draft-ietf-avtcore-rtp-circuit-breakers-00.txt   draft-ietf-avtcore-rtp-circuit-breakers-01.txt 
Network Working Group C. Perkins Network Working Group C. Perkins
Internet-Draft University of Glasgow Internet-Draft University of Glasgow
Intended status: Standards Track V. Singh Intended status: Standards Track V. Singh
Expires: April 15, 2013 Aalto University Expires: April 25, 2013 Aalto University
October 12, 2012 October 22, 2012
RTP Congestion Control: Circuit Breakers for Unicast Sessions RTP Congestion Control: Circuit Breakers for Unicast Sessions
draft-ietf-avtcore-rtp-circuit-breakers-00 draft-ietf-avtcore-rtp-circuit-breakers-01
Abstract Abstract
The Real-time Transport Protocol (RTP) is widely used in telephony, The Real-time Transport Protocol (RTP) is widely used in telephony,
video conferencing, and telepresence applications. Such applications video conferencing, and telepresence applications. Such applications
are often run on best-effort UDP/IP networks. If congestion control are often run on best-effort UDP/IP networks. If congestion control
is not implemented in the applications, then network congestion will is not implemented in the applications, then network congestion will
deteriorate the user's multimedia experience. This document does not deteriorate the user's multimedia experience. This document does not
propose a congestion control algorithm; rather, it defines a minimal propose a congestion control algorithm; rather, it defines a minimal
set of "circuit-breakers". Circuit-breakers are conditions under set of "circuit-breakers". Circuit-breakers are conditions under
which an RTP flow is expected to stop transmiting media to protect which an RTP flow is expected to stop transmitting media to protect
the network from excessive congestion. It is expected that all RTP the network from excessive congestion. It is expected that all RTP
applications running on best-effort networks will be able to run applications running on best-effort networks will be able to run
without triggering these circuit breakers in normal operation. Any without triggering these circuit breakers in normal operation. Any
future RTP congestion control specification is expected to operate future RTP congestion control specification is expected to operate
within the envelope defined by these circuit breakers. within the envelope defined by these circuit breakers.
Status of this Memo Status of this Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
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Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on April 15, 2013. This Internet-Draft will expire on April 25, 2013.
Copyright Notice Copyright Notice
Copyright (c) 2012 IETF Trust and the persons identified as the Copyright (c) 2012 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
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include Simplified BSD License text as described in Section 4.e of include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Background . . . . . . . . . . . . . . . . . . . . . . . . . . 3 3. Background . . . . . . . . . . . . . . . . . . . . . . . . . . 3
4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile . . 6 4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile . . 6
4.1. RTP/AVP Circuit Breaker #1: Timeout . . . . . . . . . . . 7 4.1. RTP/AVP Circuit Breaker #1: Media Timeout . . . . . . . . 7
4.2. RTP/AVP Circuit Breaker #2: Congestion . . . . . . . . . . 8 4.2. RTP/AVP Circuit Breaker #2: RTCP Timeout . . . . . . . . . 8
5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile . 10 4.3. RTP/AVP Circuit Breaker #3: Congestion . . . . . . . . . . 9
6. Impact of RTCP XR . . . . . . . . . . . . . . . . . . . . . . 11 5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile . 11
7. Impact of Explicit Congestion Notification (ECN) . . . . . . . 11 6. Impact of RTCP XR . . . . . . . . . . . . . . . . . . . . . . 12
8. Session Timeout . . . . . . . . . . . . . . . . . . . . . . . 11 7. Impact of Explicit Congestion Notification (ECN) . . . . . . . 12
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 12 8. Security Considerations . . . . . . . . . . . . . . . . . . . 13
10. Security Considerations . . . . . . . . . . . . . . . . . . . 12 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 13
11. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 12 10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 13
12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 12 11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 13
13. References . . . . . . . . . . . . . . . . . . . . . . . . . . 13 11.1. Normative References . . . . . . . . . . . . . . . . . . . 13
13.1. Normative References . . . . . . . . . . . . . . . . . . . 13 11.2. Informative References . . . . . . . . . . . . . . . . . . 14
13.2. Informative References . . . . . . . . . . . . . . . . . . 13 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 15
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 14
1. Introduction 1. Introduction
The Real-time Transport Protocol (RTP) [RFC3550] is widely used in The Real-time Transport Protocol (RTP) [RFC3550] is widely used in
voice-over-IP, video teleconferencing, and telepresence systems. voice-over-IP, video teleconferencing, and telepresence systems.
Many of these systems run over best-effort UDP/IP networks, and can Many of these systems run over best-effort UDP/IP networks, and can
suffer from packet loss and increased latency if network congestion suffer from packet loss and increased latency if network congestion
occurs. Designing effective RTP congestion control algorithms, to occurs. Designing effective RTP congestion control algorithms, to
adapt the transmission of RTP-based media to match the available adapt the transmission of RTP-based media to match the available
network capacity, while also maintaining the user experience, is a network capacity, while also maintaining the user experience, is a
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interpreted as carrying special significance in this memo. interpreted as carrying special significance in this memo.
3. Background 3. Background
We consider congestion control for unicast RTP traffic flows. This We consider congestion control for unicast RTP traffic flows. This
is the problem of adapting the transmission of an audio/visual data is the problem of adapting the transmission of an audio/visual data
flow, encapsulated within an RTP transport session, from one sender flow, encapsulated within an RTP transport session, from one sender
to one receiver, so that it matches the available network bandwidth. to one receiver, so that it matches the available network bandwidth.
