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Versions: (draft-perkins-avtcore-rtp-circuit-breakers)
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AVTCORE Working Group C. S. Perkins
Internet-Draft University of Glasgow
Updates: 3550 (if approved) V. Singh
Intended status: Standards Track Aalto University
Expires: September 24, 2015 March 23, 2015
Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions
draft-ietf-avtcore-rtp-circuit-breakers-10
Abstract
The Real-time Transport Protocol (RTP) is widely used in telephony,
video conferencing, and telepresence applications. Such applications
are often run on best-effort UDP/IP networks. If congestion control
is not implemented in the applications, then network congestion will
deteriorate the user's multimedia experience. This document does not
propose a congestion control algorithm; instead, it defines a minimal
set of RTP "circuit-breakers". Circuit-breakers are conditions under
which an RTP sender needs to stop transmitting media data in order to
protect the network from excessive congestion. It is expected that,
in the absence of severe congestion, all RTP applications running on
best-effort IP networks will be able to run without triggering these
circuit breakers. Any future RTP congestion control specification
will be expected to operate within the constraints defined by these
circuit breakers.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on September 24, 2015.
Copyright Notice
Copyright (c) 2015 IETF Trust and the persons identified as the
document authors. All rights reserved.
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This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Background . . . . . . . . . . . . . . . . . . . . . . . . . 3
4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile . 6
4.1. RTP/AVP Circuit Breaker #1: Media Timeout . . . . . . . . 7
4.2. RTP/AVP Circuit Breaker #2: RTCP Timeout . . . . . . . . 8
4.3. RTP/AVP Circuit Breaker #3: Congestion . . . . . . . . . 9
4.4. RTP/AVP Circuit Breaker #4: Media Usability . . . . . . . 13
4.5. Choice of Circuit Breaker Interval . . . . . . . . . . . 14
4.6. Ceasing Transmission . . . . . . . . . . . . . . . . . . 15
5. RTP Circuit Breakers and the RTP/AVPF and RTP/SAVPF Profiles 16
6. Impact of RTCP Extended Reports (XR) . . . . . . . . . . . . 17
7. Impact of RTCP Reporting Groups . . . . . . . . . . . . . . . 17
8. Impact of Explicit Congestion Notification (ECN) . . . . . . 18
9. Impact of Bundled Media and Layered Coding . . . . . . . . . 18
10. Security Considerations . . . . . . . . . . . . . . . . . . . 18
11. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 19
12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 19
13. References . . . . . . . . . . . . . . . . . . . . . . . . . 19
13.1. Normative References . . . . . . . . . . . . . . . . . . 19
13.2. Informative References . . . . . . . . . . . . . . . . . 20
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 22
1. Introduction
The Real-time Transport Protocol (RTP) [RFC3550] is widely used in
voice-over-IP, video teleconferencing, and telepresence systems.
Many of these systems run over best-effort UDP/IP networks, and can
suffer from packet loss and increased latency if network congestion
occurs. Designing effective RTP congestion control algorithms, to
adapt the transmission of RTP-based media to match the available
network capacity, while also maintaining the user experience, is a
difficult but important problem. Many such congestion control and
media adaptation algorithms have been proposed, but to date there is
no consensus on the correct approach, or even that a single standard
algorithm is desirable.
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This memo does not attempt to propose a new RTP congestion control
algorithm. Rather, it proposes a minimal set of "RTP circuit
breakers"; conditions under which there is general agreement that an
RTP flow is causing serious congestion, and ought to cease
transmission. The RTP circuit breakers proposed in this memo are a
specific instance of the general class of network transport circuit
breakers [I-D.ietf-tsvwg-circuit-breaker], designed to act as a
protection mechanism of last resort to avoid persistent congestion.
It is expected that future standards-track congestion control
algorithms for RTP will operate within the envelope defined by this
memo.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
This interpretation of these key words applies only when written in
ALL CAPS. Mixed- or lower-case uses of these key words are not to be
interpreted as carrying special significance in this memo.
3. Background
We consider congestion control for unicast RTP traffic flows. This
is the problem of adapting the transmission of an audio/visual data
flow, encapsulated within an RTP transport session, from one sender
to one receiver, so that it matches the available network bandwidth.
Such adaptation needs to be done in a way that limits the disruption
to the user experience caused by both packet loss and excessive rate
changes. Congestion control for multicast flows is outside the scope
of this memo. Multicast traffic needs different solutions, since the
available bandwidth estimator for a group of receivers will differ
from that for a single receiver, and because multicast congestion
control has to consider issues of fairness across groups of receivers
that do not apply to unicast flows.
Congestion control for unicast RTP traffic can be implemented in one
of two places in the protocol stack. One approach is to run the RTP
traffic over a congestion controlled transport protocol, for example
over TCP, and to adapt the media encoding to match the dictates of
the transport-layer congestion control algorithm. This is safe for
the network, but can be suboptimal for the media quality unless the
transport protocol is designed to support real-time media flows. We
do not consider this class of applications further in this memo, as
their network safety is guaranteed by the underlying transport.
Alternatively, RTP flows can be run over a non-congestion controlled
transport protocol, for example UDP, performing rate adaptation at
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the application layer based on RTP Control Protocol (RTCP) feedback.
With a well-designed, network-aware, application, this allows highly
effective media quality adaptation, but there is potential to disrupt
the network's operation if the application does not adapt its sending
rate in a timely and effective manner. We consider this class of
applications in this memo.
