Self-Clocked Rate Adaptation for Multimedia
draft-johansson-ccwg-rfc8298bis-screamv2-00
| Document | Type | Active Internet-Draft (individual) | |
|---|---|---|---|
| Authors | Ingemar Johansson , Magnus Westerlund | ||
| Last updated | 2024-03-04 | ||
| RFC stream | (None) | ||
| Intended RFC status | (None) | ||
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| Consensus boilerplate | Unknown | ||
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draft-johansson-ccwg-rfc8298bis-screamv2-00
CCWG I. Johansson
Internet-Draft M. Westerlund
Obsoletes: 8298 (if approved) Ericsson
Intended status: Experimental 4 March 2024
Expires: 5 September 2024
Self-Clocked Rate Adaptation for Multimedia
draft-johansson-ccwg-rfc8298bis-screamv2-00
Abstract
This memo describes a rate adaptation algorithm for conversational
media services such as interactive video. The solution conforms to
the packet conservation principle and is a hybrid loss- and delay
based congestion control/rate management algorithm that also supports
ECN and L4S. The algorithm is evaluated over both simulated Internet
bottleneck scenarios as well as in a LongTerm Evolution (LTE) and 5G
system simulator and is shown to achieve both low latency and high
video throughput in these scenarios. This specification obsoletes
RFC 8298. The algorithm supports handling of multiple media streams,
typical use cases are streaming for remote control, AR and 3D VR
googles.
About This Document
This note is to be removed before publishing as an RFC.
Status information for this document may be found at
https://datatracker.ietf.org/doc/draft-johansson-ccwg-rfc8298bis-
screamv2/.
Discussion of this document takes place on the Congestion Control
Working Group (ccwg) Working Group mailing list
(mailto:ccwg@ietf.org), which is archived at
https://mailarchive.ietf.org/arch/browse/ccwg/. Subscribe at
https://www.ietf.org/mailman/listinfo/ccwg/.
Source for this draft and an issue tracker can be found at
https://github.com/gloinul/draft-johansson-ccwg-scream-bis.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. Wireless (LTE and 5G) Access Properties . . . . . . . . . 4
1.2. Why is it a self-clocked algorithm? . . . . . . . . . . . 5
1.3. Comparison with LEDBAT and TFWC in TCP . . . . . . . . . 6
2. Requirements Language . . . . . . . . . . . . . . . . . . . . 7
3. Overview of SCReAMv2 Algorithm . . . . . . . . . . . . . . . 7
3.1. Network Congestion Control . . . . . . . . . . . . . . . 8
3.2. Sender Transmission Control . . . . . . . . . . . . . . . 9
3.3. Media Rate Control . . . . . . . . . . . . . . . . . . . 9
4. Detailed Description of SCReAMv2 sender algorithm . . . . . . 9
4.1. Constants and variables . . . . . . . . . . . . . . . . . 11
4.1.1. Constants . . . . . . . . . . . . . . . . . . . . . . 11
4.1.2. State Variables . . . . . . . . . . . . . . . . . . . 12
4.2. Network Congestion Control . . . . . . . . . . . . . . . 14
4.2.1. Reaction to Delay, Packet Loss and ECN-CE . . . . . . 16
4.2.2. Congestion Window Update . . . . . . . . . . . . . . 18
4.2.3. Lost Packet Detection . . . . . . . . . . . . . . . . 22
4.2.4. Send Window Calculation . . . . . . . . . . . . . . . 22
4.2.5. Packet Pacing . . . . . . . . . . . . . . . . . . . . 23
4.2.6. Stream Prioritization . . . . . . . . . . . . . . . . 24
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4.3. Media Rate Control . . . . . . . . . . . . . . . . . . . 24
4.4. Competing Flows Compensation . . . . . . . . . . . . . . 26
4.5. Handling of systematic errors in video coders . . . . . . 28
5. Receiver Requirements on Feedback Intensity . . . . . . . . . 29
6. Discussion . . . . . . . . . . . . . . . . . . . . . . . . . 30
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 31
8. Security Considerations . . . . . . . . . . . . . . . . . . . 31
9. References . . . . . . . . . . . . . . . . . . . . . . . . . 31
9.1. Normative References . . . . . . . . . . . . . . . . . . 31
9.2. Informative References . . . . . . . . . . . . . . . . . 32
Appendix A. Acknowledgments . . . . . . . . . . . . . . . . . . 34
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 34
1. Introduction
Congestion in the Internet occurs when the transmitted bitrate is
higher than the available capacity over a given transmission path.
Applications that are deployed in the Internet have to employ
congestion control to achieve robust performance and to avoid
congestion collapse in the Internet.
Interactive real-time communication imposes a lot of requirements on
the transport; therefore, a robust, efficient rate adaptation for all
access types is an important part of interactive real-time
communications, as the transmission channel bandwidth can vary over
time.
Wireless access such as LTE and 5G, which is an integral part of the
current Internet, increases the importance of rate adaptation as the
channel bandwidth of a default LTE bearer [QoS-3GPP] can change
considerably in a very short time frame. Thus, a rate adaptation
solution for interactive real-time media, such as WebRTC [RFC7478],
should be both quick and be able to operate over a large range in
channel capacity.
This memo describes Self-Clocked Rate Adaptation for Multimedia
version 2 (SCReAMv2), an update to SCReAM congestion control for RTP
streams [RFC3550]. While SCReAM was originally devised for WebRTC,
SCReAM as well as SCReAMv2 can also be used for other applications
where congestion control of RTP streams is necessary. SCReAM is
based on the self-clocking principle of TCP and uses techniques
similar to what is used in the rate adaptation algorithm based on Low
Extra Delay Background Transport (LEDBAT) [RFC6817].
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SCReAMv2 is not entirely self-clocked as it augments self-clocking
with pacing and a minimum send rate. Further, SCReAMv2 can take
advantage of Explicit Congestion Notification (ECN) and Low Latency
Low Loss and Scalable throughput (L4S) in cases where ECN or L4S is
supported by the network and the hosts. However, ECN or L4S is not
required for the basic congestion control functionality in SCReAMv2.
This specification replaces the previous experimental version
[RFC8298] of SCReAM with SCReAMv2. There are many and fairly
significant changes to the original SCReAM algorithm.
The algorithm in this memo differs greatly against the previous
version of SCReAM. The main differences are:
* L4S support added. The L4S algoritm has many similarities with
the DCTCP and Prague congestion control but has a few extra
modifications to make it work well with peridic sources such as
video.
* The delay based congestion control is changed to implement a
pseudo-L4S approach, this simplifies the delay based congestion
control.
* The fast increase mode is removed. The congestion window additive
increase is replaced with an adaptive multiplicative increase to
enhance convergence speed.
* The algorithm is more rate based than self-clocked. The
calculated congestion window is used mainly to calculate proper
media bitrates. Bytes in flight is however allowed to exceeed the
congestion window.
* The media bitrate calculation is dramatically changed and
simplified.
* Additional compensation is added to make SCReAMv2 handle cases
such as large changing frame sizes.
1.1. Wireless (LTE and 5G) Access Properties
[RFC8869] describes the complications that can be observed in
wireless environments. Wireless access such as LTE and 5G typically
cannot guarantee a given bandwidth; this is true especially for
default bearers. The network throughput can vary considerably, for
instance, in cases where the wireless terminal is moving around.
