draft-ietf-rmcat-scream-cc-10.txt   draft-ietf-rmcat-scream-cc-11.txt 
RMCAT WG I. Johansson RMCAT WG I. Johansson
Internet-Draft Z. Sarker Internet-Draft Z. Sarker
Intended status: Experimental Ericsson AB Intended status: Experimental Ericsson AB
Expires: January 19, 2018 July 18, 2017 Expires: April 12, 2018 October 9, 2017
Self-Clocked Rate Adaptation for Multimedia Self-Clocked Rate Adaptation for Multimedia
draft-ietf-rmcat-scream-cc-10 draft-ietf-rmcat-scream-cc-11
Abstract Abstract
This memo describes a rate adaptation algorithm for conversational This memo describes a rate adaptation algorithm for conversational
media services such as video. The solution conforms to the packet media services such as video. The solution conforms to the packet
conservation principle and uses a hybrid loss and delay based conservation principle and uses a hybrid loss and delay based
congestion control algorithm. The algorithm is evaluated over both congestion control algorithm. The algorithm is evaluated over both
simulated Internet bottleneck scenarios as well as in a Long Term simulated Internet bottleneck scenarios as well as in a Long Term
Evolution (LTE) system simulator and is shown to achieve both low Evolution (LTE) system simulator and is shown to achieve both low
latency and high video throughput in these scenarios. latency and high video throughput in these scenarios.
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on January 19, 2018. This Internet-Draft will expire on April 12, 2018.
Copyright Notice Copyright Notice
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
skipping to change at page 2, line 23 skipping to change at page 2, line 23
3. Overview of SCReAM Algorithm . . . . . . . . . . . . . . . . 4 3. Overview of SCReAM Algorithm . . . . . . . . . . . . . . . . 4
3.1. Network Congestion Control . . . . . . . . . . . . . . . 7 3.1. Network Congestion Control . . . . . . . . . . . . . . . 7
3.2. Sender Transmission Control . . . . . . . . . . . . . . . 8 3.2. Sender Transmission Control . . . . . . . . . . . . . . . 8
3.3. Media Rate Control . . . . . . . . . . . . . . . . . . . 8 3.3. Media Rate Control . . . . . . . . . . . . . . . . . . . 8
4. Detailed Description of SCReAM . . . . . . . . . . . . . . . 9 4. Detailed Description of SCReAM . . . . . . . . . . . . . . . 9
4.1. SCReAM Sender . . . . . . . . . . . . . . . . . . . . . . 9 4.1. SCReAM Sender . . . . . . . . . . . . . . . . . . . . . . 9
4.1.1. Constants and Parameter values . . . . . . . . . . . 9 4.1.1. Constants and Parameter values . . . . . . . . . . . 9
4.1.1.1. Constants . . . . . . . . . . . . . . . . . . . . 9 4.1.1.1. Constants . . . . . . . . . . . . . . . . . . . . 9
4.1.1.2. State variables . . . . . . . . . . . . . . . . . 11 4.1.1.2. State variables . . . . . . . . . . . . . . . . . 11
4.1.2. Network congestion control . . . . . . . . . . . . . 13 4.1.2. Network congestion control . . . . . . . . . . . . . 13
4.1.2.1. Congestion window update . . . . . . . . . . . . 16 4.1.2.1. Reaction to packets loss and ECN . . . . . . . . 15
4.1.2.2. Competing flows compensation . . . . . . . . . . 18 4.1.2.2. Congestion window update . . . . . . . . . . . . 16
4.1.2.3. Lost packet detection . . . . . . . . . . . . . . 20 4.1.2.3. Competing flows compensation . . . . . . . . . . 18
4.1.2.4. Send window calculation . . . . . . . . . . . . . 20 4.1.2.4. Lost packet detection . . . . . . . . . . . . . . 20
4.1.2.5. Packet pacing . . . . . . . . . . . . . . . . . . 21 4.1.2.5. Send window calculation . . . . . . . . . . . . . 20
4.1.2.6. Resuming fast increase . . . . . . . . . . . . . 21 4.1.2.6. Packet pacing . . . . . . . . . . . . . . . . . . 21
4.1.2.7. Resuming fast increase . . . . . . . . . . . . . 22
4.1.2.8. Stream prioritization . . . . . . . . . . . . . . 22
4.1.3. Media rate control . . . . . . . . . . . . . . . . . 22 4.1.3. Media rate control . . . . . . . . . . . . . . . . . 22
4.2. SCReAM Receiver . . . . . . . . . . . . . . . . . . . . . 25 4.2. SCReAM Receiver . . . . . . . . . . . . . . . . . . . . . 25
4.2.1. Requirements on feedback elements . . . . . . . . . . 25 4.2.1. Requirements on feedback elements . . . . . . . . . . 25
4.2.2. Requirements on feedback intensity . . . . . . . . . 27 4.2.2. Requirements on feedback intensity . . . . . . . . . 27
5. Discussion . . . . . . . . . . . . . . . . . . . . . . . . . 27 5. Discussion . . . . . . . . . . . . . . . . . . . . . . . . . 28
6. Implementation status . . . . . . . . . . . . . . . . . . . . 28 6. Implementation status . . . . . . . . . . . . . . . . . . . . 29
6.1. OpenWebRTC . . . . . . . . . . . . . . . . . . . . . . . 29 6.1. OpenWebRTC . . . . . . . . . . . . . . . . . . . . . . . 29
6.2. A C++ Implementation of SCReAM . . . . . . . . . . . . . 29 6.2. A C++ Implementation of SCReAM . . . . . . . . . . . . . 30
7. Suggested experiments . . . . . . . . . . . . . . . . . . . . 30 7. Suggested experiments . . . . . . . . . . . . . . . . . . . . 30
8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 30 8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 31
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 31 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 31
10. Security Considerations . . . . . . . . . . . . . . . . . . . 31 10. Security Considerations . . . . . . . . . . . . . . . . . . . 31
11. Change history . . . . . . . . . . . . . . . . . . . . . . . 31 11. Change history . . . . . . . . . . . . . . . . . . . . . . . 31
12. References . . . . . . . . . . . . . . . . . . . . . . . . . 32 12. References . . . . . . . . . . . . . . . . . . . . . . . . . 33
12.1. Normative References . . . . . . . . . . . . . . . . . . 32 12.1. Normative References . . . . . . . . . . . . . . . . . . 33
12.2. Informative References . . . . . . . . . . . . . . . . . 33 12.2. Informative References . . . . . . . . . . . . . . . . . 33
Appendix A. Additional information . . . . . . . . . . . . . . . 34
A.1. Stream prioritization . . . . . . . . . . . . . . . . . . 34
A.2. Computation of autocorrelation function . . . . . . . . . 35
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 35 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 35
1. Introduction 1. Introduction
Congestion in the Internet occurs when the transmitted bitrate is Congestion in the Internet occurs when the transmitted bitrate is
higher than the available capacity over a given transmission path. higher than the available capacity over a given transmission path.
Applications that are deployed in the Internet MUST employ congestion Applications that are deployed in the Internet have to employ
control, to achieve robust performance and to avoid congestion congestion control, to achieve robust performance and to avoid
collapse in the Internet. Interactive realtime communication imposes congestion collapse in the Internet. Interactive realtime
a lot of requirements on the transport, therefore a robust, efficient communication imposes a lot of requirements on the transport,
rate adaptation for all access types is an important part of therefore a robust, efficient rate adaptation for all access types is
interactive realtime communications as the transmission channel an important part of interactive realtime communications as the
bandwidth MAY vary over time. Wireless access such as LTE, which is transmission channel bandwidth can vary over time. Wireless access
an integral part of the current Internet, increases the importance of such as LTE, which is an integral part of the current Internet,
rate adaptation as the channel bandwidth of a default LTE bearer increases the importance of rate adaptation as the channel bandwidth
[QoS-3GPP] can change considerably in a very short time frame. Thus of a default LTE bearer [QoS-3GPP] can change considerably in a very
a rate adaptation solution for interactive realtime media, such as short time frame. Thus a rate adaptation solution for interactive
WebRTC, SHOULD be both quick and be able to operate over a large realtime media, such as WebRTC, should be both quick and be able to
range in channel capacity. This memo describes SCReAM (Self-Clocked operate over a large range in channel capacity. This memo describes
Rate Adaptation for Multimedia), a solution that is based on the SCReAM (Self-Clocked Rate Adaptation for Multimedia), a solution that
self-clocking principle of TCP and uses techniques similar to what is implements congestion control for RTP streams [RFC3550]. While
used in the LEDBAT based rate adaptation algorithm [RFC6817]. SCReAM SCReAM was originally devised for WebRTC (Web Real-Time
is not entirely self-clocked as it augments self-clocking with pacing Communication) [RFC7478], it can also be used for other applications
and a minimum send rate. where congestion control of RTP streams is necessary. SCReAM is
based on the self-clocking principle of TCP and uses techniques
similar to what is used in the LEDBAT based rate adaptation algorithm
[RFC6817]. SCReAM is not entirely self-clocked as it augments self-
clocking with pacing and a minimum send rate.
