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Versions: (draft-perkins-avtcore-rtp-circuit-breakers) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 17 18 RFC 8083

Network Working Group                                         C. Perkins
Internet-Draft                                     University of Glasgow
Intended status: Standards Track                                V. Singh
Expires: April 25, 2013                                 Aalto University
                                                        October 22, 2012


     RTP Congestion Control: Circuit Breakers for Unicast Sessions
               draft-ietf-avtcore-rtp-circuit-breakers-01

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

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on April 25, 2013.

Copyright Notice

   Copyright (c) 2012 IETF Trust and the persons identified as the
   document authors.  All rights reserved.




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   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
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   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   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  . . . . . . . .  7
     4.2.  RTP/AVP Circuit Breaker #2: RTCP Timeout . . . . . . . . .  8
     4.3.  RTP/AVP Circuit Breaker #3: Congestion . . . . . . . . . .  9
   5.  RTP Circuit Breakers for Systems Using the RTP/AVPF Profile  . 11
   6.  Impact of RTCP XR  . . . . . . . . . . . . . . . . . . . . . . 12
   7.  Impact of Explicit Congestion Notification (ECN) . . . . . . . 12
   8.  Security Considerations  . . . . . . . . . . . . . . . . . . . 13
   9.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 13
   10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 13
   11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 13
     11.1. Normative References . . . . . . . . . . . . . . . . . . . 13
     11.2. Informative References . . . . . . . . . . . . . . . . . . 14
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 15





















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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.  Rather, it proposes a minimal set of "circuit breakers";
   conditions under which there is general agreement that an RTP flow is
   causing serious congestion, and ought to cease transmission.  It is
   expected that future standards-track congestion control algorithms
   for RTP will operate within the envelope defined by this memo.


2.  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.


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.

   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



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   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 participants in the session and a configured session
      bandwidth estimate.  The interval between reports sent from each
      receiver tends to be on the order of a few seconds on average, and



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      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 partway 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],
      [I-D.ietf-xrblock-rtcp-xr-discard],
      [I-D.ietf-xrblock-rtcp-xr-discard-rle-metrics],
      [I-D.ietf-xrblock-rtcp-xr-burst-gap-discard],
      [I-D.ietf-xrblock-rtcp-xr-burst-gap-loss]; and the extended delay
      metrics [I-D.ietf-xrblock-rtcp-xr-delay],
      [I-D.ietf-xrblock-rtcp-xr-pdv].  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] in place of the more common RTP/AVP profile
      [RFC3551].  This modifies the RTCP timing rules to allow RTCP
      reports to be sent early, in some cases immediately, provided the
      average RTCP reporting interval remains unchanged.  It also
      defines new transport-layer feedback messages, including negative
      acknowledgements (NACKs), that can be used to report on specific
      congestion events.  The use of the RTP/AVPF profile is dependent
      on signalling, but is otherwise generally backwards compatible, as
      it keeps the same average RTCP reporting interval as the base RTP
      specification.  The 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.  The
      dynamics of how rapidly feedback can be sent are unchanged.

   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



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   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.

   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 frame comprises one
       RTP packet).

   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



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   to determine connection timeouts.  RTT estimates derived from RTCP SR
   and RR packets sent according to the RTP/AVP timing rules are far 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.  These are robust indicators of
   congestion in a network where packet loss is primarily due to queue
   overflows, although less accurate in networks where losses can be
   caused by non-congestive packet corruption.  TCP uses packet loss as
   a congestion signal.

   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.  We derive circuit breaker conditions for these two
   loss regimes in the following.

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

   If RTP data packets are being sent while the corresponding RTCP RR
   packets report 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 two or more consecutive RTCP RR
   packets from the same receiver that correspond to its transmission,
   and have a non-increasing extended highest sequence number received
   field (i.e., at least three RTCP RR packets that report the same
   value in the extended highest sequence number received field, when
   the sender has sent data packets that would have caused an increase
   in the reported value of the extended highest sequence number



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   received if they had reached the receiver), then that sender SHOULD
   cease transmission.  What it means to cease transmission depends on
   the application, but the intention is that the application will stop
   sending RTP data packets until the user makes an explicit attempt to
   restart the call (RTP flows halted by the circuit breaker SHOULD NOT
   be restarted automatically unless the sender has received information
   that the congestion has dissipated).

   Systems that usually send at a high data rate, but that can reduce
   their data rate significantly (i.e., by at least a factor of ten),
   MAY first reduce their sending rate to this lower value to see if
   this resolves the congestion, but MUST then cease transmission if the
   problem does not resolve itself within a further two RTCP reporting
   intervals.  An example of this 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 choice of two RTCP reporting intervals is to give enough time for
   transient 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 numerous
   transient faults, and so will not cease transmission.  Waiting for
   more than two RTCP reports before stopping a flow might avoid some
   false positives, but would lead to problematic flows running for a
   long time before being cut off.

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.

   According to RFC 3550 [RFC3550], any participant that has not sent an
   RTCP packet within the last two RTCP intervals is removed from the
   sender list.  Therefore, an RTP sender SHOULD cease transmission if
   it does not receive a single RTCP RR packet and during this period
   has sent 3 RTCP SR packets to the RTP receiver.  Similarly, the same
   circuit breaker rule applies to an RTCP receiver which has not
   received a single SR packet, and in the corresponding period it has
   sent 3 RTCP RR packets.  What it means to cease transmission depends
   on the application, but the intention is that the application will



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   stop sending RTP data packets until the user makes an explicit
   attempt to restart the call (RTP flows halted by the circuit breaker
   SHOULD NOT be restarted automatically unless the sender has received
   information that the congestion has dissipated).