Such adaptation needs to be done in a way that limits the disruption 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 to the user experience caused by both packet loss and excessive rate
changes. changes. Congestion control for multicast flows is outside the scope
of this memo.
Congestion control for unicast RTP traffic can be implemented in one 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 of two places in the protocol stack. One approach is to run the RTP
traffic over a congestion controlled transport protocol, for example traffic over a congestion controlled transport protocol, for example
over TCP, and to adapt the media encoding to match the dictates of over TCP, and to adapt the media encoding to match the dictates of
the transport-layer congestion control algorithm. This is safe for the transport-layer congestion control algorithm. This is safe for
the network, but can be suboptimal for the media quality unless the the network, but can be suboptimal for the media quality unless the
transport protocol is designed to support real-time media flows. We transport protocol is designed to support real-time media flows. We
do not consider this class of applications further in this memo, as do not consider this class of applications further in this memo, as
their network safety is guaranteed by the underlying transport. their network safety is guaranteed by the underlying transport.
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network congestion to the sender. However, if a receiver detects network congestion to the sender. However, if a receiver detects
the onset of congestion partway through a reporting interval, the the onset of congestion partway through a reporting interval, the
base RTP specification contains no provision for sending the RTCP base RTP specification contains no provision for sending the RTCP
RR packet early, and the receiver has to wait until the next RR packet early, and the receiver has to wait until the next
scheduled reporting interval. scheduled reporting interval.
o The RTCP Extended Reports (XR) [RFC3611] allow reporting of more o The RTCP Extended Reports (XR) [RFC3611] allow reporting of more
complex and sophisticated reception quality metrics, but do not complex and sophisticated reception quality metrics, but do not
change the RTCP timing rules. RTCP extended reports of potential change the RTCP timing rules. RTCP extended reports of potential
interest for congestion control purposes are the extended packet interest for congestion control purposes are the extended packet
loss, discard, and burst metrics [RFC3611], and loss, discard, and burst metrics [RFC3611],
[I-D.ietf-xrblock-rtcp-xr-discard], [I-D.ietf-xrblock-rtcp-xr-discard],
[I-D.ietf-xrblock-rtcp-xr-discard-rle-metrics], [I-D.ietf-xrblock-rtcp-xr-discard-rle-metrics],
[I-D.ietf-xrblock-rtcp-xr-burst-gap-discard], [I-D.ietf-xrblock-rtcp-xr-burst-gap-discard],
[I-D.ietf-xrblock-rtcp-xr-burst-gap-loss]; and the extended delay [I-D.ietf-xrblock-rtcp-xr-burst-gap-loss]; and the extended delay
metrics [I-D.ietf-xrblock-rtcp-xr-delay], metrics [I-D.ietf-xrblock-rtcp-xr-delay],
[I-D.ietf-xrblock-rtcp-xr-pdv]. Other RTCP Extended Reports that [I-D.ietf-xrblock-rtcp-xr-pdv]. Other RTCP Extended Reports that
could be helpful for congestion control purposes might be could be helpful for congestion control purposes might be
developed in future. developed in future.
o Rapid feedback about the occurrence of congestion events can be o Rapid feedback about the occurrence of congestion events can be
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defines new transport-layer feedback messages, including negative defines new transport-layer feedback messages, including negative
acknowledgements (NACKs), that can be used to report on specific acknowledgements (NACKs), that can be used to report on specific
congestion events. The use of the RTP/AVPF profile is dependent congestion events. The use of the RTP/AVPF profile is dependent
on signalling, but is otherwise generally backwards compatible, as on signalling, but is otherwise generally backwards compatible, as
it keeps the same average RTCP reporting interval as the base RTP it keeps the same average RTCP reporting interval as the base RTP
specification. The RTP Codec Control Messages [RFC5104] extend specification. The RTP Codec Control Messages [RFC5104] extend
the RTP/AVPF profile with additional feedback messages that can be the RTP/AVPF profile with additional feedback messages that can be
used to influence that way in which rate adaptation occurs. The used to influence that way in which rate adaptation occurs. The
dynamics of how rapidly feedback can be sent are unchanged. dynamics of how rapidly feedback can be sent are unchanged.
o Finally, the RTP and RTCP extensions for Explicit Congestion o Finally, Explicit Congestion Notification (ECN) for RTP over UDP
Notification (ECN) [I-D.ietf-avtcore-ecn-for-rtp] can be used to [RFC6679] can be used to provide feedback on the number of packets
provide feedback on the number of packets that received an ECN that received an ECN Congestion Experienced (CE) mark. This RTCP
Congestion Experienced (CE) mark. This extension builds on the extension builds on the RTP/AVPF profile to allow rapid congestion
RTP/AVPF profile to allow rapid congestion feedback when ECN is feedback when ECN is supported.
supported.
In addition to these mechanisms for providing feedback, the sender In addition to these mechanisms for providing feedback, the sender
can include an RTP header extension in each packet to record packet can include an RTP header extension in each packet to record packet
transmission times. There are two methods: [RFC5450] represents the transmission times. There are two methods: [RFC5450] represents the
transmission time in terms of a time-offset from the RTP timestamp of transmission time in terms of a time-offset from the RTP timestamp of
the packet, while [RFC6051] includes an explicit NTP-format sending the packet, while [RFC6051] includes an explicit NTP-format sending
timestamp (potentially more accurate, but a higher header overhead). timestamp (potentially more accurate, but a higher header overhead).