1.1. Wireless (LTE) access properties 1.1. Wireless (LTE) access properties
[I-D.ietf-rmcat-wireless-tests] describes the complications that can [I-D.ietf-rmcat-wireless-tests] describes the complications that can
be observed in wireless environments. Wireless access such as LTE be observed in wireless environments. Wireless access such as LTE
can typically not guarantee a given bandwidth, this is true can typically not guarantee a given bandwidth, this is true
especially for default bearers. The network throughput MAY vary especially for default bearers. The network throughput can vary
considerably for instance in cases where the wireless terminal is considerably for instance in cases where the wireless terminal is
moving around. Even though LTE can support bitrates well above moving around. Even though LTE can support bitrates well above
100Mbps, there are cases when the available bitrate can be much 100Mbps, there are cases when the available bitrate can be much
lower, examples are situations with high network load and poor lower, examples are situations with high network load and poor
coverage. An additional complication is that the network throughput coverage. An additional complication is that the network throughput
MAY drop for short time intervals at e.g. handover, these short can drop for short time intervals at e.g. handover, these short
glitches are initially very difficult to distinguish from more glitches are initially very difficult to distinguish from more
permanent reductions in throughput. permanent reductions in throughput.
Unlike wireline bottlenecks with large statistical multiplexing it is Unlike wireline bottlenecks with large statistical multiplexing it is
not possible to try to maintain a given bitrate when congestion 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 detected with the hope that other flows will yield, this is because
there are generally few other flows competing for the same there are generally few other flows competing for the same
bottleneck. Each user gets its own variable throughput bottleneck, bottleneck. Each user gets its own variable throughput bottleneck,
where the throughput depends on factors like channel quality, network where the throughput depends on factors like channel quality, network
load and historical throughput. The bottom line is, if the load and historical throughput. The bottom line is, if the
throughput drops, the sender has no other option than to reduce the throughput drops, the sender has no other option than to reduce the
bitrate. Once the radio scheduler has reduced the resource bitrate. Once the radio scheduler has reduced the resource
allocation for a bearer, an RMCAT flow in that bearer SHOULD reduce allocation for a bearer, an RMCAT flow in that bearer aims to reduce
the sending rate quite quickly (within one RTT) in order to avoid the sending rate quite quickly (within one RTT) in order to avoid
excessive queuing delay or packet loss. excessive queuing delay or packet loss.
1.2. Why is it a self-clocked algorithm? 1.2. Why is it a self-clocked algorithm?
Self-clocked congestion control algorithms provide a benefit over the Self-clocked congestion control algorithms provide a benefit over the
rate based counterparts in that the former consists of two adaptation rate based counterparts in that the former consists of two adaptation
mechanisms: mechanisms:
o A congestion window computation that evolves over a longer o A congestion window computation that evolves over a longer
skipping to change at page 10, line 10 skipping to change at page 10, line 10
The RECOMMENDED values, within (), for the constants are deduced from The RECOMMENDED values, within (), for the constants are deduced from
experiments. The units are enclosed in square brackets [ ]. experiments. The units are enclosed in square brackets [ ].
QDELAY_TARGET_LO (0.1s) QDELAY_TARGET_LO (0.1s)
Target value for the minimum qdelay. Target value for the minimum qdelay.
QDELAY_TARGET_HI (0.4s) QDELAY_TARGET_HI (0.4s)
Target value for the maximum qdelay. This parameter provides an Target value for the maximum qdelay. This parameter provides an
upper limit to how much the target qdelay (qdelay_target) can be upper limit to how much the target qdelay (qdelay_target) can be
increased in order to cope with competing loss based flows. The increased in order to cope with competing loss based flows. The
target qdelay MUST NOT be initialized to this high value however as target qdelay does not have to be initialized to this high value
it would increase e2e delay and also make the rate control and however as it would increase e2e delay and also make the rate
congestion control loop sluggish. control and congestion control loop sluggish.
QDELAY_WEIGHT (0.1) QDELAY_WEIGHT (0.1)
Averaging factor for qdelay_fraction_avg. Averaging factor for qdelay_fraction_avg.
QDELAY_TREND_TH (0.2) QDELAY_TREND_TH (0.2)
Averaging factor for qdelay_fraction_avg. Averaging factor for qdelay_fraction_avg.
QDELAY_TREND_TH (0.2)
Averaging factor for qdelay_fraction_avg.
MIN_CWND (3000byte)
Min CWND.
MAX_BYTES_IN_FLIGHT_HEAD_ROOM (1.1) MAX_BYTES_IN_FLIGHT_HEAD_ROOM (1.1)
Headroom for the limitation of CWND. Headroom for the limitation of CWND.
GAIN (1.0) GAIN (1.0)
Gain factor for congestion window adjustment. Gain factor for congestion window adjustment.
BETA_LOSS (0.8) BETA_LOSS (0.8)
CWND scale factor due to loss event. CWND scale factor due to loss event.
BETA_ECN (0.8) BETA_ECN (0.8)
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for evaluation of a real implementation. for evaluation of a real implementation.
RTP_QDELAY_TH (0.02s) RTP queue delay threshold for a target rate RTP_QDELAY_TH (0.02s) RTP queue delay threshold for a target rate
reduction. reduction.
TARGET_RATE_SCALE_RTP_QDELAY (0.95) Target rate scale when RTP TARGET_RATE_SCALE_RTP_QDELAY (0.95) Target rate scale when RTP
qdelay threshold exceeds. qdelay threshold exceeds.
QDELAY_TREND_LO (0.2) Threshold value for qdelay_trend. QDELAY_TREND_LO (0.2) Threshold value for qdelay_trend.
T_RESUME_FAST_INCREASE Time span until fast increase can be resumed, T_RESUME_FAST_INCREASE (5s) Time span until fast increase can be
given that the qdelay_trend is below QDELAY_TREND_LO. resumed, given that the qdelay_trend is below QDELAY_TREND_LO.
RATE_PACE_MIN (50000bps) Minimum pacing rate.
4.1.1.2. State variables 4.1.1.2. State variables
The values within () indicate initial values. The values within () indicate initial values.
qdelay_target (QDELAY_TARGET_LO) qdelay_target (QDELAY_TARGET_LO)
qdelay target, a variable qdelay target is introduced to manage qdelay target, a variable qdelay target is introduced to manage
cases where e.g. FTP competes for the bandwidth over the same cases where e.g. FTP competes for the bandwidth over the same
bottleneck, a fixed qdelay target would otherwise starve the RMCAT bottleneck, a fixed qdelay target would otherwise starve the RMCAT
flow under such circumstances. The qdelay target is allowed to flow under such circumstances. The qdelay target is allowed to
skipping to change at page 14, line 48 skipping to change at page 15, line 7
# of congestion in the past # of congestion in the past
qdelay_trend_mem = max(0.99*qdelay_trend_mem, qdelay_trend) qdelay_trend_mem = max(0.99*qdelay_trend_mem, qdelay_trend)
<CODE ENDS> <CODE ENDS>
The qdelay fraction is sampled every 50ms and the last 20 samples are The qdelay fraction is sampled every 50ms and the last 20 samples are
stored in a vector (qdelay_fraction_hist). This vector is used in stored in a vector (qdelay_fraction_hist). This vector is used in
the computation of an qdelay trend that gives a value between 0.0 and the computation of an qdelay trend that gives a value between 0.0 and
1.0 depending on the estimated congestion level. The prediction 1.0 depending on the estimated congestion level. The prediction
coefficient 'a' has positive values if qdelay shows an increasing coefficient 'a' has positive values if qdelay shows an increasing
trend, thus an indication of congestion is obtained before the qdelay trend, thus an indication of congestion is obtained before the qdelay
target is reached. The autocorrelation function 'R' is defined in target is reached.
