draft-ietf-avtcore-rtp-multi-stream-07.txt   draft-ietf-avtcore-rtp-multi-stream-08.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: September 10, 2015 Q. Wu Expires: January 7, 2016 Q. Wu
Huawei Huawei
C. Perkins C. Perkins
University of Glasgow University of Glasgow
March 9, 2015 July 6, 2015
Sending Multiple Media Streams in a Single RTP Session Sending Multiple Media Streams in a Single RTP Session
draft-ietf-avtcore-rtp-multi-stream-07 draft-ietf-avtcore-rtp-multi-stream-08
Abstract Abstract
This memo expands and clarifies the behaviour of Real-time Transport This memo expands and clarifies the behaviour of Real-time Transport
Protocol (RTP) endpoints that use multiple synchronization sources Protocol (RTP) endpoints that use multiple synchronization sources
(SSRCs). This occurs, for example, when an endpoint sends multiple (SSRCs). This occurs, for example, when an endpoint sends multiple
media streams in a single RTP session. This memo updates RFC 3550 media streams in a single RTP session. This memo updates RFC 3550
with regards to handling multiple SSRCs per endpoint in RTP sessions, with regards to handling multiple SSRCs per endpoint in RTP sessions,
with a particular focus on RTCP behaviour. It also updates RFC 4585 with a particular focus on RTCP behaviour. It also updates RFC 4585
to update and clarify the calculation of the timeout of SSRCs and the to update and clarify the calculation of the timeout of SSRCs and the
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Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
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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
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on September 10, 2015. This Internet-Draft will expire on January 7, 2016.
Copyright Notice Copyright Notice
Copyright (c) 2015 IETF Trust and the persons identified as the Copyright (c) 2015 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
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2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Use Cases For Multi-Stream Endpoints . . . . . . . . . . . . 3 3. Use Cases For Multi-Stream Endpoints . . . . . . . . . . . . 3
3.1. Endpoints with Multiple Capture Devices . . . . . . . . . 3 3.1. Endpoints with Multiple Capture Devices . . . . . . . . . 3
3.2. Multiple Media Types in a Single RTP Session . . . . . . 4 3.2. Multiple Media Types in a Single RTP Session . . . . . . 4
3.3. Multiple Stream Mixers . . . . . . . . . . . . . . . . . 4 3.3. Multiple Stream Mixers . . . . . . . . . . . . . . . . . 4
3.4. Multiple SSRCs for a Single Media Source . . . . . . . . 4 3.4. Multiple SSRCs for a Single Media Source . . . . . . . . 4
4. Use of RTP by endpoints that send multiple media streams . . 5 4. Use of RTP by endpoints that send multiple media streams . . 5
5. Use of RTCP by Endpoints that send multiple media streams . . 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 . . . . . . . . . . . . . . . 6
5.3. Aggregation of Reports into Compound RTCP Packets . . . . 6 5.3. Aggregation of Reports into Compound RTCP Packets . . . . 7
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 when Aggregating Multiple SSRCs . . . 9
5.4. Use of RTP/AVPF Feedback . . . . . . . . . . . . . . . . 11 5.4. Use of RTP/AVPF or RTP/SAVPF Feedback . . . . . . . . . . 11
5.4.1. Choice of SSRC for Feedback Packets . . . . . . . . . 11 5.4.1. Choice of SSRC for Feedback Packets . . . . . . . . . 11
5.4.2. Scheduling an RTCP Feedback Packet . . . . . . . . . 12 5.4.2. Scheduling an RTCP Feedback Packet . . . . . . . . . 12
6. RTCP Considerations for Streams with Disparate Rates . . . . 14 6. Adding and Removing SSRCs . . . . . . . . . . . . . . . . . . 14
6.1. Timing out SSRCs . . . . . . . . . . . . . . . . . . . . 16 6.1. Adding RTP Streams . . . . . . . . . . . . . . . . . . . 14
6.1.1. Problems with RTP/AVPF the T_rr_interval Parameter . 16 6.2. Removing RTP Streams . . . . . . . . . . . . . . . . . . 15
6.1.2. Avoiding Premature Timeout . . . . . . . . . . . . . 17 7. RTCP Considerations for Streams with Disparate Rates . . . . 16
6.1.3. Interoperability Between RTP/AVP and RTP/AVPF . . . . 18 7.1. Timing out SSRCs . . . . . . . . . . . . . . . . . . . . 17
6.1.4. Updated SSRC Timeout Rules . . . . . . . . . . . . . 18 7.1.1. Problems with the RTP/AVPF T_rr_interval Parameter . 18
6.2. Tuning RTCP transmissions . . . . . . . . . . . . . . . . 19 7.1.2. Avoiding Premature Timeout . . . . . . . . . . . . . 19
6.2.1. RTP/AVP and RTP/SAVP . . . . . . . . . . . . . . . . 19 7.1.3. Interoperability Between RTP/AVP and RTP/AVPF . . . . 19
6.2.2. RTP/AVPF and RTP/SAVPF . . . . . . . . . . . . . . . 20 7.1.4. Updated SSRC Timeout Rules . . . . . . . . . . . . . 20
7. Security Considerations . . . . . . . . . . . . . . . . . . . 22 7.2. Tuning RTCP transmissions . . . . . . . . . . . . . . . . 20
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 22 7.2.1. RTP/AVP and RTP/SAVP . . . . . . . . . . . . . . . . 21
9. References . . . . . . . . . . . . . . . . . . . . . . . . . 22 7.2.2. RTP/AVPF and RTP/SAVPF . . . . . . . . . . . . . . . 22
9.1. Normative References . . . . . . . . . . . . . . . . . . 22 8. Security Considerations . . . . . . . . . . . . . . . . . . . 24
9.2. Informative References . . . . . . . . . . . . . . . . . 23 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 24
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 24 10. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 24
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 24
11.1. Normative References . . . . . . . . . . . . . . . . . . 24
11.2. Informative References . . . . . . . . . . . . . . . . . 25
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 26
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 designed, 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. Recently, however, a number of scenarios have emerged in which type. Recently, however, a number of scenarios have emerged in which
endpoints wish to send multiple RTP media streams, distinguished by endpoints wish to send multiple RTP media streams, distinguished by
distinct RTP synchronization source (SSRC) identifiers, in a single distinct RTP synchronization source (SSRC) identifiers, in a single
RTP session. These are outlined in Section 3. Although the initial RTP session. These are outlined in Section 3. Although the initial
design of RTP did consider such scenarios, the specification was not design of RTP 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 specification
are thus somewhat unclear. is thus somewhat unclear in places.
This memo updates [RFC3550] to clarify behaviour in use cases where This memo updates [RFC3550] to clarify behaviour in use cases where
endpoints use multiple SSRCs. It also updates [RFC4585] in regards endpoints use multiple SSRCs. It also updates [RFC4585] to resolve
to the timeout of inactive SSRCs to resolve problematic behaviour as problems with regards to timeout of inactive SSRCs, and to clarify
well as clarifying the inclusion of feedback messages. behaviour around inclusion of feedback messages.
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. Endpoints with Multiple Capture Devices 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
simultaneous RTP streams in a session is the scenario where an simultaneous RTP streams in a single RTP session is when an endpoint
endpoint has multiple capture devices, and thus media sources, of the has multiple capture devices, and hence can generate multiple media
same media type and characteristics. For example, telepresence sources, of the same media type and characteristics. For example,
endpoints, of the type described by the CLUE Telepresence Framework telepresence systems of the type described by the CLUE Telepresence
[I-D.ietf-clue-framework], often have multiple cameras or microphones Framework [I-D.ietf-clue-framework] often have multiple cameras or
covering various areas of a room, and hence send several RTP streams. microphones covering various areas of a room, and hence send several
RTP streams of each type within a single RTP session.
3.2. Multiple Media Types in a Single RTP Session 3.2. Multiple Media Types in a Single RTP Session
Recent work has updated 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 remove the historical [I-D.ietf-mmusic-sdp-bundle-negotiation] to remove the historical
assumption in RTP that media sources of different media types would assumption in RTP that media sources of different media types would
always be sent on different RTP sessions. In this work, a single always be sent on different RTP sessions. In this work, a single
endpoint's audio and video RTP media streams (for example) are endpoint's audio and video RTP media streams (for example) are
instead sent in a single RTP session to reduce the number of instead sent in a single RTP session to reduce the number of
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in RTP). This sort of device is closer to being an RTP mixer than an in RTP). This sort of device is closer to being an RTP mixer than an
RTP translator, in that it terminates RTCP reporting about the mixed RTP translator, in that it terminates RTCP reporting about the mixed
streams, and it can re-write SSRCs, timestamps, and sequence numbers, streams, and it can re-write SSRCs, timestamps, and sequence numbers,
as well as the contents of the RTP payloads, and can turn sources on as well as the contents of the RTP payloads, and can turn sources on
and off at will without appearing to be generating packet loss. Each and off at will without appearing to be generating packet loss. Each
projected stream will typically preserve its original RTCP source projected stream will typically preserve its original RTCP source
description (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 multiple SSRCs can be used to send
the usage of multiple SSRCs within the same RTP session. Transport data from a single media source within a single RTP session. These
robustness tools like RTP Retransmission [RFC4588] result in multiple include, but are not limited to, transport robustness tools, such as
SSRCs, one with source data, and another with the repair data. the RTP retransmission payload format [RFC4588], that require one
Scalable encoders and their RTP payload formats, like H.264's SSRC to be used for the media data and another SSRC for the repair
extension for Scalable Video Coding(SVC) [RFC6190] can be transmitted data. Similarly, some layered media encoding schemes, for example
in a configuration where the scalable layers are distributed over H.264 SVC [RFC6190], can be used in a configuration where each layer
multiple SSRCs within the same session, to enable RTP packet stream is sent using a different SSRC within a single RTP session.
level (SSRC) selection and routing in conferencing middleboxes.
