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rfc:rfc8298

Internet Engineering Task Force (IETF) I. Johansson Request for Comments: 8298 Z. Sarker Category: Experimental Ericsson AB ISSN: 2070-1721 December 2017

            Self-Clocked Rate Adaptation for Multimedia

Abstract

 This memo describes a rate adaptation algorithm for conversational
 media services such as interactive video.  The solution conforms to
 the packet conservation principle and uses a hybrid loss-and-delay-
 based congestion control algorithm.  The algorithm is evaluated over
 both simulated Internet bottleneck scenarios as well as in a Long
 Term Evolution (LTE) system simulator and is shown to achieve both
 low latency and high video throughput in these scenarios.

Status of This Memo

 This document is not an Internet Standards Track specification; it is
 published for examination, experimental implementation, and
 evaluation.
 This document defines an Experimental Protocol for the Internet
 community.  This document is a product of the Internet Engineering
 Task Force (IETF).  It represents the consensus of the IETF
 community.  It has received public review and has been approved for
 publication by the Internet Engineering Steering Group (IESG).  Not
 all documents approved by the IESG are a candidate for any level of
 Internet Standard; see Section 2 of RFC 7841.
 Information about the current status of this document, any errata,
 and how to provide feedback on it may be obtained at
 https://www.rfc-editor.org/info/rfc8298.

Johansson & Sarker Experimental [Page 1] RFC 8298 SCReAM December 2017

Copyright Notice

 Copyright (c) 2017 IETF Trust and the persons identified as the
 document authors.  All rights reserved.
 This document is subject to BCP 78 and the IETF Trust's Legal
 Provisions Relating to IETF Documents
 (https://trustee.ietf.org/license-info) in effect on the date of
 publication of this document.  Please review these documents
 carefully, as they describe your rights and restrictions with respect
 to this document.  Code Components extracted from this document must
 include Simplified BSD License text as described in Section 4.e of
 the Trust Legal Provisions and are provided without warranty as
 described in the Simplified BSD License.

Johansson & Sarker Experimental [Page 2] RFC 8298 SCReAM December 2017

Table of Contents

 1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   4
   1.1.  Wireless (LTE) Access Properties  . . . . . . . . . . . .   4
   1.2.  Why is it a self-clocked algorithm? . . . . . . . . . . .   5
 2.  Requirements Language . . . . . . . . . . . . . . . . . . . .   5
 3.  Overview of SCReAM Algorithm  . . . . . . . . . . . . . . . .   6
   3.1.  Network Congestion Control  . . . . . . . . . . . . . . .   8
   3.2.  Sender Transmission Control . . . . . . . . . . . . . . .   9
   3.3.  Media Rate Control  . . . . . . . . . . . . . . . . . . .   9
 4.  Detailed Description of SCReAM  . . . . . . . . . . . . . . .  10
   4.1.  SCReAM Sender . . . . . . . . . . . . . . . . . . . . . .  10
     4.1.1.  Constants and Parameter Values  . . . . . . . . . . .  10
       4.1.1.1.  Constants . . . . . . . . . . . . . . . . . . . .  11
       4.1.1.2.  State Variables . . . . . . . . . . . . . . . . .  12
     4.1.2.  Network Congestion Control  . . . . . . . . . . . . .  14
       4.1.2.1.  Reaction to Packet Loss and ECN . . . . . . . . .  17
       4.1.2.2.  Congestion Window Update  . . . . . . . . . . . .  17
       4.1.2.3.  Competing Flows Compensation  . . . . . . . . . .  20
       4.1.2.4.  Lost Packet Detection . . . . . . . . . . . . . .  22
       4.1.2.5.  Send Window Calculation . . . . . . . . . . . . .  23
       4.1.2.6.  Packet Pacing . . . . . . . . . . . . . . . . . .  24
       4.1.2.7.  Resuming Fast Increase Mode . . . . . . . . . . .  24
       4.1.2.8.  Stream Prioritization . . . . . . . . . . . . . .  24
     4.1.3.  Media Rate Control  . . . . . . . . . . . . . . . . .  25
   4.2.  SCReAM Receiver . . . . . . . . . . . . . . . . . . . . .  28
     4.2.1.  Requirements on Feedback Elements . . . . . . . . . .  28
     4.2.2.  Requirements on Feedback Intensity  . . . . . . . . .  30
 5.  Discussion  . . . . . . . . . . . . . . . . . . . . . . . . .  31
 6.  Suggested Experiments . . . . . . . . . . . . . . . . . . . .  31
 7.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  32
 8.  Security Considerations . . . . . . . . . . . . . . . . . . .  32
 9.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  33
   9.1.  Normative References  . . . . . . . . . . . . . . . . . .  33
   9.2.  Informative References  . . . . . . . . . . . . . . . . .  34
 Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . .  36
 Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  36

Johansson & Sarker Experimental [Page 3] RFC 8298 SCReAM December 2017

1. Introduction

 Congestion in the Internet occurs when the transmitted bitrate is
 higher than the available capacity over a given transmission path.
 Applications that are deployed in the Internet have to employ
 congestion control to achieve robust performance and to avoid
 congestion collapse in the Internet.  Interactive real-time
 communication imposes a lot of requirements on the transport;
 therefore, a robust, efficient rate adaptation for all access types
 is an important part of interactive real-time communications, as the
 transmission channel bandwidth can vary over time.  Wireless access
 such as LTE, which is an integral part of the current Internet,
 increases the importance of rate adaptation as the channel bandwidth
 of a default LTE bearer [QoS-3GPP] can change considerably in a very
 short time frame.  Thus, a rate adaptation solution for interactive
 real-time media, such as WebRTC [RFC7478], should be both quick and
 be able to operate over a large range in channel capacity.  This memo
 describes Self-Clocked Rate Adaptation for Multimedia (SCReAM), a
 solution that implements congestion control for RTP streams
 [RFC3550].  While SCReAM was originally devised for WebRTC, it can
 also be used for other applications where congestion control of RTP
 streams is necessary.  SCReAM is based on the self-clocking principle
 of TCP and uses techniques similar to what is used in the rate
 adaptation algorithm based on Low Extra Delay Background Transport
 (LEDBAT) [RFC6817].  SCReAM is not entirely self-clocked as it
 augments self-clocking with pacing and a minimum send rate.  SCReAM
 can take advantage of Explicit Congestion Notification (ECN) in cases
 where ECN is supported by the network and the hosts.  However, ECN is
 not required for the basic congestion control functionality in
 SCReAM.

1.1. Wireless (LTE) Access Properties

 [WIRELESS-TESTS] describes the complications that can be observed in
 wireless environments.  Wireless access such as LTE typically cannot
 guarantee a given bandwidth; this is true especially for default
 bearers.  The network throughput can vary considerably, for instance,
 in cases where the wireless terminal is moving around.  Even though
 LTE can support bitrates well above 100 Mbps, there are cases when
 the available bitrate can be much lower; examples are situations with
 high network load and poor coverage.  An additional complication is
 that the network throughput can drop for short time intervals (e.g.,
 at handover); these short glitches are initially very difficult to
 distinguish from more permanent reductions in throughput.
 Unlike wireline bottlenecks with large statistical multiplexing, it
 is not possible to try to maintain a given bitrate when congestion is
 detected with the hope that other flows will yield.  This is because

Johansson & Sarker Experimental [Page 4] RFC 8298 SCReAM December 2017

 there are generally few other flows competing for the same
 bottleneck.  Each user gets its own variable throughput bottleneck,
 where the throughput depends on factors like channel quality, network
 load, and historical throughput.  The bottom line is, if the
 throughput drops, the sender has no other option than to reduce the
 bitrate.  Once the radio scheduler has reduced the resource
 allocation for a bearer, a flow (which is using RTP Media Congestion
 Avoidance Techniques (RMCAT)) in that bearer aims to reduce the
 sending rate quite quickly (within one RTT) in order to avoid
 excessive queuing delay or packet loss.

1.2. Why is it a self-clocked algorithm?

 Self-clocked congestion control algorithms provide a benefit over
 their rate-based counterparts in that the former consists of two
 adaptation mechanisms:
 o  A congestion window computation that evolves over a longer
    timescale (several RTTs) especially when the congestion window
    evolution is dictated by estimated delay (to minimize
    vulnerability to, e.g., short-term delay variations).
 o  A fine-grained congestion control given by the self-clocking; it
    operates on a shorter time scale (1 RTT).  The benefits of self-
    clocking are also elaborated upon in [TFWC].
 A rate-based congestion control algorithm typically adjusts the rate
 based on delay and loss.  The congestion detection needs to be done
 with a certain time lag to avoid overreaction to spurious congestion
 events such as delay spikes.  Despite the fact that there are two or
 more congestion indications, the outcome is that there is still only
 one mechanism to adjust the sending rate.  This makes it difficult to
 reach the goals of high throughput and prompt reaction to congestion.

