AVTCORE Working Group                                      C. S. Perkins
Internet-Draft                                     University of Glasgow
Updates: 3550 (if approved)                                     V. Singh
Intended status: Standards Track                        Aalto University
Expires: September 07, 24, 2015                               March 06, 23, 2015

Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions


   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 the applications, then network congestion will
   deteriorate the user's multimedia experience.  This document does not
   propose a congestion control algorithm; instead, it defines a minimal
   set of RTP "circuit-breakers".  Circuit-breakers are conditions under
   which an RTP sender needs to stop transmitting media data in order to
   protect the network from excessive congestion.  It is expected that,
   in the absence of severe congestion, all RTP applications running on
   best-effort IP networks will be able to run without triggering these
   circuit breakers.  Any future RTP congestion control specification
   will be expected to operate within the constraints defined by these
   circuit breakers.

Status of This Memo

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   provisions of BCP 78 and BCP 79.

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   This Internet-Draft will expire on September 07, 24, 2015.

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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   3
   3.  Background  . . . . . . . . . . . . . . . . . . . . . . . . .   3
   4.  RTP Circuit Breakers for Systems Using the RTP/AVP Profile  .   6
     4.1.  RTP/AVP Circuit Breaker #1: Media Timeout . . . . . . . .   8   7
     4.2.  RTP/AVP Circuit Breaker #2: RTCP Timeout  . . . . . . . .   8
     4.3.  RTP/AVP Circuit Breaker #3: Congestion  . . . . . . . . .  10   9
     4.4.  RTP/AVP Circuit Breaker #4: Media Usability . . . . . . .  13
     4.5.  Choice of Circuit Breaker Interval  . . . . . . . . . . .  14
     4.6.  Ceasing Transmission  . . . . . . . . . . . . . . . . . .  15
   5.  RTP Circuit Breakers and the RTP/AVPF and RTP/SAVPF Profiles   16
   6.  Impact of RTCP Extended Reports (XR)  . . . . . . . . . . . .  17
   7.  Impact of RTCP Reporting Groups . . . . . . . . . . . . . . .  17
   8.  Impact of Explicit Congestion Notification (ECN)  . . . . . .  18
   9.  Impact of Bundled Media and Layered Coding  . . . . . . . . .  18
   10. Security Considerations . . . . . . . . . . . . . . . . . . .  18
   11. IANA Considerations . . . . . . . . . . . . . . . . . . . . .  19
   12. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  19
   13. References  . . . . . . . . . . . . . . . . . . . . . . . . .  19
     13.1.  Normative References . . . . . . . . . . . . . . . . . .  19
     13.2.  Informative References . . . . . . . . . . . . . . . . .  20
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  22

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.  Rather, it proposes a minimal set of "circuit "RTP circuit
   breakers"; conditions under which there is general agreement that an
   RTP flow is causing serious congestion, and ought to cease
   transmission.  The RTP circuit breakers proposed in this memo are a
   specific instance of the general class of network transport circuit
   breakers [I-D.ietf-tsvwg-circuit-breaker], designed to act as a
   protection mechanism of last resort to avoid persistent congestion.
   It is expected that future standards-track congestion control
   algorithms for RTP will operate within the envelope defined by this

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   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.

3.  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 matches the available network bandwidth.
   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 bandwidth 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
   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 disrupt
   the network's operation 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; 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 Reception 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
      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 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 tends to be on the order
      of a few seconds on average, although it varies with the session
      bandwidth, and sub-second reporting intervals are possible in high
      bandwidth sessions, and it is randomised to avoid synchronisation
      of reports from multiple receivers.  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]; and 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 that 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 signalling.

   o  Finally, Explicit Congestion Notification (ECN) for RTP over UDP
      [RFC6679] can be used to provide feedback on the number of packets
      that received an ECN Congestion Experienced (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.  There are two methods: [RFC5450] represents the
   transmission time in terms of a time-offset from the RTP timestamp of
   the packet, while [RFC6051] includes an explicit NTP-format sending
   timestamp (potentially more accurate, but a higher header overhead).
   Accurate sending 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.

