draft-ietf-avtcore-rtp-circuit-breakers-01.txt   draft-ietf-avtcore-rtp-circuit-breakers-02.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 25, 2013 Aalto University Expires: August 26, 2013 Aalto University
October 22, 2012 February 22, 2013
RTP Congestion Control: Circuit Breakers for Unicast Sessions Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions
draft-ietf-avtcore-rtp-circuit-breakers-01 draft-ietf-avtcore-rtp-circuit-breakers-02
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; instead, it defines a minimal
set of "circuit-breakers". Circuit-breakers are conditions under set of RTP "circuit-breakers". Circuit-breakers are conditions under
which an RTP flow is expected to stop transmitting media to protect which an RTP sender needs to stop transmitting media data in order to
the network from excessive congestion. It is expected that all RTP protect the network from excessive congestion. It is expected that,
applications running on best-effort networks will be able to run in the absence of severe congestion, all RTP applications running on
without triggering these circuit breakers in normal operation. Any best-effort IP networks will be able to run without triggering these
future RTP congestion control specification is expected to operate circuit breakers. Any future RTP congestion control specification
within the envelope defined by these circuit breakers. will be expected to operate within the constraints 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.
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
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Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
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time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
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This Internet-Draft will expire on April 25, 2013. This Internet-Draft will expire on August 26, 2013.
Copyright Notice Copyright Notice
Copyright (c) 2012 IETF Trust and the persons identified as the Copyright (c) 2013 IETF Trust and the persons identified as the
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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: Media Timeout . . . . . . . . 7 4.1. RTP/AVP Circuit Breaker #1: Media Timeout . . . . . . . . 8
4.2. RTP/AVP Circuit Breaker #2: RTCP Timeout . . . . . . . . . 8 4.2. RTP/AVP Circuit Breaker #2: RTCP Timeout . . . . . . . . . 8
4.3. RTP/AVP Circuit Breaker #3: Congestion . . . . . . . . . . 9 4.3. RTP/AVP Circuit Breaker #3: Congestion . . . . . . . . . . 9
5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile . 11 4.4. Ceasing Transmission . . . . . . . . . . . . . . . . . . . 12
6. Impact of RTCP XR . . . . . . . . . . . . . . . . . . . . . . 12 5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile . 12
7. Impact of Explicit Congestion Notification (ECN) . . . . . . . 12 6. Impact of RTCP XR . . . . . . . . . . . . . . . . . . . . . . 13
8. Security Considerations . . . . . . . . . . . . . . . . . . . 13 7. Impact of Explicit Congestion Notification (ECN) . . . . . . . 14
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 13 8. Security Considerations . . . . . . . . . . . . . . . . . . . 14
10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 13 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 14
11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 13 10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 14
11.1. Normative References . . . . . . . . . . . . . . . . . . . 13 11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 15
11.2. Informative References . . . . . . . . . . . . . . . . . . 14 11.1. Normative References . . . . . . . . . . . . . . . . . . . 15
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 15 11.2. Informative References . . . . . . . . . . . . . . . . . . 15
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 17
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
skipping to change at page 3, line 44 skipping to change at page 3, line 44
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. Congestion control for multicast flows is outside the scope changes. Congestion control for multicast flows is outside the scope
of this memo. of this memo. Multicast traffic needs different solutions, since the
available bandwidth estimator for a group of receivers will differ
from that for a single receiver, and because multicast congestion
control has to consider issues of fairness across groups of receivers
that do not apply to unicast flows.