Appendix A.2. The prediction coefficient is further multiplied with
The autocorrelation function 'R' is defined as follows. Let x be a
vector constituting N values, the biased autocorrelation function for
a given lag=k for the vector x is given by.
n=N-k
R(x,k) = SUM x(n)*x(n+k)
n=1
The prediction coefficient is further multiplied with
qdelay_fraction_avg to reduce sensitivity to increasing qdelay when qdelay_fraction_avg to reduce sensitivity to increasing qdelay when
it is very small. The 50ms sampling is a simplification and MAY have it is very small. The 50ms sampling is a simplification and MAY have
the effect that the same qdelay is sampled several times, this does the effect that the same qdelay is sampled several times, this does
however not pose any problem as the vector is only used to determine however not pose any problem as the vector is only used to determine
if the qdelay is increasing or decreasing. The qdelay_trend is if the qdelay is increasing or decreasing. The qdelay_trend is
utilized in the media rate control to indicate incipient congestion utilized in the media rate control to indicate incipient congestion
and to determine when to exit from fast increase mode. and to determine when to exit from fast increase mode.
qdelay_trend_mem is used to enforce a less aggressive rate increase qdelay_trend_mem is used to enforce a less aggressive rate increase
after congestion events. The function after congestion events. The function
update_qdelay_fraction_hist(..) removes the oldest element and adds update_qdelay_fraction_hist(..) removes the oldest element and adds
the latest qdelay_fraction element to the qdelay_fraction_hist the latest qdelay_fraction element to the qdelay_fraction_hist
vector. vector.
4.1.2.1. Reaction to packets loss and ECN
A loss event is indicated if one or more RTP packets are declared A loss event is indicated if one or more RTP packets are declared
missing. The loss detection is described in Section 4.1.2.3. Once a missing. The loss detection is described in Section 4.1.2.4. Once a
loss event is detected, further detected lost RTP packets are ignored loss event is detected, further detected lost RTP packets SHOULD be
for a full smoothed round trip time, the intention of this is to ignored for a full smoothed round trip time, the intention of this is
limit the congestion window decrease to at most once per round trip. to limit the congestion window decrease to at most once per round
trip.
The congestion window back off due to loss events is deliberately a The congestion window back off due to loss events is deliberately a
bit less than is the case with e.g. TCP Reno. The reason is that bit less than is the case with e.g. TCP Reno. The reason is that
TCP is generally used to transmit whole files, which can be TCP is generally used to transmit whole files, which can be
translated to an infinite source bitrate. SCReAM on the other hand translated to an infinite source bitrate. SCReAM on the other hand
has a source whose rate is limited to a value close to the available has a source whose rate is limited to a value close to the available
transmit rate and often below that value, the effect of this is that transmit rate and often below that value, the effect of this is that
SCReAM has less opportunity to grab free capacity than a TCP based SCReAM has less opportunity to grab free capacity than a TCP based
file transfer. To compensate for this it is RECOMMENDED to let file transfer. To compensate for this it is RECOMMENDED to let
SCReAM reduce the congestion window less than what is the case with SCReAM reduce the congestion window less than what is the case with
TCP when loss events occur. TCP when loss events occur.
skipping to change at page 15, line 40 skipping to change at page 16, line 12
An ECN event is detected if the n_ECN counter in the feedback report An ECN event is detected if the n_ECN counter in the feedback report
has increased since the previous received feedback. Once an ECN has increased since the previous received feedback. Once an ECN
event is detected, the n_ECN counter is ignored for a full smoothed event is detected, the n_ECN counter is ignored for a full smoothed
round trip time, the intention of this is to limit the congestion round trip time, the intention of this is to limit the congestion
window decrease to at most once per round trip. The congestion window decrease to at most once per round trip. The congestion
window back off due to an ECN event MAY be smaller than if a loss window back off due to an ECN event MAY be smaller than if a loss
event occurs. This is in line with the idea outlined in event occurs. This is in line with the idea outlined in
[I-D.ietf-tcpm-alternativebackoff-ecn] to enable ECN marking [I-D.ietf-tcpm-alternativebackoff-ecn] to enable ECN marking
thresholds lower than the corresponding packet drop thresholds. thresholds lower than the corresponding packet drop thresholds.
4.1.2.2. Congestion window update
The update of the congestion window depends on whether loss or ECN- The update of the congestion window depends on whether loss or ECN-
marking or neither occurs. The pseudo code below describes actions marking or neither occurs. The pseudo code below describes actions
taken in case of the different events. taken in case of the different events.
<CODE BEGINS> <CODE BEGINS>
on congestion event(qdelay): on congestion event(qdelay):
# Either loss or ECN mark is detected # Either loss or ECN mark is detected
in_fast_increase = false in_fast_increase = false
if (is loss) if (is loss)
# loss is detected # loss is detected
cwnd = max(min_cwnd,cwnd*BETA_LOSS) cwnd = max(MIN_CWND,cwnd*BETA_LOSS)
else else
# No loss, so it is then an ECN mark # No loss, so it is then an ECN mark
cwnd = max(min_cwnd,cwnd*BETA_ECN) cwnd = max(MIN_CWND,cwnd*BETA_ECN)
end end
adjust_qdelay_target(qdelay) #compensating for competing flows adjust_qdelay_target(qdelay) #compensating for competing flows
calculate_send_window(qdelay,qdelay_target) calculate_send_window(qdelay,qdelay_target)
# when no congestion event # when no congestion event
on acknowledgement(qdelay): on acknowledgement(qdelay):
update_bytes_newly_acked() update_bytes_newly_acked()
update_cwnd(bytes_newly_acked) update_cwnd(bytes_newly_acked)
adjust_qdelay_target(qdelay) #compensating for competing flows adjust_qdelay_target(qdelay) #compensating for competing flows
calculate_send_window(qdelay, qdelay_target) calculate_send_window(qdelay, qdelay_target)
check_to_resume_fast_increase() check_to_resume_fast_increase()
<CODE ENDS> <CODE ENDS>
The methods are further described in detail below. The methods are further described in detail below.
4.1.2.1. Congestion window update
The congestion window update is based on qdelay, except for the The congestion window update is based on qdelay, except for the
occurrence of loss events (one or more lost RTP packets in one RTT), occurrence of loss events (one or more lost RTP packets in one RTT),
or ECN events, which was described earlier. or ECN events, which was described earlier.
Pseudo code for the update of the congestion window is found below. Pseudo code for the update of the congestion window is found below.
<CODE BEGINS> <CODE BEGINS>
update_cwnd(bytes_newly_acked): update_cwnd(bytes_newly_acked):
# in fast increase ? # in fast increase ?
skipping to change at page 18, line 19 skipping to change at page 18, line 19
the number of newly acknowledged bytes as long as the window is the number of newly acknowledged bytes as long as the window is
sufficiently used. Sparse feedback can potentially limit congestion sufficiently used. Sparse feedback can potentially limit congestion
window growth, an additional slack is therefore added, given by the window growth, an additional slack is therefore added, given by the
number of newly acknowledged bytes. number of newly acknowledged bytes.
The congestion window growth when in_fast_increase is false is The congestion window growth when in_fast_increase is false is
dictated by the relation between qdelay and qdelay_target, congestion dictated by the relation between qdelay and qdelay_target, congestion
window growth is limited if the window is not used sufficiently. window growth is limited if the window is not used sufficiently.
SCReAM calculates the GAIN in a similar way to what is specified in SCReAM calculates the GAIN in a similar way to what is specified in
[RFC6817]. There are however a few differences. [RFC6817]. However, [RFC6817] specifies that the CWND increase is
limited by an additional function controlled by a constant
o [RFC6817] specifies a constant GAIN, this specification however ALLOWED_INCREASE. This additional limitation is removed in this
limits the gain when CWND is increased dependent on near specification.
congestion state and the relation to the last known max CWND
value.
o [RFC6817] specifies that the CWND increase is limited by an
additional function controlled by a constant ALLOWED_INCREASE.
This additional limitation is removed in this specification.
Further the CWND is limited by max_bytes_in_flight and min_cwnd. The Further the CWND is limited by max_bytes_in_flight and min_cwnd. The
limitation of the congestion window by the maximum number of bytes in limitation of the congestion window by the maximum number of bytes in
flight over the last 5 seconds (max_bytes_in_flight) avoids possible flight over the last 5 seconds (max_bytes_in_flight) avoids possible
over-estimation of the throughput after for example, idle periods. over-estimation of the throughput after for example, idle periods.
An additional MAX_BYTES_IN_FLIGHT_HEAD_ROOM allows for a slack, to An additional MAX_BYTES_IN_FLIGHT_HEAD_ROOM allows for a slack, to
allow for a certain amount of media coder output rate variability. allow for a certain amount of media coder output rate variability.
