wdiff draft-ietf-rmcat-scream-cc-02.txt draft-ietf-rmcat-scream-cc-03.txt

RMCAT WG I. Johansson
Internet-Draft Z. Sarker
Intended status: Experimental Ericsson AB
Expires: April 21, August 11, 2016 February 8, 2016 October 19, 2015

Self-Clocked Rate Adaptation for Multimedia
draft-ietf-rmcat-scream-cc-02
draft-ietf-rmcat-scream-cc-03

Abstract

This memo describes a rate adaptation algorithm for conversational
media services such as 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 LTE (Long
Term Evolution) 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 21, August 11, 2016.

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Table of Contents

1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. Wireless (LTE) access properties . . . . . . . . . . . . 3
1.2. Why is it a self-clocked algorithm? . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Overview of SCReAM Algorithm . . . . . . . . . . . . . . . . 4
3.1. Network Congestion Control . . . . . . . . . . . . . . . 7
3.2. Sender Transmission Control . . . . . . . . . . . . . . . 7
3.3. Media Rate Control . . . . . . . . . . . . . . . . . . . 7
4. Detailed Description of SCReAM . . . . . . . . . . . . . . . 8
4.1. SCReAM Sender . . . . . . . . . . . . . . . . . . . . . . 8
4.1.1. Constants and Parameter values . . . . . . . . . . . 8 9
4.1.1.1. Constants . . . . . . . . . . . . . . . . . . . . 8 9
4.1.1.2. State variables . . . . . . . . . . . . . . . . . 10
4.1.2. Network congestion control . . . . . . . . . . . . . 11 12
4.1.2.1. Updating bytes_newly_acked Congestion window update . . . . . . . . . . . 14
4.1.2.2. Updating congestion window . 15
4.1.2.2. Competing flows compensation . . . . . . . . . . 14 17
4.1.2.3. Compensation for competing flows Lost packets detection . . . . . . . . . . . . . 16 18
4.1.2.4. Send window calculation . . . . . . . . . . . . . 17 18
4.1.2.5. Resuming fast increase . . . . . . . . . . . . . 18 19
4.1.3. Media rate control . . . . . . . . . . . . . . . . . 18 19
4.1.3.1. FEC and packet overhead considerations . . . . . 22 23
4.2. SCReAM Receiver . . . . . . . . . . . . . . . . . . . . . 22 23
5. Discussion . . . . . . . . . . . . . . . . . . . . . . . . . 22 23
6. Implementation status . . . . . . . . . . . . . . . . . . . . 23
6.1. OpenWebRTC . . . . . . . . . . . . . . . . . . . . . . . 23 24
6.2. A C++ Implementation of SCReAM . . . . . . . . . . . . . 24 25
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 24 25
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 25
9. Security Considerations . . . . . . . . . . . . . . . . . . . 25
10. Change history . . . . . . . . . . . . . . . . . . . . . . . 25 26
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 25 26
11.1. Normative References . . . . . . . . . . . . . . . . . . 25 26
11.2. Informative References . . . . . . . . . . . . . . . . . 26 27
Appendix A. Additional features . information . . . . . . . . . . . . . . . 28 29
A.1. Stream prioritization . . . . . . . . . . . . . . . . . . 28 29
A.2. Computation of autocorrelation function . . . . . . . . . 28 29
A.3. Sender transmission control and packet pacing . . . . . . 30
A.4. RTCP feedback considerations . . . . . . . . . . . . . . 30
A.4.1. Requirements on feedback elements . . . . . . . . . . 30
A.4.2. Requirements on feedback intensity . . . . . . . . . 32
A.5. Q-bit semantics (source quench) . . . . . . . . . . . . . 33
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 29 34

1. Introduction

Congestion in the Internet is a reality and applications that are
deployed in the Internet must have congestion control schemes in
place not only for the robustness of the service that it provides but
also to ensure the function of the currently deployed Internet. As
the interactive realtime communication imposes a great deal of
requirements on the transport, a robust, efficient rate adaptation
for all access types is considered as an important part of
interactive realtime communications as the transmission channel
bandwidth may 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, must be both quick and be able to operate over a large span
in available channel bandwidth. This memo describes a solution,named
SCReAM, that is based on the self-clocking principle of TCP and uses
techniques similar to what is used in a new delay based rate
adaptation algorithm, LEDBAT [RFC6817].

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 may vary
considerably for instance in cases where the wireless terminal is
moving around.

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. In addition, the grace time, i.e. allowed reaction time
from Once the time that radio scheduler has reduced the congestion is detected until resource
allocation for a reaction bearer, an RMCAT flow in
terms of a that bearer needs to reduce
the sending rate reduction is effected, is generally very short, quite quickly (in one RTT) in
the order of one RTT (Round Trip Time). to avoid
excessive queuing delay or packet loss.

1.2. Why is it a self-clocked algorithm?

Self-clocked congestion control algorithm provides with a benefit
over the rate based counterparts in that the former consists of two
parts; the 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) and; the 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 has 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 only one
mechanism to adjust the sending rate and that rate. This makes it more problematic difficult to
reach the goal goals of high throughput and prompt reaction to congestion and also high throughput when channel
conditions are good. 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 [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].

In case of SCReAM, the receiver of the media sends the highest
received echoes a list of received RTP
packets and the timestamp of the RTP packet with the highest sequence
number back to the sender, sender in feedback packets, the sender keeps a
list of transmitted packets and packets, their respective sizes. sizes and the time they
were transmitted. This information is then used to determine the amount
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. This is how realizes the
packet conservation principle is
realized. principle. 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 along the transmission path.
This information is used to adjust the congestion window. The use of
LEDBAT ensures that the e2e end-to-end latency is kept low. The basic
functionality is quite simple, there are however a few steps to take
to make the concept work with conversational media. In a few words
they are:

o Congestion window validation techniques. These are similar in
action as the method described in [I-D.ietf-tcpm-newcwv]. [RFC7661]. Congestion window
validation ensures that the congestion window is limited by the
amount of actual bytes in flight, this is important especially in
the context of rate limited sources which is the case when video
is transmitted. 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 draft memo is shorter than in the reference, this to
avoid excessive delays in the cases where e.g. wireless throughput
has decreased during a period where the output bitrate has been
low. Furthermore, this draft 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 for quicker bitrate increase. 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 be however be resumed resume if the
congestion level is low, this to enable a reasonably quick rate
increase in case link throughput increases.

o A delay trend is computed for earlier detection of incipient
congestion and as a result it reduces jitter.

o Addition of media a media rate control function.

o Use of inflection points to calculate congestion window and media
rate to achieve reduced jitter.

o Adjustment of delay target for better performance when competing
with other loss based congestion controlled flows flows.

The above mentioned features will be described in more detail in
sections Section 3.1 to Section 3.3.

+---------------------------+
| Media encoder |
+---------------------------+
^ |
(3)| (1)|
| RTP
| V
| +-----------+
+---------+ | |
| Media | (2) | Queue |
| rate |<------| |
| control | |RTP packets|
+---------+ | |
+-----------+
|
|
(4)|
RTP
|
v
+------------+ +--------------+
| Network | (7) | Sender |
+-->| congestion |------>| Transmission |
| | control | | Control |
| +------------+ +--------------+
| |
| (6) |(5)
|-------------RTCP----------| RTP
| |
| v
+------------+
| UDP |
| socket |
+------------+

Figure 1: SCReAM sender functional view

The SCReAM algorithm constitutes mainly of three parts: network
congestion control, sender transmission control and media rate
adaptation.
control. All these three parts reside at the sender side. Figure 1 2
shows the functional overview of a SCReAM sender. The receiver side
algorithm is very simple in comparison as it only generates feedback
containing acknowledgements to received RTP
packets, loss count packets and ECN [RFC6679] 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.

The SCReAM congestion control method, uses techniques similar to
LEDBAT [RFC6817] to measure the one-way delay (OWD). The OWD can be expressed as the
estimated queuing delay. delay, also termed qdelay in
this memo for brevity. Similar to LEDBAT, it is not necessary to use
synchronized clocks in sender and receiver in order to compute the one way
queuing delay. 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.

The SCReAM sender calculates the congestion window based on the
feedback from the SCReAM receiver. The congestion window is allowed
to increase if the OWD qdelay is below a predefined qdelay target,
otherwise the congestion window decreases. The qdelay delay target
is typically set to 50-100ms. This ensures that the OWD queuing delay is
kept low on the average. 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

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, this has the effect that 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
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 up in the RTP packet queues at 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 the
case of a RAT (Radio Access Type) handover it may be necessary to
implement additional actions, such as discarding of encoded media
frames or frame skipping in order to ensure that the RTP queues are
drained quickly. quickly or simply that stale RTP packets are removed from the
queue. Frame skipping means that the frame rate is temporarily
reduced. Which method to use is a design consideration 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 a split between the network congestion control, sender
transmission control and the media rate
adaptation. control.

