21,089 research outputs found
Delay aspects in Internet telephony
In this work, we address the transport of high quality voice over the Internet with a particular concern for delays. Transport of interactive audio over IP networks often suffers from packet loss and variations in the network delay (jitter). Forward Error Correction (FEC) mitigates the impact of packet loss at the expense of an increase of the end-to-end delay and the bit rate requirement of an audio source. Furthermore, adaptive playout buffer algorithms at the receiver compensate for jitter, but again this may come at the expense of additional delay. As a consequence, existing error control and playout adjustment schemes often have end-to-end delays exceeding 150 ms, which significantly impairs the perceived quality, while it would be more important to keep delay low and accept some small loss. We develop a joint playout buffer and FEC adjustment scheme for Internet Telephony that incorporates the impact of end-to-end delay on perceived audio quality. To this end, we take a utility function approach. We represent the perceived audio quality as a function of both the end-to-end delay and the distortion of the voice signal. We develop a joint rate/error/playout delay control algorithm which optimizes this measure of quality and is TCP-Friendly. It uses a channel model for both loss and delay. We validate our approach by simulation and show that (1) our scheme allows a source to increase its utility by avoiding increasing the playout delay when it is not really necessary and (2) it provides better quality than the adjustment schemes for playout and FEC that were previously published. We use this scheme in the framework of non-elevated services which allow applications to select a service class with reduced end-to-end delay at the expense of a higher loss rate. The tradeoff between delay and loss is not straightforward since audio sources may be forced to compensate the additional losses by more FEC and hence more delay. We show that the use of non-elevated services can lead to quality improvements, but that the choice of service depends on network conditions and on the importance that users attach to delay. Based on this observation, we propose an adaptive service choosing algorithm that allows audio sources to choose in real-time the service providing the highest audio quality. In addition, when used over the standard IP best effort service, an audio source should also control its rate in order to react to network congestion and to share the bandwidth in a fair way. Current congestion control mechanisms are based on packets (i.e., they aim to reduce or increase the number of packets sent per time interval to adjust to the current level of congestion in the network). However, voice is an inelastic traffic where packets are generated at regular intervals but packet size varies with the codec that is used. Therefore, standard congestion control is not directly applicable to this type of traffic. We present three alternative modifications to equation based congestion control protocols and evaluate them through mathematical analysis and network simulation
A QUIC Implementation for ns-3
Quick UDP Internet Connections (QUIC) is a recently proposed transport
protocol, currently being standardized by the Internet Engineering Task Force
(IETF). It aims at overcoming some of the shortcomings of TCP, while
maintaining the logic related to flow and congestion control, retransmissions
and acknowledgments. It supports multiplexing of multiple application layer
streams in the same connection, a more refined selective acknowledgment scheme,
and low-latency connection establishment. It also integrates cryptographic
functionalities in the protocol design. Moreover, QUIC is deployed at the
application layer, and encapsulates its packets in UDP datagrams. Given the
widespread interest in the new QUIC features, we believe that it is important
to provide to the networking community an implementation in a controllable and
isolated environment, i.e., a network simulator such as ns-3, in which it is
possible to test QUIC's performance and understand design choices and possible
limitations. Therefore, in this paper we present a native implementation of
QUIC for ns-3, describing the features we implemented, the main assumptions and
differences with respect to the QUIC Internet Drafts, and a set of examples.Comment: 8 pages, 4 figures. Please cite it as A. De Biasio, F. Chiariotti, M.
Polese, A. Zanella, M. Zorzi, "A QUIC Implementation for ns-3", Proceedings
of the Workshop on ns-3 (WNS3 '19), Firenze, Italy, 201
Comparative Study Of Congestion Control Techniques In High Speed Networks
Congestion in network occurs due to exceed in aggregate demand as compared to
the accessible capacity of the resources. Network congestion will increase as
network speed increases and new effective congestion control methods are
needed, especially to handle bursty traffic of todays very high speed networks.
Since late 90s numerous schemes i.e. [1]...[10] etc. have been proposed. This
paper concentrates on comparative study of the different congestion control
schemes based on some key performance metrics. An effort has been made to judge
the performance of Maximum Entropy (ME) based solution for a steady state
GE/GE/1/N censored queues with partial buffer sharing scheme against these key
performance metrics.Comment: 10 pages IEEE format, International Journal of Computer Science and
Information Security, IJCSIS November 2009, ISSN 1947 5500,
http://sites.google.com/site/ijcsis
A duality model of TCP and queue management algorithms
We propose a duality model of end-to-end congestion control and apply it to understanding the equilibrium properties of TCP and active queue management schemes. The basic idea is to regard source rates as primal variables and congestion measures as dual variables, and congestion control as a distributed primal-dual algorithm over the Internet to maximize aggregate utility subject to capacity constraints. The primal iteration is carried out by TCP algorithms such as Reno or Vegas, and the dual iteration is carried out by queue management algorithms such as DropTail, RED or REM. We present these algorithms and their generalizations, derive their utility functions, and study their interaction
Application-Oriented Flow Control: Fundamentals, Algorithms and Fairness
This paper is concerned with flow control and resource allocation problems in computer networks in which real-time applications may have hard quality of service (QoS) requirements. Recent optimal flow control approaches are unable to deal with these problems since QoS utility functions generally do not satisfy the strict concavity condition in real-time applications. For elastic traffic, we show that bandwidth allocations using the existing optimal flow control strategy can be quite unfair. If we consider different QoS requirements among network users, it may be undesirable to allocate bandwidth simply according to the traditional max-min fairness or proportional fairness. Instead, a network should have the ability to allocate bandwidth resources to various users, addressing their real utility requirements. For these reasons, this paper proposes a new distributed flow control algorithm for multiservice networks, where the application's utility is only assumed to be continuously increasing over the available bandwidth. In this, we show that the algorithm converges, and that at convergence, the utility achieved by each application is well balanced in a proportionally (or max-min) fair manner
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