Even though 5G can support bitrates well above 100 Mbps, there are
cases when the available bitrate can be much lower; examples are
situations with high network load and poor coverage. An additional
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complication is that the network throughput can drop for short time
intervals (e.g., at handover); these short glitches are initially
very difficult to distinguish from more permanent reductions in
throughput.
Unlike wireline bottlenecks with large statistical multiplexing, it
is not possible to try to maintain a given bitrate when congestion is
detected with the hope that other flows will yield. This is because
there are generally few other flows competing for the same
bottleneck. Each user gets its own variable throughput bottleneck,
where the throughput depends on factors like channel quality, network
load, and historical throughput. The bottom line is, if the
throughput drops, the sender has no other option than to reduce the
bitrate. Once the radio scheduler has reduced the resource
allocation for a bearer, a flow in that bearer aims to reduce the
sending rate quite quickly (within one RTT) in order to avoid
excessive queuing delay or packet loss.
1.2. Why is it a self-clocked algorithm?
Self-clocked congestion control algorithms provide a benefit over
their rate-based counterparts in that the former consists of two
adaptation mechanisms:
* A congestion window computation that evolves over a longer
timescale (several RTTs) especially when the congestion window
evolution is dictated by estimated delay (to minimize
vulnerability to, e.g., short-term delay variations).
* A fine-grained congestion control given by the self-clocking; it
operates on a shorter time scale (1 RTT). The benefits of self-
clocking are also elaborated upon in [TFWC]. The self-clocking
however acts more like an emergency break as bytes in flight can
exceed the congesion window only to a certain degree. The
rationale is to be able to transmit large video frames and avoid
that they are unnecessarily queued up on the sender side, but
still prevent a bootleneck queue
A rate-based congestion control algorithm typically adjusts the rate
based on delay and loss. The congestion detection needs to be done
with a certain time lag to avoid overreaction to spurious congestion
events such as delay spikes. Despite the fact that there are two or
more congestion indications, the outcome is that there is still only
one mechanism to adjust the sending rate. This makes it difficult to
reach the goals of high throughput and prompt reaction to congestion.
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1.3. Comparison with LEDBAT and TFWC in TCP
The core SCReAM algorithm, which is still similar in SCReAMv2, has
similarities to the concepts of self-clocking used in TCP-friendly
window-based congestion control [TFWC] and follows the packet
conservation principle. The packet conservation principle is
described as a key factor behind the protection of networks from
congestion [Packet-conservation].
The congestion window is determined in a way similar to LEDBAT
[RFC6817]. LEDBAT is a congestion control algorithm that uses send
and receive timestamps to estimate the queuing delay (from now on
denoted "qdelay") along the transmission path. This information is
used to adjust the congestion window. The general problem described
in the paper is that the base delay is offset by LEDBAT's own queue
buildup. The big difference with using LEDBAT in the SCReAM context
lies in the facts that the source is rate limited and that the RTP
queue must be kept short (preferably empty). In addition, the output
from a video encoder is rarely constant bitrate; static content
(talking heads, for instance) gives almost zero video bitrate. This
yields two useful properties when LEDBAT is used with SCReAM; they
help to avoid the issues described in [LEDBAT-delay-impact]:
1. There is always a certain probability that SCReAM is short of
data to transmit; this means that the network queue will become
empty every once in a while.
2. The max video bitrate can be lower than the link capacity. If
the max video bitrate is 5 Mbps and the capacity is 10 Mbps, then
the network queue will become empty.
It is sufficient that any of the two conditions above is fulfilled to
make the base delay update properly. Furthermore,
[LEDBAT-delay-impact] describes an issue with short-lived competing
flows. In SCReAM, these short-lived flows will cause the self-
clocking to slow down, thereby building up the RTP queue; in turn,
this results in a reduced media video bitrate. Thus, SCReAM slows
the bitrate more when there are competing short-lived flows than the
traditional use of LEDBAT does. The basic functionality in the use
of LEDBAT in SCReAM is quite simple; however, there are a few steps
in order to make the concept work with conversational media:
* Addition of a media rate control function.
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* Congestion window validation techniques. The congestion window is
used as a basis for the target bitrate calculation. For that
reason, various actions are taken to avoid that the congestion
window grows too much beyond the bytes in flight. Additional
contraints are applied when in congested state and when the max
target bitrate is reached.
* Use of inflection points in the congestion window calculation to
achieve reduced delay jitter (when L4S is not active).
* Adjustment of qdelay target for better performance when competing
with other loss-based congestion-controlled flows.
The above-mentioned features will be described in more detail in
Section Section 4.
The SCReAM/SCReAMv2 congestion control method uses techniques similar
to LEDBAT [RFC6817] to measure the qdelay. As is the case with
LEDBAT, it is not necessary to use synchronized clocks in the sender
and receiver in order to compute the qdelay. However, it is
necessary that they use the same clock frequency, or that the clock
frequency at the receiver can be inferred reliably by the sender.
Failure to meet this requirement leads to malfunction in the SCReAM/
SCReAMv2 congestion control algorithm due to incorrect estimation of
the network queue delay. Use of [RFC8888] as feedback ensures that
the same time base is used in sender and receiver.
2. Requirements Language
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 BCP
14 [RFC2119] [RFC8174] when, and only when, they appear in all
capitals, as shown here.
3. Overview of SCReAMv2 Algorithm
SCReAMv2 still consists of three main parts: network congestion
control, sender transmission control, and media rate control. All of
these parts reside at the sender side. Figure 1 shows the functional
overview of a SCReAMv2 sender. The receiver-side algorithm is very
simple in comparison, as it only generates feedback containing
acknowledgements of received RTP packets and indication of ECN bits.
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3.1. Network Congestion Control
The network congestion control sets an upper limit on how much data
can be in the network (bytes in flight); this limit is called CWND
(congestion window) and is used in the sender transmission control.
The sender calculates the congestion window based on the feedback
from the RTP receiver. The feedback is timestamp and ECN echo for
individual RTP packets.
The receiver of the media echoes a list of received RTP packets and
the timestamp of the RTP packet with the highest sequence number back
to the sender in feedback packets. The sender keeps a list of
transmitted packets, their respective sizes, and the time they were
transmitted. This information is used to determine the number of
bytes that can be transmitted at any given time instant. A
congestion window puts an upper limit on how many bytes can be in
flight, i.e., transmitted but not yet acknowledged. The congestion
window is however not an absolute limit as a slack is given to
efficiently transmit large video frames.
The congestion window seeks to increase by at least one segment per
RTT and this increase regardless congestion occurs or not, the
congestion window increase is restriced or relaxed based on the
current value of the congestion window and the time elapsed since
last congestion event. The congestion window update is increased by
one MSS (maximum known RTP packet size) per RTT with some variation
based on congestion window size and time elapsed since the last
congestion event. Multiplicative increase allows the congestion to
increase by a fraction of cwnd when congestion has not occured for a
while. The congestion window is thus an adaptive multiplicative
increase that is mainly additive increase when steady state is
reached but allows a faster convergence to a higher link speed.
Congestion window reduction is triggered by:
* Packet loss or Classic ECN marking is detected : The congestion
window is reduced by a predetermined fraction.
* Estimated queue delay exceeds a given threshold : The congestion
window is reduced given by how much the delay exceeds the
threshold.
* L4S ECN marking detected : The congestion window is reduced in
proportion to the fraction of packets that are marked (scalable
congestion control).