A SCReAM sender implements media rate control and a 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 sources source (a.k.a streams) are stream) is used. However,
multiple Multiple
media sources are also supported in the design, in that case the
sender transmission control will include a transmission scheduler.
The transmission scheduler can then enforce the priorities for the
different streams and then act like a coupled congestion controller
for multiple flows.

Media frames are encoded and forwarded to the RTP queue (1). The
media rate adaptation adapts to the size of the RTP queue (2) and
controls the media bitrate (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) takes care of
the transmission of RTP packets, to be written to the UDP socket (5).
In the general case all media must go through the sender transmission
controller and is allowed to be transmitted if 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.

4.1.1.1. Constants

The recommended values for the constants are deduced from
experimental results.

OWD_TARGET_LO
experimentals.

QDELAY_TARGET_LO (0.1s)
Target value for the minimum OWD

OWD_TARGET_HI qdelay.

QDELAY_TARGET_HI (0.4s)
Target value for the maximum OWD

OWD_WEIGHT qdelay.

QDELAY_WEIGHT (0.1)
Averaging factor for owd_fraction_avg qdelay_fraction_avg.

MAX_BYTES_IN_FLIGHT_HEAD_ROOM (1.1)
Headroom for the limitation of CWND CWND.

GAIN (1.0)
Gain factor for congestion window adjustment adjustment.

BETA_LOSS (0.6)
CWND scale factor due to loss event event.

BETA_ECN (0.8)
CWND scale factor due to ECN event event.

BETA_R (0.9)
Target rate scale factor due to loss event event.

MSS (1000 byte)
Maximum segment size = Max RTP packet size

BYTES_IN_FLIGHT_SLACK (10%)
Additional slack to the congestion window size.

RATE_ADJUST_INTERVAL (0.2s)
Interval between media bitrate adjustments adjustments.

TARGET_BITRATE_MIN
Min target bitrate [bps] [bps].

TARGET_BITRATE_MAX
Max target bitrate [bps] [bps].

RAMP_UP_SPEED (200kbps/s) (200000bps/s)
Maximum allowed rate increase speed speed.

PRE_CONGESTION_GUARD (0.0..0.2)
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 have however given
that 0.1 is a suitable value.

TX_QUEUE_SIZE_FACTOR (0.0..0.2) (0.0..2.0)
Guard factor against RTP queue buildup

OWD_TREND_LO buildup. This value may be subject
to tuning depending on e.g media coder characteristics, experiments
with H264 and VP8 have however given that 1.0 is a suitable value.

QDELAY_TREND_LO (0.2) Threshold value for owd_trend qdelay_trend.

T_RESUME_FAST_INCREASE Time span until fast increase can be resumed,
given that the owd_trend qdelay_trend is below OWD_TREND_LO QDELAY_TREND_LO.

4.1.1.2. State variables

owd_target (OWD_TARGET_LO)
OWD

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

owd_fraction_avg is allowed to
vary between QDELAY_TARGET_LO and QDELAY_TARGET_HI.

qdelay_fraction_avg (0.0)
EWMA filtered owd_fraction

owd_fraction_hist[20] fractional qdelay.

qdelay_fraction_hist[20] ({0,..,0})
Vector of the last 20 owd_fraction

owd_trend fractional qdelay samples.

qdelay_trend (0.0)
OWD
qdelay trend, indicates incipient congestion

owd_trend_mem congestion.

qdelay_trend_mem (0.0)
Low pass filtered version of owd_trend

owd_norm_hist[100] qdelay_trend.

qdelay_norm_hist[100] ({0,..,0})
Vector of the last 100 owd_norm normalized qdelay samples.

min_cwnd (2*MSS)
Minimum congestion window window.

in_fast_increase (true)
True if in fast increase state state.

cwnd (min_cwnd)
Congestion window window.

cwnd_last_max (1 byte)
Congestion window inflection point, i.e. the last known highest
cwnd. Used to limit cwnd increase speed close to the last known
congestion 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.
Reset after a CWND update

send_wnd (0)
Upper limit of to how many bytes that can currently be transmitted.
Updated when CWND cwnd is updated and when RTP packet is transmitted transmitted.

target_bitrate (0 bps)
Media target bitrate 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 point.

rate_transmit (0.0 bps)
Measured transmit bitrate bitrate.

rate_ack (0.0 bps)
Measured throughput based on received acknowledgements

rate_rtp acknowledgements.

rate_media (0.0 bps)
Measured bitrate from the media encoder

rate_rtp_median encoder.

rate_media_median (0.0 bps)
Median value of rate_rtp, rate_media, computed over more than 10s 10s.

s_rtt (0.0s)
Smoothed RTT [s], computed similar to method depicted in [RFC6298]

rtp_queue_size (0 bits)
Size of RTP packets in queue queue.

rtp_size (0 byte)
Size of the last transmitted RTP packet packet.

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 i.e. how many bytes that
have been transmitted but not yet acknowledged.

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.

Unlike TCP, SCReAM is not a byte oriented protocol, rather it is an
RTP packet oriented protocol. Thus a list of transmitted RTP packets
and their respective transmission times (wall-clock time) is kept for
further calculation. The feedback from congestion control is however based on
transmitted and acknowledged bytes.

SCReAM uses the receiver terminology "Bytes in flight" (bytes_in_flight) which
is assumed to consist computed as the sum of the sizes of the following
elements.

o The highest received RTP sequence number.

o The wall clock timestamp corresponding to packets ranging from
the received RTP packet most recently transmitted down to but not including
the acknowledged packet with he the highest sequence number.

o Accumulated This can
be translated to the difference between the highest transmitted byte
sequence number of lost and the highest acknowledged byte sequence number.
As an example: If RTP packets (n_loss).

o Accumulated packet with sequence number of ECN-CE marked packets (n_ECN).

When SN is transmitted
and the sender receives RTCP feedback, last acknowledgement indicates SN-5 as the OWD highest received
sequence number then bytes in flight is calculated computed as
outlined in [RFC6817] and a number the sum of variables are updated as
illustrated by the pseudo code below.

update_variables(owd):
owd_fraction = owd/owd_target
#calculate moving average
owd_fraction_avg = (1-OWD_WEIGHT)*owd_fraction_avg+
OWD_WEIGHT*owd_fraction
update_owd_fraction_hist(owd_fraction)
# R is an autocorrelation function
size of owd_fraction_hist
# at lag K
a = R(owd_fraction_hist,1)/R(owd_fraction_hist,0)
#calculate OWD trend
owd_trend = a*owd_fraction_avg
owd_trend_mem = max(0.99*owd_trend_mem, owd_trend)

The OWD fraction is sampled every 50ms 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, the last 20 samples are
stored in a vector (owd_fraction_hist). This vector is used size of RTP packet with sequence number SN-3 will
still be considered in the computation of an OWD trend that gives bytes_in_flight.

Furthermore, a value between 0.0 and 1.0
depending on the estimated congestion level. The prediction
coefficient 'a' has positive values if OWD shows an increasing trend,
thus an indication of congestion is obtained before the OWD target is
reached. The prediction coefficient variable bytes_newly_acked is further multiplied incremented with
owd_fraction_avg to reduce sensitivity to increasing OWD when OWD is
very small. The owd_trend is utilized in the media rate control to
indicate incipient congestion and to determine when to exit from fast
increase mode. owd_trend_mem is used to enforce a less aggressive
rate increase after congestion events. The function
update_owd_fraction_hist(..) removes the oldest element and adds the
latest owd_fraction element value
corresponding to how much the owd_fraction_hist vector.

A loss event is detected if the n_loss counter in the feedback highest sequence number has increased
since the previous received last feedback. Once a loss event is
detected, As an example: If the n_loss counter previous
acknowledgement indicated the highest sequence number N and the new
acknowledgement indicated N+3, then bytes_newly_acked is ignored for incremented
by a full smoothed round
trip time, value equal to the intention sum of this is to limit the congestion window
decrease to at most once per round trip.
The congestion window backoff due to loss events is deliberately a
bit less than is the case sizes of RTP packets with e.g TCP NewReno. The reason is sequence
number N+1, N+2 and N+3. Packets 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 are lost are also included,
which rate means that even though e.g packet N+2 was lost, its size is limited to a value close to the available
transmit rate and often below said value,
still included in the effect update of this bytes_newly_acked. The
bytes_newly_acked is that
SCReAM has less opportunity to grab free capacity than reset after a TCP based
file transfer. To compensate for this it is necessary to let SCReAM
reduce CWND update.