2. Requirements Language

 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
 "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
 "OPTIONAL" in this document are to be interpreted as described in
 BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all
 capitals, as shown here.

Johansson & Sarker Experimental [Page 5] RFC 8298 SCReAM December 2017

3. Overview of SCReAM Algorithm

 The core SCReAM algorithm has similarities to the concepts of self-
 clocking used in TCP-friendly window-based congestion control [TFWC]
 and follows the packet conservation principle.  The packet
 conservation principle is described as a key factor behind the
 protection of networks from congestion [Packet-conservation].
 In SCReAM, the receiver of the media echoes a list of received RTP
 packets and the timestamp of the RTP packet with the highest sequence
 number back to the sender in feedback packets.  The sender keeps a
 list of transmitted packets, their respective sizes, and the time
 they were transmitted.  This information is used to determine the
 number of bytes that can be transmitted at any given time instant.  A
 congestion window puts an upper limit on how many bytes can be in
 flight, i.e., transmitted but not yet acknowledged.
 The congestion window is determined in a way similar to LEDBAT
 [RFC6817].  LEDBAT is a congestion control algorithm that uses send
 and receive timestamps to estimate the queuing delay (from now on
 denoted "qdelay") along the transmission path.  This information is
 used to adjust the congestion window.  The use of LEDBAT ensures that
 the end-to-end latency is kept low.  [LEDBAT-delay-impact] shows that
 LEDBAT has certain inherent issues that make it counteract its
 purpose of achieving low delay.  The general problem described in the
 paper is that the base delay is offset by LEDBAT's own queue buildup.
 The big difference with using LEDBAT in the SCReAM context lies in
 the facts that the source is rate limited and that the RTP queue must
 be kept short (preferably empty).  In addition, the output from a
 video encoder is rarely constant bitrate; static content (talking
 heads, for instance) gives almost zero video bitrate.  This yields
 two useful properties when LEDBAT is used with SCReAM; they help to
 avoid the issues described in [LEDBAT-delay-impact]:
 1.  There is always a certain probability that SCReAM is short of
     data to transmit; this means that the network queue will become
     empty every once in a while.
 2.  The max video bitrate can be lower than the link capacity.  If
     the max video bitrate is 5 Mbps and the capacity is 10 Mbps, then
     the network queue will become empty.
 It is sufficient that any of the two conditions above is fulfilled to
 make the base delay update properly.  Furthermore,
 [LEDBAT-delay-impact] describes an issue with short-lived competing
 flows.  In SCReAM, these short-lived flows will cause the self-
 clocking to slow down, thereby building up the RTP queue; in turn,
 this results in a reduced media video bitrate.  Thus, SCReAM slows

Johansson & Sarker Experimental [Page 6] RFC 8298 SCReAM December 2017

 the bitrate more when there are competing short-lived flows than the
 traditional use of LEDBAT does.  The basic functionality in the use
 of LEDBAT in SCReAM is quite simple; however, there are a few steps
 in order to make the concept work with conversational media:
 o  Congestion window validation techniques.  These are similar to the
    method described in [RFC7661].  Congestion window validation
    ensures that the congestion window is limited by the actual number
    bytes in flight; this is important especially in the context of
    rate-limited sources such as video.  Lack of congestion window
    validation would lead to a slow reaction to congestion as the
    congestion window does not properly reflect the congestion state
    in the network.  The allowed idle period in this memo is shorter
    than in [RFC7661]; this to avoid excessive delays in the cases
    where, e.g., wireless throughput has decreased during a period
    where the output bitrate from the media coder has been low (for
    instance, due to inactivity).  Furthermore, this memo allows for
    more relaxed rules for when the congestion window is allowed to
    grow; this is necessary as the variable output bitrate generally
    means that the congestion window is often underutilized.
 o  Fast increase mode makes the bitrate increase faster when no
    congestion is detected.  It makes the media bitrate ramp up within
    5 to 10 seconds.  The behavior is similar to TCP slowstart.  Fast
    increase mode is exited when congestion is detected.  However,
    fast increase mode can resume if the congestion level is low; this
    enables a reasonably quick rate increase in case link throughput
    increases.
 o  A qdelay trend is computed for earlier detection of incipient
    congestion; as a result, it reduces jitter.
 o  Addition of a media rate control function.
 o  Use of inflection points in the media rate calculation to achieve
    reduced jitter.
 o  Adjustment of qdelay target for better performance when competing
    with other loss-based congestion-controlled flows.
 The above-mentioned features will be described in more detail in
 Sections 3.1 to 3.3.  The full details are described in Section 4.

Johansson & Sarker Experimental [Page 7] RFC 8298 SCReAM December 2017

                  +---------------------------+
                  |        Media encoder      |
                  +---------------------------+
                      ^                  |
                      |                  |(1)
                      |(3)              RTP
                      |                  V
                      |            +-----------+
                 +---------+       |           |
                 | Media   |  (2)  |   Queue   |
                 | rate    |<------|           |
                 | control |       |RTP packets|
                 +---------+       |           |
                                   +-----------+
                                         |
                                         |(4)
                                        RTP
                                         |
                                         v
            +------------+       +--------------+
            |  Network   |  (7)  |    Sender    |
        +-->| congestion |------>| Transmission |
        |   |  control   |       |   Control    |
        |   +------------+       +--------------+
        |                                |
        |-------------RTCP----------|    |(5)
            (6)                     |   RTP
                                    |    v
                                +------------+
                                |     UDP    |
                                |   socket   |
                                +------------+
                Figure 1: SCReAM Sender Functional View
 The SCReAM algorithm consists of three main parts: network congestion
 control, sender transmission control, and media rate control.  All of
 these parts reside at the sender side.  Figure 1 shows the functional
 overview of a SCReAM sender.  The receiver-side algorithm is very
 simple in comparison, as it only generates feedback containing
 acknowledgements of received RTP packets and an ECN count.

3.1. Network Congestion Control

 The network congestion control sets an upper limit on how much data
 can be in the network (bytes in flight); this limit is called CWND
 (congestion window) and is used in the sender transmission control.

Johansson & Sarker Experimental [Page 8] RFC 8298 SCReAM December 2017

 The SCReAM congestion control method uses techniques similar to
 LEDBAT [RFC6817] to measure the qdelay.  As is the case with LEDBAT,
 it is not necessary to use synchronized clocks in the sender and
 receiver in order to compute the qdelay.  However, it is necessary
 that they use the same clock frequency, or that the clock frequency
 at the receiver can be inferred reliably by the sender.  Failure to
 meet this requirement leads to malfunction in the SCReAM congestion
 control algorithm due to incorrect estimation of the network queue
 delay.
 The SCReAM sender calculates the congestion window based on the
 feedback from the SCReAM receiver.  The congestion window is allowed
 to increase if the qdelay is below a predefined qdelay target;
 otherwise, the congestion window decreases.  The qdelay target is
 typically set to 50-100 ms.  This ensures that the queuing delay is
 kept low.  The reaction to loss or ECN events leads to an instant
 reduction of CWND.  Note that the source rate-limited nature of real-
 time media, such as video, typically means that the queuing delay
 will mostly be below the given delay target.  This is contrary to the
 case where large files are transmitted using LEDBAT congestion
 control and the queuing delay will stay close to the delay target.

3.2. Sender Transmission Control

 The sender transmission control limits the output of data, given by
 the relation between the number of bytes in flight and the congestion
 window.  Packet pacing is used to mitigate issues with ACK
 compression that MAY cause increased jitter and/or packet loss in the
 media traffic.  Packet pacing limits the packet transmission rate
 given by the estimated link throughput.  Even if the send window
 allows for the transmission of a number of packets, these packets are
 not transmitted immediately; rather, they are transmitted in
 intervals given by the packet size and the estimated link throughput.

