CLUE WG R. Even
Internet-Draft Huawei Technologies
Intended status: Standards Track J. Lennox
Expires: April 21, 2015 Vidyo
October 18, 2014
Mapping RTP streams to CLUE media captures
draft-ietf-clue-rtp-mapping-03.txt
Abstract
This document describes how the Real Time transport Protocol (RTP) is
used in the context of the CLUE protocol. It also describes the
mechanisms and recommended practice for mapping RTP media streams
defined in SDP to CLUE media captures.
Status of This Memo
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This Internet-Draft will expire on April 21, 2015.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. RTP topologies for CLUE . . . . . . . . . . . . . . . . . . . 3
4. Mapping CLUE Capture Encodings to RTP streams . . . . . . . . 5
4.1. Review of current directions in MMUSIC, AVText and
AVTcore . . . . . . . . . . . . . . . . . . . . . . . . . 6
4.2. Requirements of a solution . . . . . . . . . . . . . . . 7
4.3. Static Mapping . . . . . . . . . . . . . . . . . . . . . 8
4.4. Dynamic mapping . . . . . . . . . . . . . . . . . . . . . 9
4.5. Recommendations . . . . . . . . . . . . . . . . . . . . . 9
5. Application to CLUE Media Requirements . . . . . . . . . . . 9
6. CaptureID definition . . . . . . . . . . . . . . . . . . . . 11
6.1. RTCP CaptureId SDES Item . . . . . . . . . . . . . . . . 11
6.2. RTP Header Extension . . . . . . . . . . . . . . . . . . 12
7. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 12
8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 12
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 12
10. Security Considerations . . . . . . . . . . . . . . . . . . . 13
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 13
11.1. Normative References . . . . . . . . . . . . . . . . . . 13
11.2. Informative References . . . . . . . . . . . . . . . . . 13
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 14
1. Introduction
Telepresence systems can send and receive multiple media streams.
The CLUE framework [I-D.ietf-clue-framework] defines media captures
as a source of Media, such as from one or more Capture Devices. A
Media Capture (MC) may be the source of one or more Media streams. 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 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.
This document makes recommendations, for the telepresence
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.
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With regards to the media (audio and video), 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
media in the different protocols there are three mapping that need to
be specified:
1. CLUE individual encodings to SDP
2. RTP media streams to SDP (this is not a CLUE specific mapping)
3. RTP media 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 compliant RTP implementations.
3. RTP topologies for CLUE
The typical RTP topologies used by Telepresence systems specify
different behaviors for RTP and RTCP distribution. A number of RTP
topologies are described in [I-D.ietf-avtcore-rtp-topologies-update].
For telepresence, the relevant topologies include point-to-point, as
well as media 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, and
each RTP session can carry multiple RTP streams identified by their
SSRC. All SSRCs will be recognized by the peers based on the
information in the RTCP SDES report that will include the CNAME and
SSRC of the sent RTP streams. There are different point to point use
cases as specified in CLUE use case [RFC7205]. There may be a
difference between the symmetric and asymmetric use cases. While in
the symmetric use case the typical mapping will be from a Media
capture device to a render device (e.g. camera to monitor) in the
asymmetric case the render device may receive different capture
information (RTP stream from different cameras) if it has fewer
rendering devices (monitors). In some cases, a CLUE session which,
at a high-level, is point-to-point may nonetheless have RTP 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 MCC.
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For the Media Mixer topology
[I-D.ietf-avtcore-rtp-topologies-update], the peers communicate only
with the mixer. The mixer provides mixed or composited media
streams, using its own SSRC for the sent streams. There are two
cases here. In the first case the mixer may have separate RTP
sessions with each peer (similar to the point to point topology)
terminating the RTCP sessions on the mixer; this is known as Topo-
RTCP-Terminating MCU in [I-D.ietf-avtcore-rtp-topologies-update]. In
the second case, the mixer can use a conference-wide RTP session
similar to [I-D.ietf-avtcore-rtp-topologies-update] Topo-mixer or
Topo-Video-switching. The major difference is that for the second
case, the mixer uses conference-wide RTP sessions, and distributes
the RTCP reports to all the RTP session participants, enabling them
to learn all the CNAMEs and SSRCs of the participants and know the
contributing source or sources (CSRCs) of the original streams from
the RTP header. In the first case, the Mixer terminates the RTCP and
the participants cannot know all the available sources based on the
RTCP information. The conference roster information including
conference participants, endpoints, media and media-id (SSRC) can be
available using the conference event package [RFC4575] element.
