Internet-Draft SCReAMv2 March 2024
Johansson & Westerlund Expires 5 September 2024 [Page]
8298 (if approved)
Intended Status:
I. Johansson
M. Westerlund

Self-Clocked Rate Adaptation for Multimedia


This memo describes a rate adaptation algorithm for conversational media services such as interactive video. The solution conforms to the packet conservation principle and is a hybrid loss- and delay based congestion control/rate management algorithm that also supports ECN and L4S. The algorithm is evaluated over both simulated Internet bottleneck scenarios as well as in a LongTerm Evolution (LTE) and 5G system simulator and is shown to achieve both low latency and high video throughput in these scenarios. This specification obsoletes RFC 8298. The algorithm supports handling of multiple media streams, typical use cases are streaming for remote control, AR and 3D VR googles.

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Table of Contents

1. Introduction

Congestion in the Internet occurs when the transmitted bitrate is higher than the available capacity over a given transmission path. Applications that are deployed in the Internet have to employ congestion control to achieve robust performance and to avoid congestion collapse in the Internet.

Interactive real-time communication imposes a lot of requirements on the transport; therefore, a robust, efficient rate adaptation for all access types is an important part of interactive real-time communications, as the transmission channel bandwidth can vary over time.

Wireless access such as LTE and 5G, which is an integral part of the current Internet, increases the importance of rate adaptation as the channel bandwidth of a default LTE bearer [QoS-3GPP] can change considerably in a very short time frame. Thus, a rate adaptation solution for interactive real-time media, such as WebRTC [RFC7478], should be both quick and be able to operate over a large range in channel capacity.

This memo describes Self-Clocked Rate Adaptation for Multimedia version 2 (SCReAMv2), an update to SCReAM congestion control for RTP streams [RFC3550]. While SCReAM was originally devised for WebRTC, SCReAM as well as SCReAMv2 can also be used for other applications where congestion control of RTP streams is necessary. SCReAM is based on the self-clocking principle of TCP and uses techniques similar to what is used in the rate adaptation algorithm based on Low Extra Delay Background Transport (LEDBAT) [RFC6817].

SCReAMv2 is not entirely self-clocked as it augments self-clocking with pacing and a minimum send rate. Further, SCReAMv2 can take advantage of Explicit Congestion Notification (ECN) and Low Latency Low Loss and Scalable throughput (L4S) in cases where ECN or L4S is supported by the network and the hosts. However, ECN or L4S is not required for the basic congestion control functionality in SCReAMv2.

This specification replaces the previous experimental version [RFC8298] of SCReAM with SCReAMv2. There are many and fairly significant changes to the original SCReAM algorithm.

The algorithm in this memo differs greatly against the previous version of SCReAM. The main differences are:

1.1. Wireless (LTE and 5G) Access Properties

[RFC8869] describes the complications that can be observed in wireless environments. Wireless access such as LTE and 5G typically cannot guarantee a given bandwidth; this is true especially for default bearers. The network throughput can vary considerably, for instance, in cases where the wireless terminal is moving around. Even though 5G can support bitrates well above 100 Mbps, there are cases when the available bitrate can be much lower; examples are situations with high network load and poor coverage. An additional complication is that the network throughput can drop for short time intervals (e.g., at handover); these short glitches are initially very difficult to distinguish from more permanent reductions in throughput.

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

1.2. Why is it a self-clocked algorithm?

Self-clocked congestion control algorithms provide a benefit over their rate-based counterparts in that the former consists of two adaptation mechanisms:

  • A congestion window computation that evolves over a longer timescale (several RTTs) especially when the congestion window evolution is dictated by estimated delay (to minimize vulnerability to, e.g., short-term delay variations).

  • A fine-grained congestion control given by the self-clocking; it operates on a shorter time scale (1 RTT). The benefits of self-clocking are also elaborated upon in [TFWC]. The self-clocking however acts more like an emergency break as bytes in flight can exceed the congesion window only to a certain degree. The rationale is to be able to transmit large video frames and avoid that they are unnecessarily queued up on the sender side, but still prevent a bootleneck queue

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

1.3. Comparison with LEDBAT and TFWC in TCP

The core SCReAM algorithm, which is still similar in SCReAMv2, has similarities to the concepts of self-clocking used in TCP-friendly window-based congestion control [TFWC] and follows the packet conservation principle. The packet conservation principle is described as a key factor behind the protection of networks from congestion [Packet-conservation].

The congestion window is determined in a way similar to LEDBAT [RFC6817]. LEDBAT is a congestion control algorithm that uses send and receive timestamps to estimate the queuing delay (from now on denoted "qdelay") along the transmission path. This information is used to adjust the congestion window. The general problem described in the paper is that the base delay is offset by LEDBAT's own queue buildup. The big difference with using LEDBAT in the SCReAM context lies in the facts that the source is rate limited and that the RTP queue must be kept short (preferably empty). In addition, the output from a video encoder is rarely constant bitrate; static content (talking heads, for instance) gives almost zero video bitrate. This yields two useful properties when LEDBAT is used with SCReAM; they help to avoid the issues described in [LEDBAT-delay-impact]:

  1. There is always a certain probability that SCReAM is short of data to transmit; this means that the network queue will become empty every once in a while.

