Internet Engineering Task Force (IETF) C. Perkins
Request for Comments: 8083 University of Glasgow
Updates: 3550 V. Singh
Category: Standards Track callstats.io
ISSN: 2070-1721 March 2017
Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions
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 these applications, then network congestion can
lead to uncontrolled packet loss and a resulting deterioration of the
user's multimedia experience. The congestion control algorithm acts
as a safety measure by stopping RTP flows from using excessive
resources and protecting the network from overload. At the time of
this writing, however, while there are several proprietary solutions,
there is no standard algorithm for congestion control of interactive
RTP flows.
This document does not propose a congestion control algorithm. It
instead defines a minimal set of RTP circuit breakers: conditions
under which an RTP sender needs to stop transmitting media data to
protect the network from excessive congestion. It is expected that,
in the absence of long-lived excessive congestion, RTP applications
running on best-effort IP networks will be able to operate without
triggering these circuit breakers. To avoid triggering the RTP
circuit breaker, any Standards Track congestion control algorithms
defined for RTP will need to operate within the envelope set by these
RTP circuit breaker algorithms.
Status of This Memo
This is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc8083.
Copyright Notice
Copyright (c) 2017 IETF Trust and the persons identified as the
document authors. All rights reserved.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Background . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile . 8
4.1. RTP/AVP Circuit Breaker #1: RTCP Timeout . . . . . . . . 10
4.2. RTP/AVP Circuit Breaker #2: Media Timeout . . . . . . . . 11
4.3. RTP/AVP Circuit Breaker #3: Congestion . . . . . . . . . 12
4.4. RTP/AVP Circuit Breaker #4: Media Usability . . . . . . . 16
4.5. Ceasing Transmission . . . . . . . . . . . . . . . . . . 17
5. RTP Circuit Breakers and the RTP/AVPF and RTP/SAVPF Profiles 18
6. Impact of RTCP Extended Reports (XR) . . . . . . . . . . . . 19
7. Impact of Explicit Congestion Notification (ECN) . . . . . . 19
8. Impact of Bundled Media and Layered Coding . . . . . . . . . 20
9. Security Considerations . . . . . . . . . . . . . . . . . . . 20
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 21
10.1. Normative References . . . . . . . . . . . . . . . . . . 21
10.2. Informative References . . . . . . . . . . . . . . . . . 22
Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . . 25
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 25
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. Instead, we propose a small set of RTP circuit breakers:
mechanisms that terminate RTP flows in conditions under which there
is general agreement that serious network congestion is occurring.
The RTP circuit breakers proposed in this memo are a specific
instance of the general class of network transport circuit breakers
[RFC8084] designed to act as a protection mechanism of last resort to
avoid persistent excessive congestion. To avoid triggering the RTP
circuit breaker, any Standards Track congestion control algorithms
defined for RTP will need to operate within the envelope set by the
RTP circuit breaker algorithms defined by this memo.
2. 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 does not use more capacity than is
available along the network path. 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 capacity estimator for
a group of receivers will differ from that for a single receiver, and
because multicast congestion control has to consider issues of
fairness across groups of receivers that do not apply to unicast
flows.
Congestion control for unicast RTP traffic can be implemented in one
of two places in the protocol stack. One approach is to run the RTP
traffic over a congestion-controlled transport protocol (for example,
over TCP), and to adapt the media encoding to match the dictates of
the transport-layer congestion control algorithm. This is safe for
the network but can be suboptimal for the media quality unless the
transport protocol is designed to support real-time media flows. We
do not consider this class of applications further in this memo, as
their network safety is guaranteed by the underlying transport.
Alternatively, RTP flows can be run over a non-congestion-controlled
transport protocol (for example, UDP) performing rate adaptation at
the application layer based on RTP Control Protocol (RTCP) feedback.
