Internet DRAFT - draft-lennox-avtcore-rtp-multi-stream
draft-lennox-avtcore-rtp-multi-stream
AVTCORE J. Lennox
Internet-Draft Vidyo
Updates: 3550 (if approved) M. Westerlund
Intended status: Standards Track Ericsson
Expires: August 29, 2013 Q. Wu
Huawei
C. Perkins
University of Glasgow
February 25, 2013
RTP Considerations for Endpoints Sending Multiple Media Streams
draft-lennox-avtcore-rtp-multi-stream-02
Abstract
This document expands and clarifies the behavior of the Real-Time
Transport Protocol (RTP) endpoints when they are sending multiple
media streams in a single RTP session. In particular, issues
involving Real-Time Transport Control Protocol (RTCP) messages are
described.
This document updates RFC 3550 in regards to handling of multiple
SSRCs per endpoint in RTP sessions.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on August 29, 2013.
Copyright Notice
Copyright (c) 2013 IETF Trust and the persons identified as the
document authors. All rights reserved.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Use Cases For Multi-Stream Endpoints . . . . . . . . . . . . 3
3.1. Multiple-Capturer Endpoints . . . . . . . . . . . . . . . 3
3.2. Multi-Media Sessions . . . . . . . . . . . . . . . . . . 4
3.3. Multi-Stream Mixers . . . . . . . . . . . . . . . . . . . 4
4. Issue Cases . . . . . . . . . . . . . . . . . . . . . . . . . 4
4.1. Cascaded Multi-party Conference with Source Projecting
Mixers . . . . . . . . . . . . . . . . . . . . . . . . . 5
5. Multi-Stream Endpoint RTP Media Recommendations . . . . . . . 5
6. Multi-Stream Endpoint RTCP Recommendations . . . . . . . . . 5
6.1. RTCP Reporting Requirement . . . . . . . . . . . . . . . 6
6.2. Initial Reporting Interval . . . . . . . . . . . . . . . 6
6.3. Compound RTCP Packets . . . . . . . . . . . . . . . . . . 6
7. RTCP Bandwidth Considerations for Sources with Disparate
Rates . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
8. Grouping of RTCP Reception Statistics and Other Feedback . . 7
8.1. Semantics and Behavior of Reporting Groups . . . . . . . 8
8.2. Determine the Report Group . . . . . . . . . . . . . . . 9
8.3. RTCP Packet Reporting Group's Reporting Sources . . . . . 9
8.4. RTCP Source Description (SDES) item for Reporting Groups 11
8.5. Middlebox Considerations . . . . . . . . . . . . . . . . 11
8.6. SDP signaling for Reporting Groups . . . . . . . . . . . 11
8.7. Bandwidth Benefits of RTCP Reporting Groups . . . . . . . 11
8.8. Consequences of RTCP Reporting Groups . . . . . . . . . . 12
9. Security Considerations . . . . . . . . . . . . . . . . . . . 13
10. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 13
11. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 13
11.1. RTCP SDES Item . . . . . . . . . . . . . . . . . . . . . 13
11.2. RTCP Packet Type . . . . . . . . . . . . . . . . . . . . 14
12. References . . . . . . . . . . . . . . . . . . . . . . . . . 14
12.1. Normative References . . . . . . . . . . . . . . . . . . 14
12.2. Informative References . . . . . . . . . . . . . . . . . 14
Appendix A. Changes From Earlier Versions . . . . . . . . . . . 15
A.1. Changes From Draft -01 . . . . . . . . . . . . . . . . . 15
A.2. Changes From Draft -00 . . . . . . . . . . . . . . . . . 16
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Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 16
1. Introduction
At the time The Real-Time Transport Protocol (RTP) [RFC3550] was
originally written, and for quite some time after, endpoints in RTP
sessions typically only transmitted a single media stream per RTP
session, where separate RTP sessions were typically used for each
distinct media type.