  2. The max video bitrate can be lower than the link capacity. If the max video bitrate is 5 Mbps and the capacity is 10 Mbps, then the network queue will become empty.

It is sufficient that any of the two conditions above is fulfilled to make the base delay update properly. Furthermore, [LEDBAT-delay-impact] describes an issue with short-lived competing flows. In SCReAM, these short-lived flows will cause the self-clocking to slow down, thereby building up the RTP queue; in turn, this results in a reduced media video bitrate. Thus, SCReAM slows the bitrate more when there are competing short-lived flows than the traditional use of LEDBAT does. The basic functionality in the use of LEDBAT in SCReAM is quite simple; however, there are a few steps in order to make the concept work with conversational media:

  • Addition of a media rate control function.

  • Congestion window validation techniques. The congestion window is used as a basis for the target bitrate calculation. For that reason, various actions are taken to avoid that the congestion window grows too much beyond the bytes in flight. Additional contraints are applied when in congested state and when the max target bitrate is reached.

  • Use of inflection points in the congestion window calculation to achieve reduced delay jitter (when L4S is not active).

  • Adjustment of qdelay target for better performance when competing with other loss-based congestion-controlled flows.

The above-mentioned features will be described in more detail in Section Section 4.

The SCReAM/SCReAMv2 congestion control method uses techniques similar to LEDBAT [RFC6817] to measure the qdelay. As is the case with LEDBAT, it is not necessary to use synchronized clocks in the sender and receiver in order to compute the qdelay. However, it is necessary that they use the same clock frequency, or that the clock frequency at the receiver can be inferred reliably by the sender. Failure to meet this requirement leads to malfunction in the SCReAM/SCReAMv2 congestion control algorithm due to incorrect estimation of the network queue delay. Use of [RFC8888] as feedback ensures that the same time base is used in sender and receiver.

2. Requirements Language

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

3. Overview of SCReAMv2 Algorithm

SCReAMv2 still consists of three main parts: network congestion control, sender transmission control, and media rate control. All of these parts reside at the sender side. Figure 1 shows the functional overview of a SCReAMv2 sender. The receiver-side algorithm is very simple in comparison, as it only generates feedback containing acknowledgements of received RTP packets and indication of ECN bits.

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 sender calculates the congestion window based on the feedback from the RTP receiver. The feedback is timestamp and ECN echo for individual RTP packets.

The receiver of the media echoes a list of received RTP packets and the timestamp of the RTP packet with the highest sequence number back to the sender in feedback packets. The sender keeps a list of transmitted packets, their respective sizes, and the time they were transmitted. This information is used to determine the number of bytes that can be transmitted at any given time instant. A congestion window puts an upper limit on how many bytes can be in flight, i.e., transmitted but not yet acknowledged. The congestion window is however not an absolute limit as a slack is given to efficiently transmit large video frames.

The congestion window seeks to increase by at least one segment per RTT and this increase regardless congestion occurs or not, the congestion window increase is restriced or relaxed based on the current value of the congestion window and the time elapsed since last congestion event. The congestion window update is increased by one MSS (maximum known RTP packet size) per RTT with some variation based on congestion window size and time elapsed since the last congestion event. Multiplicative increase allows the congestion to increase by a fraction of cwnd when congestion has not occured for a while. The congestion window is thus an adaptive multiplicative increase that is mainly additive increase when steady state is reached but allows a faster convergence to a higher link speed.

Congestion window reduction is triggered by:

  • Packet loss or Classic ECN marking is detected : The congestion window is reduced by a predetermined fraction.

  • Estimated queue delay exceeds a given threshold : The congestion window is reduced given by how much the delay exceeds the threshold.

  • L4S ECN marking detected : The congestion window is reduced in proportion to the fraction of packets that are marked (scalable congestion control).

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. The congestion window is however not a hard limit, additional slack is given to avoid that RTP packets are queued up unnecessarily on the sender side. This means that the algoritm prefers to build up a queue in the network rather than on the sender side. Additional congestion that this causes will reflect back and cause a reduction of the congestion window. Packet pacing is used to mitigate issues with ACK compression that MAY cause increased jitter and/or packet loss in the media traffic. Packet pacing limits the packet transmission rate given by the estimated link throughput. Even if the send window allows for the transmission of a number of packets, these packets are not transmitted immediately; rather, they are transmitted in intervals given by the packet size and the estimated link throughput. Packets are generally paced at a higher rate than the target bitrate, this makes it possible to transmit occasionally larger video frames in a timely manner.

