[Docs] [txt|pdf] [Tracker] [WG] [Email] [Diff1] [Diff2] [Nits]
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
AVTCORE Working Group C. S. Perkins
Internet-Draft University of Glasgow
Updates: 3550 (if approved) V. Singh
Intended status: Standards Track Aalto University
Expires: April 30, 2015 October 27, 2014
Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions
draft-ietf-avtcore-rtp-circuit-breakers-07
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 April 30, 2015.
Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved.
Perkins & Singh Expires April 30, 2015 [Page 1]
Internet-Draft RTP Circuit Breakers October 2014
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 . . . . . . . . 8
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 . . . . . . . 13
4.5. Ceasing Transmission . . . . . . . . . . . . . . . . . . 14
5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile . 14
6. Impact of RTCP Extended Reports (XR) . . . . . . . . . . . . 15
7. Impact of RTCP Reporting Groups . . . . . . . . . . . . . . . 15
8. Impact of Explicit Congestion Notification (ECN) . . . . . . 16
9. Impact of Layered Coding . . . . . . . . . . . . . . . . . . 16
10. Security Considerations . . . . . . . . . . . . . . . . . . . 17
11. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 17
12. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 17
13. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 18
14. References . . . . . . . . . . . . . . . . . . . . . . . . . 18
14.1. Normative References . . . . . . . . . . . . . . . . . . 18
14.2. Informative References . . . . . . . . . . . . . . . . . 18
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 20
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.
Perkins & Singh Expires April 30, 2015 [Page 2]
Internet-Draft RTP Circuit Breakers October 2014
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.
Perkins & Singh Expires April 30, 2015 [Page 3]
Internet-Draft RTP Circuit Breakers October 2014
Alternatively, RTP flows can be run over a non-congestion controlled
transport protocol, for example UDP, performing rate adaptation at
the application layer based on RTP Control Protocol (RTCP) feedback.
With a well-designed, network-aware, application, this allows highly
effective media quality adaptation, but there is potential to disrupt
the network's operation if the application does not adapt its sending
rate in a timely and effective manner. We consider this class of
applications in this memo.
Congestion control relies on monitoring the delivery of a media flow,
and responding to adapt the transmission of that flow when there are
signs that the network path is congested. Network congestion can be
detected in one of three ways: 1) a receiver can infer the onset of
congestion by observing an increase in one-way delay caused by queue
build-up within the network; 2) if Explicit Congestion Notification
(ECN) [RFC3168] is supported, the network can signal the presence of
congestion by marking packets using ECN Congestion Experienced (CE)
marks; or 3) in the extreme case, congestion will cause packet loss
that can be detected by observing a gap in the received RTP sequence
numbers. Once the onset of congestion is observed, the receiver has
to send feedback to the sender to indicate that the transmission rate
needs to be reduced. How the sender reduces the transmission rate is
highly dependent on the media codec being used, and is outside the
scope of this memo.
There are several ways in which a receiver can send feedback to a
media sender within the RTP framework:
o The base RTP specification [RFC3550] defines RTCP Reception Report
(RR) packets to convey reception quality feedback information, and
Sender Report (SR) packets to convey information about the media
transmission. RTCP SR packets contain data that can be used to
reconstruct media timing at a receiver, along with a count of the
total number of octets and packets sent. RTCP RR packets report
on the fraction of packets lost in the last reporting interval,
the cumulative number of packets lost, the highest sequence number
received, and the inter-arrival jitter. The RTCP RR packets also
contain timing information that allows the sender to estimate the
network round trip time (RTT) to the receivers. RTCP reports are
sent periodically, with the reporting interval being determined by
the number of SSRCs used in the session and a configured session
bandwidth estimate (the number of SSRCs used is usually two in a
unicast session, one for each participant, but can be greater if
the participants send multiple media streams). The interval
between reports sent from each receiver tends to be on the order
of a few seconds on average, although it varies with the session
bandwidth, and sub-second reporting intervals are possible in high
bandwidth sessions, and it is randomised to avoid synchronisation
Perkins & Singh Expires April 30, 2015 [Page 4]
Internet-Draft RTP Circuit Breakers October 2014
of reports from multiple receivers. RTCP RR packets allow a
receiver to report ongoing network congestion to the sender.