Congestion control relies on monitoring the delivery of a media flow,
and responding to adapt the transmission of that flow when there are
signs that the network path is congested. Network congestion can be
detected in one of three ways: 1) a receiver can infer the onset of
congestion by observing an increase in one-way delay caused by queue
build-up within the network; 2) if Explicit Congestion Notification
(ECN) [RFC3168] is supported, the network can signal the presence of
congestion by marking packets using ECN Congestion Experienced (CE)
marks; or 3) in the extreme case, congestion will cause packet loss
that can be detected by observing a gap in the received RTP sequence
numbers. Once the onset of congestion is observed, the receiver has
to send feedback to the sender to indicate that the transmission rate
needs to be reduced. How the sender reduces the transmission rate is
highly dependent on the media codec being used, and is outside the
scope of this memo.
There are several ways in which a receiver can send feedback to a
media sender within the RTP framework:
o The base RTP specification [RFC3550] defines RTCP Reception Report
(RR) packets to convey reception quality feedback information, and
Sender Report (SR) packets to convey information about the media
transmission. RTCP SR packets contain data that can be used to
reconstruct media timing at a receiver, along with a count of the
total number of octets and packets sent. RTCP RR packets report
on the fraction of packets lost in the last reporting interval,
the cumulative number of packets lost, the highest sequence number
received, and the inter-arrival jitter. The RTCP RR packets also
contain timing information that allows the sender to estimate the
network round trip time (RTT) to the receivers. RTCP reports are
sent periodically, with the reporting interval being determined by
the number of SSRCs used in the session and a configured session
bandwidth estimate (the number of SSRCs used is usually two in a
unicast session, one for each participant, but can be greater if
the participants send multiple media streams). The interval
between reports sent from each receiver tends to be on the order
of a few seconds on average, although it varies with the session
bandwidth, and sub-second reporting intervals are possible in high
bandwidth sessions, and it is randomised to avoid synchronisation
of reports from multiple receivers. RTCP RR packets allow a
receiver to report ongoing network congestion to the sender.
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However, if a receiver detects the onset of congestion part way
through a reporting interval, the base RTP specification contains
no provision for sending the RTCP RR packet early, and the
receiver has to wait until the next scheduled reporting interval.
o The RTCP Extended Reports (XR) [RFC3611] allow reporting of more
complex and sophisticated reception quality metrics, but do not
change the RTCP timing rules. RTCP extended reports of potential
interest for congestion control purposes are the extended packet
loss, discard, and burst metrics [RFC3611], [RFC7002], [RFC7097],
[RFC7003], [RFC6958]; and the extended delay metrics [RFC6843],
[RFC6798]. Other RTCP Extended Reports that could be helpful for
congestion control purposes might be developed in future.
o Rapid feedback about the occurrence of congestion events can be
achieved using the Extended RTP Profile for RTCP-Based Feedback
(RTP/AVPF) [RFC4585] (or its secure variant, RTP/SAVPF [RFC5124])
in place of the RTP/AVP profile [RFC3551]. This modifies the RTCP
timing rules to allow RTCP reports to be sent early, in some cases
immediately, provided the RTCP transmission rate keeps within its
bandwidth allocation. It also defines transport-layer feedback
messages, including negative acknowledgements (NACKs), that can be
used to report on specific congestion events. RTP Codec Control
Messages [RFC5104] extend the RTP/AVPF profile with additional
feedback messages that can be used to influence that way in which
rate adaptation occurs, but do not further change the dynamics of
how rapidly feedback can be sent. Use of the RTP/AVPF profile is
dependent on signalling.
o Finally, Explicit Congestion Notification (ECN) for RTP over UDP
[RFC6679] can be used to provide feedback on the number of packets
that received an ECN Congestion Experienced (CE) mark. This RTCP
extension builds on the RTP/AVPF profile to allow rapid congestion
feedback when ECN is supported.
In addition to these mechanisms for providing feedback, the sender
can include an RTP header extension in each packet to record packet
transmission times. There are two methods: [RFC5450] represents the
transmission time in terms of a time-offset from the RTP timestamp of
the packet, while [RFC6051] includes an explicit NTP-format sending
timestamp (potentially more accurate, but a higher header overhead).
Accurate sending timestamps can be helpful for estimating queuing
delays, to get an early indication of the onset of congestion.
Taken together, these various mechanisms allow receivers to provide
feedback on the senders when congestion events occur, with varying
degrees of timeliness and accuracy. The key distinction is between
systems that use only the basic RTCP mechanisms, without RTP/AVPF
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rapid feedback, and those that use the RTP/AVPF extensions to respond
to congestion more rapidly.
4. RTP Circuit Breakers for Systems Using the RTP/AVP Profile
The feedback mechanisms defined in [RFC3550] and available under the
RTP/AVP profile [RFC3551] are the minimum that can be assumed for a
baseline circuit breaker mechanism that is suitable for all unicast
applications of RTP. Accordingly, for an RTP circuit breaker to be
useful, it needs to be able to detect that an RTP flow is causing
excessive congestion using only basic RTCP features, without needing
RTCP XR feedback or the RTP/AVPF profile for rapid RTCP reports.
RTCP is a fundamental part of the RTP protocol, and the mechanisms
described here rely on the implementation of RTCP. Implementations
that claim to support RTP, but that do not implement RTCP, cannot use
the circuit breaker mechanisms described in this memo. Such
implementations SHOULD NOT be used on networks that might be subject
to congestion unless equivalent mechanisms are defined using some
non-RTCP feedback channel to report congestion and signal circuit
breaker conditions.