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3.2. Sender Transmission Control
The sender transmission control limits the output of data, given by
the relation between the number of bytes in flight and the congestion
window. The congestion window is however not a hard limit,
additional slack is given to avoid that RTP packets are queued up
unnecessarily on the sender side. This means that the algoritm
prefers to build up a queue in the network rather than on the sender
side. Additional congestion that this causes will reflect back and
cause a reduction of the congestion window. Packet pacing is used to
mitigate issues with ACK compression that MAY cause increased jitter
and/or packet loss in the media traffic. Packet pacing limits the
packet transmission rate given by the estimated link throughput.
Even if the send window allows for the transmission of a number of
packets, these packets are not transmitted immediately; rather, they
are transmitted in intervals given by the packet size and the
estimated link throughput. Packets are generally paced at a higher
rate than the target bitrate, this makes it possible to transmit
occasionally larger video frames in a timely manner.
3.3. Media Rate Control
The media rate control serves to adjust the media bitrate to ramp up
quickly enough to get a fair share of the system resources when link
throughput increases.
The reaction to reduced throughput must be prompt in order to avoid
getting too much data queued in the RTP packet queue(s) in the
sender. The media rate is calculated based on the congestion window
and RTT. For the case that multiple streams are enabled, the media
rate among the streams is distrubuted according to relative
priorities.
In cases where the sender's frame queues increase rapidly, such as in
the case of a Radio Access Type (RAT) handover, the SCReAMv2 sender
MAY implement additional actions, such as discarding of encoded media
frames or frame skipping in order to ensure that the RTP queues are
drained quickly. Frame skipping results in the frame rate being
temporarily reduced. Which method to use is a design choice and is
outside the scope of this algorithm description.
4. Detailed Description of SCReAMv2 sender algorithm
This section describes the sender-side algorithm in more detail. It
is split between the network congestion control, sender transmission
control, and media rate control.
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The sender implements media rate control and an RTP queue for each
media type or source, where RTP packets containing encoded media
frames are temporarily stored for transmission. Figure 1 shows the
details when a single media source (or stream) is used. A
transmission scheduler (not shown in the figure) is added to support
multiple streams. The transmission scheduler can enforce differing
priorities between the streams and act like a coupled congestion
controller for multiple flows. Support for multiple streams is
implemented in [SCReAM-CPP-implementation].
+---------------------------+
| Media encoder |
+---------------------------+
^ |(1)
| RTP
|(3) |
| V
| +-----------+
+---------+ | |
| Media | | Queue |
| rate | | |
| control | |RTP packets|
+---------+ | |
^ +-----------+
| |
| (2) |(4)
| RTP
| |
| v
+------------+ +--------------+
| Network | (7) | Sender |
+-->| congestion |------>| Transmission |
| | control | | Control |
| +------------+ +--------------+
| |(5)
+-------------RTCP----------+ RTP
(6) | |
| v
+------------+
| UDP |
| socket |
+------------+
Figure 1: Sender Functional View
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Media frames are encoded and forwarded to the RTP queue (1) in
Figure 1. The RTP packets are picked from the RTP queue (4), for
multiple flows from each RTP queue based on some defined priority
order or simply in a round-robin fashion, by the sender transmission
controller.
The sender transmission controller (in case of multiple flows a
transmission scheduler) sends the RTP packets to the UDP socket (5).
In the general case, all media SHOULD go through the sender
transmission controller and is limited so that the number of bytes in
flight is less than the congestion window albeit with a slack to
avoid that packets are unnecessarily delayed in the RTP queue.
RTCP packets are received (6) and the information about the bytes in
flight and congestion window is exchanged between the network
congestion control and the sender transmission control (7).
The congestion window and the estimated RTT is communicated to the
media rate control (2) to compute the appropriate target bitrate.
The target bitrate is updated whenever the congestion window is
updated. Additional parameters are also communicated to make the
rate control more stable when the congestion window is very small or
when L4S is not active. This is described more in detail below.
4.1. Constants and variables
Constants and state variables are listed in this section. Temporary
variables are not listed; instead, they are appended with '_t' in the
pseudocode to indicate their local scope.
4.1.1. Constants
The RECOMMENDED values, within parentheses "()", for the constants
are deduced from experiments.
* QDELAY_TARGET_LO (0.1): Target value for the minimum qdelay [s].
* QDELAY_TARGET_HI (0.4): Target value for the maximum qdelay [s].
This parameter provides an upper limit to how much the target
qdelay (qdelay_target) can be increased in order to cope with
competing loss-based flows. However, the target qdelay does not
have to be initialized to this high value, as it would increase
end-to-end delay and also make the rate control and congestion
control loops sluggish.
* MIN_CWND (3000): Minimum congestion window [byte].
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* MAX_BYTES_IN_FLIGHT_HEAD_ROOM (1.1): Headroom for the limitation
of CWND.
* BETA_LOSS (0.7): CWND scale factor due to loss event.
* BETA_ECN (0.8): CWND scale factor due to ECN event.
* MSS (1000 byte): Maximum segment size = Max RTP packet size.
* TARGET_BITRATE_MIN: Minimum target bitrate in [bps] (bits per
second).
* TARGET_BITRATE_MAX: Maximum target bitrate in [bps].
* RATE_PACE_MIN (50000): Minimum pacing rate in [bps].
* CWND_OVERHEAD (1.5): Indicates how much bytes in flight is allowed
to exceed cwnd.
* L4S_AVG_G (1.0/16): EWMA factor for l4s_alpha
* QDELAY_AVG_G (1.0/4): EWMA factor for qdelay_avg
* POST_CONGESTION_DELAY (4.0): Determines how long (seconds) after a
congestion event that the congestion window growth should be
cautious.
* MUL_INCREASE_FACTOR (0.02): Determines how much (as a fraction of
cwnd) that the cwnd can increase per RTT.
* LOW_CWND_SCALE_FACTOR (0.1): Scale factor applied to cwnd change
when CWND is very small.
* IS_L4S (false): Congestion control operates in L4S mode.
* VIRTUAL_RTT (0.025): Virtual RTT [s]
* PACKET_PACING_HEADROOM (1.5): Extra head room for packet pacing.
* BYTES_IN_FLIGHT_HEAD_ROOM (2.0): Extra headroom for bytes in
flight.
4.1.2. State Variables
The values within parentheses "()" indicate initial values.
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* qdelay_target (QDELAY_TARGET_LO): qdelay target [s], a variable
qdelay target is introduced to manage cases where a fixed qdelay
target would otherwise starve the RMCAT flow under such
circumstances (e.g., FTP competes for the bandwidth over the same
bottleneck). The qdelay target is allowed to vary between
QDELAY_TARGET_LO and QDELAY_TARGET_HI.
* qdelay_fraction_avg (0.0): Fractional qdelay filtered by the
Exponentially Weighted Moving Average (EWMA).
* qdelay_norm_hist[100] ({0,..,0}): Vector of the last 100
normalized qdelay samples.
* cwnd (MIN_CWND): Congestion window.
* cwnd_i (1): Congestion window inflection point.
* bytes_newly_acked (0): The number of bytes that was acknowledged
with the last received acknowledgement, i.e., bytes acknowledged
since the last CWND update.
* max_bytes_in_flight (0): The maximum number of bytes in flight in
the last round trip.
* max_bytes_in_flight_prev (0): The maximum number of bytes in
flight in previous round trip.