However, if a receiver detects the onset of congestion part way
through a reporting interval, the base RTP specification contains
no provision for sending the RTCP RR packet early, and the
receiver has to wait until the next scheduled reporting interval.
o The RTCP Extended Reports (XR) [RFC3611] allow reporting of more
complex and sophisticated reception quality metrics, but do not
change the RTCP timing rules. RTCP extended reports of potential
interest for congestion control purposes are the extended packet
loss, discard, and burst metrics [RFC3611], [RFC7002], [RFC7097],
[RFC7003], [RFC6958]; and the extended delay metrics [RFC6843],
[RFC6798]. Other RTCP Extended Reports that could be helpful for
congestion control purposes might be developed in future.
o Rapid feedback about the occurrence of congestion events can be
achieved using the Extended RTP Profile for RTCP-Based Feedback
(RTP/AVPF) [RFC4585] (or its secure variant, RTP/SAVPF [RFC5124])
in place of the RTP/AVP profile [RFC3551]. This modifies the RTCP
timing rules to allow RTCP reports to be sent early, in some cases
immediately, provided the RTCP transmission rate keeps within its
bandwidth allocation. It also defines transport-layer feedback
messages, including negative acknowledgements (NACKs), that can be
used to report on specific congestion events. RTP Codec Control
Messages [RFC5104] extend the RTP/AVPF profile with additional
feedback messages that can be used to influence that way in which
rate adaptation occurs, but do not further change the dynamics of
how rapidly feedback can be sent. Use of the RTP/AVPF profile is
dependent on signalling.
o Finally, Explicit Congestion Notification (ECN) for RTP over UDP
[RFC6679] can be used to provide feedback on the number of packets
that received an ECN Congestion Experienced (CE) mark. This RTCP
extension builds on the RTP/AVPF profile to allow rapid congestion
feedback when ECN is supported.
In addition to these mechanisms for providing feedback, the sender
can include an RTP header extension in each packet to record packet
transmission times. There are two methods: [RFC5450] represents the
transmission time in terms of a time-offset from the RTP timestamp of
the packet, while [RFC6051] includes an explicit NTP-format sending
timestamp (potentially more accurate, but a higher header overhead).
Accurate sending timestamps can be helpful for estimating queuing
delays, to get an early indication of the onset of congestion.
Taken together, these various mechanisms allow receivers to provide
feedback on the senders when congestion events occur, with varying
Perkins & Singh Expires April 30, 2015 [Page 5]
Internet-Draft RTP Circuit Breakers October 2014
degrees of timeliness and accuracy. The key distinction is between
systems that use only the basic RTCP mechanisms, without RTP/AVPF
rapid feedback, and those that use the RTP/AVPF extensions to respond
to congestion more rapidly.
4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile
The feedback mechanisms defined in [RFC3550] and available under the
RTP/AVP profile [RFC3551] are the minimum that can be assumed for a
baseline circuit breaker mechanism that is suitable for all unicast
applications of RTP. Accordingly, for an RTP circuit breaker to be
useful, it needs to be able to detect that an RTP flow is causing
excessive congestion using only basic RTCP features, without needing
RTCP XR feedback or the RTP/AVPF profile for rapid RTCP reports.
RTCP is a fundamental part of the RTP protocol, and the mechanisms
described here rely on the implementation of RTCP. Implementations
that claim to support RTP, but that do not implement RTCP, cannot use
the circuit breaker mechanisms described in this memo. Such
implementations SHOULD NOT be used on networks that might be subject
to congestion unless equivalent mechanisms are defined using some
non-RTCP feedback channel to report congestion and signal circuit
breaker conditions.