Accurate sending timestamps can be helpful for estimating queuing Accurate sending timestamps can be helpful for estimating queuing
delays, to get an early indication of the onset of congestion. delays, to get an early indication of the onset of congestion.
Taken together, these various mechanisms allow receivers to provide Taken together, these various mechanisms allow receivers to provide
feedback on the senders when congestion events occur, with varying feedback on the senders when congestion events occur, with varying
degrees of timeliness and accuracy. The key distinction is between degrees of timeliness and accuracy. The key distinction is between
systems that use only the basic RTCP mechanisms, without RTP/AVPF systems that use only the basic RTCP mechanisms, without RTP/AVPF
rapid feedback, and those that use the RTP/AVPF extensions and so can rapid feedback, and those that use the RTP/AVPF extensions to respond
respond to congestion more rapidly. to congestion more rapidly.
4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile 4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile
The feedback mechanisms defined in [RFC3550] and available under the The feedback mechanisms defined in [RFC3550] and available under the
RTP/AVP profile [RFC3551] are the minimum that can be assumed for a RTP/AVP profile [RFC3551] are the minimum that can be assumed for a
baseline circuit breaker mechanism that is suitable for all unicast baseline circuit breaker mechanism that is suitable for all unicast
applications of RTP. Accordingly, for an RTP circuit breaker to be 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 useful, it needs to be able to detect that an RTP flow is causing
excessive congestion using only basic RTCP features, without needing excessive congestion using only basic RTCP features, without needing
RTCP XR feedback or the RTP/AVPF profile for rapid RTCP reports. RTCP XR feedback or the RTP/AVPF profile for rapid RTCP reports.
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These congestion signals limit the possible circuit breakers, since These congestion signals limit the possible circuit breakers, since
they give only limited visibility into the behaviour of the network. they give only limited visibility into the behaviour of the network.
RTT estimates are widely used in congestion control algorithms, as a RTT estimates are widely used in congestion control algorithms, as a
proxy for queuing delay measures in delay-based congestion control or proxy for queuing delay measures in delay-based congestion control or
to determine connection timeouts. RTT estimates derived from RTCP SR to determine connection timeouts. RTT estimates derived from RTCP SR
and RR packets sent according to the RTP/AVP timing rules are far too and RR packets sent according to the RTP/AVP timing rules are far too
infrequent to be useful though, and don't give enough information to infrequent to be useful though, and don't give enough information to
distinguish a delay change due to routing updates from queuing delay distinguish a delay change due to routing updates from queuing delay
caused by congestion. Accordingly, we do not use the RTT estimate caused by congestion. Accordingly, we cannot use the RTT estimate
alone as an RTP circuit breaker. alone as an RTP circuit breaker.
Increased jitter can be a signal of transient network congestion, but Increased jitter can be a signal of transient network congestion, but
in the highly aggregated form reported in RTCP RR packets, it offers in the highly aggregated form reported in RTCP RR packets, it offers
insufficient information to estimate the extent or persistence of insufficient information to estimate the extent or persistence of
congestion. Jitter reports are a useful early warning of potential congestion. Jitter reports are a useful early warning of potential
network congestion, but provide an insufficiently strong signal to be network congestion, but provide an insufficiently strong signal to be
used as a circuit breaker. used as a circuit breaker.
The remaining congestion signals are the packet loss fraction and the The remaining congestion signals are the packet loss fraction and the
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Two packet loss regimes can be observed: 1) RTCP RR packets show a Two packet loss regimes can be observed: 1) RTCP RR packets show a
non-zero packet loss fraction, while the extended highest sequence non-zero packet loss fraction, while the extended highest sequence
number received continues to increment; and 2) RR packets show a loss number received continues to increment; and 2) RR packets show a loss
fraction of zero, but the extended highest sequence number received fraction of zero, but the extended highest sequence number received
does not increment even though the sender has been transmitting RTP does not increment even though the sender has been transmitting RTP
data packets. The former corresponds to the TCP congestion avoidance data packets. The former corresponds to the TCP congestion avoidance
state, and indicates a congested path that is still delivering data; 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 the latter corresponds to a TCP timeout, and is most likely due to a
path failure. We derive circuit breaker conditions for these two path failure. We derive circuit breaker conditions for these two
loss regimes. loss regimes in the following.