With a well-designed, network-aware application, this allows highly
effective media quality adaptation, but there is potential to cause
persistent congestion in the network 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 (this could
potentially be augmented by mechanisms such as Congestion
Exposure (ConEx) [RFC7713] or other future protocol extensions
for network signaling of congestion); 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 Receiver 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 Synchronization Sources (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 is on the order of a few seconds on average; although it
varies with the session bandwidth, it is randomized to avoid
synchronization of reports from multiple receivers. The interval
can be less than a second in a high-bandwidth session. 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] as well as 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 the 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 signaling.
o Finally, ECN for RTP over UDP [RFC6679] can be used to provide
feedback on the number of packets that received an ECN-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 [RFC5450]. Accurate transmission 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.
3. 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.
The definition of the RTP circuit breaker is specified in terms of
the following variables:
o Td is the deterministic RTCP reporting interval, as defined in
Section 6.3.1 of [RFC3550].
o Tdr is the sender's estimate of the deterministic RTCP reporting
interval, Td, calculated by a receiver of the data it is sending.
Tdr is not known at the sender but can be estimated by executing
the algorithm in Section 6.2 of [RFC3550] using the average RTCP
packet size seen at the sender, the number of members reported in
the receiver's SR/RR report blocks, and whether the receiver is
sending SR or RR packets. Tdr is recalculated when each new RTCP
SR/RR report is received, but the media timeout circuit breaker
(see Section 4.2) is only reconsidered when Tdr increases.
o Tr is the network round-trip time, which is calculated by the
sender using the algorithm in Section 6.4.1 of [RFC3550] and is
smoothed using an exponentially weighted moving average as
Tr = (0.8 * Tr) + (0.2 * Tr_new) where Tr_new is the latest RTT
estimate obtained from an RTCP report. The weight is chosen so
old estimates decay over k intervals.
o k is the non-reporting threshold (see Section 4.2).
o Tf is the media framing interval at the sender. For applications
sending at a constant frame rate, Tf is the inter-frame interval.
For applications that switch between a small set of possible frame
rates (for example, when sending speech with comfort noise, such
that comfort noise frames are sent less often than speech frames),
Tf is set to the longest of the inter-frame intervals of the
different frame rates. For applications that send periodic frames
but dynamically vary their frame rate, Tf is set to the largest
inter-frame interval used in the last 10 seconds. For
applications that send less than one frame every 10 seconds, or
that have no concept of periodic frames (e.g., text conversation
[RFC4103], or pointer events [RFC2862]), when each frame is sent,
Tf is set to the time interval since the previous frame.
o G is the frame group size. That is, the number of frames that are
coded together based on a particular sending rate setting. If the
codec used by the sender can change its rate on each frame, then G
= 1; otherwise, G is set to the number of frames before the codec
can adjust to the new rate. For codecs that have the concept of a
Group of Pictures (GOP), G is likely the GOP length.
o T_rr_interval is the minimal interval between RTCP reports, as
defined in Section 3.4 of [RFC4585]; it is only meaningful for
implementations of RTP/AVPF profile [RFC4585] or the RTP/SAVPF
profile [RFC5124].
o X is the estimated throughput a TCP connection would achieve over
a path, in bytes per second.
o s is the size of RTP packets being sent, in bytes. If the RTP
packets being sent vary in size, then the average size over the
packet comprising the last 4 * G frames MUST be used (this is
intended to be comparable to the four loss intervals used in
[RFC5348]).
o p is the loss event rate, between 0.0 and 1.0, that would be seen
by a TCP connection over a particular path. When used in the RTP
congestion circuit breaker, this is approximated as described in
Section 4.3.
o t_RTO is the retransmission timeout value that would be used by a
TCP connection over a particular path, in seconds. This MUST be
approximated using t_RTO = 4 * Tr when used as part of the RTP
congestion circuit breaker.
o b is the number of packets that are acknowledged by a single TCP
acknowledgement. Following [RFC5348], it is RECOMMENDED that the
value b = 1 is used as part of the RTP congestion circuit breaker.
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, will be
unable to 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.