* send_wnd (0): Upper limit to how many bytes can currently be
transmitted. Updated when cwnd is updated and when RTP packet is
transmitted.
* target_bitrate (0): Media target bitrate [bps].
* rate_media (0.0): Measured bitrate [bps] from the media encoder.
* s_rtt (0.0): Smoothed RTT [s], computed with a similar method to
that described in [RFC6298].
* rtp_size (0): Size [byte] of the last transmitted RTP packet.
* loss_event_rate (0.0): The estimated fraction of RTTs with lost
packets detected.
* bytes_in_flight_ratio (0.0): Ratio between the bytes in flight and
the congestion window.
* cwnd_ratio (0.0): Ratio between MSS and cwnd.
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* l4s_alpha (0.0): Average fraction of marked packets per RTT.
* l4s_active (false): Indicates that L4S is enabled and packets are
indeed marked.
* last_update_l4s_alpha_time (0): Last time l4s_alpha was updated
[s].
* last_update_qdelay_avg_time (0): Last time qdelay_avg was updated
[s].
* packets_delivered_this_rtt (0): Counter for delivered packets.
* packets_marked_this_rtt (0): Counter delivered and ECN-CE marked
packets.
* last_congestion_detected_time (0): Last time congestion event
occured [s].
* last_cwnd_i_update_time (0): Last time cwnd_i was updated [s].
* bytes_newly_acked (0): Number of bytes newly ACKed, reset to 0
when congestion window is updated [byte].
* bytes_newly_acked_ce (0): Number of bytes newly ACKed and CE
marked, reset to 0 when congestion window is updated [byte].
* pace_bitrate (1e6): Packet pacing rate [bps].
* t_pace (1e-6): Pacing interval between packets [s].
* rel_framesize_high (1.0): High percentile of frame size,
normalized by nominal frame size for the given target bitrate
* frame_period (0.02): The frame period [s].
4.2. Network Congestion Control
This section explains the network congestion control, which performs
two main functions:
* Computation of congestion window at the sender: This gives an
upper limit to the number of bytes in flight.
* Calculation of send window at the sender: RTP packets are
transmitted if allowed by the relation between the number of bytes
in flight and the congestion window. This is controlled by the
send window.
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SCReAMv2 is a window-based and byte-oriented congestion control
protocol, where the number of bytes transmitted is inferred from the
size of the transmitted RTP packets. Thus, a list of transmitted RTP
packets and their respective transmission times (wall-clock time)
MUST be kept for further calculation.
The number of bytes in flight (bytes_in_flight) is computed as the
sum of the sizes of the RTP packets ranging from the RTP packet most
recently transmitted, down to but not including the acknowledged
packet with the highest sequence number. This can be translated to
the difference between the highest transmitted byte sequence number
and the highest acknowledged byte sequence number. As an example: If
an RTP packet with sequence number SN is transmitted and the last
acknowledgement indicates SN-5 as the highest received sequence
number, then bytes_in_flight is computed as the sum of the size of
RTP packets with sequence number SN-4, SN-3, SN-2, SN-1, and SN. It
does not matter if, for instance, the packet with sequence number
SN-3 was lost -- the size of RTP packet with sequence number SN-3
will still be considered in the computation of bytes_in_flight.
bytes_in_flight_ratio is calculated as the ratio between bytes_flight
and cwnd. This value should be computed at the beginning of the ACK
processing. cwnd_ratio is computed as the relation between MSS and
cwnd.
Furthermore, a variable bytes_newly_acked is incremented with a value
corresponding to how much the highest sequence number has increased
since the last feedback. As an example: If the previous
acknowledgement indicated the highest sequence number N and the new
acknowledgement indicated N+3, then bytes_newly_acked is incremented
by a value equal to the sum of the sizes of RTP packets with sequence
number N+1, N+2, and N+3. Packets that are lost are also included,
which means that even though, e.g., packet N+2 was lost, its size is
still included in the update of bytes_newly_acked. The
bytes_newly_acked_ce is, similar to bytes_newly_acked, a counter of
bytes newly acked with the extra condition that they are ECN-CE
marked. The bytes_newly_acked and bytes_newly_acked_ce are reset to
zero after a CWND update.
The feedback from the receiver is assumed to consist of the following
elements.
* A list of received RTP packets' sequence numbers. With an
indication if packets are ECN-CE marked.
* The wall-clock timestamp corresponding to the received RTP packet
with the highest sequence number.
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It is recommended to use RFC8888 [RFC8888] for the feedback as it
supports the feedback elements described above.
When the sender receives RTCP feedback, the qdelay is calculated as
outlined in [RFC6817]. A qdelay sample is obtained for each received
acknowledgement. A number of variables are updated as illustrated by
the pseudocode below; temporary variables are appended with '_t'.
Division operation is always floating point unless otherwise noted.
l4s_alpha is calculated based in number of packets delivered (and
marked). This makes calculation of L4S alpha more accurate at very
low bitrates, given that the tail RTP packet in a video frame is
often smaller than MSS.
packets_delivered_this_rtt += packets_acked
packets_marked_this_rtt += packets_acked_ce
if (now - last_update_l4s_alpha_time >= s_rtt)
# l4s_alpha is calculated from packets marked istf bytes marked
fraction_marked_t = packets_marked_this_rtt/
packets_delivered_this_rtt
l4s_alpha = L4S_AVG_G*fraction_marked_t + (1.0-L4S_AVG_G)*l4S_alpha
last_update_l4s_alpha_time = now
packets_delivered_this_rtt = 0
packets_marked_this_rtt = 0
end
if (now - last_update_qdelay_avg_time >= s_rtt)
# qdelay_avg is updated with a slow attack, fast decay EWMA filter
if (qdelay < qdelay_avg)
qdelay_avg = qdelay
else
qdelay_avg = QDELAY_AVG_G*qdelay + (1.0-QDELAY_AVG_G)*qdelay_avg
end
last_update_qdelay_avg_time = now
end
4.2.1. Reaction to Delay, Packet Loss and ECN-CE
Congestion is detected based on three different indicators:
* Lost packets detected. The loss detection is described in
Section 4.1.2.4.
* ECN-CE marked packets detected.
* Estimated queue delay exceeds a threshold.
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A congestion event occurs if any of the above indicators are true AND
it is than one smoothed RTT (s_rtt) since the last congestion event.
This ensures that the congestion window is reduced at most once per
smoothed RTT.
4.2.1.1. Lost packets
The congestion window back-off due to loss events is deliberately a
bit less than is the case with TCP Reno, for example. TCP is
generally used to transmit whole files; the file is then like a
source with an infinite bitrate until the whole file has been
transmitted. SCReAMv2, on the other hand, has a source which rate is
limited to a value close to the available transmit rate and often
below that value; the effect is that SCReAMv2 has less opportunity to
grab free capacity than a TCP-based file transfer. To compensate for
this, it is RECOMMENDED to let SCReAMv2 reduce the congestion window
less than what is the case with TCP when loss events occur.
4.2.1.2. ECN-CE and classic ECN
In classic ECN mode the cwnd is scaled by a fixed value (BETA_ECN).
The congestion window back-off due to an ECN event MAY be smaller
than if a loss event occurs. This is in line with the idea outlined
in [RFC8511] to enable ECN marking thresholds lower than the
corresponding packet drop thresholds.