Three potential congestion signals are available from the basic RTCP
SR/RR packets and are reported for each synchronisation source (SSRC)
in the RTP session:
1. The sender can estimate the network round-trip time once per RTCP
reporting interval, based on the contents and timing of RTCP SR
and RR packets.
2. Receivers report a jitter estimate (the statistical variance of
the RTP data packet inter-arrival time) calculated over the RTCP
reporting interval. Due to the nature of the jitter calculation
([RFC3550], section 6.4.4), the jitter is only meaningful for RTP
flows that send a single data packet for each RTP timestamp value
(i.e., audio flows, or video flows where each packet comprises
one video frame).
3. Receivers report the fraction of RTP data packets lost during the
RTCP reporting interval, and the cumulative number of RTP packets
lost over the entire RTP session.
These congestion signals limit the possible circuit breakers, since
they give only limited visibility into the behaviour of the network.
Perkins & Singh Expires April 30, 2015 [Page 6]
Internet-Draft RTP Circuit Breakers October 2014
RTT estimates are widely used in congestion control algorithms, as a
proxy for queuing delay measures in delay-based congestion control or
to determine connection timeouts. RTT estimates derived from RTCP SR
and RR packets sent according to the RTP/AVP timing rules are too
infrequent to be useful though, and don't give enough information to
distinguish a delay change due to routing updates from queuing delay
caused by congestion. Accordingly, we cannot use the RTT estimate
alone as an RTP circuit breaker.
Increased jitter can be a signal of transient network congestion, but
in the highly aggregated form reported in RTCP RR packets, it offers
insufficient information to estimate the extent or persistence of
congestion. Jitter reports are a useful early warning of potential
network congestion, but provide an insufficiently strong signal to be
used as a circuit breaker.
The remaining congestion signals are the packet loss fraction and the
cumulative number of packets lost. If considered carefully, these
can be effective indicators that congestion is occurring in networks
where packet loss is primarily due to queue overflows, although loss
caused by non-congestive packet corruption can distort the result in
some networks. TCP congestion control [RFC5681] intentionally tries
to fill the router queues, and uses the resulting packet loss as
congestion feedback. An RTP flow competing with TCP traffic will
therefore expect to see a non-zero packet loss fraction that has to
be related to TCP dynamics to estimate available capacity. This
behaviour of TCP is reflected in the congestion circuit breaker
below, and will affect the design of any RTP congestion control
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.
Perkins & Singh Expires April 30, 2015 [Page 7]
Internet-Draft RTP Circuit Breakers October 2014
4.1. RTP/AVP Circuit Breaker #1: Media Timeout
If RTP data packets are being sent, but the RTCP SR or RR packets
reporting on that SSRC indicate a non-increasing extended highest
sequence number received, this is an indication that those RTP data
packets are not reaching the receiver. This could be a short-term
issue affecting only a few packets, perhaps caused by a slow-to-open
firewall or a transient connectivity problem, but if the issue
persists, it is a sign of a more ongoing and significant problem.
Accordingly, if a sender of RTP data packets receives three 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 consecutive 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 three 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 three 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
Perkins & Singh Expires April 30, 2015 [Page 8]
Internet-Draft RTP Circuit Breakers October 2014
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
participants in an RTP session will timeout and remove an RTP sender
from the list of active RTP senders if no RTP data packets have been
received from that RTP sender within the last two RTCP reporting
intervals. Using a timeout of three RTCP reporting intervals is
therefore large enough that the other participants will have timed
out the sender if a network problem stops the data packets it is
sending from reaching the receivers, even allowing for loss of some
RTCP packets.
If a sender is transmitting a large number of RTP media streams, such
that the corresponding RTCP SR or RR packets are too large to fit
into the network MTU, the receiver will generate RTCP SR or RR
packets in a round-robin manner. In this case, the sender SHOULD
treat receipt of an RTCP SR or RR packet corresponding to any SSRC it
sent on the same 5-tuple of source and destination IP address, port,
and protocol, as an indication that the receiver and return path are
working, preventing the RTCP timeout circuit breaker from triggering.