4.1. RTP/AVP Circuit Breaker #1: Timeout 4.1. RTP/AVP Circuit Breaker #1: Media Timeout
If RTP data packets are being sent while the corresponding RTCP RR If RTP data packets are being sent while the corresponding RTCP RR
packets report a non-increasing extended highest sequence number packets report a non-increasing extended highest sequence number
received, this is an indication that those RTP data packets are not received, this is an indication that those RTP data packets are not
reaching the receiver. This could be a short-term issue affecting reaching the receiver. This could be a short-term issue affecting
only a few packets, perhaps caused by a slow-to-open firewall or a only a few packets, perhaps caused by a slow-to-open firewall or a
transient connectivity problem, but if the issue persists, it is a transient connectivity problem, but if the issue persists, it is a
sign of a more ongoing and significant problem. Accordingly, if a sign of a more ongoing and significant problem. Accordingly, if a
sender of RTP data packets receives two or more consecutive RTCP RR sender of RTP data packets receives two or more consecutive RTCP RR
packets from the same receiver that correspond to its transmission, packets from the same receiver that correspond to its transmission,
and have a non-increasing extended highest sequence number received and have a non-increasing extended highest sequence number received
field (i.e., at least three RTCP RR packets that report the same field (i.e., at least three RTCP RR packets that report the same
value in the extended highest sequence number received field, when value in the extended highest sequence number received field, when
the sender has sent data packets that would have caused an increase the sender has sent data packets that would have caused an increase
in the reported value of the extended highest sequence number in the reported value of the extended highest sequence number
received if they had reached the receiver), then that sender SHOULD received if they had reached the receiver), then that sender SHOULD
cease transmission. cease transmission. What it means to cease transmission depends on
the application, but the intention is that the application will stop
sending RTP data packets until the user makes an explicit attempt to
restart the call (RTP flows halted by the circuit breaker SHOULD NOT
be restarted automatically unless the sender has received information
that the congestion has dissipated).
Systems that usually send at a high data rate, but which can reduce Systems that usually send at a high data rate, but that can reduce
their data rate significantly (i.e., by at least a factor of ten), their data rate significantly (i.e., by at least a factor of ten),
MAY first reduce their sending rate to this lower value to see if MAY first reduce their sending rate to this lower value to see if
this resolves the congestion, but MUST then cease transmission if the this resolves the congestion, but MUST then cease transmission if the
problem does not resolve itself within a further two RTCP reporting problem does not resolve itself within a further two RTCP reporting
intervals. An example of this might be a video conferencing system intervals. An example of this might be a video conferencing system
that backs off to sending audio only, before completely dropping the that backs off to sending audio only, before completely dropping the
call. If such a reduction in sending rate resolves the congestion call. If such a reduction in sending rate resolves the congestion
problem, the sender MAY gradually increase the rate at which it sends problem, the sender MAY gradually increase the rate at which it sends
data after a reasonable amount of time has passed, provided it takes data after a reasonable amount of time has passed, provided it takes
care not to cause the problem to recur ("reasonable" is intentionally care not to cause the problem to recur ("reasonable" is intentionally
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The choice of two RTCP reporting intervals is to give enough time for The choice of two RTCP reporting intervals is to give enough time for
transient problems to resolve themselves, but to stop problem flows transient problems to resolve themselves, but to stop problem flows
quickly enough to avoid causing serious ongoing network congestion. quickly enough to avoid causing serious ongoing network congestion.
A single RTCP report showing no reception could be caused by numerous A single RTCP report showing no reception could be caused by numerous
transient faults, and so will not cease transmission. Waiting for transient faults, and so will not cease transmission. Waiting for
more than two RTCP reports before stopping a flow might avoid some more than two RTCP reports before stopping a flow might avoid some
false positives, but would lead to problematic flows running for a false positives, but would lead to problematic flows running for a
long time before being cut off. long time before being cut off.
4.2. RTP/AVP Circuit Breaker #2: Congestion 4.2. RTP/AVP Circuit Breaker #2: RTCP Timeout
In addition to media timeouts, as were discussed in Section 4.1, an
RTP session has the possibility of an RTCP timeout. This can occur
when RTP data packets are being sent, but there are no RTCP reports
returned from the receiver. This is either due to a failure of the
receiver to send RTCP reports, or a failure of the return path that
is preventing those RTCP reporting from being delivered.
According to RFC 3550 [RFC3550], any participant that has not sent an
RTCP packet within the last two RTCP intervals is removed from the
sender list. Therefore, an RTP sender SHOULD cease transmission if
it does not receive a single RTCP RR packet and during this period
has sent 3 RTCP SR packets to the RTP receiver. Similarly, the same
circuit breaker rule applies to an RTCP receiver which has not
received a single SR packet, and in the corresponding period it has
sent 3 RTCP RR packets. What it means to cease transmission depends
on the application, but the intention is that the application will
stop sending RTP data packets until the user makes an explicit
attempt to restart the call (RTP flows halted by the circuit breaker
SHOULD NOT be restarted automatically unless the sender has received
information that the congestion has dissipated).
4.3. RTP/AVP Circuit Breaker #3: Congestion
If RTP data packets are being sent, and the corresponding RTCP RR If RTP data packets are being sent, and the corresponding RTCP RR
packets show non-zero packet loss fraction and increasing extended packets show non-zero packet loss fraction and increasing extended
highest sequence number received, then the RTP data packets are highest sequence number received, then those RTP data packets are
arriving at the receiver, but some degree of congestion is occurring. arriving at the receiver, but some degree of congestion is occurring.
The RTP/AVP profile [RFC3551] states that: The RTP/AVP profile [RFC3551] states that:
If best-effort service is being used, RTP receivers SHOULD monitor If best-effort service is being used, RTP receivers SHOULD monitor
packet loss to ensure that the packet loss rate is within packet loss to ensure that the packet loss rate is within
acceptable parameters. Packet loss is considered acceptable if a acceptable parameters. Packet loss is considered acceptable if a
TCP flow across the same network path and experiencing the same TCP flow across the same network path and experiencing the same
network conditions would achieve an average throughput, measured network conditions would achieve an average throughput, measured
on a reasonable timescale, that is not less than the RTP flow is on a reasonable time scale, that is not less than the RTP flow is
achieving. This condition can be satisfied by implementing achieving. This condition can be satisfied by implementing
congestion control mechanisms to adapt the transmission rate (or congestion control mechanisms to adapt the transmission rate (or
the number of layers subscribed for a layered multicast session), the number of layers subscribed for a layered multicast session),
or by arranging for a receiver to leave the session if the loss or by arranging for a receiver to leave the session if the loss
rate is unacceptably high. rate is unacceptably high.
