RMCAT WG                                                    I. Johansson
Internet-Draft                                                 Z. Sarker
Intended status: Experimental                                Ericsson AB
Expires: February 16, 2017                               August 15, 2016

              Self-Clocked Rate Adaptation for Multimedia


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

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
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   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on February 16, 2017.

Copyright Notice

   Copyright (c) 2016 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
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   include Simplified BSD License text as described in Section 4.e of

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   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
     1.1.  Wireless (LTE) access properties  . . . . . . . . . . . .   3
     1.2.  Why is it a self-clocked algorithm? . . . . . . . . . . .   4
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   4
   3.  Overview of SCReAM Algorithm  . . . . . . . . . . . . . . . .   4
     3.1.  Network Congestion Control  . . . . . . . . . . . . . . .   7
     3.2.  Sender Transmission Control . . . . . . . . . . . . . . .   7
     3.3.  Media Rate Control  . . . . . . . . . . . . . . . . . . .   7
   4.  Detailed Description of SCReAM  . . . . . . . . . . . . . . .   8
     4.1.  SCReAM Sender . . . . . . . . . . . . . . . . . . . . . .   8
       4.1.1.  Constants and Parameter values  . . . . . . . . . . .   9  Constants . . . . . . . . . . . . . . . . . . . .   9  State variables . . . . . . . . . . . . . . . . .  10
       4.1.2.  Network congestion control  . . . . . . . . . . . . .  12  Congestion window update  . . . . . . . . . . . .  15  Competing flows compensation  . . . . . . . . . .  17  Lost packets detection  . . . . . . . . . . . . .  19  Send window calculation . . . . . . . . . . . . .  19  Resuming fast increase  . . . . . . . . . . . . .  20
       4.1.3.  Media rate control  . . . . . . . . . . . . . . . . .  20  FEC and packet overhead considerations  . . . . .  23
     4.2.  SCReAM Receiver . . . . . . . . . . . . . . . . . . . . .  24
   5.  Discussion  . . . . . . . . . . . . . . . . . . . . . . . . .  24
   6.  Implementation status . . . . . . . . . . . . . . . . . . . .  24
     6.1.  OpenWebRTC  . . . . . . . . . . . . . . . . . . . . . . .  25
     6.2.  A C++ Implementation of SCReAM  . . . . . . . . . . . . .  25
   7.  Suggested experiments . . . . . . . . . . . . . . . . . . . .  26
   8.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  26
   9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  27
   10. Security Considerations . . . . . . . . . . . . . . . . . . .  27
   11. Change history  . . . . . . . . . . . . . . . . . . . . . . .  27
   12. References  . . . . . . . . . . . . . . . . . . . . . . . . .  28
     12.1.  Normative References . . . . . . . . . . . . . . . . . .  28
     12.2.  Informative References . . . . . . . . . . . . . . . . .  28
   Appendix A.  Additional information . . . . . . . . . . . . . . .  30
     A.1.  Stream prioritization . . . . . . . . . . . . . . . . . .  30
     A.2.  Computation of autocorrelation function . . . . . . . . .  31
     A.3.  Sender transmission control and packet pacing . . . . . .  31
     A.4.  RTCP feedback considerations  . . . . . . . . . . . . . .  31
       A.4.1.  Requirements on feedback elements . . . . . . . . . .  32
       A.4.2.  Requirements on feedback intensity  . . . . . . . . .  34
     A.5.  Q-bit semantics (source quench) . . . . . . . . . . . . .  34
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  35

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1.  Introduction

   Congestion in the Internet occurs when the transmitted bitrate is
   higher than the available bandwidth over a given transmission path.
   Applications that are deployed in the Internet must have congestion
   control schemes in place not only for the robustness of the service
   that it provides but also to ensure the function of the currently
   deployed Internet.  Interactive realtime communication imposes a lot
   of requirements on the transport, therefore a robust, efficient rate
   adaptation for all access types is an important part of interactive
   realtime communications as the transmission channel bandwidth may
   vary over time.  Wireless access such as LTE, which is an integral
   part of the current Internet, increases the importance of rate
   adaptation as the channel bandwidth of a default LTE bearer
   [QoS-3GPP] can change considerably in a very short time frame.  Thus
   a rate adaptation solution for interactive realtime media, such as
   WebRTC, must be both quick and be able to operate over a large span
   in available channel bandwidth.  This memo describes a solution,named
   SCReAM (Self-Clocked Rate Adaptation for Multimedia), that is based
   on the self-clocking principle of TCP and uses techniques similar to
   what is used in a new delay based rate adaptation algorithm, LEDBAT

1.1.  Wireless (LTE) access properties

   [I-D.ietf-rmcat-wireless-tests] describes the complications that can
   be observed in wireless environments.  Wireless access such as LTE
   can typically not guarantee a given bandwidth, this is true
   especially for default bearers.  The network throughput may vary
   considerably for instance in cases where the wireless terminal is
   moving around.  Even though LTE can support bitrates well above
   100Mbps, there are cases when the available bitrate can be much
   lower, examples are situations with high network load and poor

   Unlike wireline bottlenecks with large statistical multiplexing it is
   not possible to try to maintain a given bitrate when congestion is
   detected with the hope that other flows will yield, this is because
   there are generally few other flows competing for the same
   bottleneck.  Each user gets its own variable throughput bottleneck,
   where the throughput depends on factors like channel quality, network
   load and historical throughput.  The bottom line is, if the
   throughput drops, the sender has no other option than to reduce the
   bitrate.  Once the radio scheduler has reduced the resource
   allocation for a bearer, an RMCAT flow in that bearer needs to reduce
   the sending rate quite quickly (in one RTT) in order to avoid
   excessive queuing delay or packet loss.

