Internet Engineering Task Force M. Handley
INTERNET-DRAFT University College London
Intended status: Proposed Standard S. Floyd
Expires: September 2007 ICIR
J. Padhye
Microsoft
J. Widmer
University of Mannheim
4 March 2007
TCP Friendly Rate Control (TFRC): Protocol Specification
draft-ietf-dccp-rfc3448bis-01.txt
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Abstract
This document specifies TCP-Friendly Rate Control (TFRC). TFRC is a
congestion control mechanism for unicast flows operating in a best-
effort Internet environment. It is reasonably fair when competing
for bandwidth with TCP flows, but has a much lower variation of
throughput over time compared with TCP, making it more suitable for
applications such as streaming media where a relatively smooth
sending rate is of importance.
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Table of Contents
1. Introduction ...................................................8
2. Conventions ....................................................9
3. Protocol Mechanism .............................................9
3.1. TCP Throughput Equation ..................................10
3.2. Packet Contents ..........................................11
3.2.1. Data Packets ......................................12
3.2.2. Feedback Packets ..................................12
4. Data Sender Protocol ..........................................13
4.1. Measuring the Segment Size ...............................13
4.2. Sender Initialization ....................................14
4.3. Sender behavior when a feedback packet is received .......15
4.4. Expiration of nofeedback timer ...........................16
4.5. Sending a packet after an idle or data-limited period ....17
4.6. Preventing Oscillations ..................................17
4.7. Scheduling of Packet Transmissions .......................18
5. Calculation of the Loss Event Rate (p) ........................19
5.1. Detection of Lost or Marked Packets ......................19
5.2. Translation from Loss History to Loss Events .............20
5.3. Inter-loss Event Interval ................................21
5.4. Average Loss Interval ....................................22
5.5. History Discounting ......................................23
6. Data Receiver Protocol ........................................25
6.1. Receiver behavior when a data packet is received .........26
6.2. Expiration of feedback timer .............................26
6.3. Receiver initialization ..................................27
6.3.1. Initializing the Loss History after the First Loss
Event ....................................................28
7. Sender-based Variants .........................................29
8. Implementation Issues .........................................30
9. Changes from RFC 3448 .........................................31
10. Security Considerations ......................................32
11. IANA Considerations ..........................................32
12. Acknowledgments ..............................................33
13. Terminology ..................................................33
14. Normative References .........................................35
15. Informational References .....................................35
16. Authors' Addresses ...........................................36
Full Copyright Statement .........................................37
Intellectual Property ............................................38
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NOTE TO RFC EDITOR: PLEASE DELETE THIS NOTE UPON PUBLICATION.
Changes from draft-ietf-dccp-rfc3448bis-00.txt:
* When initializing the loss history after the first
data packet sent is lost or ECN-marked, TFRC uses
a minimum receive rate of 0.5 packets per second.
* For initializing the estimated packet drop rate
for the first loss interval when coming out of slow-start,
it is ok to use the maximum receive rate so far, not just
the receive rate in the last round-trip time.
Feedback from Ladan Gharai.
* General feedback from Gorry Fairhurst:
- Added a reference for TFRC-SP.
- Clarified that R_m is sender's estimate of RTT, as reported
in Section 3.2.1.
- Added a definition of terms.
- Added a discussion of why the initial value of the nofeedback
timer is two seconds, instead of three seconds for the
recommended initial value for TCP's retransmit timer.
* General feedback from Arjuna Sathiaseelan:
- Added more details about sending multiple feedback
packets per RTT.
- Added change to Section 4.3 to use the first feedback
packet, or the first feedback packet after a
nofeedback timer during slow-start, *if min_rate > X*.
* General feedback from Gerrit Renker:
- Changed "delta" to "t_delta".
- Changed X_calc to X_Bps, clarified X.
- Clarified send times in Section 4.7.
- Changed so that tld can be initialized to either 0 or -1.
- Fixed Section 5.5 to say that the most recent lost
interval has weight 1/(0.75*n) *when there have been
at least eight loss intervals*.
- Clarified introduction about fixed-size and variable-size
packets.
* Added more about sender-based variants.
Feedback from Guillaume Jourjon.
* Corrected that the loss interval I_0 includes all transmitted
packets, including lost and marked packets (as defined in Section
5.3 in the general definition.) Email from Eddie Kohler and
Gerrit Renker.
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* Open issue: Feedback from Ian about problems being limited by
X_recv after a loss event. There might not be an easy answer.
* Related open issue: Add Faster Restart to RFC3448bis? Or not?
From Ian McDonald.
* Open issue: Adopt something like DCCP's Receive Rate Length,
instead of ignoring one feedback packet? From Eddie Kohler.
* Open issue: Add possible mechanisms for limited the maximum
burst size? Using a token bucket size based on the
current rate? Or not? Email from Eddie Kohler and Gerrit
Renker.
* Related open issue: To deal with idle periods and the like,
in Section 4.7 say that t_i := max(t_i, t_now - RTT/2), to
limit bursts to RTT/2 packets? Has anyone implemented this?
Email from Eddie Kohler and Ian McDonald.
* Not done: I didn't add a minimum value for the nofeedback
timer. (Why would a nofeedback timer need to be bigger
than max(4*R, 2*s/X)? Email discussing pros and cons from
Arjuna.
* Not addressed yet: Email thread on "RFC 3448, 4.4: Modifying
X_recv if p = 0 at the time of last feedback".
* Todo: Update Section 9 on "Changes from RFC 3448" with
changes since draft-floyd-rfc3448bis-00.txt.
Changes from draft-floyd-rfc3448bis-00.txt:
* Name change to draft-ietf-dccp-rfc3448bis-00.txt.
* Specified the receiver's initialization of the feedback timer
when the first data packet doesn't have an estimate of the
RTT. From feedback from Dado Colussi.
* Added the procedure for sending receiver
feedback packets when a coarse-grained
timestamp is used. From RFC 4243.
Changes from RFC 3448:
* Incorporated changes in the RFC 3448 errata:
- "If the sender does not receive a feedback report for
four round trip times, it cuts its sending rate in half."
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("Two" changed to "four", for consistency with the rest
of the document. Reported by Joerg Widmer).
- "If the nofeedback timer expires when the sender does not
yet have an RTT sample, and has not yet received any
feedback from the receiver, or when p == 0,..."
(Added "or when p == 0,", reported by Wim Heirman).
- In Section 5.5, changed:
for (i = 1 to n) { DF_i = 1; }
to:
for (i = 0 to n) { DF_i = 1; }
Reported by Michele R.
* Changed RFC 3448 to correspond to the larger initial windows
specified in RFC 3390. This includes the following:
- Incorporated Section 5.1 from [RFC4342], saying that
when reducing the sending rate after an idle period, don't
reduce the sending rate below the initial sending rate.
- Change for a datalimited sender:
When the sender has been datalimited, the sender doesn't
let the receive rate limit it to a sending rate less than
the initial rate.
