draft-ietf-avtcore-cc-feedback-message-09.xml

<?xml version="1.0" encoding="US-ASCII"?>
<!DOCTYPE rfc SYSTEM "rfc2629.dtd">
<?rfc strict="yes" ?>
<?rfc toc="yes"?>
<?rfc tocdepth="4"?>
<?rfc symrefs="yes"?>
<?rfc sortrefs="yes" ?>
<?rfc compact="yes" ?>
<?rfc subcompact="no" ?>
<rfc category="std" docName="draft-ietf-avtcore-cc-feedback-message-09"
     ipr="trust200902">
  <front>
    <title abbrev="Congestion Control Feedback in RTCP">RTP Control Protocol
    (RTCP) Feedback for Congestion Control</title>

    <author fullname="Zaheduzzaman Sarker" initials="Z." surname="Sarker">
      <organization>Ericsson AB</organization>

      <address>
        <postal>
          <street>Torshamnsgatan 21</street>

          <city>Stockholm</city>

          <region></region>

          <code>164 40</code>

          <country>Sweden</country>
        </postal>

        <phone>+46107173743</phone>

        <email>zaheduzzaman.sarker@ericsson.com</email>
      </address>
    </author>

    <author fullname="Colin Perkins" initials="C. S." surname="Perkins">
      <organization>University of Glasgow</organization>

      <address>
        <postal>
          <street>School of Computing Science</street>

          <city>Glasgow</city>

          <code>G12 8QQ</code>

          <country>United Kingdom</country>
        </postal>

        <email>csp@csperkins.org</email>
      </address>
    </author>

    <author fullname="Varun Singh" initials="V." surname="Singh">
      <organization abbrev="callstats.io">CALLSTATS I/O Oy</organization>

      <address>
        <postal>
          <street>Annankatu 31-33 C 42</street>

          <city>Helsinki</city>

          <code>00100</code>

          <country>Finland</country>
        </postal>

        <email>varun.singh@iki.fi</email>

        <uri>http://www.callstats.io/</uri>
      </address>
    </author>

    <author fullname="Michael A. Ramalho" initials="M. A." surname="Ramalho">
      <address>
        <postal>
          <street>6310 Watercrest Way Unit 203</street>
          <city>Lakewood Ranch</city>
          <region>FL</region>
          <code>34202-5122</code>
          <country>USA</country>
        </postal>
        <phone>+1 732 832 9723</phone>
        <email>mar42@cornell.edu</email>
        <uri>http://ramalho.webhop.info/</uri>
      </address>
    </author>

    <date day="2" month="November" year="2020" />

    <area>Transport</area>

    <workgroup>IETF RMCAT Working Group</workgroup>

    <keyword>Congestion control, feedback message, RTP, RTCP</keyword>

    <abstract>
      <t>
        An effective RTP congestion control algorithm requires more fine-grained
        feedback on packet loss, timing, and ECN marks than is provided by the
        standard RTP Control Protocol (RTCP) Sender Report (SR) and Receiver
        Report (RR) packets.
        This document describes an RTCP feedback message intended to enable
        congestion control for interactive real-time traffic using RTP. The
        feedback message is designed for use with a sender-based congestion
        control algorithm, in which the receiver of an RTP flow sends RTCP
        feedback packets to the sender containing the information the sender
        needs to perform congestion control.
    </t>
    </abstract>
  </front>

  <middle>
    <section title="Introduction">
      <t>For interactive real-time traffic, such as video conferencing
      flows, the typical protocol choice is the Real-time Transport
      Protocol (RTP) <xref target="RFC3550"/> running over the User Datagram Protocol (UDP). RTP
      does not provide any guarantee of Quality of Service (QoS), reliability,
      or timely delivery, and expects the underlying transport protocol to
      do so.  UDP alone certainly does not meet that expectation. However,
      the RTP Control Protocol (RTCP) <xref target="RFC3550"/> provides a mechanism by which the
      receiver of an RTP flow can periodically send transport and media
      quality metrics to the sender of that RTP flow. This information can
      be used by the sender to perform congestion control.  In the absence
      of standardized messages for this purpose, designers of congestion
      control algorithms have developed proprietary RTCP messages that
      convey only those parameters needed for their respective designs.
      As a direct result, the different congestion control
      designs are not interoperable.  To enable algorithm
      evolution as well as interoperability across designs (e.g., different
      rate adaptation algorithms), it is highly desirable to have a generic
      congestion control feedback format.</t>