4.3. RTP/AVP Circuit Breaker #3: Congestion
If RTP data packets are being sent, and the corresponding RTCP SR or
RR packets show non-zero packet loss fraction and increasing extended
highest sequence number received, then those RTP data packets are
arriving at the receiver, but some degree of congestion is occurring.
The RTP/AVP profile [RFC3551] states that:
If best-effort service is being used, RTP receivers SHOULD monitor
packet loss to ensure that the packet loss rate is within
acceptable parameters. Packet loss is considered acceptable if a
TCP flow across the same network path and experiencing the same
network conditions would achieve an average throughput, measured
on a reasonable time scale, that is not less than the RTP flow is
achieving. This condition can be satisfied by implementing
congestion control mechanisms to adapt the transmission rate (or
the number of layers subscribed for a layered multicast session),
or by arranging for a receiver to leave the session if the loss
rate is unacceptably high.
The comparison to TCP cannot be specified exactly, but is intended
as an "order-of-magnitude" comparison in time scale and
Perkins & Singh Expires April 30, 2015 [Page 9]
Internet-Draft RTP Circuit Breakers October 2014
throughput. The time scale on which TCP throughput is measured is
the round-trip time of the connection. In essence, this
requirement states that it is not acceptable to deploy an
application (using RTP or any other transport protocol) on the
best-effort Internet which consumes bandwidth arbitrarily and does
not compete fairly with TCP within an order of magnitude.
The phase "order of magnitude" in the above means within a factor of
ten, approximately. In order to implement this, it is necessary to
estimate the throughput a TCP connection would achieve over the path.
For a long-lived TCP Reno connection, it has been shown that the TCP
throughput can be estimated using the following equation [Padhye]:
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 the second term on
the denominator is small, so this formula can be approximated with
reasonable accuracy as follows [Mathis]:
s
X = -----------------
R * sqrt(2*b*p/3)
Perkins & Singh Expires April 30, 2015 [Page 10]
Internet-Draft RTP Circuit Breakers October 2014
It is RECOMMENDED that this simplified throughout equation be used,
since the reduction in accuracy is small, and it is much simpler to
calculate than the full equation. Measurements have shown that the
simplified TCP throughput equation is effective as an RTP circuit
breaker for multimedia flows sent to hosts on residential networks
using ADSL and cable modem links [Singh]. The data shows that the
full TCP throughput equation tends to be more sensitive to packet
loss and triggers the RTP circuit breaker earlier than the simplified
equation. Implementations that desire this extra sensitivity MAY use
the full TCP throughput equation in the RTP circuit breaker. Initial
measurements in LTE networks have shown that the extra sensitivity is
helpful in that environment, with the full TCP throughput equation
giving a more balanced circuit breaker response than the simplified
TCP equation [Sarker]; other networks might see similar behaviour.
No matter what TCP throughput equation is chosen, two parameters need
to be estimated and reported to the sender in order to calculate the
throughput: the round trip time, R, and the loss event rate, p (the
packet size, s, is known to the sender). The round trip time can be
estimated from RTCP SR and RR packets. This is done too infrequently
for accurate statistics, but is the best that can be done with the
standard RTCP mechanisms.
Report blocks in RTCP SR or RR packets contain the packet loss
fraction, rather than the loss event rate, so p cannot be reported
(TCP typically treats the loss of multiple packets within a single
RTT as one loss event, but RTCP RR packets report the overall
fraction of packets lost, and does not report when the packet losses
occurred). Using the loss fraction in place of the loss event rate
can overestimate the loss. We believe that this overestimate will
not be significant, given that we are only interested in order of
magnitude comparison ([Floyd] section 3.2.1 shows that the difference
is small for steady-state conditions and random loss, but using the
loss fraction is more conservative in the case of bursty loss).
The congestion circuit breaker is therefore: when a sender 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 three consecutive RTCP reporting intervals, the sender
SHOULD cease transmission (see Section 4.5).
