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Versions: (draft-perkins-avtcore-rtp-circuit-breakers) 00 01 02 03 04 05 06

AVTCORE Working Group                                      C. S. Perkins
Internet-Draft                                     University of Glasgow
Updates: 3550 (if approved)                                     V. Singh
Intended status: Standards Track                        Aalto University
Expires: January 05, 2015                                  July 04, 2014


Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions
               draft-ietf-avtcore-rtp-circuit-breakers-06

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; 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

   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 January 05, 2015.

Copyright Notice

   Copyright (c) 2014 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
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   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 . . . . . . . .   7
     4.2.  RTP/AVP Circuit Breaker #2: RTCP Timeout  . . . . . . . .   8
     4.3.  RTP/AVP Circuit Breaker #3: Congestion  . . . . . . . . .   9
     4.4.  RTP/AVP Circuit Breaker #4: Media Usability . . . . . . .  12
     4.5.  Ceasing Transmission  . . . . . . . . . . . . . . . . . .  13
   5.  RTP Circuit Breakers for Systems Using the RTP/AVPF Profile .  13
   6.  Impact of RTCP XR . . . . . . . . . . . . . . . . . . . . . .  14
   7.  Impact of RTCP Reporting Groups . . . . . . . . . . . . . . .  15
   8.  Impact of Explicit Congestion Notification (ECN)  . . . . . .  15
   9.  Security Considerations . . . . . . . . . . . . . . . . . . .  15
   10. IANA Considerations . . . . . . . . . . . . . . . . . . . . .  16
   11. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  16
   12. References  . . . . . . . . . . . . . . . . . . . . . . . . .  16
     12.1.  Normative References . . . . . . . . . . . . . . . . . .  16
     12.2.  Informative References . . . . . . . . . . . . . . . . .  17
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  18

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.





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   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.  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.











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   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, and it is randomised to avoid
      synchronisation of reports from multiple receivers.  RTCP RR
      packets allow a receiver to report ongoing network congestion to



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      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], [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] 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
      RTCP transmission keeps within its bandwidth allocation.  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.  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
   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



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   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
   which 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



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   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.  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
   protocol.

   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



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   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 SR or RR packets from the same receiver, 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 4.5).  The extended highest
   sequence number received field is non-increasing if the sender
   receives at least three RTCP SR or RR packets that report the same
   value for this field, but it has sent RTP data packets that would
   have caused an increase in the reported value if they had reached the
   receiver.

   The reason for waiting for two 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 two 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.

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 4.5).  When
   calculating the timeout, the deterministic RTCP reporting interval,
   Td, without the randomization factor, and with a fixed minimum
   interval Tmin=5 seconds) SHOULD be used.  The rationale for this
   choice of timeout is as described in Section 6.2 of RFC 3550
   [RFC3550].

   The choice of three RTCP reporting intervals as the timeout is made
   following Section 6.3.5 of RFC 3550 [RFC3550].  This specifies that



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   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, this will force the receiver to generate RTCP
   SR or RR packets in a round-robin manner.  In this case, the sender
   MAY treat receipt of an RTCP SR or RR packet corresponding to an SSRC
   it sent using the same 5-tuple of source and destination IP address,
   port, and protocol, as an indication that the receiver and return
   path are working to prevent 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.




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   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]:

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


   where:

   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, this formula can be
   approximated as follows with reasonable accuracy [Mathis]:

                       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.  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



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   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, 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).

   The congestion circuit breaker is therefore: when a sender receives
   an RTCP SR or RR packet that contains a report block for an SSRC it
   is using, that sender has to check the fraction lost field in that
   report block to determine if there is a non-zero packet loss rate.
   If the fraction lost field is zero, then continue sending as normal.
   If the fraction lost is greater than zero, then estimate the TCP
   throughput using the simplified equation above, and the measured R, p
   (approximated by the fraction lost), 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 (see Section 4.5).  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 (see
   Section 4.5).  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,



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   provided it takes care not to cause the problem to recur
   ("reasonable" is intentionally not defined here).

   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.  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
   matters (note that RTCP requires all participants in a session to
   have similar reporting intervals, else the participant timeout rules
   in [RFC3550] will not work).

   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.

   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, applications SHOULD monitor the reported
   packet loss and delay to estimate whether the media is suitable for
   the intended purpose.  If the packet loss rate and/or latency is such



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   that the media has become unusable for the application, and has
   remained unusable for a significant time period, then the application
   SHOULD cease transmission.  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 dependent.

   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.  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 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.



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   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
   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 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 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).  Reduced-size RTCP reports sent under
   the RTP/AVPF early feedback rules that contain RTCP SR or RR packets
   MUST be processed 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
   fire 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 early 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.

6.  Impact of RTCP XR





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   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.  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.




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   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].

10.  IANA Considerations

   There are no actions for IANA.

11.  Acknowledgements

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

12.  References

12.1.  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
              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.





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12.2.  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.

   [I-D.ietf-avtcore-rtp-multi-stream-optimisation]
              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-03 (work
              in progress), July 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.

   [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.



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   [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




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   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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