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1.2.  Why is it a self-clocked algorithm?

   Self-clocked congestion control algorithm provides with a benefit
   over the rate based counterparts in that the former consists of two
   parts; the congestion window computation that evolves over a longer
   timescale (several RTTs) especially when the congestion window
   evolution is dictated by estimated delay (to minimize vulnerability
   to e.g. short term delay variations) and; the fine grained congestion
   control given by the self-clocking which operates on a shorter time
   scale (1 RTT).  The benefits of self-clocking are also elaborated
   upon in [TFWC].

   A rate based congestion control typically adjusts the rate based on
   delay and loss.  The congestion detection needs to be done with a
   certain time lag to avoid over-reaction to spurious congestion events
   such as delay spikes.  Despite the fact that there are two or more
   congestion indications, the outcome is still that there is only one
   mechanism to adjust the sending rate.  This makes it difficult to
   reach the goals of high throughput and prompt reaction to congestion.

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC2119 [RFC2119]

3.  Overview of SCReAM Algorithm

   The core SCReAM algorithm has similarities to the concepts of self-
   clocking used in TFWC [TFWC] and follows the packet conservation
   principle.  The packet conservation principle is described as an
   important key-factor behind the protection of networks from
   congestion [PACKET_CONSERVATION].

   In 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 amount
   of bytes that can be transmitted at any given time instant.  A
   congestion window puts an upper limit on how many bytes can be in
   flight, i.e transmitted but not yet acknowledged.  All this
   implements a congestion control that follows the packet conservation
   principle.  The fact that SCReAM follows the packet conservation
   principle, makes it as safe to deploy as a congestion control
   algorithm for the Internet as TCP and its most commonly used
   congestion control algorithms are.  No additional circuit breaker
   mechanisms are necessary with SCReAM as the ACK-clocking

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   automatically falls back to a very low transmission rate (1 RTP
   packet/200ms) when the acknowledgements no longer arrive at the
   sender.  Furthermore, high packet loss rates reduces the congestion
   value to very low values and thus a low transmission rate.

   The congestion window is determined in a way similar to LEDBAT

   LEDBAT is a congestion control algorithm that uses send and receive
   timestamps to estimate the queuing delay along the transmission path.
   This information is used to adjust the congestion window.  The use of
   LEDBAT ensures that the end-to-end latency is kept low.  The basic
   functionality is quite simple, there are however a few steps to take
   to make the concept work with conversational media.  In a few words
   they are:

   o  Congestion window validation techniques.  These are similar in
      action as the method described in [RFC7661].  Congestion window
      validation ensures that the congestion window is limited by the
      amount of actual bytes in flight, this is important especially in
      the context of rate limited sources such as video.  Lack of
      congestion window validation would lead to a slow reaction to
      congestion as the congestion window does not properly reflect the
      congestion state in the network.  The allowed idle period in this
      memo is shorter than in [RFC7661], this to avoid excessive delays
      in the cases where e.g. wireless throughput has decreased during a
      period where the output bitrate from the media coder has been low,
      for instance due to inactivity.  Furthermore, this memo allows for
      more relaxed rules for when the congestion window is allowed to
      grow, this is necessary as the variable output bitrate generally
      means that the congestion window is often under-utilized.

   o  Fast increase for quicker bitrate increase.  It makes the media
      bitrate ramp-up within 5 to 10 seconds.  The behavior is similar
      to TCP slowstart.  The fast increase is exited when congestion is
      detected.  The fast increase state can however resume if the
      congestion level is low, this to enable a reasonably quick rate
      increase in case link throughput increases.

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

   o  Addition of a media rate control function.

   o  Use of inflection points in the media rate calculation to achieve
      reduced jitter.

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

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

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

                  Figure 1: SCReAM sender functional view

   The SCReAM algorithm constitutes mainly three parts: network
   congestion control, sender transmission control and media rate
   control.  All these three parts reside at the sender side.  Figure 1
   shows the functional overview of a SCReAM sender.  The receiver side

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   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.

   The SCReAM congestion control method, uses techniques similar to
   LEDBAT [RFC6817] to measure the queuing delay, also termed qdelay in
   this memo for brevity.  Similar to LEDBAT, it is not necessary to use
   synchronized clocks in sender and receiver in order to compute the
   queuing delay.  It is however necessary that they use the same clock
   frequency, or that the clock frequency at the receiver can be
   inferred reliably by the sender.

   The SCReAM sender calculates the congestion window based on the
   feedback from the SCReAM receiver.  The congestion window is allowed
   to increase if the qdelay is below a predefined qdelay target,
   otherwise the congestion window decreases.  The qdelay delay target
   is typically set to 50-100ms.  This ensures that the queuing delay is
   kept low.  The reaction to loss or ECN events leads to an instant
   reduction of CWND.  Note that the source rate limited nature of real
   time media such as video, typically means that the queuing delay will
   mostly be below the given delay target, this is contrary to the case
   where large files are transmitted using LEDBAT congestion control, in
   which case the queuing delay will stay close to the delay target.

3.2.  Sender Transmission Control

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

3.3.  Media Rate Control

   The media rate control serves to adjust the media bitrate to ramp up
   quickly enough to get a fair share of the system resources when link
   throughput increases.

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   The reaction to reduced throughput must be prompt in order to avoid
   getting too much data queued up in the RTP packet queue(s) in the
   sender.  The media bitrate is decreased if the RTP queue size exceeds
   a threshold.