- Small change to slow-start:
Changed so that for the first feedback packet received,
or for the first feedback packet received after an idle
period, the receive rate is not used to limit the
sending rate. This is because the receiver might not yet
have seen an entire window of data.
* Clarified how the average loss interval is calculated when
the receiver has not yet seen eight loss intervals.
* Discussed more about estimating the average segment size:
- For initializing the loss history after the first loss event,
either the receiver knows the sender's value for s, or
the receiver uses the throughput equation for X_pps and does
not need to know an estimate for s.
- Added a discussion about estimating the average segment size
s in Section 4.1 on "Measuring the Segment Size".
- Changed "packet size" to "segment size".
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END OF NOTE TO RFC EDITOR.
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1. Introduction
This document specifies TCP-Friendly Rate Control (TFRC). TFRC is a
congestion control mechanism designed for unicast flows operating in
an Internet environment and competing with TCP traffic [FHPW00].
Instead of specifying a complete protocol, this document simply
specifies a congestion control mechanism that could be used in a
transport protocol such as DCCP (Datagram Congestion Control
Protocol) [RFC4340], in an application incorporating end-to-end
congestion control at the application level, or in the context of
endpoint congestion management [BRS99]. This document does not
discuss packet formats or reliability. Implementation-related
issues are discussed only briefly, in Section 8.
TFRC is designed to be reasonably fair when competing for bandwidth
with TCP flows, where a flow is "reasonably fair" if its sending
rate is generally within a factor of two of the sending rate of a
TCP flow under the same conditions. However, TFRC has a much lower
variation of throughput over time compared with TCP, which makes it
more suitable for applications such as telephony or streaming media
where a relatively smooth sending rate is of importance.
The penalty of having smoother throughput than TCP while competing
fairly for bandwidth is that TFRC responds slower than TCP to
changes in available bandwidth. Thus TFRC should only be used when
the application has a requirement for smooth throughput, in
particular, avoiding TCP's halving of the sending rate in response
to a single packet drop. For applications that simply need to
transfer as much data as possible in as short a time as possible we
recommend using TCP, or if reliability is not required, using an
Additive-Increase, Multiplicative-Decrease (AIMD) congestion control
scheme with similar parameters to those used by TCP.
TFRC is designed for best performance with applications that use a
fixed segment size, and vary their sending rate in packets per
second in response to congestion. TFRC can also be used, perhaps
with less optimal performance, with applications that don't have a
fixed segment size, but where the segment size varies according to
the needs of the application (e.g., video applications).
Some applications (e.g., some audio applications) require a fixed
interval of time between packets and vary their segment size instead
of their packet rate in response to congestion. The congestion
control mechanism in this document is not designed for those
applications; TFRC-SP (Small-Packet TFRC) is a variant of TFRC for
applications that have a fixed sending rate in packets per second
but either use small packets, or vary their packet size in response
to congestion. TFRC-SP will be specified in a later document [TFRC-
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SP].
This document specifies TFRC as a receiver-based mechanism, with the
calculation of the congestion control information (i.e., the loss
event rate) in the data receiver rather in the data sender. This is
well-suited to an application where the sender is a large server
handling many concurrent connections, and the receiver has more
memory and CPU cycles available for computation. In addition, a
receiver-based mechanism is more suitable as a building block for
multicast congestion control. However, it is also possible to
implement TFRC in sender-based variants, as allowed in DCCP's
Congestion Control ID 3 (CCID 3) [RFC4342].
2. Conventions
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 [RFC2119].
Appendix A gives a list of terms used in this document.
3. Protocol Mechanism
For its congestion control mechanism, TFRC directly uses a
throughput equation for the allowed sending rate as a function of
the loss event rate and round-trip time. In order to compete fairly
with TCP, TFRC uses the TCP throughput equation, which roughly
describes TCP's sending rate as a function of the loss event rate,
round-trip time, and segment size. We define a loss event as one or
more lost or marked packets from a window of data, where a marked
packet refers to a congestion indication from Explicit Congestion
Notification (ECN) [RFC3168].
Generally speaking, TFRC's congestion control mechanism works as
follows:
o The receiver measures the loss event rate and feeds this
information back to the sender.
o The sender also uses these feedback messages to measure the
round-trip time (RTT).
o The loss event rate and RTT are then fed into TFRC's throughput
equation, giving the acceptable transmit rate.
o The sender then adjusts its transmit rate to match the
calculated rate.
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The dynamics of TFRC are sensitive to how the measurements are
performed and applied. We recommend specific mechanisms below to
perform and apply these measurements. Other mechanisms are
possible, but it is important to understand how the interactions
between mechanisms affect the dynamics of TFRC.
3.1. TCP Throughput Equation
Any realistic equation giving TCP throughput as a function of loss
event rate and RTT should be suitable for use in TFRC. However, we
note that the TCP throughput equation used must reflect TCP's
retransmit timeout behavior, as this dominates TCP throughput at
higher loss rates. We also note that the assumptions implicit in
the throughput equation about the loss event rate parameter have to
be a reasonable match to how the loss rate or loss event rate is
actually measured. While this match is not perfect for the
throughput equation and loss rate measurement mechanisms given
below, in practice the assumptions turn out to be close enough.
The throughput equation we currently recommend for TFRC is a
slightly simplified version of the throughput equation for Reno TCP
from [PFTK98]. Ideally we'd prefer a throughput equation based on
SACK TCP, but no one has yet derived the throughput equation for
SACK TCP, and from both simulations and experiments, the differences
between the two equations are relatively minor.
The throughput equation is:
s
X_Bps = ----------------------------------------------------------
R*sqrt(2*b*p/3) + (t_RTO * (3*sqrt(3*b*p/8)*p*(1+32*p^2)))
Where:
X_Bps is the transmit rate in bytes/second.
s is the segment size in bytes.
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.
b is the number of packets acknowledged by a single TCP
acknowledgement.
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We further simplify this by setting t_RTO = 4*R. A more accurate
calculation of t_RTO is possible, but experiments with the current
setting have resulted in reasonable fairness with existing TCP
implementations [W00]. Another possibility would be to set t_RTO =
max(4R, one second), to match the recommended minimum of one second
on the RTO [RFC2988].
Many current TCP connections use delayed acknowledgements, sending
an acknowledgement for every two data packets received, and thus
have a sending rate modeled by b = 2. However, TCP is also allowed
to send an acknowledgement for every data packet, and this would be
modeled by b = 1. Because many TCP implementations do not use
delayed acknowledgements, we recommend b = 1.
In future, different TCP equations may be substituted for this
equation. The requirement is that the throughput equation be a
reasonable approximation of the sending rate of TCP for conformant
TCP congestion control.