Recently, however, a number of scenarios have emerged (discussed
further in Section 3) in which endpoints wish to send multiple RTP
media streams, distinguished by distinct RTP synchronization source
(SSRC) identifiers, in a single RTP session. Although RTP's initial
design did consider such scenarios, the specification was not
consistently written with such use cases in mind. The specifications
are thus somewhat unclear.
The purpose of this document is to expand and clarify [RFC3550]'s
language for these use cases. The authors believe this does not
result in any major normative changes to the RTP specification,
however this document defines how the RTP specification is to be
interpreted. In these cases, this document updates RFC3550.
The document starts with terminology and some use cases where
multiple sources will occur. This is followed by some case studies
to try to identify issues that exist and need considerations. This
is followed by RTP and RTCP recommendations to resolve issues. Next
are security considerations and remaining open issues.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in RFC
2119 [RFC2119] and indicate requirement levels for compliant
implementations.
3. Use Cases For Multi-Stream Endpoints
This section discusses several use cases that have motivated the
development of endpoints that send multiple streams in a single RTP
session.
3.1. Multiple-Capturer Endpoints
The most straightforward motivation for an endpoint to send multiple
media streams in a session is the scenario where an endpoint has
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multiple capture devices of the same media type and characteristics.
For example, telepresence endpoints, of the type described by the
CLUE Telepresence Framework [I-D.ietf-clue-framework] is designed,
often have multiple cameras or microphones covering various areas of
a room.
3.2. Multi-Media Sessions
Recent work has been done in RTP
[I-D.ietf-avtcore-multi-media-rtp-session] and SDP
[I-D.ietf-mmusic-sdp-bundle-negotiation] to update RTP's historical
assumption that media streams of different media types would always
be sent on different RTP sessions. In this work, a single endpoint's
audio and video media streams (for example) are instead sent in a
single RTP session.
3.3. Multi-Stream Mixers
There are several RTP topologies which can involve a central device
that itself generates multiple media streams in a session.
One example is a mixer providing centralized compositing for a multi-
capture scenario like that described in Section 3.1. In this case,
the centralized node is behaving much like a multi-capturer endpoint,
generating several similar and related sources.
More complicated is the Source Projecting Mixer, see Section 3.6 of
[I-D.westerlund-avtcore-rtp-topologies-update]. This is a central
box that receives media streams from several endpoints, and then
selectively forwards modified versions of some of the streams toward
the other endpoints it is connected to. Toward one destination, a
separate media source appears in the session for every other source
connected to the mixer, "projected" from the original streams, but at
any given time many of them can appear to be inactive (and thus are
receivers, not senders, in RTP). This sort of device is closer to
being an RTP mixer than an RTP translator, in that it terminates RTCP
reporting about the mixed streams, and it can re-write SSRCs,
timestamps, and sequence numbers, as well as the contents of the RTP
payloads, and can turn sources on and off at will without appearing
to be generating packet loss. Each projected stream will typically
preserve its original RTCP source description (SDES) information.
4. Issue Cases
This section illustrates some scenarios that have shown areas where
the RTP specification is unclear.
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4.1. Cascaded Multi-party Conference with Source Projecting Mixers
This issue case tries to illustrate the effect of having multiple
SSRCs sent by an endpoint, by considering the traffic between two
source-projecting mixers in a large multi-party conference.
For concreteness, consider a 200-person conference, where 16 sources
are viewed at any given time. Assuming participants are distributed
evenly among the mixers, each mixer would have 100 sources "behind"
(projected through) it, of which at any given time eight are active
senders. Thus, the RTP session between the mixers consists of two
endpoints, but 200 sources.
The RTCP bandwidth implications of this scenario are discussed
further in Section 8.7.
(TBD: Other examples? Can this section be removed?)