Congestion control for unicast RTP traffic can be implemented in one 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.
skipping to change at page 4, line 49 skipping to change at page 5, line 5
Sender Report (SR) packets to convey information about the media Sender Report (SR) packets to convey information about the media
transmission. RTCP SR packets contain data that can be used to transmission. RTCP SR packets contain data that can be used to
reconstruct media timing at a receiver, along with a count of the reconstruct media timing at a receiver, along with a count of the
total number of octets and packets sent. RTCP RR packets report total number of octets and packets sent. RTCP RR packets report
on the fraction of packets lost in the last reporting interval, on the fraction of packets lost in the last reporting interval,
the cumulative number of packets lost, the highest sequence number the cumulative number of packets lost, the highest sequence number
received, and the inter-arrival jitter. The RTCP RR packets also received, and the inter-arrival jitter. The RTCP RR packets also
contain timing information that allows the sender to estimate the contain timing information that allows the sender to estimate the
network round trip time (RTT) to the receivers. RTCP reports are network round trip time (RTT) to the receivers. RTCP reports are
sent periodically, with the reporting interval being determined by sent periodically, with the reporting interval being determined by
the number of participants in the session and a configured session the number of SSRCs used in the session and a configured session
bandwidth estimate. The interval between reports sent from each bandwidth estimate (the number of SSRCs used is usually two in a
receiver tends to be on the order of a few seconds on average, and unicast session, one for each participant, but can be greater if
it is randomised to avoid synchronisation of reports from multiple the participants send multiple media streams). The interval
receivers. RTCP RR packets allow a receiver to report ongoing between reports sent from each receiver tends to be on the order
network congestion to the sender. However, if a receiver detects of a few seconds on average, and it is randomised to avoid
the onset of congestion partway through a reporting interval, the synchronisation of reports from multiple receivers. RTCP RR
base RTP specification contains no provision for sending the RTCP packets allow a receiver to report ongoing network congestion to
RR packet early, and the receiver has to wait until the next the sender. However, if a receiver detects the onset of
scheduled reporting interval. congestion partway through a reporting interval, the base RTP
specification contains no provision for sending the RTCP RR packet
early, and the receiver has to wait until the next scheduled
reporting interval.
o The RTCP Extended Reports (XR) [RFC3611] allow reporting of more 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], 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 [RFC6843], [RFC6798]. 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
achieved using the Extended RTP Profile for RTCP-Based Feedback achieved using the Extended RTP Profile for RTCP-Based Feedback
(RTP/AVPF) [RFC4585] in place of the more common RTP/AVP profile (RTP/AVPF) [RFC4585] in place of the more common RTP/AVP profile
[RFC3551]. This modifies the RTCP timing rules to allow RTCP [RFC3551]. This modifies the RTCP timing rules to allow RTCP
reports to be sent early, in some cases immediately, provided the reports to be sent early, in some cases immediately, provided the
average RTCP reporting interval remains unchanged. It also average RTCP reporting interval remains unchanged. It also
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
it keeps the same average RTCP reporting interval as the base RTP with the RTP/AVP profile, as it keeps the same average RTCP
specification. The RTP Codec Control Messages [RFC5104] extend reporting interval as the base RTP specification. The RTP Codec
the RTP/AVPF profile with additional feedback messages that can be Control Messages [RFC5104] extend the RTP/AVPF profile with
used to influence that way in which rate adaptation occurs. The additional feedback messages that can be used to influence that
dynamics of how rapidly feedback can be sent are unchanged. way in which rate adaptation occurs. The dynamics of how rapidly
feedback can be sent are unchanged.
o Finally, Explicit Congestion Notification (ECN) for RTP over UDP o Finally, Explicit Congestion Notification (ECN) for RTP over UDP
[RFC6679] can be used to provide feedback on the number of packets [RFC6679] can be used to provide feedback on the number of packets
that received an ECN Congestion Experienced (CE) mark. This RTCP that received an ECN Congestion Experienced (CE) mark. This RTCP
extension builds on the RTP/AVPF profile to allow rapid congestion extension builds on the RTP/AVPF profile to allow rapid congestion
feedback when ECN is supported. feedback when ECN is 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
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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.
RTCP is a fundamental part of the RTP protocol, and the mechanisms
described here rely on the implementation of RTCP. Implementations
which claim to support RTP, but that do not implement RTCP, cannot
use the circuit breaker mechanisms described in this memo. Such
implementations SHOULD NOT be used on networks that might be subject
to congestion unless equivalent mechanisms are defined using some
non-RTCP feedback channel to report congestion and signal circuit
breaker conditions.