The throughput equation can also be expressed as
X_Bps = X_pps * s ,
with X_pps, the sending rate in packets per second, given as
1
X_pps = --------------------------------------------------------
R*sqrt(2*b*p/3) + (t_RTO*(3*sqrt(3*b*p/8)*p*(1+32*p^2)))
The parameters s (segment size), p (loss event rate) and R (RTT)
need to be measured or calculated by a TFRC implementation. The
measurement of s is specified in Section 4.1, measurement of R is
specified in Section 4.3, and measurement of p is specified in
Section 5. In the rest of this document all data rates are measured
in bytes/second.
3.2. Packet Contents
Before specifying the sender and receiver functionality, we describe
the contents of the data packets sent by the sender and feedback
packets sent by the receiver. As TFRC will be used along with a
transport protocol, we do not specify packet formats, as these
depend on the details of the transport protocol used.
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3.2.1. Data Packets
Each data packet sent by the data sender contains the following
information:
o A sequence number. This number is incremented by one for each
data packet transmitted. The field must be sufficiently large
that it does not wrap causing two different packets with the
same sequence number to be in the receiver's recent packet
history at the same time.
o A timestamp indicating when the packet is sent. We denote by
ts_i the timestamp of the packet with sequence number i. The
resolution of the timestamp should typically be measured in
milliseconds.
This timestamp is used by the receiver to determine which losses
belong to the same loss event. The timestamp is also echoed by
the receiver to enable the sender to estimate the round-trip
time, for senders that do not save timestamps of transmitted
data packets.
We note that as an alternative to a timestamp incremented in
milliseconds, a "timestamp" that increments every quarter of a
round-trip time would be sufficient for determining when losses
belong to the same loss event, in the context of a protocol
where this is understood by both sender and receiver, and where
the sender saves the timestamps of transmitted data packets.
o The sender's current estimate of the round trip time. The
estimate reported in packet i is denoted by R_i. The round-trip
time estimate is used by the receiver, along with the timestamp,
to determine when multiple losses belong to the same loss event.
The round-trip time estimate is also used by the receiver to
determine the interval to use for calculating the receive rate,
and to determine when to send feedback packets.
If the sender sends a coarse-grained "timestamp" that increments
every quarter of a round-trip time, as discussed above, then the
sender does not need to send its current estimate of the round
trip time.
3.2.2. Feedback Packets
Each feedback packet sent by the data receiver contains the
following information:
o The timestamp of the last data packet received. We denote this
by t_recvdata. If the last packet received at the receiver has
sequence number i, then t_recvdata = ts_i.
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This timestamp is used by the sender to estimate the round-trip
time, and is only needed if the sender does not save timestamps
of transmitted data packets.
o The amount of time elapsed between the receipt of the last data
packet at the receiver, and the generation of this feedback
report. We denote this by t_delay.
o The rate at which the receiver estimates that data was received
since the last feedback report was sent. We denote this by
X_recv.
o The receiver's current estimate of the loss event rate, p.
4. Data Sender Protocol
The data sender sends a stream of data packets to the data receiver
at a controlled rate. When a feedback packet is received from the
data receiver, the data sender changes its sending rate, based on
the information contained in the feedback report. If the sender does
not receive a feedback report for four round trip times, it cuts its
sending rate in half. This is achieved by means of a timer called
the nofeedback timer.
We specify the sender-side protocol in the following steps:
o Measurement of the mean segment size being sent.
o The sender behavior when a feedback packet is received.
o The sender behavior when the nofeedback timer expires.
o Oscillation prevention (optional)
o Scheduling of transmission on non-realtime operating systems.
4.1. Measuring the Segment Size
The parameter s (segment size) is normally known to an application.
This may not be so in two cases:
o (1) The segment size naturally varies depending on the data. In
this case, although the segment size varies, that variation is
not coupled to the transmit rate. The TFRC sender can either
compute the average segment size or use the maximum segment size
for the segment size s.
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o (2) The application needs to change the segment size rather than
the number of segments per second to perform congestion control.
This would normally be the case with packet audio applications
where a fixed interval of time needs to be represented by each
packet. Such applications need to have a completely different
way of measuring parameters.
For the first class of applications where the segment size varies
depending on the data, the sender MAY estimate the segment size s as
the average segment size over the last four loss intervals. The
sender MAY also estimate the average segment size over longer time
intervals, if so desired. The TFRC sender uses the segment size s
in the throughput equation, in the setting of the maximum receive
rate and the minimum sending rate, and in the setting of the
nofeedback timer.
The TFRC receiver may use the average segment size s in initializing
the loss history after the first loss event, but Section 6.3.1 also
gives an alternate procedure that does not use the average segment
size s.
The second class of applications are discussed separately in a
separate document on TFRC-SP. For the remainder of this section we
assume the sender can estimate the segment size, and that congestion
control is performed by adjusting the number of packets sent per
second.
4.2. Sender Initialization
The initial values for X (the allowed sending rate in bytes per
second) and tld (the Time Last Doubled during slow-start) are
undefined until they are set as described below. If the sender is
ready to send data when it does not yet have a round trip sample,
the value of X is set to 1 MSS/second (for MSS the Maximum Segment
Size), the nofeedback timer is set to expire after two seconds, and
tld is set either to 0 or to -1. Upon receiving a round trip time
measurement (e.g., after the first feedback packet), tld is set to
the current time, and the allowed transmit rate X is set to
W_init/R, for W_init below from [RFC3390]:
W_init = min(4*MSS, max(2*MSS, 4380)).
For responding to the initial feedback packet, this replaces step
(4) of Section 4.3 below.
If the sender does have a round trip sample when it is ready to
first send data (e.g., from the SYN exchange or from a previous
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connection [RFC2140]), the initial transmit rate X is set to
W_init/R, and tld is set to the current time.
Why is the initial value of TFRC's nofeedback timer set to two
seconds, instead of the recommended initial value of three seconds
for TCP's retransmit timer, from [RFC2988]? There isn't any
particular reason why TFRC's nofeedback timer should have the same
initial value as TCP's retransmit timer. TCP's retransmit timer is
used not only to reduce the sending rate in response to congestion,
but also to retransit a packet that is assumed to have been dropped
in the network. In contrast, TFRC's nofeedback timer is only used
to reduce the allowed sending rate, not to trigger the sending of a
new packet. As a result, there is no danger to the network for the
initial value of TFRC's nofeedback timer to be smaller than the
recommended initial value for TCP's retransmit timer.
4.3. Sender behavior when a feedback packet is received
The sender knows its current allowed sending rate, X, and maintains
an estimate of the current round trip time, R, and an estimate of
the timeout interval, t_RTO.
When a feedback packet is received by the sender at time t_now, the
following actions should be performed:
1) Calculate a new round trip sample.
R_sample = (t_now - t_recvdata) - t_delay.
2) Update the round trip time estimate:
If no feedback has been received before
R = R_sample;
Else
R = q*R + (1-q)*R_sample;
TFRC is not sensitive to the precise value for the filter
constant q, but we recommend a default value of 0.9.
3) Update the timeout interval:
t_RTO = 4*R.
