| Rfc | 8108 |
|---|
| Title | Sending Multiple RTP Streams in a Single RTP Session |
|---|
| Author | J. Lennox, M.
Westerlund, Q. Wu, C. Perkins |
|---|
| Date | March 2017 |
|---|
| Format: | TXT, HTML |
|---|
| Updates | RFC3550, RFC4585 |
|---|
| Status: | PROPOSED STANDARD |
|---|
|
Internet Engineering Task Force (IETF) J. Lennox
Request for Comments: 8108 Vidyo
Updates: 3550, 4585 M. Westerlund
Category: Standards Track Ericsson
ISSN: 2070-1721 Q. Wu
Huawei
C. Perkins
University of Glasgow
March 2017
Sending Multiple RTP Streams in a Single RTP Session
Abstract
This memo expands and clarifies the behavior of Real-time Transport
Protocol (RTP) endpoints that use multiple synchronization sources
(SSRCs). This occurs, for example, when an endpoint sends multiple
RTP streams in a single RTP session. This memo updates RFC 3550 with
regard to handling multiple SSRCs per endpoint in RTP sessions, with
a particular focus on RTP Control Protocol (RTCP) behavior. It also
updates RFC 4585 to change and clarify the calculation of the timeout
of SSRCs and the inclusion of feedback messages.
Status of This Memo
This is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc8108.
Copyright Notice
Copyright (c) 2017 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction ....................................................4
2. Terminology .....................................................4
3. Use Cases for Multi-Stream Endpoints ............................4
3.1. Endpoints with Multiple Capture Devices ....................4
3.2. Multiple Media Types in a Single RTP Session ...............5
3.3. Multiple Stream Mixers .....................................5
3.4. Multiple SSRCs for a Single Media Source ...................5
4. Use of RTP by Endpoints That Send Multiple Media Streams ........6
5. Use of RTCP by Endpoints That Send Multiple Media Streams .......6
5.1. RTCP Reporting Requirement .................................7
5.2. Initial Reporting Interval .................................7
5.3. Aggregation of Reports into Compound RTCP Packets ..........8
5.3.1. Maintaining AVG_RTCP_SIZE ...........................9
5.3.2. Scheduling RTCP when Aggregating Multiple SSRCs ....10
5.4. Use of RTP/AVPF or RTP/SAVPF Feedback .....................13
5.4.1. Choice of SSRC for Feedback Packets ................13
5.4.2. Scheduling an RTCP Feedback Packet .................14
6. Adding and Removing SSRCs ......................................15
6.1. Adding RTP Streams ........................................16
6.2. Removing RTP Streams ......................................16
7. RTCP Considerations for Streams with Disparate Rates ...........17
7.1. Timing Out SSRCs ..........................................19
7.1.1. Problems with the RTP/AVPF T_rr_interval
Parameter ..........................................19
7.1.2. Avoiding Premature Timeout .........................20
7.1.3. Interoperability between RTP/AVP and RTP/AVPF ......21
7.1.4. Updated SSRC Timeout Rules .........................22
7.2. Tuning RTCP Transmissions .................................22
7.2.1. RTP/AVP and RTP/SAVP ...............................22
7.2.2. RTP/AVPF and RTP/SAVPF .............................24
8. Security Considerations ........................................25
9. References .....................................................26
9.1. Normative References ......................................26
9.2. Informative References ....................................26
Acknowledgments ...................................................29
Authors' Addresses ................................................29
1. Introduction
At the time the Real-Time Transport Protocol (RTP) [RFC3550] was
originally designed, and for quite some time after, endpoints in RTP
sessions typically only transmitted a single media source and, thus,
used a single RTP stream and synchronization source (SSRC) per RTP
session, where separate RTP sessions were typically used for each
distinct media type. Recently, however, a number of scenarios have
emerged in which endpoints wish to send multiple RTP streams,
distinguished by distinct RTP synchronization source (SSRC)
identifiers, in a single RTP session. These are outlined in
Section 3. Although the initial design of RTP did consider such
scenarios, the specification was not consistently written with such
use cases in mind; thus, the specification is somewhat unclear in
places.
This memo updates [RFC3550] to clarify behavior in use cases where
endpoints use multiple SSRCs. It also updates [RFC4585] to resolve
problems with regard to timeout of inactive SSRCs and to clarify
behavior around inclusion of feedback messages.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in RFC
2119 [RFC2119] and indicate requirement levels for compliant
implementations.
3. Use Cases for Multi-Stream Endpoints
This section discusses several use cases that have motivated the
development of endpoints that sends RTP data using multiple SSRCs in
a single RTP session.
3.1. Endpoints with Multiple Capture Devices
The most straightforward motivation for an endpoint to send multiple
simultaneous RTP streams in a single RTP session is when an endpoint
has multiple capture devices and, hence, can generate multiple media
sources, of the same media type and characteristics. For example,
telepresence systems of the type described by the CLUE Telepresence
Framework [CLUE-FRAME] often have multiple cameras or 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
Recent work has updated RTP [MULTI-RTP] and Session Description
Protocol (SDP) [SDP-BUNDLE] to remove the historical assumption in
RTP that media sources of different media types would always be sent
on different RTP sessions. In this work, a single endpoint's audio
and video RTP streams (for example) are instead sent in a single RTP
session to reduce the number of transport-layer flows used.
3.3. Multiple Stream Mixers
There are several RTP topologies that can involve a central device
that itself generates multiple RTP streams in a session. An example
is a mixer providing centralized compositing for a multi-capture
scenario like that described in Section 3.1. In this case, the
centralized node is behaving much like a multi-capturer endpoint,
generating several similar and related sources.