5. Multi-Stream Endpoint RTP Media Recommendations
While an endpoint MUST (of course) stay within its share of the
available session bandwidth, as determined by signalling and
congestion control, this need not be applied independently or
uniformly to each media stream. In particular, session bandwidth MAY
be reallocated among an endpoint's media streams, for example by
varying the bandwidth use of a variable-rate codec, or changing the
codec used by the media stream, up to the constraints of the
session's negotiated (or declared) codecs. This includes enabling or
disabling media streams as more or less bandwidth becomes available.
6. Multi-Stream Endpoint RTCP Recommendations
This section contains a number of different RTCP clarifications or
recommendations that enables more efficient and simpler behavior
without loss of functionality.
The RTP Control Protocol (RTCP) is defined in Section 6 of [RFC3550],
but it is largely documented in terms of "participants". In many
cases, the specification's recommendations for "participants" are to
be interpreted as applying to individual media streams, rather than
to endpoints. This section describes several concrete cases where
this applies.
(tbd: rather than think in terms of media streams, it might be
clearer to refer to SSRC values, where a participant with multiple
active SSRC values counts as multiple participants, once per SSRC)
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6.1. RTCP Reporting Requirement
For each of an endpoint's media streams, whether or not it is
currently sending media, SR/RR and SDES packets MUST be sent at least
once per RTCP report interval. (For discussion of the content of SR
or RR packets' reception statistic reports, see Section 8.)
6.2. Initial Reporting Interval
When a new media stream is added to a unicast session, the sentence
in [RFC3550]'s Section 6.2 applies: "For unicast sessions ... the
delay before sending the initial compound RTCP packet MAY be zero."
This applies to individual media sources as well. Thus, endpoints
MAY send an initial RTCP packet for an SSRC immediately upon adding
it to a unicast session.
This allowance also applies, as written, when initially joining a
unicast session. However, in this case some caution needs to be
exercised if the end-point or mixer has a large number of sources
(SSRCs) as this can create a significant burst. How big an issue
this depends on the number of source to send initial SR or RR and
Session Description CNAME items for in relation to the RTCP
bandwidth.
(tbd: Maybe some recommendation here? The aim in restricting this to
unicast sessions was to avoid this burst of traffic, which the usual
RTCP timing and reconsideration rules will prevent)
6.3. Compound RTCP Packets
Section 6.1 gives the following advice to RTP translators and mixers:
It is RECOMMENDED that translators and mixers combine individual
RTCP packets from the multiple sources they are forwarding into
one compound packet whenever feasible in order to amortize the
packet overhead (see Section 7). An example RTCP compound packet
as might be produced by a mixer is shown in Fig. 1. If the
overall length of a compound packet would exceed the MTU of the
network path, it SHOULD be segmented into multiple shorter
compound packets to be transmitted in separate packets of the
underlying protocol. This does not impair the RTCP bandwidth
estimation because each compound packet represents at least one
distinct participant. Note that each of the compound packets MUST
begin with an SR or RR packet.
Note: To avoid confusion, an RTCP packet is an individual item, such
as a Sender Report (SR), Receiver Report (RR), Source Description
(SDES), Goodbye (BYE), Application Defined (APP), Feedback [RFC4585]
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or Extended Report (XR) [RFC3611] packet. A compound packet is the
combination of two or more such RTCP packets where the first packet
has to be an SR or an RR packet, and which contains a SDES packet
containing an CNAME item. Thus the above results in compound RTCP
packets that contain multiple SR or RR packets from different sources
as well as any of the other packet types. There are no restrictions
on the order in which the packets can occur within the compound
packet, except the regular compound rule, i.e., starting with an SR
or RR.
This advice applies to multi-media-stream endpoints as well, with the
same restrictions and considerations. (Note, however, that the last
sentence does not apply to AVPF [RFC4585] or SAVPF [RFC5124] feedback
packets if Reduced-Size RTCP [RFC5506] is in use.)
Due to RTCP's randomization of reporting times, there is a fair bit
of tolerance in precisely when an endpoint schedules RTCP to be sent.