   In cases where the sender frame queues increase rapidly such as the
   case of a RAT (Radio Access Type) handover it may be necessary to
   implement additional actions, such as discarding of encoded media
   frames or frame skipping in order to ensure that the RTP queues are
   drained quickly or simply that stale RTP packets are removed from the
   queue.  Frame skipping means that the frame rate is temporarily
   reduced.  Which method to use is a design consideration and outside
   the scope of this algorithm description.

4.  Detailed Description of SCReAM

4.1.  SCReAM Sender

   This section describes the sender side algorithm in more detail.  It
   is a split between the network congestion control, sender
   transmission control and the media rate control.

   A SCReAM sender implements media rate control and a queue for each
   media type or source, where RTP packets containing encoded media
   frames are temporarily stored for transmission.  Figure 1 shows the
   details when a single media source (a.k.a stream) is used.  Multiple
   media sources are also supported in the design, in that case the
   sender transmission control will include a transmission scheduler.
   The transmission scheduler can then enforce the priorities for the
   different streams and then act like a coupled congestion controller
   for multiple flows.

   Media frames are encoded and forwarded to the RTP queue (1) in
   Figure 1.  The media rate adaptation adapts to the size of the RTP
   queue (2) and controls the media bitrate (3).  The RTP packets are
   picked from the RTP queue (for multiple flows from each RTP queue
   based on some defined priority order or simply in a round robin
   fashion) (4) by the sender transmission controller.  The sender
   transmission controller (in case of multiple flows a transmission
   scheduler) takes care of the transmission of RTP packets, to be
   written to the UDP socket (5).  In the general case all media must go
   through the sender transmission controller and is allowed to be
   transmitted if the number of bytes in flight is less than the
   congestion window.  RTCP packets are received (6) and the information
   about bytes in flight and congestion window is exchanged between the
   network congestion control and the sender transmission control (7).

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4.1.1.  Constants and Parameter values

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

   The recommended values for the constants are deduced from

     Target value for the minimum qdelay.

     Target value for the maximum qdelay.

     Averaging factor for qdelay_fraction_avg.

     Headroom for the limitation of CWND.

   GAIN (1.0)
     Gain factor for congestion window adjustment.

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

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

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

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

     Interval between media bitrate adjustments.

     Min target bitrate [bps].

     Max target bitrate [bps].

   RAMP_UP_SPEED (200000bps/s)

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     Maximum allowed rate increase speed.

     Guard factor against early congestion onset.  A higher value gives
     less jitter, possibly at the expense of a lower link utilization.
     This value may be subject to tuning depending on e.g media coder
     characteristics, experiments with H264 and VP8 have however given
     that 0.1 is a suitable value.

   TX_QUEUE_SIZE_FACTOR (0.0..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 have however given that 1.0 is a suitable value.

   RTP_QDELAY_TH (0.02s)  RTP queue delay threshold for a target rate

   TARGET_RATE_SCALE_RTP_QDELAY (0.95)  Target rate scale when RTP queue
     delay threshold exceeded.

   QDELAY_TREND_LO (0.2)  Threshold value for qdelay_trend.

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

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

   qdelay_fraction_avg (0.0)
     EWMA filtered fractional qdelay.

   qdelay_fraction_hist[20] ({0,..,0})
     Vector of the last 20 fractional qdelay samples.

   qdelay_trend (0.0)
     qdelay trend, indicates incipient congestion.

   qdelay_trend_mem (0.0)
     Low pass filtered version of qdelay_trend.

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

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   min_cwnd (2*MSS)
     Minimum congestion window.

   in_fast_increase (true)
     True if in fast increase state.

   cwnd (min_cwnd)
     Congestion window.

   bytes_newly_acked (0)
     The number of bytes that was acknowledged with the last received
     acknowledgement i.e bytes acknowledged since the last CWND update.

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

   target_bitrate (0 bps)
     Media target bitrate.

   target_bitrate_last_max (1 bps)
     Media target bitrate inflection point i.e the last known highest
     target_bitrate.  Used to limit bitrate increase speed close to the
     last known congestion point.

   rate_transmit (0.0 bps)
     Measured transmit bitrate.

   rate_ack (0.0 bps)
     Measured throughput based on received acknowledgements.

   rate_media (0.0 bps)
     Measured bitrate from the media encoder.

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

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

   rtp_queue_size (0 bits)
     Size 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.

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4.1.2.  Network congestion control

   This section explains the network congestion control, it contains two
   main functions

   o  Computation of congestion window at the sender: Gives an upper
      limit to the number of bytes in flight i.e how many bytes that
      have been transmitted but not yet acknowledged.

   o  Calculation of send window at the sender: RTP packets are
      transmitted if allowed by the relation between the number of bytes
      in flight and the congestion window.  This is controlled by the
      send window.

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

   SCReAM uses the terminology "Bytes in flight" (bytes_in_flight) which
   is computed as the sum of the sizes of the RTP packets ranging from
   the RTP packet most recently transmitted down to but not including
   the acknowledged packet with the highest sequence number.  This can
   be translated to the difference between the highest transmitted byte
   sequence number and the highest acknowledged byte sequence number.
   As an example: If RTP packet with sequence number SN is transmitted
   and the last acknowledgement indicates SN-5 as the highest received
   sequence number then bytes in flight is computed as the sum of the
   size of RTP packets with sequence number SN-4, SN-3, SN-2, SN-1 and
   SN, it does not matter if for instance packet with sequence number
   SN-3 was lost, 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 is reset after a CWND update.

   The feedback from the receiver is assumed to consist of the following
   elements.  More details are found in Appendix A.4.