4.1.2.2. Competing flows compensation 4.1.2.3. Competing flows compensation
It is likely that a flow using SCReAM algorithm will have to share It is likely that a flow using SCReAM algorithm will have to share
congested bottlenecks with other flows that use a more aggressive congested bottlenecks with other flows that use a more aggressive
congestion control algorithm. SCReAM takes care of such situations congestion control algorithm. SCReAM takes care of such situations
by adjusting the qdelay_target. by adjusting the qdelay_target.
<CODE BEGINS> <CODE BEGINS>
adjust_qdelay_target(qdelay) adjust_qdelay_target(qdelay)
qdelay_norm_t = qdelay / QDELAY_TARGET_LOW qdelay_norm_t = qdelay / QDELAY_TARGET_LOW
update_qdelay_norm_history(qdelay_norm_t) update_qdelay_norm_history(qdelay_norm_t)
# Compute variance # Compute variance
qdelay_norm_var_t = VARIANCE(qdelay_norm_history(200)) qdelay_norm_var_t = VARIANCE(qdelay_norm_history(200))
# Compensation for competing traffic # Compensation for competing traffic
# Compute average # Compute average
qdelay_norm_avg_t = AVERAGE(qdelay_norm_history(50)) qdelay_norm_avg_t = AVERAGE(qdelay_norm_history(50))
# Compute upper limit to target delay # Compute upper limit to target delay
oh_t = qdelay_norm_avg_t + sqrt(qdelay_norm_var_t) new_target_t = qdelay_norm_avg_t + sqrt(qdelay_norm_var_t)
oh_t *= QDELAY_TARGET_LO new_target_t *= QDELAY_TARGET_LO
if (loss_event_rate > 0.002) if (loss_event_rate > 0.002)
# Packet losses detected # Packet losses detected
qdelay_target = 1.5*oh_t qdelay_target = 1.5*new_target_t
else else
if (qdelay_norm_var_t < 0.2) if (qdelay_norm_var_t < 0.2)
# Reasonably safe to set target qdelay # Reasonably safe to set target qdelay
qdelay_target = oh_t qdelay_target = new_target_t
else else
# Check if target delay can be reduced, this helps to avoid # Check if target delay can be reduced, this helps to avoid
# that the target delay is locked to high values for ever # that the target delay is locked to high values for ever
if (oh_t < QDELAY_TARGET_LO) if (new_target_t < QDELAY_TARGET_LO)
# Decrease target delay quickly as measured queueing # Decrease target delay quickly as measured queueing
# delay is lower than target # delay is lower than target
qdelay_target = max(qdelay_target*0.5,oh_t) qdelay_target = max(qdelay_target*0.5,new_target_t)
else else
# Decrease target delay slowly # Decrease target delay slowly
qdelay_target *= 0.9 qdelay_target *= 0.9
end end
end end
end end
# Apply limits # Apply limits
qdelay_target = min(QDELAY_TARGET_HI, qdelay_target) qdelay_target = min(QDELAY_TARGET_HI, qdelay_target)
qdelay_target = max(QDELAY_TARGET_LO, qdelay_target) qdelay_target = max(QDELAY_TARGET_LO, qdelay_target)
skipping to change at page 20, line 6 skipping to change at page 20, line 6
A low qdelay_norm_avg_t value indicates that the qdelay does not A low qdelay_norm_avg_t value indicates that the qdelay does not
change rapidly. It is desired to avoid the case that the qdelay change rapidly. It is desired to avoid the case that the qdelay
target is increased due to self-congestion, indicated by a changing target is increased due to self-congestion, indicated by a changing
qdelay and consequently an increased qdelay_norm_var_t. Still it qdelay and consequently an increased qdelay_norm_var_t. Still it
SHOULD be possible to increase the qdelay target if the qdelay SHOULD be possible to increase the qdelay target if the qdelay
continues to be high. This is a simple function with a certain risk continues to be high. This is a simple function with a certain risk
of both false positives and negatives. In the simulated LTE test of both false positives and negatives. In the simulated LTE test
cases it manages competing FTP flows reasonably well at the same time cases it manages competing FTP flows reasonably well at the same time
as generally avoiding accidental increases in the qdelay target. The as generally avoiding accidental increases in the qdelay target. The
algorithm can however accidentally increase the qdelay target and algorithm can however accidentally increase the qdelay target and
cause self-inflicted congestion in certain cases. It is therefore cause self-inflicted congestion in certain cases. If it is deemed
RECOMMENDED that the algorithm described in this section is turned unlikely that competing flows occur over the same bottleneck, the
off it is deemed unlikely that competing flows occur over the same algorithm described in this section MAY be turned off. However, when
bottleneck sending over the Internet, often the network conditions are not known
for sure. Therefore turning this algorithm off must be considered
with caution as that can lead to basically zero throughput if
competing with other, loss based, traffic.
4.1.2.3. Lost packet detection 4.1.2.4. Lost packet detection
Lost packet detection is based on the received sequence number list. Lost packet detection is based on the received sequence number list.
A reordering window SHOULD be applied to avoid packet reordering A reordering window SHOULD be applied to avoid that packet reordering
triggering loss events. triggers loss events.
The reordering window is specified as a time unit, similar to the The reordering window is specified as a time unit, similar to the
ideas behind RACK (Recent ACKnowledgement) [I-D.ietf-tcpm-rack]. The ideas behind RACK (Recent ACKnowledgement) [I-D.ietf-tcpm-rack]. The
computation of the reordering window is made possible by means of a computation of the reordering window is made possible by means of a
lost flag in the list of transmitted RTP packets. This flag is set lost flag in the list of transmitted RTP packets. This flag is set
if the received sequence number list indicates that the given RTP if the received sequence number list indicates that the given RTP
packet is missing. If a later feedback indicates that a previously packet is missing. If a later feedback indicates that a previously
lost marked packet was indeed received, then the reordering window is lost marked packet was indeed received, then the reordering window is
updated to reflect the reordering delay. The reordering window is updated to reflect the reordering delay. The reordering window is
given by the difference in time between the event that the packet was given by the difference in time between the event that the packet was
marked as lost and the event that it was indicated as successfully marked as lost and the event that it was indicated as successfully
received. received.
Loss is detected if a given RTP packet is not acknowledged within a Loss is detected if a given RTP packet is not acknowledged within a
time window (indicated by the reordering window) after an RTP packet time window (indicated by the reordering window) after an RTP packet
with higher sequence number was acknowledged. with higher sequence number was acknowledged.
4.1.2.4. Send window calculation 4.1.2.5. Send window calculation
The basic design principle behind packet transmission in SCReAM is to The basic design principle behind packet transmission in SCReAM is to
allow transmission only if the number of bytes in flight is less than allow transmission only if the number of bytes in flight is less than
the congestion window. There are however two reasons why this strict the congestion window. There are however two reasons why this strict
rule will not work optimally: rule will not work optimally:
o Bitrate variations: The media frame size is always varying to a o Bitrate variations: The media frame size is always varying to a
larger or smaller extent. A strict rule can lead to that the larger or smaller extent. A strict rule can lead to that the
media bitrate will have difficulties to increase as the congestion media bitrate will have difficulties to increase as the congestion
window puts a too hard restriction on the media frame size window puts a too hard restriction on the media frame size
variation. This can lead to occasional queuing of RTP packets in variation. This can lead to occasional queuing of RTP packets in
the RTP packet queue that will prevent bitrate increase. the RTP packet queue that will prevent bitrate increase.
o Reverse (feedback) path congestion: Especially in transport over o Reverse (feedback) path congestion: Especially in transport over
buffer-bloated networks, the one way delay in the reverse buffer-bloated networks, the one way delay in the reverse
direction MAY jump due to congestion. The effect of this is that direction can jump due to congestion. The effect of this is that
the acknowledgements are delayed with the result that the self- the acknowledgements are delayed with the result that the self-
clocking is temporarily halted, even though the forward path is clocking is temporarily halted, even though the forward path is
not congested. not congested.