A more complex example is the selective forwarding middlebox,
described in Section 3.7 of [RFC7667]. This is a middlebox that
receives RTP streams from several endpoints and then selectively
forwards modified versions of some RTP streams toward the other
endpoints to which it is connected. For each connected endpoint, a
separate media source appears in the session for every other source
connected to the middlebox, "projected" from the original streams,
but at any given time many of them can appear to be inactive (and
thus are receivers, not senders, in RTP). This sort of device is
closer to being an RTP mixer than an RTP translator: it terminates
RTCP reporting about the mixed streams; it can rewrite SSRCs,
timestamps, and sequence numbers, as well as the contents of the RTP
payloads; and it can turn sources on and off at will without
appearing to generate packet loss. Each projected stream will
typically preserve its original RTCP source description (SDES)
information.
3.4. Multiple SSRCs for a Single Media Source
There are also several cases where multiple SSRCs can be used to send
data from a single media source within a single RTP session. These
include, but are not limited to, transport robustness tools, such as
the RTP retransmission payload format [RFC4588], that require one
SSRC to be used for the media data and another SSRC for the repair
data. Similarly, some layered media encoding schemes, for example,
H.264 Scalable Video Coding (SVC) [RFC6190], can be used in a
configuration where each layer is sent using a different SSRC within
a single RTP session.
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 signaled, RTP endpoints can expect to receive RTP
data packets sent using 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
session bandwidth, as determined by signaling and congestion control.
The endpoint needs to keep its total media sending rate within this
share. However, endpoints that send multiple RTP streams do not
necessarily need to subdivide their share of the available bandwidth
independently or uniformly to each RTP stream and its SSRCs. In
particular, an endpoint can vary the bandwidth allocation to
different streams depending on their needs, and it can dynamically
change the bandwidth allocated to different SSRCs (for example, by
using a variable-rate codec), provided the total sending rate does
not exceed its allocated share. This includes enabling or disabling
RTP streams, or their redundancy streams, as more or less bandwidth
becomes available.
5. Use of RTCP by Endpoints That Send Multiple Media Streams
RTCP is defined in Section 6 of [RFC3550]. The description of the
protocol is phrased in terms of the behavior of "participants" in an
RTP session, under the assumption that each endpoint is a participant
with a single SSRC. However, for correct operation in cases where
endpoints have multiple SSRC values, implementations MUST treat each
SSRC as a separate participant in the RTP session, so that an
endpoint that has multiple SSRCs counts as multiple participants.
5.1. RTCP Reporting Requirement
An RTP endpoint that has multiple SSRCs MUST treat each SSRC as a
separate participant in the RTP session. Each SSRC will maintain its
own RTCP-related state information and, hence, will have its own RTCP
reporting interval that determines when it sends RTCP reports. If
the mechanism in [MULTI-STREAM-OPT] is 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
are only receivers, then they will receive different shares of the
RTCP bandwidth and calculate different base RTCP reporting intervals.
Otherwise, all SSRCs at an endpoint will calculate the same base RTCP
reporting interval. The actual reporting intervals for each SSRC are
randomized in the usual way, but reports can be aggregated as
described in Section 5.3.
5.2. Initial Reporting Interval
When a participant joins a unicast session, the following text from
Section 6.2 of [RFC3550] is relevant: "For unicast sessions... the
delay before sending the initial compound RTCP packet MAY be zero."
The basic assumption is that this also ought to apply in the case of
multiple SSRCs. Caution has to be exercised, however, when an
endpoint (or middlebox) with a large number of SSRCs joins a unicast
session, since immediate transmission of many RTCP reports can create
a significant burst of traffic, leading to transient congestion and
packet loss due to queue overflows.
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
RTP endpoint MUST NOT send more than four compound RTCP packets with
zero initial delay when it joins an RTP session, independent of the
number of SSRCs used by the endpoint. Each of those initial compound
RTCP packets MAY include aggregated reports from multiple SSRCs,
provided the total compound RTCP packet size does not exceed the MTU,
and the avg_rtcp_size is maintained as in Section 5.3.1. Aggregating
reports from several SSRCs in the initial compound RTCP packets
allows a substantial number of SSRCs to report immediately.
Endpoints SHOULD prioritize reports on SSRCs that are likely to be
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
four compound RTCP reports that can be sent immediately MUST send the
other reports later, following the usual RTCP timing rules including
timer reconsideration. Those reports MAY be aggregated as described
in Section 5.3.
Note: The above is chosen to match the TCP maximum initial window
of four packets [RFC3390], not the larger TCP initial windows for
which there is an ongoing experiment [RFC6928]. The reason for
this is a desire to be conservative, since an RTP endpoint will
also in many cases start sending RTP data packets at the same time
as these initial RTCP packets are sent.
5.3. Aggregation of Reports into Compound RTCP Packets
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
RTCP reports. This will lead to each SSRC sending a compound RTCP
packet in each reporting interval. Since these packets are coming
from the same endpoint, it might reasonably be expected that they can
be aggregated to reduce overheads. Indeed, Section 6.1 of [RFC3550]
allows RTP translators and mixers to aggregate packets in similar
circumstances:
It is RECOMMENDED that translators and mixers combine individual
RTCP packets from the multiple sources they are forwarding into
one compound packet whenever feasible in order to amortize the
packet overhead (see Section 7). An example RTCP compound packet
as might be produced by a mixer is shown in Fig. 1. If the
overall length of a compound packet would exceed the MTU of the
network path, it SHOULD be segmented into multiple shorter
compound packets to be transmitted in separate packets of the
underlying protocol. This does not impair the RTCP bandwidth
estimation because each compound packet represents at least one
distinct participant. Note that each of the compound packets MUST
begin with an SR or RR packet.