In the Media-Switching Mixer topology
[I-D.ietf-avtcore-rtp-topologies-update], the peer to mixer
communication is unicast with mixer RTCP feedback. It is
conceptually similar to a compositing mixer as described in the
previous paragraph, except that rather than compositing or mixing
multiple sources, the mixer provides one or more conceptual sources
selecting one source at a time from the original sources. The Mixer
creates a conference-wide RTP session by sharing remote SSRC values
as CSRCs to all conference participants.
In the Selective Forwarding middlebox topology, the peer to mixer
communication is unicast with RTCP mixer feedback. Every potential
sender in the conference has a source which may be "projected" by the
mixer into every other RTP session in the conference; thus, every
original source is maintained with an independent RTP identity to
every receiver, maintaining separate decoding state and its original
RTCP SDES information. However, RTCP is terminated at the mixer,
which might also perform reliability, repair, rate adaptation, or
transcoding on the stream. Senders' SSRCs may be renumbered by the
mixer. The sender may turn the projected sources on and off at any
time, depending on which sources it thinks are most relevant for the
receiver; this is the primary reason why this topology must act as an
RTP mixer rather than as a translator, as otherwise these disabled
sources would appear to have enormous packet loss. Source switching
is accomplished through this process of enabling and disabling
projected sources, with the higher-level semantic assignment of
reason for the RTP streams assigned externally.
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The above topologies demonstrate two major RTP/RTCP behaviors:
1. The mixer may either use the source SSRC when forwarding RTP
packets, or use its own created SSRC. Still the mixer will
distribute all RTCP information to all participants creating
conference-wide RTP session/s. This allows the participants to
learn the available RTP sources in each RTP session. The
original source information will be the SSRC or in the CSRC
depending on the topology. The point to point case behaves like
this.
2. The mixer terminates the RTCP from the source, creating separate
RTP sessions with the peers. In this case the participants will
not receive the source SSRC in the CSRC. Since this is usually a
mixer topology, the source information is available from the SIP
conference event package [RFC4575]. Subscribing to the
conference event package allows each participant to know the
SSRCs of all sources in the conference.
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 separate RTP sessions using, the SDP
description. For example, main and slides video sources are
separated into separate RTP sessions based on the content attribute
[RFC4796]. This solution works straightforward if the multiplexing
point is at the UDP transport level, where each RTP stream uses a
separate RTP session. 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 identifies the media capture in
the CLUE description. In this case, it does not change the mapping
even if the RTP session is switched using same or different SSRC.
(The multiplexing is not at the SSRC level).
Even though Session multiplexing is supported by CLUE, for scaling
reasons, CLUE recommends using SSRC multiplexing in a single or
multiple sessions using [I-D.ietf-mmusic-sdp-bundle-negotiation]. So
we need to look at how to map RTP streams to Captures Encodings when
SSRC multiplexing is used.
When looking at SSRC multiplexing we can see that in various
topologies, the SSRC behavior may be different:
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1. The SSRCs are static (assigned by the MCU/Mixer), and there is an
SSRC for each media capture encoding defined in the CLUE
protocol. Source information may be conveyed using CSRC, or, in
the case of topo-RTCP-Terminating MCU, is not conveyed.
2. The SSRCs are dynamic, representing the original source and are
relayed by the Mixer/MCU to the participants.
In the above two cases the MCU/Mixer may create an advertisement,
with a virtual room capture scene.
Another case we can envision is that the MCU / Mixer relays all the
capture scenes from all advertisements to all consumers. This means
that the advertisement will include multiple capture scenes, each
representing a separate TelePresence room with its own coordinate
system.
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, not the original MC, 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.
4.1. Review of current directions in MMUSIC, AVText and AVTcore
Editor's note: This section provides an overview of the RFCs and
drafts that are used as a base for a mapping solution. This section
is for information only; the normative behavior is given in the cited
documents.
The solution needs to also support the simulcast case where more than
one RTP session may be advertised for a Media Capture. Support of
such simulcast is out of scope for CLUE but a solution is needed.