4.2.1.3. ECN-CE and L4S
The cwnd is scaled down in proportion to the fraction of marked
packets per RTT. The scale down proportion is given by l4s_alpha,
which is an EWMA filtered version of the fraction of marked packets
per RTT. This is inline with how DCTCP works [RFC8257]. Additional
methods are applied to make the congestion window reduction
reasonably stable, especially when the congestion window is only a
few MSS. In addition, because SCReAMv2 can quite often be source
limited, additional steps are taken to restore the congestion window
to a proper value after a long period without congestion.
4.2.1.4. Increased queue delay
SCReAMv2 implements a delay based congestion control approach where
it mimics L4S congestion marking when the averaged queue delay
exceeds a target threshold. This threshold is set to qdelay_target/2
and the congestion backoff factor (l4s_alpha_v) increases linearly
from 0 to 100% as qdelay_avg goes from qdelay_target/2 to
qdelay_target. The averaged qdelay (qdelay_avg) is used to avoid
that the SCReAMv2 congestion control over-reacts to scheduling
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jitter, sudden delay spikes due to e.g. handover or link layer
retransmissions. Furthermore, the delay based congestion control is
inactivated when it is reasonably certain that L4S is active, i.e.
L4S is enabled and congested nodes apply L4S marking of packets. The
rationale is reduce negative effect of clockdrift that the delay
based control can introduce whenever possible.
4.2.2. Congestion Window Update
The congestion window update contains two parts. One that reduces
the congestion window when congestion events (listed above) occur,
and one part that continously increases the congestion window.
The target bitrate is updated whenever the congestion window is
updated.
Actions when congestion detected
if (now - last_congestion_detected_time >= s_rtt)
if (loss detected)
is_loss_t = true
end
if (packets marked)
is_ce_t = true
end
if (qdelay > qdelay_target/2)
# It is expected that l4s_alpha is below a given value,
l4_alpha_lim_t = 2 / target_bitrate * MSS * 8 / s_rtt
if (l4s_alpha < l4_alpha_lim_t || !l4s_active)
# L4S does not seem to be active
l4s_alpha_v_t = min(1.0, max(0.0,
(qdelay_avg - qdelay_target / 2) /
(qdelay_target / 2)));
is_virtual_ce_t = true
end
end
end
if (is_loss_t || is_ce_t || is_virtual_ce_t)
if (now - last_cwnd_i_update_time > 0.25)
last_cwnd_i_update_time = now
cwnd_i = cwnd
end
end
# Scale factor for cwnd update
cwnd_scale_factor_t =
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(LOW_CWND_SCALE_FACTOR + (MUL_INCREASE_FACTOR * cwnd) / MSS)
# Either loss, ECN mark or increased qdelay is detected
if (is_loss_t)
# Loss is detected
cwnd = cwnd * BETA_LOSS
end
if (is_ce_t)
# ECN-CE detected
if (IS_L4S)
# L4S mode
backoff_t = l4s_alpha / 2
# Increase stability for very small cwnd
backOff_t *= min(1.0, cwnd_scale_factor_t)
backOff_t *= max(0.8, 1.0 - cwnd_ratio * 2)
if (now - last_congestion_detected_time > 5)
# A long time since last congested because link throughput
# exceeds max video bitrate.
# There is a certain risk that CWND has increased way above
# bytes in flight, so we reduce it here to get it better on
# track and thus the congestion episode is shortened
cwnd = min(cwnd, max_bytes_in_flight_prev)
# Also, we back off a little extra if needed
# because alpha is quite likely very low
# This can in some cases be an over-reaction
# but as this function should kick in relatively seldom
# it should not be to too big concern
backoff_t = max(backoff_t, 0.25)
# In addition, bump up l4sAlpha to a more credible value
# This may over react but it is better than
# excessive queue delay
l4sAlpha = 0.25
end
cwnd = (1.0 - backoff_t) * cwnd
else
# Classic ECN mode
cwnd = cwnd * BETA_ECN
end
end
if (is_virtual_ce_t)
backoff_t = l4s_alpha_v_t / 2
cwnd = (1.0 - backoff_t) * cwnd
end
cwnd = max(MIN_CWND, cwnd)
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if (is_loss_t || is_ce_t || is_virtual_ce_t)
last_congestion_detected_time = now
end
The variable max_bytes_in_flight_prev indicates the maximum bytes in
flights in the previous round trip. The reason to this is that bytes
in flight can spike when congestion occurs, max_bytes_in_flight_prev
thus ensures better that an uncongested bytes in flight is used.
The cwnd_scale_factor_t scales the congestion window decrease upon
congestion as well as the increase. In a normal addititive increase
setting this would be 1.0/MSS. However to increase stability
especially when with L4S when cwnd is very small, the
cwnd_scale_factor_t can be as small as low as LOW_CWND_SCALE_FACTOR /
MSS. The result is then that the congestion window increase can be
as small as LOW_CWND_SCALE_FACTOR MSS per RTT. Because the same
restriction is applied to both decrease and increase of the
congestion window, the net effect is zero. The cwnd_scale_factor_t
is increased with larger cwnd to allow for a multiplicative increase
and thus a faster convergence when link capacity increases.
Congestion window increase
# Additional factor for cwnd update
post_congestion_scale_t = max(0.0, min(1.0,
(now - last_congestion_detected_time) / POST_CONGESTION_DELAY))
# Calculate bytes acked that are not CE marked
bytes_newly_acked_minus_ce_t = bytes_newly_acked-
bytes_newly_acked_ce
increment_t = bytes_newly_acked_minus_ce_t*cwnd_ratio
# Reduce increment for small RTTs
tmp_t = min(1.0, s_rtt / VIRTUAL_RTT)
increment *= tmp_t * tmp_t
if (!is_l4s_active)
# The increment is scaled down for more cautious
# ramp-up around the last known congestion window
# when congestion last occured.
# This is only applied when L4S is inactive
scl_t = 1.0
scl_t = (cwnd - cwnd_i) / cwnd_i
scl_t *= 4
scl_t = scl_t * scl_t
scl_t = max(0.1, min(1.0, scl_t))
increment_t *= scl_t
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end
# Slow down CWND increase when CWND is only a few MSS
# This goes hand in hand with that the down scaling is also
# slowed down then. Cwnd increase can be as slow as
# LOW_CWND_SCALE_FACTOR*MSS per RTT
float tmp_t = cwnd_scale_factor_t
# Further limit multiplicative increase when congestion occured
# recently.
if (tmp_t > 1.0)
tmp_t = 1.0 + ((tmp_t - 1.0) * post_congestion_scale_t);
end
increment *= tmp_t
# Increase CWND only if bytes in flight is large enough
# Quite a lot of slack is allowed here to avoid that bitrate
# locks to low values.
max_allowed_t = MSS + max(max_bytes_in_flight,
max_bytes_in_flight_prev) * BYTES_IN_FLIGHT_HEAD_ROOM
int cwnd_t = cwnd + increment_t
if (cwnd_t <= max_allowed_t)
cwnd = cwnd_t
end
The variable max_bytes_in_flight indicates the max bytes in flight in
the current round trip.
The multiplicative increase is restricted directly after a congestion
event and the restriction is gradually relaxed as the time since last
congested increased. The restriction makes the congestion window
growth to be no faster than additive increase when congestion
continusly occurs. For L4S operation this means that the SCReAMv2
algorithm will adhere to the 2 marked packets per RTT equilibrium at
steady state congestion.
