draft-ietf-avtcore-rtp-multi-stream-04.txt   draft-ietf-avtcore-rtp-multi-stream-05.txt 
AVTCORE J. Lennox AVTCORE J. Lennox
Internet-Draft Vidyo Internet-Draft Vidyo
Updates: 3550, 4585 (if approved) M. Westerlund Updates: 3550, 4585 (if approved) M. Westerlund
Intended status: Standards Track Ericsson Intended status: Standards Track Ericsson
Expires: November 29, 2014 Q. Wu Expires: January 04, 2015 Q. Wu
Huawei Huawei
C. Perkins C. Perkins
University of Glasgow University of Glasgow
May 28, 2014 July 03, 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-04 draft-ietf-avtcore-rtp-multi-stream-05
Abstract Abstract
This document expands and clarifies the behavior of the Real-Time This memo expands and clarifies the behaviour of Real-time Transport
Transport Protocol (RTP) endpoints when they are using multiple Protocol (RTP) endpoints that use multiple synchronization sources
synchronization sources (SSRCs), e.g. for sending multiple media (SSRCs). This occurs, for example, when an endpoint sends multiple
streams, in a single RTP session. In particular, issues involving media streams in a single RTP session. This memo updates RFC 3550
RTCP Control Protocol (RTCP) messages are described. with regards to handling multiple SSRCs per endpoint in RTP sessions,
with a particular focus on RTCP behaviour. It also updates RFC 4585
This document updates RFC 3550 in regards to handling of multiple to update and clarify the calculation of the timeout of SSRCs and the
SSRCs per endpoint in RTP sessions. It also updates RFC 4585 to inclusion of feedback messages.
update and clarify the calculation of the timeout of SSRCs and the
inclusion of feeback messages.
Status of This Memo Status of This Memo
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provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on November 29, 2014. This Internet-Draft will expire on January 04, 2015.
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 . . . . . . . . . . . . 4 3. Use Cases For Multi-Stream Endpoints . . . . . . . . . . . . 3
3.1. Multiple-Capturer Endpoints . . . . . . . . . . . . . . . 4 3.1. Endpoints with Multiple Capture Devices . . . . . . . . . 3
3.2. Multi-Media Sessions . . . . . . . . . . . . . . . . . . 4 3.2. Multiple Media Types in a Single RTP Session . . . . . . 3
3.3. Multi-Stream Mixers . . . . . . . . . . . . . . . . . . . 4 3.3. Multiple Stream Mixers . . . . . . . . . . . . . . . . . 4
3.4. Multiple SSRCs for a Single Media Source . . . . . . . . 5 3.4. Multiple SSRCs for a Single Media Source . . . . . . . . 4
4. Multi-Stream Endpoint RTP Media Recommendations . . . . . . . 5 4. Use of RTP by endpoints that send multiple media streams . . 5
5. Multi-Stream Endpoint RTCP Recommendations . . . . . . . . . 5 5. Use of RTCP by Endpoints that send multiple media streams . . 5
5.1. RTCP Reporting Requirement . . . . . . . . . . . . . . . 5 5.1. RTCP Reporting Requirement . . . . . . . . . . . . . . . 5
5.2. Initial Reporting Interval . . . . . . . . . . . . . . . 6 5.2. Initial Reporting Interval . . . . . . . . . . . . . . . 5
5.3. Compound RTCP Packets . . . . . . . . . . . . . . . . . . 6 5.3. Aggregation of Reports into Compound RTCP Packets . . . . 6
5.3.1. Maintaining AVG_RTCP_SIZE . . . . . . . . . . . . . . 7 5.3.1. Maintaining AVG_RTCP_SIZE . . . . . . . . . . . . . . 7
5.3.2. Scheduling RTCP with Multiple Reporting SSRCs . . . . 8 5.3.2. Scheduling RTCP with Multiple Reporting SSRCs . . . . 8
5.4. RTP/AVPF Feedback Packets . . . . . . . . . . . . . . . . 10 5.4. Use of RTP/AVPF Feedback . . . . . . . . . . . . . . . . 10
5.4.1. The SSRC Used . . . . . . . . . . . . . . . . . . . . 10 5.4.1. Choice of SSRC for Feedback Packets . . . . . . . . . 10
5.4.2. Scheduling a Feedback Packet . . . . . . . . . . . . 11 5.4.2. Scheduling an RTCP Feedback Packet . . . . . . . . . 11
6. RTCP Considerations for Streams with Disparate Rates . . . . 12 6. RTCP Considerations for Streams with Disparate Rates . . . . 12
6.1. Timing out SSRCs . . . . . . . . . . . . . . . . . . . . 13 6.1. Timing out SSRCs . . . . . . . . . . . . . . . . . . . . 13
6.1.1. AVPF T_rr_interval Behavior . . . . . . . . . . . . . 13 6.1.1. AVPF T_rr_interval Behaviour . . . . . . . . . . . . 13
6.1.2. Avoiding Pre-mature Timeout . . . . . . . . . . . . . 14 6.1.2. Avoiding Premature Timeout . . . . . . . . . . . . . 14
6.1.3. AVP and AVPF Interoperability . . . . . . . . . . . . 15 6.1.3. RTP/AVP and RTP/AVPF Interoperability . . . . . . . . 15
6.1.4. Specified Behavior . . . . . . . . . . . . . . . . . 16 6.1.4. Specified Behaviour . . . . . . . . . . . . . . . . . 16
6.2. Tuning RTCP transmissions . . . . . . . . . . . . . . . . 17 6.2. Tuning RTCP transmissions . . . . . . . . . . . . . . . . 16
6.2.1. RTP/AVP and RTP/SAVP . . . . . . . . . . . . . . . . 17 6.2.1. RTP/AVP and RTP/SAVP . . . . . . . . . . . . . . . . 16
6.2.2. RT/AVPF and RTP/SAVPF . . . . . . . . . . . . . . . . 18 6.2.2. RTP/AVPF and RTP/SAVPF . . . . . . . . . . . . . . . 18
7. Security Considerations . . . . . . . . . . . . . . . . . . . 19 7. Security Considerations . . . . . . . . . . . . . . . . . . . 19
8. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 20 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 19
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 20 9. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 19
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 20 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 20
10.1. Normative References . . . . . . . . . . . . . . . . . . 20 10.1. Normative References . . . . . . . . . . . . . . . . . . 20
10.2. Informative References . . . . . . . . . . . . . . . . . 21 10.2. Informative References . . . . . . . . . . . . . . . . . 20
Appendix A. Changes From Earlier Versions . . . . . . . . . . . 22 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 22
A.1. Changes From WG Draft -02 . . . . . . . . . . . . . . . . 22
A.2. Changes From WG Draft -01 . . . . . . . . . . . . . . . . 22
A.3. Changes From WG Draft -00 . . . . . . . . . . . . . . . . 22
A.4. Changes From Individual Draft -02 . . . . . . . . . . . . 23
A.5. Changes From Individual Draft -01 . . . . . . . . . . . . 23
A.6. Changes From Individual Draft -00 . . . . . . . . . . . . 23
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 24
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 designed, 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, and thus sessions typically only transmitted a single media stream, and thus
used a single synchronization source (SSRC) per RTP session, where used a single synchronization source (SSRC) per RTP session, where
separate RTP sessions were typically used for each distinct media separate RTP sessions were typically used for each distinct media
type. type. Recently, however, a number of scenarios have emerged in which
endpoints wish to send multiple RTP media streams, distinguished by
Recently, however, a number of scenarios have emerged (discussed distinct RTP synchronization source (SSRC) identifiers, in a single
further in Section 3) in which endpoints wish to send multiple RTP RTP session. These are outlined in Section 3. Although the initial
media streams, distinguished by distinct RTP synchronization source design of RTP did consider such scenarios, the specification was not
(SSRC) identifiers, in a single RTP session. Although RTP's initial
design did consider such scenarios, the specification was not
consistently written with such use cases in mind. The specifications 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 This memo updates [RFC3550] to clarify behaviour in use cases where
language for these use cases. The authors believe this does not endpoints use multiple SSRCs. It also updates [RFC4585] in regards
result in any major normative changes to the RTP specification, to the timeout of inactive SSRCs to resolve problematic behaviour as
however this document defines how the RTP specification is to be
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 to resolve problematic behavior as
well as clarifying the inclusion of feedback messages. well as clarifying the inclusion of feedback messages.