Thus, one potential way of implementing this recommendation would be
to randomize all of an endpoint's sources together, with a single
randomization schedule, so an MTU's worth of RTCP all comes out
simultaneously.
(tbd: Multiplexing RTCP packets from multiple different sources might
require some adjustment to the calculation of RTCP's avg_rtcp_size,
as the RTCP group interval is proportional to avg_rtcp_size times the
group size).
7. RTCP Bandwidth Considerations for Sources with Disparate Rates
It is possible for an RTP session to carry sources of greatly
differing bandwidths. One example is the scenario of
[I-D.ietf-avtcore-multi-media-rtp-session], when audio and video are
sent in the same session. However, this can occur even within a
single media type, for example a video session carrying both 5 fps
QCIF and 60 fps 1080p HD video, or an audio session carrying both
G.729 and L24/48000/6 audio.
(tbd: recommend how RTCP bandwidths are to be chosen in these
scenarios. Likely, these recommendations will be different for
sessions using AVPF-based profiles (where the trr-int parameter is
available) than for those using AVP.)
8. Grouping of RTCP Reception Statistics and Other Feedback
As specified by [RFC3550], an endpoint MUST send reception reports
about every active media stream it is receiving, from at least one
local source.
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However, a naive application of the RTP specification's rules could
be quite inefficient. In particular, if a session has N SSRCs
(active and inactive, i.e., participant SSRCs), and the session has S
active senders in each reporting interval, there will either be N*S
report blocks per reporting interval, or (per the round-robin
recommendations of [RFC3550] Section 6.1) reception sources would be
unnecessarily round-robinned. In a session where most media sources
become senders reasonably frequently, this results in quadratically
many reception report blocks in the conference, or reporting delays
proportional to the number of session members.
Since traffic is received by endpoints, however, rather than by media
sources, there is not actually any need for this quadratic expansion.
All that is needed is for each endpoint to report all the remote
sources it is receiving.
Thus, this document defines a new RTCP mechanism, Reporting Groups,
to indicate sources which originate from the same endpoint, and which
therefore would have identical recption reports.
8.1. Semantics and Behavior of Reporting Groups
An RTCP Reporting Group indicates that a set of sources (SSRCs) that
originate from a single entity (endpoint or middlebox) in an RTP
session, and therefore all the sources in the group's view of the
network is identical. If reporting groups are in use, two sources
SHOULD be put into the same reporting group if their view of the
network is identical; i.e., if they report on traffic received at the
same interface of an RTP endpoint. Sources with different views of
the network MUST NOT be put into the same reporting group.
If reporting groups are in use, an endpoint MUST NOT send reception
reports from one source in a reporting group about another one in the
same group ("self-reports"). Similarly, an endpoint MUST NOT send
reception reports about a remote media source from more than one
source in a reporting group ("cross-reports"). Instead, it MUST pick
one of its local media sources as the "reporting" source for each
remote media source, and use it to send reception reports about that
remote source; all the other media sources in the reporting group
MUST NOT send any reception reports about that remote media source.
This reporting source MUST also be the source for any RTP/AVPF
Feedback [RFC4585] or Extended Report (XR) [RFC3611] packets about
the corresponding remote sources as well. If a reporting source
leaves the session (i.e., if it sends a BYE, or leaves the group
without sending BYE under the rules of [RFC3550] section 6.3.7),
another reporting source MUST be chosen if any members of the group
still exist.
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An endpoint or middlebox MAY use multiple sources as reporting
sources; however, each reporting source MUST have non-overlapping
sets of remote SSRCs it reports on. This is primarily to be done
when the reporting source's number of reception report blocks is so
large that it would be forced to round robin around the sources.
Thus, by splitting the reports among several reporting SSRCs more
consistent reporting can be achieved.
If RTP/AVPF feedback is in use, a reporting source MAY send immediate
or early feedback at any point when any member of the reporting group
could validly do so.