Three potential congestion signals are available from the basic RTCP Three potential congestion signals are available from the basic RTCP
SR/RR packets and are reported for each synchronisation source (SSRC) SR/RR packets and are reported for each synchronisation source (SSRC)
in the RTP session: in the RTP session:
1. The sender can estimate the network round-trip time once per RTCP 1. The sender can estimate the network round-trip time once per RTCP
reporting interval, based on the contents and timing of RTCP SR reporting interval, based on the contents and timing of RTCP SR
and RR packets. and RR packets.
2. Receivers report a jitter estimate (the statistical variance of 2. Receivers report a jitter estimate (the statistical variance of
the RTP data packet inter-arrival time) calculated over the RTCP the RTP data packet inter-arrival time) calculated over the RTCP
reporting interval. Due to the nature of the jitter calculation reporting interval. Due to the nature of the jitter calculation
([RFC3550], section 6.4.4), the jitter is only meaningful for RTP ([RFC3550], section 6.4.4), the jitter is only meaningful for RTP
flows that send a single data packet for each RTP timestamp value flows that send a single data packet for each RTP timestamp value
(i.e., audio flows, or video flows where each frame comprises one (i.e., audio flows, or video flows where each packet comprises
RTP packet). one video frame).
3. Receivers report the fraction of RTP data packets lost during the 3. Receivers report the fraction of RTP data packets lost during the
RTCP reporting interval, and the cumulative number of RTP packets RTCP reporting interval, and the cumulative number of RTP packets
lost over the entire RTP session. lost over the entire RTP session.
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
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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
cumulative number of packets lost. These are robust indicators of cumulative number of packets lost. If considered carefully, these
congestion in a network where packet loss is primarily due to queue can be effective indicators that congestion is occurring in networks
overflows, although less accurate in networks where losses can be where packet loss is primarily due to queue overflows, although loss
caused by non-congestive packet corruption. TCP uses packet loss as caused by non-congestive packet corruption can distort the result in
a congestion signal. some networks. TCP congestion control intentionally tries to fill
the router queues, and uses the resulting packet loss as congestion
feedback. An RTP flow competing with TCP traffic will therefore
expect to see a non-zero packet loss fraction that has to be related
to TCP dynamics to estimate available capacity. This behaviour of
TCP is reflected in the congestion circuit breaker below, and will
affect the design of any RTP congestion control protocol.
Two packet loss regimes can be observed: 1) RTCP RR packets show a 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. A third condition is that data is being sent but no
loss regimes in the following. RTCP feedback is received at all, corresponding to a failure of the
reverse path. We derive circuit breaker conditions for these loss
regimes in the following.
4.1. RTP/AVP Circuit Breaker #1: Media 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, but the RTCP SR or RR packets
packets report a non-increasing extended highest sequence number reporting on that SSRC indicate a non-increasing extended highest
received, this is an indication that those RTP data packets are not sequence number received, this is an indication that those RTP data
reaching the receiver. This could be a short-term issue affecting packets are not reaching the receiver. This could be a short-term
only a few packets, perhaps caused by a slow-to-open firewall or a issue affecting only a few packets, perhaps caused by a slow-to-open
transient connectivity problem, but if the issue persists, it is a firewall or a transient connectivity problem, but if the issue
sign of a more ongoing and significant problem. Accordingly, if a persists, it is a sign of a more ongoing and significant problem.
sender of RTP data packets receives two or more consecutive RTCP RR Accordingly, if a sender of RTP data packets receives two or more
packets from the same receiver that correspond to its transmission, consecutive RTCP SR or RR packets from the same receiver, and those
and have a non-increasing extended highest sequence number received packets correspond to its transmission and have a non-increasing
field (i.e., at least three RTCP RR packets that report the same extended highest sequence number received field (i.e., the sender
value in the extended highest sequence number received field, when receivers at least three RTCP SR or RR packets that report the same
the sender has sent data packets that would have caused an increase value in the extended highest sequence number received field for an
in the reported value of the extended highest sequence number SSRC, but the sender has sent RTP data packets for that SSRC that
received if they had reached the receiver), then that sender SHOULD would have caused an increase in the reported value of the extended
cease transmission. What it means to cease transmission depends on highest sequence number received if they had reached the receiver),
the application, but the intention is that the application will stop then that sender SHOULD cease transmission (see Section 4.4).
sending RTP data packets until the user makes an explicit attempt to
restart the call (RTP flows halted by the circuit breaker SHOULD NOT
be restarted automatically unless the sender has received information
that the congestion has dissipated).