When looking at the available tools based on current work in MMUSIC,
AVTcore and AVText for supporting SSRC multiplexing the following
documents are considered to be relevant.
Negotiating Media Multiplexing Using the Session Description Protocol
in [I-D.ietf-mmusic-sdp-bundle-negotiation] defines a "bundle" SDP
grouping extension that can be used with SDP Offer/Answer mechanism
to negotiate the usage of a single 5-tuple for sending and receiving
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media associated with multiple SDP media descriptions ("m=").
[bundle] specifies how to associate a received RTP stream with the
m-line describing it. A "currently hidden" assumption in the work is
that each SDP m-line represents a single media source.
[I-D.ietf-mmusic-sdp-bundle-negotiation] specifies using the SDP mid
value and sending it as RTCP SDES and an RTP header extension in
order to be able to map the RTP stream to the SDP m-line. This is
relevant when there are multiple RTP streams with the same payload
subtype number.
SDP Source attribute [RFC5576] mechanisms to describe specific
attributes of RTP sources based on their SSRC.
Negotiation of generic image attributes in SDP [RFC6236] provides the
means to negotiate the image size. The image attribute can be used
to offer different image parameters like size but in order to offer
multiple RTP streams with different resolutions it does it using
separate RTP session for each image option
([I-D.ietf-mmusic-sdp-bundle-negotiation] provides the support of a
single RTP session but each image option will need a separate SDP
m-line).
In the next sections, the document will propose mechanisms to map the
RTP streams to media captures addressing.
4.2. Requirements of a solution
This section lists, more briefly, the requirements a media
architecture for Clue telepresence needs to achieve, summarizing the
discussion of previous sections. In this section, RFC 2119 [RFC2119]
language refers to requirements on a solution, not an implementation;
thus, requirements keywords are not written in capital letters.
Media-1: It must not be necessary for a Clue session to use more than
a single transport flow for transport of a given media type (video or
audio).
Media-2: It must, however, be possible for a Clue session to use
multiple transport flows for a given media type where it is
considered valuable (for example, for distributed media, or
differential quality-of-service).
Media-3: It must be possible for a Clue endpoint or MCU to
simultaneously send sources corresponding to static captures and to
both composited and switched multi-content captures in the same
transport flow. (Any given device might not necessarily be able send
all of these source types; but for those that can, it must be
possible for them to be sent simultaneously.)
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Media-4: It must be possible for an original source to move among
multi-content captures (i.e. at one time be sent for one MCC, and at
a later time be sent for another one).
Media-5: It must be possible for a source to be placed into a MCC
even if the source is a "late joiner", i.e. was added to the
conference after the receiver requested the MCC.
Media-6: Whenever a given source is assigned to a switched capture,
it must be immediately possible for a receiver to determine the MCC
it corresponds to, and thus that any previous source is no longer
being mapped to that switched capture.
Media-7: It must be possible for a receiver to identify the original
capture(s) that are currently being mapped to an MCC, and correlate
it with both the Clue advertisement and out-of-band (non-Clue)
information such as rosters.
Media-8: It must be possible for a source to move among MCCs without
requiring a refresh of decoder state (e.g., for video, a fresh
I-frame), when this is unnecessary. However, it must also be
possible for a receiver to indicate when a refresh of decoder state
is in fact necessary.
Media-9: If a given source is being sent on the same transport flow
for more than one reason (e.g. if it corresponds to more than one
switched capture at once, or to a static capture), it should be
possible for a sender to send only one copy of the source.
Media-10: On the network, media flows should, as much as possible,
look and behave like currently-defined usages of existing protocols;
established semantics of existing protocols must not be redefined.
Media-11: The solution should seek to minimize the processing burden
for boxes that distribute media to decoding hardware.
Media-12: If multiple sources from a single synchronization context
are being sent simultaneously, it must be possible for a receiver to
associate and synchronize them properly, even for sources that are
are mapped to switched captures.
