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RFC 8298


Self-Clocked Rate Adaptation for Multimedia

Part 2 of 2, p. 14 to 36
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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

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   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

   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.

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       qdelay_fraction_t = qdelay / qdelay_target
       # Calculate moving average
       qdelay_fraction_avg = (1 - QDELAY_WEIGHT) * qdelay_fraction_avg +
          QDELAY_WEIGHT * 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.

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

   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

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   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.  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  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.  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.

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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
       calculate_send_window(qdelay, qdelay_target)

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

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     # 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
         # 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

     # 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;

     # Apply delta
     cwnd += cwnd_delta_t
     # limit cwnd to the maximum number of bytes in flight
     cwnd = min(cwnd, max_bytes_in_flight *
     cwnd = max(cwnd, MIN_CWND)


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   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

   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.  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.

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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
       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
         if (qdelay_norm_var_t < 0.2)
           # Reasonably safe to set target qdelay
           qdelay_target = new_target_t
           # 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)
             # 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)
     <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

   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

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   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.  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

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   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.  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.

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   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.  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 :

      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.  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.  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

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   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

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

   Both estimates are updated every 200 ms.

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   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.

   # The target_bitrate is updated at a regular interval according

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

   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 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
      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)

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      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

   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))

   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

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   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.

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   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

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      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].

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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

   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).

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   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

   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

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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,

   [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, <>.

   [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,

   [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, <>.

   [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,

   [RFC8174]  Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
              2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
              May 2017, <>.

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9.2.  Informative References

              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.

              Islam, S., Welzl, M., and S. Gjessing, "Coupled congestion
              control for RTP media", Work in Progress, draft-ietf-
              rmcat-coupled-cc-07, September 2017.

              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,

              Ericsson Research, "OpenWebRTC",

              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,

   [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, <>.

   [RFC7478]  Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
              Time Communication Use Cases and Requirements", RFC 7478,
              DOI 10.17487/RFC7478, March 2015,

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

              Ericsson Research, "SCReAM - Mobile optimised congestion
              control algorithm",

              Ericsson Research, "OpenWebRTC specific GStreamer
              plugins", <

              Sarker, Z. and I. Johansson, "Updates on SCReAM: An
              implementation experience", November 2015,

   [TFWC]     Choi, S. and M. Handley, "Fairer TCP-Friendly Congestion
              Control Protocol for Multimedia Streaming Applications",
              DOI 10.1145/1364654.1364717, December 2007,

              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.

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   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

   Phone: +46 730783289

   Zaheduzzaman Sarker
   Ericsson AB
   Laboratoriegraend 11
   Luleaa  977 53

   Phone: +46 761153743