The send window is adjusted depending on qdelay and its relation to The send window is adjusted depending on qdelay and its relation to
the qdelay target and the relation between the congestion window and the qdelay target and the relation between the congestion window and
the number of bytes in flight. A strict rule is applied when qdelay the number of bytes in flight. A strict rule is applied when qdelay
is higher than qdelay_target, to avoid further queue buildup in the is higher than qdelay_target, to avoid further queue buildup in the
network. For cases when qdelay is lower than the qdelay_target, a network. For cases when qdelay is lower than the qdelay_target, a
more relaxed rule is applied. This allows the bitrate to increase more relaxed rule is applied. This allows the bitrate to increase
skipping to change at page 21, line 31 skipping to change at page 21, line 33
if (qdelay <= qdelay_target) if (qdelay <= qdelay_target)
send_wnd = cwnd+MSS-bytes_in_flight send_wnd = cwnd+MSS-bytes_in_flight
else else
send_wnd = cwnd-bytes_in_flight send_wnd = cwnd-bytes_in_flight
end end
<CODE ENDS> <CODE ENDS>
The send window is updated whenever an RTP packet is transmitted or The send window is updated whenever an RTP packet is transmitted or
an RTCP feedback messaged is received. an RTCP feedback messaged is received.
4.1.2.5. Packet pacing 4.1.2.6. Packet pacing
Packet pacing is used in order to mitigate coalescing i.e. that Packet pacing is used in order to mitigate coalescing i.e. that
packets are transmitted in bursts, with the increased risk of more packets are transmitted in bursts, with the increased risk of more
jitter and potentially increased packet loss. The time interval jitter and potentially increased packet loss. Packet pacing also
between consecutive packet transmissions is enforced to be equal to mitigates possible issues with queue overflow due to key-frame
or higher than t_pace where t_pace is given by the equations below : generation in video coders. The time interval between consecutive
packet transmissions is enforced to be equal to or higher than t_pace
where t_pace is given by the equations below :
<CODE BEGINS> <CODE BEGINS>
pace_bitrate = max (RATE_PACE_MIN, cwnd* 8 / s_rtt) pace_bitrate = max (RATE_PACE_MIN, cwnd* 8 / s_rtt)
t_pace = rtp_size * 8 / pace_bitrate t_pace = rtp_size * 8 / pace_bitrate
<CODE ENDS> <CODE ENDS>
rtp_size is the size of the last transmitted RTP packet, s_rtt is the rtp_size is the size of the last transmitted RTP packet, s_rtt is the
smoothed round trip time. RATE_PACE_MIN=50000 is the minimum pacing smoothed round trip time. RATE_PACE_MIN is the minimum pacing rate.
rate.
4.1.2.6. Resuming fast increase 4.1.2.7. Resuming fast increase
Fast increase can resume in order to speed up the bitrate increase in Fast increase can resume in order to speed up the bitrate increase in
case congestion abates. The condition to resume fast increase case congestion abates. The condition to resume fast increase
(in_fast_increase = true) is that qdelay_trend is less than (in_fast_increase = true) is that qdelay_trend is less than
QDELAY_TREND_LO for T_RESUME_FAST_INCREASE seconds or more. QDELAY_TREND_LO for T_RESUME_FAST_INCREASE seconds or more.
4.1.2.8. Stream prioritization
The SCReAM algorithm makes a good distinction between network
congestion control and the media rate control. This is easily
extended to many streams, in which case RTP packets from two or more
RTP queues are scheduled at the rate permitted by the network
congestion control.
The scheduling can be done by means of a few different scheduling
regimes. For example the method applied in
[I-D.ietf-rmcat-coupled-cc] can be used. The implementation of
SCReAM [SCReAM-CPP-implementation] use credit based scheduling. In
credit based scheduling, credit is accumulated by queues as they wait
for service and are spent while the queues are being serviced. For
instance, if one queue is allowed to transmit 1000bytes, then a
credit of 1000bytes is allocated to the other unscheduled queues.
This principle can be extended to weighted scheduling in which case
the credit allocated to unscheduled queues depends on the relative
weights. The latter is also implemented in
[SCReAM-CPP-implementation].
4.1.3. Media rate control 4.1.3. Media rate control
The media rate control algorithm is executed at regular intervals The media rate control algorithm is executed at regular intervals
RATE_ADJUSTMENT_INTERVAL, with the exception of a prompt reaction to RATE_ADJUSTMENT_INTERVAL, with the exception of a prompt reaction to
loss events. The media rate control operates based on the size of loss events. The media rate control operates based on the size of
the RTP packet send queue and observed loss events. In addition, the RTP packet send queue and observed loss events. In addition,
qdelay_trend is also considered in the media rate control to reduce qdelay_trend is also considered in the media rate control to reduce
the amount of induced network jitter. the amount of induced network jitter.
The role of the media rate control is to strike a reasonable balance The role of the media rate control is to strike a reasonable balance
between a low amount of queuing in the RTP queue(s) and a sufficient between a low amount of queuing in the RTP queue(s) and a sufficient
amount of data to send in order to keep the data path busy. A too amount of data to send in order to keep the data path busy. A too
cautious setting leads to possible under-utilization of network cautious setting leads to possible under-utilization of network
capacity leading to the flow being starved out by other more capacity leading to that the flow can become starved out by other
opportunistic traffic. On the other hand, a too aggressive setting more opportunistic traffic. On the other hand, a too aggressive
leads to increased jitter. setting leads to increased jitter.
The target_bitrate is adjusted depending on the congestion state. The target_bitrate is adjusted depending on the congestion state.
The target bitrate can vary between a minimum value The target bitrate can vary between a minimum value
(TARGET_BITRATE_MIN) and a maximum value (TARGET_BITRATE_MAX). (TARGET_BITRATE_MIN) and a maximum value (TARGET_BITRATE_MAX).
TARGET_BITRATE_MIN SHOULD be chosen to a low enough value to avoid TARGET_BITRATE_MIN SHOULD be chosen to a low enough value to avoid
RTP packets being queued up when the network throughput becomes low. that RTP packets become queued up when the network throughput is
The sender SHOULD also be equipped with a mechanism that discards RTP reduced. The sender SHOULD also be equipped with a mechanism that
packets in cases where the network throughput becomes very low and discards RTP packets in cases where the network throughput becomes
RTP packets are excessively delayed. very low and RTP packets are excessively delayed.
For the overall bitrate adjustment, two network throughput estimates For the overall bitrate adjustment, two network throughput estimates
are computed : are computed :
o rate_transmit: The measured transmit bitrate. o rate_transmit: The measured transmit bitrate.
o rate_ack: The ACKed bitrate, i.e. the volume of ACKed bits per o rate_ack: The ACKed bitrate, i.e. the volume of ACKed bits per
second. second.
Both estimates are updated every 200ms. Both estimates are updated every 200ms.
The current throughput, current_rate, is computed as the maximum The current throughput, current_rate, is computed as the maximum
value of rate_transmit and rate_ack. The rationale behind the use of value of rate_transmit and rate_ack. The rationale behind the use of
rate_ack in addition to rate_transmit is that rate_transmit is rate_ack in addition to rate_transmit is that rate_transmit is
affected also by the amount of data that is available to transmit, affected also by the amount of data that is available to transmit,
thus a lack of data to transmit can be seen as reduced throughput thus a lack of data to transmit can be seen as reduced throughput
that MAY itself cause an unnecessary rate reduction. To overcome that can itself cause an unnecessary rate reduction. To overcome
this shortcoming; rate_ack is used as well. This gives a more stable this shortcoming; rate_ack is used as well. This gives a more stable
throughput estimate. throughput estimate.
The rate change behavior depends on whether a loss or ECN event has The rate change behavior depends on whether a loss or ECN event has
occurred and if the congestion control is in fast increase or not. occurred and if the congestion control is in fast increase or not.
<CODE BEGINS> <CODE BEGINS>
# The target_bitrate is updated at a regular interval according # The target_bitrate is updated at a regular interval according
# to RATE_ADJUST_INTERVAL # to RATE_ADJUST_INTERVAL
skipping to change at page 24, line 23 skipping to change at page 24, line 48
continues. The rationale behind the rate reduction due to loss is continues. The rationale behind the rate reduction due to loss is
that a congestion window reduction will take effect, a rate reduction that a congestion window reduction will take effect, a rate reduction
pro actively avoids RTP packets being queued up when the transmit pro actively avoids RTP packets being queued up when the transmit
rate decreases due to the reduced congestion window. A similar rate rate decreases due to the reduced congestion window. A similar rate
reduction happens when ECN events are detected. reduction happens when ECN events are detected.
The rate update frequency is limited by RATE_ADJUST_INTERVAL, unless The rate update frequency is limited by RATE_ADJUST_INTERVAL, unless
a loss event occurs. The value is based on experimentation with real a loss event occurs. The value is based on experimentation with real
life limitations in video coders taken into account life limitations in video coders taken into account
[SCReAM-CPP-implementation]. A too short interval is shown to make [SCReAM-CPP-implementation]. A too short interval is shown to make
the video coder internal rate control loop more unstable, a too long the rate control loop in video coders more unstable, a too long
interval makes the overall congestion control sluggish. interval makes the overall congestion control sluggish.
