    Internet Engineering Task Force (IETF)                        C. Perkins
    Request for Comments: 8083                         University of Glasgow
    Updates: 3550                                                   V. Singh
    Category: Standards Track                                   callstats.io
    ISSN: 2070-1721                                               March 2017

    Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions

    Abstract

       The Real-time Transport Protocol (RTP) is widely used in telephony,
       video conferencing, and telepresence applications.  Such applications
       are often run on best-effort UDP/IP networks.  If congestion control
       is not implemented in these applications, then network congestion can
       lead to uncontrolled packet loss and a resulting deterioration of the
       user's multimedia experience.  The congestion control algorithm acts
       as a safety measure by stopping RTP flows from using excessive
       resources and protecting the network from overload.  At the time of
       this writing, however, while there are several proprietary solutions,
       there is no standard algorithm for congestion control of interactive
       RTP flows.

       This document does not propose a congestion control algorithm.  It
       instead defines a minimal set of RTP circuit breakers: conditions
       under which an RTP sender needs to stop transmitting media data to
       protect the network from excessive congestion.  It is expected that,
       in the absence of long-lived excessive congestion, RTP applications
       running on best-effort IP networks will be able to operate without
       triggering these circuit breakers.  To avoid triggering the RTP
       circuit breaker, any Standards Track congestion control algorithms
       defined for RTP will need to operate within the envelope set by these
       RTP circuit breaker algorithms.

    Status of This Memo

       This is an Internet Standards Track document.

       This document is a product of the Internet Engineering Task Force
       (IETF).  It represents the consensus of the IETF community.  It has
       received public review and has been approved for publication by the
       Internet Engineering Steering Group (IESG).  Further information on
       Internet Standards is available in Section 2 of RFC 7841.

       Information about the current status of this document, any errata,
       and how to provide feedback on it may be obtained at
       http://www.rfc-editor.org/info/rfc8083.

    Perkins & Singh              Standards Track                    [Page 1]

    RFC 8083                  RTP Circuit Breakers                March 2017

    Copyright Notice

       Copyright (c) 2017 IETF Trust and the persons identified as the
       document authors.  All rights reserved.

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

    Table of Contents

       1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
       2.  Background  . . . . . . . . . . . . . . . . . . . . . . . . .   3
       3.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   6
       4.  RTP Circuit Breakers for Systems Using the RTP/AVP Profile  .   8
         4.1.  RTP/AVP Circuit Breaker #1: RTCP Timeout  . . . . . . . .  10
         4.2.  RTP/AVP Circuit Breaker #2: Media Timeout . . . . . . . .  11
         4.3.  RTP/AVP Circuit Breaker #3: Congestion  . . . . . . . . .  12
         4.4.  RTP/AVP Circuit Breaker #4: Media Usability . . . . . . .  16
         4.5.  Ceasing Transmission  . . . . . . . . . . . . . . . . . .  17
       5.  RTP Circuit Breakers and the RTP/AVPF and RTP/SAVPF Profiles   18
       6.  Impact of RTCP Extended Reports (XR)  . . . . . . . . . . . .  19
       7.  Impact of Explicit Congestion Notification (ECN)  . . . . . .  19
       8.  Impact of Bundled Media and Layered Coding  . . . . . . . . .  20
       9.  Security Considerations . . . . . . . . . . . . . . . . . . .  20
       10. References  . . . . . . . . . . . . . . . . . . . . . . . . .  21
         10.1.  Normative References . . . . . . . . . . . . . . . . . .  21
         10.2.  Informative References . . . . . . . . . . . . . . . . .  22
       Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . .  25
       Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  25

    Perkins & Singh              Standards Track                    [Page 2]

    RFC 8083                  RTP Circuit Breakers                March 2017

    1.  Introduction

       The Real-time Transport Protocol (RTP) [RFC3550] is widely used in
       voice-over-IP, video teleconferencing, and telepresence systems.
       Many of these systems run over best-effort UDP/IP networks and can
       suffer from packet loss and increased latency if network congestion
       occurs.  Designing effective RTP congestion control algorithms to
       adapt the transmission of RTP-based media to match the available
       network capacity while also maintaining the user experience is a
       difficult but important problem.  Many such congestion control and
       media adaptation algorithms have been proposed, but to date there is
       no consensus on the correct approach or even that a single standard
       algorithm is desirable.

       This memo does not attempt to propose a new RTP congestion control
       algorithm.  Instead, we propose a small set of RTP circuit breakers:
       mechanisms that terminate RTP flows in conditions under which there
       is general agreement that serious network congestion is occurring.
       The RTP circuit breakers proposed in this memo are a specific
       instance of the general class of network transport circuit breakers
       [RFC8084] designed to act as a protection mechanism of last resort to
       avoid persistent excessive congestion.  To avoid triggering the RTP
       circuit breaker, any Standards Track congestion control algorithms
       defined for RTP will need to operate within the envelope set by the
       RTP circuit breaker algorithms defined by this memo.

    2.  Background

       We consider congestion control for unicast RTP traffic flows.  This
       is the problem of adapting the transmission of an audio/visual data
       flow, encapsulated within an RTP transport session, from one sender
       to one receiver so that it does not use more capacity than is
       available along the network path.  Such adaptation needs to be done
       in a way that limits the disruption to the user experience caused by
       both packet loss and excessive rate changes.  Congestion control for
       multicast flows is outside the scope of this memo.  Multicast traffic
       needs different solutions since the available capacity estimator for
       a group of receivers will differ from that for a single receiver, and
       because multicast congestion control has to consider issues of
       fairness across groups of receivers that do not apply to unicast
       flows.

       Congestion control for unicast RTP traffic can be implemented in one
       of two places in the protocol stack.  One approach is to run the RTP
       traffic over a congestion-controlled transport protocol (for example,
       over TCP), and to adapt the media encoding to match the dictates of
       the transport-layer congestion control algorithm.  This is safe for
       the network but can be suboptimal for the media quality unless the

    Perkins & Singh              Standards Track                    [Page 3]

    RFC 8083                  RTP Circuit Breakers                March 2017

       transport protocol is designed to support real-time media flows.  We
       do not consider this class of applications further in this memo, as
       their network safety is guaranteed by the underlying transport.

       Alternatively, RTP flows can be run over a non-congestion-controlled
       transport protocol (for example, UDP) performing rate adaptation at
       the application layer based on RTP Control Protocol (RTCP) feedback.
       With a well-designed, network-aware application, this allows highly
       effective media quality adaptation, but there is potential to cause
       persistent congestion in the network if the application does not
       adapt its sending rate in a timely and effective manner.  We consider
       this class of applications in this memo.

       Congestion control relies on monitoring the delivery of a media flow
       and responding to adapt the transmission of that flow when there are
       signs that the network path is congested.  Network congestion can be
       detected in one of three ways:

       1)  a receiver can infer the onset of congestion by observing an
           increase in one-way delay caused by queue build-up within the
           network;

       2)  if Explicit Congestion Notification (ECN) [RFC3168] is supported,
           the network can signal the presence of congestion by marking
           packets using ECN Congestion Experienced (CE) marks (this could
           potentially be augmented by mechanisms such as Congestion
           Exposure (ConEx) [RFC7713] or other future protocol extensions
           for network signaling of congestion); or

       3)  in the extreme case, congestion will cause packet loss that can
           be detected by observing a gap in the received RTP sequence
           numbers.