It is particularly important that the congestion window reflects the
transmitted bitrate especially in L4S mode operation. An inflated
cwnd takes extra RTTs to bring down to a correct value upon
congestion and thus causes unnecessary queue buildup. At the same
time the congestion window must be allowed to be large enough to
avoid that the SCReAMv2 algorithm begins to limit itself, given that
the target bitrate is calculated based on the cwnd. Two mechanisms
are used to manage this:
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* Restore correct value of cwnd upon congestion. This is done if is
a prolonged time since the link was congested. A typical example
is that SCReAMv2 has been rate limited, i.e the target bitrate has
reached the TARGET_BITRATE_MAX.
* Limit cwnd when the target_bitrate has reached TARGET_BITRATE_MAX.
The cwnd is restricted based on a history of the last
max_bytes_in_flight values. See [SCReAM-CPP-implementation] for
details.
The two mechanisms complement one another.
4.2.3. Lost Packet Detection
Lost packet detection is based on the received sequence number list.
A reordering window SHOULD be applied to prevent packet reordering
from triggering loss events. The reordering window is specified as a
time unit, similar to the ideas behind Recent ACKnowledgement (RACK)
[RFC8985]. The computation of the reordering window is made possible
by means of a lost flag in the list of transmitted RTP packets. This
flag is set if the received sequence number list indicates that the
given RTP packet is missing. If later feedback indicates that a
previously lost marked packet was indeed received, then the
reordering window is updated to reflect the reordering delay. The
reordering window is given by the difference in time between the
event that the packet was marked as lost and the event that it was
indicated as successfully received. Loss is detected if a given RTP
packet is not acknowledged within a time window (indicated by the
reordering window) after an RTP packet with a higher sequence number
was acknowledged.
4.2.4. Send Window Calculation
The basic design principle behind packet transmission in SCReAM was
to allow transmission only if the number of bytes in flight is less
than the congestion window. There are, however, two reasons why this
strict rule will not work optimally:
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* Bitrate variations: Media sources such as video encoders generally
produce frames whose size always vary to a larger or smaller
extent. The RTP queue absorbs the natural variations in frame
sizes. However, the RTP queue should be as short as possible to
prevent the end-to-end delay from increasing. A strict 'send only
when bytes in flight is less than the congestion window' rule can
cause the RTP queue to grow simply because the send window is
limited. The consequence is that the congestion window will not
increase, or will increase very slowly, because the congestion
window is only allowed to increase when there is a sufficient
amount of data in flight. The final effect is that the media
bitrate increases very slowly or not at all.
* Reverse (feedback) path congestion: Especially in transport over
buffer-bloated networks, the one-way delay in the reverse
direction can jump due to congestion. The effect is that the
acknowledgements are delayed, and the self-clocking is temporarily
halted, even though the forward path is not congested. The
CWND_OVERHEAD allows for some degree of reverse path congestion as
the bytes in flight is allowed to exceed cwnd.
In SCReAMv2, the send window is given by the relation between the
adjusted congestion window and the amount of bytes in flight
according to the pseudocode below. The multiplication of cwnd with
CWND_OVERHEAD and rel_framesize_high has the effect that bytes in
flight is 'around' the cwnd rather than limited by the cwnd when the
link is congested. The implementation allows the RTP queue to be
small even when the frame sizes vary and thus increased e2e delay can
be avoided.
send_wnd = cwnd * CWND_OVERHEAD * rel_framesize_high -
bytes_in_flight
The send window is updated whenever an RTP packet is transmitted or
an RTCP feedback messaged is received.
4.2.5. Packet Pacing
Packet pacing is used in order to mitigate coalescing, i.e., when
packets are transmitted in bursts, with the risks of increased jitter
and potentially increased packet loss. Packet pacing is also
recommended to be used with L4S and also mitigates possible issues
with queue overflow due to key-frame generation in video coders. The
time interval between consecutive packet transmissions is greater
than or equal to t_pace, where t_pace is given by the equations below
:
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pace_bitrate = max(RATE_PACE_MIN, target_bitrate) *
PACKET_PACING_HEADROOM
t_pace = rtp_size * 8 / pace_bitrate
rtp_size is the size of the last transmitted RTP packet, and s_rtt is
the smoothed round trip time. RATE_PACE_MIN is the minimum pacing
rate.
4.2.6. Stream Prioritization
The SCReAM algorithm makes a distinction between network congestion
control and media rate control. This is easily extended to many
streams. RTP packets from two or more RTP queues are scheduled at
the rate permitted by the network congestion control.
The scheduling can be done by means of a few different scheduling
regimes. For example, the method for coupled congestion control
specified in [RFC8699] can be used. One implementation of SCReAM and
SCReAMv2 [SCReAM-CPP-implementation] uses credit-based scheduling.
In credit-based scheduling, credit is accumulated by queues as they
wait for service and is spent while the queues are being serviced.
For instance, if one queue is allowed to transmit 1000 bytes, then a
credit of 1000 bytes is allocated to the other unscheduled queues.
This principle can be extended to weighted scheduling, where the
credit allocated to unscheduled queues depends on the relative
weights. The latter is also implemented in
[SCReAM-CPP-implementation] in which case the target bitrate for the
streams are also scaled relative to the scheduling priority.
4.3. Media Rate Control
The media rate control algorithm is executed whenever the congestion
window is updated and updates the target bitrate. The target bitrate
is essentiatlly based on the congestion window and the (smoothed) RTT
according to
target_bitrate = 8 * cwnd / s_rtt
The role of the media rate control is to strike a reasonable balance
between a low amount of queuing in the RTP queue(s) and a sufficient
amount of data to send in order to keep the data path busy. Because
the congestion window is updated based on loss, ECN-CE and delay, so
does the target rate also update.
The code above however needs some modifications to work fine in a
number of scenarios
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* L4S is inactive, i.e L4S is either not enabled or congested
bottlenecks do not L4S mark packets
* Frame sizes vary
* cwnd is very small, just a few MSS or smaller
The complete pseudo code for adjustment of the target bitrate is
shown below
tmp_t = 1.0
# limit bitrate if bytes in flight exceeds is close to or
# exceeds cwnd. This helps to avoid large rate fluctiations and
# variations in RTT
# Only applied when L4S is inactive
if (!l4s_active && bytes_in_flight_ratio > BYTES_IN_FLIGHT_LIMIT)
tmp_t /= min(BYTES_IN_FLIGHT_LIMIT_COMPENSATION,
bytesInFlightRatio / BYTES_IN_FLIGHT_LIMIT)
end
# Scale down rate slighty when the congestion window is very
# small compared to MSS
tmp_t *= 1.0 - min(0.8, max(0.0, cwnd_ratio - 0.1))
# Scale down rate when frame sizes vary much
tmp_t /= rel_framesize_high
# Calculate target bitrate and limit to min and max allowed
# values
target_bitrate = tmp_t * 8 * cwnd / s_rtt
target_bitrate = min(TARGET_BITRATE_MAX,
max(TARGET_BITRATE_MIN,target_bitrate))
The variable rel_framesize_high is based on calculation of the high
percentile of the frame sizes. The calculation is based on a
histogram of the frame sizes relative to the expected frame size
given the target bitrate and frame period. The calculation of
rel_framesize_high is done for every new video frame and is outlined
roughly with the pseudo code below. For more detailed code, see
[SCReAM-CPP-implementation].
