Internet DRAFT - draft-ietf-avtcore-rtp-circuit-breakers
draft-ietf-avtcore-rtp-circuit-breakers
AVTCORE Working Group C. Perkins
Internet-Draft University of Glasgow
Updates: 3550 (if approved) V. Singh
Intended status: Standards Track callstats.io
Expires: February 19, 2017 August 18, 2016
Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions
draft-ietf-avtcore-rtp-circuit-breakers-18
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, 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 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
Perkins & Singh Expires February 19, 2017 [Page 1]
Internet-Draft RTP Circuit Breakers August 2016
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 February 19, 2017.
Copyright Notice
Copyright (c) 2016 IETF Trust and the persons identified as the
document authors. All rights reserved.
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. Background . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile . 7
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 17
6. Impact of RTCP Extended Reports (XR) . . . . . . . . . . . . 19
7. Impact of Explicit Congestion Notification (ECN) . . . . . . 19
8. Impact of Bundled Media and Layered Coding . . . . . . . . . 19
9. Security Considerations . . . . . . . . . . . . . . . . . . . 20
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 20
11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 21
12. References . . . . . . . . . . . . . . . . . . . . . . . . . 21
12.1. Normative References . . . . . . . . . . . . . . . . . . 21
12.2. Informative References . . . . . . . . . . . . . . . . . 22
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 24
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
Perkins & Singh Expires February 19, 2017 [Page 2]
Internet-Draft RTP Circuit Breakers August 2016
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
[I-D.ietf-tsvwg-circuit-breaker], 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.
Perkins & Singh Expires February 19, 2017 [Page 3]
Internet-Draft RTP Circuit Breakers August 2016
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
ConEX [RFC7713], or other future protocol extensions for network
signalling 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 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 synchronisation sources (SSRCs)
used is usually two in a unicast session, one for each
participant, but can be greater if the participants send multiple
Perkins & Singh Expires February 19, 2017 [Page 4]
Internet-Draft RTP Circuit Breakers August 2016
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 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 [RFC5450]. Accurate transmission timestamps can
be helpful for estimating queuing delays, to get an early indication
of the onset of congestion.
Perkins & Singh Expires February 19, 2017 [Page 5]
Internet-Draft RTP Circuit Breakers August 2016
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, calculated by the sender using
the algorithm in Section 6.4.1 of [RFC3550] and 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, where
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
Perkins & Singh Expires February 19, 2017 [Page 6]
Internet-Draft RTP Circuit Breakers August 2016
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]), Tf is set to the time interval since the
previous frame when each frame is sent.
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, 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
Perkins & Singh Expires February 19, 2017 [Page 7]
Internet-Draft RTP Circuit Breakers August 2016
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
signalling that the remote endpoint implements RTCP (the presence of
an 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 [I-D.ietf-tsvwg-circuit-breaker].
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
([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 February 19, 2017 [Page 8]
Internet-Draft RTP Circuit Breakers August 2016
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 [I-D.ietf-tsvwg-circuit-breaker], 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, that needs to be accounted for when
determining the circuit breaker threshold
[I-D.ietf-tsvwg-circuit-breaker]. 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 February 19, 2017 [Page 9]
Internet-Draft RTP Circuit Breakers August 2016
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, 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"), as updated by Section 6.1.4 of
[I-D.ietf-avtcore-rtp-multi-stream] 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 [I-D.ietf-avtcore-rtp-multi-stream]).
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.
Perkins & Singh Expires February 19, 2017 [Page 10]
Internet-Draft RTP Circuit Breakers August 2016
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
Perkins & Singh Expires February 19, 2017 [Page 11]
Internet-Draft RTP Circuit Breakers August 2016
circuit breaker has not been cancelled, 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 time scale, 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 time scale and
throughput. The time scale on which TCP throughput is measured is
the round-trip time of the connection. In essence, this
requirement states that it is not acceptable to deploy an
application (using RTP or any other transport protocol) on the
best-effort Internet which consumes bandwidth arbitrarily and does
not compete fairly with TCP within an order of magnitude.