The feedback from the congestion window slightly less when loss events occur.

An ECN event receiver is detected if assumed to consist of the n_ECN counter following
elements. More details are found in the feedback report
has increased since the previous Appendix A.4.

o A list of received feedback. Once an ECN
event is detected, RTP packets.

o The wall clock timestamp corresponding to the n_ECN counter is ignored for a full smoothed
round trip time, received RTP packet
with the intention highest sequence number.

o Accumulated number of this is to limit ECN-CE marked packets (n_ECN).

When the congestion
window decrease to at most once per round trip. The congestion
window backoff due to an ECN event is deliberately smaller than if a
loss event occurs. This is inline with sender receives RTCP feedback, the idea qdelay is calculated as
outlined in
[Khademi_alternative_backoff_ECN] to enable ECN marking thresholds
lower than the corresponding packet drop thresholds.

The update of congestion window depends on whether a loss or ECN or
neither occurs. The pseudo code below describes actions taken in
case of different events.

on loss(owd):
in_fast_increase = false
cwnd_last_max = cwnd
cwnd = max(min_cwnd,cwnd*BETA_LOSS)
adjust_owd_target(owd)#compensating for competing flows
calculate_send_window(owd,owd_target)

on ECN(owd):
in_fast_increase = false
cwnd_last_max = cwnd
cwnd = max(min_cwnd,cwnd*BETA_ECN)
adjust_owd_target(owd)#compensating for competing flows
calculate_send_window(owd, owd_target)

# when no loss or ECN event is detected
on acknowledgement(owd):
update_bytes_newly_acked()
update_cwnd(bytes_newly_acked)
adjust_owd_target(owd) #compensating for competing flows
calculate_send_window(owd, owd_target)
check_to_resume_fast_increase()

The methods are further described in detail below.

4.1.2.1. Updating bytes_newly_acked

The 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.

4.1.2.2. Updating congestion window

The congestion window update is based on OWD, except for the
occurrence of loss or ECN events, which was described earlier. OWD
is obtained from the send and received timestamp of the RTP packets.
LEDBAT [RFC6817] explains the details of the computation of the OWD.
An OWD [RFC6817]. A qdelay sample is obtained for each received
acknowledgement. No smoothing of the OWD qdelay samples occur, however
some smoothing occurs anyway as the computation of the CWND is in
itself a low pass filter function.

Pseudo code for the update A number of variables are updated
as illustrated by the congestion window is found pseudo code below.

update_cwnd(bytes_newly_acked):

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)
# additional scaling factor to slow down closer to target
# The min scale factor is 0.2 to avoid that the congestion window
# growth is stalled
scale = max(0.2,min(1.0,(abs(cwnd-cwnd_last_max)/cwnd_i*4)^2))

# action depends on whether algorithm R is in fast increase
if (in_fast_increase)
if(owd_trend >= 0.2)
in_fast_increase=false
cwnd_i=cwnd
else
cwnd = cwnd + bytes_newly_acked*scale
return

# not in fast increase phase an autocorrelation function of qdelay_fraction_hist
# off_target calculated as with LEDBAT
off_target at lag K
a = (owd_target - owd) / owd_target

gain R(qdelay_fraction_hist,1)/R(qdelay_fraction_hist,0)
#calculate qdelay trend
qdelay_trend = GAIN
# adapt only increase based on scale
if (off_target > 0)
gain *= (1 - owd_trend/ 0.2) * scale

# increase/decrease the congestion window
# off_target can be positive or negative
cwnd += gain * off_target * bytes_newly_acked * MSS / cwnd min(1.0,max(0.0,a*qdelay_fraction_avg))
#calculate a 'peak-hold' qdelay_trend, this gives a memory
# Limit cwnd to the maximum number of bytes congestion in flight
cwnd = min(cwnd, max_bytes_in_flight*MAX_BYTES_IN_FLIGHT_HEAD_ROOM)
cwnd the past
qdelay_trend_mem = max(cwnd, MIN_CWND)

CWND max(0.99*qdelay_trend_mem, qdelay_trend)

The qdelay fraction is updated differently depending on whether sampled every 50ms and the congestion
control is last 20 samples are
stored in fast increase or not. A Boolean variable
in_fast_increase indicates if the congestion a vector (qdelay_fraction_hist). This vector is used in fast increase
state.

In fast increase state the congestion window is increased with
the
number computation of newly acknowledged bytes scaled by a scale factor an qdelay trend that
depends on the relation gives a value between CWND 0.0 and
1.0 depending on the last known maximum value
of CWND (cwnd_last_max). estimated congestion level. The prediction
coefficient 'a' has positive values if qdelay shows an increasing
trend, thus an indication of congestion window growth when
in_fast_increase is false is dictated by the relation between owd and
owd_target, also here obtained before the scale factor scale factor qdelay
target is applied to
limit the congestion window growth when cwnd gets close to
cwnd_last_max. reached. The scale factor as applied above makes the congestion window grow autocorrelation function 'R' is defined in
a similar way as
Appendix A.2. The prediction coefficient is the case further multiplied with the Cubic congestion control
algorithm.

SCReAM calculates the GAIN in a similar way
qdelay_fraction_avg to what is specified in
[RFC6817]. There are however a few differences.

o [RFC6817] specifies a constant GAIN, this specification however
limits the gain reduce sensitivity to increasing qdelay when CWND
it is increased dependent on near
congestion state very small. The 50ms sampling is a simplification and may have
the relation to the last known max CWND
value.

o [RFC6817] specifies effect that the CWND increased is limited by an
additional function controlled by a constant ALLOWED_INCREASE.
This additional limitation same qdelay is removed in sampled several times, this specification.

Further the CWND is limited by max_bytes_in_flight and min_cwnd. The
limitation of
however not a big issue as the congestion window by vector is only used for the maximum number
computation of bytes qdelay_trend. The qdelay_trend is utilized 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.

SCReAM uses the terminology "Bytes in flight (bytes_in_flight)" which control to indicate incipient congestion and to determine
when to exit from fast increase mode. qdelay_trend_mem is computed as used to
enforce a less aggressive rate increase after congestion events. The
function update_qdelay_fraction_hist(..) removes the sum of oldest element
and adds the sizes of latest qdelay_fraction element to the
qdelay_fraction_hist vector.

A loss event is indicated if one or more 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 are declared
missing. The loss detection is transmitted
and the last acknowledgement indicates SN-5 as the highest received
sequence number then bytes described in flight Section 4.1.2.3. Once a
loss event is computed as the sum of the
size of detected, further detected lost RTP packets with sequence number SN-4, SN-3, SN-2, SN-1 and
SN, it does not matter if are ignored
for instance packet with sequence number
SN-3 was lost, the size of RTP packet with sequence number SN-3 will
still be considered in a full smoothed round trip time, the computation intention of bytes_in_flight.

4.1.2.3. Compensation for competing flows

It this 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. SCReAM takes care of such situations
by adjusting
limit the owr_target.

adjust_owd_target(owd)
owd_norm = owd / OWD_TARGET_LOW
update_owd_norm_history(owd_norm)
# Compute variance
owd_norm_var = VARIATION(owd_norm_history(100))
# Compensation for competing traffic
if (owd_norm_var < 0.16)
# Compute average
owd_norm_avg = AVERAGE(owd_norm_history(20))
# Update target OWD
owd_target = owd_norm_avg*OWD_TARGET_LO*1.1
owd_target = min(OWD_TARGET_HI, owd_target)
owd_target = max(OWD_TARGET_LO, owd_target)

The owd_target is adjusted according congestion window decrease to the owd_norm_mean_sh whenever
owd_norm_var is below a given value. at most once per round trip.
The condition congestion window backoff due to update
owd_target is fulfilled if owd_norm_var < 0.16 (indicating that the
standard deviation loss events is deliberately a
bit less than 0.4).

owd_norm is the OWD divided by OWD_TARGET_LO. owd_norm_mean_sh case with e.g TCP NewReno. The reason is the
short term (last 20 samples) average of owd_norm. owd_norm_var that
TCP is
the variance of owd_norm over the last 100 samples.