The document starts with terminology and some use cases where
multiple sources will occur. This is followed by RTP and RTCP
recommendations to resolve issues. Next 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 sends RTP data using multiple SSRCs in development of endpoints that sends RTP data using multiple SSRCs in
a single RTP session. a single RTP session.
3.1. Multiple-Capturer Endpoints 3.1. Endpoints with Multiple Capture Devices
The most straightforward motivation for an endpoint to send multiple The most straightforward motivation for an endpoint to send multiple
RTP streams in a session is the scenario where an endpoint has simultaneous RTP streams in a session is the scenario where an
multiple capture devices, and thus media sources, of the same media endpoint has multiple capture devices, and thus media sources, of the
type and characteristics. For example, telepresence endpoints, of same media type and characteristics. For example, telepresence
the type described by the CLUE Telepresence Framework endpoints, of the type described by the CLUE Telepresence Framework
[I-D.ietf-clue-framework], often have multiple cameras or microphones [I-D.ietf-clue-framework], often have multiple cameras or microphones
covering various areas of a room. covering various areas of a room, and hence send several RTP streams.
3.2. Multi-Media Sessions 3.2. Multiple Media Types in a Single RTP Session
Recent work has been done in RTP Recent work has updated 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 remove the historical
assumption that media sources of different media types would always assumption in RTP that media sources of different media types would
be sent on different RTP sessions. In this work, a single endpoint's always be sent on different RTP sessions. In this work, a single
audio and video RTP media streams (for example) are instead sent in a endpoint's audio and video RTP media streams (for example) are
single RTP session. instead sent in a single RTP session to reduce the number of
transport layer flows used.
3.3. Multi-Stream Mixers 3.3. Multiple 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 RTP media streams in a session. that itself generates multiple RTP media streams in a session. An
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 Selective Forwarding Middlebox, see A more complex example is the selective forwarding middlebox,
Section 3.7 of [I-D.ietf-avtcore-rtp-topologies-update]. This is a described in Section 3.7 of [I-D.ietf-avtcore-rtp-topologies-update].
middlebox that receives media streams from several endpoints, and This is a middlebox that receives media streams from several
then selectively forwards modified versions of some of the streams endpoints, and then selectively forwards modified versions of some
toward the other endpoints it is connected to. Toward one RTP streams toward the other endpoints to which it is connected. For
destination, a separate media source appears in the session for every each connected endpoint, a separate media source appears in the
other source connected to the middlebox, "projected" from the session for every other source connected to the middlebox,
original streams, but at any given time many of them can appear to be "projected" from the original streams, but at any given time many of
inactive (and thus are receivers, not senders, in RTP). This sort of them can appear to be inactive (and thus are receivers, not senders,
device is closer to being an RTP mixer than an RTP translator, in in RTP). This sort of device is closer to being an RTP mixer than an
that it terminates RTCP reporting about the mixed streams, and it can RTP translator, in that it terminates RTCP reporting about the mixed
re-write SSRCs, timestamps, and sequence numbers, as well as the streams, and it can re-write SSRCs, timestamps, and sequence numbers,
contents of the RTP payloads, and can turn sources on and off at will as well as the contents of the RTP payloads, and can turn sources on
without appearing to be generating packet loss. Each projected and off at will without appearing to be generating packet loss. Each
stream will typically preserve its original RTCP source description projected stream will typically preserve its original RTCP source
(SDES) information. description (SDES) information.
3.4. Multiple SSRCs for a Single Media Source 3.4. Multiple SSRCs for a Single Media Source
There are also several cases where a single media source results in There are also several cases where a single media source results in
the usage of multiple SSRCs within the same RTP session. Transport the usage of multiple SSRCs within the same RTP session. Transport
robustification tools like RTP Retransmission [RFC4588] result in robustness tools like RTP Retransmission [RFC4588] result in multiple
multiple SSRCs, one with source data, and another with the repair SSRCs, one with source data, and another with the repair data.
data. Scalable encoders and their RTP payload foramts, like H.264's Scalable encoders and their RTP payload formats, like H.264's
extension for Scalable Video Coding(SVC) [RFC6190] can be transmitted extension for Scalable Video Coding(SVC) [RFC6190] can be transmitted
in a configuration where the scalable layers are distributed over in a configuration where the scalable layers are distributed over
multiple SSRCs within the same session, to enable RTP packet stream multiple SSRCs within the same session, to enable RTP packet stream
level (SSRC) selection and routing in conferencing middleboxes. level (SSRC) selection and routing in conferencing middleboxes.
4. Multi-Stream Endpoint RTP Media Recommendations 4. Use of RTP by endpoints that send multiple media streams
While an endpoint MUST (of course) stay within its share of the
available session bandwidth, as determined by signalling and
congestion control, this need not be applied independently or
uniformly to each media stream and its SSRCs. In particular, session
bandwidth MAY be reallocated among an endpoint's SSRCs, for example
by varying the bandwidth use of a variable-rate codec, or changing
the codec used by the media stream, up to the constraints of the
session's negotiated (or declared) codecs. This includes enabling or
disabling media streams and their redundancy streams as more or less
bandwidth becomes available.
5. Multi-Stream Endpoint RTCP Recommendations Every RTP endpoint will have an allocated share of the available
session bandwidth, as determined by signalling and congestion
control. The endpoint MUST keep its total media sending rate within
this share. However, endpoints that send multiple media streams do
not necessarily need to subdivide their share of the available
bandwidth independently or uniformly to each media stream and its
SSRCs. In particular, an endpoint can vary the allocation to
different streams depending on their needs, and can dynamically
change the bandwidth allocated to different SSRCs (for example, by
using a variable rate codec), provided the total sending rate does
not exceed its allocated share. This includes enabling or disabling
media streams and their redundancy streams as more or less bandwidth
becomes available.
This section contains a number of different RTCP clarifications or 5. Use of RTCP by Endpoints that send multiple media streams
recommendations that enables more efficient and simpler behavior
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 The description of the protocol is phrased in terms of the behaviour
cases, the specification's recommendations for "participants" are to of "participants" in an RTP session, under the assumption that each
be interpreted as applying to individual SSRCs, rather than to endpoint is a participant with a single SSRC. However, for correct
endpoints. This section describes several concrete cases where this operation in cases where endpoints can send multiple media streams,
applies. the specification needs to be interpreted with each SSRC counting as
a participant in the session, so that an endpoint that has multiple
SSRCs counts as multiple participants. The following describes
several concrete cases where this applies.
