draft-ietf-avtcore-rtp-multi-stream-02.txt   draft-ietf-avtcore-rtp-multi-stream-03.txt 
AVTCORE J. Lennox AVTCORE J. Lennox
Internet-Draft Vidyo Internet-Draft Vidyo
Updates: 3550 (if approved) M. Westerlund Updates: 3550, 4585 (if approved) M. Westerlund
Intended status: Standards Track Ericsson Intended status: Standards Track Ericsson
Expires: July 17, 2014 Q. Wu Expires: August 18, 2014 Q. Wu
Huawei Huawei
C. Perkins C. Perkins
University of Glasgow University of Glasgow
January 13, 2014 February 14, 2014
Sending Multiple Media Streams in a Single RTP Session Sending Multiple Media Streams in a Single RTP Session
draft-ietf-avtcore-rtp-multi-stream-02 draft-ietf-avtcore-rtp-multi-stream-03
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 using multiple
media streams in a single RTP session. In particular, issues synchronization sources (SSRCs), e.g. for sending multiple media
involving RTP Control Protocol (RTCP) messages are described. streams, in a single RTP session. In particular, issues involving
RTCP Control Protocol (RTCP) messages are described.
This document updates RFC 3550 in regards to handling of multiple This document updates RFC 3550 in regards to handling of multiple
SSRCs per endpoint in RTP sessions. SSRCs per endpoint in RTP sessions. It also updates RFC 4585 to
clarify the calculation of the timeout of SSRCs and the inclusion of
feeback messages.
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
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Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
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time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
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This Internet-Draft will expire on July 17, 2014. This Internet-Draft will expire on August 18, 2014.
Copyright Notice Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Use Cases For Multi-Stream Endpoints . . . . . . . . . . . . 3 3. Use Cases For Multi-Stream Endpoints . . . . . . . . . . . . 4
3.1. Multiple-Capturer Endpoints . . . . . . . . . . . . . . . 3 3.1. Multiple-Capturer Endpoints . . . . . . . . . . . . . . . 4
3.2. Multi-Media Sessions . . . . . . . . . . . . . . . . . . 3 3.2. Multi-Media Sessions . . . . . . . . . . . . . . . . . . 4
3.3. Multi-Stream Mixers . . . . . . . . . . . . . . . . . . . 4 3.3. Multi-Stream Mixers . . . . . . . . . . . . . . . . . . . 4
4. Multi-Stream Endpoint RTP Media Recommendations . . . . . . . 4 3.4. Multiple SSRCs for a Single Media Source . . . . . . . . 5
5. Multi-Stream Endpoint RTCP Recommendations . . . . . . . . . 4 4. Multi-Stream Endpoint RTP Media Recommendations . . . . . . . 5
5. Multi-Stream Endpoint RTCP Recommendations . . . . . . . . . 5
5.1. RTCP Reporting Requirement . . . . . . . . . . . . . . . 5 5.1. RTCP Reporting Requirement . . . . . . . . . . . . . . . 5
5.2. Initial Reporting Interval . . . . . . . . . . . . . . . 5 5.2. Initial Reporting Interval . . . . . . . . . . . . . . . 6
5.3. Compound RTCP Packets . . . . . . . . . . . . . . . . . . 5 5.3. Compound RTCP Packets . . . . . . . . . . . . . . . . . . 6
6. RTCP Considerations for Streams with Disparate Rates . . . . 6 5.3.1. Maintaining AVG_RTCP_SIZE . . . . . . . . . . . . . . 7
6.1. Timing out SSRCs . . . . . . . . . . . . . . . . . . . . 8 5.3.2. Scheduling RTCP with Multiple Reporting SSRCs . . . . 8
6.2. Tuning RTCP transmissions . . . . . . . . . . . . . . . . 8 5.4. RTP/AVPF Feedback Packets . . . . . . . . . . . . . . . . 10
7. Security Considerations . . . . . . . . . . . . . . . . . . . 11 5.4.1. The SSRC Used . . . . . . . . . . . . . . . . . . . . 10
8. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 11 5.4.2. Scheduling a Feedback Packet . . . . . . . . . . . . 11
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 11 6. RTCP Considerations for Streams with Disparate Rates . . . . 12
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 12 6.1. Timing out SSRCs . . . . . . . . . . . . . . . . . . . . 13
10.1. Normative References . . . . . . . . . . . . . . . . . . 12 6.2. Tuning RTCP transmissions . . . . . . . . . . . . . . . . 14
10.2. Informative References . . . . . . . . . . . . . . . . . 12 6.2.1. RTP/AVP and RTP/SAVP . . . . . . . . . . . . . . . . 14
Appendix A. Changes From Earlier Versions . . . . . . . . . . . 13 6.2.2. RT/AVPF and RTP/SAVPF . . . . . . . . . . . . . . . . 16
A.1. Changes From WG Draft -00 . . . . . . . . . . . . . . . . 13 7. Security Considerations . . . . . . . . . . . . . . . . . . . 17
A.2. Changes From Individual Draft -02 . . . . . . . . . . . . 13 8. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 17
A.3. Changes From Individual Draft -01 . . . . . . . . . . . . 14 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 18
A.4. Changes From Individual Draft -00 . . . . . . . . . . . . 14 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 18
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 14 10.1. Normative References . . . . . . . . . . . . . . . . . . 18
10.2. Informative References . . . . . . . . . . . . . . . . . 18
Appendix A. Changes From Earlier Versions . . . . . . . . . . . 19
A.1. Changes From WG Draft -02 . . . . . . . . . . . . . . . . 20
A.2. Changes From WG Draft -01 . . . . . . . . . . . . . . . . 20
A.3. Changes From WG Draft -00 . . . . . . . . . . . . . . . . 20
A.4. Changes From Individual Draft -02 . . . . . . . . . . . . 20
A.5. Changes From Individual Draft -01 . . . . . . . . . . . . 20
A.6. Changes From Individual Draft -00 . . . . . . . . . . . . 21
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 21
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, and thus
session, where separate RTP sessions were typically used for each used a single synchronization source (SSRC) per RTP session, where
distinct media type. separate RTP sessions were typically used for each distinct media
type.