This allows RTP translators and mixers to generate compound RTCP
packets that contain multiple Sender Report (SR) or Receiver Report
(RR) packets from different SSRCs, 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 packet, except the regular rule
that the compound RTCP packet starts with an SR or RR packet. Due to
this rule, correctly implemented RTP endpoints will be able to handle
compound RTCP packets that contain RTCP packets relating to multiple
SSRCs.
Accordingly, endpoints that use multiple SSRCs can aggregate the RTCP
packets sent by their different SSRCs into compound RTCP packets,
provided 1) the resulting compound RTCP packets begin with an SR or
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
aggregation as described in Section 5.3.2.
5.3.1. Maintaining AVG_RTCP_SIZE
The RTCP scheduling algorithm in [RFC3550] works on a per-SSRC basis.
Each SSRC sends a single compound RTCP packet in each RTCP reporting
interval. When an endpoint uses multiple SSRCs, it is desirable to
aggregate the compound RTCP packets sent by its SSRCs, reducing the
overhead by forming a larger compound RTCP packet. This aggregation
can be done as described in Section 5.3.2, provided the average RTCP
packet size calculation is updated as follows.
Participants in an RTP session update their estimate of the average
RTCP packet size (avg_rtcp_size) each time they send or receive an
RTCP packet (see Section 6.3.3 of [RFC3550]). When a compound RTCP
packet that contains RTCP packets from several SSRCs is sent or
received, the avg_rtcp_size estimate for each SSRC that is reported
upon is updated using div_packet_size rather than the actual packet
size:
avg_rtcp_size = (1/16) * div_packet_size + (15/16) * avg_rtcp_size
where div_packet_size is packet_size divided by the number of SSRCs
reporting in that compound packet. The number of SSRCs reporting in
a compound packet is determined by counting the number of different
SSRCs that are the source of SR or RR RTCP packets within the
compound RTCP packet. Non-compound RTCP packets (i.e., RTCP packets
that do not contain an SR or RR packet [RFC5506]) are considered to
report on a single SSRC.
A participant that doesn't follow the above rule, and instead uses
the full RTCP compound packet size to calculate avg_rtcp_size, will
derive an RTCP reporting interval that is overly large by a factor
that is proportional to the number of SSRCs aggregated into compound
RTCP packets and the size of set of SSRCs being aggregated relative
to the total number of participants. This increased RTCP reporting
interval can cause premature timeouts if it is more than five times
the interval chosen by the SSRCs that understand compound RTCP that
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
concern if endpoints aggregate RTCP reports from multiple SSRCs.
The issue raised in the previous paragraph is mitigated by the
modification in timeout behavior specified in Section 7.1.2 of this
memo. This mitigation is in place in those cases where the RTCP
bandwidth is sufficiently high that an endpoint, using avg_rtcp_size
calculated without taking into account the number of reporting SSRCs,
can transmit more frequently than approximately every 5 seconds.
Note, however, that the non-updated endpoint's RTCP reporting is
still negatively impacted even if the premature timeouts of its SSRCs
are avoided. If compatibility with non-updated endpoints is a
concern, the number of reports from different SSRCs aggregated into a
single compound RTCP packet SHOULD either be limited to two reports
or aggregation ought not be used at all. This will limit the
non-updated endpoint's RTCP reporting interval to be no larger than
twice the RTCP reporting interval that would be chosen by an endpoint
following this specification.
5.3.2. Scheduling RTCP when Aggregating Multiple SSRCs
This section revises and extends the behavior defined in Section 6.3
of [RFC3550], and in Section 3.5.3 of [RFC4585] if the RTP/AVPF
profile or the RTP/SAVPF profile is used, regarding actions to take
when scheduling and sending RTCP packets where multiple reporting
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 behavior during flash joins and other changes
in session membership.
The variables tn, tp, tc, T, and Td used in the following are defined
in Section 6.3 of [RFC3550]. The variables T_rr_interval and
T_rr_last are defined in [RFC4585].
Each endpoint MUST schedule RTCP transmission independently for each
of its SSRCs using the regular calculation of tn for the RTP profile
being used. Each time the timer tn expires for an SSRC, the endpoint
MUST perform RTCP timer reconsideration and, if applicable,
suppression based on T_rr_interval. 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 due to backwards compatibility
reasons is discussed in Section 5.3.1.
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.
When the compound RTCP packet is sent, the endpoint MUST update tp,
tn, and T_rr_last (if applicable) for each SSRC that was included.
These variables are updated as follows:
a. For the first SSRC that reported in the compound RTCP packet, set
the effective transmission time, tt, of that SSRC to tc.
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 suppression based on
T_rr_interval MUST NOT be used in this calculation.
c. Calculate average effective transmission time, tt_avg, for the
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.
d. For each of the SSRCs sent in the compound RTCP packet, calculate
new tn values based on the updated parameters and the usual RTCP
timing rules and reschedule the timers.
When using the RTP/AVPF profile or the RTP/SAVPF profile, the above
mechanism only attempts to aggregate RTCP packets when the compound
RTCP packet to be sent is not an early RTCP packet, and hence the
algorithm in Section 3.5.3 of [RFC4585] will control RTCP scheduling.
If T_rr_interval == 0, or if T_rr_interval != 0 and option 1, 2a, or
2b of the algorithm are chosen, then the above mechanism updates the
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 MUST be performed following Section 6.3.4 of
[RFC3550]. In some cases, this can lead to the value of tp after
reverse reconsideration being larger than tc. This is not a problem,
and has the desired effect of proportionally pulling the tp value
towards tc (as well as tn) as the reporting interval shrinks in
direct proportion the reduced group size.
