Internet DRAFT - draft-ietf-rmcat-scream-cc
draft-ietf-rmcat-scream-cc
RMCAT WG I. Johansson
Internet-Draft Z. Sarker
Intended status: Experimental Ericsson AB
Expires: April 29, 2018 October 26, 2017
Self-Clocked Rate Adaptation for Multimedia
draft-ietf-rmcat-scream-cc-13
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 uses a hybrid loss and delay
based congestion control algorithm. The algorithm is evaluated over
both simulated Internet bottleneck scenarios as well as in a Long
Term Evolution (LTE) system simulator and is shown to achieve both
low latency and high video throughput in these scenarios.
Status of This Memo
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provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on April 29, 2018.
Copyright Notice
Copyright (c) 2017 IETF Trust and the persons identified as the
document authors. All rights reserved.
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include Simplified BSD License text as described in Section 4.e of
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. Wireless (LTE) access properties . . . . . . . . . . . . 3
1.2. Why is it a self-clocked algorithm? . . . . . . . . . . . 4
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Overview of SCReAM Algorithm . . . . . . . . . . . . . . . . 4
3.1. Network Congestion Control . . . . . . . . . . . . . . . 7
3.2. Sender Transmission Control . . . . . . . . . . . . . . . 8
3.3. Media Rate Control . . . . . . . . . . . . . . . . . . . 8
4. Detailed Description of SCReAM . . . . . . . . . . . . . . . 9
4.1. SCReAM Sender . . . . . . . . . . . . . . . . . . . . . . 9
4.1.1. Constants and Parameter values . . . . . . . . . . . 9
4.1.1.1. Constants . . . . . . . . . . . . . . . . . . . . 10
4.1.1.2. State variables . . . . . . . . . . . . . . . . . 11
4.1.2. Network congestion control . . . . . . . . . . . . . 13
4.1.2.1. Reaction to packets loss and ECN . . . . . . . . 16
4.1.2.2. Congestion window update . . . . . . . . . . . . 16
4.1.2.3. Competing flows compensation . . . . . . . . . . 19
4.1.2.4. Lost packet detection . . . . . . . . . . . . . . 21
4.1.2.5. Send window calculation . . . . . . . . . . . . . 22
4.1.2.6. Packet pacing . . . . . . . . . . . . . . . . . . 23
4.1.2.7. Resuming fast increase . . . . . . . . . . . . . 23
4.1.2.8. Stream prioritization . . . . . . . . . . . . . . 23
4.1.3. Media rate control . . . . . . . . . . . . . . . . . 24
4.2. SCReAM Receiver . . . . . . . . . . . . . . . . . . . . . 27
4.2.1. Requirements on feedback elements . . . . . . . . . . 27
4.2.2. Requirements on feedback intensity . . . . . . . . . 29
5. Discussion . . . . . . . . . . . . . . . . . . . . . . . . . 29
6. Implementation status . . . . . . . . . . . . . . . . . . . . 30
6.1. OpenWebRTC . . . . . . . . . . . . . . . . . . . . . . . 31
6.2. A C++ Implementation of SCReAM . . . . . . . . . . . . . 31
7. Suggested experiments . . . . . . . . . . . . . . . . . . . . 32
8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 33
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 33
10. Security Considerations . . . . . . . . . . . . . . . . . . . 33
11. Change history . . . . . . . . . . . . . . . . . . . . . . . 33
12. References . . . . . . . . . . . . . . . . . . . . . . . . . 34
12.1. Normative References . . . . . . . . . . . . . . . . . . 35
12.2. Informative References . . . . . . . . . . . . . . . . . 35
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 37
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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 realtime
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 realtime communications as the
transmission channel bandwidth can vary over time. Wireless access
such as LTE, 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
realtime media, such as WebRTC, should be both quick and be able to
operate over a large range in channel capacity. This memo describes
SCReAM (Self-Clocked Rate Adaptation for Multimedia), a solution that
implements congestion control for RTP streams [RFC3550]. While
SCReAM was originally devised for WebRTC (Web Real-Time
Communication) [RFC7478], it 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 LEDBAT based rate adaptation algorithm
[RFC6817]. SCReAM is not entirely self-clocked as it augments self-
clocking with pacing and a minimum send rate.
SCReAM can take advantage of ECN (Explicit Congestion Notification)
in cases where ECN is supported by the network and the hosts. ECN is
however not required for the basic congestion control functionality
in SCReAM.
1.1. Wireless (LTE) access properties
[I-D.ietf-rmcat-wireless-tests] describes the complications that can
be observed in wireless environments. Wireless access such as LTE
can typically not 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 LTE can support bitrates well above
100Mbps, there are cases when the available bitrate can be much
lower, examples are situations with high network load and poor
coverage. An additional complication is that the network throughput
can drop for short time intervals at e.g. 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
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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, an RMCAT 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 the
rate based counterparts in that the former consists of two adaptation
mechanisms:
o 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).
o A fine grained congestion control given by the self-clocking which
operates on a shorter time scale (1 RTT). The benefits of self-
clocking are also elaborated upon in [TFWC].
A rate based congestion control typically adjusts the rate based on
delay and loss. The congestion detection needs to be done with a
certain time lag to avoid over-reaction to spurious congestion events
such as delay spikes. Despite the fact that there are two or more
congestion indications, the outcome is still 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.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
3. Overview of SCReAM Algorithm
The core SCReAM algorithm has similarities to the concepts of self-
clocking used in TFWC [TFWC] and follows the packet conservation
principle. The packet conservation principle is described as an
important key-factor behind the protection of networks from
congestion [Packet-conservation].
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In SCReAM, 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 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 use of LEDBAT ensures that
the end-to-end latency is kept low. [LEDBAT-delay-impact] shows that
LEDBAT has certain inherent issues that makes it counteract its
purpose to achieve low delay. 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 fact that the source is rate limited and that it is required that
the RTP queue is 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 gives two useful properties when LEDBAT is used with SCReAM that
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, which means that the network queue will run
empty every once in a while.
2. The max video bitrate can be lower than the link capacity. If
the max video bitrate is 5Mbps and the capacity is 10Mbps then
the network queue will run 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, the case in SCReAM is that these short lived flows will cause
the self-clocking in SCReAM to slow down with the result that the RTP
queue is built up, which will in turn result in a reduced media video
bitrate. SCReAM will thus yield more to competing short lived flows
than what is the case with traditional use of LEDBAT.