4. Use of RTP by endpoints that send multiple media streams 4. Use of RTP by endpoints that send multiple media streams
RTP is inherently a group communication protocol. Each endpoint in
an RTP session will use one or more SSRCs, as will some types of RTP
level middlebox. Accordingly, unless restrictions on the number of
SSRCs have been signalled, RTP endpoints can expect to receive RTP
data packets sent using with a number of different SSRCs, within a
single RTP session. This can occur irrespective of whether the RTP
session is running over a point-to-point connection or a multicast
group, since middleboxes can be used to connect multiple transport
connections together into a single RTP session (the RTP session is
defined by the shared SSRC space, not by the transport connections).
Furthermore, if RTP mixers are used, some SSRCs might only be visible
in the contributing source (CSRC) list of an RTP packet and in RTCP,
and might not appear directly as the SSRC of an RTP data packet.
Every RTP endpoint will have an allocated share of the available Every RTP endpoint will have an allocated share of the available
session bandwidth, as determined by signalling and congestion session bandwidth, as determined by signalling and congestion
control. The endpoint MUST keep its total media sending rate within control. The endpoint MUST keep its total media sending rate within
this share. However, endpoints that send multiple media streams do this share. However, endpoints that send multiple media streams do
not necessarily need to subdivide their share of the available not necessarily need to subdivide their share of the available
bandwidth independently or uniformly to each media stream and its bandwidth independently or uniformly to each media stream and its
SSRCs. In particular, an endpoint can vary the allocation to SSRCs. In particular, an endpoint can vary the bandwidth allocation
different streams depending on their needs, and can dynamically to different streams depending on their needs, and can dynamically
change the bandwidth allocated to different SSRCs (for example, by change the bandwidth allocated to different SSRCs (for example, by
using a variable rate codec), provided the total sending rate does using a variable rate codec), provided the total sending rate does
not exceed its allocated share. This includes enabling or disabling not exceed its allocated share. This includes enabling or disabling
media streams and their redundancy streams as more or less bandwidth media streams, or their redundancy streams, as more or less bandwidth
becomes available. becomes available.
5. Use of RTCP by Endpoints that send multiple media streams 5. Use of RTCP by Endpoints that send multiple media streams
The RTP Control Protocol (RTCP) is defined in Section 6 of [RFC3550]. The RTP Control Protocol (RTCP) is defined in Section 6 of [RFC3550].
The description of the protocol is phrased in terms of the behaviour The description of the protocol is phrased in terms of the behaviour
of "participants" in an RTP session, under the assumption that each of "participants" in an RTP session, under the assumption that each
endpoint is a participant with a single SSRC. However, for correct endpoint is a participant with a single SSRC. However, for correct
operation in cases where endpoints can send multiple media streams, operation in cases where endpoints have multiple SSRC values, the
the specification needs to be interpreted with each SSRC counting as specification MUST be interpreted as each SSRC counting as a separate
a participant in the session, so that an endpoint that has multiple participant in the RTP session, so that an endpoint that has multiple
SSRCs counts as multiple participants. The following describes SSRCs counts as multiple participants.
several concrete cases where this applies.
5.1. RTCP Reporting Requirement 5.1. RTCP Reporting Requirement
An RTP endpoint that has multiple SSRCs MUST treat each SSRC as a An RTP endpoint that has multiple SSRCs MUST treat each SSRC as a
separate participant in the RTP session, sending RTCP reports for separate participant in the RTP session. Each SSRC will maintain its
each of its SSRCs in every RTCP reporting interval. If the mechanism own RTCP-related state information, and hence will have its own RTCP
in [I-D.ietf-avtcore-rtp-multi-stream-optimisation] is not used, then reporting interval that determines when it sends RTCP reports. If
each SSRC will send RTCP reports for all other SSRCs, including those the mechanism in [I-D.ietf-avtcore-rtp-multi-stream-optimisation] is
co-located at the same endpoint. not used, then each SSRC will send RTCP reports for all other SSRCs,
including those co-located at the same endpoint.
If the endpoint has some SSRCs that are sending data and some that If the endpoint has some SSRCs that are sending data and some that
are only receivers, then they will receive different shares of the are only receivers, then they will receive different shares of the
RTCP bandwidth and calculate different base RTCP reporting intervals. RTCP bandwidth and calculate different base RTCP reporting intervals.
Otherwise, all SSRCs at an endpoint will calculate the same base RTCP Otherwise, all SSRCs at an endpoint will calculate the same base RTCP
reporting interval. The actual reporting intervals for each SSRC are reporting interval. The actual reporting intervals for each SSRC are
randomised in the usual way, but reports can be aggregated as randomised in the usual way, but reports can be aggregated as
described in Section 5.3. described in Section 5.3.
5.2. Initial Reporting Interval 5.2. Initial Reporting Interval
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The basic assumption is that this also ought to apply in the case of The basic assumption is that this also ought to apply in the case of
multiple SSRCs. Caution has to be exercised, however, when an multiple SSRCs. Caution has to be exercised, however, when an
endpoint (or middlebox) with a large number of SSRCs joins a unicast endpoint (or middlebox) with a large number of SSRCs joins a unicast
session, since immediate transmission of many RTCP reports can create session, since immediate transmission of many RTCP reports can create
a significant burst of traffic, leading to transient congestion and a significant burst of traffic, leading to transient congestion and
packet loss due to queue overflows. packet loss due to queue overflows.
To ensure that the initial burst of traffic generated by an RTP To ensure that the initial burst of traffic generated by an RTP
endpoint is no larger than would be generated by a TCP connection, an endpoint is no larger than would be generated by a TCP connection, an
RTP endpoint MUST NOT send more than four compound RTCP packets with RTP endpoint MUST NOT send more than four compound RTCP packets with
zero initial delay when it joins a session. Each of those initial zero initial delay when it joins an RTP session, independently of the
compound RTCP packets MAY include aggregated reports from multiple number of SSRCs used by the endpoint. Each of those initial compound
SSRCs, provided the total compound RTCP packet size does not exceed RTCP packets MAY include aggregated reports from multiple SSRCs,
the MTU, and the avg_rtcp_packet_size is maintained as in provided the total compound RTCP packet size does not exceed the MTU,
Section 5.3.1. Aggregating reports from several SSRCs in the initial and the avg_rtcp_size is maintained as in Section 5.3.1. Aggregating
compound RTCP packets allows a substantial number of SSRCs to report reports from several SSRCs in the initial compound RTCP packets
immediately. Endpoints SHOULD prioritize reports on SSRCs that are allows a substantial number of SSRCs to report immediately.
likely to be most immediately useful, e.g., for SSRCs that are Endpoints SHOULD prioritize reports on SSRCs that are likely to be
initially senders. most immediately useful, e.g., for SSRCs that are initially senders.
An endpoint that needs to report on more SSRCs than will fit into the An endpoint that needs to report on more SSRCs than will fit into the
four compound RTCP reports that can be sent immediately MUST send the four compound RTCP reports that can be sent immediately MUST send the
other reports later, following the usual RTCP timing rules including other reports later, following the usual RTCP timing rules including
timer reconsideration. Those reports MAY be aggregated as described timer reconsideration. Those reports MAY be aggregated as described
in Section 5.3. in Section 5.3.
Note: The above is based on an TCP initial window of 4 packets, Note: The above is based on an TCP initial window of 4 packets,
not the larger initial windows which there is an ongoing not the larger TCP initial windows for which there is an ongoing
experiment with. The reason for this is a desire to be experiment. The reason for this is a desire to be conservative,
conservative as an RTP endpoint will also in many cases commence since an RTP endpoint will also in many cases start sending RTP
RTP transmission at the same time as these initial RTCP packets data packets at the same time as these initial RTCP packets are
are sent. sent.
5.3. Aggregation of Reports into Compound RTCP Packets 5.3. Aggregation of Reports into Compound RTCP Packets
As outlined in Section 5.1, an endpoint with multiple SSRCs has to As outlined in Section 5.1, an endpoint with multiple SSRCs has to
treat each SSRC as a separate participant when it comes to sending treat each SSRC as a separate participant when it comes to sending
RTCP reports. This will lead to each SSRC sending a compound RTCP RTCP reports. This will lead to each SSRC sending a compound RTCP
packet in each reporting interval. Since these packets are coming packet in each reporting interval. Since these packets are coming
from the same endpoint, it might reasonably be expected that they can from the same endpoint, it might reasonably be expected that they can
be aggregated to reduce overheads. Indeed, Section 6.1 of [RFC3550] be aggregated to reduce overheads. Indeed, Section 6.1 of [RFC3550]
allows RTP translators and mixers to aggregate packets in similar allows RTP translators and mixers to aggregate packets in similar
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This allows RTP translators and mixers to generate compound RTCP This allows RTP translators and mixers to generate compound RTCP
packets that contain multiple SR or RR packets from different SSRCs, packets that contain multiple SR or RR packets from different SSRCs,
as well as any of the other packet types. There are no restrictions as well as any of the other packet types. There are no restrictions
on the order in which the RTCP packets can occur within the compound on the order in which the RTCP packets can occur within the compound
packet, except the regular rule that the compound RTCP packet starts packet, except the regular rule that the compound RTCP packet starts
with an SR or RR packet. Due to this rule, correctly implemented RTP with an SR or RR packet. Due to this rule, correctly implemented RTP
endpoints will be able to handle compound RTCP packets that contain endpoints will be able to handle compound RTCP packets that contain
RTCP packets relating to multiple SSRCs. RTCP packets relating to multiple SSRCs.