3.3. Media Rate Control

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

The reaction to reduced throughput must be prompt in order to avoid getting too much data queued in the RTP packet queue(s) in the sender. The media rate is calculated based on the congestion window and RTT. For the case that multiple streams are enabled, the media rate among the streams is distrubuted according to relative priorities.

In cases where the sender's frame queues increase rapidly, such as in the case of a Radio Access Type (RAT) handover, the SCReAMv2 sender MAY implement additional actions, such as discarding of encoded media frames or frame skipping in order to ensure that the RTP queues are drained quickly. Frame skipping results in the frame rate being temporarily reduced. Which method to use is a design choice and is outside the scope of this algorithm description.

4. Detailed Description of SCReAMv2 sender algorithm

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

The sender implements media rate control and an RTP queue for each media type or source, where RTP packets containing encoded media frames are temporarily stored for transmission. Figure 1 shows the details when a single media source (or stream) is used. A transmission scheduler (not shown in the figure) is added to support multiple streams. The transmission scheduler can enforce differing priorities between the streams and act like a coupled congestion controller for multiple flows. Support for multiple streams is implemented in [SCReAM-CPP-implementation].

Media encoder |(1) RTP (3) | Media Queue rate control RTP packets (2) (4) RTP Network (7) Sender congestion Transmission control Control |(5) RTCP RTP (6) UDP socket
Figure 1: Sender Functional View

Media frames are encoded and forwarded to the RTP queue (1) in Figure 1. The RTP packets are picked from the RTP queue (4), for multiple flows from each RTP queue based on some defined priority order or simply in a round-robin fashion, by the sender transmission controller.

The sender transmission controller (in case of multiple flows a transmission scheduler) sends the RTP packets to the UDP socket (5). In the general case, all media SHOULD go through the sender transmission controller and is limited so that the number of bytes in flight is less than the congestion window albeit with a slack to avoid that packets are unnecessarily delayed in the RTP queue.

RTCP packets are received (6) and the information about the bytes in flight and congestion window is exchanged between the network congestion control and the sender transmission control (7).

The congestion window and the estimated RTT is communicated to the media rate control (2) to compute the appropriate target bitrate. The target bitrate is updated whenever the congestion window is updated. Additional parameters are also communicated to make the rate control more stable when the congestion window is very small or when L4S is not active. This is described more in detail below.

4.1. Constants and variables

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

4.1.1. Constants

The RECOMMENDED values, within parentheses "()", for the constants are deduced from experiments.

  • QDELAY_TARGET_LO (0.1): Target value for the minimum qdelay [s].

  • QDELAY_TARGET_HI (0.4): Target value for the maximum qdelay [s]. This parameter provides an upper limit to how much the target qdelay (qdelay_target) can be increased in order to cope with competing loss-based flows. However, the target qdelay does not have to be initialized to this high value, as it would increase end-to-end delay and also make the rate control and congestion control loops sluggish.

  • MIN_CWND (3000): Minimum congestion window [byte].

  • MAX_BYTES_IN_FLIGHT_HEAD_ROOM (1.1): Headroom for the limitation of CWND.

  • BETA_LOSS (0.7): CWND scale factor due to loss event.

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

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

  • TARGET_BITRATE_MIN: Minimum target bitrate in [bps] (bits per second).

  • TARGET_BITRATE_MAX: Maximum target bitrate in [bps].

  • RATE_PACE_MIN (50000): Minimum pacing rate in [bps].

  • CWND_OVERHEAD (1.5): Indicates how much bytes in flight is allowed to exceed cwnd.

  • L4S_AVG_G (1.0/16): EWMA factor for l4s_alpha

  • QDELAY_AVG_G (1.0/4): EWMA factor for qdelay_avg

  • POST_CONGESTION_DELAY (4.0): Determines how long (seconds) after a congestion event that the congestion window growth should be cautious.

  • MUL_INCREASE_FACTOR (0.02): Determines how much (as a fraction of cwnd) that the cwnd can increase per RTT.

  • LOW_CWND_SCALE_FACTOR (0.1): Scale factor applied to cwnd change when CWND is very small.

  • IS_L4S (false): Congestion control operates in L4S mode.

  • VIRTUAL_RTT (0.025): Virtual RTT [s]

  • PACKET_PACING_HEADROOM (1.5): Extra head room for packet pacing.

  • BYTES_IN_FLIGHT_HEAD_ROOM (2.0): Extra headroom for bytes in flight.

4.1.2. State Variables

The values within parentheses "()" indicate initial values.

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

  • qdelay_fraction_avg (0.0): Fractional qdelay filtered by the Exponentially Weighted Moving Average (EWMA).

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

  • cwnd (MIN_CWND): Congestion window.

  • cwnd_i (1): Congestion window inflection point.

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

  • max_bytes_in_flight (0): The maximum number of bytes in flight in the last round trip.

  • max_bytes_in_flight_prev (0): The maximum number of bytes in flight in previous round trip.