       Once the onset of congestion is observed, the receiver has to send
       feedback to the sender to indicate that the transmission rate needs
       to be reduced.  How the sender reduces the transmission rate is
       highly dependent on the media codec being used and is outside the
       scope of this memo.

       There are several ways in which a receiver can send feedback to a
       media sender within the RTP framework:

       o  The base RTP specification [RFC3550] defines RTCP Receiver Report
          (RR) packets to convey reception quality feedback information and
          Sender Report (SR) packets to convey information about the media
          transmission.  RTCP SR packets contain data that can be used to
          reconstruct media timing at a receiver along with a count of the
          total number of octets and packets sent.  RTCP RR packets report

    Perkins & Singh              Standards Track                    [Page 4]

    RFC 8083                  RTP Circuit Breakers                March 2017

          on the fraction of packets lost in the last reporting interval,
          the cumulative number of packets lost, the highest sequence number
          received, and the inter-arrival jitter.  The RTCP RR packets also
          contain timing information that allows the sender to estimate the
          network Round-Trip Time (RTT) to the receivers.  RTCP reports are
          sent periodically, with the reporting interval being determined by
          the number of Synchronization Sources (SSRCs) used in the session
          and a configured session bandwidth estimate (the number of SSRCs)
          used is usually two in a unicast session, one for each
          participant, but can be greater if the participants send multiple
          media streams).  The interval between reports sent from each
          receiver is on the order of a few seconds on average; although it
          varies with the session bandwidth, it is randomized to avoid
          synchronization of reports from multiple receivers.  The interval
          can be less than a second in a high-bandwidth session.  RTCP RR
          packets allow a receiver to report ongoing network congestion to
          the sender.  However, if a receiver detects the onset of
          congestion part way through a reporting interval, the base RTP
          specification contains no provision for sending the RTCP RR packet
          early, and the receiver has to wait until the next scheduled
          reporting interval.

       o  The RTCP Extended Reports (XR) [RFC3611] allow reporting of more
          complex and sophisticated reception quality metrics but do not
          change the RTCP timing rules.  RTCP extended reports of potential
          interest for congestion control purposes are the extended packet
          loss, discard, and burst metrics [RFC3611] [RFC7002] [RFC7097]
          [RFC7003] [RFC6958] as well as the extended delay metrics
          [RFC6843] [RFC6798].  Other RTCP Extended Reports that could be
          helpful for congestion control purposes might be developed in
          future.

       o  Rapid feedback about the occurrence of congestion events can be
          achieved using the Extended RTP Profile for RTCP-Based Feedback
          (RTP/AVPF) [RFC4585] (or its secure variant, RTP/SAVPF [RFC5124])
          in place of the RTP/AVP profile [RFC3551].  This modifies the RTCP
          timing rules to allow RTCP reports to be sent early, in some cases
          immediately, provided the RTCP transmission rate keeps within its
          bandwidth allocation.  It also defines transport-layer feedback
          messages, including Negative Acknowledgements (NACKs), that can be
          used to report on specific congestion events.  RTP Codec Control
          Messages [RFC5104] extend the RTP/AVPF profile with additional
          feedback messages that can be used to influence the way in which
          rate adaptation occurs but do not further change the dynamics of
          how rapidly feedback can be sent.  Use of the RTP/AVPF profile is
          dependent on signaling.

    Perkins & Singh              Standards Track                    [Page 5]

    RFC 8083                  RTP Circuit Breakers                March 2017

       o  Finally, ECN for RTP over UDP [RFC6679] can be used to provide
          feedback on the number of packets that received an ECN-CE mark.
          This RTCP extension builds on the RTP/AVPF profile to allow rapid
          congestion feedback when ECN is supported.

       In addition to these mechanisms for providing feedback, the sender
       can include an RTP header extension in each packet to record packet
       transmission times [RFC5450].  Accurate transmission timestamps can
       be helpful for estimating queuing delays to get an early indication
       of the onset of congestion.

       Taken together, these various mechanisms allow receivers to provide
       feedback on the senders when congestion events occur, with varying
       degrees of timeliness and accuracy.  The key distinction is between
       systems that use only the basic RTCP mechanisms, without RTP/AVPF
       rapid feedback, and those that use the RTP/AVPF extensions to respond
       to congestion more rapidly.

    3.  Terminology

       The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
       "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
       document are to be interpreted as described in RFC 2119 [RFC2119].
       This interpretation of these key words applies only when written in
       ALL CAPS.  Mixed- or lower-case uses of these key words are not to be
       interpreted as carrying special significance in this memo.

       The definition of the RTP circuit breaker is specified in terms of
       the following variables:

       o  Td is the deterministic RTCP reporting interval, as defined in
          Section 6.3.1 of [RFC3550].

       o  Tdr is the sender's estimate of the deterministic RTCP reporting
          interval, Td, calculated by a receiver of the data it is sending.
          Tdr is not known at the sender but can be estimated by executing
          the algorithm in Section 6.2 of [RFC3550] using the average RTCP
          packet size seen at the sender, the number of members reported in
          the receiver's SR/RR report blocks, and whether the receiver is
          sending SR or RR packets.  Tdr is recalculated when each new RTCP
          SR/RR report is received, but the media timeout circuit breaker
          (see Section 4.2) is only reconsidered when Tdr increases.

    Perkins & Singh              Standards Track                    [Page 6]

    RFC 8083                  RTP Circuit Breakers                March 2017

       o  Tr is the network round-trip time, which is calculated by the
          sender using the algorithm in Section 6.4.1 of [RFC3550] and is
          smoothed using an exponentially weighted moving average as
          Tr = (0.8 * Tr) + (0.2 * Tr_new) where Tr_new is the latest RTT
          estimate obtained from an RTCP report.  The weight is chosen so
          old estimates decay over k intervals.

       o  k is the non-reporting threshold (see Section 4.2).

       o  Tf is the media framing interval at the sender.  For applications
          sending at a constant frame rate, Tf is the inter-frame interval.
          For applications that switch between a small set of possible frame
          rates (for example, when sending speech with comfort noise, such
          that comfort noise frames are sent less often than speech frames),
          Tf is set to the longest of the inter-frame intervals of the
          different frame rates.  For applications that send periodic frames
          but dynamically vary their frame rate, Tf is set to the largest
          inter-frame interval used in the last 10 seconds.  For
          applications that send less than one frame every 10 seconds, or
          that have no concept of periodic frames (e.g., text conversation
          [RFC4103], or pointer events [RFC2862]), when each frame is sent,
          Tf is set to the time interval since the previous frame.

       o  G is the frame group size.  That is, the number of frames that are
          coded together based on a particular sending rate setting.  If the
          codec used by the sender can change its rate on each frame, then G
          = 1; otherwise, G is set to the number of frames before the codec
          can adjust to the new rate.  For codecs that have the concept of a
          Group of Pictures (GOP), G is likely the GOP length.

       o  T_rr_interval is the minimal interval between RTCP reports, as
          defined in Section 3.4 of [RFC4585]; it is only meaningful for
          implementations of RTP/AVPF profile [RFC4585] or the RTP/SAVPF
          profile [RFC5124].

       o  X is the estimated throughput a TCP connection would achieve over
          a path, in bytes per second.

       o  s is the size of RTP packets being sent, in bytes.  If the RTP
          packets being sent vary in size, then the average size over the
          packet comprising the last 4 * G frames MUST be used (this is
          intended to be comparable to the four loss intervals used in
          [RFC5348]).

       o  p is the loss event rate, between 0.0 and 1.0, that would be seen
          by a TCP connection over a particular path.  When used in the RTP
          congestion circuit breaker, this is approximated as described in
          Section 4.3.