      <t>To help achieve interoperability for unicast RTP congestion control,
      this memo proposes a common RTCP feedback packet format that can be used by
      NADA <xref target="RFC8698"></xref>, SCReAM <xref
      target="RFC8298"></xref>, Google Congestion Control
      <xref target="I-D.ietf-rmcat-gcc"></xref> and Shared Bottleneck
      Detection <xref target="RFC8382"></xref>, and hopefully
      also by future RTP congestion control algorithms.</t>
    </section>

    <section title="Terminology">
      <t>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 <xref target="RFC2119"/> <xref target="RFC8174"/> when, and only
      when, they appear in all capitals, as shown here.</t>

      <t>In addition the terminology defined in <xref target="RFC3550"></xref>,
      <xref target="RFC4585"></xref>, and <xref target="RFC5506"></xref> applies.</t>
    </section>

    <section anchor="feedback_message" title="RTCP Feedback for Congestion Control">
      <t>Based on an analysis of NADA <xref target="RFC8698"></xref>,
      SCReAM <xref target="RFC8298"></xref>, Google
      Congestion Control <xref target="I-D.ietf-rmcat-gcc"></xref> and
      Shared Bottleneck Detection <xref target="RFC8382"></xref>,
      the following per-RTP packet congestion control feedback information
      has been determined to be necessary:</t>

      <t>
        <list style="symbols">
          <t>RTP sequence number: The receiver of an RTP flow needs to feed
          the sequence numbers of the received RTP packets back to the sender, so the
          sender can determine which packets were received and which were lost.
          Packet loss is used as an indication of congestion by many congestion
          control algorithms.</t>

          <t>Packet Arrival Time: The receiver of an RTP flow needs to feed
          the arrival time of each RTP packet back to the sender. Packet delay and/or
          delay variation (jitter) is used as a congestion signal by some congestion
          control algorithms.</t>

          <t>Packet Explicit Congestion Notification (ECN) Marking: If ECN
          <xref target="RFC3168"></xref>, <xref target="RFC6679"></xref> is
          used, it is necessary to feed back the 2-bit ECN mark in received
          RTP packets, indicating for each RTP packet whether it is marked
          not-ECT, ECT(0), ECT(1), or ECN-CE. If the path used by the RTP
          traffic is ECN capable the sender can use Congestion Experienced
          (ECN-CE) marking information as a congestion control signal.</t>
        </list>
      </t>

      <t>Every RTP flow is identified by its Synchronization Source (SSRC)
      identifier. Accordingly, the RTCP feedback format needs to group its
      reports by SSRC, sending one report block per received SSRC.</t>

      <t>As a practical matter, we note that host operating system (OS)
      process interruptions can occur at inopportune times. Accordingly,
      recording RTP packet send times at the sender, and the corresponding
      RTP packet arrival times at the
      receiver, needs to be done with deliberate care. This is because the time
      duration of host OS interruptions can be significant relative to the
      precision desired in the one-way delay estimates. Specifically, the send
      time needs to be recorded at the last opportunity prior to transmitting
      the RTP packet at the sender, and the arrival time at the receiver needs
      to be recorded at the earliest available opportunity.</t>

      <section anchor="sec:ccfb" title="RTCP Congestion Control Feedback Report">
        <t>Congestion control feedback can be sent as part of a regular
        scheduled RTCP report, or in an RTP/AVPF early feedback packet. 
        If sent as early feedback, congestion control feedback MAY be 
        sent in a non-compound RTCP packet <xref target="RFC5506"></xref> 
        if the RTP/AVPF profile <xref target="RFC4585"></xref> or the 
        RTP/SAVPF profile <xref target="RFC5124"></xref> is used.</t>

        <t>Irrespective of how it is transported, the congestion control 
        feedback is sent as a Transport Layer Feedback Message (RTCP packet 
        type 205). The format of this RTCP packet is shown in 
        <xref target="rtcp-packet"></xref>:</t>