5.1. RTCP Reporting Requirement 5.1. RTCP Reporting Requirement
For each of an endpoint's SSRCs, whether or not they are currently An RTP endpoint that has multiple SSRCs MUST treat each SSRC as a
sending media, SR/RR and SDES packets MUST be sent at least once per separate participant in the RTP session, sending RTCP reports for
RTCP report interval. (For discussion of the content of SR or RR each of its SSRCs in every RTCP reporting interval. If the mechanism
packets' reception statistic reports, see in [I-D.ietf-avtcore-rtp-multi-stream-optimisation] is not used, then
[I-D.ietf-avtcore-rtp-multi-stream-optimisation].) each SSRC will send RTCP reports for all other SSRCs, including those
co-located at the same endpoint.
5.2. Initial Reporting Interval If the endpoint has some SSRCs that are sending data and some that
are only receivers, then they will receive different shares of the
RTCP bandwidth and calculate different base RTCP reporting intervals.
Otherwise, all SSRCs at an endpoint will calculate the same base RTCP
reporting interval. The actual reporting intervals for each SSRC are
randomised in the usual way, but reports can be aggregated as
described in Section 5.3.
When a new SSRC is added to a unicast session, the sentence in 5.2. Initial Reporting Interval
[RFC3550]'s Section 6.2 applies: "For unicast sessions ... the delay When a participant joins a unicast session, the following text from
Section 6.2 of [RFC3550] applies: "For unicast sessions... the delay
before sending the initial compound RTCP packet MAY be zero." This before sending the initial compound RTCP packet MAY be zero." This
applies to individual SSRCs as well. Thus, endpoints MAY send an also applies to the individual SSRCs of an endpoint that has multiple
initial RTCP packet for an SSRC immediately upon adding it to a SSRCs, and such endpoints MAY send an initial RTCP packet for each of
unicast session. their SSRCs immediately upon joining a unicast session.
This allowance also applies, as written, when initially joining a Caution has to be exercised, however, when an endpoint (or middlebox)
unicast session. However, in this case some caution needs to be with a large number of SSRCs joins a unicast session, since immediate
exercised if the end-point or mixer has a large number of sources transmission of many RTCP reports can create a significant burst of
(SSRCs) as this can create a significant burst. How big an issue traffic, leading to transient congestion and packet loss due to queue
this is depends on the number of sources for which the initial SR or overflows. Implementers are advised to consider sending immediate
RR packets and Session Description CNAME items are to be sent, in RTCP packets for only a small number of SSRCs (e.g., the one or two
relation to the RTCP bandwidth. SSRCs they consider most important), with the initial RTCP packets
for their other SSRCs being sent after the calculated initial RTCP
reporting interval, to avoid self congestion.
(tbd: Maybe some recommendation here? The aim in restricting this to (tbd: is this recommendation sufficiently strong?)
unicast sessions was to avoid this burst of traffic, which the usual
RTCP timing and reconsideration rules will prevent.)
5.3. Compound RTCP Packets 5.3. Aggregation of Reports into Compound RTCP Packets
Section 6.1 in [RFC3550] gives the following advice to RTP As outlined in Section 5.1, an endpoint with multiple SSRCs has to
translators and mixers: treat each SSRC as a separate participant when it comes to sending
RTCP reports. This will lead to each SSRC sending a compound RTCP
packet in each reporting interval. Since these packets are coming
from the same endpoint, it might reasonably be expected that they can
be aggregated to reduce overheads. Indeed, Section 6.1 of [RFC3550]
allows RTP translators and mixers to aggregate packets in similar
circumstances:
"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, The allows RTP translators and mixers to generate compound RTCP
such as a Sender Report (SR), Receiver Report (RR), Source packets that contain multiple SR or RR packets from different SSRCs,
Description (SDES), Goodbye (BYE), Application Defined (APP), as well as any of the other packet types. There are no restrictions
Feedback [RFC4585] or Extended Report (XR) [RFC3611] packet. A on the order in which the RTCP packets can occur within the compound
compound packet is the combination of two or more such RTCP packet, except the regular rule that the compound RTCP packet starts
packets where the first packet has to be an SR or an RR packet, with an SR or RR packet. Due to this rule, correctly implemented RTP
and which contains a SDES packet containing an CNAME item. endpoints will be able to handle compound RTCP packets that contain
RTCP packets relating to multiple SSRCs.
The above results in compound RTCP packets that contain multiple SR
or RR packets from different sources (SSRCs) as well as any of the
other packet types. There are no restrictions on the order in which
the packets can occur within the compound packet, except the regular
compound rule, i.e., starting with an SR or RR.
This advice applies to multi-media-stream endpoints as well, with the Accordingly, endpoints that use multiple SSRCs MAY aggregate the RTCP
same restrictions and considerations. (Note, however, that the last packets sent by their different SSRCs into compound RTCP packets,
sentence does not apply to AVPF [RFC4585] or SAVPF [RFC5124] feedback provided they maintain the average RTCP packet size as described in
packets if Reduced-Size RTCP [RFC5506] is in use.) Section 5.3.1, and schedule packet transmission and aggregation as
described in Section 5.3.2.
5.3.1. Maintaining AVG_RTCP_SIZE 5.3.1. Maintaining AVG_RTCP_SIZE
When multiple local SSRCs are sending their RTCP packets in the same The RTCP scheduling algorithm in [RFC3550] works on a per-SSRC basis.
compound packet, this obviously results in larger RTCP compound Each SSRC sends a single compound RTCP packet in each RTCP reporting
packets. This will have an affect on the value of the average RTCP interval. When an endpoint uses multiple SSRCs, it is desirable to
packet size metering (avg_rtcp_size) that is done for the purpose of aggregate the compound RTCP packets sent by its SSRCs, reducing the
RTCP transmission scheduling calculation. This section discusses the overhead by forming a larger compound RTCP packet. This aggregation
impact of this and provide recommendations with how to deal with it. can be done as described in Section 5.3.2, provided the average RTCP
packet size calculation is updated as follows.
This section will use the concept of an 'RTCP Compound Packet' to Participants in an RTP session update their estimate of the average
represent not just proper RTCP compound packets, i.e. ones that start RTCP packet size (avg_rtcp_size) each time they send or receive an
with an SR or RR RTCP packet and include at least one SDES CNAME RTCP packet (see Section 6.3.3 of [RFC3550]). When a compound RTCP
item. For the purpose of the below calculation, other valid lower packet that contains RTCP packets from several SSRCs is sent or
layer datagram units an RTCP implementation can send or receive, received, the avg_rtcp_size estimate for each SSRC that is reported
independently if they are an aggregate or not of RTCP packets are upon is updated using div_packet_size rather than the actual packet
also considered. This especially includes Reduced-Size RTCP packets size:
[RFC5506].
The RTCP packet scheduling algorithm that is defined in RTP [RFC3550] avg_rtcp_size = (1/16) * div_packet_size + (15/16) * avg_rtcp_size
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 where div_packet_size is packet_size divided by the number of SSRCs
between the number of SSRCs sharing a RTCP compound packet. reporting in that compound packet. The number of SSRCs reporting in
a compound packet is 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 RTCP packet. Non-compound RTCP
packets (i.e., RTCP packets that do not contain an SR or RR packet
[RFC5506]) are considered report on a single SSRC.