Accordingly, endpoints that use multiple SSRCs MAY aggregate the RTCP Accordingly, endpoints that use multiple SSRCs can aggregate the RTCP
packets sent by their different SSRCs into compound RTCP packets, packets sent by their different SSRCs into compound RTCP packets,
provided 1) the resulting compound RTCP packets begin with an SR or provided 1) the resulting compound RTCP packets begin with an SR or
RR packet; 2) they maintain the average RTCP packet size as described RR packet; 2) they maintain the average RTCP packet size as described
in Section 5.3.1; and 3) they schedule packet transmission and manage in Section 5.3.1; and 3) they schedule packet transmission and manage
aggregation as described in Section 5.3.2. aggregation as described in Section 5.3.2.
5.3.1. Maintaining AVG_RTCP_SIZE 5.3.1. Maintaining AVG_RTCP_SIZE
The RTCP scheduling algorithm in [RFC3550] works on a per-SSRC basis. The RTCP scheduling algorithm in [RFC3550] works on a per-SSRC basis.
Each SSRC sends a single compound RTCP packet in each RTCP reporting Each SSRC sends a single compound RTCP packet in each RTCP reporting
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proportional to the number of SSRCs aggregated into compound RTCP proportional to the number of SSRCs aggregated into compound RTCP
packets and the size of set of SSRCs being aggregated relative to the packets and the size of set of SSRCs being aggregated relative to the
total number of participants. This increased RTCP reporting interval total number of participants. This increased RTCP reporting interval
can cause premature timeouts if it is more than five times the can cause premature timeouts if it is more than five times the
interval chosen by the SSRCs that understand compound RTCP that interval chosen by the SSRCs that understand compound RTCP that
aggregate reports from many SSRCs. A 1500 octet MTU can fit five aggregate reports from many SSRCs. A 1500 octet MTU can fit five
typical size reports into a compound RTCP packet, so this is a real typical size reports into a compound RTCP packet, so this is a real
concern if endpoints aggregate RTCP reports from multiple SSRCs. concern if endpoints aggregate RTCP reports from multiple SSRCs.
The issue raised in the previous paragraph is mitigated by the The issue raised in the previous paragraph is mitigated by the
modification in timeout behaviour specified in Section 6.1.2. This modification in timeout behaviour specified in Section 7.1.2 of this
mitigation is in place in those cases where the RTCP bandwidth is memo. This mitigation is in place in those cases where the RTCP
sufficiently high that an endpoint, using an avg_rtcp_size calculated bandwidth is sufficiently high that an endpoint, using avg_rtcp_size
without taking into account the number of reporting SSRCs, can calculated without taking into account the number of reporting SSRCs,
transmit more frequently than approximately every 5 seconds. Note, can transmit more frequently than approximately every 5 seconds.
however, that the non-modified endpoint's RTCP reporting is still Note, however, that the non-modified endpoint's RTCP reporting is
negatively impacted even if the premature timeout of its SSRCs are still negatively impacted even if the premature timeout of its SSRCs
avoided. If compatibility with non-updated endpoints is a concern, are avoided. If compatibility with non-updated endpoints is a
the number of reports from different SSRCs aggregated into a single concern, the number of reports from different SSRCs aggregated into a
compound RTCP packet SHOULD either be limited to two reports, or single compound RTCP packet SHOULD either be limited to two reports,
aggregation ought not used at all. This will limit the non-updated or aggregation ought not used at all. This will limit the non-
endpoint's RTCP reporting interval to be no larger than twice the updated endpoint's RTCP reporting interval to be no larger than twice
RTCP reporting interval that would be chosen by an endpoint following the RTCP reporting interval that would be chosen by an endpoint
this specification. following this specification.
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 behaviour 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).
The below text modifies and extends the behavior defined in
Section 6.3 of [RFC3550], and in Section 3.5.3 of [RFC4585] if the
AVPF or SAVPF profile is used, regarding actions to take when
scheduling and sending an RTCP packet. It uses the variable names
tn, tp, tc, T and Td defined in Section 6.3 of [RFC3550]. The
variable T_rr_last is defined in [RFC4585].
Schedule all the endpoint's local SSRCs individually for transmission
using the regular calculation of tn for the profile being used. Each
time an SSRC's tn timer expires, do the regular reconsideration and,
if applicable, T_rr_int based suppression. If the result indicates
that an RTCP packet is to be sent and the transmission is a regular
RTCP packet:
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 the current time (tc).
2. Calculate how much space for RTCP packets would be needed to add 5.3.2. Scheduling RTCP when Aggregating Multiple SSRCs
that SSRC.
3. If the considered SSRC's RTCP Packets fit within the lower layer This section revises and extends the behaviour defined in Section 6.3
datagram's Maximum Transmission Unit, taking the necessary of [RFC3550], and in Section 3.5.3 of [RFC4585] if the RTP/AVPF
protocol headers and the space consumed by prior SSRCs into profile or the RTP/SAVPF profile is used, regarding actions to take
account, then add that SSRC's RTCP packets to the compound packet when scheduling and sending RTCP packets where multiple reporting
and go again to Step 1. SSRCs are aggregating their RTCP packets into the same compound RTCP
packet. These changes to the RTCP scheduling rules are needed to
maintain important RTCP timing properties, including the inter-packet
distribution, and the behaviour during flash joins and other changes
in session membership.
4. Otherwise, if the considered SSRC's RTCP Packets will not fit The variables tn, tp, tc, T, and Td used in the following are defined
within the compound packet, then transmit the generated compound in Section 6.3 of [RFC3550]. The variable T_rr_last is defined in
packet. [RFC4585].
5. Update the RTCP Parameters for each SSRC that has been included Each endpoint MUST schedule RTCP transmission independently for each
in the sent RTCP packet. The previous RTCP transmit time (tp) of its SSRCs using the regular calculation of tn for the RTP profile
value for each SSRC MUST be updated as follows: being used. Each time the timer tn expires for an SSRC, the endpoint
MUST perform RTCP timer reconsideration and, if applicable, T_rr_int
based suppression. If the result indicates that a compound RTCP
packet is to be sent by that SSRC, and the transmission is not an
early RTCP packet [RFC4585], then the endpoint SHOULD try to
aggregate RTCP packets of additional SSRCs that are scheduled in the
future into the compound RTCP packet before it is sent. The reason
to limit or not aggregate at due to backwards compatibility reasons
was discussed earlier in Section 5.3.1.
A. For the first SSRC set the transmission time (tt) to tc. Aggregation proceeds as follows. The endpoint selects the SSRC that
has the smallest tn value after the current time, tc, and prepares
the RTCP packets that SSRC would send if its timer tn expired at tc.
If those RTCP packets will fit into the compound RTCP packet that is
being generated, taking into account the path MTU and the previously
added RTCP packets, then they are added to the compound RTCP packet;
otherwise they are discarded. This process is repeated for each
SSRC, in order of increasing tn, until the compound RTCP packet is
full, or all SSRCs have been aggregated. At that point, the compound
RTCP packet is sent.
B. For any additional SSRC calculate the transmission time that When the compound RTCP packet is sent, the endpoint MUST update tp,
each of these SSRCs would have had it not been aggregated and tn, and T_rr_last (if applicable) for each SSRC that was included.
given the current existing session context. This value is These variables are updated as follows:
derived by taking this SSRC's tn value and performing
reconsideration and updating tn until tp + T <= tn, then set
tt = tn. If AVPF or SAVPF is being used, then T_rr_int based
suppression MUST NOT be used in this calcualtion.
C. Calculate average transmission time (tt_avg) using the tt of a. For the first SSRC that reported in the compound RTCP packet, set
all the SSRCs included in the packet. the effective transmission time, tt, of that SSRC to tc.
D. Now update tp for all the sent SSRCs to tt_avg. b. For each additional SSRC that reported in the compound RTCP
packet, calculate the transmission time that SSRC would have had
if it had not been aggregated into the compound RTCP packet.
This is derived by taking tn for that SSRC, then performing
reconsideration and updating tn until tp + T <= tn. Once this is
done, set the effective transmission time, tt, for that SSRC to
the calculated value of tn. If the RTP/AVPF profile or the RTP/
SAVPF profile is being used, then T_rr_int based suppression MUST
NOT be used in this calculation.
E. If AVPF or SAVPF profile is being used update T_rr_last to c. Calculate average effective transmission time, tt_avg, for the
tt_avg. compound RTCP packet based on the tt values for all SSRCs sent in
the compound RTCP packet. Set tp for each of the SSRCs sent in
the compound RTCP packet to tt_avg. If the RTP/AVPF profile or
the RTP/SAVPF profile is being used, set T_tt_last for each SSRC
sent in the compound RTCP packet to tt_avg.
6. For the sent SSRCs calculate new tn values based on the updated d. For each of the SSRCs sent in the compound RTCP packet, calculate
parameters and reschedule the timers. new tn values based on the updated parameters and the usual RTCP
timing rules, and reschedule the timers.
When using AVPF or SAVPF profile, when following the scheduling When using the RTP/AVPF profile or the RTP/SAVPF profile, the above
algorithm for regular transmission in Section 3.5.3 then the case of mechanism only attempts to aggregate RTCP packets when the compound
T_rr_interval == 0, as well as option 1, 2a and 2b for T_rr_interval RTCP packet to be sent is not an early RTCP packet, and hence the
!= 0, results in transmission of a regular RTCP packet that follows algorithm in Section 3.5.3 of [RFC4585] will control RTCP scheduling.
the above and updates the necessary variables. However, when the If T_rr_interval == 0, or if T_rr_interval != 0 and option 1, 2a, or
transmission is suppressed per 2c, then tp is updated to tc, as no 2b of the algorithm are chosen, then the above mechanism updates the
aggregation has taken place. necessary variables. However, if the transmission is suppressed per
option 2c of the algorithm, then tp is updated to tc as aggregation
has not taken place.