An endpoint SHOULD NOT create single-source reporting groups, unless
it is anticipated that the group might have additional sources added
to it in the future.
8.2. Determine the Report Group
A remote RTP entity, such as an endpoint or a middlebox needs to be
able to determine the report group used by another RTP entity. To
achieve this goal two RTCP extensions has been defined. For the
SSRCs that are reporting on behalf of the reporting group an SDES
item RGRP has been defined for providing the report group with an
identifier. For SSRCs that aren't reporting on any peer SSRC a new
RTCP packet type is defined. This RTCP packet type "Reporting
Sources", lists the SSRC that are reporting on this SSRC's behalf.
This divided approach has been selected for the following reasons:
1. Enable an explicit indication of who reports on this SSRC's
behalf. Being explicit prevents the remote entity from detecting
that is missing the reports if there issues with the reporting
SSRC's RTCP packets.
2. Enable explicit identification of the SSRCs that are actively
reporting as one entity.
8.3. RTCP Packet Reporting Group's Reporting Sources
This section defines a new RTCP packet type called "Reporting Group's
Reporting Sources" (RGRS). It identifies the SSRC(s) that report on
behalf of the SSRC that is the sender of the RGRS packet.
This packet consists of the fixed RTCP packet header which indicates
the packet type, the number of reporting sources included and the
SSRC which behalf the reporting SSRCs report on. This is followed by
the list of reporting SSRCs.
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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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P| SC | PT=RGRS(TBA) | length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of packet sender |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
: SSRC for Reporting Source :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The RTCP Packets field has the following definition.
version (V): This field identifies the RTP version. The current
version is 2.
padding (P): 1 bit If set, the padding bit indicates that the packet
contains additional padding octets at the end that are not part of
the control information but are included in the length field. See
[RFC3550].
Source Count (SC): 5 bits Indicating the number of reporting SSRCs
(1-31) that are include in this RTCP packet type.
Payload type (PT): 8 bits This is the RTCP packet type that
identifies the packet as being an RTCP FB message. The RGRS RTCP
packet has the value [TBA].
Length: 16 bits The length of this packet in 32-bit words minus one,
including the header and any padding. This is in line with the
definition of the length field used in RTCP sender and receiver
reports [RFC3550].
SSRC of packet sender: 32 bits. The SSRC of the sender of this
packet which indicates which SSRCs that reports on its behalf,
instead of reporting itself.
SSRC for Reporting Source: A variable number (as indicated by Source
Count) of 32-bit SSRC values. Each SSRC is an reporting SSRC
belonging to the same Report Group.
Each RGRS packet MUST contain at least one reporting SSRC. In case
the reporting SSRC field is insufficient to list all the SSRCs that
is reporting in this report group, the SSRC SHALL round robin around
the reporting sources.
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Any RTP mixer or translator which forwards SR or RR packets from
members of a reporting group MUST forward the corresponding RGRS RTCP
packet as well.
8.4. RTCP Source Description (SDES) item for Reporting Groups
A new RTCP Source Description (SDES) item is defined for the purpose
of identifying reporting groups.
The Source Description (SDES) item "RGRP" is sent by a reporting
group's reporting SSRC. Syntactically, its format is the same as the
RTCP SDES CNAME item [RFC6222], and MUST be chosen with the same
global-uniqueness and privacy considerations as CNAME. This name
MUST be stable across the lifetime of the reporting group, even if
the SSRC of a reporting source changes.
Every source which belongs to a reporting group MUST either include
an RGRP SDES item in an SDES packet (if it is a reporting source), or
an RGRS packet (if it is not), in every compound RTCP packet in which
it sends an RR or SR packet (i.e., in every RTCP packet it sends,
unless Reduced-Size RTCP [RFC5506] is in use).
