One document matched: draft-ietf-avtcore-rtp-circuit-breakers-01.txt
Differences from draft-ietf-avtcore-rtp-circuit-breakers-00.txt
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
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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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Internet-Draft RTP Circuit Breakers October 2012
[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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Internet-Draft RTP Circuit Breakers October 2012
(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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