    Perkins & Singh              Standards Track                    [Page 7]

    RFC 8083                  RTP Circuit Breakers                March 2017

       o  t_RTO is the retransmission timeout value that would be used by a
          TCP connection over a particular path, in seconds.  This MUST be
          approximated using t_RTO = 4 * Tr when used as part of the RTP
          congestion circuit breaker.

       o  b is the number of packets that are acknowledged by a single TCP
          acknowledgement.  Following [RFC5348], it is RECOMMENDED that the
          value b = 1 is used as part of the RTP congestion circuit breaker.

    4.  RTP Circuit Breakers for Systems Using the RTP/AVP Profile

       The feedback mechanisms defined in [RFC3550] and available under the
       RTP/AVP profile [RFC3551] are the minimum that can be assumed for a
       baseline circuit breaker mechanism that is suitable for all unicast
       applications of RTP.  Accordingly, for an RTP circuit breaker to be
       useful, it needs to be able to detect that an RTP flow is causing
       excessive congestion using only basic RTCP features without needing
       RTCP XR feedback or the RTP/AVPF profile for rapid RTCP reports.

       RTCP is a fundamental part of the RTP protocol, and the mechanisms
       described here rely on the implementation of RTCP.  Implementations
       that claim to support RTP, but that do not implement RTCP, will be
       unable to use the circuit breaker mechanisms described in this memo.
       Such implementations SHOULD NOT be used on networks that might be
       subject to congestion unless equivalent mechanisms are defined using
       some non-RTCP feedback channel to report congestion and signal
       circuit breaker conditions.

       The RTCP timeout circuit breaker (Section 4.1) will trigger if an
       implementation of this memo attempts to interwork with an endpoint
       that does not support RTCP.  Implementations that sometimes need to
       interwork with endpoints that do not support RTCP need to disable the
       RTP circuit breakers if they don't receive some confirmation via
       signaling that the remote endpoint implements RTCP (the presence of a
       Session Description Protocol (SDP) "a=rtcp:" attribute in an answer
       might be such an indication).  The RTP circuit breaker SHOULD NOT be
       disabled on networks that might be subject to congestion unless
       equivalent mechanisms are defined using some non-RTCP feedback
       channel to report congestion and signal circuit breaker conditions
       [RFC8084].

       Three potential congestion signals are available from the basic RTCP
       SR/RR packets and are reported for each SSRC in the RTP session:

       1.  The sender can estimate the network round-trip time once per RTCP
           reporting interval based on the contents and timing of RTCP SR
           and RR packets.

    Perkins & Singh              Standards Track                    [Page 8]

    RFC 8083                  RTP Circuit Breakers                March 2017

       2.  Receivers report a jitter estimate (the statistical variance of
           the RTP data packet inter-arrival time) calculated over the RTCP
           reporting interval.  Due to the nature of the jitter calculation
           (Section 6.4.4. of [RFC3550]), the jitter is only meaningful for
           RTP flows that send a single data packet for each RTP timestamp
           value (i.e., audio flows, or video flows where each packet
           comprises one video frame).

       3.  Receivers report the fraction of RTP data packets lost during the
           RTCP reporting interval and the cumulative number of RTP packets
           lost over the entire RTP session.

       These congestion signals limit the possible circuit breakers since
       they give only limited visibility into the behavior of the network.

       RTT estimates are widely used in congestion control algorithms as a
       proxy for queuing delay measures in delay-based congestion control or
       to determine connection timeouts.  RTT estimates derived from RTCP SR
       and RR packets sent according to the RTP/AVP timing rules are too
       infrequent to be useful for congestion control and don't give enough
       information to distinguish a delay change due to routing updates from
       queuing delay caused by congestion.  Accordingly, we cannot use the
       RTT estimate alone as an RTP circuit breaker.

       Increased jitter can be a signal of transient network congestion, but
       in the highly aggregated form reported in RTCP RR packets, it offers
       insufficient information to estimate the extent or persistence of
       congestion.  Jitter reports are a useful early warning of potential
       network congestion but provide an insufficiently strong signal to be
       used as a circuit breaker.

       The remaining congestion signals are the packet loss fraction and the
       cumulative number of packets lost.  If considered carefully, and over
       an appropriate time frame to distinguish transient problems from long
       term issues [RFC8084], these can be effective indicators that
       persistent excessive congestion is occurring in networks where packet
       loss is primarily due to queue overflows, although loss caused by
       non-congestive packet corruption can distort the result in some
       networks.  TCP congestion control [RFC5681] intentionally tries to
       fill the router queues and uses the resulting packet loss as
       congestion feedback.  An RTP flow competing with TCP traffic will
       therefore expect to see a non-zero packet loss fraction, and some
       variation in queuing latency, in normal operation when sharing a path
       with other flows, which needs to be accounted for when determining
       the circuit breaker threshold [RFC8084].  This behavior of TCP is
       reflected in the congestion circuit breaker below and will affect the
       design of any RTP congestion control protocol.

    Perkins & Singh              Standards Track                    [Page 9]

    RFC 8083                  RTP Circuit Breakers                March 2017

       Two packet loss regimes can be observed: 1) RTCP RR packets show a
       non-zero packet loss fraction while the extended highest sequence
       number received continues to increment; and 2) RR packets show a loss
       fraction of zero, but the extended highest sequence number received
       does not increment even though the sender has been transmitting RTP
       data packets.  The former corresponds to the TCP congestion avoidance
       state and indicates a congested path that is still delivering data;
       the latter corresponds to a TCP timeout and is most likely due to a
       path failure.  A third condition is that data is being sent but no
       RTCP feedback is received at all, corresponding to a failure of the
       reverse path.  We derive circuit breaker conditions for these loss
       regimes in the following.

    4.1.  RTP/AVP Circuit Breaker #1: RTCP Timeout

       An RTCP timeout can occur when RTP data packets are being sent, but
       there are no RTCP reports returned from the receiver.  This is either
       due to a failure of the receiver to send RTCP reports or a failure of
       the return path that is preventing those RTCP reporting from being
       delivered.  In either case, it is not safe to continue transmission
       since the sender has no way of knowing if it is causing congestion.

       An RTP sender that has not received any RTCP SR or RTCP RR packets
       reporting on the SSRC it is using, for a time period of at least
       three times its deterministic RTCP reporting interval, Td (where Td
       is calculated without the randomization factor and using the fixed
       minimum interval of Tmin=5 seconds), SHOULD cease transmission (see
       Section 4.5).  The rationale for this choice of timeout is as
       described in Section 6.2 of [RFC3550] ("so that implementations which
       do not use the reduced value for transmitting RTCP packets are not
       timed out by other participants prematurely") and has been updated by
       Section 6.1.4 of [RFC8108] to account for the use of the RTP/AVPF
       profile [RFC4585] or the RTP/SAVPF profile [RFC5124].