The above algorithm has been shown in simulations [Sim88] [Sim92] to
maintain the inter-RTCP packet transmission time distribution for
each SSRC and to consume the same amount of bandwidth as
non-aggregated RTCP packets. With this algorithm, the actual
transmission interval for an SSRC triggering an RTCP compound packet
transmission is following the regular transmission rules. The value
tp is set to somewhere in the interval [0, 1.5/1.21828*Td] ahead of
tc. The actual value is the average of one instance of tc and the
randomized transmission times of the additional SSRCs; thus, the
lower range of the interval is more probable. This compensates for
the bias that is otherwise introduced by picking the shortest tn
value out of the N SSRCs included in aggregate.
The algorithm also handles the cases where the number of SSRCs that
can be included in an aggregated packet varies. An SSRC that
previously was aggregated and fails to fit in a packet still has its
own transmission scheduled according to normal rules. Thus, it will
trigger a transmission in due time, or the SSRC will be included in
another aggregate. The algorithm's behavior under SSRC group size
changes is as follows:
RTP sessions where the number of SSRCs is growing: When the group
size is growing, Td grows in proportion to the number of new SSRCs
in the group. When reconsideration is performed due to expiry of
the tn timer, that SSRC will reconsider the transmission and with
a certain probability reschedule the tn timer. This part of the
reconsideration algorithm is only impacted by the above algorithm
having tp values that were in the future instead of set to the
time of the actual last transmission at the time of updating tp.
RTP sessions where the number of SSRCs is shrinking: When the group
shrinks, reverse reconsideration moves the tp and tn values
towards tc proportionally to the number of SSRCs that leave the
session compared to the total number of participants when they
left. The setting of the tp value forward in time related to the
tc could be believed to have negative effect. However, the reason
for this setting is to compensate for bias caused by picking the
shortest tn out of the N aggregated. This bias remains over a
reduction in the number of SSRCs. The reverse reconsideration
compensates the reduction independently of whether or not
aggregation is being used. The negative effect that can occur on
removing an SSRC is that the most favorable tn belonged to the
removed SSRC. The impact of this is limited to delaying the
transmission, in the worst case, one reporting interval.
In conclusion, the investigations performed have found no significant
negative impact on the scheduling algorithm.
5.4. Use of RTP/AVPF or RTP/SAVPF Feedback
This section discusses the transmission of RTP/AVPF feedback packets
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
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
factors can affect that choice:
o RTCP feedback packets relating to a particular media type SHOULD
be sent by an SSRC that receives that media type. For example,
when audio and video are multiplexed onto a single RTP session,
endpoints will use their audio SSRC to send feedback on the audio
received from other participants.
o RTCP feedback packets and RTCP codec control messages that are
notifications or indications regarding RTP data processed by an
endpoint MUST be sent from the SSRC used for that RTP data. This
includes notifications that relate to a previously received
request or command [RFC4585][RFC5104].
o If separate SSRCs are used to send and receive media, then the
corresponding SSRC SHOULD be used for feedback, since they have
differing RTCP bandwidth fractions. This can also affect the
consideration of whether or not the SSRC can be used in immediate
mode.
o Some RTCP feedback packet types require consistency in the SSRC
used. For example, if a Temporary Maximum Media Stream Bit Rate
Request (TMMBR) limitation [RFC5104] is set by an SSRC, the same
SSRC needs to be used to remove the limitation.
o If several SSRCs are suitable for sending feedback, it might be
desirable to use an SSRC that allows the sending of feedback as an
early RTCP packet.
When an RTCP feedback packet is sent as part of a compound RTCP
packet that aggregates reports from multiple SSRCs, there is no
requirement that the compound packet contain an SR or RR packet
generated by the sender of the RTCP feedback packet. For reduced-
size RTCP packets, aggregation of RTCP feedback packets from multiple
sources is not limited further than Section 4.2.2 of [RFC5506].
5.4.2. Scheduling an RTCP Feedback Packet
When an SSRC has a need to transmit a feedback packet in early mode,
it MUST schedule that packet following the algorithm in Section 3.5
of [RFC4585] modified as follows:
o To determine whether an RTP session is considered to be a point-
to-point session or a multiparty session, an endpoint MUST count
the number of distinct RTCP SDES CNAME values used by the SSRCs
listed in the SSRC field of RTP data packets it receives and in
the "SSRC of sender" field of RTCP SR, RR, RTPFB, or PSFB packets
it receives. An RTP session is considered to be a multiparty
session if more than one CNAME is used by those SSRCs, unless
signaling indicates that the session is to be handled as point to
point or RTCP reporting groups [MULTI-STREAM-OPT] are used. If
RTCP reporting groups are used, an RTP session is considered to be
a point-to-point session if the endpoint receives only a single
reporting group and is considered to be a multiparty session if
multiple reporting groups are received or a combination of
reporting groups and SSRCs that are not part of a reporting group
are received. Endpoints MUST NOT determine whether an RTP session
is multiparty or point to point based on the type of connection
(unicast or multicast) used, or on the number of SSRCs received.
o When checking if there is already a scheduled compound RTCP packet
containing feedback messages (Step 2 in Section 3.5.2 of
[RFC4585]), that check MUST be done considering all local SSRCs.
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
early or regular scheduled transmission that can occur within the
maximum useful lifetime of the feedback message (T_max_fb_delay).
This modifies the behavior in item 4a in Section 3.5.2 of
[RFC4585].
The first bullet point above specifies a rule to determine if an RTP
session is to be considered a point-to-point session or a multiparty
session. This rule is straightforward to implement, but is known to
incorrectly classify some sessions as multiparty sessions. The known
problems are as follows:
Endpoint with multiple synchronization contexts: An endpoint that is
part of a point-to-point session can have multiple synchronization
contexts, for example, due to forwarding an external media source
into an interactive real-time conversation. In this case, the
classification will consider the peer as two endpoints, while the
actual RTP/RTCP transmission will be under the control of one
endpoint.