The RTCP timeout circuit breaker (Section 4.1) will trigger if an
implementation of this memo attempts to interwork with an endpoint
that does not support RTCP. Implementations that sometimes need to
interwork with endpoints that do not support RTCP need to disable the
RTP circuit breakers if they don't receive some confirmation via
signaling that the remote endpoint implements RTCP (the presence of a
Session Description Protocol (SDP) "a=rtcp:" attribute in an answer
might be such an indication). The RTP circuit breaker SHOULD NOT be
disabled 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
[RFC8084].
Three potential congestion signals are available from the basic RTCP
SR/RR packets and are reported for each 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
(Section 6.4.4. of [RFC3550]), 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 behavior of the network.
RTT estimates are widely used in congestion control algorithms as a
proxy for queuing delay measures in delay-based congestion control or
to determine connection timeouts. RTT estimates derived from RTCP SR
and RR packets sent according to the RTP/AVP timing rules are too
infrequent to be useful for congestion control 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, and over
an appropriate time frame to distinguish transient problems from long
term issues [RFC8084], these can be effective indicators that
persistent excessive 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, and some
variation in queuing latency, in normal operation when sharing a path
with other flows, which needs to be accounted for when determining
the circuit breaker threshold [RFC8084]. This behavior 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: RTCP Timeout
An RTCP timeout 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.
An RTP sender that has not received any RTCP SR or RTCP RR packets
reporting on the SSRC it is using, for a time period of at least
three times its deterministic RTCP reporting interval, Td (where Td
is calculated without the randomization factor and using the fixed
minimum interval of Tmin=5 seconds), SHOULD cease transmission (see
Section 4.5). The rationale for this choice of timeout is as
described in Section 6.2 of [RFC3550] ("so that implementations which
do not use the reduced value for transmitting RTCP packets are not
timed out by other participants prematurely") and has been updated by
Section 6.1.4 of [RFC8108] to account for the use of the RTP/AVPF
profile [RFC4585] or the RTP/SAVPF profile [RFC5124].
To reduce the risk of premature timeout, implementations SHOULD NOT
configure the RTCP bandwidth such that Td is larger than 5 seconds.
Similarly, implementations that use the RTP/AVPF profile [RFC4585] or
the RTP/SAVPF profile [RFC5124] SHOULD NOT configure T_rr_interval to
values larger than 4 seconds (the reduced limit for T_rr_interval
follows Section 6.1.3 of [RFC8108]).
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 and thus preventing the RTCP timeout circuit breaker from
triggering.
4.2. RTP/AVP Circuit Breaker #2: 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 RTP 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 (a
"media timeout").
The time needed to declare a media timeout depends on the parameters
Tdr, Tr, Tf, and on the non-reporting threshold k. The value of k is
chosen so that when Tdr is large compared to Tr and Tf, receipt of at
least k RTCP reports with non-increasing extended highest sequence
number received gives reasonable assurance that the forward path has
failed and that the RTP data packets have not been lost by chance.
The RECOMMENDED value for k is 5 reports.
When Tdr < Tf, then RTP data packets are being sent at a rate less
than one per RTCP reporting interval of the receiver, so the extended
highest sequence number received can be expected to be non-increasing
for some receiver RTCP reporting intervals. Similarly, when
Tdr < Tr, some receiver RTCP reporting intervals might pass before
the RTP data packets arrive at the receiver, also leading to reports
where the extended highest sequence number received is non-
increasing. Both issues require the media timeout interval to be
scaled relative to the threshold, k.
The media timeout RTP circuit breaker is therefore as follows. When
starting sending, calculate MEDIA_TIMEOUT using:
MEDIA_TIMEOUT = ceil(k * max(Tf, Tr, Tdr) / Tdr)
When a sender receives an RTCP packet that indicates reception of the
media it has been sending, then it cancels the media timeout circuit
breaker. If it is still sending, then it MUST calculate a new value
for MEDIA_TIMEOUT and set a new media timeout circuit breaker.
If a sender receives an RTCP packet indicating that its media was not
received, it MUST calculate a new value for MEDIA_TIMEOUT. If the
new value is larger than the previous, it replaces MEDIA_TIMEOUT with
the new value, extending the media timeout circuit breaker;
otherwise, it keeps the original value of MEDIA_TIMEOUT. This
process is known as reconsidering the media timeout circuit breaker.