For an RTCP end-point that doesn't follow the above rule, and instead An SSRC doesn't follow the above rule, and instead uses the full RTCP
uses the full RTCP compound packet size as input, the average RTCP compound packet size to calculate avg_rtcp_size, will derive an RTCP
reporting interval will be scaled up (i.e. become longer) with a reporting interval that is overly large by a factor that is
factor that is proportional to the number of SSRCs sourcing RTCP proportional to the number of SSRCs aggregated into compound RTCP
packets in an RTCP compound packet as well as the set of SSRCs being packets and the size of set of SSRCs being aggregated relative to the
aggregated in proportion to the total number of participants. This total number of participants. This increased RTCP reporting interval
factor can quite easily become larger than 5, e.g. with an 1500 byte can cause premature timeouts if it is more than five times the
MTU and an average per-SSRC sum of RTCP packets of 240 bytes, the MTU interval chosen by the SSRCs that understand compound RTCP that
will fit 6 packets. If the receiver end-point has a single SSRC and aggregate reports from many SSRCs. A 1500 octet MTU can fit six
all other endpoints fill their MTU fully, the factor will be close to typical size reports into a compound RTCP packet, so this is a real
6. If the RTCP configuration is such that the transmission interval concern if endpoints aggregate RTCP reports from multiple SSRCs. If
is bandwidth limited, rather than any type of minimal interval compatibility with non-updated endpoints is a concern, the number of
limitation (Tmin or T_RR_INT), then the other end-points will likely reports from different SSRCs aggregated into a single compound RTCP
time out this SSRC due to it using an regular RTCP interval is more packet SHOULD be limited.
than 5 times the rest of the endpoints.
5.3.2. Scheduling RTCP with Multiple Reporting SSRCs 5.3.2. Scheduling RTCP with Multiple Reporting SSRCs
When implementing RTCP packet scheduling for cases where multiple When implementing RTCP packet scheduling for cases where multiple
reporting SSRCs are aggregating their RTCP packets in the same reporting SSRCs are aggregating their RTCP packets in the same
compound packet there are a number of challenges. First of all, we compound packet there are a number of challenges. First of all, we
have the goal of not changing the general properties of the RTCP have the goal of not changing the general properties of the RTCP
packet transmissions, which include the general inter-packet packet transmissions, which include the general inter-packet
distribution, and the behavior for dealing with flash joins as well distribution, and the behaviour for dealing with flash joins as well
as other dynamic events. as other dynamic events.
The below specified mechanism deals with: The below specified mechanism deals with:
o That one can't have a-priori knowledge about which RTCP packets 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. are to be sent, or their size, prior to generating the packets.
In which case, the time from generation to transmission ought to In which case, the time from generation to transmission ought to
be as short as possible to minimize the information that becomes be as short as possible to minimize the information that becomes
stale. stale.
o That one has an MTU limit, that one ought to avoid exceeding, as o That one has an MTU limit, that one ought to avoid exceeding, as
that requires lower-layer fragmentation (e.g., IP fragmentation) that requires lower-layer fragmentation (e.g., IP fragmentation)
which impacts the packets' probability of reaching the which impacts the packets' probability of reaching the
receiver(s). receiver(s).
Schedule all the endpoint's local SSRCs individually for transmission Schedule all the endpoint's local SSRCs individually for transmission
using the regular calculation of Tn for the profile being used. Each using the regular calculation of Tn for the profile being used. Each
time a SSRC's Tn timer expires, do the regular reconsideration. If time a SSRC's Tn timer expires, do the regular reconsideration. If
the reconsideration indictes that an RTCP packet is to be sent: the reconsideration indicates that an RTCP packet is to be sent:
1. Consider if an additional SSRC can be added. That consideration 1. Consider if an additional SSRC can be added. That consideration
is done by picking the SSRC which has the Tn value closest in is done by picking the SSRC which has the Tn value closest in
time to now (Tc). time to now (Tc).
2. Calculate how much space for RTCP packets would be needed to add 2. Calculate how much space for RTCP packets would be needed to add
that SSRC. that SSRC.
3. If the considered SSRC's RTCP Packets fit within the lower layer 3. If the considered SSRC's RTCP Packets fit within the lower layer
datagram's Maximum Transmission Unit, taking the necessary datagram's Maximum Transmission Unit, taking the necessary
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and go again to Step 1. and go again to Step 1.
4. If the considered SSRC's RTCP Packets will not fit within the 4. If the considered SSRC's RTCP Packets will not fit within the
compound packet, then transmit the generated compound packet. compound packet, then transmit the generated compound packet.
5. Update the RTCP Parameters for each SSRC that has been included 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 in the sent RTCP packet. The Tp value for each SSRC MUST be
updated as follows: updated as follows:
For the first SSRC: As this SSRC was the one that was 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 reconsidered the tp value is set to the tc as defined in
[RFC3550]. RTP [RFC3550].
For any additional SSRC: The tp value SHALL be set to the For any additional SSRC: The tp value SHALL be set to the
transmission time this SSRC would have had it not been transmission time this SSRC would have had it not been
aggregated and given the current existing session context. aggregated and given the current existing session context.
This value is derived by taking this SSRC's Tn value and This value is derived by taking this SSRC's Tn value and
performing reconisderation and updating tn until tp + T <= tn. performing reconsideration and updating tn until tp + T <=
Then set tp to this tn value. tn. Then set tp to this tn value.
6. For the sent SSRCs calculate new tn values based on the updated 6. For the sent SSRCs calculate new tn values based on the updated
parameters and reschedule the timers. parameters and reschedule the timers.
Reverse reconsideration needs to be performed as specified in RTP Reverse reconsideration needs to be performed as specified in RTP
[RFC3550]. It is important to note that under the above algorithm [RFC3550]. It is important to note that under the above algorithm
when performing reconsideration, the value of tp can actually be when performing reconsideration, the value of tp can actually be
larger than tc. However, that still has the desired effect of larger than tc. However, that still has the desired effect of
proportionally pulling the tp value towards tc (as well as tn) as the proportionally pulling the tp value towards tc (as well as tn) as the
group size shrinks in direct proportion the reduced group size. group size shrinks in direct proportion the reduced group size.
skipping to change at page 10, line 10 skipping to change at page 9, line 46
sessions with static sets of participants. With this algorithm the sessions with static sets of participants. With this algorithm the
actual transmission interval for any SSRC triggering an RTCP compound actual transmission interval for any SSRC triggering an RTCP compound
packet transmission is following the regular transmission rules. It packet transmission is following the regular transmission rules. It
also handles the cases where the number of SSRCs that can be included also handles the cases where the number of SSRCs that can be included
in an aggregated packet varies. An SSRC that previously was in an aggregated packet varies. An SSRC that previously was
aggregated and fails to fit in a packet still has its own aggregated and fails to fit in a packet still has its own
transmission scheduled according to normal rules. Thus, it will transmission scheduled according to normal rules. Thus, it will
trigger a transmission in due time, or the SSRC will be included in trigger a transmission in due time, or the SSRC will be included in
another aggregate. another aggregate.
The algorithm's behavior under SSRC group size changes is under The algorithm's behaviour under SSRC group size changes is under
investigation. However, it is expected to be well behaved based on investigation. However, it is expected to be well behaved based on
the following analyses. the following analyses.
