draft-ietf-avtcore-rtp-multi-stream-00.txt   draft-ietf-avtcore-rtp-multi-stream-01.txt 
AVTCORE J. Lennox AVTCORE J. Lennox
Internet-Draft Vidyo Internet-Draft Vidyo
Updates: 3550 (if approved) M. Westerlund Updates: 3550 (if approved) M. Westerlund
Intended status: Standards Track Ericsson Intended status: Standards Track Ericsson
Expires: October 24, 2013 Q. Wu Expires: January 12, 2014 Q. Wu
Huawei Huawei
C. Perkins C. Perkins
University of Glasgow University of Glasgow
April 22, 2013 July 11, 2013
RTP Considerations for Endpoints Sending Multiple Media Streams Sending Multiple Media Streams in a Single RTP Session
draft-ietf-avtcore-rtp-multi-stream-00 draft-ietf-avtcore-rtp-multi-stream-01
Abstract Abstract
This document expands and clarifies the behavior of the Real-Time This document expands and clarifies the behavior of the Real-Time
Transport Protocol (RTP) endpoints when they are sending multiple Transport Protocol (RTP) endpoints when they are sending multiple
media streams in a single RTP session. In particular, issues media streams in a single RTP session. In particular, issues
involving Real-Time Transport Control Protocol (RTCP) messages are involving Real-Time Transport Control Protocol (RTCP) messages are
described. described.
This document updates RFC 3550 in regards to handling of multiple This document updates RFC 3550 in regards to handling of multiple
skipping to change at page 1, line 42 skipping to change at page 1, line 42
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material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on October 24, 2013. This Internet-Draft will expire on January 12, 2014.
Copyright Notice Copyright Notice
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Use Cases For Multi-Stream Endpoints . . . . . . . . . . . . 3 3. Use Cases For Multi-Stream Endpoints . . . . . . . . . . . . 3
3.1. Multiple-Capturer Endpoints . . . . . . . . . . . . . . . 3 3.1. Multiple-Capturer Endpoints . . . . . . . . . . . . . . . 3
3.2. Multi-Media Sessions . . . . . . . . . . . . . . . . . . 4 3.2. Multi-Media Sessions . . . . . . . . . . . . . . . . . . 3
3.3. Multi-Stream Mixers . . . . . . . . . . . . . . . . . . . 4 3.3. Multi-Stream Mixers . . . . . . . . . . . . . . . . . . . 4
4. Issue Cases . . . . . . . . . . . . . . . . . . . . . . . . . 4 4. Multi-Stream Endpoint RTP Media Recommendations . . . . . . . 4
4.1. Cascaded Multi-party Conference with Source Projecting 5. Multi-Stream Endpoint RTCP Recommendations . . . . . . . . . 4
Mixers . . . . . . . . . . . . . . . . . . . . . . . . . 5 5.1. RTCP Reporting Requirement . . . . . . . . . . . . . . . 5
5. Multi-Stream Endpoint RTP Media Recommendations . . . . . . . 5 5.2. Initial Reporting Interval . . . . . . . . . . . . . . . 5
6. Multi-Stream Endpoint RTCP Recommendations . . . . . . . . . 5 5.3. Compound RTCP Packets . . . . . . . . . . . . . . . . . . 5
6.1. RTCP Reporting Requirement . . . . . . . . . . . . . . . 6 6. RTCP Considerations for Streams with Disparate Rates . . . . 7
6.2. Initial Reporting Interval . . . . . . . . . . . . . . . 6 6.1. Timing out SSRCs . . . . . . . . . . . . . . . . . . . . 8
6.3. Compound RTCP Packets . . . . . . . . . . . . . . . . . . 6 6.2. Tuning RTCP transmissions . . . . . . . . . . . . . . . . 9
7. RTCP Bandwidth Considerations for Sources with Disparate 7. Security Considerations . . . . . . . . . . . . . . . . . . . 11
Rates . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7 8. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 12
8. Grouping of RTCP Reception Statistics and Other Feedback . . 7 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 12
8.1. Semantics and Behavior of Reporting Groups . . . . . . . 8 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 12
8.2. Determine the Report Group . . . . . . . . . . . . . . . 9 10.1. Normative References . . . . . . . . . . . . . . . . . . 12
8.3. RTCP Packet Reporting Group's Reporting Sources . . . . . 9 10.2. Informative References . . . . . . . . . . . . . . . . . 13
8.4. RTCP Source Description (SDES) item for Reporting Groups 11 Appendix A. Changes From Earlier Versions . . . . . . . . . . . 14
8.5. Middlebox Considerations . . . . . . . . . . . . . . . . 11 A.1. Changes From WG Draft -00 . . . . . . . . . . . . . . . . 14
8.6. SDP signaling for Reporting Groups . . . . . . . . . . . 11 A.2. Changes From Individual Draft -02 . . . . . . . . . . . . 14
8.7. Bandwidth Benefits of RTCP Reporting Groups . . . . . . . 11 A.3. Changes From Individual Draft -01 . . . . . . . . . . . . 14