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, cannot 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.

   Three potential congestion signals are available from the basic RTCP
   SR/RR packets and are reported for each synchronisation source (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.

   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
       ([RFC3550], section 6.4.4), 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 behaviour 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 though, 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, these
   can be effective indicators that 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 that has to
   be related to TCP dynamics to estimate available capacity.  This
   behaviour of TCP is reflected in the congestion circuit breaker
   below, and will affect the design of any RTP congestion control

   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: 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 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.
   Accordingly, if a sender of RTP data packets receives CB_INTERVAL or
   more consecutive RTCP SR or RR packets from the same receiver (see
   Section 4.5), and those packets correspond to its transmission and
   have a non-increasing extended highest sequence number received
   field, then that sender SHOULD cease transmission (see Section 4.6).
   The extended highest sequence number received field is non-increasing
   if the sender receives at least CB_INTERVAL consecutive RTCP SR or RR
   packets that report the same value for this field, but it has sent
   RTP data packets, at a rate of at least one per RTT, that would have
   caused an increase in the reported value if they had reached the

   The rationale for waiting for CB_INTERVAL or more consecutive RTCP
   packets with a non-increasing extended highest sequence number is to
   give enough time for transient reception problems to resolve
   themselves, but to stop problem flows quickly enough to avoid causing
   serious ongoing network congestion.  A single RTCP report showing no
   reception could be caused by a transient fault, and so will not cease
   transmission.  Waiting for more than CB_INTERVAL consecutive RTCP
   reports before stopping a flow might avoid some false positives, but
   could lead to problematic flows running for a long time period
   (potentially tens of seconds, depending on the RTCP reporting
   interval) before being cut off.  Equally, an application that sends
   few packets when the packet loss rate is high runs the risk that the
   media timeout circuit breaker triggers inadvertently.  The chosen
   timeout interval is a trade-off between these extremes.

   The rationale for enforcing a minimum sending rate below which the
   media timeout circuit breaker will not trigger is to avoid spurious
   circuit breaker triggers when the number of packets sent per RTCP
   reporting interval is small (e.g., a telephony application sends only
   two RTP comfort noise packets during a five second RTCP reporting
   interval, and both are lost; this is 100% packet loss, but it seems
   extreme to terminate the RTP session).  The one packet per RTT bound
   derives from [RFC5405].

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

   In addition to media timeouts, as were discussed in Section 4.1, an
   RTP session has the possibility of an RTCP timeout.  This 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.  Accordingly, an RTP
   sender that has not received any RTCP SR or RTCP RR packets reporting
   on the SSRC it is using for three or more of its RTCP reporting
   intervals SHOULD cease transmission (see Section 4.6).  When
   calculating the timeout, the deterministic RTCP reporting interval,
   Td, without the randomization factor, and using the fixed minimum
   interval of Tmin=5 seconds, MUST be used.  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"), as updated by Section 6.1.4 of
   [I-D.ietf-avtcore-rtp-multi-stream] 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 [I-D.ietf-avtcore-rtp-multi-stream]).

   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
   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, preventing the RTCP timeout circuit breaker from triggering.

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 time scale, that is not less than 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 time scale and
      throughput.  The time scale 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 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 TCP connection would achieve over the path.
   For a long-lived TCP Reno connection, it has been shown that the TCP
   throughput can be estimated using the following equation [Padhye]:

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


   X  is the transmit rate in bytes/second.

   s  is the packet size in bytes.  If data packets vary in size, then
      the average size is to be used.

   R  is the round trip time in seconds.

   p  is the loss event rate, between 0 and 1.0, of the number of loss
      events as a fraction of the number of packets transmitted.

   t_RTO  is the TCP retransmission timeout value in seconds, generally
      approximated by setting t_RTO = 4*R.

   b  is the number of packets that are acknowledged by a single TCP
      acknowledgement; [RFC3448] recommends the use of b=1 since many
      TCP implementations do not use delayed acknowledgements.