RTP sessions where the number of SSRC are growing: When the group 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 size is growing, the Td values grow in proportion to the number of
new SSRCs in the group. The reconsideration when the timer for new SSRCs in the group. The reconsideration when the timer for
the tn expires, that SSRC will reconsider the transmission and the tn expires, that SSRC will reconsider the transmission and
with a certain probability reschedule the tn timer. This part of with a certain probability reschedule the tn timer. This part of
the reconsideration algorithm is only impacted by the above the reconsideration algorithm is only impacted by the above
algorithm by having tp values that are in the future instead of algorithm by having tp values that are in the future instead of
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RTP sessions where the number of SSRC are shrinking: When the group RTP sessions where the number of SSRC are shrinking: When the group
shrinks, reverse reconsideration moves the tp and tn values shrinks, reverse reconsideration moves the tp and tn values
towards tc proportionally to the number of SSRCs that leave the towards tc proportionally to the number of SSRCs that leave the
session compared to the total number of participants when they session compared to the total number of participants when they
left. Thus the also group size reductions need to be handled. left. Thus the also group size reductions need to be handled.
In general the potential issue that might exist depends on how far 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 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 transmissions that occur. That drift can only occur for an SSRC that
never is the trigger for RTCP packet transmission and always gets never is the trigger for RTCP packet transmission and always gets
aggregated and where the calculcated packet transmission interval aggregated and where the calculated packet transmission interval
randomly occurs so that tn - tp for this SSRC is on average larger randomly occurs so that tn - tp for this SSRC is on average larger
than the ones that gets transmitted. than the ones that gets transmitted.
5.4. RTP/AVPF Feedback Packets 5.4. Use of RTP/AVPF Feedback
This section discusses the transmission of RTP/AVPF feedback packets This section discusses the transmission of RTP/AVPF feedback packets
when the transmitting endpoint has multiple SSRCs. when the transmitting endpoint has multiple SSRCs.
5.4.1. The SSRC Used 5.4.1. Choice of SSRC for Feedback Packets
When an RTP endpoint has multiple SSRCs, it can make certain choices When an RTP/AVPF endpoint has multiple SSRCs, it can choose what SSRC
on which SSRC to use as the source of an RTCP Feedback Packet. This to use as the source for the RTCP feedback packets it sends. Several
sub-section discusses some considerations of this. factors can affect that choice:
o The media type of the media the SSRC transmits is actually not a o RTCP feedback packets relating to a particular media type SHOULD
relevant factor when considering if an SSRC can transmit a be sent by an SSRC that receives that media type. For example,
particular Feedback message. when audio and video are multiplexed onto a single RTP session,
endpoints will use their audio SSRC to send feedback on the audio
received from other participants.
o Feedback messages which are Notification or Indications regarding o RTCP feedback packets and RTCP codec control messages that are
the endpoint's own RTP packet stream need to be sent using the notifications or indications regarding RTP data processed by an
SSRC transmitting the media it relates to. This also includes endpoint MUST be sent from the SSRC used by that RTP data. This
notifications that are related to a received request or command. includes notifications that relate to a previously received
request or command.
o The SSRC used to send feedback messages has a role as either a o If separate SSRCs are used to send and receive media, then the
media sender or a receiver. The bandwidth pools can be different corresponding SSRC SHOULD be used for feedback, since they have
for SSRCs that are senders and receivers. Thus feedback messages differing RTCP bandwidth fractions. This can also effect the
that expect to be more frequent can be sent from an SSRC that has consideration if the SSRC can be used in immediate mode or not.
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 o Some RTCP feedback packet types requires consistency in the SSRC
example TMMBR, if one sets a limitation, the same SSRC needs to be used. For example, if one sets a TMMBR limitation, the same SSRC
the one that increases it. Others can simply benefit from having needs to be used to remove the limitation.
this property.
Note that the source of the feedback RTCP packet does not need to be When an RTCP feedback packet is sent as part of a compound RTCP
any of the sources (SSRC) including SR/RR packets in a compound packet that aggregates reports from multiple SSRCs, there is no
packet. For Reduced-Size RTCP [RFC5506] the aggregation of feedback requirement that the compound packet contains an SR or RR packet
messages from multiple sources are not limited, beyond the generated by the sender of the RTCP feedback packet. For reduced-
consideration in Section 4.2.2 of [RFC5506]. size RTCP packets, aggregation of RTCP feedback packets from multiple
sources is not limited further than Section 4.2.2 of [RFC5506].
5.4.2. Scheduling a Feedback Packet 5.4.2. Scheduling an RTCP Feedback Packet
When an SSRC has a need to transmit a feedback packet in early mode 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 it follows the scheduling rules defined in Section 3.5 in RTP/AVPF
[RFC4585]. When following these rules the following clarifications [RFC4585]. When following these rules the following clarifications
need to be taken into account: need to be taken into account:
o That a session is considered to be point-to-point or multiparty 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 not based on the number of SSRCs, but the number of endpoints
directly seen in the RTP session by the endpoint. tbd: Clarify directly seen in the RTP session by the endpoint. tbd: Clarify
what is considered to "see" an endpoint? what is considered to "see" an endpoint?
o Note that when checking if there is already a scheduled compound o Note that when checking if there is already a scheduled compound
RTCP packet containing feedback messages (Step 2 in RTCP packet containing feedback messages (Step 2 in
Section 3.5.2), that check is done considering all local SSRCs. 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 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 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 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 which it could piggyback. Nor does it allow feedback to piggyback on
even regular RTCP packet transmissions that occur within even regular RTCP packet transmissions that occur within
T_max_fb_delay. A question is if either of these behaviours ought to T_max_fb_delay. A question is if either of these behaviours ought to
be allowed. be allowed. The latter appears simple and straight forward. Instead
of discarding a FB message in step 4a: alternative 2, one could place
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, 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 and in case any of the SSRCs schedule an RTCP packet for transmission
within that time, it includes this message. within that time, it includes this message. The former case can have
more widespread impact on the application, and possibly also on the
The former case can have more widespread impact on the application, RTCP bandwidth consumption as it allows for more massive bursts of
and possibly also on the RTCP bandwidth consumption as it allows for RTCP packets. Still, on a time scale of a regular reporting
more massive bursts of RTCP packets. Still, on a time scale of a interval, it ought to have no effect on the RTCP bandwidth as the
regular reporting interval, it ough to have no effect on the RTCP extra feedback messages increase the avg_rtcp_size.
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 13, line 36 skipping to change at page 13, line 26
endpoints need to correlate the media type of the SSRC being endpoints need to correlate the media type of the SSRC being
referenced in a message or packet and only use those that apply to referenced in a message or packet and only use those that apply to
that particular SSRC and its media type. Signalling solutions might that particular SSRC and its media type. Signalling solutions might
have shortcomings when it comes to indicating that a particular set have shortcomings when it comes to indicating that a particular set
of RTCP reports or feedback messages only apply to a particular media of RTCP reports or feedback messages only apply to a particular media
type within an RTP session. type within an RTP session.
6.1. Timing out SSRCs 6.1. Timing out SSRCs
This section discusses issues around timing out SSRCs. After the This section discusses issues around timing out SSRCs. After the
discussion, clarified and mandated behavior for SSRC timeout is discussion, clarified and mandated behaviour for SSRC timeout is
specified. specified.