Selective Forwarding Middlebox: The Selective Forwarding Middlebox
(SFM) as defined in Section 3.7 of [RFC7667] has control over the
transmission and configurations between itself and each peer
endpoint individually. It also fully controls the RTCP packets
being forwarded between the individual legs. Thus, this type of
middlebox can be compared to the RTP mixer, which uses its own
SSRCs to mix or select the media it forwards, that will be
classified as a point-to-point RTP session by the above rule.
In the above cases, it is very reasonable to use RTCP reporting
groups [MULTI-STREAM-OPT]. If that extension is used, an endpoint
can indicate that the multitude of CNAMEs are in fact under a single
endpoint or middlebox control by using only a single reporting group.
The above rules will also classify some sessions where the endpoint
is connected to an RTP mixer as being point to point. For example,
the mixer could act as gateway to an RTP session based on Any Source
Multicast for the discussed endpoint. However, this will, in most
cases, be okay, as the RTP mixer provides separation between the two
parts of the session. The responsibility falls on the mixer to act
accordingly in each domain.
Finally, we note that signaling mechanisms could be defined to
override the rules when they would result in the wrong
classification.
6. Adding and Removing SSRCs
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 RTP 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 sources or for
additional RTCP reporting. If the set of media sources 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 SSRCs than it has
active RTP 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 synchronization
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 signaled using
non-RTP means (e.g., [RFC5576]). Unless restricted by signaling, an
endpoint can, at any time, send an additional RTP stream, identified
by a new SSRC (this might be associated with a signaling 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 unnecessarily 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 reuse
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 be 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 need to
consider if the ceasing of transmission of an RTP stream is temporary
or 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 resumed immediately, knowing that the context is
in place. When permanently halting transmission, a participant needs
to send an 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 a
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
established first -- 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
session bandwidth. These are the RTCP sender and receiver fractions
(e.g., the SDP "b=RR:" and "b=RS:" lines [RFC3556]) and the
parameters of the RTP/AVPF profile [RFC4585] (e.g., trr-int) if that
profile (or its secure extension, RTP/SAVPF [RFC5124]) is used. As a
consequence, the base RTCP reporting interval, before randomization,
will be the same for every sending SSRC in an RTP session.
Similarly, every receiving SSRC in an RTP session will have the same
base reporting interval, although this can differ from the reporting
interval chosen by sending SSRCs. This uniform RTCP reporting
interval for all SSRCs can result in RTCP reports being sent more
often, or too seldom, than is considered desirable for an RTP stream.
For example, consider a scenario in which an audio flow sending at
tens of kilobits per second is multiplexed into an RTP session with a
multi-megabit high-quality video flow. If the session bandwidth is
configured based on the video sending rate, and the default RTCP
bandwidth fraction of 5% of the session bandwidth is used, it is
likely that the RTCP bandwidth will exceed the audio sending rate.
If the reduced minimum RTCP interval described in Section 6.2 of
[RFC3550] is then used in the session, as appropriate for video where
rapid feedback on damaged I-frames is wanted, the uniform reporting
interval for all senders could mean that audio sources are expected
to send RTCP packets more often than they send audio data packets.
This bandwidth mismatch can be reduced by careful tuning of the RTCP
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
RTCP timing rules, and affects all RTP sessions that contain flows
with greatly mismatched bandwidth.
Different media rates or desired RTCP behaviors can also occur with
SSRCs carrying the same media type. A common case in multiparty
conferencing is when a small number of video streams are shown in
high resolution, while the others are shown as low-resolution
thumbnails, with the choice of which is shown in high resolution
being voice-activity controlled. Here the differences are both in
actual media rate and in choices for what feedback messages might be
needed. Other examples of differences that can exist are due to the
intended usage of a media source. A media source carrying the video
of the speaker in a conference is different from a document camera.
Basic parameters that can differ in this case are frame-rate,
acceptable end-to-end delay, and the Signal-to-Noise Ratio (SNR)
fidelity of the image. These differences affect not only the needed
bitrates, but also possible transmission behaviors, usable repair
mechanisms, what feedback messages the control and repair requires,
the transmission requirements on those feedback messages, and
monitoring of the RTP stream delivery. Other similar scenarios can
also exist.
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
separate RTP session. For example, if two participants each send an
audio and a video RTP stream in a single RTP session, that session
will comprise four SSRCs; but if separate RTP sessions had been used
for audio and video, each of those two RTP sessions would comprise
only two SSRCs. Hence, sending multiple RTP streams in an RTP
session increases the amount of cross reporting between the SSRCs, as
each SSRC reports on all other SSRCs in the session. This increases
the size of the RTCP reports, causing them to be sent less often than
would be the case if separate RTP sessions where used for a given
RTCP bandwidth.
Finally, when an RTP session contains multiple media types, it is
important to note that the RTCP reception quality reports, feedback
messages, and extended report blocks used might not be applicable to
all media types. Endpoints will need to consider the media type of
each SSRC, and only send or process reports and feedback that apply
to that particular SSRC and its media type. Signaling solutions
might have shortcomings when it comes to indicating that a particular
set of RTCP reports or feedback messages only apply to a particular
media type within an RTP session.
From an RTCP perspective, therefore, it can be seen that there are
advantages to using separate RTP sessions for each media source,
rather than sending multiple media sources in a single RTP session.
However, these are frequently offset by the need to reduce port use,
to ease NAT/firewall traversal, achieved by combining media sources
into a single RTP session. The following sections consider some of
the issues with using RTCP in sessions with multiple media sources in
more detail.