        <t><figure anchor="rtcp-packet" title="RTCP Congestion Control Feedback Packet Format">
            <artwork><![CDATA[
     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| FMT=CCFB|   PT = 205    |          length               |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |                 SSRC of RTCP packet sender                    |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |                   SSRC of 1st RTP Stream                      |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |          begin_seq            |          num_reports          |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |R|ECN|  Arrival time offset    | ...                           .
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    .                                                               .
    .                                                               .
    .                                                               .
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |                   SSRC of nth RTP Stream                      |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |          begin_seq            |          num_reports          |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |R|ECN|  Arrival time offset    | ...                           |
    .                                                               .
    .                                                               .
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |                 Report Timestamp (32bits)                     |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
]]></artwork>
          </figure></t>

        <t>The first eight octets comprise a standard RTCP header, with
        PT=205 and FMT=CCFB indicating that this is a congestion control
        feedback packet, and with the SSRC set to that of the sender of
        the RTCP packet. (NOTE TO RFC EDITOR: please replace CCFB here and
        in the above diagram with the IANA assigned RTCP feedback packet
        type, and remove this note)</t>

        <t>Section 6.1 of <xref target="RFC4585"></xref> requires the RTCP
        header to be followed by the SSRC of the RTP flow being reported 
        upon.  Accordingly, the RTCP header is followed by a report block
        for each SSRC from which RTP packets have been received, followed
        by a Report Timestamp.</t>
     
        <t>Each report block begins with the SSRC of the received RTP Stream
        on which it is reporting. Following this, the report block contains a
        16-bit packet metric block for each RTP packet with sequence number
        in the range begin_seq to begin_seq+num_reports inclusive (calculated using
        arithmetic modulo 65536 to account for possible sequence number wrap-around). 
        If the number of 16-bit packet metric blocks included in the report
        block is not a multiple of two, then 16 bits of zero padding MUST be
        added after the last packet metric block, to align the end of the
        packet metric blocks with the next 32 bit boundary.
        The value of num_reports MAY be zero, indicating that there are no
        packet metric blocks included for that SSRC.
        Each report block MUST NOT include more than 16384 packet metric blocks
        (i.e., it MUST NOT report on more than one quarter of the sequence
        number space in a single report).
        </t>

        <t>
        The contents of each 16-bit packet metric block comprises the R, ECN,
        and ATO fields as follows:
          <list style="symbols">
            <t>
              Received (R, 1 bit): is a boolean to indicate if the packet was
              received. 0 represents that the packet was not yet received and
              the subsequent 15-bits (ECN and ATO) in this 16-bit packet
              metric block are also set to 0 and MUST be ignored.
              1 represents that the packet was received and the subsequent
              bits in the block need to be parsed.
            </t>

            <t>
              ECN (2 bits): is the echoed ECN mark of the packet. These 
              are set to 00 if not received, or if ECN is not used.
            </t>

            <t>
              Arrival time offset (ATO, 13 bits): is the arrival time of
              the RTP packet at the receiver, as an offset before the time
              represented by the Report Timestamp (RTS) field of this RTCP congestion control
              feedback report. The ATO field is in units of 1/1024 seconds
              (this unit is chosen to give exact offsets from the RTS field)
              so, for example, an ATO value of 512 indicates that the
              corresponding RTP packet arrived exactly half a second before
              the time instant represented by the RTS field.
              If the measured value is greater than 8189/1024 seconds (the
              value that would be coded as 0x1FFD), the value 0x1FFE MUST
              be reported to indicate an over-range measurement.
              If the measurement is unavailable, or if the arrival time of
              the RTP packet is after the time represented by the RTS field,
              then an ATO value of 0x1FFF MUST be reported for the packet.
            </t>
          </list>
        </t>

        <t>The RTCP congestion control feedback report packet concludes with
        the Report Timestamp field (RTS, 32 bits). This denotes the time
        instant on which this packet is reporting, and is the instant from
        which the arrival time offset values are calculated.
        The value of RTS field is derived from the same clock used to generate
        the NTP timestamp field in RTCP Sender Report (SR) packets. It
        is formatted as the middle 32 bits of an NTP format timestamp, as 
        described in Section 4 of <xref target="RFC3550"></xref>.</t>

        <t>RTCP congestion control feedback packets SHOULD include a report
        block for every active SSRC. The sequence
        number ranges reported on in consecutive reports for a given SSRC will
        generally be contiguous, but overlapping reports MAY be sent (and need
        to be sent in cases where RTP packet reordering occurs across the
        boundary between consecutive reports).
        If an RTP packet was reported as received in one report, that packet
        MUST also be reported as received in any overlapping reports sent
        later that cover its sequence number range.
        If reports covering overlapping sequence number ranges are sent,
        information in later reports updates that sent in previous reports
        for RTP packets included in both reports.
        </t>