When in fast increase state (in_fast_increase=true), the bitrate When in fast increase state (in_fast_increase=true), the bitrate
increase is given by the desired ramp-up speed (RAMP_UP_SPEED) . The increase is given by the desired ramp-up speed (RAMP_UP_SPEED) . The
ramp-up speed is limited when the target bitrate is low to avoid rate ramp-up speed is limited when the target bitrate is low to avoid rate
oscillation at low bottleneck bitrates. The setting of RAMP_UP_SPEED oscillation at low bottleneck bitrates. The setting of RAMP_UP_SPEED
depends on preferences, a high setting such as 1000kbps/s makes it depends on preferences, a high setting such as 1000kbps/s makes it
possible to quickly get high quality media, this is however at the possible to quickly get high quality media, this is however at the
expense of a increased jitter, which can manifest itself as e.g. expense of a increased jitter, which can manifest itself as e.g.
choppy video rendering. choppy video rendering.
skipping to change at page 24, line 45 skipping to change at page 25, line 24
When in_fast_increase is false, the bitrate increase is given by the When in_fast_increase is false, the bitrate increase is given by the
current bitrate and is also controlled by the estimated RTP queue and current bitrate and is also controlled by the estimated RTP queue and
the qdelay trend, thus it is sufficient that an increased congestion the qdelay trend, thus it is sufficient that an increased congestion
level is sensed by the network congestion control to limit the level is sensed by the network congestion control to limit the
bitrate. The target_bitrate_last_max is updated when congestion is bitrate. The target_bitrate_last_max is updated when congestion is
detected. detected.
Finally the target_bitrate is enforced to be within the defined min Finally the target_bitrate is enforced to be within the defined min
and max values. and max values.
The aware reader MAY notice the dependency on the qdelay in the The aware reader may notice the dependency on the qdelay in the
computation of the target bitrate, this manifests itself in the use computation of the target bitrate, this manifests itself in the use
of the qdelay_trend. As these parameters are used also in the of the qdelay_trend. As these parameters are used also in the
network congestion control one MAY suspect some odd interaction network congestion control one may suspect some odd interaction
between the media rate control and the network congestion control, between the media rate control and the network congestion control,
this is in fact the case if the parameter PRE_CONGESTION_GUARD is set this is in fact the case if the parameter PRE_CONGESTION_GUARD is set
to a high value. The use of qdelay_trend in the media rate control to a high value. The use of qdelay_trend in the media rate control
is solely to reduce jitter, the dependency can be removed by setting is solely to reduce jitter, the dependency can be removed by setting
PRE_CONGESTION_GUARD=0, the effect is a somewhat faster rate increase PRE_CONGESTION_GUARD=0, the effect is a somewhat faster rate increase
after congestion, at the expense of increased jitter in congested after congestion, at the expense of increased jitter in congested
situations. situations.
4.2. SCReAM Receiver 4.2. SCReAM Receiver
skipping to change at page 25, line 23 skipping to change at page 25, line 50
SCReAM sender, in addition, the receive time of the RTP packet with SCReAM sender, in addition, the receive time of the RTP packet with
the highest sequence number is echoed back. Upon reception of each the highest sequence number is echoed back. Upon reception of each
RTP packet the receiver MUST maintain enough information to send the RTP packet the receiver MUST maintain enough information to send the
aforementioned values to the SCReAM sender via a RTCP transport layer aforementioned values to the SCReAM sender via a RTCP transport layer
feedback message. The frequency of the feedback message depends on feedback message. The frequency of the feedback message depends on
the available RTCP bandwidth. The requirements on the feedback the available RTCP bandwidth. The requirements on the feedback
elements and the feedback interval is described. elements and the feedback interval is described.
4.2.1. Requirements on feedback elements 4.2.1. Requirements on feedback elements
SCReAM requires the following elements for its basic functionality, The following feedback elements are REQUIRED for the basic
i.e. only including features that are strictly necessary in order to functionality in SCReAM.
make SCReAM function. ECN is not included as basic functionality as
it regarded as an additional feature that is not strictly necessary
even though it can improve quality of experience quite considerably.
o A list of received RTP packets. This list SHOULD be sufficiently o A list of received RTP packets. This list SHOULD be sufficiently
long to cover all received RTP packets. This list can be realized long to cover all received RTP packets. This list can be realized
with the Loss RLE report block in [RFC3611]. with the Loss RLE report block in [RFC3611].
o A wall clock timestamp corresponding to the received RTP packet o A wall clock timestamp corresponding to the received RTP packet
with the highest sequence number is required in order to compute with the highest sequence number is required in order to compute
the qdelay. This can be realized by means of the Packet Receipt the qdelay. This can be realized by means of the Packet Receipt
Times Report Block in [RFC3611]. begin_seq MUST be set to the Times Report Block in [RFC3611]. begin_seq MUST be set to the
highest received (possibly wrapped around) sequence number, highest received (possibly wrapped around) sequence number,
skipping to change at page 26, line 35 skipping to change at page 27, line 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source | | SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| begin_seq | end_seq | | begin_seq | end_seq |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Receipt time of packet begin_seq | | Receipt time of packet begin_seq |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2: Basic feedback message for SCReAM, based on RFC3611 Figure 2: Basic feedback message for SCReAM, based on RFC3611
In a typical use case, no more than four Loss RLE chunks SHOULD be In a typical use case, no more than four Loss RLE chunks are needed,
needed, thus the feedback message will be 44bytes. It is obvious thus the feedback message will be 44bytes. It is obvious from the
from the figure that there is a lot of redundant information in the figure that there is a lot of redundant information in the feedback
feedback message. A more optimized feedback format, including the message. A more optimized feedback format, including the additional
additional feedback elements listed below, could reduce the feedback feedback elements listed below, could reduce the feedback message
message size a bit. size a bit.
Additional feedback elements that can improve the performance of Additional feedback elements that can improve the performance of
SCReAM are: SCReAM are:
o Accumulated number of ECN-CE marked packets (n_ECN). This can for o Accumulated number of ECN-CE marked packets (n_ECN). This can for
instance be realized with the ECN Feedback Report Format in instance be realized with the ECN Feedback Report Format in
[RFC6679]. The given feedback report format is actually a slight [RFC6679]. The given feedback report format is actually a slight
overkill as SCReAM would do quite well with only a counter that overkill as SCReAM would do quite well with only a counter that
increments by one for each received packet with the ECN-CE code increments by one for each received packet with the ECN-CE code
point set. The more bulky format MAY be nevertheless be useful point set. The more bulky format could nevertheless be useful for
for e.g ECN black-hole detection. e.g ECN black-hole detection.
4.2.2. Requirements on feedback intensity 4.2.2. Requirements on feedback intensity
SCReAM benefits from a relatively frequent feedback. The feedback SCReAM benefits from a relatively frequent feedback. It is
interval depends on the media bitrate. At low bitrates it is RECOMMENDED that a SCReAM implementation follows the guidelines
sufficient with a feedback interval of 100 to 400ms, while at high below.
bitrates a feedback interval of roughly 20ms is to prefer.
The feedback interval depends on the media bitrate. At low bitrates
it is sufficient with a feedback interval of 100 to 400ms, while at
high bitrates a feedback interval of roughly 20ms is to prefer, at
very high bitrates, even shorter feedback intervals MAY be needed in
order to keep the self-clocking in SCReAM working well. One piece of
evidence of a too sparse feedback is that the SCReAM implementation
cannot reach high bitrates, even in uncongested links. A more
frequent feedback MAY solve this issue.
The numbers above can be formulated as feedback interval function The numbers above can be formulated as feedback interval function
that can be useful for the computation of the desired RTCP bandwidth. that can be useful for the computation of the desired RTCP bandwidth.