Recently, however, a number of scenarios have emerged (discussed Recently, however, a number of scenarios have emerged (discussed
further in Section 3) in which endpoints wish to send multiple RTP further in Section 3) in which endpoints wish to send multiple RTP
media streams, distinguished by distinct RTP synchronization source media streams, distinguished by distinct RTP synchronization source
(SSRC) identifiers, in a single RTP session. Although RTP's initial (SSRC) identifiers, in a single RTP session. Although RTP's initial
design did consider such scenarios, the specification was not design did consider such scenarios, the specification was not
consistently written with such use cases in mind. The specifications consistently written with such use cases in mind. The specifications
are thus somewhat unclear. are thus somewhat unclear.
The purpose of this document is to expand and clarify [RFC3550]'s The purpose of this document is to expand and clarify [RFC3550]'s
language for these use cases. The authors believe this does not language for these use cases. The authors believe this does not
result in any major normative changes to the RTP specification, result in any major normative changes to the RTP specification,
however this document defines how the RTP specification is to be however this document defines how the RTP specification is to be
interpreted. In these cases, this document updates RFC3550. interpreted. In these cases, this document updates RFC3550. The
document also updates RFC 4585 in regards to the timeout of inactive
SSRCs as specificed in Section 6.1 as well as clarifying the
inclusion of feedback messages.
The document starts with terminology and some use cases where The document starts with terminology and some use cases where
multiple sources will occur. This is followed by some case studies multiple sources will occur. This is followed by RTP and RTCP
to try to identify issues that exist and need considerations. This recommendations to resolve issues. Next are security considerations
is followed by RTP and RTCP recommendations to resolve issues. Next and remaining open issues.
are security considerations and remaining open issues.
2. Terminology 2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in RFC "OPTIONAL" in this document are to be interpreted as described in RFC
2119 [RFC2119] and indicate requirement levels for compliant 2119 [RFC2119] and indicate requirement levels for compliant
implementations. implementations.
3. Use Cases For Multi-Stream Endpoints 3. Use Cases For Multi-Stream Endpoints
This section discusses several use cases that have motivated the This section discusses several use cases that have motivated the
development of endpoints that send multiple streams in a single RTP development of endpoints that sends RTP data using multiple SSRCs in
session. a single RTP session.
3.1. Multiple-Capturer Endpoints 3.1. Multiple-Capturer Endpoints
The most straightforward motivation for an endpoint to send multiple The most straightforward motivation for an endpoint to send multiple
media streams in a session is the scenario where an endpoint has RTP streams in a session is the scenario where an endpoint has
multiple capture devices of the same media type and characteristics. multiple capture devices, and thus media sources, of the same media
For example, telepresence endpoints, of the type described by the type and characteristics. For example, telepresence endpoints, of
CLUE Telepresence Framework [I-D.ietf-clue-framework] is designed, the type described by the CLUE Telepresence Framework
often have multiple cameras or microphones covering various areas of [I-D.ietf-clue-framework], often have multiple cameras or microphones
a room. covering various areas of a room.
3.2. Multi-Media Sessions 3.2. Multi-Media Sessions
Recent work has been done in RTP Recent work has been done in RTP
[I-D.ietf-avtcore-multi-media-rtp-session] and SDP [I-D.ietf-avtcore-multi-media-rtp-session] and SDP
[I-D.ietf-mmusic-sdp-bundle-negotiation] to update RTP's historical [I-D.ietf-mmusic-sdp-bundle-negotiation] to update RTP's historical
assumption that media streams of different media types would always assumption that media sources of different media types would always
be sent on different RTP sessions. In this work, a single endpoint's 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 audio and video RTP media streams (for example) are instead sent in a
single RTP session. single RTP session.
3.3. Multi-Stream Mixers 3.3. Multi-Stream Mixers
There are several RTP topologies which can involve a central device There are several RTP topologies which can involve a central device
that itself generates multiple media streams in a session. that itself generates multiple RTP media streams in a session.
One example is a mixer providing centralized compositing for a multi- One example is a mixer providing centralized compositing for a multi-
capture scenario like that described in Section 3.1. In this case, capture scenario like that described in Section 3.1. In this case,
the centralized node is behaving much like a multi-capturer endpoint, the centralized node is behaving much like a multi-capturer endpoint,
generating several similar and related sources. generating several similar and related sources.