Reverse reconsideration needs to be performed as specified in RTP Reverse reconsideration MUST be performed following Section 6.3.4 of
[RFC3550]. It is important to note that under the above algorithm [RFC3550]. In some cases, this can lead to the value of tp after
when performing reconsideration, the value of tp can actually be reverse reconsideration being larger than tc. This is not a problem,
larger than tc. However, that still has the desired effect of and has the desired effect of proportionally pulling the tp value
proportionally pulling the tp value towards tc (as well as tn) as the towards tc (as well as tn) as the group size shrinks in direct
group size shrinks in direct proportion the reduced group size. proportion the reduced group size.
The above algorithm has been shown in simulations to maintain the The above algorithm has been shown in simulations to maintain the
inter-RTCP-packet transmission distribution for the SSRCs and consume inter-RTCP packet transmission time distribution for each SSRC, and
the same amount of bandwidth as non-aggregated packets in RTP to consume the same amount of bandwidth as non-aggregated RTCP
sessions. With this algorithm the actual transmission interval for packets. With this algorithm the actual transmission interval for an
any SSRC triggering an RTCP compound packet transmission is following SSRC triggering an RTCP compound packet transmission is following the
the regular transmission rules. The value tp is set to somewhere in regular transmission rules. The value tp is set to somewhere in the
the interval [0,1.5/1.21828*Td] ahead of tc. The actual value is interval [0,1.5/1.21828*Td] ahead of tc. The actual value is average
average of one instance of tc and the randomized transmission times of one instance of tc and the randomized transmission times of the
of the additional SSRCs, thus the lower range of the interval is more additional SSRCs, thus the lower range of the interval is more
probable. This setting is performed to compensate for the bias that probable. This compensates for the bias that is otherwise introduced
is otherwise introduced by picking the shortest tn value out of the N by picking the shortest tn value out of the N SSRCs included in
SSRCs included in aggregate. aggregate.
The algorithm also handles the cases where the number of SSRCs that The algorithm also handles the cases where the number of SSRCs that
can be included in an aggregated packet varies. An SSRC 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 previously was aggregated and fails to fit in a packet still has its
own transmission scheduled according to normal rules. Thus, it will own 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. The algorithm's behaviour under SSRC group size another aggregate. The algorithm's behaviour under SSRC group size
changes is as follows: changes is as follows:
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, Td grows in proportion to the number of new SSRCs
new SSRCs in the group. When reconsideration is done when the in the group. When reconsideration is performed due to expiry of
timer for the tn expires, that SSRC will reconsider the the tn timer, that SSRC will reconsider the transmission and with
transmission and with a certain probability reschedule the tn a certain probability reschedule the tn timer. This part of the
timer. This part of the reconsideration algorithm is only reconsideration algorithm is only impacted by the above algorithm
impacted by the above algorithm by having tp values that were in by having tp values that were in the future instead of set to the
the future instead of set to the time of the actual last time of the actual last transmission at the time of updating tp.
transmission at the time of updating tp.
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. The setting of the tp value forward in time related to the left. The setting of the tp value forward in time related to the
tc could be believed to have negative effect. However, the reason tc could be believed to have negative effect. However, the reason
for this setting is to compensate for bias caused by picking the for this setting is to compensate for bias caused by picking the
shortest tn out of the N aggregated. This bias remains over a shortest tn out of the N aggregated. This bias remains over a
reduction in the number of SSRCs. The reverse reconsideration reduction in the number of SSRCs. The reverse reconsideration
compensates the reduction independently of aggregation being used compensates the reduction independently of aggregation being used
or not. The negative effect that can occur on removing an SSRC is or not. The negative effect that can occur on removing an SSRC is
that the most favourable tn belonged to the removed SSRC. The that the most favourable tn belonged to the removed SSRC. The
impact of this is limited to delaying the transmission, in the impact of this is limited to delaying the transmission, in the
worst case, one reporting interval. worst case, one reporting interval.
In conclusion the investigations performed has found no significant In conclusion the investigations performed has found no significant
negative impact on the scheduling algorithm. negative impact on the scheduling algorithm.
5.4. Use of RTP/AVPF Feedback 5.4. Use of RTP/AVPF or RTP/SAVPF 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. The guidelines in
this section also apply to endpoints using the RTP/SAVPF profile.
5.4.1. Choice of SSRC for Feedback Packets 5.4.1. Choice of SSRC for Feedback Packets
When an RTP/AVPF endpoint has multiple SSRCs, it can choose what SSRC When an RTP/AVPF endpoint has multiple SSRCs, it can choose what SSRC
to use as the source for the RTCP feedback packets it sends. Several to use as the source for the RTCP feedback packets it sends. Several
factors can affect that choice: factors can affect that choice:
o RTCP feedback packets relating to a particular media type SHOULD o RTCP feedback packets relating to a particular media type SHOULD
be sent by an SSRC that receives that media type. For example, be sent by an SSRC that receives that media type. For example,
when audio and video are multiplexed onto a single RTP session, when audio and video are multiplexed onto a single RTP session,
endpoints will use their audio SSRC to send feedback on the audio endpoints will use their audio SSRC to send feedback on the audio
received from other participants. received from other participants.
o RTCP feedback packets and RTCP codec control messages that are o RTCP feedback packets and RTCP codec control messages that are
notifications or indications regarding RTP data processed by an notifications or indications regarding RTP data processed by an
endpoint MUST be sent from the SSRC used by that RTP data. This endpoint MUST be sent from the SSRC used for that RTP data. This
includes notifications that relate to a previously received includes notifications that relate to a previously received
request or command [RFC4585][RFC5104]. request or command [RFC4585][RFC5104].
o If separate SSRCs are used to send and receive media, then the o If separate SSRCs are used to send and receive media, then the
corresponding SSRC SHOULD be used for feedback, since they have corresponding SSRC SHOULD be used for feedback, since they have
differing RTCP bandwidth fractions. This can also affect the differing RTCP bandwidth fractions. This can also affect the
consideration if the SSRC can be used in immediate mode or not. consideration if the SSRC can be used in immediate mode or not.
o Some RTCP feedback packet types require consistency in the SSRC o Some RTCP feedback packet types require consistency in the SSRC
used. For example, if a TMMBR limitation [RFC5104] is set by an used. For example, if a TMMBR limitation [RFC5104] is set by an
skipping to change at page 12, line 40 skipping to change at page 12, line 40
When an RTCP feedback packet is sent as part of a compound RTCP When an RTCP feedback packet is sent as part of a compound RTCP
packet that aggregates reports from multiple SSRCs, there is no packet that aggregates reports from multiple SSRCs, there is no
requirement that the compound packet contains an SR or RR packet requirement that the compound packet contains an SR or RR packet
generated by the sender of the RTCP feedback packet. For reduced- generated by the sender of the RTCP feedback packet. For reduced-
size RTCP packets, aggregation of RTCP feedback packets from multiple size RTCP packets, aggregation of RTCP feedback packets from multiple
sources is not limited further than Section 4.2.2 of [RFC5506]. sources is not limited further than Section 4.2.2 of [RFC5506].
5.4.2. Scheduling an RTCP 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 MUST schedule that packet following the algorithm in Section 3.5
[RFC4585]. When following these rules the following clarifications of [RFC4585] modified as follows:
need to be taken into account:
o Whether a session is considered to be point-to-point or multiparty o To determine whether an RTP session is considered to be a point-
is not based on the number of SSRCs, but the number of endpoints to-point session or a multiparty session, an endpoint MUST count
one directly interacts with in the RTP session. This is the number of distinct RTCP SDES CNAME values used by the SSRCs
determined by counting the number of CNAMEs used by the SSRCs listed in the SSRC field of RTP data packets it receives and in
received. A RTP session MUST be considered multiparty if more the "SSRC of sender" field of RTCP SR, RR, RTPFB, or PSFB packets
than one CNAME is received, unless signalling explicitly indicates it receives. An RTP session is considered to be a multiparty
that the session is to be handled as point to point, or RTCP session if more than one CNAME is used by those SSRCs, unless
reporting groups [I-D.ietf-avtcore-rtp-multi-stream-optimisation] signalling indicates that the session is to be handled as point to
are used. If RTCP reporting groups are used, the classification point, or RTCP reporting groups
is solely based on whether the endpoint receives a single [I-D.ietf-avtcore-rtp-multi-stream-optimisation] are used. If
reporting group, indicating point to point, or if multiple RTCP reporting groups are used, an RTP session is considered to be
reporting groups are received (or a mixture of sources using and a point-to-point session if the endpoint receives only a single
sources not using reporting groups), which is classified as reporting group, and considered to be a multiparty session if
multiparty. Note that contributing sources (CSRCs) can be bound multiple reporting groups are received, or if a combination of
to any number of different CNAMEs and do not affect the reporting groups and SSRCs that are not part of a reporting group
determination of whether the session is multiparty. Similarly, are received. Endpoints MUST NOT determine whether an RTP session
SSRC/CSRC values that are only seen in the source field of an SDES is multiparty or point-to-point based on the type of connection
packet do not affect this determination. (unicast or multicast) used, or on the number of SSRCs received.
o Note that when checking if there is already a scheduled compound o When checking if there is already a scheduled compound RTCP packet
RTCP packet containing feedback messages (Step 2 in containing feedback messages (Step 2 in Section 3.5.2 of
Section 3.5.2), that check is done considering all local SSRCs. [RFC4585]), that check MUST be done considering all local SSRCs.
o If the SSRC is not allowed to send an early RTCP packet, then the o If an SSRC is not allowed to send an early RTCP packet, then the
feedback message MAY be queued for transmission as part of any feedback message MAY be queued for transmission as part of any
early or regular scheduled transmission that can occur within the early or regular scheduled transmission that can occur within the
maximum useful lifetime of the feedback message (T_max_fb_delay). maximum useful lifetime of the feedback message (T_max_fb_delay).