If MEDIA_TIMEOUT consecutive RTCP packets are received indicating
that the media being sent was not received, and the media timeout
circuit breaker has not been canceled, then the media timeout circuit
breaker triggers. When the media timeout circuit breaker triggers,
the sender SHOULD cease transmission (see Section 4.5).
When stopping sending an RTP stream, a sender MUST cancel the
corresponding media timeout circuit breaker.
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 timescale, that is not less than [the throughput]
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 timescale and throughput.
The timescale 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 bulk TCP connection would achieve over the
path. For a long-lived TCP Reno connection, it has been shown that
the TCP throughput, X, in bytes per second, can be estimated as
follows [Padhye]:
s
X = -------------------------------------------------------------
Tr*sqrt(2*b*p/3)+(t_RTO * (3*sqrt(3*b*p/8) * p * (1+32*p*p)))
This is the same approach to estimated TCP throughput that is used in
[RFC5348]. 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 = ----------------
Tr*sqrt(2*b*p/3)
It is RECOMMENDED that this simplified throughput 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 Asymmetric Digital Subscriber Line (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 behavior.
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, Tr, 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 do 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 (Section 3.2.1 of [Floyd] 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 as follows. When a
sender that is transmitting at least one RTP packet every max(Tdr,
Tr) seconds receives an RTCP SR or RR packet that contains a report
block for an SSRC it is using, the sender MUST record the value of
the fraction lost field from the report block, and the time since the
last report block was received, for that SSRC. If more than
CB_INTERVAL (see below) report blocks have been received for that
SSRC, the sender MUST calculate the average fraction lost over the
last CB_INTERVAL reporting intervals and then estimate the TCP
throughput that would be achieved over the path using the chosen TCP
throughput equation and the measured values of the round-trip time,
Tr, the loss event rate, p (approximated by the average fraction
lost, as is described below), and the packet size, s. The estimate
of the TCP throughput, X, is then compared with the actual sending
rate of the RTP stream. If the actual sending rate of the RTP stream
is more than 10 * X, then the congestion circuit breaker is
triggered.
The average fraction lost is calculated based on the sum (over the
last CB_INTERVAL reporting intervals) of the fraction lost in each
reporting interval that is then multiplied by the duration of the
corresponding reporting interval and then divided by the total
duration of the last CB_INTERVAL reporting intervals. The
CB_INTERVAL parameter is set to:
CB_INTERVAL =
ceil(3*min(max(10*G*Tf, 10*Tr, 3*Tdr), max(15, 3*Td))/(3*Tdr))
The parameters that feed into CB_INTERVAL are chosen to give the
congestion control algorithm time to react to congestion. They give
at least three RTCP reports, ten round trip times, and ten groups of
frames to adjust the rate to reduce the congestion to a reasonable
level. It is expected that a responsive congestion control algorithm
will begin to respond with the next group of frames after it receives
indication of congestion, so CB_INTERVAL ought to be a much longer
interval than the congestion response.
If the RTP/AVPF profile [RFC4585] or the RTP/SAVPF [RFC5124] is used,
and the T_rr_interval parameter is used to reduce the frequency of
regular RTCP reports, then the value of Tdr in the above expression
for the CB_INTERVAL parameter MUST be replaced by max(T_rr_interval,
Tdr).
The CB_INTERVAL parameter is calculated on joining the session, and
recalculated on receipt of each RTCP packet, after checking whether
the media timeout circuit breaker or the congestion circuit breaker
has been triggered.
To ensure a timely response to persistent congestion, implementations
SHOULD NOT configure the RTCP bandwidth such that Tdr is larger than
5 seconds. Similarly, implementations that use the RTP/AVPF profile
[RFC4585] or the RTP/SAVPF profile [RFC5124] SHOULD NOT configure
T_rr_interval to values larger than 4 seconds (the reduced limit for
T_rr_interval follows Section 6.1.3 of [RFC8108]).