More complicated is the Source Projecting Mixer, see Section 3.6 of More complicated is the Selective Forwarding Middlebox, see
[I-D.ietf-avtcore-rtp-topologies-update]. This is a central box that Section 3.7 of [I-D.ietf-avtcore-rtp-topologies-update]. This is a
receives media streams from several endpoints, and then selectively middlebox that receives media streams from several endpoints, and
forwards modified versions of some of the streams toward the other then selectively forwards modified versions of some of the streams
endpoints it is connected to. Toward one destination, a separate toward the other endpoints it is connected to. Toward one
media source appears in the session for every other source connected destination, a separate media source appears in the session for every
to the mixer, "projected" from the original streams, but at any given other source connected to the middlebox, "projected" from the
time many of them can appear to be inactive (and thus are receivers, original streams, but at any given time many of them can appear to be
not senders, in RTP). This sort of device is closer to being an RTP inactive (and thus are receivers, not senders, in RTP). This sort of
mixer than an RTP translator, in that it terminates RTCP reporting device is closer to being an RTP mixer than an RTP translator, in
about the mixed streams, and it can re-write SSRCs, timestamps, and that it terminates RTCP reporting about the mixed streams, and it can
sequence numbers, as well as the contents of the RTP payloads, and re-write SSRCs, timestamps, and sequence numbers, as well as the
can turn sources on and off at will without appearing to be contents of the RTP payloads, and can turn sources on and off at will
generating packet loss. Each projected stream will typically without appearing to be generating packet loss. Each projected
preserve its original RTCP source description (SDES) information. stream will typically preserve its original RTCP source description
(SDES) information.
3.4. Multiple SSRCs for a Single Media Source
There are also several cases where a single media source results in
the usage of multiple SSRCs within the same RTP session. Transport
robustification tools like RTP Retransmission [RFC4588] result in
multiple SSRCs, one with source data, and another with the repair
data. Scalable encoders and their RTP payload foramts, like H.264's
extension for Scalable Video Coding(SVC) [RFC6190] can be transmitted
in a configuration where the scalable layers are distributed over
multiple SSRCs within the same session, to enable RTP packet stream
level (SSRC) selection and routing in conferencing middleboxes.
4. Multi-Stream Endpoint RTP Media Recommendations 4. 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 and its SSRCs. In particular, session
be reallocated among an endpoint's media streams, for example by bandwidth MAY be reallocated among an endpoint's SSRCs, for example
varying the bandwidth use of a variable-rate codec, or changing the by varying the bandwidth use of a variable-rate codec, or changing
codec used by the media stream, up to the constraints of the 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 and their redundancy streams as more or less
bandwidth becomes available.
5. 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 SSRCs, rather than to
to endpoints. This section describes several concrete cases where endpoints. This section describes several concrete cases where this
this applies. 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)
5.1. 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 SSRCs, whether or not they are currently
currently sending media, SR/RR and SDES packets MUST be sent at least sending media, SR/RR and SDES packets MUST be sent at least once per
once per RTCP report interval. (For discussion of the content of SR RTCP report interval. (For discussion of the content of SR or RR
or RR packets' reception statistic reports, see packets' reception statistic reports, see
[I-D.ietf-avtcore-rtp-multi-stream-optimisation].) [I-D.ietf-avtcore-rtp-multi-stream-optimisation].)
5.2. Initial Reporting Interval 5.2. Initial Reporting Interval
When a new media stream is added to a unicast session, the sentence When a new SSRC is added to a unicast session, the sentence in
in [RFC3550]'s Section 6.2 applies: "For unicast sessions ... the [RFC3550]'s Section 6.2 applies: "For unicast sessions ... the delay
delay before sending the initial compound RTCP packet MAY be zero." before sending the initial compound RTCP packet MAY be zero." This
This applies to individual media sources as well. Thus, endpoints applies to individual SSRCs as well. Thus, endpoints MAY send an
MAY send an initial RTCP packet for an SSRC immediately upon adding initial RTCP packet for an SSRC immediately upon adding it to a
it to a unicast session. 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 is depends on the number of sources for which the initial SR or
Session Description CNAME items for in relation to the RTCP RR packets and Session Description CNAME items are to be sent, in
bandwidth. relation to the RTCP 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.)
5.3. Compound RTCP Packets 5.3. Compound RTCP Packets
Section 6.1 gives the following advice to RTP translators and mixers: Section 6.1 in [RFC3550] 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
compound packets to be transmitted in separate packets of the compound packets to be transmitted in separate packets of the
underlying protocol. This does not impair the RTCP bandwidth underlying protocol. This does not impair the RTCP bandwidth
estimation because each compound packet represents at least one estimation because each compound packet represents at least one
distinct participant. Note that each of the compound packets MUST distinct participant. Note that each of the compound packets MUST
begin with an SR or RR packet. begin with an SR or RR packet."
