Internet DRAFT - draft-ietf-clue-rtp-mapping
draft-ietf-clue-rtp-mapping
CLUE WG R. Even
Internet-Draft Huawei Technologies
Intended status: Standards Track J. Lennox
Expires: August 31, 2017 Vidyo
February 27, 2017
Mapping RTP streams to CLUE Media Captures
draft-ietf-clue-rtp-mapping-14.txt
Abstract
This document describes how the Real Time transport Protocol (RTP) is
used in the context of the CLUE protocol (ControLling mUltiple
streams for tElepresence). It also describes the mechanisms and
recommended practice for mapping RTP media streams defined in Session
Description Protocol (SDP) to CLUE Media Captures and defines a new
RTP header extension (CaptureId).
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on August 31, 2017.
Copyright Notice
Copyright (c) 2017 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
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. RTP topologies for CLUE . . . . . . . . . . . . . . . . . . . 3
4. Mapping CLUE Capture Encodings to RTP streams . . . . . . . . 4
5. MCC Constituent CaptureID definition . . . . . . . . . . . . 5
5.1. RTCP CaptureID SDES Item . . . . . . . . . . . . . . . . 5
5.2. RTP Header Extension . . . . . . . . . . . . . . . . . . 6
6. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 6
7. Communication Security . . . . . . . . . . . . . . . . . . . 7
8. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 8
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8
10. Security Considerations . . . . . . . . . . . . . . . . . . . 8
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 10
11.1. Normative References . . . . . . . . . . . . . . . . . . 10
11.2. Informative References . . . . . . . . . . . . . . . . . 11
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 13
1. Introduction
Telepresence systems can send and receive multiple media streams.
The CLUE framework [I-D.ietf-clue-framework] defines Media Captures
(MC) as a source of Media, from one or more Capture Devices. A Media
Capture may also be constructed from other Media streams. A middle
box can express conceptual Media Captures that it constructs from
Media streams it receives. A Multiple Content Capture (MCC) is a
special Media Capture composed of multiple Media Captures.
SIP Offer/Answer [RFC3264] uses SDP [RFC4566] to describe the
RTP[RFC3550] media streams. Each RTP stream has a unique
Synchronization Source (SSRC) within its RTP session. The content of
the RTP stream is created by an encoder in the endpoint. This may be
an original content from a camera or a content created by an
intermediary device like an MCU (Multipoint Control Unit).
This document makes recommendations for the CLUE architecture about
how RTP and RTCP streams should be encoded and transmitted, and how
their relation to CLUE Media Captures should be communicated. The
proposed solution supports multiple RTP topologies [RFC7667].
With regards to the media (audio, video and timed text), systems that
support CLUE use RTP for the media, SDP for codec and media transport
negotiation (CLUE individual encodings) and the CLUE protocol for
Media Capture description and selection. In order to associate the
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media in the different protocols there are three mapping that need to
be specified:
1. CLUE individual encodings to SDP
2. RTP streams to SDP (this is not a CLUE specific mapping)
3. RTP streams to MC to map the received RTP steam to the current MC
in the MCC.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC2119[RFC2119] and
indicate requirement levels for RTP processing in compliant CLUE
implementations.
The definitions from the CLUE framework document
[I-D.ietf-clue-framework] section 3 are used by this document as
well.
3. RTP topologies for CLUE
The typical RTP topologies used by CLUE Telepresence systems specify
different behaviors for RTP and RTCP distribution. A number of RTP
topologies are described in [RFC7667]. For CLUE telepresence, the
relevant topologies include Point-to-Point, as well as Media-Mixing
mixers, Media- Switching mixers, and Selective Forwarding Middleboxs.
In the Point-to-Point topology, one peer communicates directly with a
single peer over unicast. There can be one or more RTP sessions,
each sent on a separate 5-tuple, and having a separate SSRC space,
with each RTP session carrying multiple RTP streams identified by
their SSRC. All SSRCs are recognized by the peers based on the
information in the RTCP Source description (SDES) report that
includes the CNAME and SSRC of the sent RTP streams. There are
different Point-to-Point use cases as specified in CLUE use case
[RFC7205]. In some cases, a CLUE session which, at a high-level, is
point-to-point may nonetheless have an RTP stream which is best
described by one of the mixer topologies. For example, a CLUE
endpoint can produce composite or switched captures for use by a
receiving system with fewer displays than the sender has cameras.
The Media Capture may be described using an MCC.