The basic functionality in the use of LEDBAT in SCReAM is quite
simple, there are however a few steps to take to make the concept
work with conversational media:
o Congestion window validation techniques. These are similar in
action as the method described in [RFC7661]. Congestion window
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validation ensures that the congestion window is limited by the
actual number bytes in flight, this is important especially in the
context of rate limited sources such as video. Lack of congestion
window validation would lead to a slow reaction to congestion as
the congestion window does not properly reflect the congestion
state in the network. The allowed idle period in this memo is
shorter than in [RFC7661], this to avoid excessive delays in the
cases where e.g. wireless throughput has decreased during a period
where the output bitrate from the media coder has been low, for
instance due to inactivity. Furthermore, this memo allows for
more relaxed rules for when the congestion window is allowed to
grow, this is necessary as the variable output bitrate generally
means that the congestion window is often under-utilized.
o Fast increase makes the bitrate increase faster when no congestion
is detected. It makes the media bitrate ramp-up within 5 to 10
seconds. The behavior is similar to TCP slowstart. The fast
increase is exited when congestion is detected. The fast increase
state can however resume if the congestion level is low, this
enables a reasonably quick rate increase in case link throughput
increases.
o A qdelay trend is computed for earlier detection of incipient
congestion and as a result it reduces jitter.
o Addition of a media rate control function.
o Use of inflection points in the media rate calculation to achieve
reduced jitter.
o 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
sections Section 3.1 to Section 3.3. The full details are described
in Section 4.
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+---------------------------+
| Media encoder |
+---------------------------+
^ |
| |(1)
|(3) RTP
| V
| +-----------+
+---------+ | |
| Media | (2) | Queue |
| rate |<------| |
| control | |RTP packets|
+---------+ | |
+-----------+
|
|(4)
RTP
|
v
+------------+ +--------------+
| Network | (7) | Sender |
+-->| congestion |------>| Transmission |
| | control | | Control |
| +------------+ +--------------+
| |
|-------------RTCP----------| |(5)
(6) | RTP
| v
+------------+
| UDP |
| socket |
+------------+
Figure 1: SCReAM sender functional view
The SCReAM algorithm consists of three main parts: network congestion
control, sender transmission control and media rate control. All of
these three parts reside at the sender side. Figure 1 shows the
functional overview of a SCReAM sender. The receiver side algorithm
is very simple in comparison as it only generates feedback containing
acknowledgements of received RTP packets and an ECN count.
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.
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The SCReAM 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 sender and receiver
in order to compute the qdelay. It is however 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 congestion
control algorithm due to incorrect estimation of the network queue
delay.
The SCReAM sender calculates the congestion window based on the
feedback from the SCReAM receiver. The congestion window is allowed
to increase if the qdelay is below a predefined qdelay target,
otherwise the congestion window decreases. The qdelay target is
typically set to 50-100ms. This ensures that the queuing delay is
kept low. The reaction to loss or ECN events leads to an instant
reduction of CWND. Note that the source rate limited nature of real
time media such as video, typically means that the queuing delay will
mostly be below the given delay target, this is contrary to the case
where large files are transmitted using LEDBAT congestion control, in
which case the queuing delay will stay close to the delay target.
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. 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, but rather they are transmitted in
intervals given by the packet size and the estimated link throughput.
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 bitrate is decreased if the RTP queue size exceeds
a threshold.
In cases where the sender frame queues increase rapidly such as in
the case of a RAT (Radio Access Type) handover it MAY be necessary to
implement additional actions, such as discarding of encoded media
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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
outside the scope of this algorithm description.
4. Detailed Description of SCReAM
4.1. SCReAM Sender
This section describes the sender side algorithm in more detail. It
is split between the network congestion control, sender transmission
control and the media rate control.
A SCReAM 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 frames are encoded and forwarded to the RTP queue (1) in
Figure 1. The media rate adaptation adapts to the size of the RTP
queue (2) and provides a target rate for the media encoder (3). The
RTP packets are picked from the RTP queue (for multiple flows from
each RTP queue based on some defined priority order or simply in a
round robin fashion) (4) 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. RTCP packets are received
(6) and the information about bytes in flight and congestion window
is exchanged between the network congestion control and the sender
transmission control (7).
4.1.1. Constants and Parameter values
Constants and state variables are listed in this section. Temporary
variables are not listed, instead they are appended with '_t' in the
pseudo code to indicate their local scope.
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4.1.1.1. Constants
The RECOMMENDED values, within (), for the constants are deduced from
experiments. The units are enclosed in square brackets [ ].
QDELAY_TARGET_LO (0.1s)
Target value for the minimum qdelay.
QDELAY_TARGET_HI (0.4s)
Target value for the maximum qdelay. 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. The
target qdelay does not have to be initialized to this high value
however as it would increase e2e delay and also make the rate
control and congestion control loop sluggish.
QDELAY_WEIGHT (0.1)
Averaging factor for qdelay_fraction_avg.
QDELAY_TREND_TH (0.2)
Threshold for the detection of incipient congestion.
MIN_CWND (3000byte)
Minimum congestion window.
MAX_BYTES_IN_FLIGHT_HEAD_ROOM (1.1)
Headroom for the limitation of CWND.
GAIN (1.0)
Gain factor for congestion window adjustment.
BETA_LOSS (0.8)
CWND scale factor due to loss event.
BETA_ECN (0.9)
CWND scale factor due to ECN event.
BETA_R (0.9)
Target rate scale factor due to loss event.
MSS (1000 byte)
Maximum segment size = Max RTP packet size.
RATE_ADJUST_INTERVAL (0.2s)
Interval between media bitrate adjustments.
TARGET_BITRATE_MIN
Min target bitrate [bps], bps is bits per second.
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TARGET_BITRATE_MAX
Max target bitrate [bps].
RAMP_UP_SPEED (200000bps/s)
Maximum allowed rate increase speed.
PRE_CONGESTION_GUARD (0.0..1.0)
Guard factor against early congestion onset. A higher value gives
less jitter, possibly at the expense of a lower link utilization.
This value MAY be subject to tuning depending on e.g media coder
characteristics, experiments with H264 and VP8 indicate that 0.1 is
a suitable value. See [SCReAM-CPP-implementation] and
[SCReAM-implementation-experience] for evaluation of a real
implementation.
TX_QUEUE_SIZE_FACTOR (0.0..2.0)
Guard factor against RTP queue buildup. This value MAY be subject
to tuning depending on e.g media coder characteristics, experiments
with H264 and VP8 indicate that 1.0 is a suitable value. See
[SCReAM-CPP-implementation] and [SCReAM-implementation-experience]
for evaluation of a real implementation.