Systems that usually send at a high data rate, but that can reduce
their data rate significantly (i.e., by at least a factor of ten),
MAY first reduce their sending rate to this lower value to see if
this resolves the congestion, but MUST then cease transmission if the
problem does not resolve itself within a further two RTCP reporting
intervals. An example of this might be a video conferencing system
that backs off to sending audio only, before completely dropping the
call. If such a reduction in sending rate resolves the congestion
problem, the sender MAY gradually increase the rate at which it sends
data after a reasonable amount of time has passed, provided it takes
care not to cause the problem to recur ("reasonable" is intentionally
not defined here).
The choice of two RTCP reporting intervals is to give enough time for The reason for waiting for two or more consecutive RTCP packets with
transient problems to resolve themselves, but to stop problem flows a non-increasing extended highest sequence number is to give enough
quickly enough to avoid causing serious ongoing network congestion. time for transient reception problems to resolve themselves, but to
A single RTCP report showing no reception could be caused by numerous stop problem flows quickly enough to avoid causing serious ongoing
transient faults, and so will not cease transmission. Waiting for network congestion. A single RTCP report showing no reception could
more than two RTCP reports before stopping a flow might avoid some be caused by a transient fault, and so will not cease transmission.
false positives, but would lead to problematic flows running for a Waiting for more than two consecutive RTCP reports before stopping a
long time before being cut off. flow might avoid some false positives, but could lead to problematic
flows running for a long time period (potentially tens of seconds,
depending on the RTCP reporting interval) before being cut off.
4.2. RTP/AVP Circuit Breaker #2: RTCP Timeout 4.2. RTP/AVP Circuit Breaker #2: RTCP Timeout
In addition to media timeouts, as were discussed in Section 4.1, an 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 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 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 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 receiver to send RTCP reports, or a failure of the return path that
is preventing those RTCP reporting from being delivered. is preventing those RTCP reporting from being delivered. In either
case, it is not safe to continue transmission, since the sender has
no way of knowing if it is causing congestion. Accordingly, an RTP
sender that has not received any RTCP SR or RTCP RR packets reporting
on the SSRC it is using for three or more RTCP reporting intervals
SHOULD cease transmission (see Section 4.4). When calculating the
timeout, the fixed minimum RTCP reporting interval SHOULD be used
(based on the rationale in Section 6.2 of RFC 3550 [RFC3550]).
According to RFC 3550 [RFC3550], any participant that has not sent an The choice of three RTCP reporting intervals as the timeout is made
RTCP packet within the last two RTCP intervals is removed from the following Section 6.3.5 of RFC 3550 [RFC3550]. This specifies that
sender list. Therefore, an RTP sender SHOULD cease transmission if participants in an RTP session will timeout and remove an RTP sender
it does not receive a single RTCP RR packet and during this period from the list of active RTP senders if no RTP data packets have been
has sent 3 RTCP SR packets to the RTP receiver. Similarly, the same received from that RTP sender within the last two RTCP reporting
circuit breaker rule applies to an RTCP receiver which has not intervals. Using a timeout of three RTCP reporting intervals is
received a single SR packet, and in the corresponding period it has therefore large enough that the other participants will have timed
sent 3 RTCP RR packets. What it means to cease transmission depends out the sender if a network problem stops the data packets it is
on the application, but the intention is that the application will sending from reaching the receivers, even allowing for loss of some
stop sending RTP data packets until the user makes an explicit RTCP packets.
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 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 those 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
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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] section only interested in order of magnitude comparison ([Floyd] section
3.2.1 shows that the difference is small for steady-state conditions 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 and random loss, but using the loss fraction is more conservative in
the case of bursty loss). the case of bursty loss).