        <t>RTCP congestion control feedback packets can be large if they are
        sent infrequently relative to the number of RTP data packets.  If an
        RTCP congestion control feedback packet is too large to fit within the
        path MTU, its sender SHOULD split it into multiple feedback packets.
        The RTCP reporting interval SHOULD be chosen such that feedback packets
        are sent often enough that they are small enough to fit within the path
        MTU (<xref target="I-D.ietf-rmcat-rtp-cc-feedback"/> discusses how to
        choose the reporting interval; specifications for RTP congestion control
        algorithms can also provide guidance).</t>


        <t>If duplicate copies of a particular RTP packet are received, then the
        arrival time of the first copy to arrive MUST be reported. If any of the
        copies of the duplicated packet are ECN-CE marked, then an ECN-CE mark
        MUST be reported that for packet; otherwise the ECN mark of the first
        copy to arrive is reported.</t>

        <t>If no packets are received from an SSRC in a reporting interval,
        a report block MAY be sent with begin_seq set to the highest sequence
        number previously received from that SSRC and num_reports set to zero
        (or, the report can simply to omitted). The corresponding SR/RR packet 
        will have a non-increased extended highest sequence number received
        field that will inform the sender that no packets have been received,
        but it can ease processing to have that information available in the
        congestion control feedback reports too.</t>

        <t>A report block indicating that certain RTP packets were lost is
        not to be interpreted as a request to retransmit the lost packets.
        The receiver of such a report might choose to retransmit such packets,
        provided a retransmission payload format has been negotiated, but
        there is no requirement that it do so.</t>
      </section>
    </section>

    <section title="Feedback Frequency and Overhead">
      <t>There is a trade-off between speed and accuracy of reporting, and the
      overhead of the reports. <xref target="I-D.ietf-rmcat-rtp-cc-feedback"/>
      discusses this trade-off, suggests desirable RTCP feedback rates, and
      provides guidance on how to configure the RTCP bandwidth fraction, etc.,
      to make appropriate use of the reporting block described in this memo.
      Specifications for RTP congestion control algorithms can also provide
      guidance.</t>

      <t>It is generally understood that congestion control algorithms work
      better with more frequent feedback.
      However, RTCP bandwidth and transmission rules put some upper limits
      on how frequently the RTCP feedback messages can be sent from an RTP
      receiver to the RTP sender.
      
      In many cases, sending feedback once per frame is an upper bound
      before the reporting overhead becomes excessive, although this will
      depend on the media rate and more frequent feedback might be needed
      with high-rate media flows <xref target="I-D.ietf-rmcat-rtp-cc-feedback"></xref>.
      Analysis <xref target="feedback-requirements"></xref> has also shown
      that some candidate congestion control algorithms can operate with less
      frequent feedback, using a feedback interval range of 50-200ms.
      Applications need to negotiate an appropriate congestion control
      feedback interval at session setup time, based on the choice of
      congestion control algorithm, the expected media bit rate, and
      the acceptable feedback overhead.
    </t>
    </section>

    <section title="Response to Loss of Feedback Packets">
      <t>
        Like all RTCP packets, RTCP congestion control feedback packets
        might be lost. All RTP congestion control algorithms MUST specify
        how they respond to the loss of feedback packets.
      </t>

      <t>
        RTCP packets do not contain a sequence number, so loss of feedback
        packets has to be inferred based on the time since the last feedback
        packet.
        If only a single congestion control feedback packet is lost, an
        appropriate response is to assume that the level of congestion
        has remained roughly the same as the previous report. However,
        if multiple consecutive congestion control feedback packets are
        lost, then the media sender SHOULD rapidly reduce its sending rate as
        this likely indicates a path failure. The RTP circuit
        breaker <xref target="RFC8083"/> provides further guidance.
      </t>
    </section>