       To reduce the risk of premature timeout, implementations SHOULD NOT
       configure the RTCP bandwidth such that Td is larger than 5 seconds.
       Similarly, implementations that use the RTP/AVPF profile [RFC4585] or
       the RTP/SAVPF profile [RFC5124] SHOULD NOT configure T_rr_interval to
       values larger than 4 seconds (the reduced limit for T_rr_interval
       follows Section 6.1.3 of [RFC8108]).

       The choice of three RTCP reporting intervals as the timeout is made
       following Section 6.3.5 of RFC 3550 [RFC3550].  This specifies that
       participants in an RTP session will timeout and remove an RTP sender
       from the list of active RTP senders if no RTP data packets have been
       received from that RTP sender within the last two RTCP reporting
       intervals.  Using a timeout of three RTCP reporting intervals is
       therefore large enough that the other participants will have timed

    Perkins & Singh              Standards Track                   [Page 10]

    RFC 8083                  RTP Circuit Breakers                March 2017

       out the sender if a network problem stops the data packets it is
       sending from reaching the receivers, even allowing for loss of some
       RTCP packets.

       If a sender is transmitting a large number of RTP media streams, such
       that the corresponding RTCP SR or RR packets are too large to fit
       into the network MTU, the receiver will generate RTCP SR or RR
       packets in a round-robin manner.  In this case, the sender SHOULD
       treat receipt of an RTCP SR or RR packet corresponding to any SSRC it
       sent on the same 5-tuple of source and destination IP address, port,
       and protocol as an indication that the receiver and return path are
       working and thus preventing the RTCP timeout circuit breaker from
       triggering.

    4.2.  RTP/AVP Circuit Breaker #2: Media Timeout

       If RTP data packets are being sent but the RTCP SR or RR packets
       reporting on that SSRC indicate a non-increasing extended highest
       sequence number received, this is an indication that those RTP data
       packets are not reaching the receiver.  This could be a short-term
       issue affecting only a few RTP packets, perhaps caused by a slow-to-
       open firewall or a transient connectivity problem, but if the issue
       persists, it is a sign of a more ongoing and significant problem (a
       "media timeout").

       The time needed to declare a media timeout depends on the parameters
       Tdr, Tr, Tf, and on the non-reporting threshold k.  The value of k is
       chosen so that when Tdr is large compared to Tr and Tf, receipt of at
       least k RTCP reports with non-increasing extended highest sequence
       number received gives reasonable assurance that the forward path has
       failed and that the RTP data packets have not been lost by chance.
       The RECOMMENDED value for k is 5 reports.

       When Tdr < Tf, then RTP data packets are being sent at a rate less
       than one per RTCP reporting interval of the receiver, so the extended
       highest sequence number received can be expected to be non-increasing
       for some receiver RTCP reporting intervals.  Similarly, when
       Tdr < Tr, some receiver RTCP reporting intervals might pass before
       the RTP data packets arrive at the receiver, also leading to reports
       where the extended highest sequence number received is non-
       increasing.  Both issues require the media timeout interval to be
       scaled relative to the threshold, k.

       The media timeout RTP circuit breaker is therefore as follows.  When
       starting sending, calculate MEDIA_TIMEOUT using:

          MEDIA_TIMEOUT = ceil(k * max(Tf, Tr, Tdr) / Tdr)

    Perkins & Singh              Standards Track                   [Page 11]

    RFC 8083                  RTP Circuit Breakers                March 2017

       When a sender receives an RTCP packet that indicates reception of the
       media it has been sending, then it cancels the media timeout circuit
       breaker.  If it is still sending, then it MUST calculate a new value
       for MEDIA_TIMEOUT and set a new media timeout circuit breaker.

       If a sender receives an RTCP packet indicating that its media was not
       received, it MUST calculate a new value for MEDIA_TIMEOUT.  If the
       new value is larger than the previous, it replaces MEDIA_TIMEOUT with
       the new value, extending the media timeout circuit breaker;
       otherwise, it keeps the original value of MEDIA_TIMEOUT.  This
       process is known as reconsidering the media timeout circuit breaker.

       If MEDIA_TIMEOUT consecutive RTCP packets are received indicating
       that the media being sent was not received, and the media timeout
       circuit breaker has not been canceled, then the media timeout circuit
       breaker triggers.  When the media timeout circuit breaker triggers,
       the sender SHOULD cease transmission (see Section 4.5).

       When stopping sending an RTP stream, a sender MUST cancel the
       corresponding media timeout circuit breaker.

    4.3.  RTP/AVP Circuit Breaker #3: Congestion

       If RTP data packets are being sent and the corresponding RTCP SR or
       RR packets show non-zero packet loss fraction and increasing extended
       highest sequence number received, then those RTP data packets are
       arriving at the receiver, but some degree of congestion is occurring.
       The RTP/AVP profile [RFC3551] states that:

          If best-effort service is being used, RTP receivers SHOULD monitor
          packet loss to ensure that the packet loss rate is within
          acceptable parameters.  Packet loss is considered acceptable if a
          TCP flow across the same network path and experiencing the same
          network conditions would achieve an average throughput, measured
          on a reasonable timescale, that is not less than [the throughput]
          the RTP flow is achieving.  This condition can be satisfied by
          implementing congestion control mechanisms to adapt the
          transmission rate (or the number of layers subscribed for a
          layered multicast session), or by arranging for a receiver to
          leave the session if the loss rate is unacceptably high.

          The comparison to TCP cannot be specified exactly, but is intended
          as an "order-of-magnitude" comparison in timescale and throughput.
          The timescale on which TCP throughput is measured is the round-
          trip time of the connection.  In essence, this requirement states
          that it is not acceptable to deploy an application (using RTP or

    Perkins & Singh              Standards Track                   [Page 12]

    RFC 8083                  RTP Circuit Breakers                March 2017

          any other transport protocol) on the best-effort Internet which
          consumes bandwidth arbitrarily and does not compete fairly with
          TCP within an order of magnitude.

       The phase "order of magnitude" in the above means within a factor of
       ten, approximately.  In order to implement this, it is necessary to
       estimate the throughput a bulk TCP connection would achieve over the
       path.  For a long-lived TCP Reno connection, it has been shown that
       the TCP throughput, X, in bytes per second, can be estimated as
       follows [Padhye]:

                                      s
          X = -------------------------------------------------------------
              Tr*sqrt(2*b*p/3)+(t_RTO * (3*sqrt(3*b*p/8) * p * (1+32*p*p)))

       This is the same approach to estimated TCP throughput that is used in
       [RFC5348].  Under conditions of low packet loss, the second term on
       the denominator is small, so this formula can be approximated with
       reasonable accuracy as follows [Mathis]:

                    s
          X = ----------------
              Tr*sqrt(2*b*p/3)

       It is RECOMMENDED that this simplified throughput equation be used
       since the reduction in accuracy is small, and it is much simpler to
       calculate than the full equation.  Measurements have shown that the
       simplified TCP throughput equation is effective as an RTP circuit
       breaker for multimedia flows sent to hosts on residential networks
       using Asymmetric Digital Subscriber Line (ADSL) and cable modem links
       [Singh].  The data shows that the full TCP throughput equation tends
       to be more sensitive to packet loss and triggers the RTP circuit
       breaker earlier than the simplified equation.  Implementations that
       desire this extra sensitivity MAY use the full TCP throughput
       equation in the RTP circuit breaker.  Initial measurements in LTE
       networks have shown that the extra sensitivity is helpful in that
       environment, with the full TCP throughput equation giving a more
       balanced circuit breaker response than the simplified TCP equation
       [Sarker]; other networks might see similar behavior.