4.1.2.4. Send window calculation

The basic design principle behind packet transmission in generally used to transmit whole files, which can be
translated to an infinite source bitrate. SCReAM on the other hand
has a source which rate is limited to a value close to
allow transmission only if the number available
transmit rate and often below said value, the effect of bytes in flight this is that
SCReAM has less opportunity to grab free capacity than
the congestion window. There are however two reasons why a TCP based
file transfer. To compensate for this strict
rule will not work optimally:

o Bitrate variations: The media frame size it is always varying to a
larger or smaller extent. A strict rule as the one given above
will have the effect that the media bitrate will have difficulties necessary to increase as let SCReAM
reduce the congestion window puts a too hard restriction
on the media frame size variation. This can lead to occasional
queuing of RTP packets in slightly less when loss events occur.

An ECN event is detected if the RTP packet queue that will further
prevent bitrate increase.

o Reverse (feedback) path congestion: Especially n_ECN counter in transport over
buffer-bloated networks, the one way delay in feedback report
has increased since the reverse
direction may jump due to congestion. The effect of this previous received feedback. Once an ECN
event is that
the acknowledgements are delayed with the result that detected, the self-
clocking n_ECN counter is temporarily halted, even though ignored for a full smoothed
round trip time, the forward path intention of this is
not congested.

The to limit the congestion
window is adjusted depending on OWD and its relation decrease to the OWD target. When OWD is greater than OWD target the at most once per round trip. The congestion
window enforces a strict rule that helps backoff due to prevent
further queue buildup. When OWD an ECN event is less deliberately smaller than or equal to OWD target
then an additional slack if a
loss event occurs. This is added inline with the idea outlined in
[Khademi_alternative_backoff_ECN] to enable ECN marking thresholds
lower than the corresponding packet drop thresholds.

The update of the congestion window that
reduces as congestion increases, BYTES_IN_FLIGHT_SLACK is depends on whether a maximum
allowed slack in percent. A large value increases the robustness to
bitrate variations loss or ECN
or neither occurs. The pseudo code below describes actions taken in
case of the source and congested feedback channel
issues. The possible drawback is increased delay different events.

on congestion event(qdelay):
# Either loss or packet ECN mark is detected
in_fast_increase = false
cwnd_last_max = cwnd
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)
adjust_qdelay_target(qdelay) #compensating for competing flows
calculate_send_window(qdelay,qdelay_target)

# when
forward path no congestion occurs. 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()

The adjusted congestion window
(cwnd_s) is used methods are further described in the send detail below.

4.1.2.1. Congestion window calculation. update

The send window is given by the relation between the adjusted congestion window and update is based on qdelay, except for the amount
occurrence of bytes loss events (one or more lost RTP packets in flight according to the
pseudo one RTT),
or ECN events, which was described earlier.

Pseudo code below.

calculate_send_window(owd, owd_target)
# compensate for backward congestion and bitrate variations
if (owd <= owd_target)
x_cwnd=1.0+BYTES_IN_FLIGHT_SLACK*(1.0-owd_trend/0.5)/100.0
cwnd_s = max(cwnd*x_cwnd, cwnd+MSS)

send_wnd = cwnd_s-bytes_in_flight

4.1.2.5. Resuming fast increase

Fast increase can be resumed in order to speed up the bitrate
increase in case congestion abates. update of the congestion window is found below.

update_cwnd(bytes_newly_acked):
# additional scaling factor to slow down closer to target
# The condition min scale factor is 0.2 to resume avoid that the congestion window
# growth is stalled when cwnd is close to cwnd_last_max
scale_t = max(0.2,min(1.0,(4*(cwnd-cwnd_last_max)/cwnd_i)^2))

# in fast increase (in_fast_increase ?
if (in_fast_increase)
if (qdelay_trend >= 0.2)
# incipient congestion detected, exit fast increase
in_fast_increase = true) false
cwnd_last_max = cwnd
else
# no congestion yet, increase cwnd
cwnd = cwnd+bytes_newly_acked*scale_t
return

# not in fast increase phase
# off_target calculated as with LEDBAT
off_target_t = (qdelay_target - qdelay) / qdelay_target

gain_t = GAIN
# adapt only increase based on scale
if (off_target_t > 0)
gain_t *= max(0.0, (1 - qdelay_trend/ 0.2)) * scale_t

# increase/decrease the congestion window
# off_target can be positive or negative
cwnd += gain_t * off_target_t * bytes_newly_acked * MSS / cwnd
# 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)

CWND is that owd_trend updated differently depending on whether the congestion
control is less than
OWD_TREND_LO for T_RESUME_FAST_INCREASE seconds in fast increase state or more.

4.1.3. Media rate control

The media rate control algorithm not, as indicated by the
variable in_fast_increase.

In fast increase state the congestion window is executed at regular intervals
RATE_ADJUSTMENT_INTERVAL, increased with the exception
number of newly acknowledged bytes scaled by a prompt reaction to
loss events. The media rate control operates based scale factor that
depends on the size of
the RTP packet send queue relation between CWND and observed loss events. In addition,
owd_trend is also considered in the media rate control, this to
reduce the amount last known maximum value
of induced network jitter. CWND (cwnd_last_max).

The role of the media rate control congestion window growth when in_fast_increase is to strike a reasonable balance
between a low amount of queuing in false is
dictated by the RTP queue relation between qdelay and qdelay_target, also here
a sufficient
amount of data to send in order scale factor is applied to keep limit the data path busy. A too
cautious setting leads congestion window growth when
cwnd gets close to possible under-utilization of network
capacity and that the flow is starved out by other, more
opportunistic traffic, on cwnd_last_max. The scale factor makes the other hand
congestion window grow in a too aggressive setting
leads to extra jitter.

A variable target_bitrate similar way as is adjusted depending on the case with the Cubic
congestion
state. The target bitrate can vary between a minimum value
(target_bitrate_min) and control algorithm i.e. a slow increase around the last
known maximum value (target_bitrate_max).

For value.

SCReAM calculates the overall bitrate adjustment, two network throughput estimates GAIN in a similar way to what is specified in
[RFC6817]. There are computed :

o rate_transmit: The measured transmit bitrate however a few differences.

o rate_ack: The ACKed bitrate, i.e. [RFC6817] specifies a constant GAIN, this specification however
limits the volume of ACKed bits per
time unit.

Both estimates are updated every 200ms.

The current throughput, current_rate, gain when CWND is computed as the maximum
value of rate_transmit increased dependent on near
congestion state and rate_ack. The rationale behind the use of
rate_ack in addition relation to rate_transmit is that rate_transmit is
affected also by the amount of data last known max CWND
value.

o [RFC6817] specifies that the CWND increase is available to transmit,
thus limited by an
additional function controlled by a lack of data to transmit can be seen as reduced throughput
that may itself cause an unnecessary rate reduction. To overcome
this shortcoming; rate_ack is used as well. constant ALLOWED_INCREASE.
This gives a more stable
throughput estimate.

Note that rate_ack additional limitation is updated removed in this specification.

Further the CWND is limited by bytes_newly_acked, which means that
even lost packets are regarded as acknowledged.

The rate change behavior depends on whether a loss event has
occurred, max_bytes_in_flight and if min_cwnd. The
limitation of the congestion control is window by the maximum number of bytes in fast increase or not.

# The target_bitrate
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.

4.1.2.2. Competing flows compensation

It is updated at likely that a regular interval according
# flow using SCReAM algorithm will have to RATE_ADJUST_INTERVAL

on loss:
target_bitrate_last_max share
congested bottlenecks with other flows that use a more aggressive
congestion control algorithm. SCReAM takes care of such situations
by adjusting the qdelay_target.

adjust_qdelay_target(qdelay)
qdelay_norm_t = target_bitrate
target_bitrate qdelay / QDELAY_TARGET_LOW
update_qdelay_norm_history(qdelay_norm_t)
# Compute variance
qdelay_norm_var_t = max(BETA_R* target_bitrate, TARGET_BITRATE_MIN)
exit VARIANCE(qdelay_norm_history(100))
# Compensation for competing traffic
if (in_fast_increase (qdelay_norm_var_t < 0.16)
# Compute average
qdelay_norm_avg_t = true)
scl_i AVERAGE(qdelay_norm_history(20))
# Update target qdelay
qdelay_target = (target_bitrate - target_bitrate_last_max)/
target_bitrate_last_max
increment qdelay_norm_avg_t*QDELAY_TARGET_LO*1.1
qdelay_target = RAMP_UP_SPEED*RATE_ADJUST_INTERVAL*
(1.0-min(1.0, owd_trend/0.2))
# Value 0.2 as min(QDELAY_TARGET_HI, qdelay_target)
qdelay_target = max(QDELAY_TARGET_LO, qdelay_target)

The qdelay_target is adjusted according to the bitrate should be allowed qdelay_norm_avg_t
whenever qdelay_norm_var_t is below a given value. The condition to increase
# at least slowly -->
update qdelay_target is fulfilled if qdelay_norm_var_t < 0.16.