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   o  A list of received RTP packets.

   o  The wall clock timestamp corresponding to the received RTP packet
      with the highest sequence number.

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

   When the sender receives RTCP feedback, the qdelay is calculated as
   outlined in [RFC6817].  A qdelay sample is obtained for each received
   acknowledgement.  No smoothing of the qdelay samples occur, however
   some smoothing occurs anyway as the computation of the CWND is in
   itself a low pass filter function.  A number of variables are updated
   as illustrated by the pseudo code below, temporary variables are
   appended with '_t'.  Note that the pseudo code does not show all
   details for reasons of readability, the reader is referred to the C++
   code in [SCReAM-Cplusplus_Implementation] for the details.

       qdelay_fraction_t = qdelay/qdelay_target
       #calculate moving average
       qdelay_fraction_avg = (1-QDELAY_WEIGHT)*qdelay_fraction_avg+
       # R is an autocorrelation function of qdelay_fraction_hist
       #  at lag K
       a = R(qdelay_fraction_hist,1)/R(qdelay_fraction_hist,0)
       #calculate qdelay trend
       qdelay_trend = min(1.0,max(0.0,a*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)

   The qdelay fraction is sampled every 50ms and the last 20 samples are
   stored in a vector (qdelay_fraction_hist).  This vector is used in
   the computation of an qdelay trend that gives a value between 0.0 and
   1.0 depending on the estimated congestion level.  The prediction
   coefficient 'a' has positive values if qdelay shows an increasing
   trend, thus an indication of congestion is obtained before the qdelay
   target is reached.  The autocorrelation function 'R' is defined in
   Appendix A.2.  The prediction coefficient is further multiplied with
   qdelay_fraction_avg to reduce sensitivity to increasing qdelay when
   it is very small.  The 50ms sampling is a simplification and may have
   the effect that the same qdelay is sampled several times, this is
   however not a big issue as the vector is only used for the
   computation of qdelay_trend.  The 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

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   function update_qdelay_fraction_hist(..) removes the oldest element
   and adds the latest qdelay_fraction element to the
   qdelay_fraction_hist vector.

   A loss event is indicated if one or more RTP packets are declared
   missing.  The loss detection is described in Section  Once a
   loss event is detected, further detected lost RTP packets are ignored
   for a full smoothed round trip time, the intention of this is to
   limit the congestion window decrease to at most once per round trip.
   The congestion window backoff due to loss events is deliberately a
   bit less than is the case with e.g.  TCP Reno.  The reason is that
   TCP is generally used to transmit whole files, which can be
   translated to an infinite source bitrate.  SCReAM on the other hand
   has a source which rate is limited to a value close to the available
   transmit rate and often below said value, the effect of this is that
   SCReAM has less opportunity to grab free capacity than a TCP based
   file transfer.  To compensate for this it is necessary to let SCReAM
   reduce the congestion window slightly less than what is the case with
   TCP when loss events occur.

   An ECN event is detected if the n_ECN counter in the feedback report
   has increased since the previous received feedback.  Once an ECN
   event is detected, the n_ECN counter is ignored for a full smoothed
   round trip time, the intention of this is to limit the congestion
   window decrease to at most once per round trip.  The congestion
   window backoff due to an ECN event is deliberately smaller than if a
   loss event occurs.  This is in line with the idea outlined in
   [Khademi_alternative_backoff_ECN] to enable ECN marking thresholds
   lower than the corresponding packet drop thresholds.

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

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     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)
         # No loss, so it is then an ECN mark
         cwnd = max(min_cwnd,cwnd*BETA_ECN)
       adjust_qdelay_target(qdelay) #compensating for competing flows

     # when no congestion event
     on acknowledgement(qdelay):
       adjust_qdelay_target(qdelay) #compensating for competing flows
       calculate_send_window(qdelay, qdelay_target)

   The methods are further described in detail below.  Congestion window update

   The congestion window update is based on qdelay, except for the
   occurrence of loss events (one or more lost RTP packets in one RTT),
   or ECN events, which was described earlier.

   Pseudo code for the update of the congestion window is found below.

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     # in fast increase ?
     if (in_fast_increase)
       if (qdelay_trend >= 0.2)
         # incipient congestion detected, exit fast increase
         in_fast_increase = false
         # no congestion yet, increase cwnd if it
         #  is sufficiently used
         # an additional slack of bytes_newly_acked is
         #  added to ensure that CWND growth occurs
         #  even when feedback is sparse
         if (bytes_in_flight*1.5+bytes_newly_acked > cwnd)
           cwnd = cwnd+bytes_newly_acked

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

     gain_t = GAIN
     # adjust congestion window
     cwnd_delta_t =
       gain_t * off_target_t * bytes_newly_acked * MSS / cwnd
     if (off_target_t > 0 && bytes_in_flight*1.25+bytes_newly_acked <= cwnd)
       # no cwnd increase if window is underutilized
       # an additional slack of bytes_newly_acked is
       #  added to ensure that CWND growth occurs
       #  even when feedback is sparse
       cwnd_delta_t = 0;

     # 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)

   CWND is updated differently depending on whether the congestion
   control is in fast increase state or not, as controlled by the
   variable in_fast_increase.

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   When in fast increase state, the congestion window is increased with
   the number of newly acknowledged bytes as long as the window is
   sufficiently used.  Sparse feedback can potentially limit congestion
   window growth, an additional slack is therefore added, given by the
   number of newly acked 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].  There are however a few differences.

   o  [RFC6817] specifies a constant GAIN, this specification however
      limits the gain when CWND is increased dependent on near
      congestion state and the relation to the last known max CWND

   o  [RFC6817] specifies that the CWND increase is limited by an
      additional function controlled by a constant ALLOWED_INCREASE.
      This additional limitation is removed in this specification.