Three potential congestion signals are available from the basic RTCP
SR/RR packets and are reported for each synchronisation source (SSRC)
in the RTP session:
1. The sender can estimate the network round-trip time once per RTCP
reporting interval, based on the contents and timing of RTCP SR
and RR packets.
2. Receivers report a jitter estimate (the statistical variance of
the RTP data packet inter-arrival time) calculated over the RTCP
reporting interval. Due to the nature of the jitter calculation
([RFC3550], section 6.4.4), the jitter is only meaningful for RTP
flows that send a single data packet for each RTP timestamp value
(i.e., audio flows, or video flows where each packet comprises
one video frame).
3. Receivers report the fraction of RTP data packets lost during the
RTCP reporting interval, and the cumulative number of RTP packets
lost over the entire RTP session.
These congestion signals limit the possible circuit breakers, since
they give only limited visibility into the behaviour of the network.
RTT estimates are widely used in congestion control algorithms, as a
proxy for queuing delay measures in delay-based congestion control or
to determine connection timeouts. RTT estimates derived from RTCP SR
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and RR packets sent according to the RTP/AVP timing rules are too
infrequent to be useful though, and don't give enough information to
distinguish a delay change due to routing updates from queuing delay
caused by congestion. Accordingly, we cannot use the RTT estimate
alone as an RTP circuit breaker.
Increased jitter can be a signal of transient network congestion, but
in the highly aggregated form reported in RTCP RR packets, it offers
insufficient information to estimate the extent or persistence of
congestion. Jitter reports are a useful early warning of potential
network congestion, but provide an insufficiently strong signal to be
used as a circuit breaker.
The remaining congestion signals are the packet loss fraction and the
cumulative number of packets lost. If considered carefully, these
can be effective indicators that congestion is occurring in networks
where packet loss is primarily due to queue overflows, although loss
caused by non-congestive packet corruption can distort the result in
some networks. TCP congestion control [RFC5681] intentionally tries
to fill the router queues, and uses the resulting packet loss as
congestion feedback. An RTP flow competing with TCP traffic will
therefore expect to see a non-zero packet loss fraction that has to
be related to TCP dynamics to estimate available capacity. This
behaviour of TCP is reflected in the congestion circuit breaker
below, and will affect the design of any RTP congestion control
protocol.
Two packet loss regimes can be observed: 1) RTCP RR packets show a
non-zero packet loss fraction, while the extended highest sequence
number received continues to increment; and 2) RR packets show a loss
fraction of zero, but the extended highest sequence number received
does not increment even though the sender has been transmitting RTP
data packets. The former corresponds to the TCP congestion avoidance
state, and indicates a congested path that is still delivering data;
the latter corresponds to a TCP timeout, and is most likely due to a
path failure. A third condition is that data is being sent but no
RTCP feedback is received at all, corresponding to a failure of the
reverse path. We derive circuit breaker conditions for these loss
regimes in the following.
4.1. RTP/AVP Circuit Breaker #1: Media Timeout
If RTP data packets are being sent, but the RTCP SR or RR packets
reporting on that SSRC indicate a non-increasing extended highest
sequence number received, this is an indication that those RTP data
packets are not reaching the receiver. This could be a short-term
issue affecting only a few packets, perhaps caused by a slow-to-open
firewall or a transient connectivity problem, but if the issue
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persists, it is a sign of a more ongoing and significant problem.
Accordingly, if a sender of RTP data packets receives CB_INTERVAL or
more consecutive RTCP SR or RR packets from the same receiver (see
Section 4.5), and those packets correspond to its transmission and
have a non-increasing extended highest sequence number received
field, then that sender SHOULD cease transmission (see Section 4.6).
The extended highest sequence number received field is non-increasing
if the sender receives at least CB_INTERVAL consecutive RTCP SR or RR
packets that report the same value for this field, but it has sent
RTP data packets, at a rate of at least one per RTT, that would have
caused an increase in the reported value if they had reached the
receiver.
The rationale for waiting for CB_INTERVAL or more consecutive RTCP
packets with a non-increasing extended highest sequence number is to
give enough time for transient reception problems to resolve
themselves, but to stop problem flows quickly enough to avoid causing
serious ongoing network congestion. A single RTCP report showing no
reception could be caused by a transient fault, and so will not cease
transmission. Waiting for more than CB_INTERVAL consecutive RTCP
reports before stopping a flow might avoid some false positives, but
could lead to problematic flows running for a long time period
(potentially tens of seconds, depending on the RTCP reporting
interval) before being cut off. Equally, an application that sends
few packets when the packet loss rate is high runs the risk that the
media timeout circuit breaker triggers inadvertently. The chosen
timeout interval is a trade-off between these extremes.
The rationale for enforcing a minimum sending rate below which the
media timeout circuit breaker will not trigger is to avoid spurious
circuit breaker triggers when the number of packets sent per RTCP
reporting interval is small (e.g., a telephony application sends only
two RTP comfort noise packets during a five second RTCP reporting
interval, and both are lost; this is 100% packet loss, but it seems
extreme to terminate the RTP session). The one packet per RTT bound
derives from [RFC5405].