   This is the same approach to estimated TCP throughput that is used in
   [RFC3448].  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]:

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

   It is RECOMMENDED that this simplified throughout 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 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 behaviour.

   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, R, 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.

   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 does 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 ([Floyd] section 3.2.1 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: when a sender that is
   transmitting more than one RTP packet per RTT 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 in the report
   block and the time since the last report block was received for that
   SSRC.  If more than CB_INTERVAL (see Section 4.5) 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, R, the loss event rate, p (as approximated by
   the average fraction lost), and the packet size, s.  This estimate of
   the TCP throughput is then compared with the actual sending rate.  If
   the actual sending rate is more than ten times the TCP throughput
   estimate, 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 multiplied by the duration of the corresponding
   reporting interval, divided by the total duration of the last
   CB_INTERVAL reporting intervals.

   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 artefacts.  The one packet per RTT bound
   derives from [RFC5405].

   When the congestion circuit breaker is triggered, the sender SHOULD
   cease transmission (see Section 4.6).  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).

   The RTCP reporting interval of the media sender does not affect how
   quickly 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.

   As in Section 4.1 and Section 4.2, we use CB_INTERVAL reporting
   intervals to avoid triggering the circuit breaker on transient
   failures.  This circuit breaker is a worst-case condition, and
   congestion control needs to be performed to keep well within this
   bound.  It is expected that the circuit breaker will only be
   triggered if the usual congestion control fails for some reason.

   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 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, nor does it specify what
   time period is deemed significant, as these are highly application

   Sending media that suffers from such high packet loss or latency that
   it is unusable at the receiver is both wasteful of resources, and 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 that the RTP
   application cease sending useless traffic.  The role of the media
   usability circuit breaker is to protect the network in such cases.

4.5.  Choice of Circuit Breaker Interval

   The CB_INTERVAL parameter determines the number of consecutive RTCP
   reporting intervals that need to suffer congestion before the media
   timeout circuit breaker (see Section 4.1) or the congestion circuit
   breaker (see Section 4.3) triggers.  It determines the sensitivity
   and responsiveness of these circuit breakers.

   The CB_INTERVAL parameter is set to min(floor(3+(2.5/Td)), 30) RTCP
   reporting intervals, where Td is the deterministic calculated RTCP
   interval described in section 6.3.1 of [RFC3550].  This expression
   gives an CB_INTERVAL that varies as follows:

            Td       |       CB_INTERVAL            | Time to trigger
       0.016 seconds |  30 RTCP reporting intervals |  0.48 seconds
       0.033 seconds |  30 RTCP reporting intervals |  0.99 seconds
       0.100 seconds |  28 RTCP reporting intervals |    2.8  2.80 seconds
       0.500 seconds |   8 RTCP reporting intervals |    4.0  4.00 seconds
       1.000 seconds |   5 RTCP reporting intervals |    5.5  5.00 seconds
       2.000 seconds |   4 RTCP reporting intervals |    8.5  8.00 seconds
       5.000 seconds |   5   3 RTCP reporting intervals |   17.5 15.00 seconds
      10.000 seconds |   3 RTCP reporting intervals |   32.5 30.00 seconds

   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 Td in the above expression for
   the CB_INTERVAL parameter MUST be replaced by T_rr_interval. max(T_rr_interval, Td).

   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 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

   Rationale: If the CB_INTERVAL was always set to the same number of
   RTCP reporting intervals, this would cause higher rate RTP sessions
   to trigger the RTP circuit breaker after a shorter time interval than
   lower rate sessions, because the RTCP reporting interval scales based
   on the RTP session bandwidth.  This is felt to penalise high rate RTP
   sessions too aggressively.  Conversely, scaling CB_INTERVAL according
   to the inverse of the RTCP reporting interval, so the RTP circuit
   breaker triggers after a constant time interval, doesn't sufficiently
   protect the network from congestion caused by high-rate flows.  The
   chosen expression for CB_INTERVAL seeks a balance between these two
   extremes.  It causes higher rate RTP sessions subject to persistent
   congestion to trigger the RTP circuit breaker after a shorter time
   interval than do lower rate RTP sessions, while also making the RTP
   circuit breaker for such sessions less sensitive by requiring the
   congestion to persist for longer numbers of RTCP reporting intervals.