4) Update the sending rate as follows:
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If (sender has been idle or data-limited
within last two round-trip times)
min_rate = max(2*X_recv, W_init/R);
Else
min_rate = 2*X_recv;
If (p > 0)
Calculate X_Bps using the TCP throughput equation.
X = max(min(X_Bps, min_rate), s/t_mbi);
Else if ((min_rate < X) and (the first feedback packet, or
the first feedback packet after a nofeedback timer))
Do nothing;
Else if (t_now - tld >= R)
X = max(min(2*X, min_rate), s/R);
tld = t_now;
The condition ``if (sender has been idle or data-limited within last
two round-trip times)'' prevents an idle or data-limited sender from
having to reduce the sending rate to less than the initial sending
rate as a result of limitations from a small receive rate. The
condition ``if (not the first feedback packet, and not the first
feedback packet after a nofeedback timer)'' prevents a sender from
reducing the sending rate in response to a feedback packet that
reports the receipt of only a few packets after start-up or after an
idle period.
Note that if p == 0, then the sender is in slow-start phase, where
it approximately doubles the sending rate each round-trip time until
a loss occurs. The s/R term gives a minimum sending rate during
slow-start of one packet per RTT. The parameter t_mbi is 64
seconds, and represents the maximum inter-packet backoff interval in
the persistent absence of feedback. Thus, when p > 0 the sender
sends at least one packet every 64 seconds.
5) Reset the nofeedback timer to expire after max(4*R, 2*s/X)
seconds.
4.4. Expiration of nofeedback timer
If the nofeedback timer expires, the sender should perform the
following actions:
1) Cut the sending rate in half. If the sender has received
feedback from the receiver, this is done by modifying the
sender's cached copy of X_recv (the receive rate). Because the
sending rate is limited to at most twice X_recv, modifying
X_recv limits the current sending rate, but allows the sender to
slow-start, doubling its sending rate each RTT, if feedback
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messages resume reporting no losses.
If (X_Bps > 2*X_recv)
X_recv = max(X_recv/2, s/(2*t_mbi));
Else
X_recv = X_Bps/4;
The term s/(2*t_mbi) limits the backoff to one packet every 64
seconds in the case of persistent absence of feedback.
2) The value of X must then be recalculated as described under
point (4) above.
If the nofeedback timer expires when the sender does not yet
have an RTT sample and has not yet received any feedback from
the receiver, or when p == 0, then step (1) can be skipped, and
the sending rate cut in half directly:
X = max(X/2, s/t_mbi)
3) Restart the nofeedback timer to expire after max(4*R, 2*s/X)
seconds.
Note that when the sender stops sending, the receiver will stop
sending feedback. When the sender's nofeedback timer expires, the
sender will decrease X_recv. If the sender subsequently starts to
send again, X_recv will limit the transmit rate, and a normal
slowstart phase will occur until the transmit rate reaches X_Bps.
4.5. Sending a packet after an idle or data-limited period
If the sender has been idle (unable to send because there is little
or no data from the application), the allowed sending rate could
have been reduced due to the nofeedback timer, as specified in the
section above. Because the sender is always restricted to sending
at most twice the receive rate reported by the receiver, the sender
will be limited to at most doubling its sending rate each round-trip
time, until the sending rate reaches the allowed sending rate
calculated by the throughput equation.
4.6. Preventing Oscillations
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To prevent oscillatory behavior in environments with a low degree of
statistical multiplexing it is useful to modify sender's transmit
rate to provide congestion avoidance behavior by reducing the
transmit rate as the queuing delay (and hence RTT) increases. To do
this the sender maintains an estimate of the long-term RTT and
modifies its sending rate depending on how the most recent sample of
the RTT differs from this value. The long-term sample is R_sqmean,
and is set as follows:
If no feedback has been received before
R_sqmean = sqrt(R_sample);
Else
R_sqmean = q2*R_sqmean + (1-q2)*sqrt(R_sample);
Thus R_sqmean gives the exponentially weighted moving average of the
square root of the RTT samples. The constant q2 should be set
similarly to q, and we recommend a value of 0.9 as the default.
The sender obtains the base allowed transmit rate, X, from the
throughput function. It then calculates a modified instantaneous
transmit rate X_inst, as follows:
X_inst = X * R_sqmean / sqrt(R_sample);
When sqrt(R_sample) is greater than R_sqmean then the queue is
typically increasing and so the transmit rate needs to be decreased
for stable operation.
Note: This modification is not always strictly required, especially
if the degree of statistical multiplexing in the network is high.
However, we recommend that it is done because it does make TFRC
behave better in environments with a low level of statistical
multiplexing. If it is not done, we recommend using a very low
value of q, such that q is close to or exactly zero.
4.7. Scheduling of Packet Transmissions
As TFRC is rate-based, and as operating systems typically cannot
schedule events precisely, it is necessary to be opportunistic about
sending data packets so that the correct average rate is maintained
despite the coarse-grain or irregular scheduling of the operating
system. Thus a typical sending loop will calculate the correct
inter-packet interval, t_ipi, as follows:
t_ipi = s/X_inst;
Let t_now be the current time and i be a natural number, i = 0, 1,
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..., with t_i the nominal send time for the i-th packet. Then the
nominal send time t_(i+1) derives recursively as
t_0 = t_now,
t_(i+1) = t_i + t_ipi.
The parameter t_delta allows a degree of flexibility in the send
time of a packet. When the application becomes idle, it requests
re-scheduling for time t_i = t_(i-1) + t_ipi, for t_(i-1) the send
time for the previous packet. When the application is re-scheduled,
it checks the current time, t_now. If (t_now > t_i - t_delta) then
packet i is sent.
In some cases, when the nominal send time, t_i, of the next packet
is calculated, it may already be the case that t_now > t_i -
t_delta. In such a case the packet should be sent immediately.
Thus if the operating system has coarse timer granularity and the
transmit rate is high, then TFRC may send short bursts of several
packets separated by intervals of the OS timer granularity.
If the operating system has a scheduling timer granularity of t_gran
seconds, then t_delta would typically be set to:
t_delta = min(t_ipi/2, t_gran/2);
t_gran is 10ms on many Unix systems. If t_gran is not known, a
value of 10ms can be safely assumed.
5. Calculation of the Loss Event Rate (p)
Obtaining an accurate and stable measurement of the loss event rate
is of primary importance for TFRC. Loss rate measurement is
performed at the receiver, based on the detection of lost or marked
packets from the sequence numbers of arriving packets. We describe
this process before describing the rest of the receiver protocol.
5.1. Detection of Lost or Marked Packets
TFRC assumes that all packets contain a sequence number that is
incremented by one for each packet that is sent. For the purposes
of this specification, we require that if a lost packet is
retransmitted, the retransmission is given a new sequence number
that is the latest in the transmission sequence, and not the same
sequence number as the packet that was lost. If a transport
protocol has the requirement that it must retransmit with the
original sequence number, then the transport protocol designer must
figure out how to distinguish delayed from retransmitted packets and
how to detect lost retransmissions.