7.1. Timing Out SSRCs
Various issues have been identified with timing out SSRC values when
sending multiple RTP streams in an RTP session.
7.1.1. Problems with the RTP/AVPF T_rr_interval Parameter
The RTP/AVPF profile includes a method to prevent regular RTCP
reports from being sent too often. This mechanism is described in
Section 3.5.3 of [RFC4585]; it is controlled by the T_rr_interval
parameter. It works as follows. When a regular RTCP report is sent,
a new random value, T_rr_current_interval, is generated, drawn evenly
in the range 0.5 to 1.5 times T_rr_interval. If a regular RTCP
packet is to be sent earlier than T_rr_current_interval seconds after
the previous regular RTCP packet, and there are no feedback messages
to be sent, then that regular RTCP packet is suppressed and the next
regular RTCP packet is scheduled. The T_rr_current_interval is
recalculated each time a regular RTCP packet is sent. The benefit of
suppression is that it avoids wasting bandwidth when there is nothing
requiring frequent RTCP transmissions, but still allows utilization
of the configured bandwidth when feedback is needed.
Unfortunately, this suppression mechanism skews the distribution of
the RTCP sending intervals compared to the regular RTCP reporting
intervals. The standard RTCP timing rules, including reconsideration
and the compensation factor, result in the intervals between sending
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
deterministic calculated RTCP reporting interval. With Td = 5 s,
this distribution covers the range [2.052 s, 6.156 s]. In
comparison, the RTP/AVPF suppression rules act in an interval that is
0.5 to 1.5 times T_rr_interval; for T_rr_interval = 5s, this is
[2.5 s, 7.5 s].
The effect of this is that the time between consecutive RTCP packets
when using T_rr_interval suppression can become large. The maximum
time interval between sending one regular RTCP packet and the next,
when T_rr_interval is being used, occurs when T_rr_current_interval
takes its maximum value and a regular RTCP packet is suppressed at
the end of the suppression period, then the next regular RTCP packet
is scheduled after its largest possible reporting interval. Taking
the worst case of the two intervals gives a maximum time between two
RTCP reports of 1.5*T_rr_interval + 1.5/1.21828*Td.
This behavior can be surprising when Td and T_rr_interval have 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 packets are sent according to their usual schedule, but
feedback packets can be sent early. However, the above-mentioned
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
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,
it raises the issue of premature timeout.
7.1.2. Avoiding Premature Timeout
In RTP/AVP [RFC3550] the timeout behavior is simple; it is 5 times
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
reporting interval shorter than 5 seconds, the timeout is 25 seconds
of no activity from the SSRC (RTP or RTCP); otherwise, the timeout is
5 average reporting intervals.
RTP/AVPF [RFC4585] introduces different timeout behaviors depending
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
non-zero, it replaces Tmin in the timeout calculation, most likely to
speed up detection of timed out SSRCs. However, using a non-zero
T_rr_interval has two consequences for RTP behavior.
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
of the issue discussed in Section 7.1.1. As the RTCP packet interval
can be as long as 2.73*Td, during a 5*Td time period, an endpoint
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
loss can sometimes result in timing out an SSRC.
Second, the RTP/AVPF changes to the timeout rules reduce robustness
to misconfiguration. It is common to use RTP/AVPF configured such
that RTCP packets can be sent frequently to allow rapid feedback;
however, this makes timeouts very sensitive to T_rr_interval. For
example, if two SSRCs are configured, one with T_rr_interval = 0.1 s
and the other with T_rr_interval = 0.6 s, then this small difference
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
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
period will be 25 s, and differences between configured RTCP
bandwidth can only cause premature timeouts when the reporting
intervals are greater than 5 s and differ by a factor of five. To
limit the scope for such problematic misconfiguration, we define an
update to the RTP/AVPF timeout rules in Section 7.1.4.
7.1.3. Interoperability between RTP/AVP and RTP/AVPF
If endpoints implementing the RTP/AVP and RTP/AVPF profiles (or their
secure variants) are combined within a single RTP session, and the
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
prematurely time out the SSRCs of the RTP/AVP endpoints, due to their
different RTCP timeout rules. Conversely, if the RTP/AVPF endpoints
use a T_rr_interval that is significantly larger than 5 seconds,
there is a risk that the RTP/AVP endpoints will time out the SSRCs of
the RTP/AVPF endpoints.
Mixing endpoints using two different RTP profiles within a single RTP
session is NOT RECOMMENDED. However, if mixed RTP profiles are used,
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
T_rr_interval = 4 seconds to avoid premature timeouts.
The choice of T_rr_interval = 4 seconds for interoperability might
appear strange. Intuitively, this value ought to be 5 seconds, to
make both the RTP/AVP and RTP/AVPF use the same timeout period.
However, the behavior outlined in Section 7.1.1 shows that actual
RTP/AVPF reporting intervals can be longer than expected. Setting
T_rr_interval = 4 seconds gives actual RTCP intervals near to those
expected by RTP/AVP, ensuring interoperability.
7.1.4. Updated SSRC Timeout Rules
To ensure interoperability and avoid premature timeouts, all SSRCs in
an RTP session MUST use the same timeout behavior. However, previous
specifications are inconsistent in this regard. To avoid
interoperability issues, this memo updates the timeout rules as
follows:
o For the RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF profiles, the
timeout interval SHALL be calculated using a multiplier of five
times the deterministic RTCP reporting interval. That is, the
timeout interval SHALL be 5*Td.
o For the RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF profiles,
calculation of Td, for the purpose of calculating the participant
timeout only, SHALL be done using a Tmin value of 5 seconds and
not the reduced minimal interval, even if the reduced minimum
interval is used to calculate RTCP packet transmission intervals.
