draft-ietf-rmcat-scream-cc-08.txt   draft-ietf-rmcat-scream-cc-09.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: November 11, 2017 May 10, 2017 Expires: November 30, 2017 May 29, 2017
Self-Clocked Rate Adaptation for Multimedia Self-Clocked Rate Adaptation for Multimedia
draft-ietf-rmcat-scream-cc-08 draft-ietf-rmcat-scream-cc-09
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.
skipping to change at page 1, line 36 skipping to change at page 1, line 36
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on November 11, 2017. This Internet-Draft will expire on November 30, 2017.
Copyright Notice Copyright Notice
Copyright (c) 2017 IETF Trust and the persons identified as the Copyright (c) 2017 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License. described in the Simplified BSD License.
This document may contain material from IETF Documents or IETF
Contributions published or made publicly available before November
10, 2008. The person(s) controlling the copyright in some of this
material may not have granted the IETF Trust the right to allow
modifications of such material outside the IETF Standards Process.
Without obtaining an adequate license from the person(s) controlling
the copyright in such materials, this document may not be modified
outside the IETF Standards Process, and derivative works of it may
not be created outside the IETF Standards Process, except to format
it for publication as an RFC or to translate it into languages other
than English.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. Wireless (LTE) access properties . . . . . . . . . . . . 3 1.1. Wireless (LTE) access properties . . . . . . . . . . . . 3
1.2. Why is it a self-clocked algorithm? . . . . . . . . . . . 4 1.2. Why is it a self-clocked algorithm? . . . . . . . . . . . 4
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Overview of SCReAM Algorithm . . . . . . . . . . . . . . . . 4 3. Overview of SCReAM Algorithm . . . . . . . . . . . . . . . . 4
3.1. Network Congestion Control . . . . . . . . . . . . . . . 8 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 . . . . . . . . . . . . . . . . . . . . 10 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. Congestion window update . . . . . . . . . . . . 16
4.1.2.2. Competing flows compensation . . . . . . . . . . 18 4.1.2.2. Competing flows compensation . . . . . . . . . . 18
4.1.2.3. Lost packet detection . . . . . . . . . . . . . . 20 4.1.2.3. Lost packet detection . . . . . . . . . . . . . . 20
4.1.2.4. Send window calculation . . . . . . . . . . . . . 20 4.1.2.4. Send window calculation . . . . . . . . . . . . . 20
4.1.2.5. Resuming fast increase . . . . . . . . . . . . . 21 4.1.2.5. Packet pacing . . . . . . . . . . . . . . . . . . 21
4.1.3. Media rate control . . . . . . . . . . . . . . . . . 21 4.1.2.6. Resuming fast increase . . . . . . . . . . . . . 21
4.1.3.1. FEC and packet overhead considerations . . . . . 24 4.1.3. Media rate control . . . . . . . . . . . . . . . . . 22
4.2. SCReAM Receiver . . . . . . . . . . . . . . . . . . . . . 25 4.2. SCReAM Receiver . . . . . . . . . . . . . . . . . . . . . 25
5. Discussion . . . . . . . . . . . . . . . . . . . . . . . . . 25 4.2.1. Requirements on feedback elements . . . . . . . . . . 25
6. Implementation status . . . . . . . . . . . . . . . . . . . . 25 4.2.2. Requirements on feedback intensity . . . . . . . . . 27
6.1. OpenWebRTC . . . . . . . . . . . . . . . . . . . . . . . 26 5. Discussion . . . . . . . . . . . . . . . . . . . . . . . . . 27
6.2. A C++ Implementation of SCReAM . . . . . . . . . . . . . 27 6. Implementation status . . . . . . . . . . . . . . . . . . . . 28
7. Suggested experiments . . . . . . . . . . . . . . . . . . . . 27 6.1. OpenWebRTC . . . . . . . . . . . . . . . . . . . . . . . 29
8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 28 6.2. A C++ Implementation of SCReAM . . . . . . . . . . . . . 29
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 28 7. Suggested experiments . . . . . . . . . . . . . . . . . . . . 30
10. Security Considerations . . . . . . . . . . . . . . . . . . . 28 8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 30
11. Change history . . . . . . . . . . . . . . . . . . . . . . . 28 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 31
12. References . . . . . . . . . . . . . . . . . . . . . . . . . 29 10. Security Considerations . . . . . . . . . . . . . . . . . . . 31
12.1. Normative References . . . . . . . . . . . . . . . . . . 29 11. Change history . . . . . . . . . . . . . . . . . . . . . . . 31
12.2. Informative References . . . . . . . . . . . . . . . . . 30 12. References . . . . . . . . . . . . . . . . . . . . . . . . . 32
Appendix A. Additional information . . . . . . . . . . . . . . . 32 12.1. Normative References . . . . . . . . . . . . . . . . . . 32
A.1. Stream prioritization . . . . . . . . . . . . . . . . . . 32 12.2. Informative References . . . . . . . . . . . . . . . . . 33
A.2. Computation of autocorrelation function . . . . . . . . . 32 Appendix A. Additional information . . . . . . . . . . . . . . . 35
A.3. Sender transmission control and packet pacing . . . . . . 33 A.1. Stream prioritization . . . . . . . . . . . . . . . . . . 35
A.4. RTCP feedback considerations . . . . . . . . . . . . . . 33 A.2. Computation of autocorrelation function . . . . . . . . . 35
A.4.1. Requirements on feedback elements . . . . . . . . . . 33 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 35
A.4.2. Requirements on feedback intensity . . . . . . . . . 35
A.5. Q-bit semantics (source quench) . . . . . . . . . . . . . 36
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 37
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 MUST employ congestion
control, to achieve robust performance and to avoid congestion control, to achieve robust performance and to avoid congestion
collapse in the Internet. Interactive realtime communication imposes collapse in the Internet. Interactive realtime communication imposes
a lot of requirements on the transport, therefore a robust, efficient a lot of requirements on the transport, therefore a robust, efficient
rate adaptation for all access types is an important part of rate adaptation for all access types is an important part of
interactive realtime communications as the transmission channel interactive realtime communications as the transmission channel
bandwidth may vary over time. Wireless access such as LTE, which is bandwidth MAY vary over time. Wireless access such as LTE, which is
an integral part of the current Internet, increases the importance of an integral part of the current Internet, increases the importance of
rate adaptation as the channel bandwidth of a default LTE bearer rate adaptation as the channel bandwidth of a default LTE bearer
[QoS-3GPP] can change considerably in a very short time frame. Thus [QoS-3GPP] can change considerably in a very short time frame. Thus
a rate adaptation solution for interactive realtime media, such as a rate adaptation solution for interactive realtime media, such as
WebRTC, must be both quick and be able to operate over a large range WebRTC, SHOULD be both quick and be able to operate over a large
in channel capacity. This memo describes SCReAM (Self-Clocked Rate range in channel capacity. This memo describes SCReAM (Self-Clocked
Adaptation for Multimedia), a solution that is based on the self- Rate Adaptation for Multimedia), a solution that is based on the
clocking principle of TCP and uses techniques similar to what is used self-clocking principle of TCP and uses techniques similar to what is
in the LEDBAT based rate adaptation algorithm [RFC6817]. SCReAM is used in the LEDBAT based rate adaptation algorithm [RFC6817]. SCReAM
not entirely self-clocked as it augments self-clocking with pacing is not entirely self-clocked as it augments self-clocking with pacing
and a minimum send rate. 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 MAY 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. coverage. An additional complication is that the network throughput
MAY drop for short time intervals at e.g. handover, these short
glitches are initially very difficult to distinguish from more
permanent reductions in throughput.