4.2. RTP/AVP Circuit Breaker #2: RTCP Timeout
In addition to media timeouts, as were discussed in Section 4.1, an
RTP session has the possibility of an RTCP timeout. This can occur
when RTP data packets are being sent, but there are no RTCP reports
returned from the receiver. This is either due to a failure of the
receiver to send RTCP reports, or a failure of the return path that
is preventing those RTCP reporting from being delivered. In either
case, it is not safe to continue transmission, since the sender has
no way of knowing if it is causing congestion. Accordingly, an RTP
sender that has not received any RTCP SR or RTCP RR packets reporting
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on the SSRC it is using for three or more of its RTCP reporting
intervals SHOULD cease transmission (see Section 4.6). When
calculating the timeout, the deterministic RTCP reporting interval,
Td, without the randomization factor, and using the fixed minimum
interval of Tmin=5 seconds, MUST be used. The rationale for this
choice of timeout is as described in Section 6.2 of [RFC3550] ("so
that implementations which do not use the reduced value for
transmitting RTCP packets are not timed out by other participants
prematurely"), as updated by Section 6.1.4 of
[I-D.ietf-avtcore-rtp-multi-stream] to account for the use of the RTP
/AVPF profile [RFC4585] or the RTP/SAVPF profile [RFC5124].
To reduce the risk of premature timeout, implementations SHOULD NOT
configure the RTCP bandwidth such that Td is larger than 5 seconds.
Similarly, implementations that use the RTP/AVPF profile [RFC4585] or
the RTP/SAVPF profile [RFC5124] SHOULD NOT configure T_rr_interval to
values larger than 4 seconds (the reduced limit for T_rr_interval
follows Section 6.1.3 of [I-D.ietf-avtcore-rtp-multi-stream]).
The choice of three RTCP reporting intervals as the timeout is made
following Section 6.3.5 of RFC 3550 [RFC3550]. This specifies that
participants in an RTP session will timeout and remove an RTP sender
from the list of active RTP senders if no RTP data packets have been
received from that RTP sender within the last two RTCP reporting
intervals. Using a timeout of three RTCP reporting intervals is
therefore large enough that the other participants will have timed
out the sender if a network problem stops the data packets it is
sending from reaching the receivers, even allowing for loss of some
RTCP packets.
If a sender is transmitting a large number of RTP media streams, such
that the corresponding RTCP SR or RR packets are too large to fit
into the network MTU, the receiver will generate RTCP SR or RR
packets in a round-robin manner. In this case, the sender SHOULD
treat receipt of an RTCP SR or RR packet corresponding to any SSRC it
sent on the same 5-tuple of source and destination IP address, port,
and protocol, as an indication that the receiver and return path are
working, preventing the RTCP timeout circuit breaker from triggering.
4.3. RTP/AVP Circuit Breaker #3: Congestion
If RTP data packets are being sent, and the corresponding RTCP SR or
RR packets show non-zero packet loss fraction and increasing extended
highest sequence number received, then those RTP data packets are
arriving at the receiver, but some degree of congestion is occurring.
The RTP/AVP profile [RFC3551] states that:
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If best-effort service is being used, RTP receivers SHOULD monitor
packet loss to ensure that the packet loss rate is within
acceptable parameters. Packet loss is considered acceptable if a
TCP flow across the same network path and experiencing the same
network conditions would achieve an average throughput, measured
on a reasonable time scale, that is not less than the RTP flow is
achieving. This condition can be satisfied by implementing
congestion control mechanisms to adapt the transmission rate (or
the number of layers subscribed for a layered multicast session),
or by arranging for a receiver to leave the session if the loss
rate is unacceptably high.
The comparison to TCP cannot be specified exactly, but is intended
as an "order-of-magnitude" comparison in time scale and
throughput. The time scale on which TCP throughput is measured is
the round-trip time of the connection. In essence, this
requirement states that it is not acceptable to deploy an
application (using RTP or any other transport protocol) on the
best-effort Internet which consumes bandwidth arbitrarily and does
not compete fairly with TCP within an order of magnitude.
The phase "order of magnitude" in the above means within a factor of
ten, approximately. In order to implement this, it is necessary to
estimate the throughput a TCP connection would achieve over the path.
For a long-lived TCP Reno connection, it has been shown that the TCP
throughput can be estimated using the following equation [Padhye]:
s
X = --------------------------------------------------------------
R*sqrt(2*b*p/3) + (t_RTO * (3*sqrt(3*b*p/8) * p * (1+32*p^2)))
where:
X is the transmit rate in bytes/second.
s is the packet size in bytes. If data packets vary in size, then
the average size is to be used.
R is the round trip time in seconds.
p is the loss event rate, between 0 and 1.0, of the number of loss
events as a fraction of the number of packets transmitted.
t_RTO is the TCP retransmission timeout value in seconds, generally
approximated by setting t_RTO = 4*R.
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b is the number of packets that are acknowledged by a single TCP
acknowledgement; [RFC3448] recommends the use of b=1 since many
TCP implementations do not use delayed acknowledgements.
This is the same approach to estimated TCP throughput that is used in
[RFC3448]. Under conditions of low packet loss the second term on
the denominator is small, so this formula can be approximated with
reasonable accuracy as follows [Mathis]:
s
X = -----------------
R * sqrt(2*b*p/3)
It is RECOMMENDED that this simplified throughout equation be used,
since the reduction in accuracy is small, and it is much simpler to
calculate than the full equation. Measurements have shown that the
simplified TCP throughput equation is effective as an RTP circuit
breaker for multimedia flows sent to hosts on residential networks
using ADSL and cable modem links [Singh]. The data shows that the
full TCP throughput equation tends to be more sensitive to packet
loss and triggers the RTP circuit breaker earlier than the simplified
equation. Implementations that desire this extra sensitivity MAY use
the full TCP throughput equation in the RTP circuit breaker. Initial
measurements in LTE networks have shown that the extra sensitivity is
helpful in that environment, with the full TCP throughput equation
giving a more balanced circuit breaker response than the simplified
TCP equation [Sarker]; other networks might see similar behaviour.