       No matter what TCP throughput equation is chosen, two parameters need
       to be estimated and reported to the sender in order to calculate the
       throughput: the round-trip time, Tr, and the loss event rate, p (the
       packet size, s, is known to the sender).  The round-trip time can be
       estimated from RTCP SR and RR packets.  This is done too infrequently
       for accurate statistics but is the best that can be done with the
       standard RTCP mechanisms.

    Perkins & Singh              Standards Track                   [Page 13]

    RFC 8083                  RTP Circuit Breakers                March 2017

       Report blocks in RTCP SR or RR packets contain the packet loss
       fraction, rather than the loss event rate, so p cannot be reported
       (TCP typically treats the loss of multiple packets within a single
       RTT as one loss event, but RTCP RR packets report the overall
       fraction of packets lost and do not report when the packet losses
       occurred).  Using the loss fraction in place of the loss event rate
       can overestimate the loss.  We believe that this overestimate will
       not be significant given that we are only interested in order of
       magnitude comparison (Section 3.2.1 of [Floyd] shows that the
       difference is small for steady-state conditions and random loss, but
       using the loss fraction is more conservative in the case of bursty
       loss).

       The congestion circuit breaker is therefore as follows.  When a
       sender that is transmitting at least one RTP packet every max(Tdr,
       Tr) seconds receives an RTCP SR or RR packet that contains a report
       block for an SSRC it is using, the sender MUST record the value of
       the fraction lost field from the report block, and the time since the
       last report block was received, for that SSRC.  If more than
       CB_INTERVAL (see below) report blocks have been received for that
       SSRC, the sender MUST calculate the average fraction lost over the
       last CB_INTERVAL reporting intervals and then estimate the TCP
       throughput that would be achieved over the path using the chosen TCP
       throughput equation and the measured values of the round-trip time,
       Tr, the loss event rate, p (approximated by the average fraction
       lost, as is described below), and the packet size, s.  The estimate
       of the TCP throughput, X, is then compared with the actual sending
       rate of the RTP stream.  If the actual sending rate of the RTP stream
       is more than 10 * X, then the congestion circuit breaker is
       triggered.

       The average fraction lost is calculated based on the sum (over the
       last CB_INTERVAL reporting intervals) of the fraction lost in each
       reporting interval that is then multiplied by the duration of the
       corresponding reporting interval and then divided by the total
       duration of the last CB_INTERVAL reporting intervals.  The
       CB_INTERVAL parameter is set to:

          CB_INTERVAL =
             ceil(3*min(max(10*G*Tf, 10*Tr, 3*Tdr), max(15, 3*Td))/(3*Tdr))

       The parameters that feed into CB_INTERVAL are chosen to give the
       congestion control algorithm time to react to congestion.  They give
       at least three RTCP reports, ten round trip times, and ten groups of
       frames to adjust the rate to reduce the congestion to a reasonable
       level.  It is expected that a responsive congestion control algorithm

    Perkins & Singh              Standards Track                   [Page 14]

    RFC 8083                  RTP Circuit Breakers                March 2017

       will begin to respond with the next group of frames after it receives
       indication of congestion, so CB_INTERVAL ought to be a much longer
       interval than the congestion response.

       If the RTP/AVPF profile [RFC4585] or the RTP/SAVPF [RFC5124] is used,
       and the T_rr_interval parameter is used to reduce the frequency of
       regular RTCP reports, then the value of Tdr in the above expression
       for the CB_INTERVAL parameter MUST be replaced by max(T_rr_interval,
       Tdr).

       The CB_INTERVAL parameter is calculated on joining the session, and
       recalculated on receipt of each RTCP packet, after checking whether
       the media timeout circuit breaker or the congestion circuit breaker
       has been triggered.

       To ensure a timely response to persistent congestion, implementations
       SHOULD NOT configure the RTCP bandwidth such that Tdr is larger than
       5 seconds.  Similarly, implementations that use the RTP/AVPF profile
       [RFC4585] or the RTP/SAVPF profile [RFC5124] SHOULD NOT configure
       T_rr_interval to values larger than 4 seconds (the reduced limit for
       T_rr_interval follows Section 6.1.3 of [RFC8108]).

       The rationale for enforcing a minimum sending rate below which the
       congestion circuit breaker will not trigger is to avoid spurious
       circuit breaker triggers when the number of packets sent per RTCP
       reporting interval is small, and hence, the fraction lost samples are
       subject to measurement artifacts.  The bound of at least one packet
       every max(Tdr, Tr) seconds is derived from the one packet per RTT
       minimum sending rate of TCP [RFC8085], which is adapted for use with
       RTP where the RTCP reporting interval is decoupled from the network
       RTT.

       When the congestion circuit breaker is triggered, the sender SHOULD
       cease transmission (see Section 4.5).  However, if the sender is able
       to reduce its sending rate by a factor of (approximately) ten, then
       it MAY first reduce its sending rate by this factor (or some larger
       amount) to see if that resolves the congestion.  If the sending rate
       is reduced in this way and the congestion circuit breaker triggers
       again after the next CB_INTERVAL RTCP reporting intervals, the sender
       MUST then cease transmission.  An example of such a rate reduction
       might be a video conferencing system that backs off to sending audio
       only before completely dropping the call.  If such a reduction in
       sending rate resolves the congestion problem, the sender MAY
       gradually increase the rate at which it sends data after a reasonable
       amount of time has passed, provided it takes care not to cause the
       problem to recur ("reasonable" is intentionally not defined here
       since it depends on the application, media codec, and congestion
       control algorithm).

    Perkins & Singh              Standards Track                   [Page 15]

    RFC 8083                  RTP Circuit Breakers                March 2017

       The RTCP reporting interval of the media sender does not affect how
       quickly the congestion circuit breaker can trigger.  The timing is
       based on the RTCP reporting interval of the receiver that generates
       the SR/RR packets from which the loss rate and RTT estimate are
       derived (note that RTCP requires all participants in a session to
       have similar reporting intervals, else the participant timeout rules
       in [RFC3550] will not work, so this interval is likely similar to
       that of the sender).  If the incoming RTCP SR or RR packets are using
       a reduced minimum RTCP reporting interval (as specified in
       Section 6.2 of RFC 3550 [RFC3550] or the RTP/AVPF profile [RFC4585]),
       then that reduced RTCP reporting interval is used when determining if
       the circuit breaker is triggered.