Unlike wireline bottlenecks with large statistical multiplexing it is 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 needs to reduce allocation for a bearer, an RMCAT flow in that bearer SHOULD 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 6, line 27 skipping to change at page 6, line 41
o Addition of a media rate control function. o Addition of a media rate control function.
o Use of inflection points in the media rate calculation to achieve o Use of inflection points in the media rate calculation to achieve
reduced jitter. reduced jitter.
o Adjustment of qdelay target for better performance when competing o Adjustment of qdelay target for better performance when competing
with other loss based congestion controlled flows. with other loss based congestion controlled flows.
The above mentioned features will be described in more detail in The above mentioned features will be described in more detail in
sections Section 3.1 to Section 3.3. sections Section 3.1 to Section 3.3. The full details are described
in Section 4.
+---------------------------+ +---------------------------+
| Media encoder | | Media encoder |
+---------------------------+ +---------------------------+
^ | ^ |
(3)| (1)| | |(1)
| RTP |(3) RTP
| V | V
| +-----------+ | +-----------+
+---------+ | | +---------+ | |
| Media | (2) | Queue | | Media | (2) | Queue |
| rate |<------| | | rate |<------| |
| control | |RTP packets| | control | |RTP packets|
+---------+ | | +---------+ | |
+-----------+ +-----------+
| |
| |(4)
(4)|
RTP RTP
| |
v v
+------------+ +--------------+ +------------+ +--------------+
| Network | (7) | Sender | | Network | (7) | Sender |
+-->| congestion |------>| Transmission | +-->| congestion |------>| Transmission |
| | control | | Control | | | control | | Control |
| +------------+ +--------------+ | +------------+ +--------------+
| | | |
| (6) |(5) |-------------RTCP----------| |(5)
|-------------RTCP----------| RTP (6) | RTP
| |
| v | v
+------------+ +------------+
| UDP | | UDP |
| socket | | socket |
+------------+ +------------+
Figure 1: SCReAM sender functional view Figure 1: SCReAM sender functional view
The SCReAM algorithm consists of three main parts: network congestion The SCReAM algorithm consists of three main parts: network congestion
control, sender transmission control and media rate control. All of control, sender transmission control and media rate control. All of
skipping to change at page 8, line 35 skipping to change at page 8, line 29
time media such as video, typically means that the queuing delay will 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 mostly be below the given delay target, this is contrary to the case
where large files are transmitted using LEDBAT congestion control, in where large files are transmitted using LEDBAT congestion control, in
which case the queuing delay will stay close to the delay target. which case the queuing delay will stay close to the delay target.
3.2. Sender Transmission Control 3.2. Sender Transmission Control
The sender transmission control limits the output of data, given by The sender transmission control limits the output of data, given by
the relation between the number of bytes in flight and the congestion the relation between the number of bytes in flight and the congestion
window. Packet pacing is used to mitigate issues with ACK window. Packet pacing is used to mitigate issues with ACK
compression that may cause increased jitter and/or packet loss in the compression that MAY cause increased jitter and/or packet loss in the
media traffic. Packet pacing limits the packet transmission rate media traffic. Packet pacing limits the packet transmission rate
given by the estimated link throughput. Even if the send window given by the estimated link throughput. Even if the send window
allows for the transmission of a number of packets, these packets are allows for the transmission of a number of packets, these packets are
not transmitted immediately, but rather they are transmitted in not transmitted immediately, but rather they are transmitted in
intervals given by the packet size and the estimated link throughput. intervals given by the packet size and the estimated link throughput.
3.3. Media Rate Control 3.3. Media Rate Control
The media rate control serves to adjust the media bitrate to ramp-up 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 quickly enough to get a fair share of the system resources when link
throughput increases. throughput increases.
The reaction to reduced throughput must be prompt in order to avoid The reaction to reduced throughput MUST be prompt in order to avoid
getting too much data queued in the RTP packet queue(s) in the getting too much data queued in the RTP packet queue(s) in the
sender. The media bitrate is decreased if the RTP queue size exceeds sender. The media bitrate is decreased if the RTP queue size exceeds
a threshold. a threshold.
In cases where the sender frame queues increase rapidly such as in In cases where the sender frame queues increase rapidly such as in
the case of a RAT (Radio Access Type) handover it may be necessary to the case of a RAT (Radio Access Type) handover it MAY be necessary to
implement additional actions, such as discarding of encoded media implement additional actions, such as discarding of encoded media
frames or frame skipping in order to ensure that the RTP queues are frames or frame skipping in order to ensure that the RTP queues are
drained quickly. Frame skipping results in the frame rate being drained quickly. Frame skipping results in the frame rate being
temporarily reduced. Which method to use is a design choice and temporarily reduced. Which method to use is a design choice and
outside the scope of this algorithm description. outside the scope of this algorithm description.