A low qdelay_norm_avg_t value indicates that the qdelay does not
change rapidly. It is desired avoid locking the rate to
# target_bitrate_last_max
increment *= max(0.2, min(1.0, (scl_i*4)^2))
target_bitrate += increment
target_bitrate *= (1.0- PRE_CONGESTION_GUARD*owd_trend)
else
pre_congestion = min(1.0, max(0.0, owd_fraction_avg-0.3)/0.7)
pre_congestion += owd_trend
target_bitrate=current_rate*(1.0-PRE_CONGESTION_GUARD*
pre_congestion)-TX_QUEUE_SIZE_FACTOR *rtp_queue_size
end

rate_rtp_limit = max(br, max(rate_rtp,rtp_rate_median))
rate_rtp_limit *= (2.0-1.0*owd_trend_mem)
target_bitrate = min(target_bitrate, rate_rtp_limit)
target_bitrate = min(TARGET_BITRATE_MAX,
max(TARGET_BITRATE_MIN,target_bitrate))

In case of a loss event that the target_bitrate qdelay target
is updated increased due to self-congestion, indicated by a changing qdelay
and consequently an increased qdelay_norm_var_t. Still it should be
possible to increase the rate
change procedure is exited. Otherwise qdelay target if the rate change procedure
continues. An ECN event does not cause any action, qdelay continues to be
high. This is a simple function with a certain risk of both false
positives and negatives but it manages competing FTP flows reasonably
well at the reason same time as it has proven to
this avoid accidental increased
qdelay target in simulated LTE test cases.

4.1.2.3. Lost packets detection

Lost packets dectection is that based on the congestion received sequence number
list. A reordering window is reduced less due should be applied to ECN events
than avoid that packet
reordering triggers loss events, events.
The reordering window is specified as a time unit, similar to the effect
ideas behind RACK (Recent ACKnowledgement) [RACK]. The computation
of the reordering window is thus that made possible by means of a lost flag in
the expected additional list of transmitted RTP
queuing delay due to ECN events packets. This flag is so small set if the
received sequence number list indicates that an additional
decrease in media rate the given RTP packet is not warranted.

When in fast increase state,
missing. If a later feedback indicates that a previously lost marked
packet was indeed received, then the bitrate increase reordering window is given by updated to
reflect the
desired ramp-up speed (RAMP_UP_SPEED) and reordering delay. The reordering window is limited given by the relation
difference in time between the current bitrate and event that the last known max bitrate.
Furthermore an increased OWD trend limits packet was marked as
lost and the bitrate increase. The
setting of RAMP_UP_SPEED depends on preferences, a high setting such
as 1000kbps/s makes event that it possible to quickly gain high quality media,
this is however at the expense of a higher risk of jitter, which can
manifest itself was indicated as e.g. choppy video rendering.

When in_fast_increase is false, the bitrate increase successfully received.
Loss is detected if a given by the
current bitrate and RTP packet is also controlled not acknowledged within a
time window (indicated by the estimated reordering window) after an RTP queue and
the OWD trend, thus it packet
with higher sequence number was ackelowledged.

4.1.2.4. Send window calculation

The basic design principle behind packet transmission in SCReAM is sufficient that an increased congestion
level to
allow transmission only if the number of bytes in flight is sensed by less than
the network congestion control window. There are however two reasons why this strict
rule will not work optimally:

o Bitrate variations: The media frame size is always varying to limit a
larger or smaller extent. A strict rule as the
bitrate.

In one given above
will have the fast effect that the media bitrate will have difficulties
to increase phase an allowed increment is computed based on as the congestion level and window puts a too hard restriction
on the relation media frame size variation. This can lead to target_bitrate_last_max and occasional
queuing of RTP packets in the target_bitrate is reduced RTP packet queue that will further if congestion is detected.

If in_fast_increase is false then
prevent bitrate increase.

o Reverse (feedback) path congestion: Especially in transport over
buffer-bloated networks, the target_bitrate_last_max is
updated to one way delay in the current value reverse
direction may jump due to congestion. The effect of target_bitrate if in_fast_increase
was true the last time the bitrate was updated. Additionally, a pre-
congestion indicator this is computed and that
the rate is adjusted
accordingly.

In cases where input stimuli to acknowledgements are delayed with the media encoder result that the self-
clocking is static, for
instance in "talking head" scenarios, temporarily halted, even though the target bitrate forward path is
not
always fully utilized. This may cause undesirable oscillations in congested.

The send window is adjusted depending on qdelay and its relation to
the qdelay target bitrate in the cases where the link throughput is limited and the media coder input stimuli changes relation between static and varying.
To overcome this issue, the target bitrate is capped to be less than
a given multiplier of a median value of congetsion window and
the history number of media coder
output bitrates, rate_rtp_limit. bytes in flight. A multiplier strict rule is applied to
rate_rtp_limit, depending on congestion history. The target_bitrate
is then limited by this rate_rtp_limit.

Finally the target_bitrate when qdelay
is enforced higher than qdelay_target, to be within the defined min
and max values.

The vary reader may notice the dependency on the OWD avoid further queue buildup in the
computation of the target bitrate, this manifests itself in
network. For cases when qdelay is lower than the use
of qdelay_target, a
more relaxed rule is applied. This allows the owd_trend and owd_fraction_avg. As these parameters are used
also bitrate to increase
fast when no congestion is detected while still being able to give a
stable behavior in congested situations.

The send window is given by the network congestion control one may suspect that some odd
interaction relation between the media rate control adjusted
congestion window and the network congestion
control, this is amount of bytes in fact the case if flight according to the parameter
PRE_CONGESTION_GUARD is set to a high value.
pseudo code below.

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

The use of owd_trend send window is updated whenever an RTP packet is transmitted or
an RTCP feedback messaged is received. More details around sender
transmission control and owd_fraction_avg packet pacing is found in Appendix A.3.

4.1.2.5. 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.

4.1.3. Media rate control

The media rate control algorithm is solely to reduce
jitter, the dependency can be removed by setting
PRE_CONGESTION_GUARD=0, executed at regular intervals
RATE_ADJUSTMENT_INTERVAL, with the effect is exception of a somewhat faster prompt reaction to
loss events. The media rate increase
at control operates based on the expense size of more jitter.

4.1.3.1. FEC and packet overhead considerations

The target bitrate given by SCReAM depicts
the bitrate including RTP packet send queue and FEC overhead. Therefore it observed loss events. In addition,
qdelay_trend is necessary that also considered in the media encoder
takes rate control, this overhead into account when to
reduce the amount of induced network jitter.

The role of the media bitrate is set.
It rate control is not strictly necessary to make strike a 100% perfect compensation for
the overhead as the SCReAM algorithm will inherently compensate
moderate errors. Under-compensation for reasonable balance
between a low amount of queuing in the overhead has RTP queue and a sufficient
amount of data to send in order to keep the effect data path busy. A too
cautious setting leads to possible under-utilization of network
capacity and that the jitter will increase somewhat while overcompensation will
have flow is starved out by other, more
opportunistic traffic, on the effect that other hand a too aggressive setting
leads to extra jitter.

A variable target_bitrate is adjusted depending on the bottleneck link becomes under-utilized.

4.2. SCReAM Receiver congestion
state. The simple task of the SCReAM receiver is to feedback
acknowledgements of received packets, total loss count target bitrate can vary between a minimum value
(TARGET_BITRATE_MIN) and total ECN
count a maximum value (TARGET_BITRATE_MAX). The
target_bitrate_min should be chosen to a low enough value to avoid
that RTP packets are queued up when the SCReAM sender. Upon reception of each network throughput becomes
low. The sender should be equipped with a mechanism that discards
RTP packet packets in cases the
receiver will simply maintain enough information to send network throughput becomes very low and RTP
packets are excessively delayed.

For the
aforementioned values to 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 SCReAM sender via RTCP transport layer
feedback message. volume of ACKed bits per
time unit.

Both estimates are updated every 200ms.

The frequency current throughput, current_rate, is computed as the maximum
value of rate_transmit and rate_ack. The rationale behind the feedback message depends on use of
rate_ack in addition to rate_transmit is that rate_transmit is
affected also by the amount of data that is available RTCP bandwidth. The details to transmit,
thus a lack of data to transmit can be seen as reduced throughput
that may itself cause an unnecessary rate reduction. To overcome
this feedback shortcoming; rate_ack is given
in another document.

5. Discussion used as well. This section covers gives a few discussion points

o RTCP feedback overhead: SCReAM benefits from more stable
throughput estimate.