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The receiver maintains a data structure that keeps track of which
packets have arrived and which are missing. For the purposes of
specification, we assume that the data structure consists of a list
of packets that have arrived along with the receiver timestamp when
each packet was received. In practice this data structure will
normally be stored in a more compact representation, but this is
implementation-specific.
The loss of a packet is detected by the arrival of at least NDUPACK
packets with a higher sequence number than the lost packet, for
NDUPACK set to 3. The requirement for NDUPACK subsequent packets is
the same as with TCP, and is to make TFRC more robust in the
presence of reordering. In contrast to TCP, if a packet arrives
late (after NDUPACK subsequent packets arrived) in TFRC, the late
packet can fill the hole in TFRC's reception record, and the
receiver can recalculate the loss event rate. Future versions of
TFRC might make the requirement for NDUPACK subsequent packets
adaptive based on experienced packet reordering, but we do not
specify such a mechanism here.
For an ECN-capable connection, a marked packet is detected as a
congestion event as soon as it arrives, without having to wait for
the arrival of subsequent packets.
5.2. Translation from Loss History to Loss Events
TFRC requires that the loss fraction be robust to several
consecutive packets lost or marked where those packets are part of
the same loss event. This is similar to TCP, which (typically) only
performs one halving of the congestion window during any single RTT.
Thus the receiver needs to map the packet loss history into a loss
event record, where a loss event is one or more packets lost or
marked in an RTT. To perform this mapping, the receiver needs to
know the RTT to use, and this is supplied periodically by the
sender, typically as control information piggy-backed onto a data
packet. TFRC is not sensitive to how the RTT measurement sent to
the receiver is made, but we recommend using the sender's calculated
RTT, R, (see Section 4.3) for this purpose.
To determine whether a lost or marked packet should start a new loss
event, or be counted as part of an existing loss event, we need to
compare the sequence numbers and timestamps of the packets that
arrived at the receiver. For a marked packet S_new, its reception
time T_new can be noted directly. For a lost packet, we can
interpolate to infer the nominal "arrival time". Assume:
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S_loss is the sequence number of a lost packet.
S_before is the sequence number of the last packet to arrive
with sequence number before S_loss.
S_after is the sequence number of the first packet to arrive
with sequence number after S_loss.
S_max is the largest sequence number.
T_loss is the nominal estimated arrival time for the lost
packet.
T_before is the reception time of S_before.
T_after is the reception time of S_after.
Note that T_before can either be before or after T_after due to
reordering.
For a lost packet S_loss, we can interpolate its nominal "arrival
time" at the receiver from the arrival times of S_before and
S_after. Thus:
T_loss = T_before + ( (T_after - T_before)
* (S_loss - S_before)/(S_after - S_before) );
Note that if the sequence space wrapped between S_before and
S_after, then the sequence numbers must be modified to take this
into account before performing this calculation. If the largest
possible sequence number is S_max, and S_before > S_after, then
modifying each sequence number S by S' = (S + (S_max + 1)/2) mod
(S_max + 1) would normally be sufficient.
If the lost packet S_old was determined to have started the previous
loss event, and we have just determined that S_new has been lost,
then we interpolate the nominal arrival times of S_old and S_new,
called T_old and T_new respectively.
If T_old + R >= T_new, then S_new is part of the existing loss
event. Otherwise S_new is the first packet in a new loss event.
5.3. Inter-loss Event Interval
If a loss interval, A, is determined to have started with packet
sequence number S_A and the next loss interval, B, started with
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packet sequence number S_B, then the number of packets in loss
interval A is given by (S_B - S_A). Thus, loss interval A contains
all of the packets transmitted by the sender starting with the first
packet transmitted in loss interval A, and ending with but not
including the first packet transmitted in loss interval B.
5.4. Average Loss Interval
To calculate the loss event rate p, we first calculate the average
loss interval. This is done using a filter that weights the n most
recent loss event intervals in such a way that the measured loss
event rate changes smoothly.
Weights w_0 to w_(n-1) are calculated as:
If (i < n/2)
w_i = 1;
Else
w_i = 1 - (i - (n/2 - 1))/(n/2 + 1);
Thus if n=8, the values of w_0 to w_7 are:
1.0, 1.0, 1.0, 1.0, 0.8, 0.6, 0.4, 0.2
The value n for the number of loss intervals used in calculating the
loss event rate determines TFRC's speed in responding to changes in
the level of congestion. As currently specified, TFRC should not be
used for values of n significantly greater than 8, for traffic that
might compete in the global Internet with TCP. At the very least,
safe operation with values of n greater than 8 would require a
slight change to TFRC's mechanisms to include a more severe response
to two or more round-trip times with heavy packet loss.
When calculating the average loss interval we need to decide whether
to include the interval since the most recent packet loss event. We
only do this if it is sufficiently large to increase the average
loss interval.
Let the most recent loss intervals be I_0 to I_k, where I_0 is the
interval starting with the most recent loss event (if there has been
one). If there have been at least n loss intervals, then k is set
to n; otherwise k is the maximum number of loss intervals seen so
far. We calculate the average loss interval I_mean is:
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I_tot0 = 0;
I_tot1 = 0;
W_tot = 0;
for (i = 0 to k-1) {
I_tot0 = I_tot0 + (I_i * w_i);
W_tot = W_tot + w_i;
}
for (i = 1 to k) {
I_tot1 = I_tot1 + (I_i * w_(i-1));
}
I_tot = max(I_tot0, I_tot1);
I_mean = I_tot/W_tot;
The loss event rate, p is simply:
p = 1 / I_mean;
5.5. History Discounting
As described in Section 5.4, when there have been at least eight
loss intervals, the most recent loss interval is only assigned
1/(0.75*n) of the total weight in calculating the average loss
interval, regardless of the size of the most recent loss interval.
This section describes an optional history discounting mechanism,
discussed further in [FHPW00a] and [W00], that allows the TFRC
receiver to adjust the weights, concentrating more of the relative
weight on the most recent loss interval, when the most recent loss
interval is more than twice as large as the computed average loss
interval.
To carry out history discounting, we associate a discount factor
DF_i with each loss interval L_i, for i > 0, where each discount
factor is a floating point number. The discount array maintains the
cumulative history of discounting for each loss interval. At the
beginning, the values of DF_i in the discount array are initialized
to 1:
for (i = 0 to n) {
DF_i = 1;
}
History discounting also uses a general discount factor DF, also a
floating point number, that is also initialized to 1. First we show
how the discount factors are used in calculating the average loss
interval, and then we describe later in this section how the
discount factors are modified over time.