Any RTP mixer or translator which forwards SR or RR packets from
members of a reporting group MUST forward the corresponding RGRP SDES
items as well, even if it otherwise strips SDES items other than
CNAME.
8.5. Middlebox Considerations
This section discusses middlebox considerations for Reporting groups.
To be expanded.
8.6. SDP signaling for Reporting Groups
TBD
8.7. Bandwidth Benefits of RTCP Reporting Groups
To understand the benefits of RTCP reporting groups, consider the
scenario described in Section 4.1. This scenario describes an
environment in which the two endpoints in a session each have a
hundred sources, of which eight each are sending within any given
reporting interval.
For ease of analysis, we can make the simplifying approximation that
the duration of the RTCP reporting interval is equal to the total
size of the RTCP packets sent during an RTCP interval, divided by the
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RTCP bandwidth. (This will be approximately true in scenarios where
the bandwidth is not so high that the minimum RTCP interval is
reached.) For further simplification, we can assume RTCP senders are
following the recommendations of Section 6.3; thus, the per-packet
transport-layer overhead will be small relative to the RTCP data.
Thus, only the actual RTCP data itself need be considered.
In a report interval in this scenario, there will, as a baseline, be
200 SDES packets, 184 RR packets, and 16 SR packets. This amounts to
approximately 6.5 kB of RTCP per report interval, assuming 16-byte
CNAMEs and no other SDES information.
Using the original [RFC3550] everyone-reports-on-every-sender
feedback rules, each of the 184 receivers will send 16 report blocks,
and each of the 16 senders will send 15. This amounts to
approximately 76 kB of report block traffic per interval; 92% of RTCP
traffic consists of report blocks.
If reporting groups are used, however, there is only 0.4 kB of
reports per interval, with no loss of useful information.
Additionally, there will be (assuming 16-byte RGRPs, and a single
reporting source per reporting group) an additional 2.4 kB per cycle
of RGRP SDES items and RGRS packets. Put another way, the unmodified
[RFC3550] reporting interval is approximately 8.9 times longer than
if reporting groups are in use.
8.8. Consequences of RTCP Reporting Groups
The RTCP traffic generated by receivers using RTCP Reporting Groups
might appear, to observers unaware of these semantics, to be
generated by receivers who are experiencing a network disconnection,
as the non-reporting sources appear not to be receiving a given
sender at all.
This could be a potentially critical problem for such a sender using
RTCP for congestion control, as such a sender might think that it is
sending so much traffic that it is causing complete congestion
collapse.
However, such an interpretation of the session statistics would
require a fairly sophisticated RTCP analysis. Any receiver of RTCP
statistics which is just interested in information about itself needs
to be prepared that any given reception report might not contain
information about a specific media source, because reception reports
in large conferences can be round-robined.
Thus, it is unclear to what extent such backward compatibility issues
would actually cause trouble in practice.
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9. Security Considerations
In the secure RTP protocol (SRTP) [RFC3711], the cryptographic
context of a compound SRTCP packet is the SSRC of the sender of the
first RTCP (sub-)packet. This could matter in some cases, especially
for keying mechanisms such as Mikey [RFC3830] which use per-SSRC
keying.
Other than that, the standard security considerations of RTP apply;
sending multiple media streams from a single endpoint does not appear
to have different security consequences than sending the same number
of streams.
10. Open Issues
At this stage this document contains a number of open issues. The
below list tries to summarize the issues:
1. Further clarifications on how to handle the RTCP scheduler when
sending multiple sources in one compound packet.
2. How is the use of reporting groups be signaled in SDP?
3. How is the RTCP avg_rtcp_size be calculated when RTCP packets are
routinely multiplexed among multiple RTCP senders?
4. Do we need to provide a recommendation for unicast session
joiners with many sources to not use 0 initial minimal interval
from bit-rate burst perspective?
11. IANA Considerations
This document make several requests to IANA for registering new RTP/
RTCP identifiers.