       If there are more media streams that can be reported in a single RTCP
       SR or RR packet, or if the size of a complete RTCP SR or RR packet
       exceeds the network MTU, then the receiver will report on a subset of
       sources in each reporting interval with the subsets selected round-
       robin across multiple intervals so that all sources are eventually
       reported [RFC3550].  When generating such round-robin RTCP reports,
       priority SHOULD be given to reports on sources that have high packet
       loss rates to ensure that senders are aware of network congestion
       they are causing (this is an update to [RFC3550]).

    4.4.  RTP/AVP Circuit Breaker #4: Media Usability

       Applications that use RTP are generally tolerant to some amount of
       packet loss.  How much packet loss can be tolerated will depend on
       the application, media codec, and the amount of error correction and
       packet loss concealment that is applied.  There is an upper bound on
       the amount of loss that can be corrected, however, beyond which the
       media becomes unusable.  Similarly, many applications have some upper
       bound on the media capture to play-out latency that can be tolerated
       before the application becomes unusable.  The latency bound will
       depend on the application, but typical values can range from the
       order of a few hundred milliseconds for voice telephony and
       interactive conferencing applications up to several seconds for some
       video-on-demand systems.

       As a final circuit breaker, RTP senders SHOULD monitor the reported
       packet loss and delay to estimate whether the media is likely to be
       suitable for the intended purpose.  If the packet loss rate and/or
       latency is such that the media has become unusable and has remained
       unusable for a significant time period, then the application SHOULD
       cease transmission.  Similarly, receivers SHOULD monitor the quality
       of the media they receive, and if the quality is unusable for a
       significant time period, they SHOULD terminate the session.  This
       memo intentionally does not define a bound on the packet loss rate or
       latency that will result in unusable media, as these are highly

    Perkins & Singh              Standards Track                   [Page 16]

    RFC 8083                  RTP Circuit Breakers                March 2017

       application dependent.  Similarly, the time period that is considered
       significant is application dependent but is likely on the order of
       seconds, or tens of seconds.

       Sending media that suffers from such high packet loss or latency that
       it is unusable at the receiver is both wasteful of resources and is
       of no benefit to the user of the application.  It also is highly
       likely to be congesting the network and disrupting other
       applications.  As such, the congestion circuit breaker will almost
       certainly trigger to stop flows where the media would be unusable due
       to high packet loss or latency.  However, in pathological scenarios
       where the congestion circuit breaker does not stop the flow, it is
       desirable to prevent the application sending unnecessary traffic that
       might disrupt other uses of the network.  The role of the media
       usability circuit breaker is to protect the network in such cases.

    4.5.  Ceasing Transmission

       What it means to cease transmission depends on the application.  This
       could mean stopping a single RTP flow or it could mean that multiple
       bundled RTP flows are stopped.  The intention is that the application
       will stop sending RTP data packets on a particular 5-tuple (transport
       protocol, source and destination ports, source and destination IP
       addresses) until whatever network problem that triggered the RTP
       circuit breaker has dissipated.  RTP flows halted by the circuit
       breaker SHOULD NOT be restarted automatically unless the sender has
       received information that the congestion has dissipated or can
       reasonably be expected to have dissipated.  What could trigger this
       expectation is necessarily application dependent, but could be, for
       example, an indication that a competing flow has finished and freed
       up some capacity, or for an application running on a mobile device it
       could indicate that the device moved to a new location so the flow
       would traverse a different path if it were restarted.  Ideally, a
       human user will be involved in the decision to try to restart the
       flow since that user will eventually give up if the flows repeatedly
       trigger the circuit breaker.  This will help avoid problems with
       automatic redial systems from congesting the network.

       It is recognized that the RTP implementation in some systems might
       not be able to determine if a flow setup request was initiated by a
       human user or automatically by some scripted higher-level component
       of the system.  These implementations MUST rate limit attempts to
       restart a flow on the same 5-tuple as used by a flow that triggered
       the circuit breaker so that the reaction to a triggered circuit
       breaker lasts for at least the triggering interval [RFC8084].

    Perkins & Singh              Standards Track                   [Page 17]

    RFC 8083                  RTP Circuit Breakers                March 2017

       The RTP circuit breaker will only trigger, and cease transmission,
       for media flows subject to long-term persistent congestion.  Such
       flows are likely to have poor quality and usability for some time
       before the circuit breaker triggers.  Implementations can monitor
       RTCP Receiver Report blocks being returned for their media flows and
       might find it beneficial to use this information to provide a user
       interface cue that problems are occurring in advance of the circuit
       breaker triggering.

    5.  RTP Circuit Breakers and the RTP/AVPF and RTP/SAVPF Profiles

       Use of the Extended RTP Profile for RTCP-based Feedback (RTP/AVPF)
       [RFC4585] allows receivers to send early RTCP reports, in some cases,
       to inform the sender about particular events in the media stream.
       There are several use cases for such early RTCP reports, including
       providing rapid feedback to a sender about the onset of congestion.
       The RTP/SAVPF Profile [RFC5124] is a secure variant of the RTP/AVPF
       profile that is treated the same in the context of the RTP circuit
       breaker.  These feedback profiles are often used with non-compound
       RTCP reports [RFC5506] to reduce the reporting overhead.

       Receiving rapid feedback about congestion events potentially allows
       congestion control algorithms to be more responsive and to better
       adapt the media transmission to the limitations of the network.  It
       is expected that many RTP congestion control algorithms will adopt
       the RTP/AVPF profile or the RTP/SAVPF profile for this reason and
       thus define new transport-layer feedback reports that suit their
       requirements.  Since these reports are not yet defined, and likely
       very specific to the details of the congestion control algorithm
       chosen, they cannot be used as part of the generic RTP circuit
       breaker.

       Reduced-size RTCP reports sent under the RTP/AVPF early feedback
       rules that do not contain an RTCP SR or RR packet MUST be ignored by
       the congestion circuit breaker (they do not contain the information
       needed by the congestion circuit breaker algorithm) but MUST be
       counted as received packets for the RTCP timeout circuit breaker.
       Reduced-size RTCP reports sent under the RTP/AVPF early feedback
       rules that contain RTCP SR or RR packets MUST be processed by the
       congestion circuit breaker as if they were sent as regular RTCP
       reports and counted towards the circuit breaker conditions specified
       in Section 4 of this memo.  This will potentially make the RTP
       circuit breaker trigger earlier than it would if the RTP/AVPF profile
       was not used.

       When using ECN with RTP (see Section 7), early RTCP feedback packets
       can contain ECN feedback reports.  The count of ECN-CE-marked packets
       contained in those ECN feedback reports is counted towards the number

    Perkins & Singh              Standards Track                   [Page 18]

    RFC 8083                  RTP Circuit Breakers                March 2017

       of lost packets reported if the ECN Feedback Report is sent in a
       compound RTCP packet along with an RTCP SR/RR report packet.  Reports
       of ECN-CE packets sent as reduced-size RTCP ECN feedback packets
       without an RTCP SR/RR packet MUST be ignored.

