Existing approaches for providing guaranteed services require routers to manage per ow states and perform per ow operations [9, 21]. Such a stateful network architecture is less scalable and robust than stateless network architectures like the original IP and the recently proposed Di serv [3]. However, services provided with current stateless solutions, Di serv included, have lower exibility, utilization, and/or assurance level as compared to the services that can be provided with per ow mechanisms. In this paper, we propose techniques that do not require per ow management (either control or data planes) at core routers, but can implement guaranteed services with levels of exibility, utilization, and assurance similar to those that can be provided with per ow mechanisms. In this way we can simultaneously achieve high quality of service, high scalability and robustness. The key technique we use is called Dynamic Packet State (DPS), which provides a lightweight and robust mechanism for routers to coordinate actions and implement distributed algorithms. We present an implementation of the proposed algorithms that has minimum incompatibility with IPv4.

In this article we present a framework for the emerging Internet quality of service (QoS). All the important components of this framework --- integrated services, RSVP, differentiated services, multiprotocol label switching (MPLS), and constraintbased routing --- are covered. We describe what integrated services and differentiated services are, how they can be implemented, and the problems they have. We then describe why MPLS and constraint-based routing have been introduced into this framework, how they differ from and relate to each other, and where they fit into the differentiated services architecture. Two likely service architectures are presented, and the end-to-end service deliveries in these two architectures are illustrated. We also compare ATM networks to router networks with differentiated services and MPLS. Putting all these together, we give the readers a grasp of the big picture of the emerging Internet QoS. 12 IEEE Network . March/April 1999 9 and differenti...

Transmission of multimedia streams imposesa minimum-bandwidth requirementon the path being used to ensureend-to-end Quality-ofService (QoS) guarantees. While any shortest-path algorithm can be used to select a feasible path, additional constraints that limit resource consumption and balance the network load are needed to achieve efficient resource utilization. We present a systematic evaluation of four routing algorithms that offer different tradeoffs between limiting the path hop count and balancing the network load. Our evaluation considers not only the call blocking rate but also the fairness to requests for different bandwidths, robustness to inaccurate routing information, and sensitivity to the routing information update frequency. It evaluates not only the performance of these algorithms for the sessions with bandwidth guarantees, but also their impact on the lower priority best-effort sessions. Our results show that a routing algorithm that gives preference to limiting the hop ...

This memo presents two recommendations to the Internet community concerning measures to improve and preserve Internet performance. It presents a strong recommendation for testing, standardization, and widespread deployment of active queue management in routers, to improve the performance of today's Internet. It also urges a concerted effort of research, measurement, and ultimate deployment of router mechanisms to protect the Internet from flows that are not sufficiently responsive to congestion notification. 1 INTRODUCTION The Internet protocol architecture is based on a connectionless end-to-end packet service using the IP protocol. The advantages of its connectionless design, flexibility and robustness, have been amply demonstrated. However, these advantages are not without cost: careful design is required to provide good service under heavy load. In fact, lack of attention to the dynamics of packet forwarding can result in severe service degradation or "Internet meltdown". T...

In packet audio applications, packets are buffered at a receiving site and their playout delayed in order to compensate for variable network delays. In this paper, we consider the problem of adaptively adjusting the playout delay in order to keep this delay as small as possible, while at the same time avoiding excessive “loss ” due to the arrival of packets at the receiver after their playout time has already passed. The contributions of this paper are twofold. First, given a trace of packet audio receptions at a receiver, we present efficient algorithms for computing a bound on the achievable performance of any playout delay adjustment algorithm. More precisely, we compute upper and lower bounds (which are shown to be tight for the range of loss and delay values of interest) on the optimum (minimum) average playout delay for a given number of packet losses (due to late arrivals) at the receiver for that trace. Second, we present a new adaptive delay adjustment algorithm that tracks the network delay of recently received packets and efficiently maintains delay percentile information. This information, together with a “delay spike ” detection algorithm based on (but extending) our earlier work [RKTS94], is used to dynamically adjust talkspurt playout delay. We show that this algorithm outperforms existing delay adjustment algorithms over a number of measured audio delay traces and performs close to the theoretical optimum over a range of parameter values of interest.