   Further the CWND is limited by max_bytes_in_flight and min_cwnd.  The
   limitation of the congestion window by the maximum number of bytes in
   flight over the last 5 seconds (max_bytes_in_flight) avoids possible
   over-estimation of the throughput after for example, idle periods.
   An additional MAX_BYTES_IN_FLIGHT_HEAD_ROOM allows for a slack, to
   allow for a certain amount of media coder output rate variability.  Competing flows compensation

   It is likely that a flow using SCReAM algorithm will have to share
   congested bottlenecks with other flows that use a more aggressive
   congestion control algorithm.  SCReAM takes care of such situations
   by adjusting the qdelay_target.

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       qdelay_norm_t = qdelay / QDELAY_TARGET_LOW
       # 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
       oh_t = qdelay_norm_avg_t + sqrt(qdelay_norm_var_t)
       oh_t *= QDELAY_TARGET_LO
       if (loss_event_rate > 0.002)
         # Packet losses detected
         qdelay_target = 1.5*oh_t
         if (qdelay_norm_var_t < 0.2)
           # Reasonably safe to set target qdelay
           qdelay_target = oh_t
           # Check if target delay can be reduced, this helps to avoid
           #  that the target delay is locked to high values for ever
           if (oh_t < QDELAY_TARGET_LO)
             # Decrease target delay quickly as measured queueing
             #  delay is lower than target
             qdelay_target = max(qdelay_target*0.5,oh_t)
             # Decrease target delay slowly
             qdelay_target *= 0.9

       # Apply limits
       qdelay_target = min(QDELAY_TARGET_HI, qdelay_target)
       qdelay_target = max(QDELAY_TARGET_LO, qdelay_target)

   The qdelay_target is adjusted differently, depending on if
   qdelay_norm_var_t is above or below a given value.
   A low qdelay_norm_avg_t value indicates that the qdelay does not
   change rapidly.  It is desired avoid the case that the qdelay target
   is increased due to self-congestion, indicated by a changing qdelay
   and consequently an increased qdelay_norm_var_t.  Still it should be
   possible to increase the qdelay target if the qdelay continues to be
   high.  This is a simple function with a certain risk of both false
   positives and negatives but it manages competing FTP flows reasonably
   well at the same time as it has proven to avoid accidental increased
   qdelay target relatively well in simulated LTE test cases.  The
   algorithm can however accidentally increase the qdelay target and

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   cause self-inflicted congestion in certain cases, therefore it is
   recommended to turn off the algorithm if is unlikely that competing
   flows will occur over the same bottleneck.  Lost packets detection

   Lost packets detection is based on the received sequence number list.
   A reordering window should be applied to avoid that packet reordering
   triggers loss events.
   The reordering window is specified as a time unit, similar to the
   ideas behind RACK (Recent ACKnowledgement) [RACK].  The computation
   of the reordering window is made possible by means of a lost flag in
   the list of transmitted RTP packets.  This flag is set if the
   received sequence number list indicates that the given RTP packet is
   missing.  If a later feedback indicates that a previously lost marked
   packet was indeed received, then the reordering window is updated to
   reflect the reordering delay.  The reordering window is given by the
   difference in time between the event that the packet was marked as
   lost and the event that it was indicated as successfully received.
   Loss is detected if a given RTP packet is not acknowledged within a
   time window (indicated by the reordering window) after an RTP packet
   with higher sequence number was acknowledged.  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: The media frame size is always varying to a
      larger or smaller extent.  A strict rule as the one given above
      will have the effect that the media bitrate will have difficulties
      to increase as the congestion window puts a too hard restriction
      on the media frame size variation.  This can lead to occasional
      queuing of RTP packets in the RTP packet queue that will further
      prevent bitrate increase.

   o  Reverse (feedback) path congestion: Especially in transport over
      buffer-bloated networks, the one way delay in the reverse
      direction may jump due to congestion.  The effect of this is that
      the acknowledgements are delayed with the result that the self-
      clocking is temporarily halted, even though the forward path is
      not congested.

   The send window is adjusted depending on qdelay and its relation to
   the qdelay target and the relation between the congestion window and
   the number of bytes in flight.  A strict rule is applied when qdelay

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   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
   fast when no congestion is detected while still being able to give a
   stable behavior in congested situations.

   The send window is given by the relation between the adjusted
   congestion window and the amount of bytes in flight according to the
   pseudo code below.

  calculate_send_window(qdelay, qdelay_target)
     # send window is computed differently depending on congestion level
     if (qdelay <= qdelay_target)
       send_wnd = cwnd+MSS-bytes_in_flight
       send_wnd = cwnd-bytes_in_flight

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

   Fast increase can resume in order to speed up the bitrate increase in
   case congestion abates.  The condition to resume fast increase
   (in_fast_increase = true) is that qdelay_trend is less than

4.1.3.  Media rate control

   The media rate control algorithm is executed at regular intervals
   RATE_ADJUSTMENT_INTERVAL, with the exception of a prompt reaction to
   loss events.  The media rate control operates based on the size of
   the RTP packet send queue and observed loss events.  In addition,
   qdelay_trend is also considered in the media rate control, this to
   reduce the amount of induced network jitter.

   The role of the media rate control is to strike a reasonable balance
   between a low amount of queuing in the RTP queue(s) and a sufficient
   amount of data to send in order to keep the data path busy.  A too
   cautious setting leads to possible under-utilization of network
   capacity and that the flow is starved out by other, more
   opportunistic traffic, on the other hand a too aggressive setting
   leads to extra jitter.

