Capacity reservation for multimedia traffic
TL;DR: A new scheduling algorithm for multimedia traffic using capacity reservation appears to give a noticeably improved quality of service to delay-sensitive traffic.
Abstract: We introduce a new scheduling algorithm for multimedia traffic using capacity reservation. We compare it with other algorithms in the literature. It has been implemented and its worst-case performance has been analysed. It appears to give a noticeably improved quality of service to delay-sensitive traffic.
Citations
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26 Mar 2000
TL;DR: A controlled load service that provides a network state with bounded and well known worst-case behavior and avoids network congestion and high packet losses even over short time scales is proposed.
Abstract: 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 error-control 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.
215 citations
Cites background from "Capacity reservation for multimedia..."
...Unused capacity is fully available to best-effort traffic so there is no waste incurred by the partition (see Fig. 1). The capacity limit for the service cannot be exceeded and may be provided by non-work conservative scheduling [ 16 ], which is suitable for real-time transmission with low tolerance to delay variations....
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01 Jan 2005
TL;DR: This work presents a new measurement-based admission control (MBAC) scheme that uses measurements of aggregate bandwidth only, without keeping the state of any per-flow information, and shows that the approach compares favorably against other admission control schemes found in the literature.
Abstract: In differentiated services (DiffServ) domains, where services are provisioned on a per-class basis, admission control is an essential control factor in order to ensure that congestion is avoided and that the quality of service (QoS) requirements of individual flows are met. We consider traffic-engineered and provisioned IP differentiated services domains able to support real-time traffic. We present a new measurement-based admission control (MBAC) scheme that uses measurements of aggregate bandwidth only, without keeping the state of any per-flow information. In our scheme there is no assumption made on the nature of the traffic characteristics of the real-time sources, which can be of any heterogeneous nature. Through simulations we show that the admission control scheme is robust with respect to traffic heterogeneity and measurement errors. We also show that our approach compares favorably against other admission control schemes found in the literature.
28 citations
Cites background from "Capacity reservation for multimedia..."
...Furthermore, the deployment of non-work conserving scheduling in routers for the real-time traffic class can be beneficial for controlling jitter [8]....
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TL;DR: This paper describes a media synchronization scheme for wireless networks, and investigates four packet scheduling algorithms: First-In-First-Out (FIFO), Highest-Priority-First (PQ), Weighted Fair-Queuing (WFQ) and Round-Robin (RR).
Abstract: Wireless multimedia synchronization is concerned with distributed multimedia packets such as video, audio, text and graphics being played-out onto the mobile clients via a base station (BS) that services the mobile client with the multimedia packets. Our focus is on improving the Quality of Service (QoS) of the mobile client's on-time-arrival of distributed multimedia packets through network multimedia synchronization. We describe a media synchronization scheme for wireless networks, and we investigate the multimedia packet scheduling algorithms at the base station to accomplish our goal. In this paper, we extend the media synchronization algorithm by investigating four packet scheduling algorithms: First-In-First-Out (FIFO), Highest-Priority-First (PQ), Weighted Fair-Queuing (WFQ) and Round-Robin (RR). We analyze the effect of the four packet scheduling algorithms in terms of multimedia packet delivery time and the delay between concurrent multimedia data streams. We show that the play-out of multimedia units on the mobile clients by the base station plays an important role in enhancing the mobile client's quality of service in terms of intra-stream synchronization and inter-stream synchronization. Our results show that the Round-Robin (RR) packet scheduling algorithm is, by far, the best of the four packet scheduling algorithms in terms of mobile client buffer usage. We analyze the four packet scheduling algorithms and make a correlation between play-out of multimedia packets, by the base station, onto the mobile clients and wireless network multimedia synchronization. We clarify the meaning of buffer usage, buffer overflow, buffer underflow, message complexity and multimedia packet delay in terms of synchronization between distributed multimedia servers, base stations and mobile clients.
25 citations
01 Jan 1999
TL;DR: The goal is to support interactive video with only small modifications to network routers and control systems by regulation of the load through sender-based admission control, and using end-to-end probes of the network state.
Abstract: One of the most interesting improvements of the Internet today is the provisioning of services for interactive audio-visual applications. Such applications have quality requirements which place limits on both transfer delays and information losses. Our goal is to support interactive video with only small modifications to network routers and control systems. The approach we favor is to ensure the quality for a session by forward-error correction. The code strength is dynamically tuned to meet a user’s quality expectation, given the experienced loss process of the transfer. To make the tuning feasible, the network state must be predictable. We accomplish this by regulation of the load through sender-based admission control. It uses end-to-end probes of the network state and a self-imposed blocking if then sender determines that a sensible transfer cannot be made. The paper outlines this procedure along with a review of source coding considerations and a description of tunable error-control coding.
19 citations
TL;DR: This work presents the analysis of this reservation discipline based on the use of generating functions and provides results for the mean value, variance and tail behaviour of the delay experienced by both the delay-sensitive and the best-effort traffic.