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As described in Section 5.4 the average loss interval is calculated
using the n previous loss intervals I_1, ..., I_n, and the interval
I_0 that represents the number of packets sent since the beginning
of the last loss event. The computation of the average loss
interval using the discount factors is a simple modification of the
procedure in Section 5.4, as follows:
I_tot0 = I_0 * w_0
I_tot1 = 0;
W_tot0 = w_0
W_tot1 = 0;
for (i = 1 to n-1) {
I_tot0 = I_tot0 + (I_i * w_i * DF_i * DF);
W_tot0 = W_tot0 + w_i * DF_i * DF;
}
for (i = 1 to n) {
I_tot1 = I_tot1 + (I_i * w_(i-1) * DF_i);
W_tot1 = W_tot1 + w_(i-1) * DF_i;
}
p = min(W_tot0/I_tot0, W_tot1/I_tot1);
The general discounting factor, DF is updated on every packet
arrival as follows. First, the receiver computes the weighted
average I_mean of the loss intervals I_1, ..., I_n:
I_tot = 0;
W_tot = 0;
for (i = 1 to n) {
W_tot = W_tot + w_(i-1) * DF_i;
I_tot = I_tot + (I_i * w_(i-1) * DF_i);
}
I_mean = I_tot / W_tot;
This weighted average I_mean is compared to I_0, the number of
packets sent since the beginning of the last loss event. If I_0 is
greater than twice I_mean, then the new loss interval is
considerably larger than the old ones, and the general discount
factor DF is updated to decrease the relative weight on the older
intervals, as follows:
if (I_0 > 2 * I_mean) {
DF = 2 * I_mean/I_0;
if (DF < THRESHOLD)
DF = THRESHOLD;
} else
DF = 1;
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A nonzero value for THRESHOLD ensures that older loss intervals from
an earlier time of high congestion are not discounted entirely. We
recommend a THRESHOLD of 0.5. Note that with each new packet
arrival, I_0 will increase further, and the discount factor DF will
be updated.
When a new loss event occurs, the current interval shifts from I_0
to I_1, loss interval I_i shifts to interval I_(i+1), and the loss
interval I_n is forgotten. The previous discount factor DF has to
be incorporated into the discount array. Because DF_i carries the
discount factor associated with loss interval I_i, the DF_i array
has to be shifted as well. This is done as follows:
for (i = 1 to n) {
DF_i = DF * DF_i;
}
for (i = n-1 to 0 step -1) {
DF_(i+1) = DF_i;
}
I_0 = 1;
DF_0 = 1;
DF = 1;
This completes the description of the optional history discounting
mechanism. We emphasize that this is an optional mechanism whose
sole purpose is to allow TFRC to response somewhat more quickly to
the sudden absence of congestion, as represented by a long current
loss interval.
6. Data Receiver Protocol
The receiver periodically sends feedback messages to the sender.
Feedback packets should normally be sent at least once per RTT,
unless the sender is sending at a rate of less than one packet per
RTT, in which case a feedback packet should be send for every data
packet received. A feedback packet should also be sent whenever a
new loss event is detected without waiting for the end of an RTT,
and whenever an out-of-order data packet is received that removes a
loss event from the history.
If the sender is transmitting at a high rate (many packets per RTT)
there may be some advantages to sending periodic feedback messages
more than once per RTT as this allows faster response to changing
RTT measurements, and more resilience to feedback packet loss. If
the receiver was sending k feedback packets per RTT, step (4) of
Section 6.2 would be modified to set the feedback timer to expire
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after R_m/k seconds. However, each feedback packet would still
report the receiver rate over the last RTT, not over a fraction of
an RTT. We note that there is little gain from sending a large
number of feedback messages per RTT.
6.1. Receiver behavior when a data packet is received
When a data packet is received, the receiver performs the following
steps:
1) Add the packet to the packet history.
2) Let the previous value of p be p_prev. Calculate the new value
of p as described in Section 5.
3) If p > p_prev, cause the feedback timer to expire, and perform
the actions described in Section 6.2
If p <= p_prev no action need be performed.
However an optimization might check to see if the arrival of the
packet caused a hole in the packet history to be filled and
consequently two loss intervals were merged into one. If this
is the case, the receiver might also send feedback immediately.
The effects of such an optimization are normally expected to be
small.
6.2. Expiration of feedback timer
When the feedback timer at the receiver expires, the action to be
taken depends on whether data packets have been received since the
last feedback was sent.
Let the maximum sequence number of a packet at the receiver so far
be S_m, and the value of the RTT measurement included in packet S_m
be R_m. As described in Section 3.2.1, R_m is the sender's current
estimate of the round trip time, reported in data packets. If data
packets have been received since the previous feedback was sent, the
receiver performs the following steps:
1) Calculate the average loss event rate using the algorithm
described above.
2) Calculate the measured receive rate, X_recv, based on the
packets received within the previous R_m seconds.
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3) Prepare and send a feedback packet containing the information
described in Section 3.2.2
4) Restart the feedback timer to expire after R_m seconds.
Note that rule 2) above gives a minimum value for the measured
receive rate X_recv of one packet per round-trip time. If the
sender is limited to a sending rate of less than one packet per
round-trip time, this will be due to the loss event rate, not from a
limit imposed by the measured receive rate at the receiver.
If no data packets have been received since the last feedback was
sent, no feedback packet is sent, and the feedback timer is
restarted to expire after R_m seconds.
6.3. Receiver initialization
The receiver is initialized by the first data packet that arrives at
the receiver. Let the sequence number of this packet be i.
When the first packet is received:
o Set p=0
o Set X_recv = 0.
o Prepare and send a feedback packet.
o Set the feedback timer to expire after R_i seconds.
If the first data packet doesn't contain an estimate R_i of the
round-trip time, then the receiver sends a feedback packet for every
arriving data packet, until a data packet arrives containing an
estimate of the round-trip time.
If the sender is using a coarse-grained timestamp that increments
every quarter of a round-trip time, then a feedback timer is not
needed, and the following procedure from RFC 4342 is used to
determine when to send feedback messages.
o Whenever the receiver sends a feedback message, the receiver
sets a local variable last_counter to the greatest received
value of the window counter since the last feedback message was
sent, if any data packets have been received since the last
feedback message was sent.
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o If the receiver receives a data packet with a window counter
value greater than or equal to last_counter + 4, then the
receiver sends a new feedback packet. ("Greater" and "greatest"
are measured in circular window counter space.)
6.3.1. Initializing the Loss History after the First Loss Event
The number of packets until the first loss can not be used to
compute the allowed sending rate directly, as the sending rate
changes rapidly during this time. TFRC assumes that the correct
data rate after the first loss is half of the maximum sending rate
before the loss occurred. TFRC approximates this target rate
X_target by the maximum X_recv so far, for X_recv the receive rate
over a single round-trip time. (For a TFRC sender that always has
data to send, it is sufficient to approximate the target rate by the
most recent X_recv. However, for a TFRC sender that is sometimes
data-limited or idle, it is best to use the maximum X_recv so far.)
After the first loss, instead of initializing the first loss
interval to the number of packets sent until the first loss, the
TFRC receiver calculates the loss interval that would be required to
produce the data rate X_target, and uses this synthetic loss
interval to seed the loss history mechanism.
