[Docs] [txt|pdf] [draft-ietf-avtcor...] [Diff1] [Diff2]

PROPOSED STANDARD

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.




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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
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   publication of this document.  Please review these documents
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   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















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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



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   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



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      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.





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   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.









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   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.



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   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.




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   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.




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   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



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   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)




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   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





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      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.




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   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





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   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).



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   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



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   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].






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   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



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   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



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   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].




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   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>.




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   [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>.




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   [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























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