3.3. Media Rate Control

 The media rate control serves to adjust the media bitrate to ramp up
 quickly enough to get a fair share of the system resources when link
 throughput increases.
 The reaction to reduced throughput MUST be prompt in order to avoid
 getting too much data queued in the RTP packet queue(s) in the
 sender.  The media bitrate is decreased if the RTP queue size exceeds
 a threshold.
 In cases where the sender's frame queues increase rapidly, such as in
 the case of a Radio Access Type (RAT) handover, the SCReAM sender MAY
 implement additional actions, such as discarding of encoded media

Johansson & Sarker Experimental [Page 9] RFC 8298 SCReAM December 2017

 frames or frame skipping in order to ensure that the RTP queues are
 drained quickly.  Frame skipping results in the frame rate being
 temporarily reduced.  Which method to use is a design choice and is
 outside the scope of this algorithm description.

4. Detailed Description of SCReAM

4.1. SCReAM Sender

 This section describes the sender-side algorithm in more detail.  It
 is split between the network congestion control, sender transmission
 control, and media rate control.
 A SCReAM sender implements media rate control and an RTP queue for
 each media type or source, where RTP packets containing encoded media
 frames are temporarily stored for transmission.  Figure 1 shows the
 details when a single media source (or stream) is used.  A
 transmission scheduler (not shown in the figure) is added to support
 multiple streams.  The transmission scheduler can enforce differing
 priorities between the streams and act like a coupled congestion
 controller for multiple flows.  Support for multiple streams is
 implemented in [SCReAM-CPP-implementation].
 Media frames are encoded and forwarded to the RTP queue (1) in
 Figure 1.  The media rate adaptation adapts to the size of the RTP
 queue (2) and provides a target rate for the media encoder (3).  The
 RTP packets are picked from the RTP queue (4), for multiple flows
 from each RTP queue based on some defined priority order or simply in
 a round-robin fashion, by the sender transmission controller.  The
 sender transmission controller (in case of multiple flows a
 transmission scheduler) sends the RTP packets to the UDP socket (5).
 In the general case, all media SHOULD go through the sender
 transmission controller and is limited so that the number of bytes in
 flight is less than the congestion window.  RTCP packets are received
 (6) and the information about the bytes in flight and congestion
 window is exchanged between the network congestion control and the
 sender transmission control (7).

4.1.1. Constants and Parameter Values

 Constants and state variables are listed in this section.  Temporary
 variables are not listed; instead, they are appended with '_t' in the
 pseudocode to indicate their local scope.

Johansson & Sarker Experimental [Page 10] RFC 8298 SCReAM December 2017

4.1.1.1. Constants

 The RECOMMENDED values, within parentheses "()", for the constants
 are deduced from experiments.
 QDELAY_TARGET_LO (0.1 s)
   Target value for the minimum qdelay.
 QDELAY_TARGET_HI (0.4 s)
   Target value for the maximum qdelay.  This parameter provides an
   upper limit to how much the target qdelay (qdelay_target) can be
   increased in order to cope with competing loss-based flows.
   However, the target qdelay does not have to be initialized to this
   high value, as it would increase end-to-end delay and also make the
   rate control and congestion control loops sluggish.
 QDELAY_WEIGHT (0.1)
   Averaging factor for qdelay_fraction_avg.
 QDELAY_TREND_TH (0.2)
   Threshold for the detection of incipient congestion.
 MIN_CWND (3000 bytes)
   Minimum congestion window.
 MAX_BYTES_IN_FLIGHT_HEAD_ROOM (1.1)
   Headroom for the limitation of CWND.
 GAIN (1.0)
   Gain factor for congestion window adjustment.
 BETA_LOSS (0.8)
   CWND scale factor due to loss event.
 BETA_ECN (0.9)
   CWND scale factor due to ECN event.
 BETA_R (0.9)
   Scale factor for target rate due to loss event.
 MSS (1000 byte)
   Maximum segment size = Max RTP packet size.
 RATE_ADJUST_INTERVAL (0.2 s)
   Interval between media bitrate adjustments.
 TARGET_BITRATE_MIN
   Minimum target bitrate in bps (bits per second).

Johansson & Sarker Experimental [Page 11] RFC 8298 SCReAM December 2017

 TARGET_BITRATE_MAX
   Maximum target bitrate in bps.
 RAMP_UP_SPEED (200000 bps/s)
   Maximum allowed rate increase speed.
 PRE_CONGESTION_GUARD  (0.0..1.0)
   Guard factor against early congestion onset.  A higher value gives
   less jitter, possibly at the expense of a lower link utilization.
   This value MAY be subject to tuning depending on e.g., media coder
   characteristics.  Experiments with H264 and VP8 indicate that 0.1
   is a suitable value.  See [SCReAM-CPP-implementation] and
   [SCReAM-implementation-experience] for evaluation of a real
   implementation.
 TX_QUEUE_SIZE_FACTOR (0.0..2.0)
   Guard factor against RTP queue buildup.  This value MAY be subject
   to tuning depending on, e.g., media coder characteristics.
   Experiments with H264 and VP8 indicate that 1.0 is a suitable
   value.  See [SCReAM-CPP-implementation] and
   [SCReAM-implementation-experience] for evaluation of a real
   implementation.
 RTP_QDELAY_TH (0.02 s)  RTP queue delay threshold for a target rate
   reduction.
 TARGET_RATE_SCALE_RTP_QDELAY (0.95)  Scale factor for target rate
   when RTP qdelay threshold exceeds RTP_QDELAY_TH.
 QDELAY_TREND_LO (0.2)  Threshold value for qdelay_trend.
 T_RESUME_FAST_INCREASE (5 s)  Time span until fast increase mode can
   be resumed, given that the qdelay_trend is below QDELAY_TREND_LO.
 RATE_PACE_MIN (50000 bps)  Minimum pacing rate.

4.1.1.2. State Variables

 The values within parentheses "()" indicate initial values.
 qdelay_target (QDELAY_TARGET_LO)
   qdelay target, a variable qdelay target is introduced to manage
   cases where a fixed qdelay target would otherwise starve the RMCAT
   flow under such circumstances (e.g., FTP competes for the bandwidth
   over the same bottleneck).  The qdelay target is allowed to vary
   between QDELAY_TARGET_LO and QDELAY_TARGET_HI.

Johansson & Sarker Experimental [Page 12] RFC 8298 SCReAM December 2017

 qdelay_fraction_avg (0.0)
   Fractional qdelay filtered by the Exponentially Weighted Moving
   Average (EWMA).
 qdelay_fraction_hist[20] ({0,..,0})
   Vector of the last 20 fractional qdelay samples.
 qdelay_trend (0.0)
   qdelay trend; indicates incipient congestion.
 qdelay_trend_mem (0.0)
   Low-pass filtered version of qdelay_trend.
 qdelay_norm_hist[100] ({0,..,0})
   Vector of the last 100 normalized qdelay samples.
 in_fast_increase (true)
   True if in fast increase mode.
 cwnd (MIN_CWND)
   Congestion window.
 bytes_newly_acked (0)
   The number of bytes that was acknowledged with the last received
   acknowledgement, i.e., bytes acknowledged since the last CWND
   update.
 max_bytes_in_flight (0)
   The maximum number of bytes in flight over a sliding time window,
   i.e., transmitted but not yet acknowledged bytes.
 send_wnd (0)
   Upper limit to how many bytes can currently be transmitted.
   Updated when cwnd is updated and when RTP packet is transmitted.
 target_bitrate (0 bps)
   Media target bitrate.
 target_bitrate_last_max (1 bps)
   Inflection point of the media target bitrate, i.e., the last known
   highest target_bitrate.  Used to limit bitrate increase speed close
   to the last known congestion point.
 rate_transmit (0.0 bps)
   Measured transmit bitrate.
 rate_ack (0.0 bps)
   Measured throughput based on received acknowledgements.

Johansson & Sarker Experimental [Page 13] RFC 8298 SCReAM December 2017

 rate_media (0.0 bps)
   Measured bitrate from the media encoder.
 rate_media_median (0.0 bps)
   Median value of rate_media, computed over more than 10 s.
 s_rtt (0.0s)
   Smoothed RTT (in seconds), computed with a similar method to that
   described in [RFC6298].
 rtp_queue_size (0 bits)
   Sum of the sizes of RTP packets in queue.
 rtp_size (0 byte)
   Size of the last transmitted RTP packet.
 loss_event_rate (0.0)
   The estimated fraction of RTTs with lost packets detected.