8.8. Consequences of RTCP Reporting Groups . . . . . . . . . . 12 A.4. Changes From Individual Draft -00 . . . . . . . . . . . . 14
9. Security Considerations . . . . . . . . . . . . . . . . . . . 13 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 15
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 Individual Draft -02 . . . . . . . . . . . . 15
A.2. Changes From Draft -01 . . . . . . . . . . . . . . . . . 15
A.3. Changes From Draft -00 . . . . . . . . . . . . . . . . . 16
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 16
1. Introduction 1. Introduction
At the time The Real-Time Transport Protocol (RTP) [RFC3550] was At the time The Real-Time Transport Protocol (RTP) [RFC3550] was
originally written, and for quite some time after, endpoints in RTP originally written, and for quite some time after, endpoints in RTP
sessions typically only transmitted a single media stream per RTP sessions typically only transmitted a single media stream per RTP
session, where separate RTP sessions were typically used for each session, where separate RTP sessions were typically used for each
distinct media type. distinct media type.
Recently, however, a number of scenarios have emerged (discussed Recently, however, a number of scenarios have emerged (discussed
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to the mixer, "projected" from the original streams, but at any given 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, 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 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 mixer than an RTP translator, in that it terminates RTCP reporting
about the mixed streams, and it can re-write SSRCs, timestamps, and about the mixed streams, and it can re-write SSRCs, timestamps, and
sequence numbers, as well as the contents of the RTP payloads, 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 can turn sources on and off at will without appearing to be
generating packet loss. Each projected stream will typically generating packet loss. Each projected stream will typically
preserve its original RTCP source description (SDES) information. preserve its original RTCP source description (SDES) information.
4. Issue Cases 4. Multi-Stream Endpoint RTP Media Recommendations
This section illustrates some scenarios that have shown areas where
the RTP specification is unclear.
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 While an endpoint MUST (of course) stay within its share of the
available session bandwidth, as determined by signalling and available session bandwidth, as determined by signalling and
congestion control, this need not be applied independently or congestion control, this need not be applied independently or
uniformly to each media stream. In particular, session bandwidth MAY uniformly to each media stream. In particular, session bandwidth MAY
be reallocated among an endpoint's media streams, for example by be reallocated among an endpoint's media streams, for example by
varying the bandwidth use of a variable-rate codec, or changing the varying the bandwidth use of a variable-rate codec, or changing the
codec used by the media stream, up to the constraints of the codec used by the media stream, up to the constraints of the
session's negotiated (or declared) codecs. This includes enabling or session's negotiated (or declared) codecs. This includes enabling or
disabling media streams as more or less bandwidth becomes available. disabling media streams as more or less bandwidth becomes available.
6. Multi-Stream Endpoint RTCP Recommendations 5. Multi-Stream Endpoint RTCP Recommendations
This section contains a number of different RTCP clarifications or This section contains a number of different RTCP clarifications or
recommendations that enables more efficient and simpler behavior recommendations that enables more efficient and simpler behavior
without loss of functionality. without loss of functionality.
The RTP Control Protocol (RTCP) is defined in Section 6 of [RFC3550], The RTP Control Protocol (RTCP) is defined in Section 6 of [RFC3550],
but it is largely documented in terms of "participants". In many but it is largely documented in terms of "participants". In many
cases, the specification's recommendations for "participants" are to cases, the specification's recommendations for "participants" are to
be interpreted as applying to individual media streams, rather than be interpreted as applying to individual media streams, rather than
to endpoints. This section describes several concrete cases where to endpoints. This section describes several concrete cases where
this applies. this applies.