This modifies the behaviour in bullet 4a) in Section 3.5.2 of This modifies the behaviour in bullet 4a) in Section 3.5.2 of
[RFC4585]. [RFC4585].
The above rule for determining if a RTP session is to be considered The first bullet point above specifies a rule to determine if an RTP
point-to-point or multiparty is simple and straightforward and works session is to be considered a point-to-point session or a multiparty
in most cases. The goal with the above classification is to session. This rule is straightforward to implement, but is known to
determine if the resources associated with RTP and RTCP are shared incorrectly classify some sessions as multiparty sessions. The known
with only one peer or multiple other endpoints. This is significant problems are as follows:
as it affects the impact and the necessary processing and resource
consumption. Relying on only CNAME will result in classifying some
few situations where one might actually have only one peer as a
multiparty situation. The known situations are the following ones:
Endpoint with multiple synchronization contexts: An endpoint that is Endpoint with multiple synchronization contexts: An endpoint that is
part of a point-to-point session can have multiple synchronization part of a point-to-point session can have multiple synchronization
contexts, for example due to forwarding an external media source contexts, for example due to forwarding an external media source
into a interactive real-time conversation. In this case the into a interactive real-time conversation. In this case the
classification will consider the peer as two endpoints, while the classification will consider the peer as two endpoints, while the
actual RTP/RTCP transmission will be under the control of one actual RTP/RTCP transmission will be under the control of one
endpoint. endpoint.
Selective Forwarding Middlebox: The SFM as defined in Section 3.7 of Selective Forwarding Middlebox: The SFM as defined in Section 3.7 of
skipping to change at page 14, line 19 skipping to change at page 14, line 13
single reporting group. single reporting group.
The above rules will also classify some sessions where the endpoint The above rules will also classify some sessions where the endpoint
is connected to an RTP mixer as being point to point. For example is connected to an RTP mixer as being point to point. For example
the mixer could act as gateway to an Any Source Multicast based RTP the mixer could act as gateway to an Any Source Multicast based RTP
session for the discussed endpoint. However, this will in most cases session for the discussed endpoint. However, this will in most cases
be okay, as the RTP mixer provides separation between the two parts be okay, as the RTP mixer provides separation between the two parts
of the session. The responsibility falls on the mixer to act of the session. The responsibility falls on the mixer to act
accordingly in each domain. accordingly in each domain.
Note: The above usage of point-to-point or multiparty as classifiers Finally, we note that signalling mechanisms could be defined to
is actually misleading, but we maintain these labels to match what is override the rules when it would result in the wrong classification.
used in [RFC4585] as this ensures that the right algorithms are
applied.
To conclude we note that in some cases signalling can be used to 6. Adding and Removing SSRCs
override the rule when it would result in the wrong classification.
6. RTCP Considerations for Streams with Disparate Rates The set of SSRCs present in a single RTP session can vary over time
due to changes in the number of endpoints in the session, or due to
changes in the number or type of media streams being sent.
Every endpoint in an RTP session will have at least one SSRC that it
uses for RTCP reporting, and for sending media if desired. It can
also have additional SSRCs, for sending extra media streams or for
additional RTCP reporting. If the set of media streams being sent
changes, then the set of SSRCs being sent will change. Changes in
the media format or clock rate might also require changes in the set
of SSRCs used. An endpoint can also have more active SSRCs than it
has active RTP media streams, and send RTCP relating to SSRCs that
are not currently sending RTP data packets so that its peers are
aware of the SSRCs, and have the associated context (e.g., clock
synchronisation and an SDES CNAME) in place to be able to play out
media as soon as they becomes active.
In the following, we describe some considerations around adding and
removing RTP streams and their associated SSRCs.
6.1. Adding RTP Streams
When an endpoint joins an RTP session it can have zero, one, or more
RTP streams it will send, or that it is prepared to send. If it has
no RTP stream it plans to send, it still needs an SSRC that will be
used to send RTCP feedback. If it will send one or more RTP streams,
it will need the corresponding number of SSRC values. The SSRCs used
by an endpoint are made known to other endpoints in the RTP session
by sending RTP and RTCP packets. SSRCs can also be signalled using
non-RTP means (e.g., [RFC5576]). Unless restricted by signalling, an
endpoint can, at any time, send an additional RTP stream, identified
by a new SSRC (this might be associated with a signalling event, but
that is outside the scope of this memo). This makes the new SSRC
visible to the other endpoints in the session, since they share the
single SSRC space inherent in the definition of an RTP session.
An endpoint that has never sent an RTP stream will have an SSRC that
it uses for RTCP reporting. If that endpoint wants to start sending
an RTP stream, it is RECOMMENDED that it use its existing SSRC for
that stream, since otherwise the participant count in the RTP session
will be unnecessary increased, leading to a longer RTCP reporting
interval and larger RTCP reports due to cross reporting. If the
endpoint wants to start sending more than one RTP stream, it will
need to generate a new SSRC for the second and any subsequent RTP
streams.
An endpoint that has previously stopped sending an RTP stream, and
that wants to start sending a new RTP stream, cannot generally re-use
the existing SSRC, and often needs to generate a new SSRC, because an
SSRC cannot change media type (e.g., audio to video) or RTP timestamp
clock rate [RFC7160], and because the SSRC might be associated with a
particular semantic by the application (note: an RTP stream can pause
and restart using the same SSRC, provided RTCP is sent for that SSRC
during the pause; these rules only apply to new RTP streams reusing
an existing SSRC).
6.2. Removing RTP Streams
An SSRC is removed from an RTP session in one of two ways. When an
endpoint stops sending RTP and RTCP packets using an SSRC, then that
SSRC will eventually time out as described in Section 6.3.5 of
[RFC3550]. Alternatively, an SSRC can be explicitly removed from use
by sending an RTCP BYE packet as described in Section 6.3.7 of
[RFC3550]. It is RECOMMENDED that SSRCs are removed from use by
sending an RTCP BYE packet. Note that [RFC3550] requires that the
RTCP BYE SHOULD be the last RTP/RTCP packet sent in the RTP session
for an SSRC. If an endpoint needs to restart an RTP stream after
sending an RTCP BYE for its SSRC, it needs to generate a new SSRC
value for that stream.
The finality of sending RTCP BYE, means that endpoints needs to
consider if the ceasing of transmission of an RTP stream is temporary
or more permanent. Temporary suspension of media transmission using
a particular RTP stream (SSRC) needs to maintain that SSRC as an
active participant, by continuing RTCP transmission for it. That way
the media sending can be resume immediately, knowing that the context
is in place. Permanent transmission halting needs to send RTCP BYE
to allow the other participants to use the RTCP bandwidth resources
and clean up their state databases.
An endpoint that ceases transmission of all its RTP streams but
remains in the RTP session MUST maintain at least one SSRC that is to
be used for RTCP reporting and feedback (i.e., it cannot send a BYE
for all SSRCs, but needs to retain at least one active SSRC). As
some Feedback packets can be bound to media type there might be need
to maintain one SSRC per media type within an RTP session. An
alternative can be to create a new SSRC to use for RTCP reporting and
feedback. However, to avoid the perception that an endpoint drops
completely out of an RTP session such a new SSRC ought to be first
established before terminating all the existing SSRCs.
7. RTCP Considerations for Streams with Disparate Rates
An RTP session has a single set of parameters that configure the An RTP session has a single set of parameters that configure the
session bandwidth. These are the RTCP sender and receiver fractions session bandwidth. These are the RTCP sender and receiver fractions
(e.g., the SDP "b=RR:" and "b=RS:" lines), and the parameters of the (e.g., the SDP "b=RR:" and "b=RS:" lines [RFC3556]), and the
RTP/AVPF profile [RFC4585] (e.g., trr-int) if that profile (or its parameters of the RTP/AVPF profile [RFC4585] (e.g., trr-int) if that
secure extension, RTP/SAVPF [RFC5124]) is used. As a consequence, profile (or its secure extension, RTP/SAVPF [RFC5124]) is used. As a
the base RTCP reporting interval, before randomisation, will be the consequence, the base RTCP reporting interval, before randomisation,
same for every sending SSRC in an RTP session. Similarly, every will be the same for every sending SSRC in an RTP session.
receiving SSRC in an RTP session will have the same base reporting Similarly, every receiving SSRC in an RTP session will have the same
interval, although this can differ from the reporting interval chosen base reporting interval, although this can differ from the reporting
by sending SSRCs. This uniform RTCP reporting interval for all SSRCs interval chosen by sending SSRCs. This uniform RTCP reporting
can result in RTCP reports being sent more often, or too seldom, than interval for all SSRCs can result in RTCP reports being sent more
is considered desirable for a RTP stream. often, or too seldom, than is considered desirable for a RTP stream.
For example, consider a scenario when an audio flow sending at tens For example, consider a scenario when an audio flow sending at tens
of kilobits per second is multiplexed into an RTP session with a of kilobits per second is multiplexed into an RTP session with a
multi-megabit high quality video flow. If the session bandwidth is multi-megabit high quality video flow. If the session bandwidth is
configured based on the video sending rate, and the default RTCP configured based on the video sending rate, and the default RTCP
bandwidth fraction of 5% of the session bandwidth is used, it is bandwidth fraction of 5% of the session bandwidth is used, it is
likely that the RTCP bandwidth will exceed the audio sending rate. likely that the RTCP bandwidth will exceed the audio sending rate.