Note: To avoid confusion, an RTCP packet is an individual item, such Note: To avoid confusion, an RTCP packet is an individual item,
as a Sender Report (SR), Receiver Report (RR), Source Description such as a Sender Report (SR), Receiver Report (RR), Source
(SDES), Goodbye (BYE), Application Defined (APP), Feedback [RFC4585] Description (SDES), Goodbye (BYE), Application Defined (APP),
or Extended Report (XR) [RFC3611] packet. A compound packet is the Feedback [RFC4585] or Extended Report (XR) [RFC3611] packet. A
combination of two or more such RTCP packets where the first packet compound packet is the combination of two or more such RTCP
has to be an SR or an RR packet, and which contains a SDES packet packets where the first packet has to be an SR or an RR packet,
containing an CNAME item. Thus the above results in compound RTCP and which contains a SDES packet containing an CNAME item.
packets that contain multiple SR or RR packets from different sources
as well as any of the other packet types. There are no restrictions The above results in compound RTCP packets that contain multiple SR
on the order in which the packets can occur within the compound or RR packets from different sources (SSRCs) as well as any of the
packet, except the regular compound rule, i.e., starting with an SR other packet types. There are no restrictions on the order in which
or RR. 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 This advice applies to multi-media-stream endpoints as well, with the
same restrictions and considerations. (Note, however, that the last same restrictions and considerations. (Note, however, that the last
sentence does not apply to AVPF [RFC4585] or SAVPF [RFC5124] feedback sentence does not apply to AVPF [RFC4585] or SAVPF [RFC5124] feedback
packets if Reduced-Size RTCP [RFC5506] is in use.) packets if Reduced-Size RTCP [RFC5506] is in use.)
Due to RTCP's randomization of reporting times, there is a fair bit 5.3.1. Maintaining AVG_RTCP_SIZE
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 When multiple local SSRCs are sending their RTCP packets in the same
require some adjustment to the calculation of RTCP's avg_rtcp_size, compound packet, this obviously results in larger RTCP compound
as the RTCP group interval is proportional to avg_rtcp_size times the packets. This will have an affect on the value of the average RTCP
group size). packet size metering (avg_rtcp_size) that is done for the purpose of
RTCP transmission scheduling calculation. This section discusses the
impact of this and provide recommendations with how to deal with it.
This section will use the concept of an 'RTCP Compound Packet' to
represent not just proper RTCP compound packets, i.e. ones that start
with an SR or RR RTCP packet and include at least one SDES CNAME
item. For the purpose of the below calculation, other valid lower
layer datagram units an RTCP implementation can send or receive,
independently if they are an aggregate or not of RTCP packets are
also considered. This especially includes Reduced-Size RTCP packets
[RFC5506].
The RTCP packet scheduling algorithm that is defined in RTP [RFC3550]
deals with individual SSRCs. These SSRCs transmit their set of RTCP
packets at each scheduled interval. Thus, to maintain this per-SSRC
property of the scheduling, the avg_rtcp_size needs to be updated
with per-SSRC average RTCP compound packet sizes. The avg_rtcp_size
value SHALL be updated for each received or sent RTCP compound packet
with the total size (including packet overhead such as IP/UDP)
divided by the number of reporting SSRCs. The number of reporting
SSRCs SHALL be determined by counting the number of different SSRCs
that are the source of Sender Report (SR) or Receiver Report (RR)
RTCP packets within the compound. A non-compound RTCP packet, i.e.
it contains no SR or RR RTCP packets at all -- as can happen with
Reduced-Size RTCP packets [RFC5506] -- the SSRC count SHALL be
considered to be 1.
Note: The above makes it possible to amortize the packet overhead
between the number of SSRCs sharing a RTCP compound packet.
For an RTCP end-point that doesn't follow the above rule, and instead
uses the full RTCP compound packet size as input, the average RTCP
reporting interval will be scaled up (i.e. become longer) with a
factor that is proportional to the number of SSRCs sourcing RTCP
packets in an RTCP compound packet as well as the set of SSRCs being
aggregated in proportion to the total number of participants. This
factor can quite easily become larger than 5, e.g. with an 1500 byte
MTU and an average per-SSRC sum of RTCP packets of 240 bytes, the MTU
will fit 6 packets. If the receiver end-point has a single SSRC and
all other endpoints fill their MTU fully, the factor will be close to
6. If the RTCP configuration is such that the transmission interval
is bandwidth limited, rather than any type of minimal interval
limitation (Tmin or T_RR_INT), then the other end-points will likely
time out this SSRC due to it using an regular RTCP interval is more
than 5 times the rest of the endpoints.
5.3.2. Scheduling RTCP with Multiple Reporting SSRCs
When implementing RTCP packet scheduling for cases where multiple
reporting SSRCs are aggregating their RTCP packets in the same
compound packet there are a number of challenges. First of all, we
have the goal of not changing the general properties of the RTCP
packet transmissions, which include the general inter-packet
distribution, and the behavior for dealing with flash joins as well
as other dynamic events.
The below specified mechanism deals with:
o That one can't have a-priori knowledge about which RTCP packets
are to be sent, or their size, prior to generating the packets.
In which case, the time from generation to transmission ought to
be as short as possible to minimize the information that becomes
stale.
o That one has an MTU limit, that one ought to avoid exceeding, as
that requires lower-layer fragmentation (e.g., IP fragmentation)
which impacts the packets' probability of reaching the
receiver(s).
Schedule all the endpoint's local SSRCs individually for transmission
using the regular calculation of Tn for the profile being used. Each
time a SSRC's Tn timer expires, do the regular reconsideration. If
the reconsideration indictes that an RTCP packet is to be sent:
1. Consider if an additional SSRC can be added. That consideration
is done by picking the SSRC which has the Tn value closest in
time to now (Tc).
2. Calculate how much space for RTCP packets would be needed to add
that SSRC.