(tbd: rather than think in terms of media streams, it might be (tbd: rather than think in terms of media streams, it might be
clearer to refer to SSRC values, where a participant with multiple clearer to refer to SSRC values, where a participant with multiple
active SSRC values counts as multiple participants, once per SSRC) active SSRC values counts as multiple participants, once per SSRC)
6.1. RTCP Reporting Requirement 5.1. RTCP Reporting Requirement
For each of an endpoint's media streams, whether or not it is 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 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 once per RTCP report interval. (For discussion of the content of SR
or RR packets' reception statistic reports, see Section 8.) or RR packets' reception statistic reports, see
[I-D.ietf-avtcore-rtp-multi-stream-optimisation].)
6.2. Initial Reporting Interval 5.2. Initial Reporting Interval
When a new media stream is added to a unicast session, the sentence When a new media stream is added to a unicast session, the sentence
in [RFC3550]'s Section 6.2 applies: "For unicast sessions ... the in [RFC3550]'s Section 6.2 applies: "For unicast sessions ... the
delay before sending the initial compound RTCP packet MAY be zero." delay before sending the initial compound RTCP packet MAY be zero."
This applies to individual media sources as well. Thus, endpoints This applies to individual media sources as well. Thus, endpoints
MAY send an initial RTCP packet for an SSRC immediately upon adding MAY send an initial RTCP packet for an SSRC immediately upon adding
it to a unicast session. it to a unicast session.
This allowance also applies, as written, when initially joining a This allowance also applies, as written, when initially joining a
unicast session. However, in this case some caution needs to be unicast session. However, in this case some caution needs to be
exercised if the end-point or mixer has a large number of sources 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 (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 this depends on the number of source to send initial SR or RR and
Session Description CNAME items for in relation to the RTCP Session Description CNAME items for in relation to the RTCP
bandwidth. bandwidth.
(tbd: Maybe some recommendation here? The aim in restricting this to (tbd: Maybe some recommendation here? The aim in restricting this to
unicast sessions was to avoid this burst of traffic, which the usual unicast sessions was to avoid this burst of traffic, which the usual
RTCP timing and reconsideration rules will prevent) RTCP timing and reconsideration rules will prevent)
6.3. Compound RTCP Packets 5.3. Compound RTCP Packets
Section 6.1 gives the following advice to RTP translators and mixers: Section 6.1 gives the following advice to RTP translators and mixers:
It is RECOMMENDED that translators and mixers combine individual It is RECOMMENDED that translators and mixers combine individual
RTCP packets from the multiple sources they are forwarding into RTCP packets from the multiple sources they are forwarding into
one compound packet whenever feasible in order to amortize the one compound packet whenever feasible in order to amortize the
packet overhead (see Section 7). An example RTCP compound packet packet overhead (see Section 7). An example RTCP compound packet
as might be produced by a mixer is shown in Fig. 1. If the 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 overall length of a compound packet would exceed the MTU of the
network path, it SHOULD be segmented into multiple shorter network path, it SHOULD be segmented into multiple shorter
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Thus, one potential way of implementing this recommendation would be Thus, one potential way of implementing this recommendation would be
to randomize all of an endpoint's sources together, with a single to randomize all of an endpoint's sources together, with a single
randomization schedule, so an MTU's worth of RTCP all comes out randomization schedule, so an MTU's worth of RTCP all comes out
simultaneously. simultaneously.
(tbd: Multiplexing RTCP packets from multiple different sources might (tbd: Multiplexing RTCP packets from multiple different sources might
require some adjustment to the calculation of RTCP's avg_rtcp_size, 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 as the RTCP group interval is proportional to avg_rtcp_size times the
group size). group size).
7. RTCP Bandwidth Considerations for Sources with Disparate Rates 6. RTCP Considerations for Streams 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.
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.
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.
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 It is possible for a single RTP session to carry streams of greatly
packet which indicates which SSRCs that reports on its behalf, differing bandwidth. There are two scenarios where this can occur.
instead of reporting itself. The first is when a single RTP session carries multiple flows of the
same media type, but with very different quality; for example a video
switching multi-point conference unit might send a full rate high-
definition video stream of the active speaker but only thumbnails for
the other participants, all sent in a single RTP session. The second
scenarios occurs when audio and video flows are sent in a single RTP
session, as discussed in [I-D.ietf-avtcore-multi-media-rtp-session].