RTP_QDELAY_TH (0.02s) RTP queue delay threshold for a target rate
reduction.
TARGET_RATE_SCALE_RTP_QDELAY (0.95) Target rate scale when RTP
qdelay threshold exceeds RTP_QDELAY_TH.
QDELAY_TREND_LO (0.2) Threshold value for qdelay_trend.
T_RESUME_FAST_INCREASE (5s) Time span until fast increase can be
resumed, given that the qdelay_trend is below QDELAY_TREND_LO.
RATE_PACE_MIN (50000bps) Minimum pacing rate.
4.1.1.2. State variables
The values within () indicate initial values.
qdelay_target (QDELAY_TARGET_LO)
qdelay target, a variable qdelay target is introduced to manage
cases where e.g. FTP competes for the bandwidth over the same
bottleneck, a fixed qdelay target would otherwise starve the RMCAT
flow under such circumstances. The qdelay target is allowed to
vary between QDELAY_TARGET_LO and QDELAY_TARGET_HI.
qdelay_fraction_avg (0.0)
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EWMA (Exponentially Weighted Moving Average) filtered fractional
qdelay.
qdelay_fraction_hist[20] ({0,..,0})
Vector of the last 20 fractional qdelay samples.
qdelay_trend (0.0)
qdelay trend, indicates incipient congestion.
qdelay_trend_mem (0.0)
Low pass filtered version of qdelay_trend.
qdelay_norm_hist[100] ({0,..,0})
Vector of the last 100 normalized qdelay samples.
in_fast_increase (true)
True if in fast increase state.
cwnd (MIN_CWND)
Congestion window.
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 over a sliding time window,
i.e. transmitted but not yet acknowledged bytes.
send_wnd (0)
Upper limit to how many bytes that can currently be transmitted.
Updated when cwnd is updated and when RTP packet is transmitted.
target_bitrate (0 bps)
Media target bitrate.
target_bitrate_last_max (1 bps)
Media target bitrate inflection point i.e. the last known highest
target_bitrate. Used to limit bitrate increase speed close to the
last known congestion point.
rate_transmit (0.0 bps)
Measured transmit bitrate.
rate_ack (0.0 bps)
Measured throughput based on received acknowledgements.
rate_media (0.0 bps)
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Measured bitrate from the media encoder.
rate_media_median (0.0 bps)
Median value of rate_media, computed over more than 10s.
s_rtt (0.0s)
Smoothed RTT [s], computed with a similar method to that described
in [RFC6298].
rtp_queue_size (0 bits)
Sum of the sizes of RTP packets in queue.
rtp_size (0 byte)
Size of the last transmitted RTP packet.
loss_event_rate (0.0)
The estimated fraction of RTTs with lost packets detected.
4.1.2. Network congestion control
This section explains the network congestion control, it contains two
main functions:
o Computation of congestion window at the sender: Gives an upper
limit to the number of bytes in flight.
o 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.
SCReAM 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
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 packet with sequence number SN-3 was lost,
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the size of RTP packet with sequence number SN-3 will still be
considered in the computation of bytes_in_flight.
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 variable is reset to zero after a CWND update.
The feedback from the receiver is assumed to consist of the following
elements.
o A list of received RTP packets' sequence numbers.
o The wall clock timestamp corresponding to the received RTP packet
with the highest sequence number.
o Accumulated number of ECN-CE marked packets (n_ECN).
When the sender receives RTCP feedback, the qdelay is calculated as
outlined in [RFC6817]. A qdelay sample is obtained for each received
acknowledgement. No smoothing of the qdelay samples occur, however
some smoothing occurs anyway as the computation of the CWND is a low
pass filter function. A number of variables are updated as
illustrated by the pseudo code below, temporary variables are
appended with '_t'. As mentioned in Section 7 , calculation of the
proper congestion window and media bitrate may benefit from
additional optimizations for handling of very high and very low
bitrates, and from additional damping to handle periodic packet
bursts. Some such optimizations are implemented in
[SCReAM-CPP-implementation], but they do not form part of the
specification of SCReAM at this time.
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<CODE BEGINS>
update_variables(qdelay):
qdelay_fraction_t = qdelay/qdelay_target
# Calculate moving average
qdelay_fraction_avg = (1-QDELAY_WEIGHT)*qdelay_fraction_avg+
QDELAY_WEIGHT*qdelay_fraction_t
update_qdelay_fraction_hist(qdelay_fraction_t)
# Compute the average of the values in qdelay_fraction_hist
avg_t = average(qdelay_fraction_hist)
# R is an autocorrelation function of qdelay_fraction_hist,
# with the mean (DC component) removed, at lag K
# The subtraction of the scalar avg_t from
# qdelay_fraction_hist is performed element-wise
a_t = R(qdelay_fraction_hist-avg_t,1)/
R(qdelay_fraction_hist-avg_t,0)
# Calculate qdelay trend
qdelay_trend = min(1.0,max(0.0,a_t*qdelay_fraction_avg))
# Calculate a 'peak-hold' qdelay_trend, this gives a memory
# of congestion in the past
qdelay_trend_mem = max(0.99*qdelay_trend_mem, qdelay_trend)
<CODE ENDS>
The qdelay fraction is sampled every 50ms and the last 20 samples are
stored in a vector (qdelay_fraction_hist). This vector is used in
the computation of an qdelay trend that gives a value between 0.0 and
1.0 depending on the estimated congestion level. The prediction
coefficient 'a_t' has positive values if qdelay shows an increasing
or decreasing trend, thus an indication of congestion is obtained
before the qdelay target is reached. As a side effect, also the case
that qdelay decreases is taken as a sign of congestion, experiments
have however shown that this is beneficial as varying queue delay up
or down is an indication that the transmit rate is very close to the
path capacity.
The autocorrelation function 'R' is defined as follows. Let x be a
vector constituting N values, the biased autocorrelation function for
a given lag=k for the vector x is given by.
n=N-k
R(x,k) = SUM x(n)*x(n+k)
n=1
The prediction coefficient is further multiplied with
qdelay_fraction_avg to reduce sensitivity to increasing qdelay when
it is very small. The 50ms sampling is a simplification that could
have the effect that the same qdelay is sampled several times, this
does however not pose any problem as the vector is only used to
determine if the qdelay is increasing or decreasing. The
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qdelay_trend is utilized in the media rate control to indicate
incipient congestion and to determine when to exit from fast increase
mode. qdelay_trend_mem is used to enforce a less aggressive rate
increase after congestion events. The function
update_qdelay_fraction_hist(..) removes the oldest element and adds
the latest qdelay_fraction element to the qdelay_fraction_hist
vector.