       These rules are intended to allow the use of low-overhead RTP/AVPF
       feedback for generic NACK messages without triggering the RTP circuit
       breaker.  This is expected to make such feedback suitable for RTP
       congestion control algorithms that need to quickly report loss events
       in between regular RTCP reports.  The reaction to reduced-size RTCP
       SR/RR packets is to allow such algorithms to send feedback that can
       trigger the circuit breaker when desired.

       The RTP/AVPF and RTP/SAVPF profiles include the T_rr_interval
       parameter that can be used to adjust the regular RTCP reporting
       interval.  The use of the T_rr_interval parameter changes the
       behavior of the RTP circuit breaker, as described in Section 4.

    6.  Impact of RTCP Extended Reports (XR)

       RTCP Extended Report (XR) blocks provide additional reception quality
       metrics, but do not change the RTCP timing rules.  Some of the RTCP
       XR blocks provide information that might be useful for congestion
       control purposes, others provide non-congestion-related metrics.
       With the exception of RTCP XR ECN Summary Reports (see Section 7),
       the presence of RTCP XR blocks in a compound RTCP packet does not
       affect the RTP circuit breaker algorithm.  For consistency and ease
       of implementation, only the receiver report blocks contained in RTCP
       SR packets, RTCP RR packets, or RTCP XR ECN Summary Report packets
       are used by the RTP circuit breaker algorithm.

    7.  Impact of Explicit Congestion Notification (ECN)

       The use of ECN for RTP flows does not affect the RTCP timeout circuit
       breaker (Section 4.1) or the media timeout circuit breaker
       (Section 4.2) since these are both connectivity checks that simply
       determinate if any packets are being received.

       At the time of this writing, there's no consensus on how the receipt
       of ECN feedback will impact the congestion circuit breaker
       (Section 4.3) or indeed whether the congestion circuit breaker ought
       to take ECN feedback into account.  A future replacement of this memo
       is expected to provide guidance for implementers.

       For the media usability circuit breaker (Section 4.4), ECN-CE-marked
       packets arrive at the receiver, and if they arrive in time, they will
       be decoded and rendered as normal.  Accordingly, receipt of such
       packets ought not affect the usability of the media, and the arrival

    Perkins & Singh              Standards Track                   [Page 19]

    RFC 8083                  RTP Circuit Breakers                March 2017

       of RTCP feedback indicating their receipt is not expected to impact
       the operation of the media usability circuit breaker.

    8.  Impact of Bundled Media and Layered Coding

       The RTP circuit breaker operates on a per-RTP session basis.  An RTP
       sender that participates in several RTP sessions MUST treat each RTP
       session independently with regards to the RTP circuit breaker.

       An RTP sender can generate several media streams within a single RTP
       session, with each stream using a different SSRC.  This can happen if
       bundled media are in use when using simulcast or when using layered
       media coding.  By default, each SSRC will be treated independently by
       the RTP circuit breaker.  However, the sender MAY choose to treat the
       flows (or a subset thereof) as a group such that a circuit breaker
       trigger for one flow applies to the group of flows as a whole and
       either causes the entire group to cease transmission or causes the
       sending rate of the group to reduce by a factor of ten, depending on
       the RTP circuit breaker triggered.  Grouping flows in this way is
       expected to be especially useful for layered flows sent using
       multiple SSRCs as it allows the layered flow to react as a whole,
       thus ceasing transmission on the enhancement layers first to reduce
       sending rate, if necessary, rather than treating each layer
       independently.  Care needs to be taken if the different media streams
       sent on a single transport-layer flow use different Differentiated
       Services Code Point (DSCP) values [RFC7657] [WebRTC-QoS] since
       congestion could be experienced differently depending on the DSCP
       marking.  Accordingly, RTP media streams with different DSCP values
       SHOULD NOT be considered as a group when evaluating the RTP circuit
       breaker conditions.

    9.  Security Considerations

       The security considerations of [RFC3550] apply.

       If the RTP/AVPF profile is used to provide rapid RTCP feedback, the
       security considerations of [RFC4585] apply.  If ECN feedback for RTP
       over UDP/IP is used, the security considerations of [RFC6679] apply.

       If non-authenticated RTCP reports are used, an on-path attacker can
       trivially generate fake RTCP packets that indicate high packet loss
       rates and thus cause the circuit breaker to trigger and disrupt an
       RTP session.  This is somewhat more difficult for an off-path
       attacker due to the need to guess the randomly chosen RTP SSRC value
       and the RTP sequence number.  This attack can be avoided if RTCP
       packets are authenticated; authentication options are discussed in
       [RFC7201].

    Perkins & Singh              Standards Track                   [Page 20]

    RFC 8083                  RTP Circuit Breakers                March 2017

       Timely operation of the RTP circuit breaker depends on the choice of
       RTCP reporting interval.  If the receiver has a reporting interval
       that is overly long, then the responsiveness of the circuit breaker
       decreases.  In the limit, the RTP circuit breaker can be disabled for
       all practical purposes by configuring an RTCP reporting interval that
       has a duration of many minutes.  This issue is not specific to the
       circuit breaker: long RTCP reporting intervals also prevent reception
       quality reports, feedback messages, codec control messages, etc.,
       from being used.  Implementations are expected to impose an upper
       limit on the RTCP reporting interval they are willing to negotiate
       (based on the session bandwidth and RTCP bandwidth fraction) when
       using the RTP circuit breaker, as discussed in Section 4.3.

    10.  References

    10.1.  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,
                  <http://www.rfc-editor.org/info/rfc2119>.

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

       [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
                  Video Conferences with Minimal Control", STD 65, RFC 3551,
                  DOI 10.17487/RFC3551, July 2003,
                  <http://www.rfc-editor.org/info/rfc3551>.

       [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,
                  <http://www.rfc-editor.org/info/rfc3611>.

       [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
                  "Extended RTP Profile for Real-time Transport Control
                  Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
                  DOI 10.17487/RFC4585, July 2006,
                  <http://www.rfc-editor.org/info/rfc4585>.

       [RFC5348]  Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
                  Friendly Rate Control (TFRC): Protocol Specification",
                  RFC 5348, DOI 10.17487/RFC5348, September 2008,
                  <http://www.rfc-editor.org/info/rfc5348>.

    Perkins & Singh              Standards Track                   [Page 21]

    RFC 8083                  RTP Circuit Breakers                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, <http://www.rfc-editor.org/info/rfc6679>.

    10.2.  Informative References

       [Floyd]    Floyd, S., Handley, M., Padhye, J., and J. Widmer,
                  "Equation-Based Congestion Control for Unicast
                  Applications", ACM SIGCOMM Computer Communication
                  Review, Volume 30, Issue 4, pages 43-56,
                  DOI 10.1145/347059.347397, August 2000.

       [Mathis]   Mathis, M., Semke, J., Mahdavi, J., and T. Ott, "The
                  Macroscopic Behavior of the TCP Congestion Avoidance
                  Algorithm", ACM SIGCOMM Computer Communication
                  Review, Volume 27, Issue 3, pages 67-82,
                  DOI 10.1145/263932.264023, July 1997.

       [Padhye]   Padhye, J., Firoiu, V., Towsley, D., and J. Kurose,
                  "Modeling TCP Throughput: A Simple Model and its Empirical
                  Validation", ACM SIGCOMM Computer Communication
                  Review Volume 30, Issue 4, pages 303-314,
                  DOI 10.1145/285237.285291, August 1998.