The rationale for enforcing a minimum sending rate below which the
congestion circuit breaker will not trigger is to avoid spurious
circuit breaker triggers when the number of packets sent per RTCP
reporting interval is small, and hence, the fraction lost samples are
subject to measurement artifacts. The bound of at least one packet
every max(Tdr, Tr) seconds is derived from the one packet per RTT
minimum sending rate of TCP [RFC8085], which is adapted for use with
RTP where the RTCP reporting interval is decoupled from the network
RTT.
When the congestion circuit breaker is triggered, the sender SHOULD
cease transmission (see Section 4.5). However, if the sender is able
to reduce its sending rate by a factor of (approximately) ten, then
it MAY first reduce its sending rate by this factor (or some larger
amount) to see if that resolves the congestion. If the sending rate
is reduced in this way and the congestion circuit breaker triggers
again after the next CB_INTERVAL RTCP reporting intervals, the sender
MUST then cease transmission. An example of such a rate reduction
might be a video conferencing system that backs off to sending audio
only before completely dropping the call. If such a reduction in
sending rate resolves the congestion problem, the sender MAY
gradually increase the rate at which it sends data after a reasonable
amount of time has passed, provided it takes care not to cause the
problem to recur ("reasonable" is intentionally not defined here
since it depends on the application, media codec, and congestion
control algorithm).
The RTCP reporting interval of the media sender does not affect how
quickly the 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.
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 that 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, as these are highly
application dependent. Similarly, the time period that is considered
significant is application dependent but is likely on the order of
seconds, or tens of seconds.
Sending media that suffers from such high packet loss or latency that
it is unusable at the receiver is both wasteful of resources and is
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 to prevent the application sending unnecessary traffic that
might disrupt other uses of the network. 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. This
could mean stopping a single RTP flow or it could mean that multiple
bundled RTP flows are stopped. The intention is that the application
will stop sending RTP data packets on a particular 5-tuple (transport
protocol, source and destination ports, source and destination IP
addresses) until whatever network problem that triggered the RTP
circuit breaker has dissipated. RTP flows halted by the circuit
breaker SHOULD NOT be restarted automatically unless the sender has
received information that the congestion has dissipated or can
reasonably be expected to have dissipated. What could trigger this
expectation is necessarily application dependent, but could be, for
example, an indication that a competing flow has finished and freed
up some capacity, or for an application running on a mobile device it
could indicate that the device moved to a new location so the flow
would traverse a different path if it were restarted. Ideally, a
human user will be involved in the decision to try to restart the
flow since that user will eventually give up if the flows repeatedly
trigger the circuit breaker. This will help avoid problems with
automatic redial systems from congesting the network.
It is recognized that the RTP implementation in some systems might
not be able to determine if a flow setup request was initiated by a
human user or automatically by some scripted higher-level component
of the system. These implementations MUST rate limit attempts to
restart a flow on the same 5-tuple as used by a flow that triggered
the circuit breaker so that the reaction to a triggered circuit
breaker lasts for at least the triggering interval [RFC8084].
The RTP circuit breaker will only trigger, and cease transmission,
for media flows subject to long-term persistent congestion. Such
flows are likely to have poor quality and usability for some time
before the circuit breaker triggers. Implementations can monitor
RTCP Receiver Report blocks being returned for their media flows and
might find it beneficial to use this information to provide a user
interface cue that problems are occurring in advance of the circuit
breaker triggering.
5. RTP Circuit Breakers and the RTP/AVPF and RTP/SAVPF Profiles
Use of the Extended RTP Profile for RTCP-based Feedback (RTP/AVPF)
[RFC4585] allows receivers to send early RTCP reports, in some cases,
to inform the sender about particular events in the media stream.
There are several use cases for such early RTCP reports, including
providing rapid feedback to a sender about the onset of congestion.
The RTP/SAVPF Profile [RFC5124] is a secure variant of the RTP/AVPF
profile that is treated the same in the context of the RTP circuit
breaker. These feedback profiles are often used with non-compound
RTCP reports [RFC5506] to reduce the reporting overhead.