3. If the considered SSRC's RTCP Packets fit within the lower layer
datagram's Maximum Transmission Unit, taking the necessary
protocol headers into account and the consumed space by prior
SSRCs, then add that SSRC's RTCP packets to the compound packet
and go again to Step 1.
4. If the considered SSRC's RTCP Packets will not fit within the
compound packet, then transmit the generated compound packet.
5. Update the RTCP Parameters for each SSRC that has been included
in the sent RTCP packet. The Tp value for each SSRC MUST be
updated as follows:
For the first SSRC: As this SSRC was the one that was
reconsidered the tp value is set to the tc as defined in RTP
[RFC3550].
For any additional SSRC: The tp value SHALL be set to the
transmission time this SSRC would have had it not been
aggregated and given the current existing session context.
This value is derived by taking this SSRC's Tn value and
performing reconisderation and updating tn until tp + T <= tn.
Then set tp to this tn value.
6. For the sent SSRCs calculate new tn values based on the updated
parameters and reschedule the timers.
Reverse reconsideration needs to be performed as specified in RTP
[RFC3550]. It is important to note that under the above algorithm
when performing reconsideration, the value of tp can actually be
larger than tc. However, that still has the desired effect of
proportionally pulling the tp value towards tc (as well as tn) as the
group size shrinks in direct proportion the reduced group size.
The above algorithm has been shown in simulations to maintain the
inter-RTCP-packet transmission distribution for the SSRCs and consume
the same amount of bandwidth as non-aggregated packets in RTP
sessions with static sets of participants. With this algorithm the
actual transmission interval for any SSRC triggering an RTCP compound
packet transmission is following the regular transmission rules. It
also handles the cases where the number of SSRCs that can be included
in an aggregated packet varies. An SSRC that previously was
aggregated and fails to fit in a packet still has its own
transmission scheduled according to normal rules. Thus, it will
trigger a transmission in due time, or the SSRC will be included in
another aggregate.
The algorithm's behavior under SSRC group size changes is under
investigation. However, it is expected to be well behaved based on
the following analyses.
RTP sessions where the number of SSRC are growing: When the group
size is growing, the Td values grow in proportion to the number of
new SSRCs in the group. The reconsideration when the timer for
the tn expires, that SSRC will reconsider the transmission and
with a certain probability reschedule the tn timer. This part of
the reconsideration algorithm is only impacted by the above
algorithm by having tp values that are in the future instead of
set to the time of the actual last transmission at the time of
updating tp. Thus the scheduling causes in worst case a plateau
effect for that SSRC. That effect depends on how far into the
future tp can advance.
RTP sessions where the number of SSRC are shrinking: When the group
shrinks, reverse reconsideration moves the tp and tn values
towards tc proportionally to the number of SSRCs that leave the
session compared to the total number of participants when they
left. Thus the also group size reductions need to be handled.
In general the potential issue that might exist depends on how far
into the future the tp value can drift compared to the actual packet
transmissions that occur. That drift can only occur for an SSRC that
never is the trigger for RTCP packet transmission and always gets
aggregated and where the calculcated packet transmission interval
randomly occurs so that tn - tp for this SSRC is on average larger
than the ones that gets transmitted.
5.4. RTP/AVPF Feedback Packets
This section discusses the transmission of RTP/AVPF feedback packets
when the transmitting endpoint has multiple SSRCs.
5.4.1. The SSRC Used
When an RTP endpoint has multiple SSRCs, it can make certain choices
on which SSRC to use as the source of an RTCP Feedback Packet. This
sub-section discusses some considerations of this.
o The media type of the media the SSRC transmits is actually not a
relevant factor when considering if an SSRC can transmit a
particular Feedback message.
o Feedback messages which are Notification or Indications regarding
the endpoint's own RTP packet stream need to be sent using the
SSRC transmitting the media it relates to. This also includes
notifications that are related to a received request or command.
o The SSRC used to send feedback messages has a role as either a
media sender or a receiver. The bandwidth pools can be different
for SSRCs that are senders and receivers. Thus feedback messages
that expect to be more frequent can be sent from an SSRC that has
the better possibility of sending frequent RTCP compound packets
or reduced size packets. This also affects the consideration if
the SSRC can be used in immediate mode or not.
o Some Feedback Types requires consistency in the sender. For
example TMMBR, if one sets a limitation, the same SSRC needs to be
the one that increases it. Others can simply benefit from having
this property.
Note that the source of the feedback RTCP packet does not need to be
any of the sources (SSRC) including SR/RR packets in a compound
packet. For Reduced-Size RTCP [RFC5506] the aggregation of feedback
messages from multiple sources are not limited, beyond the
consideration in Section 4.2.2 of [RFC5506].
5.4.2. Scheduling a Feedback Packet
When an SSRC has a need to transmit a feedback packet in early mode
it follows the scheduling rules defined in Section 3.5 in RTP/AVPF
[RFC4585]. When following these rules the following clarifications
need to be taken into account:
o That a session is considered to be point-to-point or multiparty
not based on the number of SSRCs, but the number of endpoints
directly seen in the RTP session by the endpoint. tbd: Clarify
what is considered to "see" an endpoint?
o Note that when checking if there is already a scheduled compound
RTCP packet containing feedback messages (Step 2 in
Section 3.5.2), that check is done considering all local SSRCs.