For the Media Mixer topology [RFC7667], the peers communicate only
with the mixer. The mixer provides mixed or composited media
streams, using its own SSRC for the sent streams. If needed by CLUE
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endpoint, the conference roster information including conference
participants, endpoints, media and media-id (SSRC) can be determined
using the conference event package [RFC4575] element.
Media-switching mixers and Selective Forwarding Middleboxes behave as
described in [RFC7667]
4. Mapping CLUE Capture Encodings to RTP streams
The different topologies described in Section 3 create different SSRC
distribution models and RTP stream multiplexing points.
Most video conferencing systems today can separate multiple RTP
sources by placing them into RTP sessions using the SDP description;
the video conferencing application can also have some knowledge about
the purpose of each RTP session. For example, video conferencing
applications that have a primary video source and a slides video
source can send each media source in a separate RTP session with a
content attribute [RFC4796] enabling different application behavior
for each received RTP media source. Demultiplexing is
straightforward because each media capture is sent as a single RTP
stream, with each RTP stream being sent in a separate RTP session, on
a distinct UDP 5-tuple. This will also be true for mapping the RTP
streams to Media Captures Encodings if each Media Capture Encodings
uses a separate RTP session, and the consumer can identify it based
on the receiving RTP port. In this case, SDP only needs to label the
RTP session with an identifier that can be used to identify the Media
Capture in the CLUE description. The SDP label attribute serves as
this identifier.
Each Capture Encoding MUST be sent as a separate RTP stream. CLUE
endpoints MUST support sending each such RTP stream in a separate RTP
session signalled by an SDP m= line. They MAY also support sending
some or all of the RTP streams in a single RTP session, using the
mechanism described in [I-D.ietf-mmusic-sdp-bundle-negotiation] to
relate RTP streams to SDP m= lines.
MCCs bring another mapping issue, in that an MCC represents multiple
Media Captures that can be sent as part of this MCC if configured by
the consumer. When receiving an RTP stream which is mapped to the
MCC, the consumer needs to know which original MC it is in order to
get the MC parameters from the advertisement. If a consumer
requested a MCC, the original MC does not have a capture encoding, so
it cannot be associated with an m-line using a label as described in
CLUE signaling [I-D.ietf-clue-signaling]. This is important, for
example, to get correct scaling information for the original MC,
which may be different for the various MCs that are contributing to
the MCC.
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5. MCC Constituent CaptureID definition
For a MCC which can represent multiple switched MCs there is a need
to know which MC is represented in the current RTP stream at any
given time. This requires a mapping from the SSRC of the RTP stream
conveying a particular MCC to the constituent MC. In order to
address this mapping this document defines an RTP header extension
and SDES item that includes the captureID of the original MC,
allowing the consumer to use the original source MC's attributes like
the spatial information.
This mapping temporarily associates the SSRC of the RTP stream
conveying a particular MCC with the captureID of the single original
MC that is currently switched into the MCC. This mapping cannot be
used for the composed case where more than one original MC is
composed into the MCC simultaneously.
If there is only one MC in the MCC then the media provider MUST send
the captureID of the current constituent MC in the RTP Header
Extension and as a RTCP CaptureID SDES item. When the media provider
switches the MC it sends within an MCC, it MUST send the captureID
value for the MC just switched into the MCC in an RTP Header
Extension and as a RTCP CaptureID SDES item as specified in [RFC7941]
If there is more than one MC composed into the MCC then the media
provider MUST NOT send any of the MCs' captureIDs using this
mechanism. However, if an MCC is sending contributing source (CSRC)
information in the RTP header for a composed capture, it MAY send the
captureID values in the RTCP SDES packets giving source information
for the SSRC values sent as contributing sources (CSRCs).
If the media provider sends the captureID of a single MC switched
into an MCC, then later sends one composed stream of multiple MCs in
the same MCC, it MUST send the special value "-", a single dash
character, as the captureID RTP Header Extension and RTCP CaptureID
SDES item. The single dash character indicates there is no
applicable value for the MCC constituent CaptureID. The media
consumer interprets this as meaning that any previous CaptureID value
associated with this SSRC no longer applies. As
[I-D.ietf-clue-data-model-schema] defines the captureID syntax as
"xs:ID", the single dash character is not a legal captureID value, so
there is no possibility of confusing it with an actual captureID.
5.1. RTCP CaptureID SDES Item
This document specifies a new RTCP SDES item.
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| CaptId=TBA | length | CaptureID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| .... |
+-+-+-+-+-+-+-+-+
Note to the RFC Editor: Please replace TBA with the value assigned by
IANA.
This CaptureID is a variable-length UTF-8 string corresponding either
to a CaptureID negotiated in the CLUE protocol, or the single
character "-".