       [RFC2862]  Civanlar, M. and G. Cash, "RTP Payload Format for Real-
                  Time Pointers", RFC 2862, DOI 10.17487/RFC2862, June 2000,
                  <http://www.rfc-editor.org/info/rfc2862>.

       [RFC3168]  Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
                  of Explicit Congestion Notification (ECN) to IP",
                  RFC 3168, DOI 10.17487/RFC3168, September 2001,
                  <http://www.rfc-editor.org/info/rfc3168>.

       [RFC4103]  Hellstrom, G. and P. Jones, "RTP Payload for Text
                  Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005,
                  <http://www.rfc-editor.org/info/rfc4103>.

       [RFC5104]  Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
                  "Codec Control Messages in the RTP Audio-Visual Profile
                  with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
                  February 2008, <http://www.rfc-editor.org/info/rfc5104>.

       [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
                  Real-time Transport Control Protocol (RTCP)-Based Feedback
                  (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
                  2008, <http://www.rfc-editor.org/info/rfc5124>.

    Perkins & Singh              Standards Track                   [Page 22]

    RFC 8083                  RTP Circuit Breakers                March 2017

       [RFC5450]  Singer, D. and H. Desineni, "Transmission Time Offsets in
                  RTP Streams", RFC 5450, DOI 10.17487/RFC5450, March 2009,
                  <http://www.rfc-editor.org/info/rfc5450>.

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

       [RFC5681]  Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
                  Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,
                  <http://www.rfc-editor.org/info/rfc5681>.

       [RFC6798]  Clark, A. and Q. Wu, "RTP Control Protocol (RTCP) Extended
                  Report (XR) Block for Packet Delay Variation Metric
                  Reporting", RFC 6798, DOI 10.17487/RFC6798, November 2012,
                  <http://www.rfc-editor.org/info/rfc6798>.

       [RFC6843]  Clark, A., Gross, K., and Q. Wu, "RTP Control Protocol
                  (RTCP) Extended Report (XR) Block for Delay Metric
                  Reporting", RFC 6843, DOI 10.17487/RFC6843, January 2013,
                  <http://www.rfc-editor.org/info/rfc6843>.

       [RFC6958]  Clark, A., Zhang, S., Zhao, J., and Q. Wu, Ed., "RTP
                  Control Protocol (RTCP) Extended Report (XR) Block for
                  Burst/Gap Loss Metric Reporting", RFC 6958,
                  DOI 10.17487/RFC6958, May 2013,
                  <http://www.rfc-editor.org/info/rfc6958>.

       [RFC7002]  Clark, A., Zorn, G., and Q. Wu, "RTP Control Protocol
                  (RTCP) Extended Report (XR) Block for Discard Count Metric
                  Reporting", RFC 7002, DOI 10.17487/RFC7002, September
                  2013, <http://www.rfc-editor.org/info/rfc7002>.

       [RFC7003]  Clark, A., Huang, R., and Q. Wu, Ed., "RTP Control
                  Protocol (RTCP) Extended Report (XR) Block for Burst/Gap
                  Discard Metric Reporting", RFC 7003, DOI 10.17487/RFC7003,
                  September 2013, <http://www.rfc-editor.org/info/rfc7003>.

       [RFC7097]  Ott, J., Singh, V., Ed., and I. Curcio, "RTP Control
                  Protocol (RTCP) Extended Report (XR) for RLE of Discarded
                  Packets", RFC 7097, DOI 10.17487/RFC7097, January 2014,
                  <http://www.rfc-editor.org/info/rfc7097>.

       [RFC7201]  Westerlund, M. and C. Perkins, "Options for Securing RTP
                  Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,
                  <http://www.rfc-editor.org/info/rfc7201>.

    Perkins & Singh              Standards Track                   [Page 23]

    RFC 8083                  RTP Circuit Breakers                March 2017

       [RFC7657]  Black, D., Ed. and P. Jones, "Differentiated Services
                  (Diffserv) and Real-Time Communication", RFC 7657,
                  DOI 10.17487/RFC7657, November 2015,
                  <http://www.rfc-editor.org/info/rfc7657>.

       [RFC7713]  Mathis, M. and B. Briscoe, "Congestion Exposure (ConEx)
                  Concepts, Abstract Mechanism, and Requirements", RFC 7713,
                  DOI 10.17487/RFC7713, December 2015,
                  <http://www.rfc-editor.org/info/rfc7713>.

       [RFC8084]  Fairhurst, G., "Network Transport Circuit Breakers",
                  BCP 208, RFC 8084, DOI 10.17487/RFC8084, March 2017,
                  <http://www.rfc-editor.org/info/rfc8084>.

       [RFC8085]  Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage
                  Guidelines", BCP 145, RFC 8085, DOI 10.17487/RFC8085,
                  March 2017, <http://www.rfc-editor.org/info/rfc8085>.

       [RFC8108]  Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
                  "Sending Multiple RTP Streams in a Single RTP Session",
                  RFC 8108, DOI 10.17487/RFC8108, March 2017,
                  <http://www.rfc-editor.org/info/rfc8108>.

       [Sarker]   Sarker, Z., Singh, V., and C. Perkins, "An Evaluation of
                  RTP Circuit Breaker Performance on LTE Networks",
                  Proceedings of the IEEE INFOCOM Workshop on Communication
                  and Networking Techniques for Contemporary Video,
                  DOI 10.1109/INFCOMW.2014.6849240, April 2014.

       [Singh]    Singh, V., McQuistin, S., Ellis, M., and C. Perkins,
                  "Circuit Breakers for Multimedia Congestion Control",
                  Proceedings of the 2013 20th International Packet Video
                  Workshop (PV), DOI 10.1109/PV.2013.6691439, December 2013.

       [WebRTC-QoS]
                  Jones, P., Dhesikan, S., Jennings, C., and D. Druta, "DSCP
                  Packet Markings for WebRTC QoS", Work in Progress,
                  draft-ietf-tsvwg-rtcweb-qos-18, August 2016.

    Perkins & Singh              Standards Track                   [Page 24]

    RFC 8083                  RTP Circuit Breakers                March 2017

    Acknowledgements

       The authors would like to thank Bernard Aboba, Harald Alvestrand, Ben
       Campbell, Alissa Cooper, Spencer Dawkins, Gorry Fairhurst, Stephen
       Farrell, Nazila Fough, Kevin Gross, Cullen Jennings, Randell Jesup,
       Mirja Kuehlewind, Jonathan Lennox, Matt Mathis, Stephen McQuistin,
       Simon Perreault, Eric Rescorla, Abheek Saha, Meral Shirazipour, Fabio
       Verdicchio, and Magnus Westerlund for their valuable feedback.

    Authors' Addresses

       Colin Perkins
       University of Glasgow
       School of Computing Science
       Glasgow  G12 8QQ
       United Kingdom

       Email: csp@csperkins.org

       Varun Singh
       CALLSTATS I/O Oy
       Runeberginkatu 4c A 4
       Helsinki  00100
       Finland

       Email: varun@callstats.io
       URI:   https://www.callstats.io/about

    Perkins & Singh              Standards Track                   [Page 25]