TBD: The above does not allow an SSRC that is unable to send either
an early or regular RTCP packet with the feedback message within the
T_max_fb_delay to trigger another SSRC to send an early packet to
which it could piggyback. Nor does it allow feedback to piggyback on
even regular RTCP packet transmissions that occur within
T_max_fb_delay. A question is if either of these behaviours ought to
be allowed.
The latter appears simple and straight forward. Instead of
discarding a FB message in step 4a: alternative 2, one could place
such messages in a cache with a discard time equal to T_max_fb_delay,
and in case any of the SSRCs schedule an RTCP packet for transmission
within that time, it includes this message.
The former case can have more widespread impact on the application,
and possibly also on the RTCP bandwidth consumption as it allows for
more massive bursts of RTCP packets. Still, on a time scale of a
regular reporting interval, it ough to have no effect on the RTCP
bandwidth as the extra feedback messages increase the avg_rtcp_size.
6. RTCP Considerations for Streams with Disparate Rates 6. RTCP Considerations for Streams with Disparate Rates
It is possible for a single RTP session to carry streams of greatly It is possible for a single RTP session to carry streams of greatly
differing bandwidth. There are two scenarios where this can occur. differing bandwidth. There are two scenarios where this can occur.
The first is when a single RTP session carries multiple flows of the 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 same media type, but with very different quality; for example a video
switching multi-point conference unit might send a full rate high- switching multi-point conference unit might send a full rate high-
definition video stream of the active speaker but only thumbnails for definition video stream of the active speaker but only thumbnails for
the other participants, all sent in a single RTP session. The second the other participants, all sent in a single RTP session. The second
skipping to change at page 8, line 19 skipping to change at page 13, line 45
All SSRCs used in an RTP session MUST use the same timeout behaviour All SSRCs used in an RTP session MUST use the same timeout behaviour
to avoid premature timeouts. This will depend on the RTP profile and to avoid premature timeouts. This will depend on the RTP profile and
its configuration. The RTP specification provides several options its configuration. The RTP specification provides several options
that can influence the values used when calculating the time that can influence the values used when calculating the time
interval. To avoid interoperability issues when using this interval. To avoid interoperability issues when using this
specification, this document makes several clarifications to the specification, this document makes several clarifications to the
calculations. calculations.
For RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF with T_rr_interval = For RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF with T_rr_interval =
0, the timeout interval SHALL be calculated using a multiplier of 5, 0, the timeout interval SHALL be calculated using a multiplier of 5,
i.e. the timeout interval becomes 5*Td. The Td calculation SHALL be i.e. the timeout interval becomes 5*Td. The Td calculation SHALL be
done using a Tmin value of 5 seconds, not the reduced minimal done using a Tmin value of 5 seconds, not the reduced minimal
interval even if used to calculate RTCP packet transmission interval even if used to calculate RTCP packet transmission
intervals. If using either the RTP/AVPF or RTP/SAVPF profiles with 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 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 of RFC 4585 SHALL be used with a multiplier of 5, i.e. Tmin in the Td
Td calculation is the T_rr_interval. calculation is the T_rr_interval.
Note: If endpoints implementing the RTP/AVP and RTP/AVPF profiles (or If endpoints implementing the RTP/AVP and RTP/AVPF profiles (or their
their secure variants) are combined in a single RTP session, and the secure variants) are combined in a single RTP session, and the RTP/
RTP/AVPF endpoints use a non-zero T_rr_interval that is significantly AVPF endpoints use a non-zero T_rr_interval that is significantly
lower than 5 seconds, then there is a risk that the RTP/AVP endpoints lower than 5 seconds, then there is a risk that the RTP/AVPF
will prematurely timeout the RTP/AVPF endpoints due to their endpoints will prematurely timeout the RTP/AVP SSRCs due to their
different RTCP timeout intervals. Since an RTP session can only use different RTCP timeout intervals. Conversely, if the RTP/AVPF
a single RTP profile, this issue ought never occur. If such mixed endpoints use a T_rr_interval that is significant larger than 5
RTP profiles are used, however, the RTP/AVPF session MUST NOT use a seconds, there is a risk that the RTP/AVP endpoints will timeout the
non-zero T_rr_interval that is smaller than 5 seconds. RTP/AVPF SSRCs. If such mixed RTP profiles are used, (though this is
NOT RECOMMENDED), the RTP/AVPF session SHOULD use a non-zero
T_rr_interval that is 4 seconds.
(tbd: it has been suggested that a minimum non-zero T_rr_interval of Note: It might appear strange to use a T_rr_interval of 4 seconds.
4 seconds is more appropriate, due to the nature of the timing It might be intuitive that this value ought to be 5 seconds, as
rules). then both the RTP/AVP and RTP/AVPF would use the same timeout
period. However, considering regular RTCP transmission and their
packet intervals for RTP/AVPF its mean value will (with non-zero
T_rr_interval) be larger than T_rr_interval due to the scheduling
algorithm. Thus, to enable an equal amount of regular RTCP
transmissions in each directions between RTP/AVP and RTP/AVPF
endpoints, taking the altered timeout intervals into account, the
optimal value is around four (4), where almost four transmissions
will on average occur in each direction between the different
profile types given an otherwise good configuration of parameters
in regards to T_rr_interval. If the RTCP bandwidth paramters are
selected so that Td based on bandwidth is close to 4, i.e. close
to T_rr_interval the risk increases that RTP/AVPF SSRCs will be
timed out by RTP/AVP endpoints, as the RTP/AVPF SSRC might only
manage two transmissions in the timeout period.