The comparison to TCP cannot be specified exactly, but is intended The comparison to TCP cannot be specified exactly, but is intended
as an "order-of-magnitude" comparison in timescale and throughput. as an "order-of-magnitude" comparison in time scale and
The timescale on which TCP throughput is measured is the round- throughput. The time scale on which TCP throughput is measured is
trip time of the connection. In essence, this requirement states the round-trip time of the connection. In essence, this
that it is not acceptable to deploy an application (using RTP or requirement states that it is not acceptable to deploy an
any other transport protocol) on the best-effort Internet which application (using RTP or any other transport protocol) on the
consumes bandwidth arbitrarily and does not compete fairly with best-effort Internet which consumes bandwidth arbitrarily and does
TCP within an order of magnitude. not compete fairly with TCP within an order of magnitude.
(The phase "order of magnitude" in the above means a factor of ten).
The throughput of a long-lived TCP connection can be estimated using The phase "order of magnitude" in the above means within a factor of
the TCP throughput equation: ten, approximately. In order to implement this, it is necessary to
estimate the throughput a TCP connection would achieve over the path.
For a long-lived TCP Reno connection, Padhye et al. [Padhye] showed
that the throughput can be estimated using the following equation:
s s
X = -------------------------------------------------------------- X = --------------------------------------------------------------
R*sqrt(2*b*p/3) + (t_RTO * (3*sqrt(3*b*p/8) * p * (1+32*p^2))) R*sqrt(2*b*p/3) + (t_RTO * (3*sqrt(3*b*p/8) * p * (1+32*p^2)))
Where: where:
X is the transmit rate in bytes/second. X is the transmit rate in bytes/second.
s is the packet size in bytes. If the RTP data packets vary in s is the packet size in bytes. If data packets vary in size, then
size, then the average size is to be used. the average size is to be used.
R is the round trip time in seconds. R is the round trip time in seconds.
p is the loss event rate, between 0 and 1.0, of the number of loss p is the loss event rate, between 0 and 1.0, of the number of loss
events as a fraction of the number of packets transmitted. events as a fraction of the number of packets transmitted.
t_RTO is the TCP retransmission timeout value in seconds, t_RTO is the TCP retransmission timeout value in seconds,
approximated by setting t_RTO = 4*R. approximated by setting t_RTO = 4*R.
b is the number of packets acknowledged by a single TCP b is the number of packets acknowledged by a single TCP
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approximated as follows with reasonable accuracy: approximated as follows with reasonable accuracy:
s s
X = --------------- X = ---------------
R * sqrt(p*2/3) R * sqrt(p*2/3)
It is RECOMMENDED that this simplified throughout equation be used, It is RECOMMENDED that this simplified throughout equation be used,
since the reduction in accuracy is small, and it is much simpler to since the reduction in accuracy is small, and it is much simpler to
calculate than the full equation. calculate than the full equation.
Given this TCP equation, two parameters need to be estimated in order Given this TCP equation, two parameters need to be estimated and
to calculate the throughput: the round trip time, R, and the loss reported to the sender in order to calculate the throughput: the
event rate, p (the packet size, s, is known to the sender). The round trip time, R, and the loss event rate, p (the packet size, s,
round trip time can be estimated from RTCP SR and RR packets. This is known to the sender). The round trip time can be estimated from
is done too infrequently for accurate statistics, but is the best RTCP SR and RR packets. This is done too infrequently for accurate
that can be done with the standard RTCP mechanisms. statistics, but is the best that can be done with the standard RTCP
mechanisms.
RTCP RR packets contain the packet loss fraction, rather than the RTCP RR packets contain the packet loss fraction, rather than the
loss event rate, so p cannot be reported (TCP typically treats 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 loss of multiple packets within a single RTT as one loss event, but
RTCP RR packets report the overall fraction of packets lost, not RTCP RR packets report the overall fraction of packets lost, not
caring about when the losses occurred). Using the loss fraction in caring about when the losses occurred). Using the loss fraction in
place of the loss event rate can overestimate the loss. We believe place of the loss event rate can overestimate the loss. We believe
that this overestimate will not be significant, given that we are that this overestimate will not be significant, given that we are
only interested in order of magnitude comparison (Floyd et al, only interested in order of magnitude comparison ([Floyd] section
"Equation-Based Congestion Control for Unicast Applications", Proc. 3.2.1 shows that the difference is small for steady-state conditions
SIGCOMM 2000, section 3.2.1, show that the difference is small for and random loss, but using the loss fraction is more conservative in
steady-state conditions and random loss, but using the loss fraction the case of bursty loss).
is more conservative in the case of bursty loss).