4.1.2. Network Congestion Control

 This section explains the network congestion control, which performs
 two main functions:
 o  Computation of congestion window at the sender: This gives an
    upper limit to the number of bytes in flight.
 o  Calculation of send window at the sender: RTP packets are
    transmitted if allowed by the relation between the number of bytes
    in flight and the congestion window.  This is controlled by the
    send window.
 SCReAM is a window-based and byte-oriented congestion control
 protocol, where the number of bytes transmitted is inferred from the
 size of the transmitted RTP packets.  Thus, a list of transmitted RTP
 packets and their respective transmission times (wall-clock time)
 MUST be kept for further calculation.
 The number of bytes in flight (bytes_in_flight) is computed as the
 sum of the sizes of the RTP packets ranging from the RTP packet most
 recently transmitted, down to but not including the acknowledged
 packet with the highest sequence number.  This can be translated to
 the difference between the highest transmitted byte sequence number
 and the highest acknowledged byte sequence number.  As an example: If
 an RTP packet with sequence number SN is transmitted and the last
 acknowledgement indicates SN-5 as the highest received sequence
 number, then bytes_in_flight is computed as the sum of the size of
 RTP packets with sequence number SN-4, SN-3, SN-2, SN-1, and SN.  It

Johansson & Sarker Experimental [Page 14] RFC 8298 SCReAM December 2017

 does not matter if, for instance, the packet with sequence number
 SN-3 was lost -- the size of RTP packet with sequence number SN-3
 will still be considered in the computation of bytes_in_flight.
 Furthermore, a variable bytes_newly_acked is incremented with a value
 corresponding to how much the highest sequence number has increased
 since the last feedback.  As an example: If the previous
 acknowledgement indicated the highest sequence number N and the new
 acknowledgement indicated N+3, then bytes_newly_acked is incremented
 by a value equal to the sum of the sizes of RTP packets with sequence
 number N+1, N+2, and N+3.  Packets that are lost are also included,
 which means that even though, e.g., packet N+2 was lost, its size is
 still included in the update of bytes_newly_acked.  The
 bytes_newly_acked variable is reset to zero after a CWND update.
 The feedback from the receiver is assumed to consist of the following
 elements.
 o  A list of received RTP packets' sequence numbers.
 o  The wall-clock timestamp corresponding to the received RTP packet
    with the highest sequence number.
 o  The accumulated number of ECN-CE-marked packets (n_ECN).  Here,
    "CE" refers to "Congestion Experienced".
 When the sender receives RTCP feedback, the qdelay is calculated as
 outlined in [RFC6817].  A qdelay sample is obtained for each received
 acknowledgement.  No smoothing of the qdelay is performed; however,
 some smoothing occurs anyway because the CWND computation is a low-
 pass filter function.  A number of variables are updated as
 illustrated by the pseudocode below; temporary variables are appended
 with '_t'.  As mentioned in Section 6, calculation of the proper
 congestion window and media bitrate may benefit from additional
 optimizations to handle very high and very low bitrates, and from
 additional damping to handle periodic packet bursts.  Some such
 optimizations are implemented in [SCReAM-CPP-implementation], but
 they do not form part of the specification of SCReAM at this time.

Johansson & Sarker Experimental [Page 15] RFC 8298 SCReAM December 2017

   <CODE BEGINS>
   update_variables(qdelay):
     qdelay_fraction_t = qdelay / qdelay_target
     # Calculate moving average
     qdelay_fraction_avg = (1 - QDELAY_WEIGHT) * qdelay_fraction_avg +
        QDELAY_WEIGHT * qdelay_fraction_t
     update_qdelay_fraction_hist(qdelay_fraction_t)
     # Compute the average of the values in qdelay_fraction_hist
     avg_t = average(qdelay_fraction_hist)
     # R is an autocorrelation function of qdelay_fraction_hist,
     #  with the mean (DC component) removed, at lag K
     # The subtraction of the scalar avg_t from
     #  qdelay_fraction_hist is performed element-wise
     a_t = R(qdelay_fraction_hist-avg_t, 1) /
           R(qdelay_fraction_hist-avg_t, 0)
     # Calculate qdelay trend
     qdelay_trend = min(1.0, max(0.0, a_t * qdelay_fraction_avg))
     # Calculate a 'peak-hold' qdelay_trend; this gives a memory
     #  of congestion in the past
     qdelay_trend_mem = max(0.99 * qdelay_trend_mem, qdelay_trend)
    <CODE ENDS>
 The qdelay fraction is sampled every 50 ms, and the last 20 samples
 are stored in a vector (qdelay_fraction_hist).  This vector is used
 in the computation of a qdelay trend that gives a value between 0.0
 and 1.0 depending on the estimated congestion level.  The prediction
 coefficient 'a_t' has positive values if qdelay shows an increasing
 or decreasing trend; thus, an indication of congestion is obtained
 before the qdelay target is reached.  As a side effect, if qdelay
 decreases, it's taken as a sign of congestion; however, experiments
 have shown that this is beneficial, as increasing or decreasing queue
 delay is an indication that the transmit rate is very close to the
 path capacity.
 The autocorrelation function 'R' is defined as follows.  Let x be a
 vector constituting N values, the biased autocorrelation function for
 a given lag=k for the vector x is given by.
               n=N-k
       R(x,k) = SUM x(n) * x(n + k)
               n=1
 The prediction coefficient is further multiplied with
 qdelay_fraction_avg to reduce sensitivity to increasing qdelay when
 it is very small.  The 50 ms sampling is a simplification that could
 have the effect that the same qdelay is sampled several times;
 however, this does not pose any problem, as the vector is only used
 to determine if the qdelay is increasing or decreasing.  The

Johansson & Sarker Experimental [Page 16] RFC 8298 SCReAM December 2017

 qdelay_trend is utilized in the media rate control to indicate
 incipient congestion and to determine when to exit from fast increase
 mode. qdelay_trend_mem is used to enforce a less aggressive rate
 increase after congestion events.  The function
 update_qdelay_fraction_hist(..) removes the oldest element and adds
 the latest qdelay_fraction element to the qdelay_fraction_hist
 vector.

4.1.2.1. Reaction to Packet Loss and ECN

 A loss event is indicated if one or more RTP packets are declared
 missing.  The loss detection is described in Section 4.1.2.4.  Once a
 loss event is detected, further detected lost RTP packets SHOULD be
 ignored for a full smoothed round-trip time; the intention is to
 limit the congestion window decrease to at most once per round trip.
 The congestion window back-off due to loss events is deliberately a
 bit less than is the case with TCP Reno, for example.  TCP is
 generally used to transmit whole files; the file is then like a
 source with an infinite bitrate until the whole file has been
 transmitted.  SCReAM, on the other hand, has a source whose rate is
 limited to a value close to the available transmit rate and often
 below that value; the effect is that SCReAM has less opportunity to
 grab free capacity than a TCP-based file transfer.  To compensate for
 this, it is RECOMMENDED to let SCReAM reduce the congestion window
 less than what is the case with TCP when loss events occur.
 An ECN event is detected if the n_ECN counter in the feedback report
 has increased since the previous received feedback.  Once an ECN
 event is detected, the n_ECN counter is ignored for a full smoothed
 round-trip time; the intention is to limit the congestion window
 decrease to at most once per round trip.  The congestion window back-
 off due to an ECN event MAY be smaller than if a loss event occurs.
 This is in line with the idea outlined in [ALT-BACKOFF] to enable ECN
 marking thresholds lower than the corresponding packet drop
 thresholds.

4.1.2.2. Congestion Window Update

 The update of the congestion window depends on if loss, ECN-marking,
 or neither of the two occurs.  The pseudocode below describes the
 actions for each case.

Johansson & Sarker Experimental [Page 17] RFC 8298 SCReAM December 2017

   <CODE BEGINS>
   on congestion event(qdelay):
     # Either loss or ECN mark is detected
     in_fast_increase = false
     if (is loss)
       # Loss is detected
       cwnd = max(MIN_CWND, cwnd * BETA_LOSS)
     else
       # No loss, so it is then an ECN mark
       cwnd = max(MIN_CWND, cwnd * BETA_ECN)
     end
     adjust_qdelay_target(qdelay) #compensating for competing flows
     calculate_send_window(qdelay, qdelay_target)
   # When no congestion event
   on acknowledgement(qdelay):
     update_bytes_newly_acked()
     update_cwnd(bytes_newly_acked)
     adjust_qdelay_target(qdelay) # compensating for competing flows
     calculate_send_window(qdelay, qdelay_target)
     check_to_resume_fast_increase()
   <CODE ENDS>
 The methods are described in detail below.
 The congestion window update is based on qdelay, except for the
 occurrence of loss events (one or more lost RTP packets in one RTT)
 or ECN events, which were described earlier.
 Pseudocode for the update of the congestion window is found below.