TFRC does this by finding some value p for which the throughput
equation in Section 3.1 gives a sending rate within 5% of X_target,
given the round-trip time R, and the first loss interval is then set
to 1/p. If the receiver knows the segment size s used by the
sender, then the receiver can use the throughput equation for X;
otherwise, the receiver can measure the receive rate in packets per
second instead of bytes per second for this purpose, and use the
throughput equation for X_pps. (The 5% tolerance is introduced
simply because the throughput equation is difficult to invert, and
we want to reduce the costs of calculating p numerically.)
Special care is needed for initializing the first loss interval when
the first data packet is lost or marked. When the first data packet
is lost in TCP, the TCP sender retransmits the packet after the
retransmit timer expires. If TCP's first data packet is ECN-marked,
the TCP sender resets the retransmit timer, and sends a new data
packet only when the retransmit timer expires [RFC3168] (Section
6.1.2). For TFRC, if the first data packet is lost or ECN-marked,
then the first loss interval consists of the null interval with no
data packets. In this case, the loss interval length for this
(null) loss interval should be set to give a similar sending rate to
that of TCP.
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When the first TFRC loss interval is null, meaning that the first
data packet is lost or ECN-marked, in order to follow the behavior
of TCP, TFRC wants the allowed sending rate to be 1 packet every two
round-trip times, or equivalently, 0.5 packets per RTT. Thus, the
TFRC receiver calculates the loss interval that would be required to
produce the target rate X_target of 0.5/R packets per second, for
the round-trip time R, and uses this synthetic loss interval for the
first loss interval. The TFRC receiver uses 0.5/R packets per
second as the minimum value for X_target when initializing the first
loss interval.
7. Sender-based Variants
In a sender-based variant of TFRC, the receiver would use reliable
delivery to send information about packet losses to the sender, and
the sender would compute the packet loss rate and the acceptable
transmit rate.
The main advantages of a sender-based variant of TFRC would be that
the sender would not have to trust the receiver's calculation of the
packet loss rate. However, with the requirement of reliable
delivery of loss information from the receiver to the sender, a
sender-based TFRC would have much tighter constraints on the
transport protocol in which it is embedded.
In contrast, the receiver-based variant of TFRC specified in this
document is robust to the loss of feedback packets, and therefore
does not require the reliable delivery of feedback packets. It is
also better suited for applications where it is desirable to offload
work from the server to the client as much as possible.
RFC 4340 and RFC 4342 together specify CCID 3, which can be used as
a sender-based variant of TFRC. In CCID 3, each feedback packet
from the receiver contains a Loss Intervals option, reporting the
lengths of the most recent loss intervals. Feedback packets may
also include the Ack Vector option, allowing the sender to determine
exactly which packets were dropped or marked, and to check the
information reported in the Loss Intervals options. The Ack Vector
option can also include ECN Nonce Echoes, allowing the sender to
verify the receiver's report of having received a data packet. The
Ack Vector option allows the sender to determine for itself which
data packets were lost or ECN-marked, to determine loss intervals,
and to calculate the loss event rate. Section 9.2 of RFC 4342
discusses issues in the sender verifying information reported by the
receiver.
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8. Implementation Issues
This document has specified the TFRC congestion control mechanism,
for use by applications and transport protocols. This section
mentions briefly some of the few implementation issues.
For t_RTO = 4*R and b = 1, the throughput equation in Section 3.1
can be expressed as follows:
s
X_Bps = --------
R * f(p)
for
f(p) = sqrt(2*p/3) + (12*sqrt(3*p/8) * p * (1+32*p^2)).
A table lookup could be used for the function f(p).
Many of the multiplications (e.g., q and 1-q for the round-trip time
average, a factor of 4 for the timeout interval) are or could be by
powers of two, and therefore could be implemented as simple shift
operations.
We note that the optional sender mechanism for preventing
oscillations described in Section 4.6 uses a square-root
computation.
For the calculation of the nominal arrival time T_loss for a lost
packet from Section 5.2, one way to implement this that would avoid
concerns about wrapped sequence space would be to use the following:
T_loss = T_before + (T_after - T_before) * Dist(S_loss,
S_before)/Dist(S_after, S_before)
where
Dist(Seqno_A, Seqno_B) = (Seqno_A + 2^48 - Seqno_B) % 2^48
The calculation of the average loss interval in Section 5.4 involves
multiplications by the weights w_0 to w_(n-1), which for n=8 are:
1.0, 1.0, 1.0, 1.0, 0.8, 0.6, 0.4, 0.2.
With a minor loss of smoothness, it would be possible to use weights
that were powers of two or sums of powers of two, e.g.,
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1.0, 1.0, 1.0, 1.0, 0.75, 0.5, 0.25, 0.25.
The optional history discounting mechanism described in Section 5.5
is used in the calculation of the average loss rate. The history
discounting mechanism is invoked only when there has been an
unusually long interval with no packet losses. For a more efficient
operation, the discount factor DF_i could be restricted to be a
power of two.
9. Changes from RFC 3448
The changes from RFC 3448 are as follows:
o Changes to the initial sending rate: In RFC 3448, the initial
sending rate was two packets per round trip time. In this
document, the initial sending rate can be as high as four
packets per round trip time, following RFC 3390.
Following Section 5.1 from [RFC4342], this document also
specifies that when the sending rate is reduced after an idle
period, it is not reduced below the initial sending rate. In
addition, when the sender has been data-limited and the sender
is reducing the allowed transmit rate to twice the receive
rate,, the sender doesn't reduce the allowed transmit rate to
less than the initial sending rate.
A larger initial sending rate is of little use if the receiver
sends a feedback packet after the first packet is received, and
the sender in response reduces the allowed sending rate to at
most twice the receive rate. In the current document, the
sender does not reduce the allowed sending rate to at most twice
the receive rate in response to the first feedback packet.
o RFC 3448 had contradictory text about whether the sender halved
its sending rate after *two* round-trip times without receiving
a feedback report, or after *four* round-trip times. This
document clarifies that the sender halves its sending rate after
four round-trip times without receiving a feedback report
[RFC3448Err].
o Section 4.4 was clarified to specify that on the expiration of
the nofeedback timer, if p = 0, step (2) applies instead of step
(1) [RFC3448Err].
o A line in Section 5.5 was changed from ``for (i = 1 to n) { DF_i
= 1; }'' to ``for (i = 0 to n) { DF_i = 1; }'' [RFC3448Err].
Handley/Floyd/Padhye/Widmer Section 9. [Page 31]
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o Section 5.4 was modified to clarify the receiver's calculation
of the average loss interval when the receiver has not yet seen
eight loss intervals.
o Section 4.1 was modified to give a specific algorithm that could
be used for estimating the average segment size.
10. Security Considerations
TFRC is not a transport protocol in its own right, but a congestion
control mechanism that is intended to be used in conjunction with a
transport protocol. Therefore security primarily needs to be
considered in the context of a specific transport protocol and its
authentication mechanisms.
