factors such as topology, high level admission control, and characteristics of network traffic.

In just a few years the "Internet Multicast Backbone", or MBone, has risen from a small, research curiosity to a large scale and widely used communications infrastructure. A driving force behind this growth was our development of multipoint audio, video, and shared whiteboard conferencing applications that are now used daily by the large and growing MBone community. Because these real-time media are transmitted at a uniform rate to all the receivers in the network, the source must either run below the bottleneck rate or overload portions of the multicast distribution tree. In this dissertation, we propose a solution to this problem by moving the burden of rate-adaptation from the source to the receivers with a scheme we call Receiver-driven Layered Multicast, or RLM. In RLM, a source distr...

Many distributed multimedia applications have the ability to adapt to fluctuations in the network conditions. By adjusting temporal and spatial quality to available bandwidth, or manipulating the playout time of continuous media in response to variations in delay, multimedia flows can keep an acceptable QoS level at the end systems. In this paper, we present a new scheme called the loss-delay based adjustment algorithm (LDA) for adapting the transmission rate of multimedia applications to the congestion level of the network. The LDA algorithm was designed to reduce losses and improve utilization in a TCP-friendly way that avoids starving competing TCP connections. It relies on the end-toend Real Time Transport Protocol (RTP) for feedback information. In addition, we enhanced RTP with functionalities for determining the bottleneck bandwidth of a connection. The bottleneck bandwidth is then used for dynamically determining the adaptation parameters. Simulations and measurements of the LD...