The congestion circuit breaker is therefore: when RTCP RR packets are The congestion circuit breaker is therefore: when RTCP RR packets are
received, estimate the TCP throughput using the simplified equation received, estimate the TCP throughput using the simplified equation
above, and the measured R, p (approximated by the loss fraction), and above, and the measured R, p (approximated by the loss fraction), and
s. Compare this with the actual sending rate. If the actual sending s. Compare this with the actual sending rate. If the actual sending
rate has been more than a factor of ten greater than the throughput rate is more than ten times the estimated sending rate derived from
equation estimate for two or more RTCP reporting intervals, stop the TCP throughput equation for two consecutive RTCP reporting
transmitting. intervals, the sender SHOULD cease transmission. What it means to
cease transmission depends on the application, but the intention is
Again, we use two reporting intervals to avoid triggering the circuit that the application will stop sending RTP data packets until the
breaker on transient failures. This circuit breaker is a worst-case user makes an explicit attempt to restart the call (RTP flows halted
condition, and congestion control needs to be performed to keep well by the circuit breaker SHOULD NOT be restarted automatically unless
within this bound. It is expected that the circuit breaker will only the sender has received information that the congestion has
be triggered if the usual congestion control fails for some reason. dissipated).
5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile
More rapid feedback allows more responsiveness. The receiver SHOULD
provide feedback more often during, or at onset of, congestion, and
provide feedback less often when there is no congestion.
(tbd -- mechanisms probably need to be designed in conjunction with Systems that usually send at a high data rate, but that can reduce
the different classes of congestion control that can leverage RTP/ their data rate significantly (i.e., by at least a factor of ten),
AVPF; e.g., we might need to specify limits for TFRC-like or delay- MAY first reduce their sending rate to this lower value to see if
based algorithms using RTP/AVPF feedback.) this resolves the congestion, but MUST then cease transmission if the
problem does not resolve itself within a further two RTCP reporting
intervals. An example of this might be a video conferencing system
that backs off to sending audio only, before completely dropping the
call. If such a reduction in sending rate resolves the congestion
problem, the sender MAY gradually increase the rate at which it sends
data after a reasonable amount of time has passed, provided it takes
care not to cause the problem to recur ("reasonable" is intentionally
not defined here).
(tbd -- a high-level question to be answered is whether we need to As in Section 4.1, we use two reporting intervals to avoid triggering
specify anything different for the circuit breaker for AVPF, or if we the circuit breaker on transient failures. This circuit breaker is a
leave that unchanged, and focus solely on the dynamics, to ensure the worst-case condition, and congestion control needs to be performed to
circuit breaker is never triggered.) keep well within this bound. It is expected that the circuit breaker
will only be triggered if the usual congestion control fails for some
reason.
6. Impact of RTCP XR 5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile
(tbd) 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.
This improves the information, but doesn't change the dynamics of the Receiving rapid feedback about congestion events potentially allows
congestion control loop. Suspect the impact will actually be quite congestion control algorithms to be more responsive, and to better
small. adapt the media transmission to the limitations of the network. It
is expected that many RTP congestion control algorithms will adopt
the RTP/AVPF profile for this reason, defining new transport layer
feedback reports that suit their requirements. Since these reports
are not yet defined, and likely very specific to the details of the
congestion control algorithm chosen, they cannot be used as part of
the generic RTP circuit breaker.
Packets discarded [I-D.ietf-xrblock-rtcp-xr-discard] or bytes If the extension for Reduced-Size RTCP [RFC5506] is not used, early
discarded [I-D.ietf-xrblock-rtcp-xr-discard-rle-metrics] due to late RTCP feedback packets sent according to the RTP/AVPF profile will be
arrival by the receiver might indicate congestion. Congestion compound RTCP packets that include an RTCP SR/RR packet. That RTCP
control needs to consider the discarded packets as if they were lost SR/RR packet MUST be processed as if it were sent as a regular RTCP
packets. report and counted towards the circuit breaker conditions specified
in Section 4.1 and Section 4.3 of this memo. This will potentially
make the RTP circuit breaker fire earlier than it would if the RTP/
AVPF profile was not used.
The RTCP RR reports the loss fraction over an RTCP interval which is Reduced-size RTCP reports sent under to the RTP/AVPF early feedback
insufficient to distinguish between solitary or bursty losses. To rules that do not contain an RTCP SR or RR packet MUST be ignored by
provide rough sense of duration of losses or discards, an endpoint the RTP circuit breaker (they do not contain the information used by
can use burst/gap reporting for loss the circuit breaker algorithm). In this case, the circuit breaker
[I-D.ietf-xrblock-rtcp-xr-burst-gap-loss] and discard will only use the information contained in the periodic RTCP SR/RR
[I-D.ietf-xrblock-rtcp-xr-burst-gap-discard]. For more accurate packets. This allows the use of low-overhead early RTP/AVPF feedback
reporting the receiver can use Run-length encoded (RLE) lost without triggering the RTP circuit breaker, and so is suitable for
[RFC3611] or discarded [I-D.ietf-xrblock-rtcp-xr-discard-rle-metrics] RTP congestion control algorithms that need to quickly report loss
packets. events in between regular RTCP reports.
For precise measurement of network roundtrip delay the receiver can 6. Impact of RTCP XR
signal its end-system delay [I-D.ietf-xrblock-rtcp-xr-delay]
[RFC3611].
A receiver can also indicate onset or end of congestion by reporting RTCP Extended Report (XR) blocks provide additional reception quality
the distribution of the inter-packet delay variation metrics, but do not change the RTCP timing rules. Some of the RTCP
[I-D.ietf-xrblock-rtcp-xr-pdv] [RFC3611]. XR blocks provide information that might be useful for congestion
control purposes, others provided non-congestion-related metrics.