4. Detailed Description of SCReAM 4. Detailed Description of SCReAM
4.1. SCReAM Sender 4.1. SCReAM Sender
skipping to change at page 9, line 39 skipping to change at page 9, line 33
implemented in [SCReAM-CPP-implementation]. implemented in [SCReAM-CPP-implementation].
Media frames are encoded and forwarded to the RTP queue (1) in Media frames are encoded and forwarded to the RTP queue (1) in
Figure 1. The media rate adaptation adapts to the size of the RTP Figure 1. The media rate adaptation adapts to the size of the RTP
queue (2) and provides a target rate for the media encoder (3). The queue (2) and provides a target rate for the media encoder (3). The
RTP packets are picked from the RTP queue (for multiple flows from 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 each RTP queue based on some defined priority order or simply in a
round robin fashion) (4) by the sender transmission controller. The round robin fashion) (4) by the sender transmission controller. The
sender transmission controller (in case of multiple flows a sender transmission controller (in case of multiple flows a
transmission scheduler) sends the RTP packets to the UDP socket (5). transmission scheduler) sends the RTP packets to the UDP socket (5).
In the general case all media must go through the sender transmission In the general case all media SHOULD go through the sender
controller and is limited so that the number of bytes in flight is transmission controller and is limited so that the number of bytes in
less than the congestion window. RTCP packets are received (6) and flight is less than the congestion window. RTCP packets are received
the information about bytes in flight and congestion window is (6) and the information about bytes in flight and congestion window
exchanged between the network congestion control and the sender is exchanged between the network congestion control and the sender
transmission control (7). transmission control (7).
4.1.1. Constants and Parameter values 4.1.1. Constants and Parameter values
Constants and state variables are listed in this section. Temporary Constants and state variables are listed in this section. Temporary
variables are not listed, instead they are appended with '_t' in the variables are not listed, instead they are appended with '_t' in the
pseudo code to indicate their local scope. pseudo code to indicate their local scope.
4.1.1.1. Constants 4.1.1.1. Constants
The recommended values, within (), for the constants are deduced from The RECOMMENDED values, within (), for the constants are deduced from
experiments. 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 should not be initialized to this high value however target qdelay MUST not be initialized to this high value however as
as it would increase e2e delay and also make the rate control and it would increase e2e delay and also make the rate control and
congestion control loop sluggish. 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.
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 (0.6) 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)
CWND scale factor due to ECN event. CWND scale factor due to ECN event.
BETA_R (0.9) BETA_R (0.9)
Target rate scale factor due to loss event. Target rate scale factor due to loss event.
MSS (1000 byte) MSS (1000 byte)
Maximum segment size = Max RTP packet size. Maximum segment size = Max RTP packet size.
RATE_ADJUST_INTERVAL (0.2s) RATE_ADJUST_INTERVAL (0.2s)
Interval between media bitrate adjustments. Interval between media bitrate adjustments.
TARGET_BITRATE_MIN TARGET_BITRATE_MIN
Min target bitrate [bps]. Min target bitrate [bps], bps is bits per second.
TARGET_BITRATE_MAX TARGET_BITRATE_MAX
Max target bitrate [bps]. Max target bitrate [bps].
RAMP_UP_SPEED (200000bps/s) RAMP_UP_SPEED (200000bps/s)
Maximum allowed rate increase speed. Maximum allowed rate increase speed.
PRE_CONGESTION_GUARD (0.0..1.0) PRE_CONGESTION_GUARD (0.0..1.0)
Guard factor against early congestion onset. A higher value gives Guard factor against early congestion onset. A higher value gives
less jitter, possibly at the expense of a lower link utilization. less jitter, possibly at the expense of a lower link utilization.
This value may be subject to tuning depending on e.g media coder This value MAY be subject to tuning depending on e.g media coder
characteristics, experiments with H264 and VP8 indicate that 0.1 is characteristics, experiments with H264 and VP8 indicate that 0.1 is
a suitable value. See [SCReAM-implementation-experience] for a suitable value. See [SCReAM-CPP-implementation] and
evaluation of a real implementation. [SCReAM-implementation-experience] for evaluation of a real
implementation.
TX_QUEUE_SIZE_FACTOR (0.0..2.0) TX_QUEUE_SIZE_FACTOR (0.0..2.0)
Guard factor against RTP queue buildup. This value may be subject Guard factor against RTP queue buildup. This value MAY be subject
to tuning depending on e.g media coder characteristics, experiments to tuning depending on e.g media coder characteristics, experiments
with H264 and VP8 indicate that 1.0 is a suitable value. See with H264 and VP8 indicate that 1.0 is a suitable value. See
[SCReAM-implementation-experience] for evaluation of a real [SCReAM-CPP-implementation] and [SCReAM-implementation-experience]
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 Time span until fast increase can be resumed,
skipping to change at page 13, line 28 skipping to change at page 13, line 25
limit to the number of bytes in flight. limit to the number of bytes in flight.
o Calculation of send window at the sender: RTP packets are o Calculation of send window at the sender: RTP packets are
transmitted if allowed by the relation between the number of bytes transmitted if allowed by the relation between the number of bytes
in flight and the congestion window. This is controlled by the in flight and the congestion window. This is controlled by the
send window. send window.
SCReAM is a window based and byte oriented congestion control SCReAM is a window based and byte oriented congestion control
protocol, where the number of bytes transmitted is inferred from the protocol, where the number of bytes transmitted is inferred from the
size of the transmitted RTP packets. Thus a list of transmitted RTP size of the transmitted RTP packets. Thus a list of transmitted RTP
packets and their respective transmission times (wall-clock time) is packets and their respective transmission times (wall-clock time)
kept for further calculation. MUST be kept for further calculation.