18 citations
References
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IBM1
TL;DR: Worst-case bounds on delay and backlog are derived for leaky bucket constrained sessions in arbitrary topology networks of generalized processor sharing (GPS) servers and the effectiveness of PGPS in guaranteeing worst-case session delay is demonstrated under certain assignments.
Abstract: Worst-case bounds on delay and backlog are derived for leaky bucket constrained sessions in arbitrary topology networks of generalized processor sharing (GPS) servers. The inherent flexibility of the service discipline is exploited to analyze broad classes of networks. When only a subset of the sessions are leaky bucket constrained, we give succinct per-session bounds that are independent of the behavior of the other sessions and also of the network topology. However, these bounds are only shown to hold for each session that is guaranteed a backlog clearing rate that exceeds the token arrival rate of its leaky bucket. A much broader class of networks, called consistent relative session treatment (CRST) networks is analyzed for the case in which all of the sessions are leaky bucket constrained. First, an algorithm is presented that characterizes the internal traffic in terms of average rate and burstiness, and it is shown that all CRST networks are stable. Next, a method is presented that yields bounds on session delay and backlog given this internal traffic characterization. The links of a route are treated collectively, yielding tighter bounds than those that result from adding the worst-case delays (backlogs) at each of the links in the route. The bounds on delay and backlog for each session are efficiently computed from a universal service curve, and it is shown that these bounds are achieved by "staggered" greedy regimes when an independent sessions relaxation holds. Propagation delay is also incorporated into the model. Finally, the analysis of arbitrary topology GPS networks is related to Packet GPS networks (PGPS). The PGPS scheme was first proposed by Demers, Shenker and Keshav (1991) under the name of weighted fair queueing. For small packet sizes, the behavior of the two schemes is seen to be virtually identical, and the effectiveness of PGPS in guaranteeing worst-case session delay is demonstrated under certain assignments. >
3,967 citations
TL;DR: In this article, a fair gateway queueing algorithm based on an earlier suggestion by Nagle is proposed to control congestion in datagram networks, based on the idea of fair queueing.
Abstract: We discuss gateway queueing algorithms and their role in controlling congestion in datagram networks. A fair queueing algorithm, based on an earlier suggestion by Nagle, is proposed. Analysis and s...
2,639 citations
PARC1
TL;DR: It is found that fair queueing provides several important advantages over the usual first-come-first-serve queueing algorithm: fair allocation of bandwidth, lower delay for sources using less than their full share of bandwidth and protection from ill-behaved sources.
Abstract: We discuss gateway queueing algorithms and their role in controlling congestion in datagram networks. A fair queueing algorithm, based on an earlier suggestion by Nagle, is proposed. Analysis and simulations are used to compare this algorithm to other congestion control schemes. We find that fair queueing provides several important advantages over the usual first-come-first-serve queueing algorithm: fair allocation of bandwidth, lower delay for sources using less than their full share of bandwidth, and protection from ill-behaved sources.
2,480 citations
TL;DR: A calculus is developed for obtaining bounds on delay and buffering requirements in a communication network operating in a packet switched mode under a fixed routing strategy, and burstiness constraints satisfied by the traffic that exits the element are derived.
Abstract: A calculus is developed for obtaining bounds on delay and buffering requirements in a communication network operating in a packet switched mode under a fixed routing strategy. The theory developed is different from traditional approaches to analyzing delay because the model used to describe the entry of data into the network is nonprobabilistic. It is supposed that the data stream entered into the network by any given user satisfies burstiness constraints. A data stream is said to satisfy a burstiness constraint if the quantity of data from the stream contained in any interval of time is less than a value that depends on the length of the interval. Several network elements are defined that can be used as building blocks to model a wide variety of communication networks. Each type of network element is analyzed by assuming that the traffic entering it satisfies bursting constraints. Under this assumption, bounds are obtained on delay and buffering requirements for the network element; burstiness constraints satisfied by the traffic that exits the element are derived. >
2,049 citations
"Capacity reservation for multimedia..." refers methods in this paper
...…simplied call acceptance procedure may allow suciently fast connection establishment t o handle transfer of bulk data, such as images. orst-case erfor ance anal sis The performance of this algorithm with respect to a particular trac stream was analysed using techniques similar to those in [1] [2]....
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01 Oct 1995
TL;DR: Several service disciplines that are proposed in the literature to provide per-connection end-to-end performance guarantees in packet-switching networks are surveyed and a general framework for studying and comparing these disciplines is presented.
Abstract: While today's computer networks support only best-effort service, future packet-switching integrated-services networks will have to support real-time communication services that allow clients to transport information with performance guarantees expressed in terms of delay, delay jitter, throughput, and loss rate. An important issue in providing guaranteed performance service is the choice of the packet service discipline at switching nodes. In this paper, we survey several service disciplines that are proposed in the literature to provide per-connection end-to-end performance guarantees in packet-switching networks. We describe their mechanisms, their similarities and differences and the performance guarantees they can provide. Various issues and tradeoffs in designing service disciplines for guaranteed performance service are discussed, and a general framework for studying and comparing these disciplines are presented. >
1,226 citations