Perkins & Singh Expires April 30, 2015 [Page 11]
Internet-Draft RTP Circuit Breakers October 2014
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, provided it takes care not to cause the problem to recur
("reasonable" is intentionally not defined here).
The congestion circuit breaker depends on the fraction of RTP data
packets lost in a reporting interval. If the number of packets sent
in the reporting interval is too low, this statistic loses meaning,
and it is possible that a sampling error can give the appearance of
high packet loss rates. Following the guidelines in [RFC5405], an
RTP sender that sends not more than one RTP packet per RTT MAY ignore
a single trigger of the congestion circuit breaker, on the basis that
the packet loss rate estimate is unreliable with so few samples.
However, if the congestion circuit breaker triggers again after the
following three RTCP reporting intervals (i.e., if there have been
six or more consecutive RTCP reporting intervals where the actual
sending rate is more than ten times the estimated sending rate
derived from the TCP throughput equation), then the sender SHOULD
cease transmission (see Section 4.5).
The RTCP reporting interval of the media sender does not affect how
quickly congestion circuit breaker can trigger. The timing is based
on the RTCP reporting interval of the receiver that generates the SR/
RR packets from which the loss rate and RTT estimate are derived
(note that RTCP requires all participants in a session to have
similar reporting intervals, else the participant timeout rules in
[RFC3550] will not work, so this interval is likely similar to that
of the sender). If the incoming RTCP SR or RR packets are using a
reduced minimum RTCP reporting interval (as specified in Section 6.2
of RFC 3550 [RFC3550] or the RTP/AVPF profile [RFC4585]), then that
reduced RTCP reporting interval is used when determining if the
circuit breaker is triggered.
As in Section 4.1 and Section 4.2, we use three 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.
Perkins & Singh Expires April 30, 2015 [Page 12]
Internet-Draft RTP Circuit Breakers October 2014
If there are more media streams that can be reported in a single RTCP
SR or RR packet, or if the size of a complete RTCP SR or RR packet
exceeds the network MTU, then the receiver will report on a subset of
sources in each reporting interval, with the subsets selected round-
robin across multiple intervals so that all sources are eventually
reported [RFC3550]. When generating such round-robin RTCP reports,
priority SHOULD be given to reports on sources that have high packet
loss rates, to ensure that senders are aware of network congestion
they are causing (this is an update to [RFC3550]).
4.4. RTP/AVP Circuit Breaker #4: Media Usability
Applications that use RTP are generally tolerant to some amount of
packet loss. How much packet loss can be tolerated will depend on
the application, media codec, and the amount of error correction and
packet loss concealment that is applied. There is an upper bound on
the amount of loss can be corrected, however, beyond which the media
becomes unusable. Similarly, many applications have some upper bound
on the media capture to play-out latency that can be tolerated before
the application becomes unusable. The latency bound will depend on
the application, but typical values can range from the order of a few
hundred milliseconds for voice telephony and interactive conferencing
applications, up to several seconds for some video-on-demand systems.
As a final circuit breaker, RTP senders SHOULD monitor the reported
packet loss and delay to estimate whether the media is likely to be
suitable for the intended purpose. If the packet loss rate and/or
latency is such that the media has become unusable, and has remained
unusable for a significant time period, then the application SHOULD
cease transmission. Similarly, receivers SHOULD monitor the quality
of the media they receive, and if the quality is unusable for a
significant time period, they SHOULD terminate the session. This
memo intentionally does not define a bound on the packet loss rate or
latency that will result in unusable media, nor does it specify what
time period is deemed significant, as these are highly application
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.
Perkins & Singh Expires April 30, 2015 [Page 13]
Internet-Draft RTP Circuit Breakers October 2014
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.
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.
Reduced-size RTCP reports sent under the RTP/AVPF early feedback
rules that do not contain an RTCP SR or RR packet MUST be ignored by
the congestion circuit breaker (they do not contain the information
needed by the congestion circuit breaker algorithm), but MUST be
counted as received packets for the RTCP timeout circuit breaker.
