A Real-time Transport Protocol (RTP) Header Extension for Mixer-to-Client Audio Level Indication
RFC 6465
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RFC - Proposed Standard
(December 2011; No errata)
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Last updated |
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2015-10-14
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Replaces |
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draft-ivov-avt-slic
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IETF
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plain text
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bibtex
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WG state
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Submitted to IESG for Publication
Revised I-D Needed - Issue raised by WGLC
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Document shepherd |
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Keith Drage
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IESG |
IESG state |
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RFC 6465 (Proposed Standard)
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Consensus Boilerplate |
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Unknown
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Telechat date |
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Responsible AD |
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Gonzalo Camarillo
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IESG note |
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Keith Drage (keith.drage@alcatel-lucent.com) is the document shepherd.
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Send notices to |
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(None)
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Internet Engineering Task Force (IETF) E. Ivov, Ed.
Request for Comments: 6465 Jitsi
Category: Standards Track E. Marocco, Ed.
ISSN: 2070-1721 Telecom Italia
J. Lennox
Vidyo
December 2011
A Real-time Transport Protocol (RTP) Header Extension for
Mixer-to-Client Audio Level Indication
Abstract
This document describes a mechanism for RTP-level mixers in audio
conferences to deliver information about the audio level of
individual participants. Such audio level indicators are transported
in the same RTP packets as the audio data they pertain to.
Status of This Memo
This is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 5741.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc6465.
Copyright Notice
Copyright (c) 2011 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Ivov, et al. Standards Track [Page 1]
RFC 6465 Mixer-to-Client Audio Level Indication December 2011
Table of Contents
1. Introduction ....................................................2
2. Terminology .....................................................4
3. Protocol Operation ..............................................4
4. Audio Levels ....................................................5
5. Signaling Information ...........................................7
6. Security Considerations .........................................9
7. IANA Considerations ............................................10
8. Acknowledgments ................................................10
9. References .....................................................10
9.1. Normative References ......................................10
9.2. Informative References ....................................11
Appendix A. Reference Implementation ..............................12
A.1. AudioLevelCalculator.java .................................12
1. Introduction
"A Framework for Conferencing with the Session Initiation Protocol
(SIP)" [RFC4353] presents an overall architecture for multi-party
conferencing. Among others, the framework borrows from RTP [RFC3550]
and extends the concept of a mixer entity "responsible for combining
the media streams that make up a conference, and generating one or
more output streams that are delivered to recipients". Every
participant would hence receive, in a flat single stream, media
originating from all the others.
Using such centralized mixer-based architectures simplifies support
for conference calls on the client side, since they would hardly
differ from one-to-one conversations. However, the method also
introduces a few limitations. The flat nature of the streams that a
mixer would output and send to participants makes it difficult for
users to identify the original source of what they are hearing.
Mechanisms that allow the mixer to send to participants cues on
current speakers (e.g., the contributing source (CSRC) fields in RTP
[RFC3550]) only work for speaking/silent binary indications. There
are, however, a number of use cases where one would require more
detailed information. Possible examples include the presence of
background chat/noise/music/typing, someone breathing noisily in
their microphone, or other cases where identifying the source of the
disturbance would make it easy to remove it (e.g., by sending a
private IM to the concerned party asking them to mute their
microphone). A more advanced scenario could involve an intense
discussion between multiple participants that the user does not
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