The presence of RTCP XR blocks in a compound RTCP packet does not
affect the RTP circuit breaker algorithm; for consistency and ease of
implementation, only the reception report blocks contained in RTCP SR
or RR packets are used by the RTP circuit breaker algorithm.
7. Impact of Explicit Congestion Notification (ECN) 7. Impact of Explicit Congestion Notification (ECN)
ECN-CE marked packets SHOULD be treated as if it were lost for the ECN-CE marked packets SHOULD be treated as if it were lost for the
purposes of congestion control, when determining the optimal rate at purposes of congestion control, when determining the optimal media
which to send. However, it seems unwise to treat the receipt of sending rate for an RTP flow. If an RTP sender has negotiated ECN
multiple ECN-CE marked packets as a circuit breaker, since it is support for an RTP session, and has successfully initiated ECN use on
likely that ECN-capable and non-ECN-capable paths will exist for a the path to the receiver [RFC6679], then ECN-CE marked packets SHOULD
long time to come. Rather, consider packet loss as the circuit be treated as if they were lost when calculating if the congestion-
breaker condition as for non-ECN flows. based RTP circuit breaker (Section 4.3) has been met.
8. Session Timeout The use of ECN for RTP flows does not affect the media timeout RTP
circuit breaker (Section 4.1) or the RTCP timeout circuit breaker
(Section 4.2), since these are both connectivity checks that simply
determinate if any packets are being received.
From a usability perspective, if there is no audio or video response 8. Security Considerations
from the other peer, it is likely that the user will terminate the
session.
According to RFC 3550 [RFC3550], any participant that has not sent an The security considerations of [RFC3550] apply.
RTP packet within the last two RTCP interval is removed from the
sender list. To avoid timing out the specific flow, the endpoint
MUST send corresponding RTCP reports. Interactive Connectivity
Establishment (ICE) [RFC5245] recommends that the timeout MUST NOT be
less than 15 seconds.
If no RTCP RR arrives for two complete SR intervals, the sender If the RTP/AVPF profile is used to provide rapid RTCP feedback, the
SHOULD cease transmission. However, if the endpoint can reduce the security considerations of [RFC4585] apply. If ECN feedback for RTP
media rate then it MAY first reduce the rate to the lower value, but over UDP/IP is used, the security considerations of [RFC6679] apply.
terminate the transmission if still no RTCP RR is received in the
next two SR intervals. If non-authenticated RTCP reports are used, an on-path attacker can
trivially generate fake RTCP packets that indicate high packet loss
rates, causing the circuit breaker to trigger and disrupting an RTP
session. This is somewhat more difficult for an off-path attacker,
due to the need to guess the randomly chosen RTP SSRC value and the
RTP sequence number. This attack can be avoided if RTCP packets are
authenticated, for example using the Secure RTP profile [RFC3711].
9. IANA Considerations 9. IANA Considerations
There are no actions for IANA. There are no actions for IANA.
10. Security Considerations 10. Acknowledgements
(tbd: Security considerations: how to protect against fake RTCP
reports being used to force sessions to close? SRTCP is one option,
but are there any lighter weight options?)
11. Open Issues
o Clarify: when will the recipient end a call, if it receives no
data?
o When we say "cease transmission", do we need some minimum interval
before we're allowed to restart?
o What does "cease transmission" mean? Do we send an RTCP BYE and
leave the session, or is it more temporary than that?
o Add a receiver-based circuit-breaker condition. Note that this is
dependent on the signalling still working, since the receiver
needs to be able to inform the sender.
12. Acknowledgements
The authors would like to thank Harald Alvestrand, Randell Jesup, and The authors would like to thank Harald Alvestrand, Randell Jesup,
Abheek Saha for their valuable feedback. Matt Mathis, and Abheek Saha for their valuable feedback.
13. References 11. References
13.1. Normative References 11.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997. Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3448] Handley, M., Floyd, S., Padhye, J., and J. Widmer, "TCP [RFC3448] Handley, M., Floyd, S., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification", Friendly Rate Control (TFRC): Protocol Specification",
RFC 3448, January 2003. RFC 3448, January 2003.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time Jacobson, "RTP: A Transport Protocol for Real-Time
skipping to change at page 13, line 33 skipping to change at page 14, line 18
[RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control [RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control
Protocol Extended Reports (RTCP XR)", RFC 3611, Protocol Extended Reports (RTCP XR)", RFC 3611,
November 2003. November 2003.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control "Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
July 2006. July 2006.
13.2. Informative References 11.2. Informative References
[I-D.ietf-avtcore-ecn-for-rtp] [Floyd] Floyd, S., Handley, M., Padhye, J., and J. Widmer,
Westerlund, M., Johansson, I., Perkins, C., and K. "Equation-Based Congestion Control for Unicast
Carlberg, "Explicit Congestion Notification (ECN) for RTP Applications", Proc. ACM SIGCOMM 2000, DOI 10.1145/
over UDP", draft-ietf-avtcore-ecn-for-rtp-06 (work in 347059.347397, August 2000.
progress), February 2012.