SSRC for Reporting Source: A variable number (as indicated by Source An RTP session has a single set of parameters that configure the
Count) of 32-bit SSRC values. Each SSRC is an reporting SSRC session bandwidth, the RTCP sender and receiver fractions (e.g., via
belonging to the same Report Group. the SDP "b=RR:" and "b=RS:" lines), and the parameters of the RTP/
AVPF profile [RFC4585] (e.g., trr-int) if that profile (or its secure
extension, RTP/SAVPF [RFC5124]) is used. As a consequence, the RTCP
reporting interval will be the same for every SSRC in an RTP session.
This uniform RTCP reporting interval can result in RTCP reports being
sent more often than is considered desirable for a particular media
type. For example, if an audio flow is multiplexed with a high
quality video flow where the session bandwidth is configured to match
the video bandwidth, this can result in the RTCP packets having a
greater bandwidth allocation than the audio data rate. If the
reduced minimum RTCP interval described in Section 6.2 of [RFC3550]
is used in the session, which might be appropriate for video where
rapid feedback is wanted, the audio sources could be expected to send
RTCP packets more often than they send audio data packets. This is
most likely undesirable, and while the mismatch can be reduced
through careful tuning of the RTCP parameters, particularly trr_int
in RTP/AVPF sessions, it is inherent in the design of the RTCP timing
rules, and affects all RTP sessions containing flows with mismatched
bandwidth.
Each RGRS packet MUST contain at least one reporting SSRC. In case Having multiple media types in one RTP session also results in more
the reporting SSRC field is insufficient to list all the SSRCs that SSRCs being present in this RTP session. This increasing the amount
is reporting in this report group, the SSRC SHALL round robin around of cross reporting between the SSRCs. From an RTCP perspective, two
the reporting sources. RTP sessions with half the number of SSRCs in each will be slightly
more efficient. If someone needs either the higher efficiency due to
the lesser number of SSRCs or the fact that one can't tailor RTCP
usage per media type, they need to use independent RTP sessions.
Any RTP mixer or translator which forwards SR or RR packets from When it comes to configuring RTCP the need for regular periodic
members of a reporting group MUST forward the corresponding RGRS RTCP reporting needs to be weighted against any feedback or control
packet as well. messages being sent. Applications using RTP/AVPF or RTP/SAVPF are
RECOMMENDED to consider setting the trr-int parameter to a value
suitable for the application's needs, thus potentially reducing the
need for regular reporting and thus releasing more bandwidth for use
for feedback or control.
8.4. RTCP Source Description (SDES) item for Reporting Groups Another aspect of an RTP session with multiple media types is that
the RTCP packets, RTCP Feedback Messages, or RTCP XR metrics used
might not be applicable to all media types. Instead, all RTP/RTCP
endpoints need to correlate the media type of the SSRC being
referenced in a message or packet and only use those that apply to
that particular SSRC and its media type. Signalling solutions might
have shortcomings when it comes to indicating that a particular set
of RTCP reports or feedback messages only apply to a particular media
type within an RTP session.
A new RTCP Source Description (SDES) item is defined for the purpose 6.1. Timing out SSRCs
of identifying reporting groups.
The Source Description (SDES) item "RGRP" is sent by a reporting All SSRCs used in an RTP session MUST use the same timeout behaviour
group's reporting SSRC. Syntactically, its format is the same as the to avoid premature timeouts. This will depend on the RTP profile and
RTCP SDES CNAME item [RFC6222], and MUST be chosen with the same its configuration. The RTP specification provides several options
global-uniqueness and privacy considerations as CNAME. This name that can influence the values used when calculating the time
MUST be stable across the lifetime of the reporting group, even if interval. To avoid interoperability issues when using this
the SSRC of a reporting source changes. specification, this document makes several clarifications to the
calculations.
Every source which belongs to a reporting group MUST either include For RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF with T_rr_interval =
an RGRP SDES item in an SDES packet (if it is a reporting source), or 0, the timeout interval SHALL be calculated using a multiplier of 5,
an RGRS packet (if it is not), in every compound RTCP packet in which i.e. the timeout interval becomes 5*Td. The Td calculation SHALL be
it sends an RR or SR packet (i.e., in every RTCP packet it sends, done using a Tmin value of 5 seconds, not the reduced minimal
unless Reduced-Size RTCP [RFC5506] is in use). interval even if used to calculate RTCP packet transmission
intervals. If using either the RTP/AVPF or RTP/SAVPF profiles with
T_rr_interval != 0 then the calculation as specified in Section 3.5.4
of RFC 4585 SHALL be used with a multiplier of 5, i.e. Tmin in the
Td calculation is the T_rr_interval.