4.1.2.1. Reaction to packets loss and ECN
A loss event is indicated if one or more RTP packets are declared
missing. The loss detection is described in Section 4.1.2.4. Once a
loss event is detected, further detected lost RTP packets SHOULD be
ignored for a full smoothed round trip time, the intention of this is
to limit the congestion window decrease to at most once per round
trip.
The congestion window back off due to loss events is deliberately a
bit less than is the case with e.g. TCP Reno. The reason is that
TCP is generally used to transmit whole files, which can be
translated to an infinite source bitrate. SCReAM on the other hand
has a source whose rate is limited to a value close to the available
transmit rate and often below that value, the effect of this is that
SCReAM has less opportunity to grab free capacity than a TCP based
file transfer. To compensate for this it is RECOMMENDED to let
SCReAM reduce the congestion window less than what is the case with
TCP when loss events occur.
An ECN event is detected if the n_ECN counter in the feedback report
has increased since the previous received feedback. Once an ECN
event is detected, the n_ECN counter is ignored for a full smoothed
round trip time, the intention of this is to limit the congestion
window decrease to at most once per round trip. 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
[I-D.ietf-tcpm-alternativebackoff-ecn] to enable ECN marking
thresholds lower than the corresponding packet drop thresholds.
4.1.2.2. Congestion window update
The update of the congestion window depends on whether loss or ECN-
marking or neither occurs. The pseudo code below describes actions
taken in case of the different events.
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<CODE BEGINS>
on congestion event(qdelay):
# Either loss or ECN mark is detected
in_fast_increase = false
if (is loss)
# Loss is detected
cwnd = max(MIN_CWND,cwnd*BETA_LOSS)
else
# No loss, so it is then an ECN mark
cwnd = max(MIN_CWND,cwnd*BETA_ECN)
end
adjust_qdelay_target(qdelay) #compensating for competing flows
calculate_send_window(qdelay,qdelay_target)
# When no congestion event
on acknowledgement(qdelay):
update_bytes_newly_acked()
update_cwnd(bytes_newly_acked)
adjust_qdelay_target(qdelay) #compensating for competing flows
calculate_send_window(qdelay, qdelay_target)
check_to_resume_fast_increase()
<CODE ENDS>
The methods are further described in detail below.
The congestion window update is based on qdelay, except for the
occurrence of loss events (one or more lost RTP packets in one RTT),
or ECN events, which was described earlier.
Pseudo code for the update of the congestion window is found below.
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<CODE BEGINS>
update_cwnd(bytes_newly_acked):
# In fast increase ?
if (in_fast_increase)
if (qdelay_trend >= QDELAY_TREND_TH)
# Incipient congestion detected, exit fast increase
in_fast_increase = false
else
# No congestion yet, increase cwnd if it
# is sufficiently used
# An additional slack of bytes_newly_acked is
# added to ensure that CWND growth occurs
# even when feedback is sparse
if (bytes_in_flight*1.5+bytes_newly_acked > cwnd)
cwnd = cwnd+bytes_newly_acked
end
return
end
end
# Not in fast increase phase
# off_target calculated as with LEDBAT
off_target_t = (qdelay_target - qdelay) / qdelay_target
gain_t = GAIN
# Adjust congestion window
cwnd_delta_t =
gain_t * off_target_t * bytes_newly_acked * MSS / cwnd
if (off_target_t > 0 &&
bytes_in_flight*1.25+bytes_newly_acked <= cwnd)
# No cwnd increase if window is underutilized
# An additional slack of bytes_newly_acked is
# added to ensure that CWND growth occurs
# even when feedback is sparse
cwnd_delta_t = 0;
end
# Apply delta
cwnd += cwnd_delta_t
# limit cwnd to the maximum number of bytes in flight
cwnd = min(cwnd, max_bytes_in_flight*MAX_BYTES_IN_FLIGHT_HEAD_ROOM)
cwnd = max(cwnd, MIN_CWND)
<CODE ENDS>
CWND is updated differently depending on whether the congestion
control is in fast increase state or not, as controlled by the
variable in_fast_increase.
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When in fast increase state, the congestion window is increased with
the number of newly acknowledged bytes as long as the window is
sufficiently used. Sparse feedback can potentially limit congestion
window growth, an additional slack is therefore added, given by the
number of newly acknowledged bytes.
The congestion window growth when in_fast_increase is false is
dictated by the relation between qdelay and qdelay_target, congestion
window growth is limited if the window is not used sufficiently.
SCReAM calculates the GAIN in a similar way to what is specified in
[RFC6817]. However, [RFC6817] specifies that the CWND increase is
limited by an additional function controlled by a constant
ALLOWED_INCREASE. This additional limitation is removed in this
specification.
Further the CWND is limited by max_bytes_in_flight and MIN_CWND. The
limitation of the congestion window by the maximum number of bytes in
flight over the last 5 seconds (max_bytes_in_flight) avoids possible
over-estimation of the throughput after for example, idle periods.
An additional MAX_BYTES_IN_FLIGHT_HEAD_ROOM allows for a slack, to
allow for a certain amount of media coder output rate variability.
4.1.2.3. Competing flows compensation
It is likely that a flow using SCReAM algorithm will have to share
congested bottlenecks with other flows that use a more aggressive
congestion control algorithm, examples are 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.
SCReAM takes care of such situations by adjusting the qdelay_target
when loss based flows are detected, as given by the pseudo code
below.
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<CODE BEGINS>
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 to avoid
# that the target delay is locked to high values for ever
if (new_target_t < QDELAY_TARGET_LO)
# Decrease target delay quickly as measured queueing
# 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)
<CODE ENDS>
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 of reduced bottleneck
bandwidth or that a competing flow has just entered. Thus, it
indicates that it is not safe to adjust the queue delay target.
A low qdelay_norm_var_t indicates that the queue delay is relatively
stable, the reason can be that the queue delay is low but it can also
be an indication that a competing flow is filling up the bottleneck
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to the limit where packet losses may 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 that 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 has a
certain risk to produce both false positive and negatives, The case
that self-inflicted congestion by the SCReAM 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. It can
however not be completely ruled out that this algorithm can fail.
Especially the false positives can be problematic as the end to end
delay can increase dramatically if the target queue delay is
increased by accident as a result of self-inflicted congestion.
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 can be 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 and it is in general 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 that
can lead to basically zero throughput if competing with other, loss
based, traffic.
4.1.2.4. Lost packet detection
Lost packet detection is based on the received sequence number list.
A reordering window SHOULD be applied to avoid that packet reordering
triggers loss events.