The number of bytes in flight (bytes_in_flight) is computed as the The number of bytes in flight (bytes_in_flight) is computed as the
sum of the sizes of the RTP packets ranging from the RTP packet most sum of the sizes of the RTP packets ranging from the RTP packet most
recently transmitted down to but not including the acknowledged recently transmitted down to but not including the acknowledged
packet with the highest sequence number. This can be translated to packet with the highest sequence number. This can be translated to
the difference between the highest transmitted byte sequence number the difference between the highest transmitted byte sequence number
and the highest acknowledged byte sequence number. As an example: If and the highest acknowledged byte sequence number. As an example: If
RTP packet with sequence number SN is transmitted and the last RTP packet with sequence number SN is transmitted and the last
acknowledgement indicates SN-5 as the highest received sequence acknowledgement indicates SN-5 as the highest received sequence
number then bytes in flight is computed as the sum of the size of RTP number then bytes in flight is computed as the sum of the size of RTP
skipping to change at page 14, line 6 skipping to change at page 13, line 51
Furthermore, a variable bytes_newly_acked is incremented with a value Furthermore, a variable bytes_newly_acked is incremented with a value
corresponding to how much the highest sequence number has increased corresponding to how much the highest sequence number has increased
since the last feedback. As an example: If the previous since the last feedback. As an example: If the previous
acknowledgement indicated the highest sequence number N and the new acknowledgement indicated the highest sequence number N and the new
acknowledgement indicated N+3, then bytes_newly_acked is incremented 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 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, 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 which means that even though e.g packet N+2 was lost, its size is
still included in the update of bytes_newly_acked. The still included in the update of bytes_newly_acked. The
bytes_newly_acked variable is reset after a CWND update. bytes_newly_acked variable is reset to zero after a CWND update.
The feedback from the receiver is assumed to consist of the following The feedback from the receiver is assumed to consist of the following
elements. More details are found in Appendix A.4. elements.
o A list of received RTP packets. o A list of received RTP packets' sequence numbers.
o The wall clock timestamp corresponding to the received RTP packet o The wall clock timestamp corresponding to the received RTP packet
with the highest sequence number. with the highest sequence number.
o Accumulated number of ECN-CE marked packets (n_ECN). o Accumulated number of ECN-CE marked packets (n_ECN).
When the sender receives RTCP feedback, the qdelay is calculated as When the sender receives RTCP feedback, the qdelay is calculated as
outlined in [RFC6817]. A qdelay sample is obtained for each received outlined in [RFC6817]. A qdelay sample is obtained for each received
acknowledgement. No smoothing of the qdelay samples occur, however acknowledgement. No smoothing of the qdelay samples occur, however
some smoothing occurs anyway as the computation of the CWND is a low some smoothing occurs anyway as the computation of the CWND is a low
pass filter function. A number of variables are updated as pass filter function. A number of variables are updated as
illustrated by the pseudo code below, temporary variables are illustrated by the pseudo code below, temporary variables are
appended with '_t'. Note that the pseudo code does not show all appended with '_t'. Note that the pseudo code does not show all
details for reasons of readability, the reader is encouraged to look details for reasons of readability, the reader is encouraged to look
into the C++ code in [SCReAM-CPP-implementation] for the details. into the C++ code in [SCReAM-CPP-implementation] for the details.
<CODE BEGINS>
update_variables(qdelay): update_variables(qdelay):
qdelay_fraction_t = qdelay/qdelay_target qdelay_fraction_t = qdelay/qdelay_target
#calculate moving average #calculate moving average
qdelay_fraction_avg = (1-QDELAY_WEIGHT)*qdelay_fraction_avg+ qdelay_fraction_avg = (1-QDELAY_WEIGHT)*qdelay_fraction_avg+
QDELAY_WEIGHT*qdelay_fraction_t QDELAY_WEIGHT*qdelay_fraction_t
update_qdelay_fraction_hist(qdelay_fraction_t) update_qdelay_fraction_hist(qdelay_fraction_t)
# R is an autocorrelation function of qdelay_fraction_hist # R is an autocorrelation function of qdelay_fraction_hist
# at lag K # at lag K
a = R(qdelay_fraction_hist,1)/R(qdelay_fraction_hist,0) a = R(qdelay_fraction_hist,1)/R(qdelay_fraction_hist,0)
#calculate qdelay trend #calculate qdelay trend
qdelay_trend = min(1.0,max(0.0,a*qdelay_fraction_avg)) qdelay_trend = min(1.0,max(0.0,a*qdelay_fraction_avg))
#calculate a 'peak-hold' qdelay_trend, this gives a memory #calculate a 'peak-hold' qdelay_trend, this gives a memory
# 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>
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. The autocorrelation function 'R' is defined in
Appendix A.2. The prediction coefficient is further multiplied with Appendix A.2. 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 a 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.
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
skipping to change at page 15, line 28 skipping to change at page 15, line 26
loss event is detected, further detected lost RTP packets are ignored loss event is detected, further detected lost RTP packets are ignored
for a full smoothed round trip time, the intention of this is to for a full smoothed round trip time, the intention of this is to
limit the congestion window decrease to at most once per round trip. 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 necessary to let SCReAM file transfer. To compensate for this it is RECOMMENDED to let
reduce the congestion window slightly 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.
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 is deliberately smaller than if a window back off due to an ECN event MAY be smaller than if a loss
loss event occurs. This is in line with the idea outlined in event occurs. This is in line with the idea outlined in
[Khademi-alternative-backoff-ECN] to enable ECN marking thresholds [I-D.ietf-tcpm-alternativebackoff-ecn] to enable ECN marking
lower than the corresponding packet drop thresholds. thresholds lower than the corresponding packet drop thresholds.
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>
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>
The methods are further described in detail below. The methods are further described in detail below.