No matter what TCP throughput equation is chosen, two parameters need
to be estimated and reported to the sender in order to calculate the
throughput: the round trip time, R, and the loss event rate, p (the
packet size, s, is known to the sender). The round trip time can be
estimated from RTCP SR and RR packets. This is done too infrequently
for accurate statistics, but is the best that can be done with the
standard RTCP mechanisms.
Report blocks in RTCP SR or RR packets contain the packet loss
fraction, rather than the loss event rate, so p cannot be reported
(TCP typically treats the loss of multiple packets within a single
RTT as one loss event, but RTCP RR packets report the overall
fraction of packets lost, and does not report when the packet losses
occurred). Using the loss fraction in place of the loss event rate
can overestimate the loss. We believe that this overestimate will
not be significant, given that we are only interested in order of
magnitude comparison ([Floyd] section 3.2.1 shows that the difference
is small for steady-state conditions and random loss, but using the
loss fraction is more conservative in the case of bursty loss).
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The congestion circuit breaker is therefore: when a sender that is
transmitting more than one RTP packet per RTT receives an RTCP SR or
RR packet that contains a report block for an SSRC it is using, the
sender MUST record the value of the fraction lost field in the report
block and the time since the last report block was received for that
SSRC. If more than CB_INTERVAL (see Section 4.5) report blocks have
been received for that SSRC, the sender MUST calculate the average
fraction lost over the last CB_INTERVAL reporting intervals, and then
estimate the TCP throughput that would be achieved over the path
using the chosen TCP throughput equation and the measured values of
the round-trip time, R, the loss event rate, p (as approximated by
the average fraction lost), and the packet size, s. This estimate of
the TCP throughput is then compared with the actual sending rate. If
the actual sending rate is more than ten times the TCP throughput
estimate, then the congestion circuit breaker is triggered.
The average fraction lost is calculated based on the sum, over the
last CB_INTERVAL reporting intervals, of the fraction lost in each
reporting interval multiplied by the duration of the corresponding
reporting interval, divided by the total duration of the last
CB_INTERVAL reporting intervals.
The rationale for enforcing a minimum sending rate below which the
congestion circuit breaker will not trigger is to avoid spurious
circuit breaker triggers when the number of packets sent per RTCP
reporting interval is small, and hence the fraction lost samples are
subject to measurement artefacts. The one packet per RTT bound
derives from [RFC5405].
When the congestion circuit breaker is triggered, the sender SHOULD
cease transmission (see Section 4.6). However, if the sender is able
to reduce its sending rate by a factor of (approximately) ten, then
it MAY first reduce its sending rate by this factor (or some larger
amount) to see if that resolves the congestion. If the sending rate
is reduced in this way and the congestion circuit breaker triggers
again after the next CB_INTERVAL RTCP reporting intervals, the sender
MUST then cease transmission. An example of such a rate reduction
might be a video conferencing system that backs off to sending audio
only, before completely dropping the call. If such a reduction in
sending rate resolves the congestion problem, the sender MAY
gradually increase the rate at which it sends data after a reasonable
amount of time has passed, provided it takes care not to cause the
problem to recur ("reasonable" is intentionally not defined here).
The RTCP reporting interval of the media sender does not affect how
quickly congestion circuit breaker can trigger. The timing is based
on the RTCP reporting interval of the receiver that generates the SR/
RR packets from which the loss rate and RTT estimate are derived
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(note that RTCP requires all participants in a session to have
similar reporting intervals, else the participant timeout rules in
[RFC3550] will not work, so this interval is likely similar to that
of the sender). If the incoming RTCP SR or RR packets are using a
reduced minimum RTCP reporting interval (as specified in Section 6.2
of RFC 3550 [RFC3550] or the RTP/AVPF profile [RFC4585]), then that
reduced RTCP reporting interval is used when determining if the
circuit breaker is triggered.
As in Section 4.1 and Section 4.2, we use CB_INTERVAL reporting
intervals to avoid triggering the circuit breaker on transient
failures. This circuit breaker is a worst-case condition, and
congestion control needs to be performed to keep well within this
bound. It is expected that the circuit breaker will only be
triggered if the usual congestion control fails for some reason.
If there are more media streams that can be reported in a single RTCP
SR or RR packet, or if the size of a complete RTCP SR or RR packet
exceeds the network MTU, then the receiver will report on a subset of
sources in each reporting interval, with the subsets selected round-
robin across multiple intervals so that all sources are eventually
reported [RFC3550]. When generating such round-robin RTCP reports,
priority SHOULD be given to reports on sources that have high packet
loss rates, to ensure that senders are aware of network congestion
they are causing (this is an update to [RFC3550]).
4.4. RTP/AVP Circuit Breaker #4: Media Usability
Applications that use RTP are generally tolerant to some amount of
packet loss. How much packet loss can be tolerated will depend on
the application, media codec, and the amount of error correction and
packet loss concealment that is applied. There is an upper bound on
the amount of loss can be corrected, however, beyond which the media
becomes unusable. Similarly, many applications have some upper bound
on the media capture to play-out latency that can be tolerated before
the application becomes unusable. The latency bound will depend on
the application, but typical values can range from the order of a few
hundred milliseconds for voice telephony and interactive conferencing
applications, up to several seconds for some video-on-demand systems.