The reordering window is specified as a time unit, similar to the
ideas behind RACK (Recent ACKnowledgement) [I-D.ietf-tcpm-rack]. 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 a 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
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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 higher sequence number was acknowledged.
4.1.2.5. Send window calculation
The basic design principle behind packet transmission in SCReAM is 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:
o 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. The RTP queue should however be as short as possible, to
avoid that the end to end delay increases. To achieve that, the
media rate control takes the RTP queue size into account when the
target bitrate for the media is computed. A strict 'send only
when bytes in flight is less than the congestion window' rule can
lead to that the RTP queue grows simply because the send window is
limited, the effect of which would be that the target bitrate is
pushed down. The consequence of this 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 end effect is then that
the media bitrate increases very slowly or not at all.
o 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 of this is that
the acknowledgements are delayed with the result that the self-
clocking is temporarily halted, even though the forward path is
not congested.
The send window is adjusted depending on qdelay and its relation to
the qdelay target and the relation between the congestion window and
the number of bytes in flight. A strict rule is applied when qdelay
is higher than qdelay_target, to avoid further queue buildup in the
network. For cases when qdelay is lower than the qdelay_target, a
more relaxed rule is applied. This allows the bitrate to increase
quickly when no congestion is detected while still being able to give
a stable behavior in congested situations.
The send window is given by the relation between the adjusted
congestion window and the amount of bytes in flight according to the
pseudo code below.
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<CODE BEGINS>
calculate_send_window(qdelay, qdelay_target)
# send window is computed differently depending on congestion level
if (qdelay <= qdelay_target)
send_wnd = cwnd+MSS-bytes_in_flight
else
send_wnd = cwnd-bytes_in_flight
end
<CODE ENDS>
The send window is updated whenever an RTP packet is transmitted or
an RTCP feedback messaged is received.
4.1.2.6. Packet pacing
Packet pacing is used in order to mitigate coalescing i.e. that
packets are transmitted in bursts, with the increased risk of more
jitter and potentially increased packet loss. Packet pacing also
mitigates possible issues with queue overflow due to key-frame
generation in video coders. The time interval between consecutive
packet transmissions is enforced to be equal to or higher than t_pace
where t_pace is given by the equations below :
<CODE BEGINS>
pace_bitrate = max (RATE_PACE_MIN, cwnd* 8 / s_rtt)
t_pace = rtp_size * 8 / pace_bitrate
<CODE ENDS>
rtp_size is the size of the last transmitted RTP packet, s_rtt is the
smoothed round trip time. RATE_PACE_MIN is the minimum pacing rate.
4.1.2.7. Resuming fast increase
Fast increase can resume in order to speed up the bitrate increase in
case congestion abates. The condition to resume fast increase
(in_fast_increase = true) is that qdelay_trend is less than
QDELAY_TREND_LO for T_RESUME_FAST_INCREASE seconds or more.
4.1.2.8. Stream prioritization
The SCReAM algorithm makes a good distinction between network
congestion control and the media rate control. This is easily
extended to many streams, in which case 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 applied in
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[I-D.ietf-rmcat-coupled-cc] can be used. The implementation of
SCReAM [SCReAM-CPP-implementation] use credit based scheduling. In
credit based scheduling, credit is accumulated by queues as they wait
for service and are spent while the queues are being serviced. For
instance, if one queue is allowed to transmit 1000bytes, then a
credit of 1000bytes is allocated to the other unscheduled queues.
This principle can be extended to weighted scheduling in which case
the credit allocated to unscheduled queues depends on the relative
weights. The latter is also implemented in
[SCReAM-CPP-implementation].
4.1.3. Media rate control
The media rate control algorithm is executed at regular intervals
RATE_ADJUSTMENT_INTERVAL, with the exception of a prompt reaction to
loss events. The media rate control operates based on the size of
the RTP packet send queue and observed loss events. In addition,
qdelay_trend is also considered in the media rate control to reduce
the amount of induced network jitter.
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. A too
cautious setting leads to possible under-utilization of network
capacity leading to that the flow can become starved out by other
more opportunistic traffic. On the other hand, a too aggressive
setting leads to increased jitter.
The target_bitrate is adjusted depending on the congestion state.
The target bitrate can vary between a minimum value
(TARGET_BITRATE_MIN) and a maximum value (TARGET_BITRATE_MAX).
TARGET_BITRATE_MIN SHOULD be chosen to a low enough value to avoid
that RTP packets become queued up when the network throughput is
reduced. The sender SHOULD also be equipped with a mechanism that
discards RTP packets in cases where the network throughput becomes
very low and RTP packets are excessively delayed.
For the overall bitrate adjustment, two network throughput estimates
are computed :
o rate_transmit: The measured transmit bitrate.
o rate_ack: The ACKed bitrate, i.e. the volume of ACKed bits per
second.
Both estimates are updated every 200ms.
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The current throughput, current_rate, is computed as the maximum
value of rate_transmit and rate_ack. The rationale behind the use of
rate_ack in addition to rate_transmit is that rate_transmit is
affected also by the amount of data that is available to transmit,
thus a lack of data to transmit can be seen as reduced throughput
that can itself cause an unnecessary rate reduction. To overcome
this shortcoming; rate_ack is used as well. This gives a more stable
throughput estimate.
The rate change behavior depends on whether a loss or ECN event has
occurred and if the congestion control is in fast increase or not.