The congestion circuit breaker is therefore: when RTCP RR packets are The congestion circuit breaker is therefore: when a sender receives
received, estimate the TCP throughput using the simplified equation an RTCP SR or RR packet that contains a report block for an SSRC it
above, and the measured R, p (approximated by the loss fraction), and is using, that sender has to check the fraction lost field in that
s. Compare this with the actual sending rate. If the actual sending report block to determine if there is a non-zero packet loss rate.
rate is more than ten times the estimated sending rate derived from If the fraction lost field is zero, then continue sending as normal.
the TCP throughput equation for two consecutive RTCP reporting If the fraction lost is greater than zero, then estimate the TCP
intervals, the sender SHOULD cease transmission. What it means to throughput using the simplified equation above, and the measured R, p
cease transmission depends on the application, but the intention is (approximated by the fraction lost), and s. Compare this with the
that the application will stop sending RTP data packets until the actual sending rate. If the actual sending rate is more than ten
user makes an explicit attempt to restart the call (RTP flows halted times the estimated sending rate derived from the TCP throughput
by the circuit breaker SHOULD NOT be restarted automatically unless equation for two consecutive RTCP reporting intervals, the sender
the sender has received information that the congestion has SHOULD cease transmission (see Section 4.4). If the RTP sender is
dissipated). 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, since that interval scales with the
media data rate.
Systems that usually send at a high data rate, but that 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 (see Section 4.4). An example of this might be a video
that backs off to sending audio only, before completely dropping the conferencing system that backs off to sending audio only, before
call. If such a reduction in sending rate resolves the congestion completely dropping the call. If such a reduction in sending rate
problem, the sender MAY gradually increase the rate at which it sends resolves the congestion problem, the sender MAY gradually increase
data after a reasonable amount of time has passed, provided it takes the rate at which it sends data after a reasonable amount of time has
care not to cause the problem to recur ("reasonable" is intentionally passed, provided it takes care not to cause the problem to recur
not defined here). ("reasonable" is intentionally not defined here).
As in Section 4.1, we use two reporting intervals to avoid triggering As in Section 4.1, we use two reporting intervals to avoid triggering
the circuit breaker on transient failures. This circuit breaker is a the circuit breaker on transient failures. This circuit breaker is a
worst-case condition, and congestion control needs to be performed to worst-case condition, and congestion control needs to be performed to
keep well within this bound. It is expected that the circuit breaker keep well within this bound. It is expected that the circuit breaker
will only be triggered if the usual congestion control fails for some will only be triggered if the usual congestion control fails for some
reason. reason.
4.4. Ceasing Transmission
What it means to cease transmission depends on the application, but
the intention is that the application will stop sending RTP data
packets to a particular destination 3-tuple (transport protocol,
destination port, IP address), until the user makes an explicit
attempt to restart the call. It is important that a human user is
involved in the decision to try to restart the call, since that user
will eventually give up if the calls repeatedly trigger the circuit
breaker. This will help avoid problems with automatic redial systems
from congesting the network. Accordingly, RTP flows halted by the
circuit breaker SHOULD NOT be restarted automatically unless the
sender has received information that the congestion has dissipated.
It is recognised that the RTP implementation in some systems might
not be able to determine if a call set-up request was initiated by a
human user, or automatically by some scripted higher-level component
of the system. These implementations SHOULD rate limit attempts to
restart a call to the same destination 3-tuple as used by a previous
call that was recently halted by the circuit breaker. The chosen
rate limit ought to not exceed the rate at which an annoyed human
caller might redial a misbehaving phone.
5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile 5. RTP Circuit Breakers for Systems Using the RTP/AVPF Profile
Use of the Extended RTP Profile for RTCP-based Feedback (RTP/AVPF) Use of the Extended RTP Profile for RTCP-based Feedback (RTP/AVPF)
[RFC4585] allows receivers to send early RTCP reports in some cases, [RFC4585] allows receivers to send early RTCP reports in some cases,
to inform the sender about particular events in the media stream. to inform the sender about particular events in the media stream.