The following equation expresses the feedback rate: The following equation expresses the feedback rate:
rate_fb = min(50,max(2.5,rate_media/10000)) rate_fb = min(50,max(2.5,rate_media/10000))
rate_media is the RTP media bitrate expressed in [bits/s], rate_fb is rate_media is the RTP media bitrate expressed in [bits/s], rate_fb is
the feedback rate expressed in [packets/s]. Converted to feedback the feedback rate expressed in [packets/s]. Converted to feedback
interval we get: interval we get:
skipping to change at page 27, line 32 skipping to change at page 28, line 13
fb_int = 1.0/min(50,max(2.5,rate_media/10000)) fb_int = 1.0/min(50,max(2.5,rate_media/10000))
The transmission interval is not critical, this means that in the The transmission interval is not critical, this means that in the
case of multi-stream handling between two hosts, the feedback for two case of multi-stream handling between two hosts, the feedback for two
or more SSRCs can be bundled to save UDP/IP overhead, the final or more SSRCs can be bundled to save UDP/IP overhead, the final
realized feedback interval SHOULD however not exceed 2*fb_int in such realized feedback interval SHOULD however not exceed 2*fb_int in such
cases meaning that a scheduled feedback transmission event should not cases meaning that a scheduled feedback transmission event should not
be delayed more that fb_int. be delayed more that fb_int.
SCReAM works with AVPF regular mode, immediate or early mode is not SCReAM works with AVPF regular mode, immediate or early mode is not
REQUIRED by SCReAM but MAY nonetheless be useful for e.g RTCP required by SCReAM but can nonetheless be useful for e.g RTCP
messages not directly related to SCReAM, such as those specified in messages not directly related to SCReAM, such as those specified in
[RFC4585]. It is RECOMMENDED to use reduced size RTCP [RFC5506] [RFC4585]. It is RECOMMENDED to use reduced size RTCP [RFC5506]
where regular full compound RTCP transmission is controlled by trr- where regular full compound RTCP transmission is controlled by trr-
int as described in [RFC4585]. int as described in [RFC4585].
5. Discussion 5. Discussion
This section covers a few discussion points This section covers a few discussion points
o Clock drift: SCReAM can suffer from the same issues with clock o Clock drift: SCReAM can suffer from the same issues with clock
drift as is the case with LEDBAT [RFC6817]. Section A.2 in drift as is the case with LEDBAT [RFC6817]. Section A.2 in
[RFC6817] however describes ways to mitigate issues with clock [RFC6817] however describes ways to mitigate issues with clock
drift. drift.
o Support for alternate ECN semantics: This specification adopts the o Support for alternate ECN semantics: This specification adopts the
proposal in [I-D.ietf-tcpm-alternativebackoff-ecn] to reduce the proposal in [I-D.ietf-tcpm-alternativebackoff-ecn] to reduce the
congestion window less when ECN based congestion events are congestion window less when ECN based congestion events are
detected. Future work on Low Loss Low Latency for Scalable detected. Future work on Low Loss Low Latency for Scalable
throughput (L4S) MAY lead to updates in a future RFC that throughput (L4S) may lead to updates in a future RFC that
describes SCReAM support for L4S. describes SCReAM support for L4S.
o A new RFC4585 transport layer feedback message MAY to be o A new RFC4585 transport layer feedback message could to be
standardized if the use of the already existing RTCP extensions as standardized if the use of the already existing RTCP extensions as
described in Section 4.2 is not deemed sufficient. described in Section 4.2 is not deemed sufficient.
o The target bitrate given by SCReAM depicts the bitrate including o The target bitrate given by SCReAM depicts the bitrate including
RTP and FEC overhead. The media encoder SHOULD take this overhead RTP and FEC overhead. The media encoder SHOULD take this overhead
into account when the media bitrate is set. This means that the into account when the media bitrate is set. This means that the
media coder bitrate SHOULD be computed as media coder bitrate SHOULD be computed as
media_rate = target_bitrate - rtp_plus_fec_overhead_bitrate media_rate = target_bitrate - rtp_plus_fec_overhead_bitrate
skipping to change at page 30, line 36 skipping to change at page 31, line 12
o Trials with different kinds of media: Audio, Video, slide show o Trials with different kinds of media: Audio, Video, slide show
content. Evaluation of multi stream handling in SCReAM. content. Evaluation of multi stream handling in SCReAM.
o Evaluation of functionality of competing flows compensation o Evaluation of functionality of competing flows compensation
mechanism: Evaluate how SCReAM performs with competing TCP like mechanism: Evaluate how SCReAM performs with competing TCP like
traffic and to what extent the competing flows compensation causes traffic and to what extent the competing flows compensation causes
self-inflicted congestion. self-inflicted congestion.
o Determine proper parameters: A set of default parameters are given o Determine proper parameters: A set of default parameters are given
that makes SCReAM work over a reasonably large operation range, that makes SCReAM work over a reasonably large operation range,
however for instance for very low or very high bitrates it MAY be however for instance for very low or very high bitrates it may be
necessary to use different values for instance for the necessary to use different values for instance for the
RAMP_UP_SPEED. RAMP_UP_SPEED.
8. Acknowledgements 8. Acknowledgements
We would like to thank the following persons for their comments, We would like to thank the following persons for their comments,
questions and support during the work that led to this memo: Markus questions and support during the work that led to this memo: Markus
Andersson, Bo Burman, Tomas Frankkila, Frederic Gabin, Laurits Hamm, Andersson, Bo Burman, Tomas Frankkila, Frederic Gabin, Laurits Hamm,
Hans Hannu, Nikolas Hermanns, Stefan Haakansson, Erlendur Karlsson, Hans Hannu, Nikolas Hermanns, Stefan Haakansson, Erlendur Karlsson,
Daniel Lindstroem, Mats Nordberg, Jonathan Samuelsson, Rickard Daniel Lindstroem, Mats Nordberg, Jonathan Samuelsson, Rickard
Sjoeberg, Robert Swain, Magnus Westerlund, Stefan Aalund. Many Sjoeberg, Robert Swain, Magnus Westerlund, Stefan Aalund. Many
additional thanks to RMCAT chairs Karen E. E. Nielsen and Mirja additional thanks to RMCAT chairs Karen E. E. Nielsen and Mirja
Kuehlewind for patiently reading, suggesting improvements and also Kuehlewind for patiently reading, suggesting improvements and also
for asking all the difficult but necessary questions. Thanks to for asking all the difficult but necessary questions. Thanks to
Stefan Holmer, Xiaoqing Zhu, Safiqul Islam and David Hayes for the Stefan Holmer, Xiaoqing Zhu, Safiqul Islam and David Hayes for the
additional review of this document. Thanks to Ralf Globisch for additional review of this document. Thanks to Ralf Globisch for
taking time to try out SCReAM in his challenging low bitrate use taking time to try out SCReAM in his challenging low bitrate use
cases. cases, Robert Hedman for finding a few additional flaws in the
running code, and Gustavo Garcia and 'miseri' for code contributions.
9. IANA Considerations 9. IANA Considerations
There is currently no request to IANA There is currently no request to IANA
10. Security Considerations 10. Security Considerations
The feedback can be vulnerable to attacks similar to those that can The feedback can be vulnerable to attacks similar to those that can
affect TCP. It is therefore RECOMMENDED that the RTCP feedback is at affect TCP. It is therefore RECOMMENDED that the RTCP feedback is at
least integrity protected. Furthermore, as SCReAM is self-clocked, a least integrity protected. Furthermore, as SCReAM is self-clocked, a
malicious middlebox can drop RTCP feedback packets and thus cause the malicious middlebox can drop RTCP feedback packets and thus cause the
self-clocking in SCReAM to stall. This attack is however mitigated self-clocking in SCReAM to stall. This attack is however mitigated
by the minimum send rate maintained by SCReAM when no feedback is by the minimum send rate maintained by SCReAM when no feedback is
received. received.
11. Change history 11. Change history
A list of changes: A list of changes:
o WG-10 to WG-11: Review comments from Mirja
o WG-9 to WG-10: Minor edits
o WG-08 to WG-09: Updated based shepherd review by Martin o WG-08 to WG-09: Updated based shepherd review by Martin
Stiemerling, Q-bit semantics are removed as this is superfluous Stiemerling, Q-bit semantics are removed as this is superfluous
for the moment. Pacing and RTCP considerations are moved up from for the moment. Pacing and RTCP considerations are moved up from
the appendix, FEC discussion moved to discussion section. the appendix, FEC discussion moved to discussion section.
o WG-07 to WG-08: Avoid draft expiry o WG-07 to WG-08: Avoid draft expiry
o WG-06 to WG-07: Updated based on WGLC review by David Hayes and o WG-06 to WG-07: Updated based on WGLC review by David Hayes and
Safiqul Islam Safiqul Islam
skipping to change at page 32, line 22 skipping to change at page 33, line 4
o -04 to -05 : ACK vector is replaced by a loss counter, PT is o -04 to -05 : ACK vector is replaced by a loss counter, PT is
removed from feedback, references to source code added removed from feedback, references to source code added
o -03 to -04 : Extensive changes due to review comments, code o -03 to -04 : Extensive changes due to review comments, code
somewhat modified, frame skipping made optional somewhat modified, frame skipping made optional
o -02 to -03 : Added algorithm description with equations, removed o -02 to -03 : Added algorithm description with equations, removed
pseudo code and simulation results pseudo code and simulation results
o -01 to -02 : Updated GCC simulation results o -01 to -02 : Updated GCC simulation results
o -00 to -01 : Fixed a few bugs in example code o -00 to -01 : Fixed a few bugs in example code
12. References 12. References
12.1. Normative References 12.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997, DOI 10.17487/RFC2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>. <https://www.rfc-editor.org/info/rfc2119>.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control "Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
DOI 10.17487/RFC4585, July 2006, DOI 10.17487/RFC4585, July 2006,
<http://www.rfc-editor.org/info/rfc4585>. <https://www.rfc-editor.org/info/rfc4585>.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, DOI 10.17487/RFC5506, April and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
2009, <http://www.rfc-editor.org/info/rfc5506>. 2009, <https://www.rfc-editor.org/info/rfc5506>.
[RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent, [RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent,
"Computing TCP's Retransmission Timer", RFC 6298, "Computing TCP's Retransmission Timer", RFC 6298,
DOI 10.17487/RFC6298, June 2011, DOI 10.17487/RFC6298, June 2011,
<http://www.rfc-editor.org/info/rfc6298>. <https://www.rfc-editor.org/info/rfc6298>.
[RFC6817] Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind, [RFC6817] Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind,
"Low Extra Delay Background Transport (LEDBAT)", RFC 6817, "Low Extra Delay Background Transport (LEDBAT)", RFC 6817,
DOI 10.17487/RFC6817, December 2012, DOI 10.17487/RFC6817, December 2012,
<http://www.rfc-editor.org/info/rfc6817>. <https://www.rfc-editor.org/info/rfc6817>.
12.2. Informative References 12.2. Informative References
[I-D.ietf-rmcat-coupled-cc] [I-D.ietf-rmcat-coupled-cc]
Islam, S., Welzl, M., and S. Gjessing, "Coupled congestion Islam, S., Welzl, M., and S. Gjessing, "Coupled congestion
control for RTP media", draft-ietf-rmcat-coupled-cc-06 control for RTP media", draft-ietf-rmcat-coupled-cc-07
(work in progress), March 2017. (work in progress), September 2017.
[I-D.ietf-rmcat-wireless-tests] [I-D.ietf-rmcat-wireless-tests]
Sarker, Z., Johansson, I., Zhu, X., Fu, J., Tan, W., and Sarker, Z., Johansson, I., Zhu, X., Fu, J., Tan, W., and
M. Ramalho, "Evaluation Test Cases for Interactive Real- M. Ramalho, "Evaluation Test Cases for Interactive Real-
Time Media over Wireless Networks", draft-ietf-rmcat- Time Media over Wireless Networks", draft-ietf-rmcat-
wireless-tests-04 (work in progress), May 2017. wireless-tests-04 (work in progress), May 2017.
[I-D.ietf-tcpm-alternativebackoff-ecn] [I-D.ietf-tcpm-alternativebackoff-ecn]
Khademi, N., Welzl, M., Armitage, G., and G. Fairhurst, Khademi, N., Welzl, M., Armitage, G., and G. Fairhurst,
"TCP Alternative Backoff with ECN (ABE)", draft-ietf-tcpm- "TCP Alternative Backoff with ECN (ABE)", draft-ietf-tcpm-
skipping to change at page 34, line 5 skipping to change at page 34, line 33
[Packet-conservation] [Packet-conservation]
"Congestion Avoidance and Control, ACM SIGCOMM Computer "Congestion Avoidance and Control, ACM SIGCOMM Computer
Communication Review 1988", 1988. Communication Review 1988", 1988.
[QoS-3GPP] [QoS-3GPP]
TS 23.203, 3GPP., "Policy and charging control TS 23.203, 3GPP., "Policy and charging control
architecture", June 2011, <http://www.3gpp.org/ftp/specs/ architecture", June 2011, <http://www.3gpp.org/ftp/specs/
archive/23_series/23.203/23203-990.zip>. archive/23_series/23.203/23203-990.zip>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <https://www.rfc-editor.org/info/rfc3550>.
[RFC3611] Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed., [RFC3611] Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed.,
"RTP Control Protocol Extended Reports (RTCP XR)", "RTP Control Protocol Extended Reports (RTCP XR)",
RFC 3611, DOI 10.17487/RFC3611, November 2003, RFC 3611, DOI 10.17487/RFC3611, November 2003,
<http://www.rfc-editor.org/info/rfc3611>. <https://www.rfc-editor.org/info/rfc3611>.
[RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P., [RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
and K. Carlberg, "Explicit Congestion Notification (ECN) and K. Carlberg, "Explicit Congestion Notification (ECN)
for RTP over UDP", RFC 6679, DOI 10.17487/RFC6679, August for RTP over UDP", RFC 6679, DOI 10.17487/RFC6679, August
2012, <http://www.rfc-editor.org/info/rfc6679>. 2012, <https://www.rfc-editor.org/info/rfc6679>.
[RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use Cases and Requirements", RFC 7478,
DOI 10.17487/RFC7478, March 2015,
<https://www.rfc-editor.org/info/rfc7478>.
[RFC7661] Fairhurst, G., Sathiaseelan, A., and R. Secchi, "Updating [RFC7661] Fairhurst, G., Sathiaseelan, A., and R. Secchi, "Updating
TCP to Support Rate-Limited Traffic", RFC 7661, TCP to Support Rate-Limited Traffic", RFC 7661,
DOI 10.17487/RFC7661, October 2015, DOI 10.17487/RFC7661, October 2015,
<http://www.rfc-editor.org/info/rfc7661>. <https://www.rfc-editor.org/info/rfc7661>.
[RFC7942] Sheffer, Y. and A. Farrel, "Improving Awareness of Running [RFC7942] Sheffer, Y. and A. Farrel, "Improving Awareness of Running
Code: The Implementation Status Section", BCP 205, Code: The Implementation Status Section", BCP 205,
RFC 7942, DOI 10.17487/RFC7942, July 2016, RFC 7942, DOI 10.17487/RFC7942, July 2016,
<http://www.rfc-editor.org/info/rfc7942>. <https://www.rfc-editor.org/info/rfc7942>.
[SCReAM-CPP-implementation] [SCReAM-CPP-implementation]
"C++ Implementation of SCReAM", "C++ Implementation of SCReAM",
<https://github.com/EricssonResearch/scream>. <https://github.com/EricssonResearch/scream>.
[SCReAM-implementation] [SCReAM-implementation]
"SCReAM Implementation", "SCReAM Implementation",
<https://github.com/EricssonResearch/openwebrtc-gst- <https://github.com/EricssonResearch/
plugins>. openwebrtc-gst-plugins>.
[SCReAM-implementation-experience] [SCReAM-implementation-experience]
"Updates on SCReAM : An implementation experience", "Updates on SCReAM : An implementation experience",
<https://www.ietf.org/proceedings/94/slides/slides-94- <https://www.ietf.org/proceedings/94/slides/
rmcat-8.pdf>. slides-94-rmcat-8.pdf>.
[TFWC] University College London, "Fairer TCP-Friendly Congestion [TFWC] University College London, "Fairer TCP-Friendly Congestion
Control Protocol for Multimedia Streaming", December 2007, Control Protocol for Multimedia Streaming", December 2007,
<http://www-dept.cs.ucl.ac.uk/staff/M.Handley/papers/ <http://www-dept.cs.ucl.ac.uk/staff/M.Handley/papers/
tfwc-conext.pdf>. 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. 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 [SCReAM-CPP-implementation] use credit based scheduling. In
credit based scheduling, credit is accumulated by queues as they wait
for service and are spent while the queues are being serviced. For
instance, if one queue is allowed to transmit 1000bytes, then a
credit of 1000bytes is allocated to the other unscheduled queues.
This principle can be extended to weighted scheduling in which case
the credit allocated to unscheduled queues depends on the relative
weights.
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
Authors' Addresses Authors' Addresses
Ingemar Johansson Ingemar Johansson
Ericsson AB Ericsson AB
Laboratoriegraend 11 Laboratoriegraend 11
Luleaa 977 53 Luleaa 977 53
Sweden Sweden
Phone: +46 730783289 Phone: +46 730783289
Email: ingemar.s.johansson@ericsson.com Email: ingemar.s.johansson@ericsson.com
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