<CODE BEGINS>
# The target_bitrate is updated at a regular interval according
# to RATE_ADJUST_INTERVAL
on loss:
# Loss event detected
target_bitrate = max(BETA_R* target_bitrate, TARGET_BITRATE_MIN)
exit
on ecn_mark:
# ECN event detected
target_bitrate = max(BETA_ECN* target_bitrate, TARGET_BITRATE_MIN)
exit
ramp_up_speed_t = min(RAMP_UP_SPEED, target_bitrate/2.0)
scale_t = (target_bitrate - target_bitrate_last_max)/
target_bitrate_last_max
scale_t = max(0.2, min(1.0, (scale_t*4)^2))
# min scale_t value 0.2 as the bitrate should be allowed to
# increase at least slowly --> avoid locking the rate to
# target_bitrate_last_max
if (in_fast_increase = true)
increment_t = ramp_up_speed_t*RATE_ADJUST_INTERVAL
increment_t *= scale_t
target_bitrate += increment_t
else
current_rate_t = max(rate_transmit, rate_ack)
# Compute a bitrate change
delta_rate_t = current_rate_t*(1.0-PRE_CONGESTION_GUARD*
queue_delay_trend)-TX_QUEUE_SIZE_FACTOR *rtp_queue_size
# Limit a positive increase if close to target_bitrate_last_max
if (delta_rate_t > 0)
delta_rate_t *= scale_t
delta_rate_t =
min(delta_rate_t,ramp_up_speed_t*RATE_ADJUST_INTERVAL)
end
target_bitrate += delta_rate_t
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# Force a slight reduction in bitrate if RTP queue
# builds up
rtp_queue_delay_t = rtp_queue_size/current_rate_t
if (rtp_queue_delay_t > RTP_QDELAY_TH)
target_bitrate *= TARGET_RATE_SCALE_RTP_QDELAY
end
end
rate_media_limit_t =
max(current_rate_t, max(rate_media,rtp_rate_median))
rate_media_limit_t *= (2.0-qdelay_trend_mem)
target_bitrate = min(target_bitrate, rate_media_limit_t)
target_bitrate = min(TARGET_BITRATE_MAX,
max(TARGET_BITRATE_MIN,target_bitrate))
<CODE ENDS>
In case of a loss event the target_bitrate is updated and the rate
change procedure is exited. Otherwise the rate change procedure
continues. The rationale behind the rate reduction due to loss is
that a congestion window reduction will take effect, a rate reduction
pro actively avoids RTP packets being queued up when the transmit
rate decreases due to the reduced congestion window. A similar rate
reduction happens when ECN events are detected.
The rate update frequency is limited by RATE_ADJUST_INTERVAL, unless
a loss event occurs. The value is based on experimentation with real
life limitations in video coders taken into account
[SCReAM-CPP-implementation]. A too short interval is shown to make
the rate control loop in video coders more unstable, a too long
interval makes the overall congestion control sluggish.
When in fast increase state (in_fast_increase=true), the bitrate
increase is given by the desired ramp-up speed (RAMP_UP_SPEED) . The
ramp-up speed is limited when the target bitrate is low to avoid rate
oscillation at low bottleneck bitrates. The setting of RAMP_UP_SPEED
depends on preferences, a high setting such as 1000kbps/s makes it
possible to quickly get high quality media, this is however at the
expense of a increased jitter, which can manifest itself as e.g.
choppy video rendering.
When in_fast_increase is false, the bitrate increase is given by the
current bitrate and is also controlled by the estimated RTP queue and
the qdelay trend, thus it is sufficient that an increased congestion
level is sensed by the network congestion control to limit the
bitrate. The target_bitrate_last_max is updated when congestion is
detected.
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Finally the target_bitrate is enforced to be within the defined min
and max values.
The aware reader may notice the dependency on the qdelay in the
computation of the target bitrate, this manifests itself in the use
of the qdelay_trend. As these parameters are used also in the
network congestion control one may suspect some odd interaction
between the media rate control and the network congestion control,
this is in fact the case if the parameter PRE_CONGESTION_GUARD is set
to a high value. The use of qdelay_trend in the media rate control
is solely to reduce jitter, the dependency can be removed by setting
PRE_CONGESTION_GUARD=0, the effect is a somewhat faster rate increase
after congestion, at the expense of increased jitter in congested
situations.
4.2. SCReAM Receiver
The simple task of the SCReAM receiver is to feedback
acknowledgements of received packets and total ECN count to the
SCReAM sender, in addition, the receive time of the RTP packet with
the highest sequence number is echoed back. Upon reception of each
RTP packet the receiver MUST maintain enough information to send the
aforementioned values to the SCReAM sender via a 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 is described.
4.2.1. Requirements on feedback elements
The following feedback elements are REQUIRED for the basic
functionality in SCReAM.
o A list of received RTP packets. This list SHOULD be sufficiently
long to cover all received RTP packets. This list can be realized
with the Loss RLE report block in [RFC3611].
o A wall clock timestamp corresponding to the received RTP packet
with the highest sequence number is required in order to compute
the qdelay. This can be realized by means of the Packet Receipt
Times Report Block in [RFC3611]. begin_seq MUST be set to the
highest received (possibly wrapped around) sequence number,
end_seq MUST be set to begin_seq+1 % 65536. The timestamp clock
MAY be set according to [RFC3611] i.e. equal to the RTP timestamp
clock. Detailed individual packet receive times is not necessary
as SCReAM does currently not describe how this can be used.
The basic feedback needed for SCReAM involves the use of the Loss RLE
report block and the Packet Receipt Times block defined in Figure 2.
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|reserved | PT=XR=207 | length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| BT=2 | rsvd. | T=0 | block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| begin_seq | end_seq |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| chunk 1 | chunk 2 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| chunk n-1 | chunk n |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| BT=3 | rsvd. | T=0 | block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| begin_seq | end_seq |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Receipt time of packet begin_seq |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2: Basic feedback message for SCReAM, based on RFC3611
In a typical use case, no more than four Loss RLE chunks are needed,
thus the feedback message will be 44bytes. It is obvious from the
figure that there is a lot of redundant information in the feedback
message. A more optimized feedback format, including the additional
feedback elements listed below, could reduce the feedback message
size a bit.
Additional feedback elements that can improve the performance of
SCReAM are:
o Accumulated number of ECN-CE marked packets (n_ECN). This can for
instance be realized with the ECN Feedback Report Format in
[RFC6679]. The given feedback report format is actually a slight
overkill as SCReAM would do quite well with only a counter that
increments by one for each received packet with the ECN-CE code
point set. The more bulky format could nevertheless be useful for
e.g ECN black-hole detection.
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4.2.2. Requirements on feedback intensity
SCReAM benefits from a relatively frequent feedback. It is
RECOMMENDED that a SCReAM implementation follows the guidelines
below.
The feedback interval depends on the media bitrate. At low bitrates
it is sufficient with a feedback interval of 100 to 400ms, while at
high bitrates a feedback interval of roughly 20ms is to prefer, at
very high bitrates, even shorter feedback intervals MAY be needed in
order to keep the self-clocking in SCReAM working well. One piece of
evidence of a too sparse feedback is that the SCReAM implementation
cannot reach high bitrates, even in uncongested links. A more
frequent feedback might solve this issue.
The numbers above can be formulated as feedback interval function
that can be useful for the computation of the desired RTCP bandwidth.
The following equation expresses the feedback rate:
rate_fb = min(50,max(2.5,rate_media/10000))
rate_media is the RTP media bitrate expressed in [bits/s], rate_fb is
the feedback rate expressed in [packets/s]. Converted to feedback
interval we get:
fb_int = 1.0/min(50,max(2.5,rate_media/10000))
The transmission interval is not critical, this means that in the
case of multi-stream handling between two hosts, the feedback for two
or more SSRCs can be bundled to save UDP/IP overhead, the final
realized feedback interval SHOULD however not exceed 2*fb_int in such
cases meaning that a scheduled feedback transmission event should not
be delayed more that fb_int.