There are several use cases for such early RTCP reports, including There are several use cases for such early RTCP reports, including
providing rapid feedback to a sender about the onset of congestion. providing rapid feedback to a sender about the onset of congestion.
Receiving rapid feedback about congestion events potentially allows Receiving rapid feedback about congestion events potentially allows
congestion control algorithms to be more responsive, and to better congestion control algorithms to be more responsive, and to better
skipping to change at page 12, line 14 skipping to change at page 13, line 5
feedback reports that suit their requirements. Since these reports feedback reports that suit their requirements. Since these reports
are not yet defined, and likely very specific to the details of the are not yet defined, and likely very specific to the details of the
congestion control algorithm chosen, they cannot be used as part of congestion control algorithm chosen, they cannot be used as part of
the generic RTP circuit breaker. the generic RTP circuit breaker.
If the extension for Reduced-Size RTCP [RFC5506] is not used, early If the extension for Reduced-Size RTCP [RFC5506] is not used, early
RTCP feedback packets sent according to the RTP/AVPF profile will be RTCP feedback packets sent according to the RTP/AVPF profile will be
compound RTCP packets that include an RTCP SR/RR packet. That RTCP compound RTCP packets that include an RTCP SR/RR packet. That RTCP
SR/RR packet MUST be processed as if it were sent as a regular RTCP SR/RR packet MUST be processed as if it were sent as a regular RTCP
report and counted towards the circuit breaker conditions specified report and counted towards the circuit breaker conditions specified
in Section 4.1 and Section 4.3 of this memo. This will potentially in Section 4 of this memo. This will potentially make the RTP
make the RTP circuit breaker fire earlier than it would if the RTP/ circuit breaker fire earlier than it would if the RTP/AVPF profile
AVPF profile was not used. was not used.
Reduced-size RTCP reports sent under to the RTP/AVPF early feedback 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 rules that do not contain an RTCP SR or RR packet MUST be ignored by
the RTP circuit breaker (they do not contain the information used by the RTP circuit breaker (they do not contain the information used by
the circuit breaker algorithm). In this case, the circuit breaker the circuit breaker algorithm). Reduced-size RTCP reports sent under
will only use the information contained in the periodic RTCP SR/RR the RTP/AVPF early feedback rules that contain RTCP SR or RR packets
packets. This allows the use of low-overhead early RTP/AVPF feedback MUST be processed as if they were sent as regular RTCP reports, and
without triggering the RTP circuit breaker, and so is suitable for counted towards the circuit breaker conditions specified in Section 4
of this memo. This will potentially make the RTP circuit breaker
fire earlier than it would if the RTP/AVPF profile was not used.
When using ECN with RTP (see Section 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 report is sent in
an compound RTCP packet along with an RTCP SR/RR report packet.
Reports of ECN-CE packets sent as reduced-size RTCP ECN feedback
packets without an RTCP SR/RR packet MUST be ignored.
These rules are intended to allow the use of low-overhead early 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 RTP congestion control algorithms that need to quickly report loss
events in between regular RTCP reports. events in between regular RTCP reports. The reaction to reduced-size
RTCP SR/RR packets is to allow such algorithms to send feedback that
can trigger the circuit breaker, when desired.
6. Impact of RTCP XR 6. Impact of RTCP XR
RTCP Extended Report (XR) blocks provide additional reception quality RTCP Extended Report (XR) blocks provide additional reception quality
metrics, but do not change the RTCP timing rules. Some of the RTCP metrics, but do not change the RTCP timing rules. Some of the RTCP
XR blocks provide information that might be useful for congestion XR blocks provide information that might be useful for congestion
control purposes, others provided non-congestion-related metrics. control purposes, others provided non-congestion-related metrics.