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

  • target_bitrate (0): Media target bitrate [bps].

  • rate_media (0.0): Measured bitrate [bps] from the media encoder.

  • s_rtt (0.0): Smoothed RTT [s], computed with a similar method to that described in [RFC6298].

  • rtp_size (0): Size [byte] of the last transmitted RTP packet.

  • loss_event_rate (0.0): The estimated fraction of RTTs with lost packets detected.

  • bytes_in_flight_ratio (0.0): Ratio between the bytes in flight and the congestion window.

  • cwnd_ratio (0.0): Ratio between MSS and cwnd.

  • l4s_alpha (0.0): Average fraction of marked packets per RTT.

  • l4s_active (false): Indicates that L4S is enabled and packets are indeed marked.

  • last_update_l4s_alpha_time (0): Last time l4s_alpha was updated [s].

  • last_update_qdelay_avg_time (0): Last time qdelay_avg was updated [s].

  • packets_delivered_this_rtt (0): Counter for delivered packets.

  • packets_marked_this_rtt (0): Counter delivered and ECN-CE marked packets.

  • last_congestion_detected_time (0): Last time congestion event occured [s].

  • last_cwnd_i_update_time (0): Last time cwnd_i was updated [s].

  • bytes_newly_acked (0): Number of bytes newly ACKed, reset to 0 when congestion window is updated [byte].

  • bytes_newly_acked_ce (0): Number of bytes newly ACKed and CE marked, reset to 0 when congestion window is updated [byte].

  • pace_bitrate (1e6): Packet pacing rate [bps].

  • t_pace (1e-6): Pacing interval between packets [s].

  • rel_framesize_high (1.0): High percentile of frame size, normalized by nominal frame size for the given target bitrate

  • frame_period (0.02): The frame period [s].

4.2. Network Congestion Control

This section explains the network congestion control, which performs two main functions:

  • Computation of congestion window at the sender: This gives an upper limit to the number of bytes in flight.

  • 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.

SCReAMv2 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 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.

bytes_in_flight_ratio is calculated as the ratio between bytes_flight and cwnd. This value should be computed at the beginning of the ACK processing. cwnd_ratio is computed as the relation between MSS and cwnd.

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_ce is, similar to bytes_newly_acked, a counter of bytes newly acked with the extra condition that they are ECN-CE marked. The bytes_newly_acked and bytes_newly_acked_ce are reset to zero after a CWND update.

The feedback from the receiver is assumed to consist of the following elements.

  • A list of received RTP packets' sequence numbers. With an indication if packets are ECN-CE marked.

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

It is recommended to use RFC8888 [RFC8888] for the feedback as it supports the feedback elements described above.

When the sender receives RTCP feedback, the qdelay is calculated as outlined in [RFC6817]. A qdelay sample is obtained for each received acknowledgement. A number of variables are updated as illustrated by the pseudocode below; temporary variables are appended with '_t'. Division operation is always floating point unless otherwise noted. l4s_alpha is calculated based in number of packets delivered (and marked). This makes calculation of L4S alpha more accurate at very low bitrates, given that the tail RTP packet in a video frame is often smaller than MSS.

    packets_delivered_this_rtt += packets_acked
    packets_marked_this_rtt += packets_acked_ce
    if (now - last_update_l4s_alpha_time >= s_rtt)
      # l4s_alpha is calculated from packets marked istf bytes marked
      fraction_marked_t = packets_marked_this_rtt/
      l4s_alpha = L4S_AVG_G*fraction_marked_t + (1.0-L4S_AVG_G)*l4S_alpha

      last_update_l4s_alpha_time = now
      packets_delivered_this_rtt = 0
      packets_marked_this_rtt = 0

    if (now - last_update_qdelay_avg_time >= s_rtt)
      # qdelay_avg is updated with a slow attack, fast decay EWMA filter
      if (qdelay < qdelay_avg)
        qdelay_avg = qdelay
        qdelay_avg = QDELAY_AVG_G*qdelay + (1.0-QDELAY_AVG_G)*qdelay_avg
      last_update_qdelay_avg_time = now

4.2.1. Reaction to Delay, Packet Loss and ECN-CE

Congestion is detected based on three different indicators:

  • Lost packets detected. The loss detection is described in Section

  • ECN-CE marked packets detected.

  • Estimated queue delay exceeds a threshold.

A congestion event occurs if any of the above indicators are true AND it is than one smoothed RTT (s_rtt) since the last congestion event. This ensures that the congestion window is reduced at most once per smoothed RTT. Lost packets

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. SCReAMv2, on the other hand, has a source which rate is limited to a value close to the available transmit rate and often below that value; the effect is that SCReAMv2 has less opportunity to grab free capacity than a TCP-based file transfer. To compensate for this, it is RECOMMENDED to let SCReAMv2 reduce the congestion window less than what is the case with TCP when loss events occur. ECN-CE and classic ECN

In classic ECN mode the cwnd is scaled by a fixed value (BETA_ECN).