4.3. Static Mapping
Static mapping is widely used in current MCU implementations. It is
also common for a point to point symmetric use case when both
endpoints have the same capabilities. For capture encodings with
static SSRCs, it is most straightforward to indicate this mapping
outside the media stream, in the CLUE or SDP signaling. When using
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SSRC multiplexing [I-D.ietf-mmusic-sdp-bundle-negotiation] defines
the use of the SDP mid attribute value to associate between the
received RTP stream and the SDP m-line. The mid is carried as an RTP
header extension and RTCP SDES message defined in
[I-D.ietf-mmusic-sdp-bundle-negotiation] .
4.4. Dynamic mapping
Dynamic mapping by tagging each media packet with the SDP mid value.
This means that a receiver immediately knows how to interpret
received media, even when an unknown SSRC is seen. As long as the
media carries a known mid, it can be assumed that this media stream
will replace the stream currently being received with that mid.
This gives significant advantages to switching latency, as a switch
between sources can be achieved without any form of negotiation with
the receiver.
However, the disadvantage in using a mid in the stream that it
introduces additional processing costs for every media packet, as mid
are scoped only within one hop (i.e., within a cascaded conference a
mid that is used from the source to the first MCU is not meaningful
between two MCUs, or between an MCU and a receiver), and so they may
need to be added or modified at every stage.
An additional issue with putting mid in the RTP packets comes from
cases where a non-bundle aware endpoint is being switched by an MCU
to a bundle endpoint. In this case, we may require up to an
additional 12 bytes in the RTP header, which may push a media packet
over the MTU. However, as the MTU on either side of the switch may
not match, it is possible that this could happen even without adding
extra data into the RTP packet. The 12 additional bytes per packet
could also be a significant bandwidth increase in the case of very
low bandwidth audio codecs.
4.5. Recommendations
The recommendation is that CLUE endpoint using SSRC multiplexing MUST
support [[I-D.ietf-mmusic-sdp-bundle-negotiation] and use the SDP mid
attribute for mapping.
5. Application to CLUE Media Requirements
The requirement section Section 4.2 offers a number of requirements
that are believed to be necessary for a CLUE RTP mapping. The
solutions described in this document are believed to meet these
requirements, though some of them are only possible for some of the
topologies. (Since the requirements are generally of the form "it
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must be possible for a sender to do something", this is adequate; a
sender which wishes to perform that action needs to choose a topology
which allows the behavior it wants.
In this section we address only those requirements where the
topologies or the association mechanisms treat the requirements
differently.
Media-4: It must be possible for an original source to move among
switched captures (i.e. at one time be sent for one switched capture,
and at a later time be sent for another one).
This applies naturally for static sources with a Switched Mixer. For
dynamic sources with a Selective Forwarding middlebox, this just
requires the mid in the header extension element to be updated
appropriately.
Media-6: Whenever a given source is transmitted for a switched
capture, it must be immediately possible for a receiver to determine
the switched capture it corresponds to, and thus that any previous
source is no longer being mapped to that switched capture.
For a Switched Mixer, this applies naturally. For a Selective
Forwarding middlebox, this is done based on the mid.
Media-7: It must be possible for a receiver to identify the original
source that is currently being mapped to a switched capture, and
correlate it with out-of-band (non-Clue) information such as rosters.
For a Switched Mixer, this is done based on the CSRC, if the mixer is
providing CSRCs; For a Selective Forwarding middlebox, this is done
based on the SSRC.
For MCC which can represent multiple switched MCs there is a need to
know which MC represents the current RTP stream, requires a mapping
from an RTP stream to an MC. In order to address this mapping we
will define an RTP header extension that will include the CaptureID
in order to map to the original MC allowing the consumer to use the
MC attributes like the spatial information.
Media-8: It must be possible for a source to move among switched
captures without requiring a refresh of decoder state (e.g., for
video, a fresh I-frame), when this is unnecessary. However, it must
also be possible for a receiver to indicate when a refresh of decoder
state is in fact necessary.
This can be done by a Selective Forwarding middlebox, but not by a
Switching Mixer. The last requirement can be accomplished through an
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FIR message [RFC5104], though potentially a faster mechanism (not
requiring a round-trip time from the receiver) would be preferable.
Media-9: If a given source is being sent on the same transport flow
to satisfy more than one capture (e.g. if it corresponds to more than
one switched capture at once, or to a static capture as well as a
switched capture), it should be possible for a sender to send only
one copy of the source.