The presence of RTCP XR blocks in a compound RTCP packet does not With the exception of RTCP XR ECN Summary Reports (see Section 7),
affect the RTP circuit breaker algorithm; for consistency and ease of the presence of RTCP XR blocks in a compound RTCP packet does not
implementation, only the reception report blocks contained in RTCP SR affect the RTP circuit breaker algorithm. For consistency and ease
or RR packets are used by the RTP circuit breaker algorithm. 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) 7. Impact of Explicit Congestion Notification (ECN)
The use of ECN for RTP flows does not affect the media timeout RTP
circuit breaker (Section 4.1) or the RTCP timeout circuit breaker
(Section 4.2), since these are both connectivity checks that simply
determinate if any packets are being received.
ECN-CE marked packets SHOULD be treated as if it were lost for the ECN-CE marked packets SHOULD be treated as if it were lost for the
purposes of congestion control, when determining the optimal media purposes of congestion control, when determining the optimal media
sending rate for an RTP flow. If an RTP sender has negotiated ECN sending rate for an RTP flow. If an RTP sender has negotiated ECN
support for an RTP session, and has successfully initiated ECN use on support for an RTP session, and has successfully initiated ECN use on
the path to the receiver [RFC6679], then ECN-CE marked packets SHOULD the path to the receiver [RFC6679], then ECN-CE marked packets SHOULD
be treated as if they were lost when calculating if the congestion- be treated as if they were lost when calculating if the congestion-
based RTP circuit breaker (Section 4.3) has been met. based RTP circuit breaker (Section 4.3) has been met. The count of
ECN-CE marked RTP packets is returned in RTCP XR ECN summary report
The use of ECN for RTP flows does not affect the media timeout RTP packets if support for ECN has been initiated for an RTP session.
circuit breaker (Section 4.1) or the RTCP timeout circuit breaker
(Section 4.2), since these are both connectivity checks that simply
determinate if any packets are being received.
8. Security Considerations 8. Security Considerations
The security considerations of [RFC3550] apply. The security considerations of [RFC3550] apply.
If the RTP/AVPF profile is used to provide rapid RTCP feedback, the If the RTP/AVPF profile is used to provide rapid RTCP feedback, the
security considerations of [RFC4585] apply. If ECN feedback for RTP security considerations of [RFC4585] apply. If ECN feedback for RTP
over UDP/IP is used, the security considerations of [RFC6679] apply. over UDP/IP is used, the security considerations of [RFC6679] apply.
If non-authenticated RTCP reports are used, an on-path attacker can If non-authenticated RTCP reports are used, an on-path attacker can
skipping to change at page 13, line 30 skipping to change at page 14, line 44
due to the need to guess the randomly chosen RTP SSRC value and the 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 RTP sequence number. This attack can be avoided if RTCP packets are
authenticated, for example using the Secure RTP profile [RFC3711]. 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. Acknowledgements 10. Acknowledgements
The authors would like to thank Harald Alvestrand, Randell Jesup, The authors would like to thank Bernard Aboba, Harald Alvestrand,
Matt Mathis, and Abheek Saha for their valuable feedback. Kevin Gross, Cullen Jennings, Randell Jesup, Jonathan Lennox, Matt
Mathis, Stephen McQuistin, Eric Rescorla, and Abheek Saha for their
valuable feedback.
11. References 11. References
11.1. 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",
skipping to change at page 14, line 27 skipping to change at page 15, line 42
11.2. Informative References 11.2. Informative References
[Floyd] Floyd, S., Handley, M., Padhye, J., and J. Widmer, [Floyd] Floyd, S., Handley, M., Padhye, J., and J. Widmer,
"Equation-Based Congestion Control for Unicast "Equation-Based Congestion Control for Unicast
Applications", Proc. ACM SIGCOMM 2000, DOI 10.1145/ Applications", Proc. ACM SIGCOMM 2000, DOI 10.1145/
347059.347397, August 2000. 347059.347397, August 2000.
[I-D.ietf-xrblock-rtcp-xr-burst-gap-discard] [I-D.ietf-xrblock-rtcp-xr-burst-gap-discard]
Clark, A., Huang, R., and W. Wu, "RTP Control Clark, A., Huang, R., and W. Wu, "RTP Control
Protocol(RTCP) Extended Report (XR) Block for Discard Protocol(RTCP) Extended Report (XR) Block for Burst/Gap
Count metric Reporting", Discard metric Reporting",
draft-ietf-xrblock-rtcp-xr-burst-gap-discard-06 (work in draft-ietf-xrblock-rtcp-xr-burst-gap-discard-10 (work in
progress), October 2012. progress), January 2013.