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 [RFC8511] to enable ECN marking thresholds lower than the corresponding packet drop thresholds. ECN-CE and L4S

The cwnd is scaled down in proportion to the fraction of marked packets per RTT. The scale down proportion is given by l4s_alpha, which is an EWMA filtered version of the fraction of marked packets per RTT. This is inline with how DCTCP works [RFC8257]. Additional methods are applied to make the congestion window reduction reasonably stable, especially when the congestion window is only a few MSS. In addition, because SCReAMv2 can quite often be source limited, additional steps are taken to restore the congestion window to a proper value after a long period without congestion. Increased queue delay

SCReAMv2 implements a delay based congestion control approach where it mimics L4S congestion marking when the averaged queue delay exceeds a target threshold. This threshold is set to qdelay_target/2 and the congestion backoff factor (l4s_alpha_v) increases linearly from 0 to 100% as qdelay_avg goes from qdelay_target/2 to qdelay_target. The averaged qdelay (qdelay_avg) is used to avoid that the SCReAMv2 congestion control over-reacts to scheduling jitter, sudden delay spikes due to e.g. handover or link layer retransmissions. Furthermore, the delay based congestion control is inactivated when it is reasonably certain that L4S is active, i.e. L4S is enabled and congested nodes apply L4S marking of packets. The rationale is reduce negative effect of clockdrift that the delay based control can introduce whenever possible.

4.2.2. Congestion Window Update

The congestion window update contains two parts. One that reduces the congestion window when congestion events (listed above) occur, and one part that continously increases the congestion window.

The target bitrate is updated whenever the congestion window is updated.

Actions when congestion detected

      if (now - last_congestion_detected_time >= s_rtt)
        if (loss detected)
          is_loss_t = true
        if (packets marked)
          is_ce_t = true
        if (qdelay > qdelay_target/2)
          # It is expected that l4s_alpha is below a given value,
          l4_alpha_lim_t = 2 / target_bitrate * MSS * 8 / s_rtt
          if (l4s_alpha < l4_alpha_lim_t || !l4s_active)
            # L4S does not seem to be active
            l4s_alpha_v_t = min(1.0, max(0.0,
               (qdelay_avg - qdelay_target / 2) /
               (qdelay_target / 2)));
            is_virtual_ce_t = true

      if (is_loss_t || is_ce_t || is_virtual_ce_t)
        if (now - last_cwnd_i_update_time > 0.25)
          last_cwnd_i_update_time = now
          cwnd_i = cwnd

      # Scale factor for cwnd update
      cwnd_scale_factor_t =

      # Either loss, ECN mark or increased qdelay is detected
      if (is_loss_t)
        # Loss is detected
        cwnd = cwnd * BETA_LOSS
      if (is_ce_t)
        # ECN-CE detected
        if (IS_L4S)
          # L4S mode
          backoff_t = l4s_alpha / 2
          # Increase stability for very small cwnd
          backOff_t *= min(1.0, cwnd_scale_factor_t)
          backOff_t *= max(0.8, 1.0 - cwnd_ratio * 2)

          if (now - last_congestion_detected_time > 5)
            # A long time since last congested because link throughput
            # exceeds max video bitrate.
            # There is a certain risk that CWND has increased way above
            # bytes in flight, so we reduce it here to get it better on
            # track and thus the congestion episode is shortened
            cwnd = min(cwnd, max_bytes_in_flight_prev)

            # Also, we back off a little extra if needed
            # because alpha is quite likely very low
            # This can in some cases be an over-reaction
            # but as this function should kick in relatively seldom
            # it should not be to too big concern
            backoff_t = max(backoff_t, 0.25)

            # In addition, bump up l4sAlpha to a more credible value
            # This may over react but it is better than
            # excessive queue delay
            l4sAlpha = 0.25
          cwnd = (1.0 - backoff_t) * cwnd
          # Classic ECN mode
          cwnd = cwnd * BETA_ECN
      if (is_virtual_ce_t)
        backoff_t = l4s_alpha_v_t / 2
        cwnd = (1.0 - backoff_t) * cwnd
      cwnd = max(MIN_CWND, cwnd)

      if (is_loss_t || is_ce_t || is_virtual_ce_t)
        last_congestion_detected_time = now

The variable max_bytes_in_flight_prev indicates the maximum bytes in flights in the previous round trip. The reason to this is that bytes in flight can spike when congestion occurs, max_bytes_in_flight_prev thus ensures better that an uncongested bytes in flight is used.