This SDES item MUST be sent in an SDES packet within a compound RTCP
packet unless support for Reduced-size RTCP has been negotiated as
specified in RFC 5506 [RFC5506], in which case it can be sent as an
SDES packet in a non-compound RTCP packet.
5.2. RTP Header Extension
The CaptureID is also carried in an RTP header extension [RFC5285],
using the mechanism defined in [RFC7941].
Support is negotiated within SDP using the URN "urn:ietf:params:rtp-
hdrext:sdes:CaptureID".
The CaptureID is sent in a RTP Header Extension because for switched
captures, receivers need to know which original MC corresponds to the
media being sent for an MCC, in order to correctly apply geometric
adjustments to the received media.
As discussed in [RFC7941], there is no need to send the CaptId Header
Extension with all RTP packets. Senders MAY choose to send it only
when a new MC is sent. If such a mode is being used, the header
extension SHOULD be sent in the first few RTP packets to reduce the
risk of losing it due to packet loss. See [RFC7941] for more
discussion of this.
6. Examples
In this partial advertisement the Media Provider advertises a
composed capture VC7 made of a big picture representing the current
speaker (VC3) and two picture-in-picture boxes representing the
previous speakers (the previous one -VC5- and the oldest one -VC6).
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<ns2:mediaCapture xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
xsi:type="ns2:videoCaptureType" captureID="VC7" mediaType="video">
<ns2:captureSceneIDREF>CS1</ns2:captureSceneIDREF>
<ns2:nonSpatiallyDefinable>true</ns2:nonSpatiallyDefinable>
<ns2:content>
<ns2:captureIDREF>VC3</ns2:captureIDREF>
<ns2:captureIDREF>VC5</ns2:captureIDREF>
<ns2:captureIDREF>VC6</ns2:captureIDREF>
</ns2:content>
<ns2:maxCaptures>3</ns2:maxCaptures>
<ns2:allowSubsetChoice>false</ns2:allowSubsetChoice>
<ns2:description lang="en">big picture of the current speaker
pips about previous speakers</ns2:description>
<ns2:priority>1</ns2:priority>
<ns2:lang>it</ns2:lang>
<ns2:mobility>static</ns2:mobility>
<ns2:view>individual</ns2:view>
</ns2:mediaCapture>
In this case the media provider will send capture IDs VC3, VC5 or VC6
as an RTP header extension and RTCP SDES message for the RTP stream
associated with the MC.
Note that this is part of the full advertisement message example from
CLUE data model[I-D.ietf-clue-data-model-schema] example and is not a
valid xml document.
7. Communication Security
CLUE endpoints MUST support RTP/SAVPF profile and SRTP [RFC3711].
CLUE endpoints MUST support DTLS [RFC6347] and DTLS-SRTP [RFC5763]
[RFC5764] for SRTP keying.
All media channels SHOULD be secure via SRTP and the RTP/SAVPF
profile unless the RTP media and its associated RTCP are secure by
other means (see [RFC7201] [RFC7202]).
All CLUE implementations MUST implement DTLS 1.0, with the cipher
suite TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA with the the P-256 curve
[FIPS186]. The DTLS-SRTP protection profile
SRTP_AES128_CM_HMAC_SHA1_80 MUST be supported for SRTP.Encrypted SRTP
Header extensions [RFC6904] MUST be supported.
Implementations SHOULD implement DTLS 1.2 with the
TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite.
Implementations MUST favor cipher suites which support PFS over non-
PFS cipher suites and SHOULD favor AEAD over non-AEAD cipher suites.
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NULL Protection profiles MUST NOT be used for RTP or RTCP.
CLUE endpoint MUST generate short-term persistent RTCP CNAMES, as
specified in [RFC7022], and thus can't be used for long term tracking
of the users.
8. Acknowledgments
The authors would like to thanks Allyn Romanow and Paul Witty for
contributing text to this work. Magnus Westerlund helped drafting
the security section.
9. IANA Considerations
This document defines a new extension URI in the RTP SDES Compact
Header Extensions subregistry of the Real-Time Transport Protocol
(RTP) Parameters registry, according to the following data:
Extension URI: urn:ietf:params:rtp-hdrext:sdes:CaptId
Description: CLUE CaptId
Contact: ron.even.tlv@gmail.com
Reference: RFC XXXX
The IANA is requested to register one new RTCP SDES items in the
"RTCP SDES Item Types" registry, as follows:
Value Abbrev Name Reference
TBA CCID CLUE CaptId [RFCXXXX]
Note to the RFC Editor: Please replace RFCXXXX with this RFC number.
