Home  |   Login  |   Logout  |   Access Information  |   Alerts  |   Purchase History  |   Cart  |   Sitemap  |   Help   
 
Login
BROWSE SEARCH IEEE XPLORE GUIDE SUPPORT
Article Information

Simulation of FEC-based error control for packet audio on theInternet
Podolsky, M.; Romer, C.; McCanne, S.
INFOCOM apos;98. Seventeenth Annual Joint Conference of the IEEE Computer and Communications Societies. Proceedings. IEEE
Volume 2, Issue , 29 Mar-2 Apr 1998 Page(s):505 - 515 vol.2
Digital Object Identifier   10.1109/INFCOM.1998.665068
Summary:Real-time audio over a best-effort network, such as the Internet, frequently suffers from packet loss. To mitigate the impact of such packet loss, several research efforts and implementation studies advocate the use of forward error correction (FEC) coding. Although these prior works have pioneered promising and novel applications of FEC to Internet audio, they do not definitively demonstrate the advantages of FEC because they do not evaluate aggregate performance that results from multiplexing many like flows. We build on previous landmark works with a systematic study of FEC for packet audio that characterizes the aggregate performance across all audio sources in the network. We refine the novel but ad hoc coding techniques proposed by Hardman, Sasse, Handley and Watson (see Proc. INET, 1995) into a formal framework that we call “signal processing-based FEC” (SFEC) and use our framework to more rigorously evaluate the relative merits of this approach. Through extensive simulation, we evaluate the “scalability” of SFEC for packet audio-i.e., the ability for a coding algorithm to improve aggregate performance when used by all sources in the network-and find that optimal signal quality is achieved when sources react to network congestion not by blindly adding FEC, but rather by adding FEC in a controlled fashion that simultaneously constrains the source-coding rate. As a result, packet loss is mitigated without introducing more congestion, thus admitting a more scalable and effective approach than successively adding redundancy to a constant bit-rate source. While this result may seem intuitive, it has not been previously suggested in the context of Internet audio, and until now, has not been systematically studied

» View citation and abstract

IEEE Members

Log in by entering your IEEE Web Account Username and Password.

IEEE Communications Society members: If you subscribe to the IEEE Electronic Periodicals Package or IEEE Electronic Periodicals Package Plus, you must access your subscription at www.comsoc.org.

Users at Subscribing Institutions

Check with your librarian, information professional, or system manager to determine if you need to log in. Please complete the online Technical Support Form if you need assistance.

Already Purchased This Article?

Select the Purchase History link to access the document. You will have 5 Days after purchase to access the Full Text PDF. Please complete the online Technical Support Form if you need assistance.

Guests

• Search and access Abstract records free of charge
Register for table of contents alerts
• Purchase Full Text PDF documents

» Learn more about subscription options or how to become an IEEE Member.

You are not logged in.
LOGIN
Username
Password
GO
» Forgot your password?
Please remember to log out when you have finished your session.
You must log in to access:
• Advanced or Author Search
• CrossRef Search
• AbstractPlus Records
• Full Text PDF
• Full Text HTML
Access this document
» Buy this document now
» Learn more about
» Learn more about
   purchasing articles
   and standards
Learn more about IEEE Subscriptions
Indexed by IEE Inspec
© Copyright 2010 IEEE – All Rights Reserved