Any RTP mixer or translator which forwards SR or RR packets from Note: If endpoints implementing the RTP/AVP and RTP/AVPF profiles (or
members of a reporting group MUST forward the corresponding RGRP SDES their secure variants) are combined in a single RTP session, and the
items as well, even if it otherwise strips SDES items other than RTP/AVPF endpoints use a non-zero T_rr_interval that is significantly
CNAME. lower than 5 seconds, then there is a risk that the RTP/AVP endpoints
will prematurely timeout the RTP/AVPF endpoints due to their
different RTCP timeout intervals. Since an RTP session can only use
a single RTP profile, this issue ought never occur. If such mixed
RTP profiles are used, however, the RTP/AVPF session MUST NOT use a
non-zero T_rr_interval that is smaller than 5 seconds.
8.5. Middlebox Considerations (tbd: it has been suggested that a minimum non-zero T_rr_interval of
4 seconds is more appropriate, due to the nature of the timing
rules).
This section discusses middlebox considerations for Reporting groups. 6.2. Tuning RTCP transmissions
To be expanded. This sub-section discusses what tuning can be done to reduce the
downsides of the shared RTCP packet intervals.
8.6. SDP signaling for Reporting Groups When using the RTP/AVP or RTP/SAVP profiles the tuning one can do is
very limited. The controls one has are limited to the RTCP bandwidth
values and whether the minimum RTCP interval is scaled according to
the bandwidth. As the scheduling algorithm includes both random
factors and reconsideration, one can't simply calculate the expected
average transmission interval using the formula for Td. But it does
indicate the important factors affecting the transmission interval,
namely the RTCP bandwidth available for the role (Active Sender or
Participant), the average RTCP packet size, and the number of SSRCs
classified in the relevant role. Note that if the ratio of senders
to total number of session participants is larger than the ratio of
RTCP bandwidth for senders in relation to the total RTCP bandwidth,
then senders and receivers are treated together.
TBD Let's start with some basic observations:
8.7. Bandwidth Benefits of RTCP Reporting Groups a. Unless the scaled minimum RTCP interval is used, then Td prior to
randomization and reconsideration can never be less than 5
seconds (assuming default Tmin of 5 seconds).
To understand the benefits of RTCP reporting groups, consider the b. If the scaled minimum RTCP interval is used, Td can become as low
scenario described in Section 4.1. This scenario describes an as 360 divided by RTP Session bandwidth in kilobits. In SDP the
environment in which the two endpoints in a session each have a RTP session bandwidth is signalled using b=AS. An RTP Session
hundred sources, of which eight each are sending within any given bandwidth of 72 kbps results in Tmin being 5 seconds. An RTP
reporting interval. session bandwidth of 360 kbps of course gives a Tmin of 1 second,
and to achieve a Tmin equal to once every frame for a 25 Hz video
stream requires an RTP session bandwidth of 9 Mbps! (The use of
the RTP/AVPF or RTP/SAVPF profile allows a smaller Tmin, and
hence more frequent RTCP reports, as discussed below).
For ease of analysis, we can make the simplifying approximation that c. Let's calculate the number (n) of SSRCs in the RTP session that
the duration of the RTCP reporting interval is equal to the total 5% of the session bandwidth can support to yield a Td value equal
size of the RTCP packets sent during an RTCP interval, divided by the to Tmin with minimal scaling. For this calculation we have to
RTCP bandwidth. (This will be approximately true in scenarios where make two assumptions. The first is that we will consider most or
the bandwidth is not so high that the minimum RTCP interval is all SSRC being senders, resulting in everyone sharing the
reached.) For further simplification, we can assume RTCP senders are available bandwidth. Secondly we will select an average RTCP
following the recommendations of Section 6.3; thus, the per-packet packet size. This packet will consist of an SR, containing (n-1)
transport-layer overhead will be small relative to the RTCP data. report blocks up to 31 report blocks, and an SDES item with at
Thus, only the actual RTCP data itself need be considered. least a CNAME (17 bytes in size) in it. Such a basic packet will
be 800 bytes for n>=32. With these parameters, and as the
bandwidth goes up the time interval is proportionally decreased
(due to minimal scaling), thus all the example bandwidths 72
kbps, 360 kbps and 9 Mbps all support 9 SSRCs.