Congestion control mechanisms can potentially be exploited to create
denial of service. This may occur through spoofed feedback. Thus
any transport protocol that uses TFRC should take care to ensure
that feedback is only accepted from the receiver of the data. The
precise mechanism to achieve this will however depend on the
transport protocol itself.
In addition, congestion control mechanisms may potentially be
manipulated by a greedy receiver that wishes to receive more than
its fair share of network bandwidth. A receiver might do this by
claiming to have received packets that in fact were lost due to
congestion. Possible defenses against such a receiver would
normally include some form of nonce that the receiver must feed back
to the sender to prove receipt. However, the details of such a
nonce would depend on the transport protocol, and in particular on
whether the transport protocol is reliable or unreliable.
We expect that protocols incorporating ECN with TFRC will also want
to incorporate feedback from the receiver to the sender using the
ECN nonce [RFC3540]. The ECN nonce is a modification to ECN that
protects the sender from the accidental or malicious concealment of
marked packets. Again, the details of such a nonce would depend on
the transport protocol, and are not addressed in this document.
11. IANA Considerations
There are no IANA actions required for this document.
Handley/Floyd/Padhye/Widmer Section 11. [Page 32]
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12. Acknowledgments
We would like to acknowledge feedback and discussions on equation-
based congestion control with a wide range of people, including
members of the Reliable Multicast Research Group, the Reliable
Multicast Transport Working Group, and the End-to-End Research
Group. We would like to thank Dado Colussi, Gorry Fairhurst, Ladan
Gharai, Wim Heirman, Eddie Kohler, Ken Lofgren, Mike Luby, Ian
McDonald, Michele R., Gerrit Renker, Arjuna Sathiaseelan, Vladica
Stanisic, Randall Stewart, Eduardo Urzaiz, Shushan Wen, and Wendy
Lee (lhh@zsu.edu.cn) for feedback on earlier versions of this
document, and to thank Mark Allman for his extensive feedback from
using the document to produce a working implementation.
13. Terminology
This document uses the following terms:
Handley/Floyd/Padhye/Widmer Section 13. [Page 33]
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DF: discount factor for a loss interval
last_counter : greatest received value of the window counter
min_rate : minimum transmit rate
MSS : Maximum Segment Size (constant)
n : number of loss intervals
NDUPACK : number of dupacks for inferring loss (constant)
nofeedback timer : sender-side timer
p : measured Loss Event Rate
p_prev : previous value of p
q : filter constant for RTT (constant)
q2 : filter constant for long-term RTT (constant)
R : estimated path round-trip time
R_sample : measured path RTT
R_sqmean : estimated long-term RTT
s : nominal packet size in bytes (constant)
S : sequence number
t_delta : parameter for flexibility in send time
t_gran : schedular granularity (constant)
t_ipi : calculated inter-packet interval for sending packets
t_mbi : maximum RTO value of TCP (constant)
tld : Time Last Doubled
t_now : current time
t_RTO : estimated RTO of TCP
X : allowed transmit rate
Handley/Floyd/Padhye/Widmer Section 13. [Page 34]
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X_Bps : calculated sending rate in bytes per second
X_pps : calculated sending rate in packets per second
X_recv : estimated receive rate at the receiver
X_inst : instantaneous transmit rate
W_init : TCP initial window (constant)
14. Normative References
15. Informational References
[BRS99] Balakrishnan, H., Rahul, H., and Seshan, S., "An
Integrated Congestion Management Architecture for
Internet Hosts," Proc. ACM SIGCOMM, Cambridge, MA,
September 1999.
[FHPW00] S. Floyd, M. Handley, J. Padhye, and J. Widmer,
"Equation-Based Congestion Control for Unicast
Applications", August 2000, Proc SIGCOMM 2000.
[FHPW00a] S. Floyd, M. Handley, J. Padhye, and J. Widmer,
"Equation-Based Congestion Control for Unicast
Applications: the Extended Version", ICSI tech
report TR-00-03, March 2000.
[PFTK98] Padhye, J. and Firoiu, V. and Towsley, D. and
Kurose, J., "Modeling TCP Throughput: A Simple Model
and its Empirical Validation", Proc ACM SIGCOMM
1998.
[RFC2119] S. Bradner, Key Words For Use in RFCs to Indicate
Requirement Levels, RFC 2119.
[RFC2140] J. Touch, "TCP Control Block Interdependence", RFC
2140, April 1997.
[RFC2988] V. Paxson and M. Allman, "Computing TCP's
Retransmission Timer", RFC 2988, November 2000.
[RFC3168] K. Ramakrishnan and S. Floyd, "The Addition of
Explicit Congestion Notification (ECN) to IP", RFC
3168, September 2001.
Handley/Floyd/Padhye/Widmer Section 15. [Page 35]
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[RFC3390] Allman, M., Floyd, S., and C. Partridge, "Increasing
TCP's Initial Window", RFC 3390, October 2002.
[RFC3448Err] RFC 3448 Errata, URL
``http://www.icir.org/tfrc/rfc3448.errata''.
[RFC3540] Wetherall, D., Ely, D., and Spring, N., "Robust ECN
Signaling with Nonces", RFC 3540, Experimental, June
2003
[RFC4340] Kohler, E., Handley, M., and S. Floyd, "Datagram
Congestion Control Protocol (DCCP)", RFC 4340, March
2006.
[RFC4342] Floyd, S., Kohler, E., and J. Padhye, "Profile for
Datagram Congestion Control Protocol (DCCP)
Congestion Control ID 3: TCP-Friendly Rate Control
(TFRC)", RFC 4342, March 2006.
[TFRC-SP] Floyd, S., and E. Kohler, TCP Friendly Rate Control
(TFRC): the Small-Packet (SP) Variant, Internet
draft draft-ietf-dccp-tfrc-voip-07.txt, work in
progress, November 2006. Approved for Experimental.
.
[W00] Widmer, J., "Equation-Based Congestion Control",
Diploma Thesis, University of Mannheim, February
2000. URL "http://www.icir.org/tfrc/".
16. Authors' Addresses
Handley/Floyd/Padhye/Widmer Section 16. [Page 36]
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Mark Handley,
Department of Computer Science
University College London
Gower Street
London WC1E 6BT
UK
EMail: M.Handley@cs.ucl.ac.uk
Sally Floyd
ICIR/ICSI
1947 Center St, Suite 600
Berkeley, CA 94708
floyd@icir.org
Jitendra Padhye
Microsoft Research
padhye@microsoft.com
Joerg Widmer
Lehrstuhl Praktische Informatik IV
Universitat Mannheim
L 15, 16 - Room 415
D-68131 Mannheim
Germany
widmer@informatik.uni-mannheim.de
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Handley/Floyd/Padhye/Widmer [Page 37]
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Handley/Floyd/Padhye/Widmer [Page 38]