If the reduced minimum RTCP interval described in Section 6.2 of If the reduced minimum RTCP interval described in Section 6.2 of
[RFC3550] is then used in the session, as appropriate for video where [RFC3550] is then used in the session, as appropriate for video where
rapid feedback on damaged I-frames is wanted, the uniform reporting rapid feedback on damaged I-frames is wanted, the uniform reporting
skipping to change at page 15, line 4 skipping to change at page 16, line 37
of kilobits per second is multiplexed into an RTP session with a of kilobits per second is multiplexed into an RTP session with a
multi-megabit high quality video flow. If the session bandwidth is multi-megabit high quality video flow. If the session bandwidth is
configured based on the video sending rate, and the default RTCP configured based on the video sending rate, and the default RTCP
bandwidth fraction of 5% of the session bandwidth is used, it is bandwidth fraction of 5% of the session bandwidth is used, it is
likely that the RTCP bandwidth will exceed the audio sending rate. likely that the RTCP bandwidth will exceed the audio sending rate.
If the reduced minimum RTCP interval described in Section 6.2 of If the reduced minimum RTCP interval described in Section 6.2 of
[RFC3550] is then used in the session, as appropriate for video where [RFC3550] is then used in the session, as appropriate for video where
rapid feedback on damaged I-frames is wanted, the uniform reporting rapid feedback on damaged I-frames is wanted, the uniform reporting
interval for all senders could mean that audio sources are expected interval for all senders could mean that audio sources are expected
to send RTCP packets more often than they send audio data packets. to send RTCP packets more often than they send audio data packets.
This bandwidth mismatch can be reduced by careful tuning of the RTCP This bandwidth mismatch can be reduced by careful tuning of the RTCP
parameters, especially trr_int when the RTP/AVPF profile is used, parameters, especially trr_int when the RTP/AVPF profile is used, but
cannot be avoided entirely, as it is inherent in the design of the cannot be avoided entirely as it is inherent in the design of the
RTCP timing rules, and affects all RTP sessions that contain flows RTCP timing rules, and affects all RTP sessions that contain flows
with greatly mismatched bandwidth. with greatly mismatched bandwidth.
Different media rates or desired RTCP behaviours can also occur Different media rates or desired RTCP behaviours can also occur with
between SSRCs carrying the same media type. A common case in SSRCs carrying the same media type. A common case in multiparty
multiparty conferencing is when only one or two video source are conferencing is when a small number of video streams are shown in
shown in higher resolution, while the others are shown as small high resolution, while the others are shown as low resolution
thumbnails, with the choice of which is shown in high resolution thumbnails, with the choice of which is shown in high resolution
being voice activity controlled. Here the differences are both in being voice activity controlled. Here the differences are both in
actual media rate and in choices for what feedback messages might be actual media rate and in choices for what feedback messages might be
needed. Other examples of differences that can exist are due to the needed. Other examples of differences that can exist are due to the
intended usage of a media source. A media source carrying the video intended usage of a media source. A media source carrying the video
of the speaker in a conference is different from a document camera. of the speaker in a conference is different from a document camera.
Basic parameters that can differ in this case are frame-rate, Basic parameters that can differ in this case are frame-rate,
acceptable end-to-end delay, and the SNR fidelity of the image. acceptable end-to-end delay, and the SNR fidelity of the image.
These differences affect not only the needed bit-rates, but also These differences affect not only the needed bit-rates, but also
possible transmission behaviours, usable repair mechanisms, what possible transmission behaviours, usable repair mechanisms, what
feedback messages the control and repair requires, the transmission feedback messages the control and repair requires, the transmission
requirements on those feedback messages, and monitoring of the RTP requirements on those feedback messages, and monitoring of the RTP
stream delivery. stream delivery. Other similar scenarios can also exist.
Sending multiple media types in a single RTP session causes that Sending multiple media types in a single RTP session causes that
session to contain more SSRCs than if each media type was sent in a session to contain more SSRCs than if each media type was sent in a
separate RTP session. For example, if two participants each send an separate RTP session. For example, if two participants each send an
audio and a video flow in a single RTP session, that session will audio and a video flow in a single RTP session, that session will
comprise four SSRCs, but if separate RTP sessions had been used for comprise four SSRCs, but if separate RTP sessions had been used for
audio and video, each of those two RTP sessions would comprise only audio and video, each of those two RTP sessions would comprise only
two SSRCs. Sending multiple media streams in an RTP session hence two SSRCs. Sending multiple media streams in an RTP session hence
increases the amount of cross reporting between the SSRCs, as each increases the amount of cross reporting between the SSRCs, as each
SSRC reports on all other SSRCs in the session. This increases the SSRC reports on all other SSRCs in the session. This increases the
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From an RTCP perspective, therefore, it can be seen that there are From an RTCP perspective, therefore, it can be seen that there are
advantages to using separate RTP sessions for each media stream, advantages to using separate RTP sessions for each media stream,
rather than sending multiple media streams in a single RTP session. rather than sending multiple media streams in a single RTP session.
However, these are frequently offset by the need to reduce port use, However, these are frequently offset by the need to reduce port use,
to ease NAT/firewall traversal, achieved by combining media streams to ease NAT/firewall traversal, achieved by combining media streams
into a single RTP session. The following sections consider some of into a single RTP session. The following sections consider some of
the issues with using RTCP in sessions with multiple media streams in the issues with using RTCP in sessions with multiple media streams in
more detail. more detail.
6.1. Timing out SSRCs 7.1. Timing out SSRCs
Various issues have been identified with timing out SSRC values when Various issues have been identified with timing out SSRC values when
sending multiple media streams in an RTP session. sending multiple media streams in an RTP session.
6.1.1. Problems with RTP/AVPF the T_rr_interval Parameter 7.1.1. Problems with the RTP/AVPF T_rr_interval Parameter
The RTP/AVPF profile includes a method to prevent RTCP reports from The RTP/AVPF profile includes a method to prevent regular RTCP
being sent too often. This mechanism is described in Section 3.5.3 reports from being sent too often. This mechanism is described in
of [RFC4585], and is controlled by the T_rr_interval parameter. It Section 3.5.3 of [RFC4585], and is controlled by the T_rr_interval
works as follows. When a regular RTCP report is sent, a new random parameter. It works as follows. When a regular RTCP report is sent,
value, T_rr_current_interval, is generated, drawn evenly in the range a new random value, T_rr_current_interval, is generated, drawn evenly
0.5 to 1.5 times T_rr_interval. If a regular RTCP packet is to be in the range 0.5 to 1.5 times T_rr_interval. If a regular RTCP
sent earlier then T_rr_current_interval seconds after the previous packet is to be sent earlier then T_rr_current_interval seconds after
regular RTCP packet, and there are no feedback messages to be sent, the previous regular RTCP packet, and there are no feedback messages
then that regular RTCP packet is suppressed, and the next regular to be sent, then that regular RTCP packet is suppressed, and the next
RTCP packet is scheduled. The T_rr_current_interval is recalculated regular RTCP packet is scheduled. The T_rr_current_interval is
each time a regular RTCP packet is sent. The benefit of suppression recalculated each time a regular RTCP packet is sent. The benefit of
is that it avoids wasting bandwidth when there is nothing requiring suppression is that it avoids wasting bandwidth when there is nothing
frequent RTCP transmissions, but still allows utilization of the requiring frequent RTCP transmissions, but still allows utilization
configured bandwidth when feedback is needed. of the configured bandwidth when feedback is needed.
Unfortunately this suppression mechanism skews the distribution of Unfortunately this suppression mechanism skews the distribution of
the RTCP sending intervals compared to the regular RTCP reporting the RTCP sending intervals compared to the regular RTCP reporting
intervals. The standard RTCP timing rules, including reconsideration intervals. The standard RTCP timing rules, including reconsideration
and the compensation factor, result in the intervals between sending and the compensation factor, result in the intervals between sending
RTCP packets having a distribution that is skewed towards the upper RTCP packets having a distribution that is skewed towards the upper
end of the range [0.5/1.21828, 1.5/1.21828]*Td, where Td is the end of the range [0.5/1.21828, 1.5/1.21828]*Td, where Td is the
deterministic calculated RTCP reporting interval. With Td = 5s this deterministic calculated RTCP reporting interval. With Td = 5s this
distribution covers the range [2.052s, 6.156s]. In comparison, the distribution covers the range [2.052s, 6.156s]. In comparison, the
RTP/AVPF suppression rules act in an interval that is 0.5 to 1.5 RTP/AVPF suppression rules act in an interval that is 0.5 to 1.5
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same value. That is, when T_rr_interval is configured to match the same value. That is, when T_rr_interval is configured to match the
regular RTCP reporting interval. In this case, one might expect that regular RTCP reporting interval. In this case, one might expect that
regular RTCP packets are sent according to their usual schedule, but regular RTCP packets are sent according to their usual schedule, but
feedback packets can be sent early. However, the above-mentioned feedback packets can be sent early. However, the above-mentioned
issue results in the RTCP packets actually being sent in the range issue results in the RTCP packets actually being sent in the range
[0.5*Td, 2.731*Td] with a highly non-uniform distribution, rather [0.5*Td, 2.731*Td] with a highly non-uniform distribution, rather
than the range [0.41*Td, 1.23*Td]. This is perhaps unexpected, but than the range [0.41*Td, 1.23*Td]. This is perhaps unexpected, but
is not a problem in itself. However, when coupled with packet loss, is not a problem in itself. However, when coupled with packet loss,
it raises the issue of premature timeout. it raises the issue of premature timeout.
6.1.2. Avoiding Premature Timeout 7.1.2. Avoiding Premature Timeout
In RTP/AVP [RFC3550] the timeout behaviour 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 configured RTCP bandwidth allows for an average RTCP words, if the configured RTCP bandwidth allows for an average RTCP
reporting interval shorter than 5 seconds, the timeout is 25 seconds reporting interval shorter than 5 seconds, the timeout is 25 seconds
of no activity from the SSRC (RTP or RTCP), otherwise the timeout is of no activity from the SSRC (RTP or RTCP), otherwise the timeout is
5 average reporting intervals. 5 average reporting intervals.