In a report interval in this scenario, there will, as a baseline, be d. The actual transmission interval for a Td value is [0.5*Td/
200 SDES packets, 184 RR packets, and 16 SR packets. This amounts to 1.21828,1.5*Td/1.21828], which means that for Td = 5 seconds, the
approximately 6.5 kB of RTCP per report interval, assuming 16-byte interval is actually [2.052,6.156] and the distribution is not
CNAMEs and no other SDES information. uniform, but rather exponentially-increasing. The probability
for sending at time X, given it is within the interval, is
probability of picking X in the interval times the probability to
randomly picking a number that is <=X within the interval with an
uniform probability distribution. This results in that the
majority of the probability mass is above the Td value.
Using the original [RFC3550] everyone-reports-on-every-sender To conclude, with RTP/AVP and RTP/SAVP the key limitation for small
feedback rules, each of the 184 receivers will send 16 report blocks, unicast sessions is going to be the Tmin value. Thus the RTP session
and each of the 16 senders will send 15. This amounts to bandwidth configured in RTCP has to be sufficiently high to reach the
approximately 76 kB of report block traffic per interval; 92% of RTCP reporting goals the application has following the rules for the
traffic consists of report blocks. scaled minimal RTCP interval.
If reporting groups are used, however, there is only 0.4 kB of When using RTP/AVPF or RTP/SAVPF we get a quite powerful additional
reports per interval, with no loss of useful information. tool, the setting of the T_rr_interval which has several effects on
Additionally, there will be (assuming 16-byte RGRPs, and a single the RTCP reporting. First of all as Tmin is set to 0 after the
reporting source per reporting group) an additional 2.4 kB per cycle initial transmission, the regular reporting interval is instead
of RGRP SDES items and RGRS packets. Put another way, the unmodified determined by the regular bandwidth based calculation and the
[RFC3550] reporting interval is approximately 8.9 times longer than T_rr_interval. This has the effect that we are no longer restricted
if reporting groups are in use. by the minimal interval or even the scaling rule for the minimal
rule. Instead the RTCP bandwidth and the T_rr_interval are the
governing factors. Now it also becomes important to separate between
the application's need for regular reports and RTCP feedback packet
types. In both regular RTCP mode, as in Early RTCP Mode, the usage
of the T_rr_interval prevents regular RTCP packets, i.e. packets
without any Feedback packets, to be sent more often than
T_rr_interval. This value is a hard as no regular RTCP packet can be
sent less than T_rr_interval after the previous regular packet
packet.
8.8. Consequences of RTCP Reporting Groups So applications that have a use for feedback packets for some media
streams, for example video streams, but don't want frequent regular
reporting for audio, could configure the T_rr_interval to a value so
that the regular reporting for both audio and video is at a level
that is considered acceptable for the audio. They could then use
feedback packets, which will include RTCP SR/RR packets, unless
reduced-size RTCP feedback packets [RFC5506] are used, and can
include other report information in addition to the feedback packet
that needs to be sent. That way the available RTCP bandwidth can be
focused for the use which provides the most utility for the
application.
The RTCP traffic generated by receivers using RTCP Reporting Groups Using T_rr_interval still requires one to determine suitable values
might appear, to observers unaware of these semantics, to be for the RTCP bandwidth value, in fact it might make it even more
generated by receivers who are experiencing a network disconnection, important, as this is more likely to affect the RTCP behaviour and
as the non-reporting sources appear not to be receiving a given performance than when using RTP/AVP, as there are fewer limitations
sender at all. affecting the RTCP transmission.
This could be a potentially critical problem for such a sender using When using T_rr_interval, i.e. having it be non zero, there are
RTCP for congestion control, as such a sender might think that it is configurations that have to be avoided. If the resulting Td value is
sending so much traffic that it is causing complete congestion smaller but close to T_rr_interval then the interval in which the
collapse. actual regular RTCP packet transmission falls into becomes very
large, from 0.5 times T_rr_interval up to 2.73 times the
T_rr_interval. Therefore for configuration where one intends to have
Td smaller than T_rr_interval, then Td is RECOMMENDED to be targeted
at values less than 1/4th of T_rr_interval which results in that the
range becomes [0.5*T_rr_interval, 1.81*T_rr_interval].