Johansson & Sarker Experimental [Page 18] RFC 8298 SCReAM December 2017

 <CODE BEGINS>
 update_cwnd(bytes_newly_acked):
   # In fast increase mode?
   if (in_fast_increase)
     if (qdelay_trend >= QDELAY_TREND_TH)
       # Incipient congestion detected; exit fast increase mode
       in_fast_increase = false
     else
       # No congestion yet; increase cwnd if it
       #  is sufficiently used
       # Additional slack of bytes_newly_acked is
       #  added to ensure that CWND growth occurs
       #  even when feedback is sparse
       if (bytes_in_flight * 1.5 + bytes_newly_acked > cwnd)
         cwnd = cwnd + bytes_newly_acked
       end
       return
     end
   end
   # Not in fast increase mode
   # off_target calculated as with LEDBAT
   off_target_t = (qdelay_target - qdelay) / qdelay_target
   gain_t = GAIN
   # Adjust congestion window
   cwnd_delta_t =
     gain_t * off_target_t * bytes_newly_acked * MSS / cwnd
   if (off_target_t > 0 &&
       bytes_in_flight * 1.25 + bytes_newly_acked <= cwnd)
     # No cwnd increase if window is underutilized
     # Additional slack of bytes_newly_acked is
     #  added to ensure that CWND growth occurs
     #  even when feedback is sparse
     cwnd_delta_t = 0;
   end
   # Apply delta
   cwnd += cwnd_delta_t
   # limit cwnd to the maximum number of bytes in flight
   cwnd = min(cwnd, max_bytes_in_flight *
              MAX_BYTES_IN_FLIGHT_HEAD_ROOM)
   cwnd = max(cwnd, MIN_CWND)
 <CODE ENDS>

Johansson & Sarker Experimental [Page 19] RFC 8298 SCReAM December 2017

 CWND is updated differently depending on whether or not the
 congestion control is in fast increase mode, as controlled by the
 variable in_fast_increase.
 When in fast increase mode, the congestion window is increased with
 the number of newly acknowledged bytes as long as the window is
 sufficiently used.  Sparse feedback can potentially limit congestion
 window growth; therefore, additional slack is added, given by the
 number of newly acknowledged bytes.
 The congestion window growth when in_fast_increase is false is
 dictated by the relation between qdelay and qdelay_target; congestion
 window growth is limited if the window is not used sufficiently.
 SCReAM calculates the GAIN in a similar way to what is specified in
 [RFC6817].  However, [RFC6817] specifies that the CWND increase is
 limited by an additional function controlled by a constant
 ALLOWED_INCREASE.  This additional limitation is removed in this
 specification.
 Further, the CWND is limited by max_bytes_in_flight and MIN_CWND.
 The limitation of the congestion window by the maximum number of
 bytes in flight over the last 5 seconds (max_bytes_in_flight) avoids
 possible overestimation of the throughput after, for example, idle
 periods.  An additional MAX_BYTES_IN_FLIGHT_HEAD_ROOM provides slack
 to allow for a certain amount of variability in the media coder
 output rate.

4.1.2.3. Competing Flows Compensation

 It is likely that a flow using the SCReAM algorithm will have to
 share congested bottlenecks with other flows that use a more
 aggressive congestion control algorithm (for example, large FTP flows
 using loss-based congestion control).  The worst condition occurs
 when the bottleneck queues are of tail-drop type with a large buffer
 size.  SCReAM takes care of such situations by adjusting the
 qdelay_target when loss-based flows are detected, as shown in the
 pseudocode below.

Johansson & Sarker Experimental [Page 20] RFC 8298 SCReAM December 2017

   <CODE BEGINS>
   adjust_qdelay_target(qdelay)
     qdelay_norm_t = qdelay / QDELAY_TARGET_LOW
     update_qdelay_norm_history(qdelay_norm_t)
     # Compute variance
     qdelay_norm_var_t = VARIANCE(qdelay_norm_history(200))
     # Compensation for competing traffic
     # Compute average
     qdelay_norm_avg_t = AVERAGE(qdelay_norm_history(50))
     # Compute upper limit to target delay
     new_target_t = qdelay_norm_avg_t + sqrt(qdelay_norm_var_t)
     new_target_t *= QDELAY_TARGET_LO
     if (loss_event_rate > 0.002)
       # Packet losses detected
       qdelay_target = 1.5 * new_target_t
     else
       if (qdelay_norm_var_t < 0.2)
         # Reasonably safe to set target qdelay
         qdelay_target = new_target_t
       else
         # Check if target delay can be reduced; this helps prevent
         #  the target delay from being locked to high values forever
         if (new_target_t < QDELAY_TARGET_LO)
           # Decrease target delay quickly, as measured queuing
           #  delay is lower than target
           qdelay_target = max(qdelay_target * 0.5, new_target_t)
         else
           # Decrease target delay slowly
           qdelay_target *= 0.9
         end
       end
     end
     # Apply limits
     qdelay_target = min(QDELAY_TARGET_HI, qdelay_target)
     qdelay_target = max(QDELAY_TARGET_LO, qdelay_target)
   <CODE ENDS>
 Two temporary variables are calculated. qdelay_norm_avg_t is the
 long-term average queue delay, qdelay_norm_var_t is the long-term
 variance of the queue delay.  A high qdelay_norm_var_t indicates that
 the queue delay changes; this can be an indication that bottleneck
 bandwidth is reduced or that a competing flow has just entered.
 Thus, it indicates that it is not safe to adjust the queue delay
 target.
 A low qdelay_norm_var_t indicates that the queue delay is relatively
 stable.  The reason could be that the queue delay is low, but it

Johansson & Sarker Experimental [Page 21] RFC 8298 SCReAM December 2017

 could also be that a competing flow is causing the bottleneck to
 reach the point that packet losses start to occur, in which case the
 queue delay will stay relatively high for a longer time.
 The queue delay target is allowed to be increased if either the loss
 event rate is above a given threshold or qdelay_norm_var_t is low.
 Both these conditions indicate that a competing flow may be present.
 In all other cases, the queue delay target is decreased.
 The function that adjusts the qdelay_target is simple and could
 produce false positives and false negatives.  The case that self-
 inflicted congestion by the SCReAM algorithm may be falsely
 interpreted as the presence of competing loss-based FTP flows is a
 false positive.  The opposite case -- where the algorithm fails to
 detect the presence of a competing FTP flow -- is a false negative.
 Extensive simulations have shown that the algorithm performs well in
 LTE test cases and that it also performs well in simple bandwidth-
 limited bottleneck test cases with competing FTP flows.  However, the
 potential failure of the algorithm cannot be completely ruled out.  A
 false positive (i.e., when self-inflicted congestion is mistakenly
 identified as competing flows) is especially problematic when it
 leads to increasing the target queue delay, which can cause the end-
 to-end delay to increase dramatically.
 If it is deemed unlikely that competing flows occur over the same
 bottleneck, the algorithm described in this section MAY be turned
 off.  One such case is QoS-enabled bearers in 3GPP-based access such
 as LTE.  However, when sending over the Internet, often the network
 conditions are not known for sure, so in general it is not possible
 to make safe assumptions on how a network is used and whether or not
 competing flows share the same bottleneck.  Therefore, turning this
 algorithm off must be considered with caution, as it can lead to
 basically zero throughput if competing with loss-based traffic.

4.1.2.4. Lost Packet Detection

 Lost packet detection is based on the received sequence number list.
 A reordering window SHOULD be applied to prevent packet reordering
 from triggering loss events.  The reordering window is specified as a
 time unit, similar to the ideas behind Recent ACKnowledgement (RACK)
 [RACK].  The computation of the reordering window is made possible by
 means of a lost flag in the list of transmitted RTP packets.  This
 flag is set if the received sequence number list indicates that the
 given RTP packet is missing.  If later feedback indicates that a
 previously lost marked packet was indeed received, then the
 reordering window is updated to reflect the reordering delay.  The
 reordering window is given by the difference in time between the

Johansson & Sarker Experimental [Page 22] RFC 8298 SCReAM December 2017

 event that the packet was marked as lost and the event that it was
 indicated as successfully received.  Loss is detected if a given RTP
 packet is not acknowledged within a time window (indicated by the
 reordering window) after an RTP packet with a higher sequence number
 was acknowledged.