4.1.2.1. Congestion window update 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>
update_cwnd(bytes_newly_acked): update_cwnd(bytes_newly_acked):
# in fast increase ? # in fast increase ?
if (in_fast_increase) if (in_fast_increase)
if (qdelay_trend >= QDELAY_TREND_TH) if (qdelay_trend >= QDELAY_TREND_TH)
# incipient congestion detected, exit fast increase # incipient congestion detected, exit fast increase
in_fast_increase = false in_fast_increase = false
else else
# no congestion yet, increase cwnd if it # no congestion yet, increase cwnd if it
# is sufficiently used # is sufficiently used
skipping to change at page 17, line 47 skipping to change at page 17, line 48
# even when feedback is sparse # even when feedback is sparse
cwnd_delta_t = 0; cwnd_delta_t = 0;
end end
# apply delta # apply delta
cwnd += cwnd_delta_t cwnd += cwnd_delta_t
# limit cwnd to the maximum number of bytes in flight # limit cwnd to the maximum number of bytes in flight
cwnd = min(cwnd, max_bytes_in_flight*MAX_BYTES_IN_FLIGHT_HEAD_ROOM) cwnd = min(cwnd, max_bytes_in_flight*MAX_BYTES_IN_FLIGHT_HEAD_ROOM)
cwnd = max(cwnd, MIN_CWND) cwnd = max(cwnd, MIN_CWND)
<CODE ENDS>
CWND is updated differently depending on whether the congestion CWND is updated differently depending on whether the congestion
control is in fast increase state or not, as controlled by the control is in fast increase state or not, as controlled by the
variable in_fast_increase. variable in_fast_increase.
When in fast increase state, the congestion window is increased with When in fast increase state, the congestion window is increased with
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.
skipping to change at page 19, line 5 skipping to change at page 19, line 5
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 4.1.2.2. 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>
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) oh_t = qdelay_norm_avg_t + sqrt(qdelay_norm_var_t)
skipping to change at page 19, line 40 skipping to change at page 19, line 41
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)
<CODE ENDS>
The qdelay_target is adjusted differently, depending on if The qdelay_target is adjusted differently, depending on if
qdelay_norm_var_t is above or below a given value. qdelay_norm_var_t is above or below a given value.
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. It is therefore
recommended that the algorithm described in this section is turned RECOMMENDED that the algorithm described in this section is turned
off it is deemed unlikely that competing flows occur over the same off it is deemed unlikely that competing flows occur over the same
bottleneck bottleneck
4.1.2.3. Lost packet detection 4.1.2.3. 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 packet reordering
triggering loss events. triggering 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) [RACK]. The computation ideas behind RACK (Recent ACKnowledgement) [I-D.ietf-tcpm-rack]. The
of the reordering window is made possible by means of a lost flag in computation of the reordering window is made possible by means of a
the list of transmitted RTP packets. This flag is set if the lost flag in the list of transmitted RTP packets. This flag is set
received sequence number list indicates that the given RTP packet is if the received sequence number list indicates that the given RTP
missing. If a later feedback indicates that a previously lost marked packet is missing. If a later feedback indicates that a previously
packet was indeed received, then the reordering window is updated to lost marked packet was indeed received, then the reordering window is
reflect the reordering delay. The reordering window is given by the updated to reflect the reordering delay. The reordering window is
difference in time between the event that the packet was marked as given by the difference in time between the event that the packet was
lost and the event that it was indicated as successfully received. marked as lost and the event that it was indicated as successfully
received.
Loss is detected if a given RTP packet is not acknowledged within a 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 4.1.2.4. 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 MAY 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
quickly when no congestion is detected while still being able to give quickly when no congestion is detected while still being able to give
a stable behavior in congested situations. a stable behavior in congested situations.
The send window is given by the relation between the adjusted The send window is given by the relation between the adjusted
congestion window and the amount of bytes in flight according to the congestion window and the amount of bytes in flight according to the
pseudo code below. pseudo code below.
<CODE BEGINS>
calculate_send_window(qdelay, qdelay_target) calculate_send_window(qdelay, qdelay_target)
# send window is computed differently depending on congestion level # send window is computed differently depending on congestion level
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>
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. More details around sender an RTCP feedback messaged is received.
transmission control and packet pacing are found in Appendix A.3.
4.1.2.5. Resuming fast increase 4.1.2.5. 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. The time interval
between consecutive packet transmissions is enforced to be equal to
or higher than t_pace where t_pace is given by the equations below :
<CODE BEGINS>
pace_bitrate = max (RATE_PACE_MIN, cwnd* 8 / s_rtt)
t_pace = rtp_size * 8 / pace_bitrate
<CODE ENDS>
rtp_size is the size of the last transmitted RTP packet, s_rtt is the
smoothed round trip time. RATE_PACE_MIN=50000 is the minimum pacing
rate.
4.1.2.6. 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.
4.1.3. 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 the flow being starved out by other more
opportunistic traffic. On the other hand too aggressive a setting opportunistic traffic. On the other hand, a too aggressive setting
leads to extra jitter 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. RTP packets being queued up when the network throughput becomes low.
The sender should also be equipped with a mechanism that discards RTP The sender SHOULD also be equipped with a mechanism that discards RTP
packets in cases where the network throughput becomes very low and packets in cases where the network throughput becomes very low and
RTP packets are excessively delayed. 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
time unit. 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 MAY 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>
# 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
on loss: on loss:
# Loss event detected # Loss event detected
target_bitrate = max(BETA_R* target_bitrate, TARGET_BITRATE_MIN) target_bitrate = max(BETA_R* target_bitrate, TARGET_BITRATE_MIN)
exit exit
on ecn_mark: on ecn_mark:
# ECN event detected # ECN event detected
target_bitrate = max(BETA_ECN* target_bitrate, TARGET_BITRATE_MIN) target_bitrate = max(BETA_ECN* target_bitrate, TARGET_BITRATE_MIN)
skipping to change at page 23, line 33 skipping to change at page 24, line 8
if (rtp_queue_delay_t > RTP_QDELAY_TH) if (rtp_queue_delay_t > RTP_QDELAY_TH)
target_bitrate *= TARGET_RATE_SCALE_RTP_QDELAY target_bitrate *= TARGET_RATE_SCALE_RTP_QDELAY
end end
end end
rate_media_limit_t = max(current_rate_t, max(rate_media,rtp_rate_median)) rate_media_limit_t = max(current_rate_t, max(rate_media,rtp_rate_median))
rate_media_limit_t *= (2.0-qdelay_trend_mem) rate_media_limit_t *= (2.0-qdelay_trend_mem)
target_bitrate = min(target_bitrate, rate_media_limit_t) target_bitrate = min(target_bitrate, rate_media_limit_t)
target_bitrate = min(TARGET_BITRATE_MAX, target_bitrate = min(TARGET_BITRATE_MAX,
max(TARGET_BITRATE_MIN,target_bitrate)) max(TARGET_BITRATE_MIN,target_bitrate))
<CODE ENDS>
In case of a loss event the target_bitrate is updated and the rate In case of a loss event the target_bitrate is updated and the rate
change procedure is exited. Otherwise the rate change procedure change procedure is exited. Otherwise the rate change procedure
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-implementation-experience]. A too short interval is shown to [SCReAM-CPP-implementation]. A too short interval is shown to make
make the video coder internal rate control loop more unstable, a too the video coder internal rate control loop more unstable, a too long
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 higher risk of 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.