The cwnd_scale_factor_t scales the congestion window decrease upon congestion as well as the increase. In a normal addititive increase setting this would be 1.0/MSS. However to increase stability especially when with L4S when cwnd is very small, the cwnd_scale_factor_t can be as small as low as LOW_CWND_SCALE_FACTOR / MSS. The result is then that the congestion window increase can be as small as LOW_CWND_SCALE_FACTOR MSS per RTT. Because the same restriction is applied to both decrease and increase of the congestion window, the net effect is zero. The cwnd_scale_factor_t is increased with larger cwnd to allow for a multiplicative increase and thus a faster convergence when link capacity increases.

Congestion window increase

      # Additional factor for cwnd update
      post_congestion_scale_t = max(0.0, min(1.0,
        (now - last_congestion_detected_time) / POST_CONGESTION_DELAY))

      # Calculate bytes acked that are not CE marked
      bytes_newly_acked_minus_ce_t = bytes_newly_acked-

      increment_t = bytes_newly_acked_minus_ce_t*cwnd_ratio

      # Reduce increment for small RTTs
      tmp_t = min(1.0, s_rtt / VIRTUAL_RTT)
      increment *= tmp_t * tmp_t

      if (!is_l4s_active)
        # The increment is scaled down for more cautious
        # ramp-up around the last known congestion window
        # when congestion last occured.
        # This is only applied when L4S is inactive
        scl_t = 1.0
        scl_t = (cwnd - cwnd_i) / cwnd_i
        scl_t *= 4
        scl_t = scl_t * scl_t
        scl_t = max(0.1, min(1.0, scl_t))
        increment_t *= scl_t

      # Slow down CWND increase when CWND is only a few MSS
      # This goes hand in hand with that the down scaling is also
      # slowed down then. Cwnd increase can be as slow as
      float tmp_t = cwnd_scale_factor_t

      # Further limit multiplicative increase when congestion occured
      # recently.
      if (tmp_t > 1.0)
        tmp_t = 1.0 + ((tmp_t - 1.0) * post_congestion_scale_t);
      increment *= tmp_t

      # Increase CWND only if bytes in flight is large enough
      # Quite a lot of slack is allowed here to avoid that bitrate
      # locks to low values.
      max_allowed_t = MSS + max(max_bytes_in_flight,
        max_bytes_in_flight_prev) * BYTES_IN_FLIGHT_HEAD_ROOM
      int cwnd_t = cwnd + increment_t
      if (cwnd_t <= max_allowed_t)
        cwnd = cwnd_t

The variable max_bytes_in_flight indicates the max bytes in flight in the current round trip.

The multiplicative increase is restricted directly after a congestion event and the restriction is gradually relaxed as the time since last congested increased. The restriction makes the congestion window growth to be no faster than additive increase when congestion continusly occurs. For L4S operation this means that the SCReAMv2 algorithm will adhere to the 2 marked packets per RTT equilibrium at steady state congestion.

It is particularly important that the congestion window reflects the transmitted bitrate especially in L4S mode operation. An inflated cwnd takes extra RTTs to bring down to a correct value upon congestion and thus causes unnecessary queue buildup. At the same time the congestion window must be allowed to be large enough to avoid that the SCReAMv2 algorithm begins to limit itself, given that the target bitrate is calculated based on the cwnd. Two mechanisms are used to manage this:

  • Restore correct value of cwnd upon congestion. This is done if is a prolonged time since the link was congested. A typical example is that SCReAMv2 has been rate limited, i.e the target bitrate has reached the TARGET_BITRATE_MAX.

  • Limit cwnd when the target_bitrate has reached TARGET_BITRATE_MAX. The cwnd is restricted based on a history of the last max_bytes_in_flight values. See [SCReAM-CPP-implementation] for details.

The two mechanisms complement one another.

4.2.3. 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) [RFC8985]. 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 event that the packet was marked as lost and the event that it was indicated as successfully received. Loss is detected if a given RTP packet is not acknowledged within a time window (indicated by the reordering window) after an RTP packet with a higher sequence number was acknowledged.

4.2.4. Send Window Calculation

The basic design principle behind packet transmission in SCReAM was 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:

  • 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. 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. 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.

  • 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 CWND_OVERHEAD allows for some degree of reverse path congestion as the bytes in flight is allowed to exceed cwnd.

In SCReAMv2, 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. The multiplication of cwnd with CWND_OVERHEAD and rel_framesize_high has the effect that bytes in flight is 'around' the cwnd rather than limited by the cwnd when the link is congested. The implementation allows the RTP queue to be small even when the frame sizes vary and thus increased e2e delay can be avoided.

       send_wnd = cwnd * CWND_OVERHEAD * rel_framesize_high -

The send window is updated whenever an RTP packet is transmitted or an RTCP feedback messaged is received.

4.2.5. 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 is also recommended to be used with L4S and 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, target_bitrate) *
      t_pace = rtp_size * 8 / pace_bitrate

rtp_size is the size of the last transmitted RTP packet, and s_rtt is the smoothed round trip time. RATE_PACE_MIN is the minimum pacing rate.