   The target_bitrate is adjusted depending on the congestion state.
   The target bitrate can vary between a minimum value

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   (TARGET_BITRATE_MIN) and a maximum value (TARGET_BITRATE_MAX).
   TARGET_BITRATE_MIN should be chosen to a low enough value to avoid
   that RTP packets are queued up when the network throughput becomes
   low.  The sender should also be equipped with a mechanism that
   discards RTP packets in cases 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 time

   Both estimates are updated every 200ms.

   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 may itself cause an unnecessary rate reduction.  To overcome
   this shortcoming; rate_ack is used as well.  This gives a more stable
   throughput estimate.

   The rate change behavior depends on whether a loss or ECN event has
   occurred and if the congestion control is in fast increase or not.

   # The target_bitrate is updated at a regular interval according

   on loss:
      # Loss event detected
      target_bitrate = max(BETA_R* target_bitrate, TARGET_BITRATE_MIN)
   on ecn_mark:
      # ECN event detected
      target_bitrate = max(BETA_ECN* target_bitrate, TARGET_BITRATE_MIN)

   ramp_up_speed_t = min(RAMP_UP_SPEED, target_bitrate/2.0)
   scale_t = (target_bitrate - target_bitrate_last_max)/
   scale_t = max(0.2, min(1.0, (scale_t*4)^2))
   # min scale_t value 0.2 as the bitrate should be allowed to
   #  increase at least slowly --> avoid locking the rate to
   #  target_bitrate_last_max

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   if (in_fast_increase = true)
      increment_t = ramp_up_speed_t*RATE_ADJUST_INTERVAL
      increment_t *= scale_t
      target_bitrate += increment_t
      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 =
      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 > 0.02)
        target_bitrate *= 0.95

   rate_media_limit_t = max(current_rate_t, max(rate_media,rtp_rate_median))
   rate_media_limit_t *= (2.0-1.0*qdelay_trend_mem)
   target_bitrate = min(target_bitrate, rate_media_limit_t)
   target_bitrate = min(TARGET_BITRATE_MAX,

   In case of a loss event the target_bitrate is updated and the rate
   change procedure is exited.  Otherwise the rate change procedure
   continues.  The rationale behind the rate reduction due to loss is
   that a congestion window reduction will take effect, a rate reduction
   pro actively avoids that RTP packets are 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.  A too short
   interval has shown to make the video coder internal rate control loop
   more unstable, a too long interval makes the overall congestion
   control sluggish.

   When in fast increase state (in_fast_increase=true), the bitrate
   increase is given by the desired ramp-up speed (RAMP_UP_SPEED) . The

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   ramp-up speed is limited when the target bitrate is low to avoid rate
   oscillation at low bottleneck bitrates.  The setting of RAMP_UP_SPEED
   depends on preferences, a high setting such as 1000kbps/s makes it
   possible to quickly get high quality media, this is however at the
   expense of a higher risk of jitter, which can manifest itself as e.g.
   choppy video rendering.

   When in_fast_increase is false, the bitrate increase is given by the
   current bitrate and is also controlled by the estimated RTP queue and
   the qdelay trend, thus it is sufficient that an increased congestion
   level is sensed by the network congestion control to limit the
   bitrate.  The target_bitrate_last_max is updated when congestion is

   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
   and max values.

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

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   It is not strictly necessary to make a 100% perfect compensation for
   the overhead as the SCReAM algorithm will inherently compensate
   moderate errors.  Under-compensation of the overhead has the effect
   that the jitter will increase somewhat while overcompensation will
   have the effect that the bottleneck link becomes under-utilized.

4.2.  SCReAM Receiver

   The simple task of the SCReAM receiver is to feedback
   acknowledgements of received packets and total ECN count to the
   SCReAM sender, in addition, the receive time of the RTP packet with
   the highest sequence number is echoed back.  Upon reception of each
   RTP packet the receiver will simply maintain enough information to
   send the aforementioned values to the SCReAM sender via RTCP
   transport layer feedback message.  The frequency of the feedback
   message depends on the available RTCP bandwidth.  More details of the
   feedback and the frequency is found in Appendix A.4.

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].  Section A.2 in said
      RFC however describes ways to mitigate issues with clock drift.

6.  Implementation status

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

   This section records the status of known implementations of the
   protocol defined by this specification at the time of posting of this
   Internet-Draft, and is based on a proposal described in [RFC6982].
   The description of implementations in this section is intended to
   assist the IETF in its decision processes in progressing drafts to
   RFCs.  Please note that the listing of any individual implementation
   here does not imply endorsement by the IETF.  Furthermore, no effort
   has been spent to verify the information presented here that was
   supplied by IETF contributors.  This is not intended as, and must not
   be construed to be, a catalog of available implementations or their
   features.  Readers are advised to note that other implementations may

   According to [RFC6982], "this will allow reviewers and working groups
   to assign due consideration to documents that have the benefit of
   running code, which may serve as evidence of valuable experimentation
   and feedback that have made the implemented protocols more mature.

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   It is up to the individual working groups to use this information as
   they see it".