Receiving rapid feedback about congestion events potentially allows
congestion control algorithms to be more responsive and to better
adapt the media transmission to the limitations of the network. It
is expected that many RTP congestion control algorithms will adopt
the RTP/AVPF profile or the RTP/SAVPF profile for this reason and
thus define 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
congestion circuit breaker as if they were sent as regular RTCP
reports and counted towards the circuit breaker conditions specified
in Section 4 of this memo. This will potentially make the RTP
circuit breaker trigger earlier than it would if the RTP/AVPF profile
was not used.
When using ECN with RTP (see Section 7), 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 is sent in a
compound RTCP packet along with an RTCP SR/RR report packet. Reports
of ECN-CE packets sent as reduced-size RTCP ECN feedback packets
without an RTCP SR/RR packet MUST be ignored.
These rules are intended to allow the use of low-overhead RTP/AVPF
feedback for generic NACK messages without triggering the RTP circuit
breaker. This is expected to make such feedback suitable for RTP
congestion control algorithms that need to quickly report loss events
in between regular RTCP reports. The reaction to reduced-size RTCP
SR/RR packets is to allow such algorithms to send feedback that can
trigger the circuit breaker when desired.
The RTP/AVPF and RTP/SAVPF profiles include the T_rr_interval
parameter that can be used to adjust the regular RTCP reporting
interval. The use of the T_rr_interval parameter changes the
behavior of the RTP circuit breaker, as described in Section 4.
6. Impact of RTCP Extended Reports (XR)
RTCP Extended Report (XR) blocks provide additional reception quality
metrics, but do not change the RTCP timing rules. Some of the RTCP
XR blocks provide information that might be useful for congestion
control purposes, others provide non-congestion-related metrics.
With the exception of RTCP XR ECN Summary Reports (see Section 7),
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 receiver 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 Explicit Congestion Notification (ECN)
The use of ECN for RTP flows does not affect the RTCP timeout circuit
breaker (Section 4.1) or the media timeout circuit breaker
(Section 4.2) since these are both connectivity checks that simply
determinate if any packets are being received.
At the time of this writing, there's no consensus on how the receipt
of ECN feedback will impact the congestion circuit breaker
(Section 4.3) or indeed whether the congestion circuit breaker ought
to take ECN feedback into account. A future replacement of this memo
is expected to provide guidance for implementers.
For the media usability circuit breaker (Section 4.4), ECN-CE-marked
packets arrive at the receiver, and if they arrive in time, they will
be decoded and rendered as normal. Accordingly, receipt of such
packets ought not affect the usability of the media, and the arrival
of RTCP feedback indicating their receipt is not expected to impact
the operation of the media usability circuit breaker.
8. Impact of Bundled Media and Layered Coding
The RTP circuit breaker operates on a per-RTP session basis. An RTP
sender that participates in several RTP sessions MUST treat each RTP
session independently with regards to the RTP circuit breaker.
An RTP sender can generate several media streams within a single RTP
session, with each stream using a different SSRC. This can happen if
bundled media are in use when using simulcast or when using layered
media coding. By default, each SSRC will be treated independently by
the RTP circuit breaker. However, the sender MAY choose to treat the
flows (or a subset thereof) as a group such that a circuit breaker
trigger for one flow applies to the group of flows as a whole and
either causes the entire group to cease transmission or causes the
sending rate of the group to reduce by a factor of ten, depending on
the RTP circuit breaker triggered. Grouping flows in this way is
expected to be especially useful for layered flows sent using
multiple SSRCs as it allows the layered flow to react as a whole,
thus ceasing transmission on the enhancement layers first to reduce
sending rate, if necessary, rather than treating each layer
independently. Care needs to be taken if the different media streams
sent on a single transport-layer flow use different Differentiated
Services Code Point (DSCP) values [RFC7657] [WebRTC-QoS] since
congestion could be experienced differently depending on the DSCP
marking. Accordingly, RTP media streams with different DSCP values
SHOULD NOT be considered as a group when evaluating the RTP circuit
breaker conditions.