6.1.1. AVPF T_rr_interval Behavior 6.1.1. AVPF T_rr_interval Behaviour
The RTP/AVPF profile includes a mechanism for suppressing regular The RTP/AVPF profile includes a mechanism for suppressing regular
RTCP reporting from being sent unnecessarily frequently if sufficient RTCP reporting from being sent unnecessarily frequently if sufficient
RTCP bandwidth is configured. This mechanism is defined in RTCP bandwidth is configured. This mechanism is defined in
Section 3.5.3 of [RFC4585], and can be summarized as follows: if less Section 3.5.3 of [RFC4585], and can be summarized as follows: if less
than a randomized T_rr_interval value has passed since the last than a randomized T_rr_interval value has passed since the last
regular report, and no feedback messages need to be sent, then the regular report, and no feedback messages need to be sent, then the
RTCP regular report is suppressed. The randomization is done by a RTCP regular report is suppressed. The randomization is done
linear randomizer in the interval 0.5 to 1.5 times T_rr_interval. linearly in the interval 0.5 to 1.5 times T_rr_interval. The
The randomized T_rr_interval is recalculated after every transmitted randomized T_rr_interval is recalculated after every transmitted
regular packet, i.e when t_rr_last was updated. The benefit of the regular packet, i.e when t_rr_last was updated. The benefit of the
suppression mechanism is that it avoids wasting bandwidth when there suppression mechanism is that it avoids wasting bandwidth when there
is nothing requiring frequent RTCP transmissions, but still allows is nothing requiring frequent RTCP transmissions, but still allows
utilization of the configured bandwidth when feedback is needed. utilization of the configured bandwidth when feedback is needed.
Unfortunately this suppression mechanism has some behaviors that are Unfortunately this suppression mechanism has some behaviour that is
less than ideal. First of all, the randomized T_rr_interval is less than ideal. First of all, the randomized T_rr_interval is
distributed over a larger range than the actual transmission interval distributed over a larger range than the actual transmission interval
for RTCP would be if T_rr_interval and Td had the same value. The for RTCP would be if T_rr_interval and Td had the same value. The
reconsideration mechanism and its compensation factor result in the reconsideration mechanism and its compensation factor result in the
actual RTCP transmission intervals for a Td having a distribution actual RTCP transmission intervals for a Td having a distribution
that is exponentially growing more likely with higher values, and is that is exponentially growing more likely with higher values, and is
bounded to the interval [0.5/1.21828, 1.5/1.21828]*Td, i.e. with a Td bounded to the interval [0.5/1.21828, 1.5/1.21828]*Td, i.e. with a
value of 5 s [2.052, 6.156]. In comparison, the suppression acts in Td value of 5 s [2.052, 6.156]. In comparison, the suppression acts
an interval that is 0.5 to 1.5 times the T_rr_interval, i.e. for in an interval that is 0.5 to 1.5 times the T_rr_interval, i.e. for
T_rr_interval = 5 s this is [2.5, 7.5]. T_rr_interval = 5 s this is [2.5, 7.5].
The effect of the above is that the time period between two RTCP The effect of the above is that the time period between two RTCP
packets when using T_rr_interval suppression can become very long packets when using T_rr_interval suppression can become very long
compared to the average input values. The longest time interval compared to the average input values. The longest time interval
between one transmitted regular RTCP compound packet and the next between one transmitted regular RTCP compound packet and the next
when T_rr_interval suppression is being used are: First the maximum when T_rr_interval suppression is being used are: First the maximum
T_rr_interval, i.e. 1.5*T_rr_interval. Assuming that the last T_rr_interval, i.e. 1.5*T_rr_interval. Assuming that the last
suppressed packet would have been sent at 1.5*T_rr_interval, the suppressed packet would have been sent at 1.5*T_rr_interval, the
maximum interval until a packet can be sent under the regular maximum interval until a packet can be sent under the regular
scheduling is 1.5/1.21828*Td. Thus, the maximum time in total is scheduling is 1.5/1.21828*Td. Thus, the maximum time in total is
1.5*T_rr_interval + 1.5/1.21828*Td. 1.5*T_rr_interval + 1.5/1.21828*Td.
If Td and T_rr_interval have the same value, i.e. the minimal If Td and T_rr_interval have the same value, i.e. the minimal
interval desired (T_rr_interval) and the actual actual average interval desired (T_rr_interval) and the actual actual average
interval specified by the RTCP scheduling algorithm (Td) are the interval specified by the RTCP scheduling algorithm (Td) are the
same, one might expect that RTCP packets would be sent according to same, one might expect that RTCP packets would be sent according to
the regular mechanism. Instead, this algorithm results in the RTCP the regular mechanism. Instead, this algorithm results in the RTCP
packets being sent anywhere from 0.5*Td to ~2.731*Td. The packets being sent anywhere from 0.5*Td to ~2.731*Td. The
probability distribution over that time is also non-trivial in its probability distribution over that time is also non-trivial in its
shape, somewhat similar to a saw tooth. shape, somewhat similar to a saw tooth.
Thus, we recommend that the AVPF regular transmission mechanism is Thus, we recommend that the AVPF regular transmission mechanism is
revised in the future. This issue also has further implications as revised in the future. This issue also has further implications as
discussed in the next section. discussed in the next section.
6.1.2. Avoiding Pre-mature Timeout 6.1.2. Avoiding Premature Timeout
In RTP/AVP [RFC3550] the timeout behavior is simple and is 5 times In RTP/AVP [RFC3550] the timeout behaviour is simple and is 5 times
Td, where Td is calculated with a Tmin value of 5 seconds. In other Td, where Td is calculated with a Tmin value of 5 seconds. In other
words, if the RTCP bandwidth allowed for an RTCP interval more words, if the RTCP bandwidth allowed for an RTCP interval more
frequent than every 5 seconds on average, then timeout happened after frequent than every 5 seconds on average, then timeout happened after
5*Td = 25 seconds of no activity from the SSRC (RTP or RTCP), 5*Td = 25 seconds of no activity from the SSRC (RTP or RTCP),
otherwise it was 5 average reporting intervals. otherwise it was 5 average reporting intervals.
RTP/AVPF [RFC4585] introduced two different behaviors depending on RTP/AVPF [RFC4585] introduced two different behaviours depending on
the value of T_rr_interval. When T_rr_interval was 0, it defaulted the value of T_rr_interval. When T_rr_interval was 0, it defaulted
to the same Td calculation in RTP/AVP [RFC3550]. However, when to the same Td calculation in RTP/AVP [RFC3550]. However, when
T_rr_interval is non-zero the Tmin value become T_rr_interval in that T_rr_interval is non-zero the Tmin value become T_rr_interval in that
calculation, most likely to enable speed up the detection of timed calculation, most likely to enable speed up the detection of timed
out SSRCs. However, using a non-zero T_rr_interval has two out SSRCs. However, using a non-zero T_rr_interval has two
consequences for RTP's behavior. consequences for RTP behaviour.
First, the number of actually sent RTCP packets for an SSRC that First, the number of actually sent RTCP packets for an SSRC that
currently is not an active RTP sender can become very low due to the currently is not an active RTP sender can become very low due to the
issue discussed above in Section 6.1.1. As the RTCP packet interval issue discussed above in Section 6.1.1. As the RTCP packet interval
can be as long as 2.73*Td, then during a 5*Td time period an endpoint can be as long as 2.73*Td, then during a 5*Td time period an endpoint
may in fact transmit only a single RTCP packet. The long intervals may in fact transmit only a single RTCP packet. The long intervals
result in fewer RTCP packets, to a point where a one or two packet result in fewer RTCP packets, to a point where a one or two packet
losses in RTCP result in timing out an SSRC. losses in RTCP result in timing out an SSRC.