4.1.2.5. Send Window Calculation

 The basic design principle behind packet transmission in SCReAM is to
 allow transmission only if the number of bytes in flight is less than
 the congestion window.  There are, however, two reasons why this
 strict rule will not work optimally:
 o  Bitrate variations: Media sources such as video encoders generally
    produce frames whose size always vary to a larger or smaller
    extent.  The RTP queue absorbs the natural variations in frame
    sizes.  However, the RTP queue should be as short as possible to
    prevent the end-to-end delay from increasing.  To achieve that,
    the media rate control takes the RTP queue size into account when
    the target bitrate for the media is computed.  A strict 'send only
    when bytes in flight is less than the congestion window' rule can
    cause the RTP queue to grow simply because the send window is
    limited; in turn, this can cause the target bitrate to be pushed
    down.  The consequence is that the congestion window will not
    increase, or will increase very slowly, because the congestion
    window is only allowed to increase when there is a sufficient
    amount of data in flight.  The final effect is that the media
    bitrate increases very slowly or not at all.
 o  Reverse (feedback) path congestion: Especially in transport over
    buffer-bloated networks, the one-way delay in the reverse
    direction can jump due to congestion.  The effect is that the
    acknowledgements are delayed, and the self-clocking is temporarily
    halted, even though the forward path is not congested.
 The send window is adjusted depending on qdelay, its relation to the
 qdelay target, and the relation between the congestion window and the
 number of bytes in flight.  A strict rule is applied when qdelay is
 higher than qdelay_target, to avoid further queue buildup in the
 network.  For cases when qdelay is lower than the qdelay_target, a
 more relaxed rule is applied.  This allows the bitrate to increase
 quickly when no congestion is detected while still being able to
 exhibit stable behavior in congested situations.
 The send window is given by the relation between the adjusted
 congestion window and the amount of bytes in flight according to the
 pseudocode below.

Johansson & Sarker Experimental [Page 23] RFC 8298 SCReAM December 2017

 <CODE BEGINS>
 calculate_send_window(qdelay, qdelay_target)
   # send window is computed differently depending on congestion level
   if (qdelay <= qdelay_target)
     send_wnd = cwnd + MSS - bytes_in_flight
   else
     send_wnd = cwnd - bytes_in_flight
   end
 <CODE ENDS>
 The send window is updated whenever an RTP packet is transmitted or
 an RTCP feedback messaged is received.

4.1.2.6. Packet Pacing

 Packet pacing is used in order to mitigate coalescing, i.e., when
 packets are transmitted in bursts, with the risks of increased jitter
 and potentially increased packet loss.  Packet pacing also mitigates
 possible issues with queue overflow due to key-frame generation in
 video coders.  The time interval between consecutive packet
 transmissions is greater than or equal to 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, and s_rtt is
 the smoothed round trip time.  RATE_PACE_MIN is the minimum pacing
 rate.

4.1.2.7. Resuming Fast Increase Mode

 Fast increase mode can resume in order to speed up the bitrate
 increase if congestion abates.  The condition to resume fast increase
 mode (in_fast_increase = true) is that qdelay_trend is less than
 QDELAY_TREND_LO for T_RESUME_FAST_INCREASE seconds or more.

4.1.2.8. Stream Prioritization

 The SCReAM algorithm makes a good distinction between network
 congestion control and media rate control.  This is easily extended
 to many streams -- RTP packets from two or more RTP queues are
 scheduled at the rate permitted by the network congestion control.
 The scheduling can be done by means of a few different scheduling
 regimes.  For example, the method for coupled congestion control

Johansson & Sarker Experimental [Page 24] RFC 8298 SCReAM December 2017

 specified in [COUPLED-CC] can be used.  One implementation of SCReAM
 [SCReAM-CPP-implementation] uses credit-based scheduling.  In credit-
 based scheduling, credit is accumulated by queues as they wait for
 service and is spent while the queues are being serviced.  For
 instance, if one queue is allowed to transmit 1000 bytes, then a
 credit of 1000 bytes is allocated to the other unscheduled queues.
 This principle can be extended to weighted scheduling, where the
 credit allocated to unscheduled queues depends on the relative
 weights.  The latter is also implemented in
 [SCReAM-CPP-implementation].

4.1.3. Media Rate Control

 The media rate control algorithm is executed at regular intervals,
 indicated by RATE_ADJUSTMENT_INTERVAL, with the exception of a prompt
 reaction to loss events.  The media rate control operates based on
 the size of the RTP packet send queue and observed loss events.  In
 addition, qdelay_trend is also considered in the media rate control
 in order to reduce the amount of induced network jitter.
 The role of the media rate control is to strike a reasonable balance
 between a low amount of queuing in the RTP queue(s) and a sufficient
 amount of data to send in order to keep the data path busy.  Setting
 the media rate control too cautiously leads to possible
 underutilization of network capacity; this can cause the flow to
 become starved out by other more opportunistic traffic.  On the other
 hand, setting it too aggressively leads to increased jitter.
 The target_bitrate is adjusted depending on the congestion state.
 The target bitrate can vary between a minimum value
 (TARGET_BITRATE_MIN) and a maximum value (TARGET_BITRATE_MAX).
 TARGET_BITRATE_MIN SHOULD be set to a low enough value to prevent RTP
 packets from becoming queued up when the network throughput is
 reduced.  The sender SHOULD also be equipped with a mechanism that
 discards RTP packets when the network throughput becomes very low and
 RTP packets are excessively delayed.
 For the overall bitrate adjustment, two network throughput estimates
 are computed :
 o  rate_transmit: The measured transmit bitrate.
 o  rate_ack: The ACKed bitrate, i.e., the volume of ACKed bits per
    second.
 Both estimates are updated every 200 ms.

Johansson & Sarker Experimental [Page 25] RFC 8298 SCReAM December 2017

 The current throughput, current_rate, is computed as the maximum
 value of rate_transmit and rate_ack.  The rationale behind the use of
 rate_ack in addition to rate_transmit is that rate_transmit is
 affected also by the amount of data that is available to transmit,
 thus a lack of data to transmit can be seen as reduced throughput
 that can cause an unnecessary rate reduction.  To overcome this
 shortcoming, rate_ack is used as well.  This gives a more stable
 throughput estimate.
 The rate change behavior depends on whether a loss or ECN event has
 occurred and whether the congestion control is in fast increase mode.
 <CODE BEGINS>
 # The target_bitrate is updated at a regular interval according
 # to RATE_ADJUST_INTERVAL
 on loss:
    # Loss event detected
    target_bitrate = max(BETA_R * target_bitrate,
                         TARGET_BITRATE_MIN)
    exit
 on ecn_mark:
    # ECN event detected
    target_bitrate = max(BETA_ECN * target_bitrate,
                         TARGET_BITRATE_MIN)
    exit
 ramp_up_speed_t = min(RAMP_UP_SPEED, target_bitrate / 2.0)
 scale_t = (target_bitrate - target_bitrate_last_max) /
      target_bitrate_last_max
 scale_t = max(0.2, min(1.0, (scale_t * 4)^2))
 # min scale_t value 0.2, as the bitrate should be allowed to
 #  increase slowly. This prevents locking the rate to
 #  target_bitrate_last_max
 if (in_fast_increase = true)
    increment_t = ramp_up_speed_t * RATE_ADJUST_INTERVAL
    increment_t *= scale_t
    target_bitrate += increment_t
 else
    current_rate_t = max(rate_transmit, rate_ack)
    # Compute a bitrate change
    delta_rate_t = current_rate_t * (1.0 - PRE_CONGESTION_GUARD *
         queue_delay_trend) - TX_QUEUE_SIZE_FACTOR * rtp_queue_size
    # Limit a positive increase if close to target_bitrate_last_max
    if (delta_rate_t > 0)
      delta_rate_t *= scale_t
      delta_rate_t =
        min(delta_rate_t, ramp_up_speed_t * RATE_ADJUST_INTERVAL)