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# frame_size is that frame size for the last encoded frame
tmp_t = frame_size / (target_bitrate * frame_period / 8)
if (tmp_t > 1.0)
# Insert sample into histogram
insert_into_histogram(tmp_t)
# Get high percentile
rel_framesize_high = get_histogram_high_percentile()
end
A 75%-ile is used in [SCReAM-CPP-implementation], the histogram can
be made leaky such that old samples are gradually forgotten.
4.4. Competing Flows Compensation
It is likely that a flow will have to share congested bottlenecks
with other flows that use a more aggressive congestion control
algorithm (for example, large FTP flows using loss-based congestion
control). The worst condition occurs when the bottleneck queues are
of tail-drop type with a large buffer size. SCReAMv2 takes care of
such situations by adjusting the qdelay_target when loss-based flows
are detected, as shown in the pseudocode below.
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adjust_qdelay_target(qdelay)
qdelay_norm_t = qdelay / QDELAY_TARGET_LOW
update_qdelay_norm_history(qdelay_norm_t)
# Compute variance
qdelay_norm_var_t = VARIANCE(qdelay_norm_history(200))
# Compensation for competing traffic
# Compute average
qdelay_norm_avg_t = AVERAGE(qdelay_norm_history(50))
# Compute upper limit to target delay
new_target_t = qdelay_norm_avg_t + sqrt(qdelay_norm_var_t)
new_target_t *= QDELAY_TARGET_LO
if (loss_event_rate > 0.002)
# Packet losses detected
qdelay_target = 1.5 * new_target_t
else
if (qdelay_norm_var_t < 0.2)
# Reasonably safe to set target qdelay
qdelay_target = new_target_t
else
# Check if target delay can be reduced; this helps prevent
# the target delay from being locked to high values forever
if (new_target_t < QDELAY_TARGET_LO)
# Decrease target delay quickly, as measured queuing
# delay is lower than target
qdelay_target = max(qdelay_target * 0.5, new_target_t)
else
# Decrease target delay slowly
qdelay_target *= 0.9
end
end
end
# Apply limits
qdelay_target = min(QDELAY_TARGET_HI, qdelay_target)
qdelay_target = max(QDELAY_TARGET_LO, qdelay_target)
Two temporary variables are calculated. qdelay_norm_avg_t is the
long-term average queue delay, qdelay_norm_var_t is the long-term
variance of the queue delay. A high qdelay_norm_var_t indicates that
the queue delay changes; this can be an indication that bottleneck
bandwidth is reduced or that a competing flow has just entered.
Thus, it indicates that it is not safe to adjust the queue delay
target.
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A low qdelay_norm_var_t indicates that the queue delay is relatively
stable. The reason could be that the queue delay is low, but it
could also be that a competing flow is causing the bottleneck to
reach the point that packet losses start to occur, in which case the
queue delay will stay relatively high for a longer time.
The queue delay target is allowed to be increased if either the loss
event rate is above a given threshold or qdelay_norm_var_t is low.
Both these conditions indicate that a competing flow may be present.
In all other cases, the queue delay target is decreased.
The function that adjusts the qdelay_target is simple and could
produce false positives and false negatives. The case that self-
inflicted congestion by the SCReAMv2 algorithm may be falsely
interpreted as the presence of competing loss-based FTP flows is a
false positive. The opposite case -- where the algorithm fails to
detect the presence of a competing FTP flow -- is a false negative.
Extensive simulations have shown that the algorithm performs well in
LTE test cases and that it also performs well in simple bandwidth-
limited bottleneck test cases with competing FTP flows. However, the
potential failure of the algorithm cannot be completely ruled out. A
false positive (i.e., when self-inflicted congestion is mistakenly
identified as competing flows) is especially problematic when it
leads to increasing the target queue delay, which can cause the end-
to-end delay to increase dramatically.
If it is deemed unlikely that competing flows occur over the same
bottleneck, the algorithm described in this section MAY be turned
off. One such case is QoS-enabled bearers in 3GPP-based access such
as LTE. However, when sending over the Internet, often the network
conditions are not known for sure, so in general it is not possible
to make safe assumptions on how a network is used and whether or not
competing flows share the same bottleneck. Therefore, turning this
algorithm off must be considered with caution, as it can lead to
basically zero throughput if competing with loss-based traffic.
4.5. Handling of systematic errors in video coders
Some video encoders are prone to systematically generate an output
bitrate that is systematically larger or smaller than the target
bitrate. SCReAMv2 can handle some deviation inherently but for
larger devation it becomes necessary to compensate for this. The
algorithm for this is detailed in [SCReAM-CPP-implementation].
ToDo: A future draft version will describe this in more detail as it
has been fully integrated into SCReAMv2.
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5. Receiver Requirements on Feedback Intensity
The simple task of the receiver is to feed back acknowledgements with
with time stamp and ECN bits indication for received packets to the
sender. Upon reception of each RTP packet, the receiver MUST
maintain enough information to send the aforementioned values to the
sender via an RTCP transport- layer feedback message. The frequency
of the feedback message depends on the available RTCP bandwidth. The
requirements on the feedback elements and the feedback interval are
described below.
SCReAMv2 benefits from relatively frequent feedback. It is
RECOMMENDED that a SCReAMv2 implementation follows the guidelines
below.
The feedback interval depends on the media bitrate. At low bitrates,
it is sufficient with a feedback every frame; while at high bitrates,
a feedback interval of roughly 5ms ms is preferred. At very high
bitrates, even shorter feedback intervals MAY be needed in order to
keep the self-clocking in SCReAMv2 working well. One indication that
feedback is too sparse is that the SCReAMv2 implementation cannot
reach high bitrates, even in uncongested links. More frequent
feedback might solve this issue.
The numbers above can be formulated as a feedback interval function
that can be useful for the computation of the desired RTCP bandwidth.
The following equation expresses the feedback rate:
# Assume 100 byte RTCP packets
rate_fb = 0.02 * [average received rate] / (100.0 * 8.0);
rate_fb = min(1000, max(10, rate_fb))
# Calculate feedback intervals
fb_int = 1.0/rate_fb
Feedback should also forcibly be transmitted in any of these cases:
* More than N packets received since last RTCP feedback has been
transmitted
* An RTP packet with marker bit set is received
The transmission interval is not critical. So, in the case of multi-
stream handling between two hosts, the feedback for two or more
synchronization sources (SSRCs) can be bundled to save UDP/IP
overhead. However, the final realized feedback interval SHOULD
notexceed 2*fb_int in such cases, meaning that a scheduled feedback
transmission event should not be delayed more than fb_int.
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SCReAMv2 works with AVPF regular mode; immediate or early mode is not
required by SCReAMv2 but can nonetheless be useful for RTCP messages
not directly related to SCReAMv2, such as those specified in
[RFC4585]. It is RECOMMENDED to use reduced-size RTCP [RFC5506],
where regular full compound RTCP transmission is controlled by trr-
int as described in [RFC4585].
6. Discussion
This section covers a few discussion points.
* Clock drift: SCReAM/SCReAMv2 can suffer from the same issues with
clock drift as is the case with LEDBAT [RFC6817]. However,
Appendix A.2 in [RFC6817] describes ways to mitigate issues with
clock drift. A clockdrift compensation method is also implemented
in [SCReAM-CPP-implementation].