RTP/AVPF [RFC4585] introduces different timeout behaviours depending RTP/AVPF [RFC4585] introduces different timeout behaviours depending
on the value of T_rr_interval. When T_rr_interval is 0, it uses the on the value of T_rr_interval. When T_rr_interval is 0, it uses the
same timeout calculation as RTP/AVP. However, when T_rr_interval is same timeout calculation as RTP/AVP. However, when T_rr_interval is
non-zero, it replaces Tmin in the timeout calculation, most likely to non-zero, it replaces Tmin in the timeout calculation, most likely to
speed up detection of timed out SSRCs. However, using a non-zero speed up detection of timed out SSRCs. However, using a non-zero
T_rr_interval has two consequences for RTP behaviour. T_rr_interval has two consequences for RTP behaviour.
First, due to suppression, the number of RTP and RTCP packets sent by First, due to suppression, the number of RTP and RTCP packets sent by
an SSRC that is not an active RTP sender can become very low, because an SSRC that is not an active RTP sender can become very low, because
of the issue discussed in Section 6.1.1. As the RTCP packet interval of the issue discussed in Section 7.1.1. 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
might in fact transmit only a single RTCP packet. The long intervals might in fact transmit only a single RTCP packet. The long intervals
result in fewer RTCP packets, to a point where a single RTCP packet result in fewer RTCP packets, to a point where a single RTCP packet
loss can sometimes result in timing out an SSRC. loss can sometimes result in timing out an SSRC.
Second, the RTP/AVPF changes to the timeout rules reduce robustness Second, the RTP/AVPF changes to the timeout rules reduce robustness
to misconfiguration. It is common to use RTP/AVPF configured such to misconfiguration. It is common to use RTP/AVPF configured such
that RTCP packets can be sent frequently, to allow rapid feedback, that RTCP packets can be sent frequently, to allow rapid feedback,
however this makes timeouts very sensitive to T_rr_interval. For however this makes timeouts very sensitive to T_rr_interval. For
example, if two SSRCs are configured one with T_rr_interval = 0.1s example, if two SSRCs are configured one with T_rr_interval = 0.1s
and the other with T_rr_interval = 0.6s, then this small difference and the other with T_rr_interval = 0.6s, then this small difference
will result in the SSRC with the shorter T_rr_interval timing out the will result in the SSRC with the shorter T_rr_interval timing out the
other if it stops sending RTP packets, since the other RTCP reporting other if it stops sending RTP packets, since the other RTCP reporting
interval is more than five times its own. When RTP/AVP is used, or interval is more than five times its own. When RTP/AVP is used, or
RTP/AVPF with T_rr_interval = 0, this is a non-issue, as the timeout RTP/AVPF with T_rr_interval = 0, this is a non-issue, as the timeout
period will be 25s, and differences between configured RTCP bandwidth period will be 25s, and differences between configured RTCP bandwidth
can only cause premature timeouts when the reporting intervals are can only cause premature timeouts when the reporting intervals are
greater than 5s and differ by a factor of five. To limit the scope greater than 5s and differ by a factor of five. To limit the scope
for such problematic misconfiguration, we propose an update to the for such problematic misconfiguration, we propose an update to the
RTP/AVPF timeout rules in Section 6.1.4. RTP/AVPF timeout rules in Section 7.1.4.
6.1.3. Interoperability Between RTP/AVP and RTP/AVPF 7.1.3. Interoperability Between RTP/AVP and RTP/AVPF
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 within a single RTP session, and the secure variants) are combined within a single RTP session, and the
RTP/AVPF endpoints use a non-zero T_rr_interval that is significantly RTP/AVPF endpoints use a non-zero T_rr_interval that is significantly
below 5 seconds, there is a risk that the RTP/AVPF endpoints will below 5 seconds, there is a risk that the RTP/AVPF endpoints will
prematurely timeout the SSRCs of the RTP/AVP endpoints, due to their prematurely timeout the SSRCs of the RTP/AVP endpoints, due to their
different RTCP timeout rules. Conversely, if the RTP/AVPF endpoints different RTCP timeout rules. Conversely, if the RTP/AVPF endpoints
use a T_rr_interval that is significant larger than 5 seconds, there use a T_rr_interval that is significant larger than 5 seconds, there
is a risk that the RTP/AVP endpoints will timeout the SSRCs of the is a risk that the RTP/AVP endpoints will timeout the SSRCs of the
RTP/AVPF endpoints. RTP/AVPF endpoints.
Mixing endpoints using two different RTP profiles within a single RTP Mixing endpoints using two different RTP profiles within a single RTP
session is NOT RECOMMENDED. However, if mixed RTP profiles are used, session is NOT RECOMMENDED. However, if mixed RTP profiles are used,
and the RTP/AVPF endpoints are not updated to follow Section 6.1.4 of and the RTP/AVPF endpoints are not updated to follow Section 7.1.4 of
this memo, then the RTP/AVPF session SHOULD be configured to use this memo, then the RTP/AVPF session SHOULD be configured to use
T_rr_interval = 4 seconds to avoid premature timeouts. T_rr_interval = 4 seconds to avoid premature timeouts.
The choice of T_rr_interval = 4 seconds for interoperability might The choice of T_rr_interval = 4 seconds for interoperability might
appear strange. Intuitively, this value ought to be 5 seconds, to appear strange. Intuitively, this value ought to be 5 seconds, to
make both the RTP/AVP and RTP/AVPF use the same timeout period. make both the RTP/AVP and RTP/AVPF use the same timeout period.
However, the behaviour outlined in Section 6.1.1 shows that actual However, the behaviour outlined in Section 7.1.1 shows that actual
RTP/AVPF reporting intervals can be longer than expected. Setting RTP/AVPF reporting intervals can be longer than expected. Setting
T_rr_interval = 4 seconds gives actual RTCP intervals near to those T_rr_interval = 4 seconds gives actual RTCP intervals near to those
expected by RTP/AVP, ensuring interoperability. expected by RTP/AVP, ensuring interoperability.
6.1.4. Updated SSRC Timeout Rules 7.1.4. Updated SSRC Timeout Rules
To ensure interoperability and avoid premature timeouts, all SSRCs in To ensure interoperability and avoid premature timeouts, all SSRCs in
an RTP session MUST use the same timeout behaviour. However, an RTP session MUST use the same timeout behaviour. However,
previous specification are inconsistent in this regard. To avoid previous specification are inconsistent in this regard. To avoid
interoperability issues, this memo updates the timeout rules as interoperability issues, this memo updates the timeout rules as
follows: follows:
o For the RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF profiles, the o For the RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF profiles, the
timeout interval SHALL be calculated using a multiplier of five timeout interval SHALL be calculated using a multiplier of five
times the deterministic RTCP reporting interval. That is, the times the deterministic RTCP reporting interval. That is, the
timeout interval SHALL be 5*Td. timeout interval SHALL be 5*Td.
o For the RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF profiles, o For the RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF profiles,
calculation of Td, for the purpose of calculating the participant calculation of Td, for the purpose of calculating the participant
timeout only, SHALL be done using a Tmin value of 5 seconds and timeout only, SHALL be done using a Tmin value of 5 seconds and
not the reduced minimal interval, even if the reduced minimum not the reduced minimal interval, even if the reduced minimum
interval is used to calculate RTCP packet transmission intervals. interval is used to calculate RTCP packet transmission intervals.
This changes the behaviour for the RTP/AVPF or RTP/SAVPF profiles This changes the behaviour for the RTP/AVPF or RTP/SAVPF profiles
when T_rr_interval != 0, a behaviour defined in Section 3.5.4 of RFC when T_rr_interval != 0. Specifically, the first paragraph of
4585, i.e. Tmin in the Td calculation is the T_rr_interval. Section 3.5.4 of [RFC4585] is updated to use Tmin instead of
T_rr_interval in the timeout calculation for RTP/AVPF entities.
6.2. Tuning RTCP transmissions 7.2. 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 possibilities exist for the RTP/AVP [RFC3551] considered what possibilities exist for the RTP/AVP [RFC3551]
profile, then what additional tools are provided by RTP/AVPF profile, then what additional tools are provided by RTP/AVPF
[RFC4585]. [RFC4585].
6.2.1. RTP/AVP and RTP/SAVP 7.2.1. RTP/AVP and RTP/SAVP
When using the RTP/AVP or RTP/SAVP profiles, the options for tuning When using the RTP/AVP or RTP/SAVP profiles, the options for tuning
the RTCP reporting intervals are limited to the RTCP sender and the RTCP reporting intervals are limited to the RTCP sender and
receiver bandwidth, and whether the minimum RTCP interval is scaled receiver bandwidth, and whether the minimum RTCP interval is scaled
according to the bandwidth. As the scheduling algorithm includes according to the bandwidth. As the scheduling algorithm includes
both randomisation and reconsideration, one cannot simply calculate both randomisation and reconsideration, one cannot simply calculate
the expected average transmission interval using the formula for Td the expected average transmission interval using the formula for Td
given in Section 6.3.1 of [RFC3550]. However, by considering the given in Section 6.3.1 of [RFC3550]. However, by considering the
inputs to that expression, and the randomisation and reconsideration inputs to that expression, and the randomisation and reconsideration
rules, we can begin to understand the behaviour of the RTCP rules, we can begin to understand the behaviour of the RTCP
skipping to change at page 20, line 34 skipping to change at page 22, line 20
all SSRCs are senders, and that they all send compound RTCP packets all SSRCs are senders, and that they all send compound RTCP packets
comprising an SR packet with n-1 report blocks, followed by an SDES comprising an SR packet with n-1 report blocks, followed by an SDES
packet containing a 16 octet CNAME value [RFC7022] (such RTCP packets packet containing a 16 octet CNAME value [RFC7022] (such RTCP packets
will vary in size between 54 and 798 octets depending on n, up to the will vary in size between 54 and 798 octets depending on n, up to the
maximum of 31 report blocks that can be included in an SR packet). maximum of 31 report blocks that can be included in an SR packet).