Johansson & Sarker Experimental [Page 26] RFC 8298 SCReAM December 2017

    end
    target_bitrate += delta_rate_t
    # Force a slight reduction in bitrate if RTP queue
    #  builds up
    rtp_queue_delay_t = rtp_queue_size / current_rate_t
    if (rtp_queue_delay_t > RTP_QDELAY_TH)
      target_bitrate *= TARGET_RATE_SCALE_RTP_QDELAY
    end
 end
 rate_media_limit_t =
    max(current_rate_t, max(rate_media, rtp_rate_median))
 rate_media_limit_t *= (2.0 - qdelay_trend_mem)
 target_bitrate = min(target_bitrate, rate_media_limit_t)
 target_bitrate = min(TARGET_BITRATE_MAX,
    max(TARGET_BITRATE_MIN, target_bitrate))
 <CODE ENDS>
 In case of a loss event, the target_bitrate is updated and the rate
 change procedure is exited.  Otherwise, the rate change procedure
 continues.  The rationale behind the rate reduction due to loss is
 that a congestion window reduction will take effect, and a rate
 reduction proactively prevents RTP packets from being queued up when
 the transmit rate decreases due to the reduced congestion window.  A
 similar rate reduction happens when ECN events are detected.
 The rate update frequency is limited by RATE_ADJUST_INTERVAL, unless
 a loss event occurs.  The value is based on experimentation with
 real-life limitations in video coders taken into account
 [SCReAM-CPP-implementation].  A too short interval is shown to make
 the rate control loop in video coders more unstable; a too long
 interval makes the overall congestion control sluggish.
 When in fast increase mode (in_fast_increase = true), the bitrate
 increase is given by the desired ramp-up speed (RAMP_UP_SPEED).  The
 ramp-up speed is limited when the target bitrate is low to avoid rate
 oscillation at low bottleneck bitrates.  The setting of RAMP_UP_SPEED
 depends on preferences.  A high setting such as 1000 kbps/s makes it
 possible to quickly get high-quality media; however, this is at the
 expense of increased jitter, which can manifest itself as choppy
 video rendering, for example.
 When in_fast_increase is false, the bitrate increase is given by the
 current bitrate and is also controlled by the estimated RTP queue and
 the qdelay trend, thus it is sufficient that an increased congestion
 level is sensed by the network congestion control to limit the
 bitrate.  The target_bitrate_last_max is updated when congestion is
 detected.

Johansson & Sarker Experimental [Page 27] RFC 8298 SCReAM December 2017

 Finally, the target_bitrate is within the defined min and max values.
 The aware reader may notice the dependency on the qdelay in the
 computation of the target bitrate; this manifests itself in the use
 of the qdelay_trend.  As these parameters are used also in the
 network congestion control, one may suspect some odd interaction
 between the media rate control and the network congestion control.
 This is in fact the case if the parameter PRE_CONGESTION_GUARD is set
 to a high value.  The use of qdelay_trend in the media rate control
 is solely to reduce jitter; the dependency can be removed by setting
 PRE_CONGESTION_GUARD=0.  The effect is a somewhat larger rate
 increase after congestion, at the expense of increased jitter in
 congested situations.

4.2. SCReAM Receiver

 The simple task of the SCReAM receiver is to feed back
 acknowledgements of received packets and total ECN count to the
 SCReAM sender.  In addition, the receive time of the RTP packet with
 the highest sequence number is echoed back.  Upon reception of each
 RTP packet, the receiver MUST maintain enough information to send the
 aforementioned values to the SCReAM sender via an RTCP transport-
 layer feedback message.  The frequency of the feedback message
 depends on the available RTCP bandwidth.  The requirements on the
 feedback elements and the feedback interval are described below.

4.2.1. Requirements on Feedback Elements

 The following feedback elements are REQUIRED for basic functionality
 in SCReAM.
 o  A list of received RTP packets.  This list SHOULD be sufficiently
    long to cover all received RTP packets.  This list can be realized
    with the Loss RLE (Run Length Encoding) 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 sequence number (which has possibly wrapped
    around); end_seq MUST be set to begin_seq+1 modulo 65536.  The
    timestamp clock MAY be set according to [RFC3611], i.e., equal to
    the RTP timestamp clock.  Detailed individual packet receive times
    are not necessary, as SCReAM does currently not describe how they
    can be used.

Johansson & Sarker Experimental [Page 28] RFC 8298 SCReAM December 2017

 The basic feedback needed for SCReAM involves the use of the Loss RLE
 Report Block and the Packet Receipt Times Report Block as shown 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 RFC 3611
 In a typical use case, no more than four Loss RLE chunks are needed,
 thus the feedback message will be 44 bytes.  It is obvious from
 Figure 2 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.
 An additional feedback element that can improve the performance of
 SCReAM is:
 o  Accumulated number of ECN-CE-marked packets (n_ECN).  For
    instance, this can be realized with the ECN Feedback Report Format
    in [RFC6679].  The given feedback report format is slightly
    overkill, as SCReAM would do quite well with only a counter that

Johansson & Sarker Experimental [Page 29] RFC 8298 SCReAM December 2017

    increments by one for each received packet with the ECN-CE
    codepoint set.  The more bulky format could nevertheless be useful
    for, e.g., ECN black-hole detection.

4.2.2. Requirements on Feedback Intensity

 SCReAM benefits from relatively frequent feedback.  It is RECOMMENDED
 that a SCReAM implementation follows the guidelines below.
 The feedback interval depends on the media bitrate.  At low bitrates,
 it is sufficient with a feedback interval of 100 to 400 ms; while at
 high bitrates, a feedback interval of roughly 20 ms is preferred.  At
 very high bitrates, even shorter feedback intervals MAY be needed in
 order to keep the self-clocking in SCReAM working well.  One
 indication that feedback is too sparse is that the SCReAM
 implementation cannot reach high bitrates, even in uncongested links.
 More frequent feedback might solve this issue.
 The numbers above can be formulated as a feedback interval function
 that can be useful for the computation of the desired RTCP bandwidth.
 The following equation expresses the feedback rate:
    rate_fb = min(50, max(2.5, rate_media / 10000))
 rate_media is the RTP media bitrate expressed in bps; rate_fb is the
 feedback rate expressed in packets/s.  Converting to feedback
 interval, we get:
    fb_int = 1.0 / min(50, max(2.5, rate_media / 10000))
 The transmission interval is not critical.  So, in the case of multi-
 stream handling between two hosts, the feedback for two or more
 synchronization sources (SSRCs) can be bundled to save UDP/IP
 overhead.  However, the final realized feedback interval SHOULD not
 exceed 2*fb_int in such cases, meaning that a scheduled feedback
 transmission event should not be delayed more than fb_int.
 SCReAM works with AVPF regular mode; immediate or early mode is not
 required by SCReAM but can nonetheless be useful for 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].

Johansson & Sarker Experimental [Page 30] RFC 8298 SCReAM December 2017

5. Discussion

 This section covers a few discussion points.
 o  Clock drift: SCReAM can suffer from the same issues with clock
    drift as is the case with LEDBAT [RFC6817].  However, Appendix A.2
    in [RFC6817] describes ways to mitigate issues with clock drift.
 o  Support for alternate ECN semantics: This specification adopts the
    proposal in [ALT-BACKOFF] to reduce the congestion window less
    when ECN-based congestion events are detected.  Future work on Low
    Loss, Low Latency for Scalable throughput (L4S) may lead to
    updates in a future document that describes SCReAM support for
    L4S.
 o  A new transport-layer feedback message (as specified in RFC 4585)
    could 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 is the bitrate including the
    RTP and Forward Error Correction (FEC) overhead.  The media
    encoder SHOULD take this overhead into account when the media
    bitrate is set.  This means that the media coder bitrate SHOULD be
    computed as
    media_rate = target_bitrate - rtp_plus_fec_overhead_bitrate
    It is not necessary to make a 100% perfect compensation for the
    overhead, as the SCReAM algorithm will inherently compensate for
    moderate errors.  Under-compensating for the overhead has the
    effect of increasing jitter, while overcompensating will cause the
    bottleneck link to become underutilized.

6. Suggested Experiments

 SCReAM has been evaluated in a number of different ways, mostly in a
 simulator.  The OpenWebRTC implementation work ([OpenWebRTC] and
 [SCReAM-implementation]) involved extensive testing with artificial
 bottlenecks with varying bandwidths and using two different video
 coders (OpenH264 and VP9).