6.1.  OpenWebRTC

   The SCReAM algorithm has been implemented in the OpenWebRTC project
   [OpenWebRTC], an open source WebRTC implementation from Ericsson
   Research.  This SCReAM implementation is usable with any WebRTC
   endpoint using OpenWebRTC.

   o  Organization : Ericsson Research, Ericsson.

   o  Name : OpenWebRTC gst plug-in.

   o  Implementation link : The GStreamer plug-in code for SCReAM can be
      found at github repository [SCReAM-Implementation] The wiki
      (https://github.com/EricssonResearch/openwebrtc/wiki) contains
      required information for building and using OpenWebRTC.

   o  Coverage : The code implements [I-D.ietf-rmcat-scream-cc].  The
      current implementation has been tuned and tested to adapt a video
      stream and does not adapt the audio streams.

   o  Implementation experience : The implementation of the algorithm in
      the OpenWebRTC has given great insight into the algorithm itself
      and its interaction with other involved modules such as encoder,
      RTP queue etc.  In fact it proves the usability of a self-clocked
      rate adaptation algorithm in the real WebRTC system.  The
      implementation experience has led to various algorithm
      improvements both in terms of stability and design.  The current
      implementation use an n_loss counter for lost packets indication,
      this is subject to change in later versions to a list of received
      RTP packets.

   o  Contact : irc://chat.freenode.net/openwebrtc

6.2.  A C++ Implementation of SCReAM

   o  Organization : Ericsson Research, Ericsson.

   o  Name : SCReAM.

   o  Implementation link : A C++ implementation of SCReAM is also
      available [SCReAM-Cplusplus_Implementation].  The code includes
      full support for congestion control, rate control and multi stream
      handling, it can be integrated in web clients given the addition
      of extra code to implement the RTCP feedback and RTP queue(s).

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      The code also includes a rudimentary implementation of a simulator
      that allows for some initial experiments.

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

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

7.  Suggested experiments

   SCReAM has been evaluated in a number of different ways, most of the
   evaluation has been in simulator.  The OpenWebRTC implementation work
   involved extensive testing with artificial bottlenecks with varying
   bandwidths and using two different video coders (OpenH264 and VP9),
   the experience of this lead to further improvements of the media rate
   control logic.

   Further experiments are preferably done by means of implementation in
   real clients and web browsers.  Recommended experiments are:

   o  Trials with various access technologies: EDGE/3G/4G, WiFi, DSL.

   o  Trials with different kinds of media: Audio, Video, slide show
      content.  Evaluation of multi stream handling in SCReAM.

   o  Evaluation of functionality of competing flows compensation
      mechanism: Evaluate how SCReAM performs with competing TCP like
      traffic and to what extent the competing flows compensation causes
      self-inflicted congestion.

   o  Determine proper parameters: A set of default parameters are given
      that makes SCReAM work over a reasonably large operation range,
      however for instance for very low or very high bitrates it may be
      necessary to use different values for instance for RAMP_UP_SPEED.

8.  Acknowledgements

   We would like to thank the following persons for their comments,
   questions and support during the work that led to this memo: Markus
   Andersson, Bo Burman, Tomas Frankkila, Frederic Gabin, Laurits Hamm,
   Hans Hannu, Nikolas Hermanns, Stefan Haakansson, Erlendur Karlsson,
   Daniel Lindstroem, Mats Nordberg, Jonathan Samuelsson, Rickard
   Sjoeberg, Robert Swain, Magnus Westerlund, Stefan Aalund.  Many
   additional thanks to RMCAT chairs Karen and Mirja for patiently
   reading, suggesting improvements and also for asking all the
   difficult but necessary questions.  Thanks to Stefan Holmer and
   Xiaoqing Zhu for the review.  Thanks to Ralf Globisch for taking time
   to try out SCReAM in his challenging low bitrate use cases.

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9.  IANA Considerations

   A new RFC4585 transport layer feedback message needs to be

10.  Security Considerations

   The feedback can be vulnerable to attacks similar to those that can
   affect TCP.  It is therefore recommended that the RTCP feedback is at
   least integrity protected.  Furthermore, as SCReAM is self-clocked, a
   malicious middlebox can drop RTCP feedback packets and thus cause the
   self-clocking in SCReAM to stall.

11.  Change history

   A list of changes:

   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-03 to WG-04: Editorial fixes

   o  WG-02 to WG-03: Review comments from Stefan Holmer and Xiaoqing
      Zhu addressed, owd changed to qdelay for clarity.  Added appendix
      section with RTCP feedback requirements, including a suggested
      basic feedback format based Loss RLE report block and the Packet
      Receipt Times blocks in [RFC3611].  Loss detection added as a
      section.  Transmission scheduling and packet pacing explained in
      appendix.  Source quench semantics added to appendix.

   o  WG-01 to WG-02: Complete restructuring of the document.  Moved
      feedback message to a separate draft.

   o  WG-00 to WG-01 : Changed the Source code section to Implementation
      status section.

   o  -05 to WG-00 : First version of WG doc, moved additional features
      section to Appendix.  Added description of prioritization in
      SCReAM.  Added description of additional cap on target bitrate

   o  -04 to -05 : ACK vector is replaced by a loss counter, PT is
      removed from feedback, references to source code added

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

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   o  -02 to -03 : Added algorithm description with equations, removed
      pseudo code and simulation results

   o  -01 to -02 : Updated GCC simulation results

   o  -00 to -01 : Fixed a few bugs in example code

12.  References

12.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,

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

   [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., 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,

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

   [RFC6298]  Paxson, V., Allman, M., Chu, J., and M. Sargent,
              "Computing TCP's Retransmission Timer", RFC 6298,
              DOI 10.17487/RFC6298, June 2011,

   [RFC6817]  Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind,
              "Low Extra Delay Background Transport (LEDBAT)", RFC 6817,
              DOI 10.17487/RFC6817, December 2012,

12.2.  Informative References

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

              Zanaty, M., Singh, V., Nandakumar, S., and Z. Sarker,
              "Congestion Control and Codec interactions in RTP
              Applications", draft-ietf-rmcat-cc-codec-interactions-02
              (work in progress), March 2016.