6.2. Tuning RTCP transmissions 6.2. Tuning RTCP transmissions
This sub-section discusses what tuning can be done to reduce the This sub-section discusses what tuning can be done to reduce the
downsides of the shared RTCP packet intervals. downsides of the shared RTCP packet intervals. First, it is
considered what possibilites exist for the RTP/AVP [RFC3551] profile,
then what additional tools are provided by RTP/AVPF [RFC4585].
6.2.1. RTP/AVP and RTP/SAVP
When using the RTP/AVP or RTP/SAVP profiles the tuning one can do is 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 very limited. The controls one has are limited to the RTCP bandwidth
values and whether the minimum RTCP interval is scaled according to values and whether the minimum RTCP interval is scaled according to
the bandwidth. As the scheduling algorithm includes both random the bandwidth. As the scheduling algorithm includes both random
factors and reconsideration, one can't simply calculate the expected factors and reconsideration, one can't simply calculate the expected
average transmission interval using the formula for Td. But it does average transmission interval using the formula for Td. But it does
indicate the important factors affecting the transmission interval, indicate the important factors affecting the transmission interval,
namely the RTCP bandwidth available for the role (Active Sender or namely the RTCP bandwidth available for the role (Active Sender or
Participant), the average RTCP packet size, and the number of SSRCs Participant), the average RTCP packet size, and the number of SSRCs
skipping to change at page 10, line 11 skipping to change at page 16, line 7
randomly picking a number that is <=X within the interval with an randomly picking a number that is <=X within the interval with an
uniform probability distribution. This results in that the uniform probability distribution. This results in that the
majority of the probability mass is above the Td value. majority of the probability mass is above the Td value.
To conclude, with RTP/AVP and RTP/SAVP the key limitation for small To conclude, with RTP/AVP and RTP/SAVP the key limitation for small
unicast sessions is going to be the Tmin value. Thus the RTP session unicast sessions is going to be the Tmin value. Thus the RTP session
bandwidth configured in RTCP has to be sufficiently high to reach the bandwidth configured in RTCP has to be sufficiently high to reach the
reporting goals the application has following the rules for the reporting goals the application has following the rules for the
scaled minimal RTCP interval. scaled minimal RTCP interval.
6.2.2. RT/AVPF and RTP/SAVPF
When using RTP/AVPF or RTP/SAVPF we get a quite powerful additional When using RTP/AVPF or RTP/SAVPF we get a quite powerful additional
tool, the setting of the T_rr_interval which has several effects on tool, the setting of the T_rr_interval which has several effects on
the RTCP reporting. First of all as Tmin is set to 0 after the the RTCP reporting. First of all as Tmin is set to 0 after the
initial transmission, the regular reporting interval is instead initial transmission, the regular reporting interval is instead
determined by the regular bandwidth based calculation and the determined by the regular bandwidth based calculation and the
T_rr_interval. This has the effect that we are no longer restricted T_rr_interval. This has the effect that we are no longer restricted
by the minimal interval or even the scaling rule for the minimal by the minimal interval or even the scaling rule for the minimal
rule. Instead the RTCP bandwidth and the T_rr_interval are the rule. Instead the RTCP bandwidth and the T_rr_interval are the
governing factors. Now it also becomes important to separate between governing factors.
the application's need for regular reports and RTCP feedback packet
types. In both regular RTCP mode, as in Early RTCP Mode, the usage Now it also becomes important to separate between the application's
of the T_rr_interval prevents regular RTCP packets, i.e. packets need for regular reports and RTCP feedback packet types. In both
without any Feedback packets, to be sent more often than regular RTCP mode, as in Early RTCP Mode, the usage of the
T_rr_interval. This value is a hard as no regular RTCP packet can be T_rr_interval prevents regular RTCP packets, i.e. packets without any
sent less than T_rr_interval after the previous regular packet Feedback packets, to be sent more often than T_rr_interval. This
packet. value is applied to prevent any regular RTCP packet to be sent less
than T_rr_interval times a uniformly distributed random value from
the interval [0.5,1.5] after the previous regular packet packet. The
random value recalculated after each regular RTCP packet
transmission.
So applications that have a use for feedback packets for some media So applications that have a use for feedback packets for some media
streams, for example video streams, but don't want frequent regular streams, for example video streams, but don't want frequent regular
reporting for audio, could configure the T_rr_interval to a value so 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 the regular reporting for both audio and video is at a level
that is considered acceptable for the audio. They could then use that is considered acceptable for the audio. They could then use
feedback packets, which will include RTCP SR/RR packets, unless feedback packets, which will include RTCP SR/RR packets, unless
reduced-size RTCP feedback packets [RFC5506] are used, and can reduced-size RTCP feedback packets [RFC5506] are used, and can
include other report information in addition to the feedback packet include other report information in addition to the feedback packet
that needs to be sent. That way the available RTCP bandwidth can be that needs to be sent. That way the available RTCP bandwidth can be
focused for the use which provides the most utility for the focused for the use which provides the most utility for the
application. application.