For a Selective Forwarding middlebox, this may be a problem since an
encoding can be used by a single MC, it will require using the same
SDP label for multiple MC (example middle camera and active speaker
MC) this can also be done for an environment with a hybrid of mixer
topologies and static and dynamic captures. It is not possible for
static captures from a Switched Mixer.
Media-12: If multiple sources from a single synchronization context
are being sent simultaneously, it must be possible for a receiver to
associate and synchronize them properly, even for sources that are
mapped to switched captures.
For a Mixed or Switched Mixer topology, receivers will see only a
single synchronization context (CNAME), corresponding to the mixer.
For a Selective Forwarding middlebox, separate projecting sources
keep separate synchronization contexts based on their original
CNAMEs, thus allowing independent synchronization of sources from
independent rooms without needing global synchronization. In hybrid
cases, however (e.g. if audio is mixed), all sources which need to be
synchronized with the mixed audio must get the same CNAME (and thus a
mixer-provided timebase) as the mixed audio.
6. CaptureID definition
For mapping an RTP stream to a specific MC in the MCC the CLUE
captureId is used. The media sender MUST send for MCC the captureID
in the RTP header and as a RTCP SDES message.
6.1. RTCP CaptureId SDES Item
This document specifies a new RTCP SDES message
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| CaptureId = XXX | length |CaptureId
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ....
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This CaptureID is the same as in the CLUE MC and is also used in the
RTP header extension.
This SDES message MAY be sent in a compound RTCP packet based on the
application need.
6.2. RTP Header Extension
The CaptureId is carried within the RTP header extension field, using
[RFC5285] two bytes header extension.
Support is negotiated within the SDP, i.e.
a=extmap:1 urn:ietf:params:rtp-hdrext:CaptureId
Packets tagged by the sender with the CapturId then contain a header
extension as shown below
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ID | Len-1 | CaptureId
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| CaptureId .. |
+-+-+-+-+-+-+-+-+
There is no need to send the CaptureId 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.
7. Examples
TBD
8. Acknowledgements
The authors would like to thanks Allyn Romanow and Paul Witty for
contributing text to this work.
9. IANA Considerations
This document defines a new extension URI in the RTP 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:CaptureId
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Description: CLUE CaptureId
Contact: roni.even@mail01.huawei.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 CaptureId [RFCXXXX]
10. Security Considerations
TBD.
11. References
11.1. Normative References
[I-D.ietf-clue-framework]
Duckworth, M., Pepperell, A., and S. Wenger, "Framework
for Telepresence Multi-Streams", draft-ietf-clue-
framework-17 (work in progress), September 2014.
[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-12 (work in progress), October 2014.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
11.2. Informative References
[I-D.ietf-avtcore-rtp-topologies-update]
Westerlund, M. and S. Wenger, "RTP Topologies", draft-
ietf-avtcore-rtp-topologies-update-04 (work in progress),
August 2014.
[I-D.ietf-clue-signaling]
Kyzivat, P., Xiao, L., Groves, C., and R. Hansen, "CLUE
Signaling", draft-ietf-clue-signaling-03 (work in
progress), August 2014.
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Internet-Draft RTP mapping to CLUE October 2014
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264, June
2002.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session
Initiation Protocol (SIP) Event Package for Conference
State", RFC 4575, August 2006.
[RFC4796] Hautakorpi, J. and G. Camarillo, "The Session Description
Protocol (SDP) Content Attribute", RFC 4796, February
2007.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, February 2008.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
January 2008.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, July 2008.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009.
[RFC6236] Johansson, I. and K. Jung, "Negotiation of Generic Image
Attributes in the Session Description Protocol (SDP)", RFC
6236, May 2011.
[RFC7205] Romanow, A., Botzko, S., Duckworth, M., and R. Even, "Use
Cases for Telepresence Multistreams", RFC 7205, April
2014.
Authors' Addresses
Even & Lennox Expires April 21, 2015 [Page 14]
Internet-Draft RTP mapping to CLUE October 2014
Roni Even
Huawei Technologies
Tel Aviv
Israel
Email: roni.even@mail01.huawei.com
Jonathan Lennox
Vidyo, Inc.
433 Hackensack Avenue
Seventh Floor
Hackensack, NJ 07601
US
Email: jonathan@vidyo.com
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