The rate change behavior depends on whether a relatively frequent
feedback. Experiments have shown that loss event has occurred
and if the congestion control is in fast increase or not.

# The target_bitrate is updated at a feedback rate roughly
equal regular interval according
# to RATE_ADJUST_INTERVAL

on loss:
target_bitrate_last_max = target_bitrate
target_bitrate = max(BETA_R* target_bitrate, TARGET_BITRATE_MIN)
exit

if (in_fast_increase = true)
scale_t = (target_bitrate - target_bitrate_last_max)/
target_bitrate_last_max
increment_t = RAMP_UP_SPEED*RATE_ADJUST_INTERVAL*
(1.0-min(1.0, qdelay_trend/0.2))
# Value 0.2 as the bitrate should be allowed to increase
# at least slowly --> avoid locking the frame rate gives a stable self-clocking and
robustness against loss to
# target_bitrate_last_max
increment_t *= max(0.2, min(1.0, (scale_t*4)^2))
target_bitrate += increment_t
target_bitrate *= (1.0- PRE_CONGESTION_GUARD*qdelay_trend)
else
current_rate_t = max(rate_transmit, rate_ack)
pre_congestion = min(1.0, max(0.0, qdelay_fraction_avg-0.3)/0.7)
pre_congestion += qdelay_trend
target_bitrate=current_rate_t*(1.0-PRE_CONGESTION_GUARD*
pre_congestion)-TX_QUEUE_SIZE_FACTOR *rtp_queue_size
end

rate_media_limit = max(br, max(rate_media,rtp_rate_median))
rate_media_limit *= (2.0-1.0*qdelay_trend_mem)
target_bitrate = min(target_bitrate, rate_media_limit)
target_bitrate = min(TARGET_BITRATE_MAX,
max(TARGET_BITRATE_MIN,target_bitrate))

In case of feedback. With a maximum bitrate of
1500kbps loss event the RTCP feedback overhead target_bitrate is in the range 10-15kbps with
reduced size RTCP [RFC5506], including IP updated and UDP framing, in
other words the RTCP overhead rate
change procedure is quite modest and should not pose
a problem in exited. Otherwise the general case. Other solutions may be required in
highly asymmetrical link capacity cases. Worth notice rate change procedure
continues. The rationale behind the rate reduction due to loss is
that
SCReAM can work with as low feedback rates as once every 200ms,
this however comes with a higher sensitivity to loss of feedback
and also congestion window reduction will take effect, a potential rate reduction in throughput.

o AVPF mode: The RTCP feedback is based on AVPF regular mode. The
SCReAM feedback is transmitted as reduced size RTCP so save
overhead, it is however required to
pro actively avoids that RTP packets are queued up when the transmit full compound RTCP at
regular intervals, this interval can be controlled by trr-int
depicted in [RFC4585].

o Clock drift: SCReAM can suffer from
rate decreases due to the same issues with clock
drift as is reduced congestion window. An ECN event
does not cause any action, the case with LEDBAT [RFC6817]. Section A.2 in said
RFC however describes ways reason to mitigate issues with clock drift.

6. Implementation status

[Editor's note: Please remove this is that the whole section before publication,
as well reference congestion
window is reduced less due to RFC 6982]

This section records ECN events than loss events, the status of known implementations of effect
is thus that the
protocol defined expected additional RTP queuing delay due to ECN
events is so small that an additional decrease in media rate is not
warranted.

The rate update frequency is limited by this specification at the time of posting of this
Internet-Draft, and RATE_ADJUST_INTERVAL, unless
a loss event occurs. The value is based on a proposal described in [RFC6982].
The description of implementations experimentation with real
life limitations in this section is intended video coders taken into account. A too short
interval has shown to
assist make the IETF in its decision processes video coder internal rate control loop
more unstable, a too long interval makes the overall congestion
control sluggish.

When in progressing drafts to
RFCs. Please note that fast increase state (in_fast_increase=true), the listing of any individual implementation
here does not imply endorsement bitrate
increase is given by the IETF. Furthermore, no effort
has been spent to verify the information presented here that was
supplied desired ramp-up speed (RAMP_UP_SPEED) and is
limited by IETF contributors. This the relation between the current bitrate and the last
known max bitrate. Furthermore an increased qdelay trend limits the
bitrate increase, an allowed increment is not intended as, computed based on the
congestion level (given by qdelay_trend) and must not
be construed the relation to be, a catalog
target_bitrate_last_max. The target_bitrate is reduced if congestion
is detected. The setting of available implementations or their
features. Readers are advised to note that other implementations may
exist.

According to [RFC6982], "this will allow reviewers and working groups
to assign due consideration RAMP_UP_SPEED depends on preferences, a
high setting such as 1000kbps/s makes it possible to documents that have quickly get high
quality media, this is however at the benefit expense of
running code, a higher risk of
jitter, which may serve can manifest itself as evidence of valuable experimentation e.g. choppy video rendering.

When in_fast_increase is false, the bitrate increase is given by the
current bitrate and feedback that have made is also controlled by the implemented protocols more mature.
It estimated RTP queue and
the qdelay trend, thus it is up to sufficient that an increased congestion
level is sensed by the individual working groups network congestion control to use this information as
they see it".

6.1. OpenWebRTC limit the
bitrate. The SCReAM algorithm has been implemented in target_bitrate_last_max is updated to the OpenWebRTC project
[OpenWebRTC], an open source WebRTC implementation from Ericsson
Research. This SCReAM implementation current value
of target_bitrate if in_fast_increase was true the last time the
bitrate was updated. Additionally, a pre-congestion indicator 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]
computed and the rate is waiting adjusted accordingly.

In cases where input stimuli to be merged with the master branch of OpebWebRTC repository
(https://github.com/EricssonResearch/openwebrtc/pull/413).
However, people are encouraged to have look at it and send
feedback. This wiki
(https://github.com/EricssonResearch/openwebrtc/wiki) contains
required information media encoder is static, for building
instance in "talking head" scenarios, the target bitrate is not
always fully utilized. This may cause undesirable oscillations in
the target bitrate in the cases where the link throughput is limited
and using OpenWebRTC. Note that
to get all the SCReAM related code media coder input stimuli changes between static and build them, one has to use varying.
To overcome this issue, the cerbero fork from DanielLindstrm' s repository
(https://github.com/DanielLindstrm/cerbero/tree/scream) instead target bitrate is capped to be less than
a given multiplier of
EricssonResearch fork a median value of cerbero.

o Coverage : The code implements [I-D.ietf-rmcat-scream-cc]. the history of media coder
output bitrates, rate_media_limit. A multiplier is applied to
rate_media_limit, depending on congestion history. The
current implementation has been tuned and tested
target_bitrate is then limited by this rate_media_limit.

Finally the target_bitrate is enforced to adapt a video
stream and does not adapt be within the audio streams.

o Implementation experience : defined min
and max values.

The implementation of aware reader may notice the algorithm dependency on the qdelay in the OpenWebRTC has given great insight into
computation of the algorithm target bitrate, this manifests itself
and its interaction with other involved modules such as encoder,
RTP queue etc. In fact it proves in the usability use
of a self-clocked
rate adaptation algorithm the qdelay_trend and qdelay_fraction_avg. As these parameters are
used also in the real WebRTC system. The
implementation experience has led to various algorithm
improvements both in terms of stability and design. For example,
improved network congestion control one may suspect that some
odd interaction between the media rate increase behavior control and removal of the ACK vector from the feedback message.

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 also
available which network
congestion control, this is aimed for doing quick
experiments[SCReAM-Cplusplus_Implementation]. This repository
also includes a rudimentary implementation of a simulator. This
code can be included in other simulators like NS-3.

o Coverage : The code implements [I-D.ietf-rmcat-scream-cc]

o Contact : ingemar.s.johansson@ericsson.com,
zaheduzzaman.sarker@ericsson.com

7. Acknowledgements

We would like to thank fact the following persons for their comments,
questions and support during case if 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 parameter
PRE_CONGESTION_GUARD is set to Karen and Mirja for patiently reading,
suggesting improvements a high value. The use of qdelay_trend
and also for asking all qdelay_fraction_avg in the difficult but
necessary questions.

8. IANA Considerations

A new RFC4585 transport layer feedback message needs media rate control is solely to be
standardized.

9. Security Considerations

The feedback reduce
jitter, the dependency can be vulnerable to attacks similar to those that can
affect TCP. It is therefore recommended that removed by setting
PRE_CONGESTION_GUARD=0, the RTCP feedback effect is a somewhat faster rate increase
at
least integrity protected.

10. Change history

A list of changes:

o WG-01 to WG-02: Complete restructuring the expense of more jitter.