    <section title="SDP Signalling">
      <t>
        A new "ack" feedback parameter, "ccfb", is defined for use with the
        "a=rtcp-fb:" SDP extension to indicate the use of the RTP Congestion
        Control feedback packet format defined in <xref target="feedback_message"/>.
        The ABNF definition of this SDP parameter extension is:
      </t>
      <figure>
        <artwork><![CDATA[
        rtcp-fb-ack-param = <See Section 4.2 of [RFC4585]>
        rtcp-fb-ack-param =/ ccfb-par
        ccfb-par          = SP "ccfb"
        ]]></artwork>
      </figure>
      <t>
        The payload type used with "ccfb" feedback MUST be the wildcard type ("*").
        This implies that the congestion control feedback is sent
        for all payload types in use in the session, including any FEC and
        retransmission payload types.
        An example of the resulting SDP attribute is:
      </t>
      <figure>
        <artwork><![CDATA[
        a=rtcp-fb:* ack ccfb
        ]]></artwork>
      </figure>

      <t>
        The offer/answer rules for these SDP feedback parameters are
        specified in Section 4.2 of the RTP/AVPF profile <xref target="RFC4585"/>.
      </t>

      <t>
        An SDP offer might indicate support for both the congestion control
        feedback mechanism specified in this memo and one or more alternative
        congestion control feedback mechanisms that offer substantially the
        same semantics. In this case, the answering party SHOULD include
        only one of the offered congestion control feedback mechanisms in its
        answer.  If a re-invite offering the same set of congestion control
        feedback mechanisms is received, the generated answer SHOULD choose
        the same congestion control feedback mechanism as in the original
        answer where possible.
      </t>


      <t>
        When the SDP BUNDLE extension <xref target="I-D.ietf-mmusic-sdp-bundle-negotiation"/>
        is used for multiplexing, the "a=rtcp-fb:" attribute has multiplexing category
        IDENTICAL-PER-PT <xref target="I-D.ietf-mmusic-sdp-mux-attributes"/>.
      </t>
    </section>

    <section title="Relation to RFC 6679">
      <t>
        Use of Explicit Congestion Notification (ECN) with RTP is
        described in <xref target="RFC6679"/>. That specifies how
        to negotiate the use of ECN with RTP, and defines an RTCP ECN
        Feedback Packet to carry ECN feedback reports. It uses an SDP
        "a=ecn-capable-rtp:" attribute to negotiate use of ECN, and
        the "a=rtcp-fb:" attributes with the "nack" parameter "ecn" to
        negotiate the use of RTCP ECN Feedback Packets.
      </t>

      <t>
        The RTCP ECN Feedback Packet is not useful when ECN is used with
        the RTP Congestion Control Feedback Packet defined in this memo
        since it provides duplicate information. When
        congestion control feedback is to be used with RTP and ECN,
        the SDP offer generated MUST include an "a=ecn-capable-rtp:"
        attribute to negotiate ECN support, along with an "a=rtcp-fb:"
        attribute with the "ack" parameter "ccfb" to indicate that the
        RTP Congestion Control Feedback Packet can be used.
        The "a=rtcp-fb:" attribute MAY also include the "nack" parameter
        "ecn", to indicate that the RTCP ECN Feedback Packet is also
        supported. If an SDP offer signals support for both RTP
        Congestion Control Feedback Packets and the RTCP ECN Feedback
        Packet, the answering party SHOULD signal support for one, but
        not both, formats in its SDP answer to avoid sending duplicate
        feedback.
      </t>

      <t>
        When using ECN with RTP, the guidelines in Section 7.2 of 
        <xref target="RFC6679"/> MUST be followed to initiate the use of ECN in an
        RTP session. The guidelines in Section 7.3 of <xref target="RFC6679"/>
        MUST also be followed about ongoing use of ECN within an RTP
        session, with the exception that feedback is sent using the RTCP
        Congestion Control Feedback Packets described in this memo rather
        than using RTP ECN Feedback Packets. Similarly, the guidance
        in Section 7.4 of <xref target="RFC6679"/> around detecting failures
        MUST be followed, with the exception that the necessary information is
        retrieved from the RTCP Congestion Control Feedback Packets rather than
        from RTP ECN Feedback Packets.
      </t>
    </section>

    <section title="Design Rationale">
      <t>The primary function of RTCP SR/RR packets is to report statistics
      on the reception of RTP packets. The reception report blocks sent in
      these packets contain information about observed jitter, fractional
      packet loss, and cumulative packet loss. It was intended that this
      information could be used to  support congestion control algorithms,
      but experience has shown that it is not sufficient for that purpose.
      An efficient congestion control algorithm requires more fine-grained
      information on per-packet reception quality than is provided by SR/RR
      packets to react effectively. The feedback format defined in this memo
      provides such fine-grained feedback.</t> 