              Islam, S., Welzl, M., and S. Gjessing, "Coupled congestion
              control for RTP media", draft-ietf-rmcat-coupled-cc-03
              (work in progress), July 2016.

              Johansson, I. and Z. Sarker, "Self-Clocked Rate Adaptation
              for Multimedia", draft-ietf-rmcat-scream-cc-05 (work in
              progress), June 2016.

              Sarker, Z., Johansson, I., Zhu, X., Fu, J., Tan, W., and
              D. Ramalho, "Evaluation Test Cases for Interactive Real-
              Time Media over Wireless Networks", draft-ietf-rmcat-
              wireless-tests-02 (work in progress), May 2016.

              "TCP Alternative Backoff with ECN (ABE)",

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

              "Congestion Avoidance and Control", 1988.

              TS 23.203, 3GPP., "Policy and charging control
              architecture", June 2011, <http://www.3gpp.org/ftp/specs/

   [RACK]     "RACK: a time-based fast loss detection algorithm for
              TCP", <https://http://tools.ietf.org/id/

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   [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,

   [RFC6679]  Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
              and K. Carlberg, "Explicit Congestion Notification (ECN)
              for RTP over UDP", RFC 6679, DOI 10.17487/RFC6679, August
              2012, <http://www.rfc-editor.org/info/rfc6679>.

   [RFC6982]  Sheffer, Y. and A. Farrel, "Improving Awareness of Running
              Code: The Implementation Status Section", RFC 6982,
              DOI 10.17487/RFC6982, July 2013,

   [RFC7661]  Fairhurst, G., Sathiaseelan, A., and R. Secchi, "Updating
              TCP to Support Rate-Limited Traffic", RFC 7661,
              DOI 10.17487/RFC7661, October 2015,

              "C++ Implementation of SCReAM",

              "SCReAM Implementation",

   [TFWC]     University College London, "Fairer TCP-Friendly Congestion
              Control Protocol for Multimedia Streaming", December 2007,

Appendix A.  Additional information

A.1.  Stream prioritization

   The SCReAM algorithm makes a good distinction between network
   congestion control and the media rate control, an RTP queue queues up
   RTP packets pending transmission.  This is easily extended to many
   streams, in which case RTP packets from two or more RTP queues are
   scheduled at the rate permitted by the network congestion control.

   The scheduling can be done by means of a few different scheduling
   regimes.  For example the method applied in
   [I-D.ietf-rmcat-coupled-cc] can be used.  The implementation of
   SCReAM use something that is referred to as credit based scheduling.

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   Credit based scheduling is for instance implemented in IEEE 802.17.
   The short description is that credit is accumulated by queues as they
   wait for service and are spent while the queues are being services.

   For instance, if one queue is allowed to transmit 1000bytes, then a
   credit of 1000bytes is allocated to the other unscheduled queues.
   This principle can be extended to weighted scheduling in which case
   the credit allocated to unscheduled queues depends on the weight

A.2.  Computation of autocorrelation function

   The autocorrelation function is computed over a vector of values.

   Let x be a vector constituting N values, the biased autocorrelation
   function for a given lag=k for the vector x is given by .

      R(x,k) = SUM x(n)*x(n+k)

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 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 enforced to equal or higher
   than t_pace where t_pace is given by the equations below :

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

   rtp_size is the size of the last transmitted RTP packet, s_rtt is the
   smoothed round trip time.

A.4.  RTCP feedback considerations

   This section describes the requirements on the RTCP feedback to make
   SCReAM function well.  Parts of this section may be moved to a
   separate draft.  First is described the requirements on the feedback
   elements, second is described the requirements on the feedback

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   intensity to keep SCReAM self-clocking and rate control loops
   function properly.

A.4.1.  Requirements on feedback elements

   SCReAM requires the following elements for its basic functionality,
   i.e only including features that are 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 queueing delay.  This can be realized by means of the Packet
      Receipt Times Report Block in [RFC3611]. begin_seq should be set
      to the highest received (possibly wrapped around) sequence number,
      end_seq should be set to begin_seq+1 % 65536.  The timestamp clock
      may be set according to the specification i.e equal to the RTP
      timestamp clock.  Detailed individual packet receive times is not
      necessary as SCReAM does currently not describe how this can be

   The basic feedback needed for SCReAM involves the use of the Loss RLE
   report block and the Packet Receipt Times block defined in Figure 2.

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        0                   1                   2                   3
        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
       |V=2|P|reserved |   PT=XR=207   |             length            |
       |                              SSRC                             |
       |     BT=2      | rsvd. |  T=0  |         block length          |
       |                        SSRC of source                         |
       |          begin_seq            |             end_seq           |
       |          chunk 1              |             chunk 2           |
       :                              ...                              :
       |          chunk n-1            |             chunk n           |
       |     BT=3      | rsvd. |  T=0  |         block length          |
       |                        SSRC of source                         |
       |          begin_seq            |             end_seq           |
       |       Receipt time of packet begin_seq                        |

       Figure 2: Basic feedback message for SCReAM, based on RFC3611

   In a typical use case, no more than four Loss RLE chunks 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 ECE-CE code
      point set.  The more bulky format may be nevertheless be useful
      for e.g ECN black-hole detection.

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   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 200ms, while at high
   bitrates a feedback interval of ~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(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(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

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   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

   Ingemar Johansson
   Ericsson AB
   Laboratoriegraend 11
   Luleaa  977 53

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

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   Zaheduzzaman Sarker
   Ericsson AB
   Laboratoriegraend 11
   Luleaa  977 53

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

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