SCReAM works with AVPF regular mode, immediate or early mode is not
required by SCReAM but can nonetheless be useful for e.g RTCP
messages not directly related to SCReAM, 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].
5. Discussion
This section covers a few discussion points
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o Clock drift: SCReAM can suffer from the same issues with clock
drift as is the case with LEDBAT [RFC6817]. Section A.2 in
[RFC6817] however describes ways to mitigate issues with clock
drift.
o Support for alternate ECN semantics: This specification adopts the
proposal in [I-D.ietf-tcpm-alternativebackoff-ecn] to reduce the
congestion window less when ECN based congestion events are
detected. Future work on Low Loss Low Latency for Scalable
throughput (L4S) may lead to updates in a future RFC that
describes SCReAM support for L4S.
o A new RFC4585 transport layer feedback message could to be
standardized if the use of the already existing RTCP extensions as
described in Section 4.2 is not deemed sufficient.
o The target bitrate given by SCReAM depicts the bitrate including
RTP and 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 strictly necessary to make a 100% perfect compensation
for the overhead as the SCReAM algorithm will inherently
compensate for moderate errors. Under-compensation of the
overhead has the effect of increasing jitter while
overcompensation will have the effect of causing the bottleneck
link to become under-utilized.
6. Implementation status
[Editor's note: Please remove the whole section before publication,
as well reference to RFC 7942]
This section records the status of known implementations of the
protocol defined by this specification at the time of posting of this
Internet-Draft, and is based on a proposal described in [RFC7942].
The description of implementations in this section is intended to
assist the IETF in its decision processes in progressing drafts to
RFCs. Please note that the listing of any individual implementation
here does not imply endorsement by the IETF. Furthermore, no effort
has been spent to verify the information presented here that was
supplied by IETF contributors. This is not intended as, and MUST NOT
be construed to be, a catalog of available implementations or their
features. Readers are advised to note that other implementations MAY
exist.
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According to [RFC7942], "this will allow reviewers and working groups
to assign due consideration to documents that have the benefit of
running code, which may serve as evidence of valuable experimentation
and feedback that have made the implemented protocols more mature.
It is up to the individual working groups to use this information as
they see it".
6.1. OpenWebRTC
The SCReAM algorithm has been implemented in the OpenWebRTC project
[OpenWebRTC], an open source WebRTC implementation from Ericsson
Research. This SCReAM implementation is usable with any WebRTC
endpoint using OpenWebRTC.
o Organization : Ericsson Research, Ericsson.
o Name : OpenWebRTC gst plug-in.
o Implementation link : The GStreamer plug-in code for SCReAM can be
found at github repository [SCReAM-implementation] The wiki
(https://github.com/EricssonResearch/openwebrtc/wiki) contains
required information for building and using OpenWebRTC.
o Coverage : The code implements the specification in this memo.
The current implementation has been tuned and tested to adapt a
video stream and does not adapt the audio streams.
o Implementation experience : The implementation of the algorithm in
the OpenWebRTC has given great insight into the algorithm itself
and its interaction with other involved modules such as encoder,
RTP queue etc. In fact it proves the usability of a self-clocked
rate adaptation algorithm in the real WebRTC system. The
implementation experience has led to various algorithm
improvements both in terms of stability and design. The current
implementation use an n_loss counter for lost packets indication,
this is subject to change in later versions to a list of received
RTP packets.
o Contact : irc://chat.freenode.net/openwebrtc
6.2. A C++ Implementation of SCReAM
o Organization : Ericsson Research, Ericsson.
o Name : SCReAM.
o Implementation link : A C++ implementation of SCReAM is available
at[SCReAM-CPP-implementation]. The code includes full support for
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congestion control, rate control and multi stream handling, it can
be integrated in web clients given the addition of extra code to
implement the RTCP feedback and RTP queue(s). The code also
includes a rudimentary implementation of a simulator that allows
for some initial experiments. An additional experiment with
SCReAM in a remote control arrangement is also documented.
o Coverage : The code implements the specification in this memo.
o Contact : ingemar.s.johansson@ericsson.com
7. Suggested experiments
SCReAM has been evaluated in a number of different ways, most of the
evaluation has been in simulator. The OpenWebRTC implementation work
involved extensive testing with artificial bottlenecks with varying
bandwidths and using two different video coders (OpenH264 and VP9),
the experience of this lead to further improvements of the media rate
control logic.
Further experiments are preferably done by means of implementation in
real clients and web browsers. RECOMMENDED experiments are:
o Trials with various access technologies: EDGE/3G/4G, WiFi, DSL.
Some experiments have already been carried out with LTE access,
see e.g. [SCReAM-CPP-implementation] and
[SCReAM-implementation-experience]
o Trials with different kinds of media: Audio, Video, slide show
content. Evaluation of multi stream handling in SCReAM.
o Evaluation of functionality of competing flows compensation
mechanism: Evaluate how SCReAM performs with competing TCP like
traffic and to what extent the competing flows compensation causes
self-inflicted congestion.
o Determine proper parameters: A set of default parameters are given
that makes SCReAM work over a reasonably large operation range,
however for instance for very low or very high bitrates it may be
necessary to use different values for instance for the
RAMP_UP_SPEED.
o Experimentation with further improvements to the congestion window
and media bitrate calculation. [SCReAM-CPP-implementation]
implements some optimizations, not described in this memo, that
improve performance slightly. Further experiments are likely to
lead to more optimizations of the algorithm.
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8. Acknowledgements
We would like to thank the following persons for their comments,
questions and support during the work that led to this memo: Markus
Andersson, Bo Burman, Tomas Frankkila, Frederic Gabin, Laurits Hamm,
Hans Hannu, Nikolas Hermanns, Stefan Haakansson, Erlendur Karlsson,
Daniel Lindstroem, Mats Nordberg, Jonathan Samuelsson, Rickard
Sjoeberg, Robert Swain, Magnus Westerlund, Stefan Aalund. Many
additional thanks to RMCAT chairs Karen E. E. Nielsen and Mirja
Kuehlewind for patiently reading, suggesting improvements and also
for asking all the difficult but necessary questions. Thanks to
Stefan Holmer, Xiaoqing Zhu, Safiqul Islam and David Hayes for the
additional review of this document. Thanks to Ralf Globisch for
taking time to try out SCReAM in his challenging low bitrate use
cases, Robert Hedman for finding a few additional flaws in the
running code, and Gustavo Garcia and 'miseri' for code contributions.