      <t>Several other RTCP extensions also provide more detailed feedback
      than SR/RR packets:</t>

      <t>
        <list style="hanging">
          <t hangText="TMMBR:">The Codec Control Messages for the RTP/AVPF profile 
          <xref target="RFC5104"></xref> include a Temporary Maximum Media Bit
          Rate (TMMBR) message. This is used to convey a temporary maximum bit
          rate limitation from a receiver of RTP packets to their sender. Even
          though it was not designed to replace congestion control, TMMBR has
          been used as a means to do receiver based congestion control where
          the session bandwidth is high enough to send frequent TMMBR messages,
            especially when used with non-compound RTCP packets <xref target="RFC5506"></xref>.
          This approach requires the receiver of the RTP packets to monitor
          their reception, determine the level of congestion, and recommend 
          a maximum bit rate suitable for current available bandwidth on the
          path; it also assumes that the RTP sender can/will respect that bit
          rate.  This is the opposite of the sender-based congestion control
          approach suggested in this memo, so TMMBR cannot be used to convey
          the information needed for a sender-based congestion control.  TMMBR
          could, however, be viewed a complementary mechanism that can inform
          the sender of the receiver's current view of acceptable maximum bit
          rate. Mechanisms that convey the receiver's estimate of the maximum
          available bit-rate provide similar feedback.
          </t>


          <t hangText="RTCP Extended Reports (XR):">Numerous RTCP extended
          report (XR) blocks have been defined to report details of packet
          loss, arrival times <xref target="RFC3611"/>, delay 
          <xref target="RFC6843"/>, and ECN marking <xref target="RFC6679"/>. 
          It is possible to combine several such XR blocks into a compound
          RTCP packet, to report the detailed loss, arrival time, and ECN
          marking information needed for effective sender-based
          congestion control. However, the result has high overhead both
          in terms of bandwidth and complexity, due to the need to stack
          multiple reports.</t>

          <t hangText="Transport-wide Congestion Control:">The format
          defined in this memo provides individual feedback on each SSRC.
          An alternative is to add a header extension to each RTP packet,
          containing a single, transport-wide, packet sequence number,
          then have the receiver send RTCP reports giving feedback on
          these additional sequence numbers
          <xref target="I-D.holmer-rmcat-transport-wide-cc-extensions"/>.
          Such an approach adds the per-packet overhead of the header
          extension (8 octets per packet in the referenced format),
          but reduces the size of the feedback packets, and can simplify
          the rate calculation at the sender if it maintains a single
          rate limit that applies to all RTP packets sent irrespective
          of their SSRC. Equally, the use of transport-wide feedback makes
          it more difficult to adapt the sending rate, or respond to lost
          packets, based on the reception and/or loss patterns observed
          on a per-SSRC basis (for example, to perform differential rate
          control and repair for audio and video flows, based on knowledge
          of what packets from each flow were lost). Transport-wide 
          feedback is also a less natural fit with the wider RTP framework,
          which makes extensive use of per-SSRC sequence numbers and
          feedback.</t>

        </list>
      </t>

      <t>Considering these issues, we believe it appropriate to design a
      new RTCP feedback mechanism to convey information for sender-based
      congestion control algorithms. The new congestion control feedback
      RTCP packet described in <xref target="feedback_message"></xref>
      provides such a mechanism.</t> 

    </section>

    <section anchor="Acknowledgements" title="Acknowledgements">
      <t>This document is based on the outcome of a design team discussion
      in the RTP Media Congestion Avoidance Techniques (RMCAT) working group.
      The authors would like to thank
      David Hayes,
      Stefan Holmer,
      Randell Jesup,
      Ingemar Johansson,
      Jonathan Lennox,
      Sergio Mena,
      Nils Ohlmeier,
      Magnus Westerlund,
      and
      Xiaoqing Zhu
      for their valuable feedback.</t>
    </section>

    <!-- Possibly a 'Contributors' section ... -->

    <section anchor="IANA" title="IANA Considerations">
      <t>
        The IANA is requested to register one new RTP/AVPF Transport-Layer
        Feedback Message in the table for FMT values for RTPFB Payload Types
        <xref target="RFC4585"/> as defined in <xref target="sec:ccfb"/>:
      </t>

      <figure>
        <artwork><![CDATA[
      Name:      CCFB
      Long name: RTP Congestion Control Feedback
      Value:     (to be assigned by IANA)
      Reference: (RFC number of this document, when published)
        ]]></artwork>
      </figure>