Using T_rr_interval still requires one to determine suitable values Using T_rr_interval still requires one to determine suitable values
for the RTCP bandwidth value, in fact it might make it even more for the RTCP bandwidth value, in fact it might make it even more
important, as this is more likely to affect the RTCP behaviour and important, as this is more likely to affect the RTCP behaviour and
performance than when using RTP/AVP, as there are fewer limitations performance than when using RTP/AVP, as there are fewer limitations
affecting the RTCP transmission. affecting the RTCP transmission.
When using T_rr_interval, i.e. having it be non zero, there are When using T_rr_interval, i.e. having it be non zero, there are
configurations that have to be avoided. If the resulting Td value is configurations that have to be avoided. If the resulting Td value is
smaller but close to T_rr_interval then the interval in which the smaller but close to T_rr_interval then the interval in which the
actual regular RTCP packet transmission falls into becomes very actual regular RTCP packet transmission falls into becomes very
large, from 0.5 times T_rr_interval up to 2.73 times the 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 T_rr_interval. Therefore for configuration where one intends to have
Td smaller than T_rr_interval, then Td is RECOMMENDED to be targeted 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 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]. range becomes [0.5*T_rr_interval, 1.81*T_rr_interval].
With RTP/AVPF, using a T_rr_interval of 0 or with another low value With RTP/AVPF, using a T_rr_interval of 0 or with another low value
skipping to change at page 11, line 26 skipping to change at page 17, line 29
There exists no method within the specification for using different There exists no method within the specification for using different
regular RTCP reporting intervals depending on the media type or regular RTCP reporting intervals depending on the media type or
individual media stream. individual media stream.
7. Security Considerations 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 allow use of per-
keying. SSRC 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.
8. 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. Do we need to provide a recommendation for unicast session
sending multiple sources in one compound packet.
2. How is the RTCP avg_rtcp_size be calculated when RTCP packets are
routinely multiplexed among multiple RTCP senders?
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?
2. RTCP parameters for common scenarios in Section 6.2?
3. Is scheduling algorithm working well with dynamic changes?
4. Are the scheduling algorithm changes impacting previous
implementations in such a way that the report aggregation has to
be agreed on, and thus needs to be considered as an optimization?
5. An open question is if any improvements or clarifications ought
to be allowed regarding FB message scheduling in multi-SSRC
endpoints.
9. IANA Considerations 9. IANA Considerations
No IANA actions needed. No IANA actions needed.
10. References 10. References
10.1. Normative References 10.1. 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-06
(work in progress), July 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)",
RFC 3711, March 2004. RFC 3711, March 2004.
skipping to change at page 12, line 44 skipping to change at page 18, line 49
[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.
10.2. Informative References 10.2. 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, "Sending Westerlund, M., Perkins, C., and J. Lennox, "Sending
Multiple Types of Media in a Single RTP Session", draft- Multiple Types of Media in a Single RTP Session", draft-
ietf-avtcore-multi-media-rtp-session-03 (work in ietf-avtcore-multi-media-rtp-session-04 (work in
progress), July 2013. progress), January 2014.
[I-D.ietf-avtcore-rtp-multi-stream-optimisation] [I-D.ietf-avtcore-rtp-multi-stream-optimisation]
Lennox, J., Westerlund, M., Wu, Q., and C. Perkins, Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session: "Sending Multiple Media Streams in a Single RTP Session:
Grouping RTCP Reception Statistics and Other Feedback ", Grouping RTCP Reception Statistics and Other Feedback",
draft-ietf-avtcore-rtp-multi-stream-optimisation-00 (work draft-ietf-avtcore-rtp-multi-stream-optimisation-01 (work
in progress), July 2013. in progress), January 2014.
[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-01 (work in progress), ietf-avtcore-rtp-topologies-update-01 (work in progress),
October 2013. October 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-12 (work in progress), October 2013. framework-14 (work in progress), February 2014.
[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-05 (work in progress), October 2013. bundle-negotiation-05 (work in progress), October 2013.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[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.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006.
[RFC6190] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
"RTP Payload Format for Scalable Video Coding", RFC 6190,
May 2011.
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 WG Draft -00 A.1. Changes From WG Draft -02
o Changed usage of Media Stream
o Added Updates RFC 4585
o Added rules for how to deal with RTCP when aggregating multiple
SSRCs report in same compound packet:
* avg_rtcp_size calcualtion
* Scheduling rules to maintain timing
o Started a section clarifying and discsussing RTP/AVPF Feedback
Packets and their scheduling.
A.2. Changes From WG Draft -01
o None, a keep-alive version
A.3. Changes From WG Draft -00
o Split the Reporting Group Extension from this draft into draft- o Split the Reporting Group Extension from this draft into draft-
ietf-avtcore-rtp-multi-stream-optimization-00. ietf-avtcore-rtp-multi-stream-optimization-00.
o Added RTCP tuning considerations from draft-ietf-avtcore-multi- o Added RTCP tuning considerations from draft-ietf-avtcore-multi-
media-rtp-session-02. media-rtp-session-02.
A.2. Changes From Individual Draft -02 A.4. Changes From Individual Draft -02
o Resubmitted as working group draft. o Resubmitted as working group draft.
o Updated references. o Updated references.
A.3. Changes From Individual Draft -01 A.5. 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.4. Changes From Individual Draft -00 A.6. 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. 55 change blocks. 
168 lines changed or deleted 480 lines changed or added

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