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.
In cases where input stimuli to the media encoder is static, for
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 the media coder input stimuli changes 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 the history of media coder
output bitrates, rate_media_limit. A multiplier is applied to
rate_media_limit, depending on congestion history. The
target_bitrate is then limited by this rate_media_limit.
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 more jitter in congested after congestion, at the expense of increased jitter in congested
situations. situations.
4.1.3.1. FEC and packet overhead considerations
The target bitrate given by SCReAM depicts the 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. This
means that the media coder bitrate should be computed as
media_rate = target_bitrate - rtp_plus_fec_overhead_bitrate
It is not strictly necessary to make a 100% perfect compensation for
the overhead as the SCReAM algorithm will inherently compensate for
moderate errors. Under-compensation of the overhead has the effect
of increasing jitter while overcompensation will have the effect of
causing the bottleneck link to become under-utilized.
4.2. SCReAM Receiver 4.2. SCReAM Receiver
The simple task of the SCReAM receiver is to feedback The simple task of the SCReAM receiver is to feedback
acknowledgements of received packets and total ECN count to the acknowledgements of received packets and total ECN count to the
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. More details of the feedback and the the available RTCP bandwidth. The requirements on the feedback
frequency is found in Appendix A.4. elements and the feedback interval is described.
4.2.1. Requirements on feedback elements
SCReAM requires the following elements for its basic functionality,
i.e. only including features that are strictly necessary in order to
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
long to cover all received RTP packets. This list can be realized
with the Loss RLE report block in [RFC3611].
o A wall clock timestamp corresponding to the received RTP packet
with the highest sequence number is required in order to compute
the qdelay. This can be realized by means of the Packet Receipt
Times Report Block in [RFC3611]. begin_seq MUST be set to the
highest received (possibly wrapped around) sequence number,
end_seq MUST be set to begin_seq+1 % 65536. The timestamp clock
MAY be set according to [RFC3611] i.e. equal to the RTP timestamp
clock. Detailed individual packet receive times is not necessary
as SCReAM does currently not describe how this can be used.
The basic feedback needed for SCReAM involves the use of the Loss RLE
report block and the Packet Receipt Times block defined in Figure 2.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|reserved | PT=XR=207 | length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| BT=2 | rsvd. | T=0 | block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| begin_seq | end_seq |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| chunk 1 | chunk 2 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| chunk n-1 | chunk n |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| BT=3 | rsvd. | T=0 | block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| begin_seq | end_seq |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Receipt time of packet begin_seq |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2: Basic feedback message for SCReAM, based on RFC3611
In a typical use case, no more than four Loss RLE chunks 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, could reduce the feedback
message size a bit.
Additional feedback elements that can improve the performance of
SCReAM are:
o Accumulated number of ECN-CE marked packets (n_ECN). This can for
instance be realized with the ECN Feedback Report Format in
[RFC6679]. The given feedback report format is actually a slight
overkill as SCReAM would do quite well with only a counter that
increments by one for each received packet with the ECN-CE code
point set. The more bulky format MAY be nevertheless be useful
for e.g ECN black-hole detection.
4.2.2. Requirements on feedback intensity
SCReAM benefits from a relatively frequent feedback. 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.
The numbers above can be formulated as feedback interval function
that can be useful for the computation of the desired RTCP bandwidth.
The following equation expresses the feedback rate:
rate_fb = min(50,max(2.5,rate_media/10000))
rate_media is the RTP media bitrate expressed in [bits/s], rate_fb is
the feedback rate expressed in [packets/s]. Converted to feedback
interval we get:
fb_int = 1.0/min(50,max(2.5,rate_media/10000))
The transmission interval is not critical, this means that in the
case of multi-stream handling between two hosts, the feedback for two
or more SSRCs can be bundled to save UDP/IP overhead, the final
realized feedback interval SHOULD however not exceed 2*fb_int in such
cases meaning that a scheduled feedback transmission event should not
be delayed more that fb_int.
SCReAM works with AVPF regular mode, immediate or early mode is not
REQUIRED by SCReAM but MAY nonetheless be useful for e.g RTCP
messages not directly related to SCReAM, such as those specified in
[RFC4585]. It is RECOMMENDED to use reduced size RTCP [RFC5506]
where regular full compound RTCP transmission is controlled by trr-
int as described in [RFC4585].
5. Discussion 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 [Khademi-alternative-backoff-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 Latency Low Loss 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
standardized if the use of the already existing RTCP extensions as
described in Section 4.2 is not deemed sufficient.
o The target bitrate given by SCReAM depicts the bitrate including
RTP and FEC overhead. The media encoder SHOULD take this overhead
into account when the media bitrate is set. This means that the
media coder bitrate SHOULD be computed as
media_rate = target_bitrate - rtp_plus_fec_overhead_bitrate
It is not strictly necessary to make a 100% perfect compensation
for the overhead as the SCReAM algorithm will inherently
compensate for moderate errors. Under-compensation of the
overhead has the effect of increasing jitter while
overcompensation will have the effect of causing the bottleneck
link to become under-utilized.
6. Implementation status 6. Implementation status
[Editor's note: Please remove the whole section before publication, [Editor's note: Please remove the whole section before publication,
as well reference to RFC 6982] as well reference to RFC 6982]
This section records the status of known implementations of the This section records the status of known implementations of the
protocol defined by this specification at the time of posting of this protocol defined by this specification at the time of posting of this
Internet-Draft, and is based on a proposal described in [RFC6982]. Internet-Draft, and is based on a proposal described in [RFC6982].