However, such an interpretation of the session statistics would With RTP/AVPF, using a T_rr_interval of 0 or with another low value
require a fairly sophisticated RTCP analysis. Any receiver of RTCP significantly lower than Td still has utility, and different
statistics which is just interested in information about itself needs behaviour compared to RTP/AVP. This avoids the Tmin limitations of
to be prepared that any given reception report might not contain RTP/AVP, thus allowing more frequent regular RTCP reporting. In fact
information about a specific media source, because reception reports this will result that the RTCP traffic becomes as high as the
in large conferences can be round-robined. configured values.
Thus, it is unclear to what extent such backward compatibility issues (tbd: a future version of this memo will include examples of how to
would actually cause trouble in practice. choose RTCP parameters for common scenarios)
9. Security Considerations There exists no method within the specification for using different
regular RTCP reporting intervals depending on the media type or
individual media stream.
7. Security Considerations
In the secure RTP protocol (SRTP) [RFC3711], the cryptographic In the secure RTP protocol (SRTP) [RFC3711], the cryptographic
context of a compound SRTCP packet is the SSRC of the sender of the 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 first RTCP (sub-)packet. This could matter in some cases, especially
for keying mechanisms such as Mikey [RFC3830] which use per-SSRC for keying mechanisms such as Mikey [RFC3830] which use per-SSRC
keying. keying.
Other than that, the standard security considerations of RTP apply; Other than that, the standard security considerations of RTP apply;
sending multiple media streams from a single endpoint does not appear sending multiple media streams from a single endpoint does not appear
to have different security consequences than sending the same number to have different security consequences than sending the same number
of streams. of streams.
10. Open Issues 8. Open Issues
At this stage this document contains a number of open issues. The At this stage this document contains a number of open issues. The
below list tries to summarize the issues: below list tries to summarize the issues:
1. Further clarifications on how to handle the RTCP scheduler when 1. Further clarifications on how to handle the RTCP scheduler when
sending multiple sources in one compound packet. sending multiple sources in one compound packet.
2. How is the use of reporting groups be signaled in SDP? 2. How is the RTCP avg_rtcp_size be calculated when RTCP packets are
3. How is the RTCP avg_rtcp_size be calculated when RTCP packets are
routinely multiplexed among multiple RTCP senders? routinely multiplexed among multiple RTCP senders?
4. Do we need to provide a recommendation for unicast session 3. Do we need to provide a recommendation for unicast session
joiners with many sources to not use 0 initial minimal interval joiners with many sources to not use 0 initial minimal interval
from bit-rate burst perspective? from bit-rate burst perspective?
11. IANA Considerations 9. 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
11.2. RTCP Packet Type
This document defines one new RTCP Control Packet types (PT) to be
registered as follows:
Value Abbrev Name Reference No IANA actions needed.
TBA RGRR Reporting Group Reporting Sources [RFCXXXX]
Figure 2: Item for the IANA RTCP Control Packet Types (PT) Registry 10. References
12. References 10.1. Normative References
12.1. Normative References [I-D.ietf-avtcore-6222bis]
Begen, A., Perkins, C., Wing, D., and E. Rescorla,
"Guidelines for Choosing RTP Control Protocol (RTCP)
Canonical Names (CNAMEs)", draft-ietf-avtcore-6222bis-04
(work in progress), June 2013.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997. Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003. Applications", STD 64, RFC 3550, July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)", Norrman, "The Secure Real-time Transport Protocol (SRTP)",
skipping to change at page 14, line 43 skipping to change at page 13, line 22
2006. 2006.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008. (RTP/SAVPF)", RFC 5124, February 2008.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009. and Consequences", RFC 5506, April 2009.
[RFC6222] Begen, A., Perkins, C., and D. Wing, "Guidelines for 10.2. Informative References
Choosing RTP Control Protocol (RTCP) Canonical Names
(CNAMEs)", RFC 6222, April 2011.