As a final circuit breaker, RTP senders SHOULD monitor the reported
packet loss and delay to estimate whether the media is likely to be
suitable for the intended purpose. If the packet loss rate and/or
latency is such that the media has become unusable, and has remained
unusable for a significant time period, then the application SHOULD
cease transmission. Similarly, receivers SHOULD monitor the quality
of the media they receive, and if the quality is unusable for a
significant time period, they SHOULD terminate the session. This
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memo intentionally does not define a bound on the packet loss rate or
latency that will result in unusable media, nor does it specify what
time period is deemed significant, as these are highly application
dependent.
Sending media that suffers from such high packet loss or latency that
it is unusable at the receiver is both wasteful of resources, and of
no benefit to the user of the application. It also is highly likely
to be congesting the network, and disrupting other applications. As
such, the congestion circuit breaker will almost certainly trigger to
stop flows where the media would be unusable due to high packet loss
or latency. However, in pathological scenarios where the congestion
circuit breaker does not stop the flow, it is desirable that the RTP
application cease sending useless traffic. The role of the media
usability circuit breaker is to protect the network in such cases.
4.5. Choice of Circuit Breaker Interval
The CB_INTERVAL parameter determines the number of consecutive RTCP
reporting intervals that need to suffer congestion before the media
timeout circuit breaker (see Section 4.1) or the congestion circuit
breaker (see Section 4.3) triggers. It determines the sensitivity
and responsiveness of these circuit breakers.
The CB_INTERVAL parameter is set to min(floor(3+(2.5/Td)), 30) RTCP
reporting intervals, where Td is the deterministic calculated RTCP
interval described in section 6.3.1 of [RFC3550]. This expression
gives an CB_INTERVAL that varies as follows:
Td | CB_INTERVAL | Time to trigger
--------------+------------------------------+-----------------
0.016 seconds | 30 RTCP reporting intervals | 0.48 seconds
0.033 seconds | 30 RTCP reporting intervals | 0.99 seconds
0.100 seconds | 28 RTCP reporting intervals | 2.80 seconds
0.500 seconds | 8 RTCP reporting intervals | 4.00 seconds
1.000 seconds | 5 RTCP reporting intervals | 5.00 seconds
2.000 seconds | 4 RTCP reporting intervals | 8.00 seconds
5.000 seconds | 3 RTCP reporting intervals | 15.00 seconds
10.000 seconds | 3 RTCP reporting intervals | 30.00 seconds
If the RTP/AVPF profile [RFC4585] or the RTP/SAVPF [RFC5124] is used,
and the T_rr_interval parameter is used to reduce the frequency of
regular RTCP reports, then the value Td in the above expression for
the CB_INTERVAL parameter MUST be replaced by max(T_rr_interval, Td).
The CB_INTERVAL parameter is calculated on joining the session, and
recalculated on receipt of each RTCP packet, after checking whether
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the media timeout circuit breaker or the congestion circuit breaker
has been triggered.
To ensure a timely response to persistent congestion, implementations
SHOULD NOT configure the RTCP bandwidth such that Td is larger than 5
seconds. Similarly, implementations that use the RTP/AVPF profile
[RFC4585] or the RTP/SAVPF profile [RFC5124] SHOULD NOT configure
T_rr_interval to values larger than 4 seconds (the reduced limit for
T_rr_interval follows Section 6.1.3 of
[I-D.ietf-avtcore-rtp-multi-stream]).
Rationale: If the CB_INTERVAL was always set to the same number of
RTCP reporting intervals, this would cause higher rate RTP sessions
to trigger the RTP circuit breaker after a shorter time interval than
lower rate sessions, because the RTCP reporting interval scales based
on the RTP session bandwidth. This is felt to penalise high rate RTP
sessions too aggressively. Conversely, scaling CB_INTERVAL according
to the inverse of the RTCP reporting interval, so the RTP circuit
breaker triggers after a constant time interval, doesn't sufficiently
protect the network from congestion caused by high-rate flows. The
chosen expression for CB_INTERVAL seeks a balance between these two
extremes. It causes higher rate RTP sessions subject to persistent
congestion to trigger the RTP circuit breaker after a shorter time
interval than do lower rate RTP sessions, while also making the RTP
circuit breaker for such sessions less sensitive by requiring the
congestion to persist for longer numbers of RTCP reporting intervals.
4.6. 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.
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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 and the RTP/AVPF and RTP/SAVPF Profiles
Use of the Extended RTP Profile for RTCP-based Feedback (RTP/AVPF)
[RFC4585] allows receivers to send early RTCP reports in some cases,
to inform the sender about particular events in the media stream.
There are several use cases for such early RTCP reports, including
providing rapid feedback to a sender about the onset of congestion.
The RTP/SAVPF Profile [RFC5124] is a secure variant of the RTP/AVPF
profile, that is treated the same in the context of the RTP circuit
breaker. These feedback profiles are often used with non-compound
RTCP reports [RFC5506] to reduce the reporting overhead.
Receiving rapid feedback about congestion events potentially allows
congestion control algorithms to be more responsive, and to better
adapt the media transmission to the limitations of the network. It
is expected that many RTP congestion control algorithms will adopt
the RTP/AVPF profile or the RTP/SAVPF profile for this reason,
defining new transport layer feedback reports that suit their
requirements. Since these reports are not yet defined, and likely
very specific to the details of the congestion control algorithm
chosen, they cannot be used as part of the generic RTP circuit
breaker.