[I-D.ietf-xrblock-rtcp-xr-burst-gap-loss] [I-D.ietf-xrblock-rtcp-xr-burst-gap-loss]
Clark, A., Zhang, S., Zhao, J., and W. Wu, "RTP Control Clark, A., Zhang, S., Zhao, J., and W. Wu, "RTP Control
Protocol (RTCP) Extended Report (XR) Block for Burst/Gap Protocol (RTCP) Extended Report (XR) Block for Burst/Gap
Loss metric Reporting", Loss metric Reporting",
draft-ietf-xrblock-rtcp-xr-burst-gap-loss-04 (work in draft-ietf-xrblock-rtcp-xr-burst-gap-loss-08 (work in
progress), October 2012. progress), January 2013.
[I-D.ietf-xrblock-rtcp-xr-delay]
Clark, A., Gross, K., and W. Wu, "RTP Control Protocol
(RTCP) Extended Report (XR) Block for Delay metric
Reporting", draft-ietf-xrblock-rtcp-xr-delay-10 (work in
progress), October 2012.
[I-D.ietf-xrblock-rtcp-xr-discard] [I-D.ietf-xrblock-rtcp-xr-discard]
Clark, A., Zorn, G., and W. Wu, "RTP Control Protocol Clark, A., Zorn, G., and W. Wu, "RTP Control Protocol
(RTCP) Extended Report (XR) Block for Discard Count metric (RTCP) Extended Report (XR) Block for Discard Count metric
Reporting", draft-ietf-xrblock-rtcp-xr-discard-09 (work in Reporting", draft-ietf-xrblock-rtcp-xr-discard-11 (work in
progress), October 2012. progress), December 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, "RTP Control Protocol Ott, J., Singh, V., and I. Curcio, "RTP Control Protocol
(RTCP) Extended Reports (XR) for Run Length Encoding (RLE) (RTCP) Extended Reports (XR) for Run Length Encoding (RLE)
of Discarded Packets", of Discarded Packets",
draft-ietf-xrblock-rtcp-xr-discard-rle-metrics-04 (work in draft-ietf-xrblock-rtcp-xr-discard-rle-metrics-05 (work in
progress), July 2012. progress), December 2012.
[I-D.ietf-xrblock-rtcp-xr-pdv]
Clark, A. and W. Wu, "RTP Control Protocol (RTCP) Extended
Report (XR) Block for Packet Delay Variation Metric
Reporting", draft-ietf-xrblock-rtcp-xr-pdv-08 (work in
progress), September 2012.
[Padhye] Padhye, J., Firoiu, V., Towsley, D., and J. Kurose, [Padhye] Padhye, J., Firoiu, V., Towsley, D., and J. Kurose,
"Modeling TCP Throughput: A Simple Model and its Empirical "Modeling TCP Throughput: A Simple Model and its Empirical
Validation", Proc. ACM SIGCOMM 1998, DOI 10.1145/ Validation", Proc. ACM SIGCOMM 1998, DOI 10.1145/
285237.285291, August 1998. 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.
skipping to change at page 16, line 5 skipping to change at page 16, line 49
Real-Time Transport Control Protocol (RTCP): Opportunities Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009. 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., [RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
and K. Carlberg, "Explicit Congestion Notification (ECN) and K. Carlberg, "Explicit Congestion Notification (ECN)
for RTP over UDP", RFC 6679, August 2012. for RTP over UDP", RFC 6679, August 2012.
[RFC6798] Clark, A. and Q. Wu, "RTP Control Protocol (RTCP) Extended
Report (XR) Block for Packet Delay Variation Metric
Reporting", RFC 6798, November 2012.
[RFC6843] Clark, A., Gross, K., and Q. Wu, "RTP Control Protocol
(RTCP) Extended Report (XR) Block for Delay Metric
Reporting", RFC 6843, January 2013.
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
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