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RFC 8083

Proposed STD
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Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions

Part 1 of 2, p. 1 to 12
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Updates:    3550


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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
   (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

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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.



(page 12 continued on part 2)

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