4.6.  Ceasing Transmission

   What it means to cease transmission depends on the application, but
   the intention is that the application will stop sending RTP data
   packets to a particular destination 3-tuple (transport protocol,
   destination port, IP address), until the user makes an explicit
   attempt to restart the call.  It is important that a human user is
   involved in the decision to try to restart the call, since that user
   will eventually give up if the calls repeatedly trigger the circuit
   breaker.  This will help avoid problems with automatic redial systems
   from congesting the network.  Accordingly, RTP flows halted by the
   circuit breaker SHOULD NOT be restarted automatically unless the
   sender has received information that the congestion has dissipated.

   It is recognised that the RTP implementation in some systems might
   not be able to determine if a call set-up request was initiated by a
   human user, or automatically by some scripted higher-level component
   of the system.  These implementations SHOULD rate limit attempts to
   restart a call to the same destination 3-tuple as used by a previous
   call that was recently halted by the circuit breaker.  The chosen
   rate limit ought to not exceed the rate at which an annoyed human
   caller might redial a misbehaving phone.

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,
   defining 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

   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 8), 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
   of lost packets reported if the ECN Feedback Report report is sent in
   an 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
   behaviour 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 provided non-congestion-related metrics.
   With the exception of RTCP XR ECN Summary Reports (see Section 8),
   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 reception 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 RTCP Reporting Groups

   An optimisation for grouping RTCP reception statistics and other
   feedback in RTP sessions with large numbers of participants is given
   in [I-D.ietf-avtcore-rtp-multi-stream-optimisation].  This allows one
   SSRC to act as a representative that sends reports on behalf of other
   SSRCs that are co-located in the same endpoint and see identical
   reception quality.  When running the circuit breaker algorithms, an
   endpoint MUST treat a reception report from the representative of the
   reporting group as if a reception report was received from all
   members of that group.

8.  Impact of Explicit Congestion Notification (ECN)

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

   ECN-CE marked packets SHOULD be treated as if it were lost for the
   purposes of congestion control, when determining the optimal media
   sending rate for an RTP flow.  If an RTP sender has negotiated ECN
   support for an RTP session, and has successfully initiated ECN use on
   the path to the receiver [RFC6679], then ECN-CE marked packets SHOULD
   be treated as if they were lost when calculating if the congestion-
   based RTP circuit breaker (Section 4.3) has been met.  The count of
   ECN-CE marked RTP packets is returned in RTCP XR ECN summary report
   packets if support for ECN has been initiated for an RTP session.

9.  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 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, ceasing transmission
   on the enhancement layers first to reduce sending rate if necessary,
   rather than treating each layer independently.

10.  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, causing the circuit breaker to trigger and disrupting 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].

   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
   is many minutes duration.  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 4.5.

11.  IANA Considerations

   There are no actions for IANA.

12.  Acknowledgements

   The authors would like to thank Bernard Aboba, Harald Alvestrand,
   Gorry Fairhurst, Nazila Fough, Kevin Gross, Cullen Jennings, Randell
   Jesup, Jonathan Lennox, Matt Mathis, Stephen McQuistin, Eric
   Rescorla, Abheek Saha, Fabio Verdicchio, and Magnus Westerlund for
   their valuable feedback.

13.  References

13.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3448]  Handley, M., Floyd, S., Padhye, J., and J. Widmer, "TCP
              Friendly Rate Control (TFRC): Protocol Specification", RFC
              3448, January 2003.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              July 2003.

   [RFC3611]  Friedman, T., Caceres, R., and A. Clark, "RTP Control
              Protocol Extended Reports (RTCP XR)", RFC 3611, November

   [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, July

13.2.  Informative References

   [Floyd]    Floyd, S., Handley, M., Padhye, J., and J. Widmer,
              "Equation-Based Congestion Control for Unicast
              Applications", Proceedings of the ACM SIGCOMM conference,
              2000, DOI 10.1145/347059.347397, August 2000.

              Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
              "Sending Multiple Media Streams in a Single RTP Session:
              Grouping RTCP Reception Statistics and Other Feedback",
              draft-ietf-avtcore-rtp-multi-stream-optimisation-05 (work
              in progress), February 2015.

              Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
              "Sending Multiple Media Streams in a Single RTP Session",
              draft-ietf-avtcore-rtp-multi-stream-07 (work in progress),
              March 2015.

              Fairhurst, G., "Network Transport Circuit Breakers",
              draft-ietf-tsvwg-circuit-breaker-00 (work in progress),
              September 2014.

   [Mathis]   Mathis, M., Semke, J., Mahdavi, J., and T. Ott, "The
              macroscopic behavior of the TCP congestion avoidance
              algorithm", ACM SIGCOMM Computer Communication Review
              27(3), 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", Proceedings of the ACM SIGCOMM conference,
              1998, DOI 10.1145/285237.285291, August 1998.

   [RFC3168]  Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
              of Explicit Congestion Notification (ECN) to IP", RFC
              3168, September 2001.

   [RFC5104]  Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
              "Codec Control Messages in the RTP Audio-Visual Profile
              with Feedback (AVPF)", RFC 5104, February 2008.

   [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/SAVPF)", RFC 5124, February 2008.

   [RFC5405]  Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
              for Application Designers", BCP 145, RFC 5405, November

   [RFC5450]  Singer, D. and H. Desineni, "Transmission Time Offsets in
              RTP Streams", RFC 5450, March 2009.

   [RFC5506]  Johansson, I. and M. Westerlund, "Support for Reduced-Size
              Real-Time Transport Control Protocol (RTCP): Opportunities
              and Consequences", RFC 5506, April 2009.

   [RFC5681]  Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
              Control", RFC 5681, September 2009.

   [RFC6051]  Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
              Flows", RFC 6051, November 2010.

   [RFC6679]  Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
              and K. Carlberg, "Explicit Congestion Notification (ECN)
              for RTP over UDP", RFC 6679, August 2012.

   [RFC6798]  Clark, A. and Q. Wu, "RTP Control Protocol (RTCP) Extended
              Report (XR) Block for Packet Delay Variation Metric
              Reporting", RFC 6798, November 2012.

   [RFC6843]  Clark, A., Gross, K., and Q. Wu, "RTP Control Protocol
              (RTCP) Extended Report (XR) Block for Delay Metric
              Reporting", RFC 6843, January 2013.

   [RFC6958]  Clark, A., Zhang, S., Zhao, J., and Q. Wu, "RTP Control
              Protocol (RTCP) Extended Report (XR) Block for Burst/Gap
              Loss Metric Reporting", RFC 6958, May 2013.

   [RFC7002]  Clark, A., Zorn, G., and Q. Wu, "RTP Control Protocol
              (RTCP) Extended Report (XR) Block for Discard Count Metric
              Reporting", RFC 7002, September 2013.

   [RFC7003]  Clark, A., Huang, R., and Q. Wu, "RTP Control Protocol
              (RTCP) Extended Report (XR) Block for Burst/Gap Discard
              Metric Reporting", RFC 7003, September 2013.

   [RFC7097]  Ott, J., Singh, V., and I. Curcio, "RTP Control Protocol
              (RTCP) Extended Report (XR) for RLE of Discarded Packets",
              RFC 7097, January 2014.

   [RFC7201]  Westerlund, M. and C. Perkins, "Options for Securing RTP
              Sessions", RFC 7201, April 2014.

   [Sarker]   Sarker, Z., Singh, V., and C.S. 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, 2014,
              April 2014.

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

Authors' Addresses

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

   Email: csp@csperkins.org

   Varun Singh
   Aalto University
   School of Electrical Engineering
   Otakaari 5 A
   Espoo, FIN  02150

   Email: varun@comnet.tkk.fi
   URI:   http://www.netlab.tkk.fi/~varun/