(Note to the RFC-Editor: please replace "TBA" with the IANA-assigned
value, and "XXXX" with the number of this document, prior to
publication as an RFC.)
11.1. RTCP SDES Item
This document adds an additional SDES types to the IANA "RTCP SDES
Item Types" Registry, as follows:
Value Abbrev Name Reference
TBA RGRP Reporting Group [RFCXXXX]
Figure 1: Item for the IANA Source Attribute Registry
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11.2. RTCP Packet Type
This document defines one new RTCP Control Packet types (PT) to be
registered as follows:
Value Abbrev Name Reference
TBA RGRR Reporting Group Reporting Sources [RFCXXXX]
Figure 2: Item for the IANA RTCP Control Packet Types (PT) Registry
12. References
12.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[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, July
2006.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009.
[RFC6222] Begen, A., Perkins, C., and D. Wing, "Guidelines for
Choosing RTP Control Protocol (RTCP) Canonical Names
(CNAMEs)", RFC 6222, April 2011.
12.2. Informative References
[I-D.ietf-avtcore-multi-media-rtp-session]
Westerlund, M., Perkins, C., and J. Lennox, "Multiple
Media Types in an RTP Session", draft-ietf-avtcore-multi-
media-rtp-session-01 (work in progress), October 2012.
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Internet-Draft RTP Multi-Stream Considerations February 2013
[I-D.ietf-clue-framework]
Duckworth, M., Pepperell, A., and S. Wenger, "Framework
for Telepresence Multi-Streams", draft-ietf-clue-
framework-09 (work in progress), February 2013.
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Multiplexing Negotiation Using Session Description
Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp-
bundle-negotiation-03 (work in progress), February 2013.
[I-D.westerlund-avtcore-rtp-topologies-update]
Westerlund, M. and S. Wenger, "RTP Topologies", draft-
westerlund-avtcore-rtp-topologies-update-01 (work in
progress), October 2012.
[RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control
Protocol Extended Reports (RTCP XR)", RFC 3611, November
2003.
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
August 2004.
Appendix A. Changes From Earlier Versions
Note to the RFC-Editor: please remove this section prior to
publication as an RFC.
A.1. Changes From Draft -01
o Merged with draft-wu-avtcore-multisrc-endpoint-adver.
o Changed how Reporting Groups are indicated in RTCP, to make it
clear which source(s) is the group's reporting sources.
o Clarified the rules for when sources can be placed in the same
reporting group.
o Clarified that mixers and translators need to pass reporting group
SDES information if they are forwarding RR and SR traffic from
members of a reporting group.
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A.2. Changes From Draft -00
o Added the Reporting Group semantic to explicitly indicate which
sources come from a single endpoint, rather than leaving it
implicit.
o Specified that Reporting Group semantics (as they now are) apply
to AVPF and XR, as well as to RR/SR report blocks.
o Added a description of the cascaded source-projecting mixer, along
with a calculation of its RTCP overhead if reporting groups are
not in use.
o Gave some guidance on how the flexibility of RTCP randomization
allows some freedom in RTCP multiplexing.
o Clarified the language of several of the recommendations.
o Added an open issue discussing how avg_rtcp_size ought to be
calculated for multiplexed RTCP.
o Added an open issue discussing how RTCP bandwidths are to be
chosen for sessions where source bandwidths greatly differ.
Authors' Addresses
Jonathan Lennox
Vidyo, Inc.
433 Hackensack Avenue
Seventh Floor
Hackensack, NJ 07601
US
Email: jonathan@vidyo.com
Magnus Westerlund
Ericsson
Farogatan 6
SE-164 80 Kista
Sweden
Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com
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Qin Wu
Huawei
101 Software Avenue, Yuhua District
Nanjing, Jiangsu 210012
China
Email: sunseawq@huawei.com
Colin Perkins
University of Glasgow
School of Computing Science
Glasgow G12 8QQ
United Kingdom
Email: csp@csperkins.org
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