If we put this packet size, and a 5% RTCP bandwidth fraction into the If we put this packet size, and a 5% RTCP bandwidth fraction into the
RTCP interval calculation in Section 6.3.1 of [RFC3550], and RTCP interval calculation in Section 6.3.1 of [RFC3550], and
calculate the value of n needed to give Td = Tmin for the scaled calculate the value of n needed to give Td = Tmin for the scaled
minimum interval, we find n=9 SSRCs can be supported (irrespective of minimum interval, we find n=9 SSRCs can be supported (irrespective of
the interval, due to the way the reporting interval scales with the the interval, due to the way the reporting interval scales with the
session bandwidth). We see that to support more SSRCs, we need to session bandwidth). We see that to support more SSRCs without
increase the RTCP bandwidth fraction from 5%; changing the session changing the scaled minimum interval, we need to increase the RTCP
bandwidth does not help due to the limit of Tmin. bandwidth fraction from 5%; changing the session bandwidth to a
higher value would reduce the Tmin. However, if using the default 5%
allocation of RTCP bandwidth, an increase will result in more SSRCs
being supported given a fixed Td target.
To conclude, with RTP/AVP and RTP/SAVP the key limitation for small Based on the above, when using the RTP/AVP profile or the RTP/SAVP
unicast sessions is going to be the Tmin value. Thus the RTP session profile, the key limitation for rapid RTCP reporting in small unicast
bandwidth configured in RTCP has to be sufficiently high to reach the sessions is going to be the Tmin value. The RTP session bandwidth
reporting goals the application has following the rules for the configured in RTCP has to be sufficiently high to reach the reporting
scaled minimal RTCP interval. goals the application has following the rules for the scaled minimal
RTCP interval.
6.2.2. RTP/AVPF and RTP/SAVPF 7.2.2. RTP/AVPF and RTP/SAVPF
When using RTP/AVPF or RTP/SAVPF, we have a powerful additional tool When using RTP/AVPF or RTP/SAVPF, we have a powerful additional tool
for tuning RTCP transmissions: the T_rr_interval parameter. Use of for tuning RTCP transmissions: the T_rr_interval parameter. Use of
this parameter allows short RTCP reporting intervals; alternatively this parameter allows short RTCP reporting intervals; alternatively
it gives the ability to sent frequent RTCP feedback without sending it gives the ability to sent frequent RTCP feedback without sending
frequent regular RTCP reports. frequent regular RTCP reports.
The use of the RTP/AVPF or RTP/SAVPF profile with T_rr_interval set The use of the RTP/AVPF or RTP/SAVPF profile with T_rr_interval set
to a value greater than zero but smaller than Tmin allows more to a value greater than zero but smaller than Tmin allows more
frequent RTCP feedback than the RTP/AVP or RTP/SAVP profiles, for a frequent RTCP feedback than the RTP/AVP or RTP/SAVP profiles, for a
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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. Indeed, it might make this choice even for the RTCP bandwidth value. Indeed, it might make this choice even
more important, as this is more likely to affect the RTCP behaviour more important, as this is more likely to affect the RTCP behaviour
and performance than when using the RTP/AVP or RTP/SAVP profile, as and performance than when using the RTP/AVP or RTP/SAVP profile, as
there are fewer limitations affecting the RTCP transmission. there are fewer limitations affecting the RTCP transmission.
When T_rr_interval is non-zero, there are configurations that need to When T_rr_interval is non-zero, there are configurations that need to
be avoided. If the RTCP bandwidth chosen is such that the Td value be avoided. If the RTCP bandwidth chosen is such that the Td value
is smaller than, but close to, T_rr_interval, then the actual regular is smaller than, but close to, T_rr_interval, then the actual regular
RTCP packet transmission interval can become very large, as discussed RTCP packet transmission interval can become very large, as discussed
in Section 6.1.1. Therefore, for configuration where one intends to in Section 7.1.1. Therefore, for configuration where one intends to
have Td smaller than T_rr_interval, then Td is RECOMMENDED to be have Td smaller than T_rr_interval, then Td is RECOMMENDED to be
targeted at values less than 1/4th of T_rr_interval which results in targeted at values less than 1/4th of T_rr_interval which results in
that the range becomes [0.5*T_rr_interval, 1.81*T_rr_interval]. that the range becomes [0.5*T_rr_interval, 1.81*T_rr_interval].
With the RTP/AVPF or RTP/SAVPF profiles, using T_rr_interval = 0 has With the RTP/AVPF or RTP/SAVPF profiles, using T_rr_interval = 0 has
utility, and results in a behaviour where the RTCP transmission is utility, and results in a behaviour where the RTCP transmission is
only limited by the bandwidth, i.e., no Tmin limitations at all. only limited by the bandwidth, i.e., no Tmin limitations at all.
This allows more frequent regular RTCP reporting than can be achieved This allows more frequent regular RTCP reporting than can be achieved
using the RTP/AVP profile. Many configurations of RTCP will not using the RTP/AVP profile. Many configurations of RTCP will not
consume all the bandwidth that they have been configured to use, but consume all the bandwidth that they have been configured to use, but
this configuration will consume what it has been given. Note that this configuration will consume what it has been given. Note that
the same behaviour will be achieved as long as T_rr_interval is the same behaviour will be achieved as long as T_rr_interval is
smaller than 1/3 of Td as that prevents T_rr_interval from affecting smaller than 1/3 of Td as that prevents T_rr_interval from affecting
the transmission. the transmission.
There exists no method for using different regular RTCP reporting There exists no method for using different regular RTCP reporting
intervals depending on the media type or individual media stream, intervals depending on the media type or individual media stream,
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consume all the bandwidth that they have been configured to use, but consume all the bandwidth that they have been configured to use, but
this configuration will consume what it has been given. Note that this configuration will consume what it has been given. Note that
the same behaviour will be achieved as long as T_rr_interval is the same behaviour will be achieved as long as T_rr_interval is
smaller than 1/3 of Td as that prevents T_rr_interval from affecting smaller than 1/3 of Td as that prevents T_rr_interval from affecting
the transmission. the transmission.
There exists no method for using different regular RTCP reporting There exists no method for using different regular RTCP reporting
intervals depending on the media type or individual media stream, intervals depending on the media type or individual media stream,
other than using a separate RTP session for each type or stream. other than using a separate RTP session for each type or stream.
7. Security Considerations 8. Security Considerations
When using the secure RTP protocol (RTP/SAVP) [RFC3711], or the When using the secure RTP protocol (RTP/SAVP) [RFC3711], or the
secure variant of the feedback profile (RTP/SAVPF) [RFC5124], the secure variant of the feedback profile (RTP/SAVPF) [RFC5124], the
cryptographic context of a compound secure RTCP packet is the SSRC of cryptographic context of a compound secure RTCP packet is the SSRC of
the sender of the first RTCP (sub-)packet. This could matter in some the sender of the first RTCP (sub-)packet. This could matter in some
cases, especially for keying mechanisms such as Mikey [RFC3830] which cases, especially for keying mechanisms such as Mikey [RFC3830] which
allow use of per-SSRC keying. allow use of per-SSRC keying.
Otherwise, the standard security considerations of RTP apply; sending Otherwise, the standard security considerations of RTP apply; sending
multiple media streams from a single endpoint in a single RTP session multiple media streams from a single endpoint in a single RTP session
does not appear to have different security consequences than sending does not appear to have different security consequences than sending
the same number of media streams spread across different RTP the same number of media streams spread across different RTP
sessions. sessions.
8. IANA Considerations 9. IANA Considerations
No IANA actions are needed. No IANA actions are needed.
9. References 10. Acknowledgments
9.1. Normative References The authors like to thank Harald Alvestrand and everyone else who has
been involved in the development of this document.
11. References
11.1. 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.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)", Norrman, "The Secure Real-time Transport Protocol (SRTP)",
skipping to change at page 23, line 18 skipping to change at page 25, line 9
2006. 2006.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008. (RTP/SAVPF)", RFC 5124, February 2008.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009. and Consequences", RFC 5506, April 2009.
9.2. Informative References 11.2. 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-07 (work in ietf-avtcore-multi-media-rtp-session-07 (work in
progress), March 2015. progress), March 2015.
[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, 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-05 (work draft-ietf-avtcore-rtp-multi-stream-optimisation-06 (work
in progress), February 2015. in progress), July 2015.
[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-06 (work in progress), ietf-avtcore-rtp-topologies-update-10 (work in progress),
March 2015. July 2015.
[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-21 (work in progress), March 2015. framework-22 (work in progress), April 2015.
[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-18 (work in progress), March 2015. negotiation-22 (work in progress), June 2015.
[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.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC
3556, July 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.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile "Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, February 2008. with Feedback (AVPF)", RFC 5104, February 2008.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009.
[RFC6190] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis, [RFC6190] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
"RTP Payload Format for Scalable Video Coding", RFC 6190, "RTP Payload Format for Scalable Video Coding", RFC 6190,
May 2011. May 2011.
[RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla, [RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla,
"Guidelines for Choosing RTP Control Protocol (RTCP) "Guidelines for Choosing RTP Control Protocol (RTCP)
Canonical Names (CNAMEs)", RFC 7022, September 2013. Canonical Names (CNAMEs)", RFC 7022, September 2013.
[RFC7160] Petit-Huguenin, M. and G. Zorn, "Support for Multiple
Clock Rates in an RTP Session", RFC 7160, April 2014.
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
USA USA
Email: jonathan@vidyo.com Email: jonathan@vidyo.com
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