      <t>
        The IANA is also requested to register one new SDP "rtcp-fb" attribute
        "ack" parameter, "ccfb", in the SDP ("ack" and "nack" Attribute Values)
        registry:
      </t>

      <figure>
        <artwork><![CDATA[
      Value name:  ccfb
      Long name:   Congestion Control Feedback
      Usable with: ack
      Mux:         IDENTICAL-PER-PT
      Reference:   (RFC number of this document, when published)
        ]]></artwork>
      </figure>
    </section>

    <section anchor="Security" title="Security Considerations">
      <t>The security considerations of the RTP specification
      <xref target="RFC3550"/>, the applicable RTP profile (e.g.,
      <xref target="RFC3551"/>, <xref target="RFC3711"/>, or
      <xref target="RFC4585"/>), and the RTP congestion control algorithm
      that is in use (e.g., <xref target="RFC8698"/>,
      <xref target="RFC8298"/>,
      <xref target="I-D.ietf-rmcat-gcc"/>, or
      <xref target="RFC8382"/>) apply.</t>

      <t>A receiver that intentionally generates inaccurate RTCP congestion
      control feedback reports might be able trick the sender into sending
      at a greater rate than the path can support, thereby causing congestion on the
      path. This will negatively impact the quality of experience of that
      receiver, and potentially cause denial of service to other traffic
      sharing the path and excessive resource usage at the media sender.
      Since RTP is an unreliable transport, a sender can intentionally drop a packet,
      leaving a gap in the RTP sequence number space without causing serious harm, to
      check that the receiver is correctly reporting losses (this needs to be done with
      care and some awareness of the media data being sent, to limit impact on the user
      experience).</t>

      <t>An on-path attacker that can modify RTCP congestion control feedback
      packets can change the reports to trick the sender into sending at either
      an excessively high or excessively low rate, leading to denial of service.
      The secure RTCP profile <xref target="RFC3711"/> can be used to authenticate
      RTCP packets to protect against this attack.</t>

      <t>An off-patch attacker that can spoof RTCP congestion control feedback
      packets can similarly trick a sender into sending at an incorrect
      rate, leading to denial of service.  This attack is difficult, since the
      attacker needs to guess the SSRC and sequence number in addition to the
      destination transport address. As with on-patch attacks, the secure RTCP
      profile <xref target="RFC3711"/> can be used to authenticate RTCP packets
      to protect against this attack.</t>
    </section>
  </middle>

  <back>
    <references title="Normative References">
      <?rfc include='reference.RFC.2119'?>

      <?rfc include='reference.RFC.3168'?>

      <?rfc include='reference.RFC.3550'?>

      <?rfc include='reference.RFC.3551'?>

      <?rfc include='reference.RFC.3711'?>

      <?rfc include='reference.RFC.4585'?>

      <?rfc include='reference.RFC.5124'?>

      <?rfc include='reference.RFC.5506'?>

      <?rfc include='reference.RFC.6679'?>

      <?rfc include='reference.RFC.8083'?>

      <?rfc include='reference.RFC.8174'?>

      <?rfc include='reference.I-D.ietf-mmusic-sdp-bundle-negotiation'?>

      <?rfc include='reference.I-D.ietf-mmusic-sdp-mux-attributes'?>
    </references>

    <references title="Informative References">
      <?rfc include='reference.I-D.ietf-rmcat-rtp-cc-feedback'?>

      <?rfc include="reference.I-D.ietf-rmcat-gcc"?>

      <?rfc include='reference.RFC.3611'?>

      <?rfc include='reference.RFC.5104'?>

      <?rfc include='reference.RFC.6843'?>

      <?rfc include="reference.RFC.8298"?>

      <?rfc include="reference.RFC.8382"?>

      <?rfc include='reference.RFC.8698"?>

      <?rfc include="reference.I-D.holmer-rmcat-transport-wide-cc-extensions"?>

      <?rfc include="reference.I-D.alvestrand-rmcat-remb"?>

      <reference anchor="feedback-requirements"
                 target="://www.ietf.org/proceedings/95/slides/slides-95-rmcat-1.pdf">
        <front>
          <title>RMCAT Feedback Requirements</title>

          <author>
            <organization></organization>
          </author>

          <date />
        </front>
      </reference>
    </references>
  </back>
</rfc>