Reduced-size RTCP reports sent under the RTP/AVPF early feedback
rules that do not contain an RTCP SR or RR packet MUST be ignored by
the congestion circuit breaker (they do not contain the information
needed by the congestion circuit breaker algorithm), but MUST be
counted as received packets for the RTCP timeout circuit breaker.
Reduced-size RTCP reports sent under the RTP/AVPF early feedback
rules that contain RTCP SR or RR packets MUST be processed by the
congestion circuit breaker as if they were sent as regular RTCP
reports, and counted towards the circuit breaker conditions specified
in Section 4 of this memo. This will potentially make the RTP
circuit breaker trigger earlier than it would if the RTP/AVPF profile
was not used.
When using ECN with RTP (see Section 8), early RTCP feedback packets
can contain ECN feedback reports. The count of ECN-CE marked packets
contained in those ECN feedback reports is counted towards the number
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of lost packets reported if the ECN Feedback Report report is sent in
an compound RTCP packet along with an RTCP SR/RR report packet.
Reports of ECN-CE packets sent as reduced-size RTCP ECN feedback
packets without an RTCP SR/RR packet MUST be ignored.
These rules are intended to allow the use of low-overhead RTP/AVPF
feedback for generic NACK messages without triggering the RTP circuit
breaker. This is expected to make such feedback suitable for RTP
congestion control algorithms that need to quickly report loss events
in between regular RTCP reports. The reaction to reduced-size RTCP
SR/RR packets is to allow such algorithms to send feedback that can
trigger the circuit breaker, when desired.
The RTP/AVPF and RTP/SAVPF profiles include the T_rr_interval
parameter that can be used to adjust the regular RTCP reporting
interval. The use of the T_rr_interval parameter changes the
behaviour of the RTP circuit breaker, as described in Section 4.
6. Impact of RTCP Extended Reports (XR)
RTCP Extended Report (XR) blocks provide additional reception quality
metrics, but do not change the RTCP timing rules. Some of the RTCP
XR blocks provide information that might be useful for congestion
control purposes, others provided non-congestion-related metrics.
With the exception of RTCP XR ECN Summary Reports (see Section 8),
the presence of RTCP XR blocks in a compound RTCP packet does not
affect the RTP circuit breaker algorithm. For consistency and ease
of implementation, only the reception report blocks contained in RTCP
SR packets, RTCP RR packets, or RTCP XR ECN Summary Report packets,
are used by the RTP circuit breaker algorithm.
7. Impact of RTCP Reporting Groups
An optimisation for grouping RTCP reception statistics and other
feedback in RTP sessions with large numbers of participants is given
in [I-D.ietf-avtcore-rtp-multi-stream-optimisation]. This allows one
SSRC to act as a representative that sends reports on behalf of other
SSRCs that are co-located in the same endpoint and see identical
reception quality. When running the circuit breaker algorithms, an
endpoint MUST treat a reception report from the representative of the
reporting group as if a reception report was received from all
members of that group.
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8. Impact of Explicit Congestion Notification (ECN)
The use of ECN for RTP flows does not affect the media timeout RTP
circuit breaker (Section 4.1) or the RTCP timeout circuit breaker
(Section 4.2), since these are both connectivity checks that simply
determinate if any packets are being received.
ECN-CE marked packets SHOULD be treated as if it were lost for the
purposes of congestion control, when determining the optimal media
sending rate for an RTP flow. If an RTP sender has negotiated ECN
support for an RTP session, and has successfully initiated ECN use on
the path to the receiver [RFC6679], then ECN-CE marked packets SHOULD
be treated as if they were lost when calculating if the congestion-
based RTP circuit breaker (Section 4.3) has been met. The count of
ECN-CE marked RTP packets is returned in RTCP XR ECN summary report
packets if support for ECN has been initiated for an RTP session.
9. Impact of Bundled Media and Layered Coding
The RTP circuit breaker operates on a per-RTP session basis. An RTP
sender that participates in several RTP sessions MUST treat each RTP
session independently with regards to the RTP circuit breaker.
An RTP sender can generate several media streams within a single RTP
session, with each stream using a different SSRC. This can happen if
bundled media are in use, when using simulcast, or when using layered
media coding. By default, each SSRC will be treated independently by
the RTP circuit breaker. However, the sender MAY choose to treat the
flows (or a subset thereof) as a group, such that a circuit breaker
trigger for one flow applies to the group of flows as a whole, and
either causes the entire group to cease transmission, or the sending
rate of the group to reduce by a factor of ten, depending on the RTP
circuit breaker triggered. Grouping flows in this way is expected to
be especially useful for layered flows sent using multiple SSRCs, as
it allows the layered flow to react as a whole, ceasing transmission
on the enhancement layers first to reduce sending rate if necessary,
rather than treating each layer independently.
10. Security Considerations
The security considerations of [RFC3550] apply.
If the RTP/AVPF profile is used to provide rapid RTCP feedback, the
security considerations of [RFC4585] apply. If ECN feedback for RTP
over UDP/IP is used, the security considerations of [RFC6679] apply.
If non-authenticated RTCP reports are used, an on-path attacker can
trivially generate fake RTCP packets that indicate high packet loss
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rates, causing the circuit breaker to trigger and disrupting an RTP
session. This is somewhat more difficult for an off-path attacker,
due to the need to guess the randomly chosen RTP SSRC value and the
RTP sequence number. This attack can be avoided if RTCP packets are
authenticated; authentication options are discussed in [RFC7201].