Second, the change also increased RTP/AVPF's brittleness to both Second, the change also increased RTP/AVPF's brittleness to both
packet loss and configuration errors. In many cases, when one packet loss and configuration errors. In many cases, when one
desires to use RTP/AVPF for its feedback, one will ensure that RTCP desires to use RTP/AVPF for its feedback, one will ensure that RTCP
is configured for more frequent transmissions on average than every 5 is configured for more frequent transmissions on average than every 5
seconds. Thus, many more RTP and RTCP packets can be transmitted seconds. Thus, many more RTP and RTCP packets can be transmitted
during the time interval. Lets consider an implementation that would during the time interval. Lets consider an implementation that would
follow the AVP or AVPF with T_rr_interval = 0 rules for timeout, also follow the RTP/AVP or RTP/AVPF with T_rr_interval = 0 rules for
when T_rr_interval is not zero. In such a case when the configured timeout, also when T_rr_interval is not zero. In such a case when
value of the T_rr_interval is significantly smaller than 5 seconds, the configured value of the T_rr_interval is significantly smaller
e.g. less than 1 second, then a difference between using 0.1 seconds than 5 seconds, e.g. less than 1 second, then a difference between
and 0.6 seconds has no significant impact on when an SSRC will be using 0.1 seconds and 0.6 seconds has no significant impact on when
timed out. However, such a configuration difference between two an SSRC will be timed out. However, such a configuration difference
endpoints following RFC 4585 will result in that the endpoint between two endpoints following RFC 4585 will result in that the
configured with T_rr_interval = 0.1 will frequently timeout SSRCs endpoint configured with T_rr_interval = 0.1 will frequently timeout
currently not sending RTP, from the endpoint configured with 0.6, as SSRCs currently not sending RTP, from the endpoint configured with
that is six times the Td value used by the endpoint configured with 0.6, as that is six times the Td value used by the endpoint
T_rr_interval=0.1, assuming sufficient bandwidth. For this reason configured with T_rr_interval=0.1, assuming sufficient bandwidth.
such a change is implemented below in Section 6.1.4. For this reason such a change is implemented below in Section 6.1.4.
6.1.3. AVP and AVPF Interoperability 6.1.3. RTP/AVP and RTP/AVPF Interoperability
If endpoints implementing the RTP/AVP and RTP/AVPF profiles (or their If endpoints implementing the RTP/AVP and RTP/AVPF profiles (or their
secure variants) are combined in a single RTP session, and the RTP/ secure variants) are combined in a single RTP session, and the 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/AVPF lower than 5 seconds, then there is a risk that the RTP/AVPF
endpoints will prematurely timeout the RTP/AVP SSRCs due to their endpoints will prematurely timeout the RTP/AVP SSRCs due to their
different RTCP timeout intervals. Conversely, if the RTP/AVPF different RTCP timeout intervals. Conversely, if the RTP/AVPF
endpoints use a T_rr_interval that is significant larger than 5 endpoints use a T_rr_interval that is significant larger than 5
seconds, there is a risk that the RTP/AVP endpoints will timeout the seconds, there is a risk that the RTP/AVP endpoints will timeout the
RTP/AVPF SSRCs. RTP/AVPF SSRCs.
skipping to change at page 16, line 16 skipping to change at page 15, line 51
RECOMMENDED), and the AVPF endpoint is not updated to follow this RECOMMENDED), and the AVPF endpoint is not updated to follow this
specification, then the RTP/AVPF session SHOULD use a non-zero specification, then the RTP/AVPF session SHOULD use a non-zero
T_rr_interval that is 4 seconds. T_rr_interval that is 4 seconds.
It might appear strange to use a T_rr_interval of 4 seconds. It It might appear strange to use a T_rr_interval of 4 seconds. It
might be intuitive that this value ought to be 5 seconds, as then might be intuitive that this value ought to be 5 seconds, as then
both the RTP/AVP and RTP/AVPF would use the same timeout period. both the RTP/AVP and RTP/AVPF would use the same timeout period.
However, considering regular RTCP transmission and their packet However, considering regular RTCP transmission and their packet
intervals for RTP/AVPF its mean value will (with non-zero intervals for RTP/AVPF its mean value will (with non-zero
T_rr_interval) be larger than T_rr_interval due to the scheduling T_rr_interval) be larger than T_rr_interval due to the scheduling
algorithm's behavior as discussed in Section 6.1.1. Thus, to enable algorithm's behaviour as discussed in Section 6.1.1. Thus, to enable
an equal amount of regular RTCP transmissions in each directions an equal amount of regular RTCP transmissions in each directions
between RTP/AVP and RTP/AVPF endpoints, taking the altered timeout between RTP/AVP and RTP/AVPF endpoints, taking the altered timeout
intervals into account, the optimal value is around four (4), where intervals into account, the optimal value is around four (4), where
almost four transmissions will on average occur in each direction almost four transmissions will on average occur in each direction
between the different profile types given an otherwise good between the different profile types given an otherwise good
configuration of parameters in regards to T_rr_interval. If the RTCP configuration of parameters in regards to T_rr_interval. If the RTCP
bandwidth parameters are selected so that Td based on bandwidth is bandwidth parameters are selected so that Td based on bandwidth is
close to 4, i.e. close to T_rr_interval the risk increases that RTP/ 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 AVPF SSRCs will be timed out by RTP/AVP endpoints, as the RTP/AVPF
SSRC might only manage two transmissions in the timeout period. SSRC might only manage two transmissions in the timeout period.
6.1.4. Specified Behavior 6.1.4. Specified Behaviour
The above considerations result in the following clarification and The above considerations result in the following clarification and
RTP/AVPF specification change. RTP/AVPF specification change.
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, the timeout interval For RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF, the timeout interval
SHALL be calculated using a multiplier of 5, i.e. the timeout SHALL be calculated using a multiplier of 5, i.e. the timeout
interval becomes 5*Td. The Td calculation SHALL be done using a Tmin interval becomes 5*Td. The Td calculation SHALL be done using a Tmin
value of 5 seconds, not the reduced minimal interval even if used to value of 5 seconds, not the reduced minimal interval even if used to
calculate RTCP packet transmission intervals. This changes the calculate RTCP packet transmission intervals. This changes the
behavior for the RTP/AVPF or RTP/SAVPF profiles when T_rr_interval != behaviour for the RTP/AVPF or RTP/SAVPF profiles when T_rr_interval
0, a behavior defined in Section 3.5.4 of RFC 4585, i.e. Tmin in the != 0, a behaviour defined in Section 3.5.4 of RFC 4585, i.e. Tmin in
Td calculation is the T_rr_interval. the Td calculation is the T_rr_interval.
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. First, it is downsides of the shared RTCP packet intervals. First, it is
considered what possibilites exist for the RTP/AVP [RFC3551] profile, considered what possibilities exist for the RTP/AVP [RFC3551]
then what additional tools are provided by RTP/AVPF [RFC4585]. profile, then what additional tools are provided by RTP/AVPF
[RFC4585].