Johansson & Sarker Experimental [Page 31] RFC 8298 SCReAM December 2017

 Preferably, further experiments will be done by means of
 implementation in real clients and web browsers.  RECOMMENDED
 experiments are:
 o  Trials with various access technologies: EDGE/3G/4G, Wi-Fi, DSL.
    Some experiments have already been carried out with LTE access;
    see [SCReAM-CPP-implementation] and
    [SCReAM-implementation-experience].
 o  Trials with different kinds of media: Audio, video, slideshow
    content.  Evaluation of multi-stream handling in SCReAM.
 o  Evaluation of functionality of the compensation mechanism when
    there are competing flows: Evaluate how SCReAM performs with
    competing TCP-like traffic and to what extent the compensation for
    competing flows causes self-inflicted congestion.
 o  Determine proper parameters: A set of default parameters are given
    that makes SCReAM work over a reasonably large operation range.
    However, for very low or very high bitrates, it may be necessary
    to use different values for the RAMP_UP_SPEED, for instance.
 o  Experimentation with further improvements to the congestion window
    and media bitrate calculation.  [SCReAM-CPP-implementation]
    implements some optimizations, not described in this memo, that
    improve performance slightly.  Further experiments are likely to
    lead to more optimizations of the algorithm.

7. IANA Considerations

 This document does not require any IANA actions.

8. Security Considerations

 The feedback can be vulnerable to attacks similar to those that can
 affect TCP.  It is therefore RECOMMENDED that the RTCP feedback is at
 least integrity protected.  Furthermore, as SCReAM is self-clocked, a
 malicious middlebox can drop RTCP feedback packets and thus cause the
 self-clocking in SCReAM to stall.  However, this attack is mitigated
 by the minimum send rate maintained by SCReAM when no feedback is
 received.

Johansson & Sarker Experimental [Page 32] RFC 8298 SCReAM December 2017

9. References

9.1. Normative References

 [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
            Requirement Levels", BCP 14, RFC 2119,
            DOI 10.17487/RFC2119, March 1997,
            <https://www.rfc-editor.org/info/rfc2119>.
 [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
            Jacobson, "RTP: A Transport Protocol for Real-Time
            Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
            July 2003, <https://www.rfc-editor.org/info/rfc3550>.
 [RFC3611]  Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed.,
            "RTP Control Protocol Extended Reports (RTCP XR)",
            RFC 3611, DOI 10.17487/RFC3611, November 2003,
            <https://www.rfc-editor.org/info/rfc3611>.
 [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
            "Extended RTP Profile for Real-time Transport Control
            Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
            DOI 10.17487/RFC4585, July 2006,
            <https://www.rfc-editor.org/info/rfc4585>.
 [RFC5506]  Johansson, I. and M. Westerlund, "Support for Reduced-Size
            Real-Time Transport Control Protocol (RTCP): Opportunities
            and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
            2009, <https://www.rfc-editor.org/info/rfc5506>.
 [RFC6298]  Paxson, V., Allman, M., Chu, J., and M. Sargent,
            "Computing TCP's Retransmission Timer", RFC 6298,
            DOI 10.17487/RFC6298, June 2011,
            <https://www.rfc-editor.org/info/rfc6298>.
 [RFC6817]  Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind,
            "Low Extra Delay Background Transport (LEDBAT)", RFC 6817,
            DOI 10.17487/RFC6817, December 2012,
            <https://www.rfc-editor.org/info/rfc6817>.
 [RFC8174]  Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
            2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
            May 2017, <https://www.rfc-editor.org/info/rfc8174>.

Johansson & Sarker Experimental [Page 33] RFC 8298 SCReAM December 2017

9.2. Informative References

 [ALT-BACKOFF]
            Khademi, N., Welzl, M., Armitage, G., and G. Fairhurst,
            "TCP Alternative Backoff with ECN (ABE)", Work in
            Progress, draft-ietf-tcpm-alternativebackoff-ecn-04,
            November 2017.
 [COUPLED-CC]
            Islam, S., Welzl, M., and S. Gjessing, "Coupled congestion
            control for RTP media", Work in Progress, draft-ietf-
            rmcat-coupled-cc-07, September 2017.
 [LEDBAT-delay-impact]
            Ros, D. and M. Welzl, "Assessing LEDBAT's Delay Impact",
            IEEE Communications Letters, Vol. 17, No. 5,
            DOI 10.1109/LCOMM.2013.040213.130137, May 2013,
            <http://home.ifi.uio.no/michawe/research/publications/
            ledbat-impact-letters.pdf>.
 [OpenWebRTC]
            Ericsson Research, "OpenWebRTC",
            <http://www.openwebrtc.org>.
 [Packet-conservation]
            Jacobson, V., "Congestion Avoidance and Control", ACM
            SIGCOMM Computer Communication Review,
            DOI 10.1145/52325.52356, August 1988.
 [QoS-3GPP] 3GPP, "Policy and charging control architecture", 3GPP TS
            23.203, July 2017,
            <http://www.3gpp.org/ftp/specs/archive/23_series/23.203/>.
 [RACK]     Cheng, Y., Cardwell, N., and N. Dukkipati, "RACK: a time-
            based fast loss detection algorithm for TCP", Work in
            Progress, draft-ietf-tcpm-rack-02, March 2017.
 [RFC6679]  Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
            and K. Carlberg, "Explicit Congestion Notification (ECN)
            for RTP over UDP", RFC 6679, DOI 10.17487/RFC6679, August
            2012, <https://www.rfc-editor.org/info/rfc6679>.
 [RFC7478]  Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
            Time Communication Use Cases and Requirements", RFC 7478,
            DOI 10.17487/RFC7478, March 2015,
            <https://www.rfc-editor.org/info/rfc7478>.

Johansson & Sarker Experimental [Page 34] RFC 8298 SCReAM December 2017

 [RFC7661]  Fairhurst, G., Sathiaseelan, A., and R. Secchi, "Updating
            TCP to Support Rate-Limited Traffic", RFC 7661,
            DOI 10.17487/RFC7661, October 2015,
            <https://www.rfc-editor.org/info/rfc7661>.
 [SCReAM-CPP-implementation]
            Ericsson Research, "SCReAM - Mobile optimised congestion
            control algorithm",
            <https://github.com/EricssonResearch/scream>.
 [SCReAM-implementation]
            Ericsson Research, "OpenWebRTC specific GStreamer
            plugins", <https://github.com/EricssonResearch/
            openwebrtc-gst-plugins>.
 [SCReAM-implementation-experience]
            Sarker, Z. and I. Johansson, "Updates on SCReAM: An
            implementation experience", November 2015,
            <https://www.ietf.org/proceedings/94/slides/
            slides-94-rmcat-8.pdf>.
 [TFWC]     Choi, S. and M. Handley, "Fairer TCP-Friendly Congestion
            Control Protocol for Multimedia Streaming Applications",
            DOI 10.1145/1364654.1364717, December 2007,
            <http://www-dept.cs.ucl.ac.uk/staff/M.Handley/papers/
            tfwc-conext.pdf>.
 [WIRELESS-TESTS]
            Sarker, Z., Johansson, I., Zhu, X., Fu, J., Tan, W., and
            M. Ramalho, "Evaluation Test Cases for Interactive Real-
            Time Media over Wireless Networks", Work in Progress,
            draft-ietf-rmcat-wireless-tests-04, May 2017.

Johansson & Sarker Experimental [Page 35] RFC 8298 SCReAM December 2017

Acknowledgements

 We would like to thank the following people for their comments,
 questions, and support during the work that led to this memo: Markus
 Andersson, Bo Burman, Tomas Frankkila, Frederic Gabin, Laurits Hamm,
 Hans Hannu, Nikolas Hermanns, Stefan Haakansson, Erlendur Karlsson,
 Daniel Lindstroem, Mats Nordberg, Jonathan Samuelsson, Rickard
 Sjoeberg, Robert Swain, Magnus Westerlund, and Stefan Aalund.  Many
 additional thanks to RMCAT chairs Karen E. E. Nielsen and Mirja
 Kuehlewind for patiently reading, suggesting improvements and also
 for asking all the difficult but necessary questions.  Thanks to
 Stefan Holmer, Xiaoqing Zhu, Safiqul Islam, and David Hayes for the
 additional review of this document.  Thanks to Ralf Globisch for
 taking time to try out SCReAM in his challenging low-bitrate use
 cases, Robert Hedman for finding a few additional flaws in the
 running code, and Gustavo Garcia and 'miseri' for code contributions.

Authors' Addresses

 Ingemar Johansson
 Ericsson AB
 Laboratoriegraend 11
 Luleaa  977 53
 Sweden
 Phone: +46 730783289
 Email: ingemar.s.johansson@ericsson.com
 Zaheduzzaman Sarker
 Ericsson AB
 Laboratoriegraend 11
 Luleaa  977 53
 Sweden
 Phone: +46 761153743
 Email: zaheduzzaman.sarker@ericsson.com

Johansson & Sarker Experimental [Page 36]

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