4.3.  RTP/AVP Circuit Breaker #3: Congestion

   If RTP data packets are being sent, and the corresponding RTCP 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, Padhye et al.  [Padhye] showed
   that the throughput can be estimated using the following equation:

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

   where:






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   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,
      approximated by setting t_RTO = 4*R.

   b  is the number of packets 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, this formula can be
   approximated as follows with reasonable accuracy:

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

   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.

   Given this TCP equation, 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.

   RTCP 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, not
   caring about when the 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).



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   The congestion circuit breaker is therefore: when RTCP RR packets are
   received, estimate the TCP throughput using the simplified equation
   above, and the measured R, p (approximated by the loss fraction), and
   s.  Compare this with the actual sending rate.  If the actual sending
   rate is more than ten times the estimated sending rate derived from
   the TCP throughput equation for two consecutive RTCP reporting
   intervals, the sender SHOULD cease transmission.  What it means to
   cease transmission depends on the application, but the intention is
   that the application will stop sending RTP data packets until the
   user makes an explicit attempt to restart the call (RTP flows halted
   by the circuit breaker SHOULD NOT be restarted automatically unless
   the sender has received information that the congestion has
   dissipated).

   Systems that usually send at a high data rate, but that can reduce
   their data rate significantly (i.e., by at least a factor of ten),
   MAY first reduce their sending rate to this lower value to see if
   this resolves the congestion, but MUST then cease transmission if the
   problem does not resolve itself within a further two RTCP reporting
   intervals.  An example of this 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).

   As in Section 4.1, we use two 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.


5.  RTP Circuit Breakers for Systems Using the RTP/AVPF Profile

   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.

   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 for this reason, defining new transport layer



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   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.

   If the extension for Reduced-Size RTCP [RFC5506] is not used, early
   RTCP feedback packets sent according to the RTP/AVPF profile will be
   compound RTCP packets that include an RTCP SR/RR packet.  That RTCP
   SR/RR packet MUST be processed as if it were sent as a regular RTCP
   report and counted towards the circuit breaker conditions specified
   in Section 4.1 and Section 4.3 of this memo.  This will potentially
   make the RTP circuit breaker fire earlier than it would if the RTP/
   AVPF profile was not used.

   Reduced-size RTCP reports sent under to the RTP/AVPF early feedback
   rules that do not contain an RTCP SR or RR packet MUST be ignored by
   the RTP circuit breaker (they do not contain the information used by
   the circuit breaker algorithm).  In this case, the circuit breaker
   will only use the information contained in the periodic RTCP SR/RR
   packets.  This allows the use of low-overhead early RTP/AVPF feedback
   without triggering the RTP circuit breaker, and so is suitable for
   RTP congestion control algorithms that need to quickly report loss
   events in between regular RTCP reports.


6.  Impact of RTCP 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.
   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
   or RR packets are used by the RTP circuit breaker algorithm.


7.  Impact of Explicit Congestion Notification (ECN)

   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 use of ECN for RTP flows does not affect the media timeout RTP



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   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.


8.  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, for example using the Secure RTP profile [RFC3711].


9.  IANA Considerations

   There are no actions for IANA.


10.  Acknowledgements

   The authors would like to thank Harald Alvestrand, Randell Jesup,
   Matt Mathis, and Abheek Saha for their valuable feedback.


11.  References

11.1.  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.




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   [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 2003.

   [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 2006.

11.2.  Informative References

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

   [I-D.ietf-xrblock-rtcp-xr-burst-gap-discard]
              Clark, A., Huang, R., and W. Wu, "RTP Control
              Protocol(RTCP) Extended Report (XR) Block for Discard
              Count metric Reporting",
              draft-ietf-xrblock-rtcp-xr-burst-gap-discard-06 (work in
              progress), October 2012.

   [I-D.ietf-xrblock-rtcp-xr-burst-gap-loss]
              Clark, A., Zhang, S., Zhao, J., and W. Wu, "RTP Control
              Protocol (RTCP) Extended Report (XR) Block for Burst/Gap
              Loss metric Reporting",
              draft-ietf-xrblock-rtcp-xr-burst-gap-loss-04 (work in
              progress), October 2012.

   [I-D.ietf-xrblock-rtcp-xr-delay]
              Clark, A., Gross, K., and W. Wu, "RTP Control Protocol
              (RTCP) Extended Report (XR) Block for Delay metric
              Reporting", draft-ietf-xrblock-rtcp-xr-delay-10 (work in
              progress), October 2012.

   [I-D.ietf-xrblock-rtcp-xr-discard]
              Clark, A., Zorn, G., and W. Wu, "RTP Control Protocol
              (RTCP) Extended Report (XR) Block for Discard Count metric
              Reporting", draft-ietf-xrblock-rtcp-xr-discard-09 (work in
              progress), October 2012.

   [I-D.ietf-xrblock-rtcp-xr-discard-rle-metrics]
              Ott, J., Singh, V., and I. Curcio, "RTP Control Protocol



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              (RTCP) Extended Reports (XR) for Run Length Encoding (RLE)
              of Discarded Packets",
              draft-ietf-xrblock-rtcp-xr-discard-rle-metrics-04 (work in
              progress), July 2012.

   [I-D.ietf-xrblock-rtcp-xr-pdv]
              Clark, A. and W. Wu, "RTP Control Protocol (RTCP) Extended
              Report (XR) Block for Packet Delay Variation Metric
              Reporting", draft-ietf-xrblock-rtcp-xr-pdv-08 (work in
              progress), September 2012.

   [Padhye]   Padhye, J., Firoiu, V., Towsley, D., and J. Kurose,
              "Modeling TCP Throughput: A Simple Model and its Empirical
              Validation", Proc. ACM SIGCOMM 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.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.

   [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.

   [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.

   [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.










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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
   Finland

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































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