Reduced-size RTCP reports sent under the RTP/AVPF early feedback
rules that contain RTCP SR or RR packets MUST be processed by the
Perkins & Singh Expires April 30, 2015 [Page 14]
Internet-Draft RTP Circuit Breakers October 2014
congestion circuit breaker as if they were sent as regular RTCP
reports, and counted towards the circuit breaker conditions specified
in Section 4 of this memo. This will potentially make the RTP
circuit breaker 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 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 Extended Reports (XR)
RTCP Extended Report (XR) blocks provide additional reception quality
metrics, but do not change the RTCP timing rules. Some of the RTCP
XR blocks provide information that might be useful for congestion
control purposes, others provided non-congestion-related metrics.
With the exception of RTCP XR ECN Summary Reports (see Section 8),
the presence of RTCP XR blocks in a compound RTCP packet does not
affect the RTP circuit breaker algorithm. For consistency and ease
of implementation, only the reception report blocks contained in RTCP
SR packets, RTCP RR packets, or RTCP XR ECN Summary Report packets,
are used by the RTP circuit breaker algorithm.
7. Impact of RTCP Reporting Groups
An optimisation for grouping RTCP reception statistics and other
feedback in RTP sessions with large numbers of participants is given
in [I-D.ietf-avtcore-rtp-multi-stream-optimisation]. This allows one
SSRC to act as a representative that sends reports on behalf of other
SSRCs that are co-located in the same endpoint and see identical
reception quality. When running the circuit breaker algorithms, an
endpoint MUST treat a reception report from the representative of the
reporting group as if a reception report was received from all
members of that group.
Perkins & Singh Expires April 30, 2015 [Page 15]
Internet-Draft RTP Circuit Breakers October 2014
8. Impact of Explicit Congestion Notification (ECN)
The use of ECN for RTP flows does not affect the media timeout RTP
circuit breaker (Section 4.1) or the RTCP timeout circuit breaker
(Section 4.2), since these are both connectivity checks that simply
determinate if any packets are being received.
ECN-CE marked packets SHOULD be treated as if it were lost for the
purposes of congestion control, when determining the optimal media
sending rate for an RTP flow. If an RTP sender has negotiated ECN
support for an RTP session, and has successfully initiated ECN use on
the path to the receiver [RFC6679], then ECN-CE marked packets SHOULD
be treated as if they were lost when calculating if the congestion-
based RTP circuit breaker (Section 4.3) has been met. The count of
ECN-CE marked RTP packets is returned in RTCP XR ECN summary report
packets if support for ECN has been initiated for an RTP session.
9. Impact of Layered Coding
Layered coding is a method of encoding a single media stream into
disparate layers, such that a receiver can decode a subset of the
layers to vary the quality of the media. Layered coding is often
used to aid congestion control in group communication systems, where
a different subset of the layers is sent to each receiver, depending
on the available network capacity.
Media using layered coding can be transported within RTP in several
ways: each layer can be sent as a separate RTP session; each layer
can be sent using a separate SSRC within a single RTP session; or
each layer can be identified by some payload-specific header field,
with all layers being sent by a single SSRC within a single RTP
session. The choice depends on the features provided by the RTP
payload format for the layered encoding, and on the application
requirements.
The RTP circuit breaker operates on a per-RTP session basis. If a
layered encoding is split across multiple RTP sessions, then each
session MUST be treated independently for the RTP circuit breaker.
Within an RTP session, if an application that sends a layered media
encoding using a single SSRC, with the layers identified using some
payload-specific mechanism, then it MUST apply the RTP circuit
breaker to that layered flow as a whole, considering RTCP feedback
for the SSRC sending the layered flow and applying the RTP circuit
breaker as usual.