4.2.6. Stream Prioritization

The SCReAM algorithm makes a 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 specified in [RFC8699] can be used. One implementation of SCReAM and SCReAMv2 [SCReAM-CPP-implementation] uses credit-based scheduling. In credit-based scheduling, credit is accumulated by queues as they wait for service and is spent while the queues are being serviced. For instance, if one queue is allowed to transmit 1000 bytes, then a credit of 1000 bytes is allocated to the other unscheduled queues. This principle can be extended to weighted scheduling, where the credit allocated to unscheduled queues depends on the relative weights. The latter is also implemented in [SCReAM-CPP-implementation] in which case the target bitrate for the streams are also scaled relative to the scheduling priority.

4.3. Media Rate Control

The media rate control algorithm is executed whenever the congestion window is updated and updates the target bitrate. The target bitrate is essentiatlly based on the congestion window and the (smoothed) RTT according to

         target_bitrate = 8 * cwnd / s_rtt

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. Because the congestion window is updated based on loss, ECN-CE and delay, so does the target rate also update.

The code above however needs some modifications to work fine in a number of scenarios

  • L4S is inactive, i.e L4S is either not enabled or congested bottlenecks do not L4S mark packets

  • Frame sizes vary

  • cwnd is very small, just a few MSS or smaller

The complete pseudo code for adjustment of the target bitrate is shown below

        tmp_t = 1.0

        # limit bitrate if bytes in flight exceeds is close to or
        # exceeds cwnd. This helps to avoid large rate fluctiations and
        # variations in RTT
        # Only applied when L4S is inactive
        if (!l4s_active && bytes_in_flight_ratio > BYTES_IN_FLIGHT_LIMIT)
            bytesInFlightRatio / BYTES_IN_FLIGHT_LIMIT)

        # Scale down rate slighty when the congestion window is very
        # small compared to MSS
        tmp_t *= 1.0 - min(0.8, max(0.0, cwnd_ratio - 0.1))

        # Scale down rate when frame sizes vary much
        tmp_t /= rel_framesize_high

        # Calculate target bitrate and limit to min and max allowed
        # values
        target_bitrate = tmp_t * 8 * cwnd / s_rtt
        target_bitrate = min(TARGET_BITRATE_MAX,

The variable rel_framesize_high is based on calculation of the high percentile of the frame sizes. The calculation is based on a histogram of the frame sizes relative to the expected frame size given the target bitrate and frame period. The calculation of rel_framesize_high is done for every new video frame and is outlined roughly with the pseudo code below. For more detailed code, see [SCReAM-CPP-implementation].

        # frame_size is that frame size for the last encoded frame
        tmp_t = frame_size / (target_bitrate * frame_period / 8)

        if (tmp_t > 1.0)
          # Insert sample into histogram
          # Get high percentile
          rel_framesize_high = get_histogram_high_percentile()

A 75%-ile is used in [SCReAM-CPP-implementation], the histogram can be made leaky such that old samples are gradually forgotten.

4.4. Competing Flows Compensation

It is likely that a flow 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. SCReAMv2 takes care of such situations by adjusting the qdelay_target when loss-based flows are detected, as shown in the pseudocode below.

          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)

Two temporary variables are calculated. qdelay_norm_avg_t is the long-term average queue delay, qdelay_norm_var_t is the long-term variance of the queue delay. A high qdelay_norm_var_t indicates that the queue delay changes; this can be an indication that bottleneck bandwidth is reduced or that a competing flow has just entered. Thus, it indicates that it is not safe to adjust the queue delay target.

A low qdelay_norm_var_t indicates that the queue delay is relatively stable. The reason could be that the queue delay is low, but it 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 SCReAMv2 algorithm may be falsely interpreted as the presence of competing loss-based FTP flows is a false positive. The opposite case -- where the algorithm fails to detect the presence of a competing FTP flow -- is a false negative.

Extensive simulations have shown that the algorithm performs well in LTE test cases and that it also performs well in simple bandwidth-limited bottleneck test cases with competing FTP flows. However, the potential failure of the algorithm cannot be completely ruled out. A false positive (i.e., when self-inflicted congestion is mistakenly identified as competing flows) is especially problematic when it leads to increasing the target queue delay, which can cause the end-to-end delay to increase dramatically.

If it is deemed unlikely that competing flows occur over the same bottleneck, the algorithm described in this section MAY be turned off. One such case is QoS-enabled bearers in 3GPP-based access such as LTE. However, when sending over the Internet, often the network conditions are not known for sure, so in general it is not possible to make safe assumptions on how a network is used and whether or not competing flows share the same bottleneck. Therefore, turning this algorithm off must be considered with caution, as it can lead to basically zero throughput if competing with loss-based traffic.

4.5. Handling of systematic errors in video coders

Some video encoders are prone to systematically generate an output bitrate that is systematically larger or smaller than the target bitrate. SCReAMv2 can handle some deviation inherently but for larger devation it becomes necessary to compensate for this. The algorithm for this is detailed in [SCReAM-CPP-implementation].