This changes the behavior for the RTP/AVPF or RTP/SAVPF profiles when
T_rr_interval != 0. Specifically, the first paragraph of
Section 3.5.4 of [RFC4585] is updated to use Tmin instead of
T_rr_interval in the timeout calculation for RTP/AVPF entities.
7.2. Tuning RTCP Transmissions
This subsection discusses what tuning can be done to reduce the
downsides of the shared RTCP packet intervals. First, what
possibilities exist for the RTP/AVP [RFC3551] profile are listed
followed by what additional tools are provided by RTP/AVPF [RFC4585].
7.2.1. RTP/AVP and RTP/SAVP
When using the RTP/AVP or RTP/SAVP profiles, the options for tuning
the RTCP reporting intervals are limited to the RTCP sender and
receiver bandwidth, and whether the minimum RTCP interval is scaled
according to the bandwidth. As the scheduling algorithm includes
both randomization and reconsideration, one cannot simply calculate
the expected average transmission interval using the formula for Td
given in Section 6.3.1 of [RFC3550]. However, by considering the
inputs to that expression, and the randomization and reconsideration
rules, we can begin to understand the behavior of the RTCP
transmission interval.
Let's start with some basic observations:
a. Unless the scaled minimum RTCP interval is used, Td prior to
randomization and reconsideration can never be less than Tmin.
The default value of Tmin is 5 seconds.
b. If the scaled minimum RTCP interval is used, Td can become as low
as 360 divided by RTP Session bandwidth in kilobits per second.
In SDP, the RTP session bandwidth is signaled using a "b=AS"
line. An RTP Session bandwidth of 72 kbps results in Tmin being
5 seconds. An RTP session bandwidth of 360 kbps of course gives
a Tmin of 1 second, and to achieve a Tmin equal to once every
frame for a 25 frame-per-second video stream requires an RTP
session bandwidth of 9 Mbps. Use of the RTP/AVPF or RTP/SAVPF
profile allows more frequent RTCP reports for the same bandwidth,
as discussed below.
c. The value of Td scales with the number of SSRCs and the average
size of the RTCP reports to keep the overall RTCP bandwidth
constant.
d. The actual transmission interval for a Td value is in the range
[0.5*Td/1.21828, 1.5*Td/1.21828], and the distribution is skewed,
due to reconsideration, with the majority of the probability mass
being above Td. This means, for example, that for Td = 5 s, the
actual transmission interval will be distributed in the range
[2.052 s, 6.156 s], and tending towards the upper half of the
interval. Note that Tmin parameter limits the value of Td before
randomization and reconsideration are applied, so the actual
transmission interval will cover a range extending below Tmin.
Given the above, we can calculate the number of SSRCs, n, that an RTP
session with 5% of the session bandwidth assigned to RTCP can support
while maintaining Td equal to Tmin. This will tell us how many RTP
streams we can report on, keeping the RTCP overhead within acceptable
bounds. We make two assumptions that simplify the calculation: that
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
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
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
RTCP interval calculation in Section 6.3.1 of [RFC3550], and
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
the interval, due to the way the reporting interval scales with the
session bandwidth). We see that to support more SSRCs without
changing the scaled minimum interval, we need to increase the RTCP
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.
Based on the above, when using the RTP/AVP profile or the RTP/SAVP
profile, the key limitation for rapid RTCP reporting in small unicast
sessions is going to be the Tmin value. The RTP session bandwidth
configured in RTCP has to be sufficiently high to reach the reporting
goals the application has following the rules for the scaled minimal
RTCP interval.
7.2.2. RTP/AVPF and RTP/SAVPF
When using RTP/AVPF or RTP/SAVPF, we have a powerful additional tool
for tuning RTCP transmissions: the T_rr_interval parameter. Use of
this parameter allows short RTCP reporting intervals; alternatively
it gives the ability to sent frequent RTCP feedback without sending
frequent regular RTCP reports.
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
frequent RTCP feedback than the RTP/AVP or RTP/SAVP profiles, for a
given RTCP bandwidth. This happens because Tmin is set to zero after
the transmission of the initial RTCP report, causing the reporting
interval for later packet to be determined by the usual RTCP
bandwidth-based calculation, with Tmin=0, and the T_rr_interval.
This has the effect that we are no longer restricted by the minimal
interval (whether the default 5-second minimum or the reduced minimum
interval). Rather, the RTCP bandwidth and the T_rr_interval are the
governing factors, allowing faster feedback. Applications that care
about rapid regular RTCP feedback ought to consider using the RTP/
AVPF or RTP/SAVPF profile, even if they don't use the feedback
features of that profile.
The use of the RTP/AVPF or RTP/SAVPF profile allows RTCP feedback
packets to be sent frequently, without also requiring regular RTCP
reports to be sent frequently, since T_rr_interval limits the rate at
which regular RTCP packets can be sent, while still permitting RTCP
feedback packets to be sent. Applications that can use feedback
packets for some RTP streams, e.g., video streams, but don't want
frequent regular reporting for other RTP streams, can configure the
T_rr_interval to a value so that the regular reporting for both audio
and video is at a level that is considered acceptable for the audio.
They could then use feedback packets, which will include RTCP SR/RR
packets unless reduced size RTCP feedback packets [RFC5506] are used,
for the video reporting. This allows the available RTCP bandwidth to
be devoted on the feedback that provides the most utility for the
application.
Using T_rr_interval still requires one to determine suitable values
for the RTCP bandwidth value. Indeed, it might make this choice even
more important, as this is more likely to affect the RTCP behavior
and performance than when using the RTP/AVP or RTP/SAVP profile, as
there are fewer limitations affecting the RTCP transmission.