Within an RTP session, if the layered coding is sent using several
SSRC values within a single RTP session, the flows for those SSRCs
Perkins & Singh Expires April 30, 2015 [Page 16]
Internet-Draft RTP Circuit Breakers October 2014
MAY be treated together, so that a circuit breaker trigger for any
SSRC in the layered media flow causes the entire layered flow to
either cease transmission or reduce its sending rate by a factor of
ten. The intent of this is to allow a layered flow to reduce its
sending rate by dropping higher layers if the circuit breaker fails,
rather than requiring the layer that triggered the RTP circuit
breaker to cease transmission (layers are additive in many layered
codecs, so forcing a lower layer to cease transmission while allowing
higher layers to continue is pointless).
10. Security Considerations
The security considerations of [RFC3550] apply.
If the RTP/AVPF profile is used to provide rapid RTCP feedback, the
security considerations of [RFC4585] apply. If ECN feedback for RTP
over UDP/IP is used, the security considerations of [RFC6679] apply.
If non-authenticated RTCP reports are used, an on-path attacker can
trivially generate fake RTCP packets that indicate high packet loss
rates, causing the circuit breaker to trigger and disrupting an RTP
session. This is somewhat more difficult for an off-path attacker,
due to the need to guess the randomly chosen RTP SSRC value and the
RTP sequence number. This attack can be avoided if RTCP packets are
authenticated; authentication options are discussed in [RFC7201].
Timely operation of the RTP circuit breaker depends on the choice of
RTCP reporting interval. If the receiver has a reporting interval
that is overly long, then the responsiveness of the circuit breaker
decreases. In the limit, the RTP circuit breaker can be disabled for
all practical purposes by configuring an RTCP reporting interval that
is many minutes duration. This issue is not specific to the circuit
breaker: long RTCP reporting intervals also prevent reception quality
reports, feedback messages, codec control messages, etc., from being
used. Implementations SHOULD impose an upper limit on the RTCP
reporting interval they are willing to negotiate (based on the
session bandwidth and RTCP bandwidth fraction) when using the RTP
circuit breaker. An upper limit on the reporting interval on the
order of 10 seconds is a reasonable bound.
11. IANA Considerations
There are no actions for IANA.
12. Open Issues
o Should the number of RTCP reporting intervals needed to trigger
the media timeout and congestion circuit breakers scale with the
Perkins & Singh Expires April 30, 2015 [Page 17]
Internet-Draft RTP Circuit Breakers October 2014
duration of the RTCP reporting interval, so the circuit breaker
triggers after a fixed duration, rather than after a fixed number
of reporting intervals?
13. Acknowledgements
The authors would like to thank Bernard Aboba, Harald Alvestrand,
Gorry Fairhurst, Kevin Gross, Cullen Jennings, Randell Jesup,
Jonathan Lennox, Matt Mathis, Stephen McQuistin, Eric Rescorla,
Abheek Saha, and Fabio Verdicchio, for their valuable feedback.
14. References
14.1. 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.
14.2. 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]
Perkins & Singh Expires April 30, 2015 [Page 18]
Internet-Draft RTP Circuit Breakers October 2014
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-04 (work
in progress), August 2014.
[Mathis] Mathis, M., Semke, J., Mahdavi, J., and T. Ott, "The
macroscopic behavior of the TCP congestion avoidance
algorithm", ACM SIGCOMM Computer Communication Review
27(3), DOI 10.1145/263932.264023, July 1997.
[Padhye] Padhye, J., Firoiu, V., Towsley, D., and J. Kurose,
"Modeling TCP Throughput: A Simple Model and its Empirical
Validation", Proceedings of the ACM SIGCOMM conference,
1998, DOI 10.1145/285237.285291, August 1998.
[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP", RFC
3168, September 2001.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, February 2008.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008.
[RFC5405] Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
for Application Designers", BCP 145, RFC 5405, November
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.
Perkins & Singh Expires April 30, 2015 [Page 19]
Internet-Draft RTP Circuit Breakers October 2014
[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
Perkins & Singh Expires April 30, 2015 [Page 20]
Internet-Draft RTP Circuit Breakers October 2014
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/
Perkins & Singh Expires April 30, 2015 [Page 21]
Html markup produced by rfcmarkup 1.129c, available from
https://tools.ietf.org/tools/rfcmarkup/