4.1.3.1. FEC and packet overhead considerations

The target bitrate given by SCReAM depicts the document. Moved
feedback message bitrate including RTP
and FEC overhead. Therefore it is necessary that the media encoder
takes this overhead into account when the media bitrate is set.
It is not strictly necessary to make a separate draft.

o WG-00 to WG-01 : Changed 100% perfect compensation for
the Source code section to Implementation
status section.

o -05 to WG-00 : First version overhead as the SCReAM algorithm will inherently compensate
moderate errors. Under-compensation for the overhead has the effect
that the jitter will increase somewhat while overcompensation will
have the effect that the bottleneck link becomes under-utilized.

4.2. SCReAM Receiver

The simple task of WG doc, moved additional features
section the SCReAM receiver is to Appendix. Added description feedback
acknowledgements of prioritization received packets and total ECN count to the
SCReAM sender, in
SCReAM. Added description addition, the reveive time of additional cap on target bitrate

o -04 the RTP packet with
the highest sequence number is echoed back. Upon reception of each
RTP packet the receiver will simply maintain enough information to -05 : ACK vector
send the aforementioned values to the SCReAM sender via RTCP
transport layer feedback message. The frequency of the feedback
message depends on the available RTCP bandwidth. More details of the
feedback and the frequency is replaced by found in Appendix A.4.

5. Discussion

This section covers a loss counter, PT is
removed from feedback, references to source code added few discussion points

o -03 to -04 : Extensive changes due to review comments, code
somewhat modified, frame skipping made optional

o -02 to -03 : Added algorithm description Clock drift: SCReAM can suffer from the same issues with equations, removed
pseudo code and simulation results

o -01 clock
drift as is the case with LEDBAT [RFC6817]. Section A.2 in said
RFC however describes ways to -02 : Updated GCC simulation results

o -00 mitigate issues with clock drift.

6. Implementation status

[Editor's note: Please remove the whole section before publication,
as well reference to -01 : Fixed RFC 6982]

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 few bugs proposal described in example code

11. References

11.1. Normative References

[RFC2119] Bradner, S., "Key words for use [RFC6982].
The description of implementations in RFCs this section is intended to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>.

[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>.

[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C.,
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 J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
DOI 10.17487/RFC4585, July 2006,
<http://www.rfc-editor.org/info/rfc4585>.

[RFC5506] Johansson, I. must not
be construed to be, a catalog of available implementations or their
features. Readers are advised to note that other implementations may
exist.

According to [RFC6982], "this will allow reviewers and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities working groups
to assign due consideration to documents that have the benefit of
running code, which may serve as evidence of valuable experimentation
and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
2009, <http://www.rfc-editor.org/info/rfc5506>.

[RFC6298] Paxson, V., Allman, M., Chu, J., 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 M. Sargent,
"Computing TCP's Retransmission Timer", RFC 6298,
DOI 10.17487/RFC6298, June 2011,
<http://www.rfc-editor.org/info/rfc6298>.

[RFC6817] Shalunov, S., Hazel, G., Iyengar, J., using OpenWebRTC.

o Coverage : The code implements [I-D.ietf-rmcat-scream-cc]. The
current implementation has been tuned and M. Kuehlewind,
"Low Extra Delay Background Transport (LEDBAT)", RFC 6817,
DOI 10.17487/RFC6817, December 2012,
<http://www.rfc-editor.org/info/rfc6817>.

11.2. Informative References

[I-D.ietf-rmcat-app-interaction]
Zanaty, M., Singh, V., Nandakumar, S., tested to adapt a video
stream and Z. Sarker, "RTP
Application Interaction 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 also
available [SCReAM-Cplusplus_Implementation] The code includes full
support for 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. 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 Coverage : The code implements [I-D.ietf-rmcat-scream-cc]

o Contact : ingemar.s.johansson@ericsson.com

7. 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 chairs Karen and Mirja for patiently reading,
suggesting improvements and also for asking all the difficult but
necessary questions. Thanks to Stefan Holmer and Xiaoqing Zhu for
the review.

8. IANA Considerations

A new RFC4585 transport layer feedback message needs to be
standardized.

9. 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.

10. Change history

A list of changes:

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

11. References

11.1. Normative References

[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>.

[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
DOI 10.17487/RFC4585, July 2006,
<http://www.rfc-editor.org/info/rfc4585>.

[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
2009, <http://www.rfc-editor.org/info/rfc5506>.

[RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent,
"Computing TCP's Retransmission Timer", RFC 6298,
DOI 10.17487/RFC6298, June 2011,
<http://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,
<http://www.rfc-editor.org/info/rfc6817>.

11.2. Informative References

[I-D.ietf-rmcat-app-interaction]
Zanaty, M., Singh, V., Nandakumar, S., and Z. Sarker, "RTP
Application Interaction with Congestion Control", draft-
ietf-rmcat-app-interaction-01 (work in progress), October
2014.

[I-D.ietf-rmcat-cc-codec-interactions]
Zanaty, M., Singh, V., Nandakumar, S., and Z. Sarker,
"Congestion Control and Codec interactions in RTP
Applications", draft-ietf-rmcat-cc-codec-interactions-01
(work in progress), October 2015.

[I-D.ietf-rmcat-coupled-cc]
Islam, S., Welzl, M., and S. Gjessing, "Coupled congestion
control for RTP media", draft-ietf-rmcat-coupled-cc-00
(work in progress), September 2015.

[I-D.ietf-rmcat-scream-cc]
Johansson, I. and Z. Sarker, "Self-Clocked Rate Adaptation
for Multimedia", draft-ietf-rmcat-scream-cc-02 (work in
progress), October 2015.

[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-01 (work in progress), November 2015.

[Khademi_alternative_backoff_ECN]
"TCP Alternative Backoff with ECN (ABE)",
<https://tools.ietf.org/html/draft-khademi-
alternativebackoff-ecn-00>.

[OpenWebRTC]
"Open WebRTC project.", <http://www.openwebrtc.io/>.

[PACKET_CONSERVATION]
"Congestion Avoidance and Control", 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>.

[RACK] "RACK: a time-based fast loss detection algorithm for
TCP", <https://http://tools.ietf.org/id/
draft-cheng-tcpm-rack-00.txt>.

[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,
<http://www.rfc-editor.org/info/rfc3611>.

[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, <http://www.rfc-editor.org/info/rfc6679>.

[RFC6982] Sheffer, Y. and A. Farrel, "Improving Awareness of Running
Code: The Implementation Status Section", RFC 6982,
DOI 10.17487/RFC6982, July 2013,
<http://www.rfc-editor.org/info/rfc6982>.

[RFC7661] Fairhurst, G., Sathiaseelan, A., and R. Secchi, "Updating
TCP to Support Rate-Limited Traffic", RFC 7661,
DOI 10.17487/RFC7661, October 2015,
<http://www.rfc-editor.org/info/rfc7661>.

[SCReAM-Cplusplus_Implementation]
"C++ Implementation of SCReAM",
<https://github.com/EricssonResearch/scream>.

[SCReAM-Implementation]
"SCReAM Implementation",
<https://github.com/EricssonResearch/openwebrtc-gst-
plugins>.

[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>.

Appendix A. Additional information

A.1. Stream prioritization

The SCReAM algorithm makes a good distinction between network
congestion control and the media rate control, an RTP queue queues up
RTP packets pending transmission. 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
[I-D.ietf-rmcat-coupled-cc] can be used. The implementation of
SCReAM use something that is referred to as credit based scheduling.
Credit based scheduling is for instance implemented in IEEE 802.17.
The short description is that credit is accumulated by queues as they
wait for service and are spent while the queues are being services.

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 weight
allocation.

A.2. Computation of autocorrelation function

The autocorrelation function is computed over a vector of values.

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

A.3. Sender transmission control and packet pacing

RTP packet transmission is allowed whenever the size of the next RTP
packet in the sender queue is less than or equal to send window. As
explained in Section 4.1.2.4 the send window is updated whenever an
RTP packet is transmitted or RTCP feedback is received, the packet
transmission rate is however restricted by means of 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 is enforced when qdelay_fraction_avg is greater than
0.1. The time interval between consecutive packet transmissions is
then enforced to equal or higher than t_pace where t_pace is given by
the equations below.

pace_bitrate = max (50000, cwnd* 8 / s_rtt)

t_pace = rtp_size * 8 / pace_bitrate

rtp_size is the size of the last transmitted RTP packet

A.4. RTCP feedback considerations

This section describes the requrements on the RTCP feedback to make
SCReAM function well. Parts of this section may be moved to a
separate draft. First is described the requrements on the feedback
elements, second is decribed the requirements on the feedback
intensity to keep SCReAM self-clocking and rate control loops
function properly.