6.2.1. RTP/AVP and RTP/SAVP 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,
skipping to change at page 18, line 9 skipping to change at page 17, line 43
all SSRC being senders, resulting in everyone sharing the all SSRC being senders, resulting in everyone sharing the
available bandwidth. Secondly we will select an average RTCP available bandwidth. Secondly we will select an average RTCP
packet size. This packet will consist of an SR, containing (n-1) packet size. This packet will consist of an SR, containing (n-1)
report blocks up to 31 report blocks, and an SDES item with at report blocks up to 31 report blocks, and an SDES item with at
least a CNAME (17 bytes in size) in it. Such a basic packet will 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 be 800 bytes for n>=32. With these parameters, and as the
bandwidth goes up the time interval is proportionally decreased bandwidth goes up the time interval is proportionally decreased
(due to minimal scaling), thus all the example bandwidths 72 (due to minimal scaling), thus all the example bandwidths 72
kbps, 360 kbps and 9 Mbps all support 9 SSRCs. kbps, 360 kbps and 9 Mbps all support 9 SSRCs.
d. The actual transmission interval for a Td value is d. The actual transmission interval for a Td value is [0.5*Td/
[0.5*Td/1.21828,1.5*Td/1.21828], which means that for Td = 5 1.21828,1.5*Td/1.21828], which means that for Td = 5 seconds, the
seconds, the interval is actually [2.052,6.156] and the interval is actually [2.052,6.156] and the distribution is not
distribution is not uniform, but rather exponentially-increasing. uniform, but rather exponentially-increasing. The probability
The probability for sending at time X, given it is within the for sending at time X, given it is within the interval, is
interval, is probability of picking X in the interval times the probability of picking X in the interval times the probability to
probability to randomly picking a number that is <=X within the randomly picking a number that is <=X within the interval with an
interval with an uniform probability distribution. This results uniform probability distribution. This results in that the
in that the majority of the probability mass is above the Td majority of the probability mass is above the Td value.
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 6.2.2. RTP/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. governing factors.
Now it also becomes important to separate between the application's Now it also becomes important to separate between the application's
need for regular reports and RTCP feedback packet types. In both need for regular reports and RTCP feedback packet types. In both
regular RTCP mode, as in Early RTCP Mode, the usage of the regular RTCP mode, as in Early RTCP Mode, the usage of the
T_rr_interval prevents regular RTCP packets, i.e. packets without any T_rr_interval prevents regular RTCP packets, i.e. packets without
Feedback packets, to be sent more often than T_rr_interval. This any Feedback packets, to be sent more often than T_rr_interval. This
value is applied to prevent any regular RTCP packet to be sent less value is applied to prevent any regular RTCP packet to be sent less
than T_rr_interval times a uniformly distributed random value from than T_rr_interval times a uniformly distributed random value from
the interval [0.5,1.5] after the previous regular packet packet. The the interval [0.5,1.5] after the previous regular packet packet. The
random value recalculated after each regular RTCP packet random value recalculated after each regular RTCP packet
transmission. 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
skipping to change at page 19, line 19 skipping to change at page 19, line 5
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 20, line 10 skipping to change at page 19, line 42
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 allow use of per- for keying mechanisms such as Mikey [RFC3830] which allow use of per-
SSRC 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. IANA Considerations
No IANA actions needed.
9. 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. Do we need to provide a recommendation for unicast session 1. 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? 2. RTCP parameters for common scenarios in Section 6.2?
3. Is scheduling algorithm working well with dynamic changes? 3. Is scheduling algorithm working well with dynamic changes?
4. Are the scheduling algorithm changes impacting previous 4. Are the scheduling algorithm changes impacting previous
implementations in such a way that the report aggregation has to implementations in such a way that the report aggregation has to
be agreed on, and thus needs to be considered as an optimization? be agreed on, and thus needs to be considered as an optimization?
5. An open question is if any improvements or clarifications ought 5. An open question is if any improvements or clarifications ought
to be allowed regarding FB message scheduling in multi-SSRC to be allowed regarding FB message scheduling in multi-SSRC
endpoints. endpoints.
9. IANA Considerations
No IANA actions needed.
10. References 10. References
10.1. Normative References 10.1. Normative References
[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.
skipping to change at page 21, line 27 skipping to change at page 21, line 10
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-05 (work in ietf-avtcore-multi-media-rtp-session-05 (work in
progress), February 2014. progress), February 2014.
[I-D.ietf-avtcore-rtp-multi-stream-optimisation] [I-D.ietf-avtcore-rtp-multi-stream-optimisation]
Lennox, J., Westerlund, M., Wu, W., and C. Perkins, Lennox, J., Westerlund, M., Wu, Q., 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-02 (work draft-ietf-avtcore-rtp-multi-stream-optimisation-00 (work
in progress), February 2014. 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-02 (work in progress), ietf-avtcore-rtp-topologies-update-02 (work in progress),
May 2014. May 2014.
[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-15 (work in progress), May 2014. framework-16 (work in progress), June 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,
"Negotiating Media Multiplexing Using the Session "Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
negotiation-07 (work in progress), April 2014. negotiation-07 (work in progress), April 2014.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551, Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003. July 2003.
skipping to change at page 22, line 17 skipping to change at page 21, line 48
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. [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588, Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006. July 2006.
[RFC6190] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis, [RFC6190] Wenger, S., Wang, Y.-K., Schierl, T., and A.
"RTP Payload Format for Scalable Video Coding", RFC 6190, Eleftheriadis, "RTP Payload Format for Scalable Video
May 2011. Coding", RFC 6190, May 2011.
Appendix A. Changes From Earlier Versions
Note to the RFC-Editor: please remove this section prior to
publication as an RFC.
A.1. Changes From 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-
ietf-avtcore-rtp-multi-stream-optimization-00.
o Added RTCP tuning considerations from draft-ietf-avtcore-multi-
media-rtp-session-02.
A.4. Changes From Individual Draft -02
o Resubmitted as working group draft.
o Updated references.
A.5. Changes From Individual Draft -01
o Merged with draft-wu-avtcore-multisrc-endpoint-adver.
o Changed how Reporting Groups are indicated in RTCP, to make it
clear which source(s) is the group's reporting sources.
o Clarified the rules for when sources can be placed in the same
reporting group.
o Clarified that mixers and translators need to pass reporting group
SDES information if they are forwarding RR and SR traffic from
members of a reporting group.
A.6. Changes From Individual Draft -00
o Added the Reporting Group semantic to explicitly indicate which
sources come from a single endpoint, rather than leaving it
implicit.
o Specified that Reporting Group semantics (as they now are) apply
to AVPF and XR, as well as to RR/SR report blocks.
o Added a description of the cascaded source-projecting mixer, along
with a calculation of its RTCP overhead if reporting groups are
not in use.
o Gave some guidance on how the flexibility of RTCP randomization
allows some freedom in RTCP multiplexing.
o Clarified the language of several of the recommendations.
o Added an open issue discussing how avg_rtcp_size ought to be
calculated for multiplexed RTCP.
o Added an open issue discussing how RTCP bandwidths are to be
chosen for sessions where source bandwidths greatly differ.
Authors' Addresses Authors' Addresses
Jonathan Lennox Jonathan Lennox
Vidyo, Inc. Vidyo, Inc.
433 Hackensack Avenue 433 Hackensack Avenue
Seventh Floor Seventh Floor
Hackensack, NJ 07601 Hackensack, NJ 07601
US USA
Email: jonathan@vidyo.com Email: jonathan@vidyo.com
Magnus Westerlund Magnus Westerlund
Ericsson Ericsson
Farogatan 6 Farogatan 6
SE-164 80 Kista SE-164 80 Kista
Sweden Sweden
Phone: +46 10 714 82 87 Phone: +46 10 714 82 87
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