* Clock skipping: The sender or receiver clock can occasionally
skip. Handling of this is implemented in
[SCReAM-CPP-implementation].
* The target bitrate given by SCReAMv2 is the bitrate including the
RTP and Forward Error Correction (FEC) overhead. The media
encoder SHOULD take this overhead into account when the media
bitrate is set. This means that the media coder bitrate SHOULD be
computed as: media_rate = target_bitrate -
rtp_plus_fec_overhead_bitrate It is not necessary to make a 100%
perfect compensation for the overhead, as the SCReAM algorithm
will inherently compensate for moderate errors. Under-
compensating for the overhead has the effect of increasing jitter,
while overcompensating will cause the bottleneck link to become
underutilized.
* The link utilization with SCReAMv2 can be lower than 100%. There
are several possible reasons to this:
- Large variations in frame sizes: Large variations in frame size
makes SCReAMv2 push down the target_bitrate to give sufficient
headroom and avoid queue buildup in the network. It is in
general recommended to operate video coders in low latency mode
and enable GDR (Gradual Decoding Refresh) if possible to
minimize frame size variations.
- Link layer properties: Media transport in 5G in uplink
typically requires to transmit a scheduling request (SR) to get
persmission to transmit data. Because transmission of video is
frame based, there is a high likelihood that the channel
becomes idle between frames (especially with L4S), in which
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case a new SR/grant exchange is needed. This potentially means
that uplink transmission slots are unused with a lower link
utilization as a result.
* Packet pacing is recommended, it is however possible to operate
SCReAMv2 with packet pacing disabled. The code in
[SCReAM-CPP-implementation] implements additonal mechanisms to
achieve a high link utilization when packet pacing is disabled.
* RFC8888 Feedback issues: RTCP feedback packets can be lost, this
means that the loss detection in SCReAMv2 may trigger even though
packets arrive safely on the receiver side.
[SCReAM-CPP-implementation] solves this by using overlapping RTCP
feedback, i.e RTCP feedback is transmitted no more seldom than
every 16th packet, and where each RTCP feedback spans the last 64
received packets. This however creates unnecessary overhead.
[RFC3550] RR (Receiver Reports) can possibly be another solution
to achieve better robustness with less overhead.
7. IANA Considerations
This document does not require any IANA actions.
8. Security Considerations
The feedback can be vulnerable to attacks similar to those that can
affect TCP. It is therefore RECOMMENDED that the RTCP feedback is at
least integrity protected. Furthermore, as SCReAM/SCReAMv2 is self-
clocked, a malicious middlebox can drop RTCP feedback packets and
thus cause the self-clocking to stall. However, this attack is
mitigated by the minimum send rate maintained by SCReAM/SCReAMv2 when
no feedback is received.
9. References
9.1. Normative References
[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, <https://www.rfc-editor.org/info/rfc3550>.
[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,
<https://www.rfc-editor.org/info/rfc4585>.
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[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, <https://www.rfc-editor.org/info/rfc5506>.
[RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent,
"Computing TCP's Retransmission Timer", RFC 6298,
DOI 10.17487/RFC6298, June 2011,
<https://www.rfc-editor.org/info/rfc6298>.
[RFC6817] Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind,
"Low Extra Delay Background Transport (LEDBAT)", RFC 6817,
DOI 10.17487/RFC6817, December 2012,
<https://www.rfc-editor.org/info/rfc6817>.
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>.
[RFC8888] Sarker, Z., Perkins, C., Singh, V., and M. Ramalho, "RTP
Control Protocol (RTCP) Feedback for Congestion Control",
RFC 8888, DOI 10.17487/RFC8888, January 2021,
<https://www.rfc-editor.org/info/rfc8888>.
9.2. Informative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
[RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use Cases and Requirements", RFC 7478,
DOI 10.17487/RFC7478, March 2015,
<https://www.rfc-editor.org/info/rfc7478>.
[RFC8298] Johansson, I. and Z. Sarker, "Self-Clocked Rate Adaptation
for Multimedia", RFC 8298, DOI 10.17487/RFC8298, December
2017, <https://www.rfc-editor.org/info/rfc8298>.
[RFC8511] Khademi, N., Welzl, M., Armitage, G., and G. Fairhurst,
"TCP Alternative Backoff with ECN (ABE)", RFC 8511,
DOI 10.17487/RFC8511, December 2018,
<https://www.rfc-editor.org/info/rfc8511>.
[RFC8699] Islam, S., Welzl, M., and S. Gjessing, "Coupled Congestion
Control for RTP Media", RFC 8699, DOI 10.17487/RFC8699,
January 2020, <https://www.rfc-editor.org/info/rfc8699>.
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[RFC8869] Sarker, Z., Zhu, X., and J. Fu, "Evaluation Test Cases for
Interactive Real-Time Media over Wireless Networks",
RFC 8869, DOI 10.17487/RFC8869, January 2021,
<https://www.rfc-editor.org/info/rfc8869>.
[RFC8985] Cheng, Y., Cardwell, N., Dukkipati, N., and P. Jha, "The
RACK-TLP Loss Detection Algorithm for TCP", RFC 8985,
DOI 10.17487/RFC8985, February 2021,
<https://www.rfc-editor.org/info/rfc8985>.
[RFC8257] Bensley, S., Thaler, D., Balasubramanian, P., Eggert, L.,
and G. Judd, "Data Center TCP (DCTCP): TCP Congestion
Control for Data Centers", RFC 8257, DOI 10.17487/RFC8257,
October 2017, <https://www.rfc-editor.org/info/rfc8257>.
[Packet-conservation]
Jacobson, V., "Congestion Avoidance and Control", ACM
SIGCOMM Computer Communication Review,
DOI 10.1145/52325.52356, August 1988,
<https://doi.org/10.1145/52325.52356>.
[LEDBAT-delay-impact]
Ros, D. and M. Welzl, "Assessing LEDBAT's Delay Impact",
IEEE Communications Letters, Vol. 17, No. 5,,
DOI 10.1109/LCOMM.2013.040213.130137, May 2013,
<http://home.ifi.uio.no/michawe/research/publications/
ledbat-impact-letters.pdf>.
[QoS-3GPP] "Policy and charging control architecture", 3GPP TS
23.203, July 2017,
<http://www.3gpp.org/ftp/specs/archive/23_series/23.203/>.
[SCReAM-CPP-implementation]
Ericsson Research, "SCReAM - Mobile optimised congestion
control algorithm", n.d.,
<https://github.com/EricssonResearch/scream>.
[SCReAM-evaluation-L4S]
Ericsson Research, "SCReAM - evaluations with L4S", n.d.,
<https://github.com/EricssonResearch/scream/blob/master/
L4S-Results.pdf?raw=true>.
[TFWC] Choi, S. and M. Handley, "Fairer TCP-Friendly Congestion
Control Protocol for Multimedia Streaming Applications",
DOI 10.1145/1364654.1364717, December 2007, <http://www-
dept.cs.ucl.ac.uk/staff/M.Handley/papers/tfwc-conext.pdf>.
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Appendix A. Acknowledgments
We would like to thank the following people for their comments,
questions, and support during the work that led to this memo: Mirja
Kuehlewind
Authors' Addresses
Ingemar Johansson
Ericsson
Email: ingemar.s.johansson@ericsson.com
Magnus Westerlund
Ericsson
Email: magnus.westerlund@ericsson.com
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