9. IANA Considerations
There is currently no request to IANA
10. 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 is self-clocked, a
malicious middlebox can drop RTCP feedback packets and thus cause the
self-clocking in SCReAM to stall. This attack is however mitigated
by the minimum send rate maintained by SCReAM when no feedback is
received.
11. Change history
A list of changes:
o WG-12 to WG-13: IESG comments addressed
o WG-11 to WG-12: Review comments from Joel Halpern and Mirja
o WG-10 to WG-11: Review comments from Mirja
o WG-9 to WG-10: Minor edits
o WG-08 to WG-09: Updated based shepherd review by Martin
Stiemerling, Q-bit semantics are removed as this is superfluous
for the moment. Pacing and RTCP considerations are moved up from
the appendix, FEC discussion moved to discussion section.
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o WG-07 to WG-08: Avoid draft expiry
o WG-06 to WG-07: Updated based on WGLC review by David Hayes and
Safiqul Islam
o WG-05 to WG-06: Added list of suggested experiments
o WG-04 to WG-05: Congestion control and rate control simplified
somewhat
o WG-03 to WG-04: Editorial fixes
o WG-02 to WG-03: Review comments from Stefan Holmer and Xiaoqing
Zhu addressed, owd changed to qdelay for clarity. Added appendix
section with RTCP feedback requirements, including a suggested
basic feedback format based Loss RLE report block and the Packet
Receipt Times blocks in [RFC3611]. Loss detection added as a
section. Transmission scheduling and packet pacing explained in
appendix. Source quench semantics added to appendix.
o WG-01 to WG-02: Complete restructuring of the document. Moved
feedback message to a separate draft.
o WG-00 to WG-01 : Changed the Source code section to Implementation
status section.
o -05 to WG-00 : First version of WG doc, moved additional features
section to Appendix. Added description of prioritization in
SCReAM. Added description of additional cap on target bitrate
o -04 to -05 : ACK vector is replaced by a loss counter, PT is
removed from feedback, references to source code added
o -03 to -04 : Extensive changes due to review comments, code
somewhat modified, frame skipping made optional
o -02 to -03 : Added algorithm description with equations, removed
pseudo code and simulation results
o -01 to -02 : Updated GCC simulation results
o -00 to -01 : Fixed a few bugs in example code
12. References
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12.1. 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,
<https://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, <https://www.rfc-editor.org/info/rfc3550>.
[RFC3611] Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed.,
"RTP Control Protocol Extended Reports (RTCP XR)",
RFC 3611, DOI 10.17487/RFC3611, November 2003,
<https://www.rfc-editor.org/info/rfc3611>.
[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>.
[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>.
12.2. Informative References
[I-D.ietf-rmcat-coupled-cc]
Islam, S., Welzl, M., and S. Gjessing, "Coupled congestion
control for RTP media", draft-ietf-rmcat-coupled-cc-07
(work in progress), September 2017.
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[I-D.ietf-rmcat-wireless-tests]
Sarker, Z., Johansson, I., Zhu, X., Fu, J., Tan, W., and
M. Ramalho, "Evaluation Test Cases for Interactive Real-
Time Media over Wireless Networks", draft-ietf-rmcat-
wireless-tests-04 (work in progress), May 2017.
[I-D.ietf-tcpm-alternativebackoff-ecn]
Khademi, N., Welzl, M., Armitage, G., and G. Fairhurst,
"TCP Alternative Backoff with ECN (ABE)", draft-ietf-tcpm-
alternativebackoff-ecn-02 (work in progress), October
2017.
[I-D.ietf-tcpm-rack]
Cheng, Y., Cardwell, N., and N. Dukkipati, "RACK: a time-
based fast loss detection algorithm for TCP", draft-ietf-
tcpm-rack-02 (work in progress), March 2017.
[LEDBAT-delay-impact]
"Assessing LEDBAT's Delay Impact, IEEE communications
letters, vol. 17, no. 5, May 2013", May 2013,
<http://home.ifi.uio.no/michawe/research/publications/
ledbat-impact-letters.pdf>.
[OpenWebRTC]
"Open WebRTC project.", <http://www.openwebrtc.io/>.
[Packet-conservation]
"Congestion Avoidance and Control, ACM SIGCOMM Computer
Communication Review 1988", 1988.
[QoS-3GPP]
TS 23.203, 3GPP., "Policy and charging control
architecture", June 2011, <http://www.3gpp.org/ftp/specs/
archive/23_series/23.203/23203-990.zip>.
[RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
and K. Carlberg, "Explicit Congestion Notification (ECN)
for RTP over UDP", RFC 6679, DOI 10.17487/RFC6679, August
2012, <https://www.rfc-editor.org/info/rfc6679>.
[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>.
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[RFC7661] Fairhurst, G., Sathiaseelan, A., and R. Secchi, "Updating
TCP to Support Rate-Limited Traffic", RFC 7661,
DOI 10.17487/RFC7661, October 2015,
<https://www.rfc-editor.org/info/rfc7661>.
[RFC7942] Sheffer, Y. and A. Farrel, "Improving Awareness of Running
Code: The Implementation Status Section", BCP 205,
RFC 7942, DOI 10.17487/RFC7942, July 2016,
<https://www.rfc-editor.org/info/rfc7942>.
[SCReAM-CPP-implementation]
"C++ Implementation of SCReAM",
<https://github.com/EricssonResearch/scream>.
[SCReAM-implementation]
"SCReAM Implementation",
<https://github.com/EricssonResearch/
openwebrtc-gst-plugins>.
[SCReAM-implementation-experience]
"Updates on SCReAM : An implementation experience",
<https://www.ietf.org/proceedings/94/slides/
slides-94-rmcat-8.pdf>.
[TFWC] University College London, "Fairer TCP-Friendly Congestion
Control Protocol for Multimedia Streaming", December 2007,
<http://www-dept.cs.ucl.ac.uk/staff/M.Handley/papers/
tfwc-conext.pdf>.
Authors' Addresses
Ingemar Johansson
Ericsson AB
Laboratoriegraend 11
Luleaa 977 53
Sweden
Phone: +46 730783289
Email: ingemar.s.johansson@ericsson.com
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Zaheduzzaman Sarker
Ericsson AB
Laboratoriegraend 11
Luleaa 977 53
Sweden
Phone: +46 761153743
Email: zaheduzzaman.sarker@ericsson.com
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