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
is smaller than, but close to, T_rr_interval, then the actual regular
RTCP packet transmission interval can become very large, as discussed
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
targeted at values less than 1/4th of T_rr_interval, which results in
the range becoming [0.5*T_rr_interval, 1.81*T_rr_interval].
With the RTP/AVPF or RTP/SAVPF profiles, using T_rr_interval = 0 has
utility and results in a behavior where the RTCP transmission is only
limited by the bandwidth, i.e., no Tmin limitations at all. This
allows more frequent regular RTCP reporting than can be achieved
using the RTP/AVP profile. Many configurations of RTCP will not
consume all the bandwidth that they have been configured to use, but
this configuration will consume what it has been given. Note that
the same behavior will be achieved as long as T_rr_interval is
smaller than 1/3 of Td as that prevents T_rr_interval from affecting
the transmission.
There exists no method for using different regular RTCP reporting
intervals depending on the media type or individual RTP stream, other
than using a separate RTP session for each type or stream.
8. Security Considerations
When using the secure RTP protocol (RTP/SAVP) [RFC3711], or the
secure variant of the feedback profile (RTP/SAVPF) [RFC5124], the
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
cases, especially for keying mechanisms such as MIKEY [RFC3830] that
allow use of per-SSRC keying.
Otherwise, the standard security considerations of RTP apply; sending
multiple RTP streams from a single endpoint in a single RTP session
does not appear to have different security consequences than sending
the same number of RTP streams spread across different RTP sessions.
9. References
9.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, DOI 10.17487/RFC3711, March 2004,
<http://www.rfc-editor.org/info/rfc3711>.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
DOI 10.17487/RFC4585, July 2006,
<http://www.rfc-editor.org/info/rfc4585>.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
2008, <http://www.rfc-editor.org/info/rfc5124>.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
2009, <http://www.rfc-editor.org/info/rfc5506>.
9.2. Informative References
[CLUE-FRAME]
Duckworth, M., Ed., Pepperell, A., and S. Wenger,
"Framework for Telepresence Multi-Streams", Work in
Progress, draft-ietf-clue-framework-25, January 2016.
[MULTI-RTP]
Westerlund, M., Perkins, C., and J. Lennox, "Sending
Multiple Types of Media in a Single RTP Session", Work in
Progress, draft-ietf-avtcore-multi-media-rtp-session-13,
December 2015.
[MULTI-STREAM-OPT]
Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session:
Grouping RTCP Reception Statistics and Other Feedback",
Work in Progress, draft-ietf-avtcore-rtp-multi-
stream-optimisation-12, March 2016.
[RFC3390] Allman, M., Floyd, S., and C. Partridge, "Increasing TCP's
Initial Window", RFC 3390, DOI 10.17487/RFC3390, October
2002, <http://www.rfc-editor.org/info/rfc3390>.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
DOI 10.17487/RFC3551, July 2003,
<http://www.rfc-editor.org/info/rfc3551>.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth",
RFC 3556, DOI 10.17487/RFC3556, July 2003,
<http://www.rfc-editor.org/info/rfc3556>.
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
DOI 10.17487/RFC3830, August 2004,
<http://www.rfc-editor.org/info/rfc3830>.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
DOI 10.17487/RFC4588, July 2006,
<http://www.rfc-editor.org/info/rfc4588>.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
February 2008, <http://www.rfc-editor.org/info/rfc5104>.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,
<http://www.rfc-editor.org/info/rfc5576>.
[RFC6190] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
"RTP Payload Format for Scalable Video Coding", RFC 6190,
DOI 10.17487/RFC6190, May 2011,
<http://www.rfc-editor.org/info/rfc6190>.
[RFC6928] Chu, J., Dukkipati, N., Cheng, Y., and M. Mathis,
"Increasing TCP's Initial Window", RFC 6928,
DOI 10.17487/RFC6928, April 2013,
<http://www.rfc-editor.org/info/rfc6928>.
[RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla,
"Guidelines for Choosing RTP Control Protocol (RTCP)
Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022,
September 2013, <http://www.rfc-editor.org/info/rfc7022>.
[RFC7160] Petit-Huguenin, M. and G. Zorn, Ed., "Support for Multiple
Clock Rates in an RTP Session", RFC 7160,
DOI 10.17487/RFC7160, April 2014,
<http://www.rfc-editor.org/info/rfc7160>.
[RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
DOI 10.17487/RFC7667, November 2015,
<http://www.rfc-editor.org/info/rfc7667>.
[SDP-BUNDLE]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", Work in Progress,
draft-ietf-mmusic-sdp-bundle-negotiation-36, October 2016.
[Sim88] Westerlund, M., "SIMULATION RESULTS FOR MULTI-STREAM",
IETF 88 Proceedings, November 2013,
<https://www.ietf.org/proceedings/88/slides/
slides-88-avtcore-0.pdf>.
[Sim92] Westerlund, M., Lennox, J., Perkins, C., and Q. Wu,
"Changes in RTP Multi-stream", IETF 92 Proceedings, March
2015, <https://www.ietf.org/proceedings/92/slides/
slides-92-avtcore-0.pdf>.
Acknowledgments
The authors like to thank Harald Alvestrand and everyone else who has
been involved in the development of this document.
Authors' Addresses
Jonathan Lennox
Vidyo, Inc.
433 Hackensack Avenue
Seventh Floor
Hackensack, NJ 07601
United States of America
Email: jonathan@vidyo.com
Magnus Westerlund
Ericsson
Farogatan 2
SE-164 80 Kista
Sweden
Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com
Qin Wu
Huawei
101 Software Avenue, Yuhua District
Nanjing, Jiangsu 210012
China
Email: bill.wu@huawei.com
Colin Perkins
University of Glasgow
School of Computing Science
Glasgow G12 8QQ
United Kingdom
Email: csp@csperkins.org