The growing use of multimedia communication applications with specific bandwidth and real-time delivery requirements has created the need for an Integrated Services Internet in which traditional best-effort datagram delivery can coexist with additional enhanced Quality Of Service(QOS) delivery classes. Such classes provide data flows with QOS commitments with regard to bandwidth, packet loss and delay through the reservation of network resources along the data path which can be done using the Resource ReSerVation Protocol (RSVP). This paper is a tutorial on how RSVP can be used by end-applications to ensure that they receive the end-to-end QOS that they require. Keywords Controlled-Load Service Data Flow Guaranteed Service Integrated Services (IS) One Pass with Advertising (OPWA) Quality of Service (QOS) ReSerVation setup Protocol (RSVP) This paper appeared in the May 1997 Issue of IEEE Communications magazine 2 1.0 Introduction The current Internet consists of a multitude of networ...

RSVP has been designed to support resource reservation in the Internet. However, it has two major problems: complexity and scalability. The former results in large message processing overhead at end systems and routers, and inefficient firewall processing at the edge of the network. The latter implies that in a backbone environment, the amount of bandwidth consumed by refresh messages and the storage space that is needed to support a large number of flows at a router are too large. We have developed a new reservation mechanism that simplifies the process of establishing reserved flows while preserving many unique features introduced by RSVP. Simplicity is measured in terms of control message processing, data packet processing, and user-level flexibility. Features such as robustness, advertising network service availability and resource sharing among multiple senders are also supported in the proposal. The proposed mechanism, YESSIR (YEt another Sender Session Internet Reservations) generates reservation requests by senders to reduce the processing overhead, builds on top of RTCP, uses soft state to maintain reservation states, supports shared reservation and associated flow merging and is compatible with the IETF Integrated Services models. YESSIR extends the all-or-nothing reservation model to support partial reservations that improve over the duration of the session. To address the scalability issue, we investigate the possibility of using YESSIR for per-stream reservation and RSVP for aggregate reservation.

Relaxed real-time services that do not provide guaranteed loss rates or delay bounds are of considerable interest in the Internet, since these services can achieve higher utilization than hard real-time services while still providing adequate service to adaptive real-time applications. Achieving this higher level of utilization depends on an admission control algorithm that does not rely on worst-case bounds to guide its admission decisions. Measurement-based admission control is one such approach, and several measurement-based admission control algorithms have been proposed in the literature. In this paper, we use simulation to compare the performance of several of these algorithms. We find that all of them achieve nearly the same utilization for a given packet loss rate, and that none of them are capable of accurately meeting loss targets. I. INTRODUCTION In an effort to better support applications with real-time constraints, several new per-flow packet delivery services have been ...

The traditional approach to implementing admission control, as exemplified by the Integrated Services proposal in the IETF, uses a signalling protocol to establish reservations at all routers along the path. While providing excellent quality-of-service, this approach has limited scalability because it requires routers to keep per-flow state and to process per-flow reservation messages. In an attempt to implement admission control without these scalability problems, several recent papers have proposed various forms of endpoint admission control. In these designs, the hosts (the endpoints) probe the network to detect the level of congestion; the host admits the flow only if the detected level of congestion is sufficiently low. This paper is devoted to the study of endpoint admission control. We first consider several architectural issues that guide (and constrain) the design of such systems. We then use simulations to evaluate the performance of endpoint admission control in various settings. The modest performance degradation between traditional router-based admission control and endpoint admission control suggests that a real-time service based on endpoint probing may be viable. 1.

This paper proposes a controlled-load service that provides a network state with bounded and well known worst-case behavior. The service is primarily developed for real-time applications. The full system for achieving quality of service to the application consists of an admission control combined with forward-error correction. The admission control is used to limit the packet-loss probability to a known value; the errorcontrol coding (i.e., FEC) is then used to raise the quality above the level enforced by the admission control. The basic idea for the admission control is that a host must probe the path to the receiver before sending actual data. It accepts the session if the probe is received with no or at most a moderate amount of loss. The performance evaluation shows clearly that the proposed scheme avoids network congestion and high packet losses even over short time scales.