10. Security Considerations
The security considerations of the RTP specification, the RTP/SAVPF
profile, and the various RTP/RTCP extensions and RTP payload formats
that form the complete protocol suite described in this memo apply.
It is not believed there are any new security considerations
resulting from the combination of these various protocol extensions.
The Extended Secure RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides
handling of fundamental issues by offering confidentiality, integrity
and partial source authentication. A mandatory to implement and use
media security solution is created by combining this secured RTP
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profile and DTLS-SRTP keying [RFC5764] as defined in the
communication security section of this memo Section 7
RTCP packets convey a Canonical Name (CNAME) identifier that is used
to associate RTP packet streams that need to be synchronised across
related RTP sessions. Inappropriate choice of CNAME values can be a
privacy concern, since long-term persistent CNAME identifiers can be
used to track users across multiple calls. The communication
security section of this memo Section 7 mandates generation of short-
term persistent RTCP CNAMES, as specified in [RFC7022] so they can't
be used for long term tracking of the users.
Some potential denial of service attacks exist if the RTCP reporting
interval is configured to an inappropriate value. This could be done
by configuring the RTCP bandwidth fraction to an excessively large or
small value using the SDP "b=RR:" or "b=RS:" lines [RFC3556], or some
similar mechanism, or by choosing an excessively large or small value
for the RTP/AVPF minimal receiver report interval (if using SDP, this
is the "a=rtcp-fb:... trr-int" parameter) [RFC4585] The risks are as
follows:
1. the RTCP bandwidth could be configured to make the regular
reporting interval so large that effective congestion control
cannot be maintained, potentially leading to denial of service
due to congestion caused by the media traffic;
2. the RTCP interval could be configured to a very small value,
causing endpoints to generate high rate RTCP traffic, potentially
leading to denial of service due to the non-congestion controlled
RTCP traffic; and
3. RTCP parameters could be configured differently for each
endpoint, with some of the endpoints using a large reporting
interval and some using a smaller interval, leading to denial of
service due to premature participant timeouts due to mismatched
timeout periods which are based on the reporting interval (this
is a particular concern if endpoints use a small but non-zero
value for the RTP/AVPF minimal receiver report interval (trr-int)
[RFC4585], as discussed in [I-D.ietf-avtcore-rtp-multi-stream]).
Premature participant timeout can be avoided by using the fixed (non-
reduced) minimum interval when calculating the participant timeout
([I-D.ietf-avtcore-rtp-multi-stream]). To address the other
concerns, endpoints SHOULD ignore parameters that configure the RTCP
reporting interval to be significantly longer than the default five
second interval specified in [RFC3550] (unless the media data rate is
so low that the longer reporting interval roughly corresponds to 5%
of the media data rate), or that configure the RTCP reporting
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interval small enough that the RTCP bandwidth would exceed the media
bandwidth.
The guidelines in [RFC6562] apply when using variable bit rate (VBR)
audio codecs such as Opus.
The use of the encryption of the header extensions are RECOMMENDED,
unless there are known reasons, like RTP middleboxes performing voice
activity based source selection or third party monitoring that will
greatly benefit from the information, and this has been expressed
using API or signalling. If further evidence are produced to show
that information leakage is significant from audio level indications,
then use of encryption needs to be mandated at that time.
In multi-party communication scenarios using RTP Middleboxes; this
middleboxes are REQUIRED, by this protocol, to not weaken the
sessions' security. The middlebox SHOULD maintain the
confidentiality, integrity and perform source authentication. The
middlebox MAY perform checks that prevents any endpoint participating
in a conference to impersonate another. Some additional security
considerations regarding multi-party topologies can be found in
[RFC7667]
The CaptureID is created as part of the CLUE protocol. The CaptId
SDES item is used to convey the same CaptureID value in the SDES
item. When sending the SDES item the security consideration
specified in the security section of [RFC7941] and in the
communication security section of this memo Section 7 are applicable.
Note that since the CaptureID is carried also in CLUE protocol
messages it is RECOMMENDED that this SDES item use at least similar
protection profiles as the CLUE protocol messages carried in the CLUE
data channel. .
11. References
11.1. Normative References
[I-D.ietf-clue-data-model-schema]
Presta, R. and S. Romano, "An XML Schema for the CLUE data
model", draft-ietf-clue-data-model-schema-17 (work in
progress), August 2016.