The description of implementations in this section is intended to The description of implementations in this section is intended to
assist the IETF in its decision processes in progressing drafts to assist the IETF in its decision processes in progressing drafts to
RFCs. Please note that the listing of any individual implementation RFCs. Please note that the listing of any individual implementation
here does not imply endorsement by the IETF. Furthermore, no effort here does not imply endorsement by the IETF. Furthermore, no effort
has been spent to verify the information presented here that was has been spent to verify the information presented here that was
supplied by IETF contributors. This is not intended as, and must not supplied by IETF contributors. This is not intended as, and MUST not
be construed to be, a catalog of available implementations or their be construed to be, a catalog of available implementations or their
features. Readers are advised to note that other implementations may features. Readers are advised to note that other implementations MAY
exist. exist.
According to [RFC6982], "this will allow reviewers and working groups According to [RFC6982], "this will allow reviewers and working groups
to assign due consideration to documents that have the benefit of to assign due consideration to documents that have the benefit of
running code, which may serve as evidence of valuable experimentation running code, which may serve as evidence of valuable experimentation
and feedback that have made the implemented protocols more mature. and feedback that have made the implemented protocols more mature.
It is up to the individual working groups to use this information as It is up to the individual working groups to use this information as
they see it". they see it".
6.1. OpenWebRTC 6.1. OpenWebRTC
skipping to change at page 27, line 17 skipping to change at page 29, line 50
o Organization : Ericsson Research, Ericsson. o Organization : Ericsson Research, Ericsson.
o Name : SCReAM. o Name : SCReAM.
o Implementation link : A C++ implementation of SCReAM is available o Implementation link : A C++ implementation of SCReAM is available
at[SCReAM-CPP-implementation]. The code includes full support for at[SCReAM-CPP-implementation]. The code includes full support for
congestion control, rate control and multi stream handling, it can congestion control, rate control and multi stream handling, it can
be integrated in web clients given the addition of extra code to be integrated in web clients given the addition of extra code to
implement the RTCP feedback and RTP queue(s). The code also implement the RTCP feedback and RTP queue(s). The code also
includes a rudimentary implementation of a simulator that allows includes a rudimentary implementation of a simulator that allows
for some initial experiments. for some initial experiments. An additional experiment with
SCReAM in a remote control arrangement is also documented.
o Coverage : The code implements the specification in this memo. o Coverage : The code implements the specification in this memo.
o Contact : ingemar.s.johansson@ericsson.com o Contact : ingemar.s.johansson@ericsson.com
7. Suggested experiments 7. Suggested experiments
SCReAM has been evaluated in a number of different ways, most of the SCReAM has been evaluated in a number of different ways, most of the
evaluation has been in simulator. The OpenWebRTC implementation work evaluation has been in simulator. The OpenWebRTC implementation work
involved extensive testing with artificial bottlenecks with varying involved extensive testing with artificial bottlenecks with varying
bandwidths and using two different video coders (OpenH264 and VP9), bandwidths and using two different video coders (OpenH264 and VP9),
the experience of this lead to further improvements of the media rate the experience of this lead to further improvements of the media rate
control logic. control logic.
Further experiments are preferably done by means of implementation in Further experiments are preferably done by means of implementation in
real clients and web browsers. Recommended experiments are: real clients and web browsers. RECOMMENDED experiments are:
o Trials with various access technologies: EDGE/3G/4G, WiFi, DSL. o Trials with various access technologies: EDGE/3G/4G, WiFi, DSL.
Some experiments have already been carried out with LTE access,
see e.g. [SCReAM-CPP-implementation] and
[SCReAM-implementation-experience]
o Trials with different kinds of media: Audio, Video, slide show 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
skipping to change at page 28, line 23 skipping to change at page 31, line 9
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.
9. IANA Considerations 9. IANA Considerations
A new RFC4585 transport layer feedback message may to be standardized There is currently no request to IANA
if the use of the already existing RTCP extensions as described in
Appendix A.4 is not deemed sufficient.
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-08 to WG-09: Updated based shepherd review by Martin
Stiemerling, Q-bit semantics are removed as this is superfluous
for the moment. Pacing and RTCP considerations are moved up from
the appendix, FEC discussion moved to discussion section.
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
o WG-05 to WG-06: Added list of suggested experiments o WG-05 to WG-06: Added list of suggested experiments
o WG-04 to WG-05: Congestion control and rate control simplified o WG-04 to WG-05: Congestion control and rate control simplified
somewhat somewhat
o WG-03 to WG-04: Editorial fixes o WG-03 to WG-04: Editorial fixes
skipping to change at page 30, line 39 skipping to change at page 33, line 34
Zanaty, M., Singh, V., Nandakumar, S., and Z. Sarker, Zanaty, M., Singh, V., Nandakumar, S., and Z. Sarker,
"Congestion Control and Codec interactions in RTP "Congestion Control and Codec interactions in RTP
Applications", draft-ietf-rmcat-cc-codec-interactions-02 Applications", draft-ietf-rmcat-cc-codec-interactions-02
(work in progress), March 2016. (work in progress), March 2016.
[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-06
(work in progress), March 2017. (work in progress), March 2017.
[I-D.ietf-rmcat-scream-cc]
Johansson, I. and Z. Sarker, "Self-Clocked Rate Adaptation
for Multimedia", draft-ietf-rmcat-scream-cc-07 (work in
progress), November 2016.
[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-03 (work in progress), November 2016. wireless-tests-04 (work in progress), May 2017.
[Khademi-alternative-backoff-ECN] [I-D.ietf-tcpm-alternativebackoff-ecn]
"Alternative Backoff: Achieving Low Latency and High Khademi, N., Welzl, M., Armitage, G., and G. Fairhurst,
Throughput with ECN and AQM , CAIA Technical Report", "TCP Alternative Backoff with ECN (ABE)", draft-ietf-tcpm-
<https://tools.ietf.org/html/draft-khademi- alternativebackoff-ecn-01 (work in progress), May 2017.
alternativebackoff-ecn-00>.