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.
If non-authenticated RTCP reports are used, an on-path attacker can
trivially generate fake RTCP packets that indicate high packet loss
rates and thus cause the circuit breaker to trigger and disrupt 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
has a duration of many minutes. This issue is not specific to the
circuit breaker: long RTCP reporting intervals also prevent reception
quality reports, feedback messages, codec control messages, etc.,
from being used. Implementations are expected to impose an upper
limit on the RTCP reporting interval they are willing to negotiate
(based on the session bandwidth and RTCP bandwidth fraction) when
using the RTP circuit breaker, as discussed in Section 4.3.
10. References
10.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
DOI 10.17487/RFC3551, July 2003,
<http://www.rfc-editor.org/info/rfc3551>.
[RFC3611] Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed.,
"RTP Control Protocol Extended Reports (RTCP XR)",
RFC 3611, DOI 10.17487/RFC3611, November 2003,
<http://www.rfc-editor.org/info/rfc3611>.
[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,
DOI 10.17487/RFC4585, July 2006,
<http://www.rfc-editor.org/info/rfc4585>.
[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification",
RFC 5348, DOI 10.17487/RFC5348, September 2008,
<http://www.rfc-editor.org/info/rfc5348>.
[RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
and K. Carlberg, "Explicit Congestion Notification (ECN)
for RTP over UDP", RFC 6679, DOI 10.17487/RFC6679, August
2012, <http://www.rfc-editor.org/info/rfc6679>.
10.2. Informative References
[Floyd] Floyd, S., Handley, M., Padhye, J., and J. Widmer,
"Equation-Based Congestion Control for Unicast
Applications", ACM SIGCOMM Computer Communication
Review, Volume 30, Issue 4, pages 43-56,
DOI 10.1145/347059.347397, August 2000.
[Mathis] Mathis, M., Semke, J., Mahdavi, J., and T. Ott, "The
Macroscopic Behavior of the TCP Congestion Avoidance
Algorithm", ACM SIGCOMM Computer Communication
Review, Volume 27, Issue 3, pages 67-82,
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", ACM SIGCOMM Computer Communication
Review Volume 30, Issue 4, pages 303-314,
DOI 10.1145/285237.285291, August 1998.
[RFC2862] Civanlar, M. and G. Cash, "RTP Payload Format for Real-
Time Pointers", RFC 2862, DOI 10.17487/RFC2862, June 2000,
<http://www.rfc-editor.org/info/rfc2862>.
[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP",
RFC 3168, DOI 10.17487/RFC3168, September 2001,
<http://www.rfc-editor.org/info/rfc3168>.
[RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text
Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005,
<http://www.rfc-editor.org/info/rfc4103>.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
February 2008, <http://www.rfc-editor.org/info/rfc5104>.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
2008, <http://www.rfc-editor.org/info/rfc5124>.
[RFC5450] Singer, D. and H. Desineni, "Transmission Time Offsets in
RTP Streams", RFC 5450, DOI 10.17487/RFC5450, March 2009,
<http://www.rfc-editor.org/info/rfc5450>.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
2009, <http://www.rfc-editor.org/info/rfc5506>.
[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,
<http://www.rfc-editor.org/info/rfc5681>.
[RFC6798] Clark, A. and Q. Wu, "RTP Control Protocol (RTCP) Extended
Report (XR) Block for Packet Delay Variation Metric
Reporting", RFC 6798, DOI 10.17487/RFC6798, November 2012,
<http://www.rfc-editor.org/info/rfc6798>.
[RFC6843] Clark, A., Gross, K., and Q. Wu, "RTP Control Protocol
(RTCP) Extended Report (XR) Block for Delay Metric
Reporting", RFC 6843, DOI 10.17487/RFC6843, January 2013,
<http://www.rfc-editor.org/info/rfc6843>.