A.4.1. Requirements on feedback elements

SCReAM requires the following elements for its basic functionality,
i.e only including features that are sctrictly necessary in order to
make SCReAM function. ECN is not included as basic functionality as
it regarded as an additional feature that is not strickly necessary
even though it can improve quality of experience quite considerably.

o A list of received RTP packets. This list should be suffciently
long to cover all received RTP packets. This list may 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 queueing delay. This can be realized by means of the Packet
Receipt Times Report Block in [RFC3611]. begin_seq should be set
to the highest received (possibly wrapped around) sequence number,
end_seq should be set to begin_seq+1 % 65536. The timestamp clock
may be set according to the specification 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.

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

In a typical use case, no more than four Loss RLE chunks should be
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, should 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 Congestion Control", draft-
ietf-rmcat-app-interaction-01 (work the ECN Feedback Report Format in progress), October
2014.

[I-D.ietf-rmcat-cc-codec-interactions]
Zanaty, M., Singh, V., Nandakumar, S.,
[RFC6679]. The given feedback report format is actually a slight
overkill as SCReAM would do quite well with only an 8 bit counter
that increments by one for each received packet with the ECE-CE
code point set. The more bulky format may be nevertheless be
useful for e.g ECN black-hole detection.

o Source quench bit (Q): Makes it possible to request the sender to
reduce its congestion window. This is useful if WebRTC media is
received from many hosts and Z. Sarker,
"Congestion Control it becomes necessary to balance the
bitrates between the streams. This can currently not be realized
with any standardized feedback format.

A.4.2. Requirements on feedback intensity

SCReAM benefits from a relatively frequent feedback. Experiments
have shown that a feedback rate roughly equal to the frame rate gives
a stable self-clocking and Codec interactions in RTP
Applications", draft-ietf-rmcat-cc-codec-interactions-01
(work robustness against loss of feedback. With
a maximum bitrate of 1500kbps the RTCP feedback overhead is in progress), October 2015.

[I-D.ietf-rmcat-coupled-cc]
Islam, S., Welzl, M., the
range 10-15kbps with reduced size RTCP [RFC5506], including IP and S. Gjessing, "Coupled congestion
control for RTP media", draft-ietf-rmcat-coupled-cc-00
(work
UDP framing and a reasonable compact RTCP feedback format. In other
words the RTCP overhead is quite modest and should not pose a problem
in progress), September 2015.

[I-D.ietf-rmcat-scream-cc]
Johansson, I. the general case. Other solutions may be required in highly
asymmetrical link capacity cases. Worth notice is that SCReAM can
work with as low feedback rates as once every 200ms, this however
comes with a higher sensitivity to loss of feedback and Z. Sarker, "Self-Clocked Rate Adaptation
for Multimedia", draft-ietf-rmcat-scream-cc-01 (work also a
potential reduction in
progress), July 2015.

[I-D.ietf-rmcat-wireless-tests]
Sarker, Z. and I. Johansson, "Evaluation Test Cases throughput.

SCReAM works with AVPF regular mode, immediate or early mode is not
required by SCReAM but may nontheless be useful for
Interactive Real-Time Media over Wireless Networks",
draft-ietf-rmcat-wireless-tests-00 (work e.g CCM messages
specified in progress),
June 2015.

[I-D.ietf-tcpm-newcwv]
Fairhurst, G., Sathiaseelan, A., and R. Secchi, "Updating
TCP [RFC4585]. It is recommended to support Rate-Limited Traffic", draft-ietf-tcpm-
newcwv-13 (work use reduced size RTCP
[RFC5506]where regular full compound RTCP transmission is controlled
by trr-int as described in progress), June 2015.

[Khademi_alternative_backoff_ECN]
"TCP Alternative Backoff [RFC4585].

The feedback interval is somewhat depending on the media bitrate. At
low bitrates it is sufficient with ECN (ABE)",
<https://tools.ietf.org/html/draft-khademi-
alternativebackoff-ecn-00>.

[OpenWebRTC]
"Open WebRTC project.", <http://www.openwebrtc.io/>.

[PACKET_CONSERVATION]
"Congestion Avoidance and Control", 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>.

[RFC6982] Sheffer, Y. and A. Farrel, "Improving Awareness a feedback interval of Running
Code: The Implementation Status Section", RFC 6982,
DOI 10.17487/RFC6982, July 2013,
<http://www.rfc-editor.org/info/rfc6982>.

[SCReAM-Cplusplus_Implementation]
"C++ Implementation 100 to
200ms, while at high bitrates a feedback interval of SCReAM",
<https://github.com/EricssonResearch/scream>.

[SCReAM-Implementation]
"SCReAM Implementation",
<https://github.com/DanielLindstrm/openwebrtc-gst-
plugins/tree/scream>.

[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>.

Appendix A. Additional features ~20ms is to
prefer.

This section describes additional features. They are leads to a feedback rate according to the following equation

rate_fb = min(50,max(10,rate_media/20000))
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(10,rate_media/20000))

The transmission interval is not required
for critical, this means that in the basic functionality
case of SCReAM but multi-stream handling between two hosts, the feedback for two
or more SSRCs can improve performance be bundled to save UDP/IP overhead, the final
realized feedback interval should however not exceed 2*fb_int in
certain scenarios and topologies.

A.1. Stream prioritization such
cases meaning that a scheduled feedback transmission event should not
be delayed more that fb_int.

A.5. Q-bit semantics (source quench)

The SCReAM algorithm makes Q bit in the feedback is set by a good distinction between network receiver to signal that the
sender should reduce the bitrate. The sender will in response to
this reduce the congestion control and window with the media consequence that the video
bitrate decreases. A typical use case for source quench is when a
receiver receives streams from sources located at different hosts and
they all share a common bottleneck, typically it is difficult to
apply any rate control, an RTP queue queues up
RTP packets pending transmission. This distribution signaling between the sending hosts. The
solution is easily extended then that the receiver sets the Q bit in the feedback to many
streams,
the sender that should reduce its rate, if the streams share a common
bottleneck then the released bandwidth due to the reduction of the
congestion window for the flow that had the Q bit set in which case RTP packets from two the feedback
will be grabbed by the other flows that did not have the Q bit set.
This is ensured by the opportunistic behavior of SCReAM's congestion
control. The source quench will have no or more RTP queues are
scheduled at little effect if the
flows do not share the same bottleneck.

The reduction in congestion window is proportional to the amount of
SCReAM RTCP feedback with the rate permitted by Q bit set, the network congestion control. below steps outline how
the sender should react to RTCP feedback with the Q bit set. The scheduling can be
reduction is done by means once per RTT. Let :

o n = Number of a few different scheduling
regimes. For example the method applied received RTCP feedback messages in
[I-D.ietf-rmcat-coupled-cc] can be used. The implementation one RTT

o n_q = Number of
SCReAM use something that is referred to as credit based scheduling.
Credit based scheduling is for instance implemented received RTCP feedback messages in IEEE 802.17. one RTT, with Q
bit set.

The short description new congestion window is then expressed as:

cwnd = max(MIN_CWND, cwnd*(1.0-0.5* n_q /n))

Note that credit CWND is accumulated by queues as they
wait adjusted at most once per RTT. Furthermore The
CWND increase should be inhibited for service and are spent while the queues are being services.

For instance, if one queue is allowed to transmit 1000bytes, then RTT if CWND has been
decreased as a
credit result of 1000bytes is allocated to Q bits set in the other unscheduled queues.
This principle can be extended to weighted scheduling feedback.

The required intensity of the Q-bit set in which case the credit allocated feedback in order to unscheduled queues
achieve a given rate distribution depends on the weight
allocation.

A.2. Computation of autocorrelation function many factors such as
RTT, video source material etc. The autocorrelation function is computed over a vector of values.

Let x be a vector constituting N values, receiver thus need to monitor
the autocorrelation function
for a given lag=k for change in the vector x is given by .

n=N-k
R(x,k) = SUM x(n)*x(n+k)
n=1

Figure 2: Autocorrelation function received video bitrate on the different streams and
adjust the intensity of the Q-bit accordingly.

Authors' Addresses

Ingemar Johansson
Ericsson AB
Laboratoriegraend 11
Luleaa 977 53
Sweden

Phone: +46 730783289
Email: ingemar.s.johansson@ericsson.com

Zaheduzzaman Sarker
Ericsson AB
Laboratoriegraend 11
Luleaa 977 53
Sweden

Phone: +46 761153743
Email: zaheduzzaman.sarker@ericsson.com