ToDo: A future draft version will describe this in more detail as it has been fully integrated into SCReAMv2.

5. Receiver Requirements on Feedback Intensity

The simple task of the receiver is to feed back acknowledgements with with time stamp and ECN bits indication for received packets to the sender. Upon reception of each RTP packet, the receiver MUST maintain enough information to send the aforementioned values to the 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.

SCReAMv2 benefits from relatively frequent feedback. It is RECOMMENDED that a SCReAMv2 implementation follows the guidelines below.

The feedback interval depends on the media bitrate. At low bitrates, it is sufficient with a feedback every frame; while at high bitrates, a feedback interval of roughly 5ms ms is preferred. At very high bitrates, even shorter feedback intervals MAY be needed in order to keep the self-clocking in SCReAMv2 working well. One indication that feedback is too sparse is that the SCReAMv2 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:

      # Assume 100 byte RTCP packets
      rate_fb = 0.02 * [average received rate] / (100.0 * 8.0);
      rate_fb = min(1000, max(10, rate_fb))

      # Calculate feedback intervals
      fb_int = 1.0/rate_fb

Feedback should also forcibly be transmitted in any of these cases:

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 notexceed 2*fb_int in such cases, meaning that a scheduled feedback transmission event should not be delayed more than fb_int.

SCReAMv2 works with AVPF regular mode; immediate or early mode is not required by SCReAMv2 but can nonetheless be useful for RTCP messages not directly related to SCReAMv2, 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].

6. Discussion

This section covers a few discussion points.

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/SCReAMv2 is self-clocked, a malicious middlebox can drop RTCP feedback packets and thus cause the self-clocking to stall. However, this attack is mitigated by the minimum send rate maintained by SCReAM/SCReAMv2 when no feedback is received.

9. References

9.1. Normative References

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, , <>.
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, , <>.
Johansson, I. and M. Westerlund, "Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences", RFC 5506, DOI 10.17487/RFC5506, , <>.
Paxson, V., Allman, M., Chu, J., and M. Sargent, "Computing TCP's Retransmission Timer", RFC 6298, DOI 10.17487/RFC6298, , <>.
Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind, "Low Extra Delay Background Transport (LEDBAT)", RFC 6817, DOI 10.17487/RFC6817, , <>.
Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, , <>.
Sarker, Z., Perkins, C., Singh, V., and M. Ramalho, "RTP Control Protocol (RTCP) Feedback for Congestion Control", RFC 8888, DOI 10.17487/RFC8888, , <>.

9.2. Informative References

Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, , <>.
Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-Time Communication Use Cases and Requirements", RFC 7478, DOI 10.17487/RFC7478, , <>.
Johansson, I. and Z. Sarker, "Self-Clocked Rate Adaptation for Multimedia", RFC 8298, DOI 10.17487/RFC8298, , <>.
Khademi, N., Welzl, M., Armitage, G., and G. Fairhurst, "TCP Alternative Backoff with ECN (ABE)", RFC 8511, DOI 10.17487/RFC8511, , <>.
Islam, S., Welzl, M., and S. Gjessing, "Coupled Congestion Control for RTP Media", RFC 8699, DOI 10.17487/RFC8699, , <>.
Sarker, Z., Zhu, X., and J. Fu, "Evaluation Test Cases for Interactive Real-Time Media over Wireless Networks", RFC 8869, DOI 10.17487/RFC8869, , <>.
Cheng, Y., Cardwell, N., Dukkipati, N., and P. Jha, "The RACK-TLP Loss Detection Algorithm for TCP", RFC 8985, DOI 10.17487/RFC8985, , <>.
Bensley, S., Thaler, D., Balasubramanian, P., Eggert, L., and G. Judd, "Data Center TCP (DCTCP): TCP Congestion Control for Data Centers", RFC 8257, DOI 10.17487/RFC8257, , <>.
Jacobson, V., "Congestion Avoidance and Control", ACM SIGCOMM Computer Communication Review, DOI 10.1145/52325.52356, , <>.
Ros, D. and M. Welzl, "Assessing LEDBAT's Delay Impact", IEEE Communications Letters, Vol. 17, No. 5,, DOI 10.1109/LCOMM.2013.040213.130137, , <>.
"Policy and charging control architecture", 3GPP TS 23.203, , <>.
Ericsson Research, "SCReAM - Mobile optimised congestion control algorithm", n.d., <>.
Ericsson Research, "SCReAM - evaluations with L4S", n.d., <>.
Choi, S. and M. Handley, "Fairer TCP-Friendly Congestion Control Protocol for Multimedia Streaming Applications", DOI 10.1145/1364654.1364717, , <>.

Appendix A. Acknowledgments

We would like to thank the following people for their comments, questions, and support during the work that led to this memo: Mirja Kuehlewind

Authors' Addresses

Ingemar Johansson
Magnus Westerlund