Timely operation of the RTP circuit breaker depends on the choice of
RTCP reporting interval. If the receiver has a reporting interval
that is overly long, then the responsiveness of the circuit breaker
decreases. In the limit, the RTP circuit breaker can be disabled for
all practical purposes by configuring an RTCP reporting interval that
is many minutes duration. This issue is not specific to the circuit
breaker: long RTCP reporting intervals also prevent reception quality
reports, feedback messages, codec control messages, etc., from being
used. Implementations are expected to impose an upper limit on the
RTCP reporting interval they are willing to negotiate (based on the
session bandwidth and RTCP bandwidth fraction) when using the RTP
circuit breaker, as discussed in Section 4.5.
11. IANA Considerations
There are no actions for IANA.
12. Acknowledgements
The authors would like to thank Bernard Aboba, Harald Alvestrand,
Gorry Fairhurst, Nazila Fough, Kevin Gross, Cullen Jennings, Randell
Jesup, Jonathan Lennox, Matt Mathis, Stephen McQuistin, Eric
Rescorla, Abheek Saha, Fabio Verdicchio, and Magnus Westerlund for
their valuable feedback.
13. References
13.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3448] Handley, M., Floyd, S., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification", RFC
3448, January 2003.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
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[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control
Protocol Extended Reports (RTCP XR)", RFC 3611, November
2003.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
2006.
13.2. Informative References
[Floyd] Floyd, S., Handley, M., Padhye, J., and J. Widmer,
"Equation-Based Congestion Control for Unicast
Applications", Proceedings of the ACM SIGCOMM conference,
2000, DOI 10.1145/347059.347397, August 2000.
[I-D.ietf-avtcore-rtp-multi-stream-optimisation]
Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session:
Grouping RTCP Reception Statistics and Other Feedback",
draft-ietf-avtcore-rtp-multi-stream-optimisation-05 (work
in progress), February 2015.
[I-D.ietf-avtcore-rtp-multi-stream]
Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session",
draft-ietf-avtcore-rtp-multi-stream-07 (work in progress),
March 2015.
[I-D.ietf-tsvwg-circuit-breaker]
Fairhurst, G., "Network Transport Circuit Breakers",
draft-ietf-tsvwg-circuit-breaker-00 (work in progress),
September 2014.
[Mathis] Mathis, M., Semke, J., Mahdavi, J., and T. Ott, "The
macroscopic behavior of the TCP congestion avoidance
algorithm", ACM SIGCOMM Computer Communication Review
27(3), DOI 10.1145/263932.264023, July 1997.
[Padhye] Padhye, J., Firoiu, V., Towsley, D., and J. Kurose,
"Modeling TCP Throughput: A Simple Model and its Empirical
Validation", Proceedings of the ACM SIGCOMM conference,
1998, DOI 10.1145/285237.285291, August 1998.
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[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP", RFC
3168, September 2001.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, February 2008.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008.
[RFC5405] Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
for Application Designers", BCP 145, RFC 5405, November
2008.
[RFC5450] Singer, D. and H. Desineni, "Transmission Time Offsets in
RTP Streams", RFC 5450, March 2009.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009.
[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, September 2009.
[RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
Flows", RFC 6051, November 2010.
[RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
and K. Carlberg, "Explicit Congestion Notification (ECN)
for RTP over UDP", RFC 6679, August 2012.
[RFC6798] Clark, A. and Q. Wu, "RTP Control Protocol (RTCP) Extended
Report (XR) Block for Packet Delay Variation Metric
Reporting", RFC 6798, November 2012.
[RFC6843] Clark, A., Gross, K., and Q. Wu, "RTP Control Protocol
(RTCP) Extended Report (XR) Block for Delay Metric
Reporting", RFC 6843, January 2013.
[RFC6958] Clark, A., Zhang, S., Zhao, J., and Q. Wu, "RTP Control
Protocol (RTCP) Extended Report (XR) Block for Burst/Gap
Loss Metric Reporting", RFC 6958, May 2013.
[RFC7002] Clark, A., Zorn, G., and Q. Wu, "RTP Control Protocol
(RTCP) Extended Report (XR) Block for Discard Count Metric
Reporting", RFC 7002, September 2013.
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[RFC7003] Clark, A., Huang, R., and Q. Wu, "RTP Control Protocol
(RTCP) Extended Report (XR) Block for Burst/Gap Discard
Metric Reporting", RFC 7003, September 2013.
[RFC7097] Ott, J., Singh, V., and I. Curcio, "RTP Control Protocol
(RTCP) Extended Report (XR) for RLE of Discarded Packets",
RFC 7097, January 2014.
[RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP
Sessions", RFC 7201, April 2014.
[Sarker] Sarker, Z., Singh, V., and C.S. Perkins, "An Evaluation of
RTP Circuit Breaker Performance on LTE Networks",
Proceedings of the IEEE Infocom workshop on Communication
and Networking Techniques for Contemporary Video, 2014,
April 2014.
[Singh] Singh, V., McQuistin, S., Ellis, M., and C.S. Perkins,
"Circuit Breakers for Multimedia Congestion Control",
Proceedings of the International Packet Video Workshop,
2013, DOI 10.1109/PV.2013.6691439, December 2013.
Authors' Addresses
Colin Perkins
University of Glasgow
School of Computing Science
Glasgow G12 8QQ
United Kingdom
Email: csp@csperkins.org
Varun Singh
Aalto University
School of Electrical Engineering
Otakaari 5 A
Espoo, FIN 02150
Finland
Email: varun@comnet.tkk.fi
URI: http://www.netlab.tkk.fi/~varun/
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