[RFC6958] Clark, A., Zhang, S., Zhao, J., and Q. Wu, Ed., "RTP
Control Protocol (RTCP) Extended Report (XR) Block for
Burst/Gap Loss Metric Reporting", RFC 6958,
DOI 10.17487/RFC6958, May 2013,
<http://www.rfc-editor.org/info/rfc6958>.
[RFC7002] Clark, A., Zorn, G., and Q. Wu, "RTP Control Protocol
(RTCP) Extended Report (XR) Block for Discard Count Metric
Reporting", RFC 7002, DOI 10.17487/RFC7002, September
2013, <http://www.rfc-editor.org/info/rfc7002>.
[RFC7003] Clark, A., Huang, R., and Q. Wu, Ed., "RTP Control
Protocol (RTCP) Extended Report (XR) Block for Burst/Gap
Discard Metric Reporting", RFC 7003, DOI 10.17487/RFC7003,
September 2013, <http://www.rfc-editor.org/info/rfc7003>.
[RFC7097] Ott, J., Singh, V., Ed., and I. Curcio, "RTP Control
Protocol (RTCP) Extended Report (XR) for RLE of Discarded
Packets", RFC 7097, DOI 10.17487/RFC7097, January 2014,
<http://www.rfc-editor.org/info/rfc7097>.
[RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP
Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,
<http://www.rfc-editor.org/info/rfc7201>.
[RFC7657] Black, D., Ed. and P. Jones, "Differentiated Services
(Diffserv) and Real-Time Communication", RFC 7657,
DOI 10.17487/RFC7657, November 2015,
<http://www.rfc-editor.org/info/rfc7657>.
[RFC7713] Mathis, M. and B. Briscoe, "Congestion Exposure (ConEx)
Concepts, Abstract Mechanism, and Requirements", RFC 7713,
DOI 10.17487/RFC7713, December 2015,
<http://www.rfc-editor.org/info/rfc7713>.
[RFC8084] Fairhurst, G., "Network Transport Circuit Breakers",
BCP 208, RFC 8084, DOI 10.17487/RFC8084, March 2017,
<http://www.rfc-editor.org/info/rfc8084>.
[RFC8085] Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage
Guidelines", BCP 145, RFC 8085, DOI 10.17487/RFC8085,
March 2017, <http://www.rfc-editor.org/info/rfc8085>.
[RFC8108] Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session",
RFC 8108, DOI 10.17487/RFC8108, March 2017,
<http://www.rfc-editor.org/info/rfc8108>.
[Sarker] Sarker, Z., Singh, V., and C. 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,
DOI 10.1109/INFCOMW.2014.6849240, April 2014.
[Singh] Singh, V., McQuistin, S., Ellis, M., and C. Perkins,
"Circuit Breakers for Multimedia Congestion Control",
Proceedings of the 2013 20th International Packet Video
Workshop (PV), DOI 10.1109/PV.2013.6691439, December 2013.
[WebRTC-QoS]
Jones, P., Dhesikan, S., Jennings, C., and D. Druta, "DSCP
Packet Markings for WebRTC QoS", Work in Progress,
draft-ietf-tsvwg-rtcweb-qos-18, August 2016.
Acknowledgements
The authors would like to thank Bernard Aboba, Harald Alvestrand, Ben
Campbell, Alissa Cooper, Spencer Dawkins, Gorry Fairhurst, Stephen
Farrell, Nazila Fough, Kevin Gross, Cullen Jennings, Randell Jesup,
Mirja Kuehlewind, Jonathan Lennox, Matt Mathis, Stephen McQuistin,
Simon Perreault, Eric Rescorla, Abheek Saha, Meral Shirazipour, Fabio
Verdicchio, and Magnus Westerlund for their valuable feedback.
Authors' Addresses
Colin Perkins
University of Glasgow
School of Computing Science
Glasgow G12 8QQ
United Kingdom
Email: csp@csperkins.org
Varun Singh
CALLSTATS I/O Oy
Runeberginkatu 4c A 4
Helsinki 00100
Finland
Email: varun@callstats.io
URI: https://www.callstats.io/about
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