[I-D.ietf-tcpm-rack]
Cheng, Y., Cardwell, N., and N. Dukkipati, "RACK: a time-
based fast loss detection algorithm for TCP", draft-ietf-
tcpm-rack-02 (work in progress), March 2017.
[LEDBAT-delay-impact] [LEDBAT-delay-impact]
"Assessing LEDBAT's Delay Impact, IEEE communications "Assessing LEDBAT's Delay Impact, IEEE communications
letters, vol. 17, no. 5, May 2013", May 2013, letters, vol. 17, no. 5, May 2013", May 2013,
<http://home.ifi.uio.no/michawe/research/publications/ <http://home.ifi.uio.no/michawe/research/publications/
ledbat-impact-letters.pdf>. ledbat-impact-letters.pdf>.
[OpenWebRTC] [OpenWebRTC]
"Open WebRTC project.", <http://www.openwebrtc.io/>. "Open WebRTC project.", <http://www.openwebrtc.io/>.
[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>.
[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., [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>. <http://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, <http://www.rfc-editor.org/info/rfc6679>.
skipping to change at page 33, line 9 skipping to change at page 35, line 48
The autocorrelation function is computed over a vector of values. The autocorrelation function is computed over a vector of values.
Let x be a vector constituting N values, the biased autocorrelation Let x be a vector constituting N values, the biased autocorrelation
function for a given lag=k for the vector x is given by . function for a given lag=k for the vector x is given by .
n=N-k n=N-k
R(x,k) = SUM x(n)*x(n+k) R(x,k) = SUM x(n)*x(n+k)
n=1 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. 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 :
pace_bitrate = max (RATE_PACE_MIN, cwnd* 8 / s_rtt)
t_pace = rtp_size * 8 / pace_bitrate
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
rate.
A.4. RTCP feedback considerations
This section describes the requirements on the RTCP feedback to make
SCReAM function well. First is described the requirements on the
feedback elements, second is described the requirements on the
feedback intensity to keep the 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 strictly necessary in order to
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
long to cover all received RTP packets. This list can be realized
with the Loss RLE report block in [RFC3611].
o A wall clock timestamp corresponding to the received RTP packet
with the highest sequence number is required in order to compute
the qdelay. This can be realized by means of the Packet Receipt
Times Report Block in [RFC3611]. begin_seq 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 [RFC3611] i.e. equal to the RTP timestamp
clock. Detailed individual packet receive times is not necessary
as SCReAM does currently not describe how this can be used.
The basic feedback needed for SCReAM involves the use of the Loss RLE
report block and the Packet Receipt Times block defined in Figure 2.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|reserved | PT=XR=207 | length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| BT=2 | rsvd. | T=0 | block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| begin_seq | end_seq |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| chunk 1 | chunk 2 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| chunk n-1 | chunk n |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| BT=3 | rsvd. | T=0 | block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| begin_seq | end_seq |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Receipt time of packet begin_seq |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2: Basic feedback message for SCReAM, based on RFC3611
In a typical use case, no more than four Loss RLE chunks 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, could reduce the feedback
message size a bit.
Additional feedback elements that can improve the performance of
SCReAM are:
o Accumulated number of ECN-CE marked packets (n_ECN). This can for
instance be realized with the ECN Feedback Report Format in
[RFC6679]. The given feedback report format is actually a slight
overkill as SCReAM would do quite well with only a counter that
increments by one for each received packet with the ECN-CE code
point set. The more bulky format 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 it becomes necessary to balance the
bitrates between the streams. This can currently not be realized
with any standardized feedback format, however the ECN counter can
be artificially incremented, even though no ECN-CE marked packets
are received to achieve a similar behavior.
A.4.2. Requirements on feedback intensity
SCReAM benefits from a relatively frequent feedback. 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.
The numbers above can be formulated as feedback interval function
that can be useful for the computation of the desired RTCP bandwidth.
The following equation expresses the feedback rate:
rate_fb = min(50,max(2.5,rate_media/10000))
rate_media is the RTP media bitrate expressed in [bits/s], rate_fb is
the feedback rate expressed in [packets/s]. Converted to feedback
interval we get:
fb_int = 1.0/min(50,max(2.5,rate_media/10000))
The transmission interval is not critical, this means that in the
case of multi-stream handling between two hosts, the feedback for two
or more SSRCs can be bundled to save UDP/IP overhead, the final
realized feedback interval should however not exceed 2*fb_int in such
cases meaning that a scheduled feedback transmission event should not
be delayed more that fb_int.
SCReAM works with AVPF regular mode, immediate or early mode is not
required by SCReAM but may nonetheless be useful for e.g RTCP
messages not directly related to SCReAM, such as those specified in
[RFC4585]. It is recommended to use reduced size RTCP [RFC5506]
where regular full compound RTCP transmission is controlled by trr-
int as described in [RFC4585].
A.5. Q-bit semantics (source quench)
The Q bit in the feedback is set by a receiver to signal that the
sender should reduce the bitrate. The sender will in response to
this reduce the congestion window with the 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 distribution signaling between the sending hosts. The
solution is then that the receiver sets the Q bit in the feedback to
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 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 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 Q bit set, the below steps outline how
the sender should react to RTCP feedback with the Q bit set. The
reduction is done once per RTT. Let :
o n = Number of received RTCP feedback messages in one RTT
o n_q = Number of received RTCP feedback messages in one RTT, with Q
bit set.
The new congestion window is then expressed as:
cwnd = max(MIN_CWND, cwnd*(1.0-0.5*n_q/n))
Note that CWND is adjusted at most once per RTT. Furthermore The
CWND increase should be inhibited for one RTT if CWND has been
decreased as a result of Q bits set in the feedback.
The required intensity of the Q-bit set in the feedback in order to
achieve a given rate distribution depends on many factors such as
RTT, video source material etc. The receiver thus need to monitor
the change in the received video bitrate on the different streams and
adjust the intensity of the Q-bit accordingly.
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
Zaheduzzaman Sarker Zaheduzzaman Sarker
 End of changes. 89 change blocks. 
351 lines changed or deleted 310 lines changed or added

This html diff was produced by rfcdiff 1.45. The latest version is available from http://tools.ietf.org/tools/rfcdiff/