12.2. Informative References
[I-D.ietf-avtcore-multi-media-rtp-session] [I-D.ietf-avtcore-multi-media-rtp-session]
Westerlund, M., Perkins, C., and J. Lennox, "Multiple Westerlund, M., Perkins, C., and J. Lennox, "Multiple
Media Types in an RTP Session", draft-ietf-avtcore-multi- Media Types in an RTP Session", draft-ietf-avtcore-multi-
media-rtp-session-02 (work in progress), February 2013. media-rtp-session-02 (work in progress), February 2013.
[I-D.ietf-avtcore-rtp-multi-stream-optimisation]
Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session:
Grouping RTCP Reception Statistics and Other Feedback ",
draft-ietf-avtcore-rtp-multi-stream-optimisation-00 (work
in progress), July 2013.
[I-D.ietf-avtcore-rtp-topologies-update] [I-D.ietf-avtcore-rtp-topologies-update]
Westerlund, M. and S. Wenger, "RTP Topologies", draft- Westerlund, M. and S. Wenger, "RTP Topologies", draft-
ietf-avtcore-rtp-topologies-update-00 (work in progress), ietf-avtcore-rtp-topologies-update-00 (work in progress),
April 2013. April 2013.
[I-D.ietf-clue-framework] [I-D.ietf-clue-framework]
Duckworth, M., Pepperell, A., and S. Wenger, "Framework Duckworth, M., Pepperell, A., and S. Wenger, "Framework
for Telepresence Multi-Streams", draft-ietf-clue- for Telepresence Multi-Streams", draft-ietf-clue-
framework-09 (work in progress), February 2013. framework-10 (work in progress), May 2013.
[I-D.ietf-mmusic-sdp-bundle-negotiation] [I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings, Holmberg, C., Alvestrand, H., and C. Jennings,
"Multiplexing Negotiation Using Session Description "Multiplexing Negotiation Using Session Description
Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp- Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp-
bundle-negotiation-03 (work in progress), February 2013. bundle-negotiation-04 (work in progress), June 2013.
[RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control [RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control
Protocol Extended Reports (RTCP XR)", RFC 3611, November Protocol Extended Reports (RTCP XR)", RFC 3611, November
2003. 2003.
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K. [RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830, Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
August 2004. August 2004.
Appendix A. Changes From Earlier Versions Appendix A. Changes From Earlier Versions
Note to the RFC-Editor: please remove this section prior to Note to the RFC-Editor: please remove this section prior to
publication as an RFC. publication as an RFC.
A.1. Changes From Individual Draft -02 A.1. Changes From WG Draft -00
o Split the Reporting Group Extension from this draft into draft-
ietf-avtcore-rtp-multi-stream-optimization-00.
o Added RTCP tuning considerations from draft-ietf-avtcore-multi-
media-rtp-session-02.
A.2. Changes From Individual Draft -02
o Resubmitted as working group draft. o Resubmitted as working group draft.
o Updated references. o Updated references.
A.2. Changes From Draft -01 A.3. Changes From Individual Draft -01
o Merged with draft-wu-avtcore-multisrc-endpoint-adver. o Merged with draft-wu-avtcore-multisrc-endpoint-adver.
o Changed how Reporting Groups are indicated in RTCP, to make it o Changed how Reporting Groups are indicated in RTCP, to make it
clear which source(s) is the group's reporting sources. clear which source(s) is the group's reporting sources.
o Clarified the rules for when sources can be placed in the same o Clarified the rules for when sources can be placed in the same
reporting group. reporting group.
o Clarified that mixers and translators need to pass reporting group o Clarified that mixers and translators need to pass reporting group
SDES information if they are forwarding RR and SR traffic from SDES information if they are forwarding RR and SR traffic from
members of a reporting group. members of a reporting group.
A.3. Changes From Draft -00 A.4. Changes From Individual Draft -00
o Added the Reporting Group semantic to explicitly indicate which o Added the Reporting Group semantic to explicitly indicate which
sources come from a single endpoint, rather than leaving it sources come from a single endpoint, rather than leaving it
implicit. implicit.
o Specified that Reporting Group semantics (as they now are) apply o Specified that Reporting Group semantics (as they now are) apply
to AVPF and XR, as well as to RR/SR report blocks. to AVPF and XR, as well as to RR/SR report blocks.
o Added a description of the cascaded source-projecting mixer, along o Added a description of the cascaded source-projecting mixer, along
with a calculation of its RTCP overhead if reporting groups are with a calculation of its RTCP overhead if reporting groups are
 End of changes. 56 change blocks. 
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