The phase "order of magnitude" in the above means within a factor of
ten, approximately. In order to implement this, it is necessary to
estimate the throughput a 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 using
[Padhye]:
s
X = -------------------------------------------------------------
Tr*sqrt(2*b*p/3)+(t_RTO * (3*sqrt(3*b*p/8) * p * (1+32*p*p)))
Perkins & Singh Expires February 19, 2017 [Page 12]
Internet-Draft RTP Circuit Breakers August 2016
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 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, 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 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 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
Perkins & Singh Expires February 19, 2017 [Page 13]
Internet-Draft RTP Circuit Breakers August 2016
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 multiplied by the duration of the corresponding
reporting interval, 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 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
Perkins & Singh Expires February 19, 2017 [Page 14]
Internet-Draft RTP Circuit Breakers August 2016
T_rr_interval follows Section 6.1.3 of
[I-D.ietf-avtcore-rtp-multi-stream]).
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 artefacts. 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 [RFC5405], 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 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,
Perkins & Singh Expires February 19, 2017 [Page 15]
Internet-Draft RTP Circuit Breakers August 2016
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 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.
Perkins & Singh Expires February 19, 2017 [Page 16]
Internet-Draft RTP Circuit Breakers August 2016
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,
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 recognised that the RTP implementation in some systems might
not be able to determine if a flow set-up 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
[I-D.ietf-tsvwg-circuit-breaker].
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 Reception 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
Perkins & Singh Expires February 19, 2017 [Page 17]
Internet-Draft RTP Circuit Breakers August 2016
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,
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
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
behaviour of the RTP circuit breaker, as described in Section 4.
Perkins & Singh Expires February 19, 2017 [Page 18]
Internet-Draft RTP Circuit Breakers August 2016
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 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 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 version 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 the sending
rate of the group to reduce by a factor of ten, depending on the RTP
Perkins & Singh Expires February 19, 2017 [Page 19]
Internet-Draft RTP Circuit Breakers August 2016
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, 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 DSCP values [RFC7657],
[I-D.ietf-tsvwg-rtcweb-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, causing 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
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 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. IANA Considerations
There are no actions for IANA.
Perkins & Singh Expires February 19, 2017 [Page 20]
Internet-Draft RTP Circuit Breakers August 2016
11. 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.
12. References
12.1. 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>.
Perkins & Singh Expires February 19, 2017 [Page 21]
Internet-Draft RTP Circuit Breakers August 2016
12.2. Informative References
[Floyd] Floyd, S., Handley, M., Padhye, J., and J. Widmer,
"Equation-Based Congestion Control for Unicast
Applications", Proceedings of the ACM SIGCOMM
conference, 2000, DOI 10.1145/347059.347397, August 2000.
[I-D.ietf-avtcore-rtp-multi-stream]
Lennox, J., Westerlund, M., Wu, Q., and D. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session",
draft-ietf-avtcore-rtp-multi-stream-11 (work in progress),
December 2015.
[I-D.ietf-tsvwg-circuit-breaker]
Fairhurst, G., "Network Transport Circuit Breakers",
draft-ietf-tsvwg-circuit-breaker-15 (work in progress),
April 2016.
[I-D.ietf-tsvwg-rtcweb-qos]
Jones, P., Dhesikan, S., Jennings, C., and D. Druta, "DSCP
Packet Markings for WebRTC QoS", draft-ietf-tsvwg-rtcweb-
qos-17 (work in progress), May 2016.
[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.
[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>.
Perkins & Singh Expires February 19, 2017 [Page 22]
Internet-Draft RTP Circuit Breakers August 2016
[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>.
[RFC5405] Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
for Application Designers", BCP 145, RFC 5405,
DOI 10.17487/RFC5405, November 2008,
<http://www.rfc-editor.org/info/rfc5405>.
[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>.
Perkins & Singh Expires February 19, 2017 [Page 23]
Internet-Draft RTP Circuit Breakers August 2016
[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>.
[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, 2014,
April 2014.
[Singh] Singh, V., McQuistin, S., Ellis, M., and C. 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 February 19, 2017 [Page 24]
Internet-Draft RTP Circuit Breakers August 2016
Varun Singh
Nemu Dialogue Systems Oy
Runeberginkatu 4c A 4
Helsinki 00100
Finland
Email: varun.singh@iki.fi
URI: http://www.callstats.io/
Perkins & Singh Expires February 19, 2017 [Page 25]