[I-D.ietf-xrblock-rtcp-xr-burst-gap-discard] [I-D.ietf-xrblock-rtcp-xr-burst-gap-discard]
Clark, A., Hunt, G., Wu, W., and R. Huang, "RTCP XR Report Clark, A., Huang, R., and W. Wu, "RTP Control
Block for Burst/Gap Discard metric Reporting", Protocol(RTCP) Extended Report (XR) Block for Discard
draft-ietf-xrblock-rtcp-xr-burst-gap-discard-02 (work in Count metric Reporting",
progress), January 2012. draft-ietf-xrblock-rtcp-xr-burst-gap-discard-06 (work in
progress), October 2012.
[I-D.ietf-xrblock-rtcp-xr-burst-gap-loss] [I-D.ietf-xrblock-rtcp-xr-burst-gap-loss]
Clark, A., Hunt, G., Zhao, J., Wu, W., and S. Zhang, "RTCP Clark, A., Zhang, S., Zhao, J., and W. Wu, "RTP Control
XR Report Block for Burst/Gap Loss metric Reporting", Protocol (RTCP) Extended Report (XR) Block for Burst/Gap
draft-ietf-xrblock-rtcp-xr-burst-gap-loss-01 (work in Loss metric Reporting",
progress), January 2012. draft-ietf-xrblock-rtcp-xr-burst-gap-loss-04 (work in
progress), October 2012.
[I-D.ietf-xrblock-rtcp-xr-delay] [I-D.ietf-xrblock-rtcp-xr-delay]
Hunt, G., Gross, K., and A. Clark, "RTCP XR Report Block Clark, A., Gross, K., and W. Wu, "RTP Control Protocol
for Delay metric Reporting", (RTCP) Extended Report (XR) Block for Delay metric
draft-ietf-xrblock-rtcp-xr-delay-01 (work in progress), Reporting", draft-ietf-xrblock-rtcp-xr-delay-10 (work in
December 2011. progress), October 2012.
[I-D.ietf-xrblock-rtcp-xr-discard] [I-D.ietf-xrblock-rtcp-xr-discard]
Hunt, G., Clark, A., Zorn, G., and W. Wu, "RTCP XR Report Clark, A., Zorn, G., and W. Wu, "RTP Control Protocol
Block for Discard metric Reporting", (RTCP) Extended Report (XR) Block for Discard Count metric
draft-ietf-xrblock-rtcp-xr-discard-01 (work in progress), Reporting", draft-ietf-xrblock-rtcp-xr-discard-09 (work in
December 2011. progress), October 2012.
[I-D.ietf-xrblock-rtcp-xr-discard-rle-metrics] [I-D.ietf-xrblock-rtcp-xr-discard-rle-metrics]
Ott, J., Singh, V., and I. Curcio, "Real-time Transport Ott, J., Singh, V., and I. Curcio, "RTP Control Protocol
Control Protocol (RTCP) Extension Report (XR) for Run (RTCP) Extended Reports (XR) for Run Length Encoding (RLE)
Length Encoding of Discarded Packets", of Discarded Packets",
draft-ietf-xrblock-rtcp-xr-discard-rle-metrics-03 (work in draft-ietf-xrblock-rtcp-xr-discard-rle-metrics-04 (work in
progress), February 2012. progress), July 2012.
[I-D.ietf-xrblock-rtcp-xr-pdv] [I-D.ietf-xrblock-rtcp-xr-pdv]
Hunt, G. and A. Clark, "RTCP XR Report Block for Packet Clark, A. and W. Wu, "RTP Control Protocol (RTCP) Extended
Delay Variation Metric Reporting", Report (XR) Block for Packet Delay Variation Metric
draft-ietf-xrblock-rtcp-xr-pdv-02 (work in progress), Reporting", draft-ietf-xrblock-rtcp-xr-pdv-08 (work in
December 2011. progress), September 2012.
[Padhye] Padhye, J., Firoiu, V., Towsley, D., and J. Kurose,
"Modeling TCP Throughput: A Simple Model and its Empirical
Validation", Proc. ACM SIGCOMM 1998, DOI 10.1145/
285237.285291, August 1998.
[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition [RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP", of Explicit Congestion Notification (ECN) to IP",
RFC 3168, September 2001. RFC 3168, September 2001.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile "Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, February 2008. with Feedback (AVPF)", RFC 5104, February 2008.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
April 2010.
[RFC5450] Singer, D. and H. Desineni, "Transmission Time Offsets in [RFC5450] Singer, D. and H. Desineni, "Transmission Time Offsets in
RTP Streams", RFC 5450, March 2009. RTP Streams", RFC 5450, March 2009.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009.
[RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP [RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
Flows", RFC 6051, November 2010. Flows", RFC 6051, November 2010.
[RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
and K. Carlberg, "Explicit Congestion Notification (ECN)
for RTP over UDP", RFC 6679, August 2012.
Authors' Addresses Authors' Addresses
Colin Perkins Colin Perkins
University of Glasgow University of Glasgow
School of Computing Science School of Computing Science
Glasgow G12 8QQ Glasgow G12 8QQ
United Kingdom United Kingdom
Email: csp@csperkins.org Email: csp@csperkins.org
Varun Singh Varun Singh
Aalto University Aalto University
School of Science and Technology School of Electrical Engineering
Otakaari 5 A Otakaari 5 A
Espoo, FIN 02150 Espoo, FIN 02150
Finland Finland
Email: varun@comnet.tkk.fi Email: varun@comnet.tkk.fi
URI: http://www.netlab.tkk.fi/~varun/ URI: http://www.netlab.tkk.fi/~varun/
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