[I-D.ietf-clue-framework]
Duckworth, M., Pepperell, A., and S. Wenger, "Framework
for Telepresence Multi-Streams", draft-ietf-clue-
framework-25 (work in progress), January 2016.
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[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
negotiation-36 (work in progress), October 2016.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, DOI 10.17487/RFC3711, March 2004,
<http://www.rfc-editor.org/info/rfc3711>.
[RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
for Establishing a Secure Real-time Transport Protocol
(SRTP) Security Context Using Datagram Transport Layer
Security (DTLS)", RFC 5763, DOI 10.17487/RFC5763, May
2010, <http://www.rfc-editor.org/info/rfc5763>.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764,
DOI 10.17487/RFC5764, May 2010,
<http://www.rfc-editor.org/info/rfc5764>.
[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347,
January 2012, <http://www.rfc-editor.org/info/rfc6347>.
[RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure
Real-time Transport Protocol (SRTP)", RFC 6904,
DOI 10.17487/RFC6904, April 2013,
<http://www.rfc-editor.org/info/rfc6904>.
[RFC7941] Westerlund, M., Burman, B., Even, R., and M. Zanaty, "RTP
Header Extension for the RTP Control Protocol (RTCP)
Source Description Items", RFC 7941, DOI 10.17487/RFC7941,
August 2016, <http://www.rfc-editor.org/info/rfc7941>.
11.2. Informative References
[FIPS186] National Institute of Standards and Technology, "Digital
Signature Standard", FIPS PUB 186-4, July 2013.
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[I-D.ietf-avtcore-rtp-multi-stream]
Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session",
draft-ietf-avtcore-rtp-multi-stream-11 (work in progress),
December 2015.
[I-D.ietf-clue-signaling]
Kyzivat, P., Xiao, L., Groves, C., and R. Hansen, "CLUE
Signaling", draft-ietf-clue-signaling-10 (work in
progress), January 2017.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
DOI 10.17487/RFC3264, June 2002,
<http://www.rfc-editor.org/info/rfc3264>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth",
RFC 3556, DOI 10.17487/RFC3556, July 2003,
<http://www.rfc-editor.org/info/rfc3556>.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
July 2006, <http://www.rfc-editor.org/info/rfc4566>.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A
Session Initiation Protocol (SIP) Event Package for
Conference State", RFC 4575, DOI 10.17487/RFC4575, August
2006, <http://www.rfc-editor.org/info/rfc4575>.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
DOI 10.17487/RFC4585, July 2006,
<http://www.rfc-editor.org/info/rfc4585>.
[RFC4796] Hautakorpi, J. and G. Camarillo, "The Session Description
Protocol (SDP) Content Attribute", RFC 4796,
DOI 10.17487/RFC4796, February 2007,
<http://www.rfc-editor.org/info/rfc4796>.
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[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
2008, <http://www.rfc-editor.org/info/rfc5124>.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July
2008, <http://www.rfc-editor.org/info/rfc5285>.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
2009, <http://www.rfc-editor.org/info/rfc5506>.
[RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of
Variable Bit Rate Audio with Secure RTP", RFC 6562,
DOI 10.17487/RFC6562, March 2012,
<http://www.rfc-editor.org/info/rfc6562>.
[RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla,
"Guidelines for Choosing RTP Control Protocol (RTCP)
Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022,
September 2013, <http://www.rfc-editor.org/info/rfc7022>.
[RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP
Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,
<http://www.rfc-editor.org/info/rfc7201>.
[RFC7202] Perkins, C. and M. Westerlund, "Securing the RTP
Framework: Why RTP Does Not Mandate a Single Media
Security Solution", RFC 7202, DOI 10.17487/RFC7202, April
2014, <http://www.rfc-editor.org/info/rfc7202>.
[RFC7205] Romanow, A., Botzko, S., Duckworth, M., and R. Even, Ed.,
"Use Cases for Telepresence Multistreams", RFC 7205,
DOI 10.17487/RFC7205, April 2014,
<http://www.rfc-editor.org/info/rfc7205>.
[RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
DOI 10.17487/RFC7667, November 2015,
<http://www.rfc-editor.org/info/rfc7667>.
Authors' Addresses
Even & Lennox Expires August 31, 2017 [Page 13]
Internet-Draft RTP mapping to CLUE February 2017
Roni Even
Huawei Technologies
Tel Aviv
Israel
Email: roni.even@huawei.com
Jonathan Lennox
Vidyo, Inc.
433 Hackensack Avenue
Seventh Floor
Hackensack, NJ 07601
US
Email: jonathan@vidyo.com
Even & Lennox Expires August 31, 2017 [Page 14]
