Internet Engineering Task Force                                   TSV WG
INTERNET-DRAFT                                        Mark Handley/ACIRI
draft-ietf-tsvwg-tfrc-02.txt                       Jitendra Padhye/ACIRI
                                                       Sally Floyd/ACIRI

                                             Joerg Widmer/Univ. Mannheim
                                                             18 May 2001
                                                  Expires: November 2001

                   TCP Friendly Rate Control (TFRC):
                         Protocol Specification

Status of this Document

This document is an Internet-Draft and is in full conformance with all
provisions of Section 10 of RFC2026.

Internet-Drafts are working documents of the Internet Engineering Task
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This document is a product of the IETF TSV WG.  Comments should be
addressed to the authors.


     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,

Handley/Padhye/Floyd/Widmer                                     [Page 1]

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     but has a much lower variation of throughput over time
     compared with TCP, making it more suitable for applications
     such as telephony or streaming media where a relatively smooth
     sending rate is of importance.

Handley/Padhye/Floyd/Widmer                                     [Page 2]

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                           Table of Contents

     1. Introduction. . . . . . . . . . . . . . . . . . . . . .   4
     2. Terminology . . . . . . . . . . . . . . . . . . . . . .   5
     3. Protocol Mechanism. . . . . . . . . . . . . . . . . . .   5
      3.1. TCP Throughput Equation. . . . . . . . . . . . . . .   5
      3.2. Packet Contents. . . . . . . . . . . . . . . . . . .   7
       3.2.1. Data Packets. . . . . . . . . . . . . . . . . . .   7
       3.2.2. Feedback Packets. . . . . . . . . . . . . . . . .   8
     4. Data Sender Protocol. . . . . . . . . . . . . . . . . .   8
      4.1. Measuring the Packet Size. . . . . . . . . . . . . .   8
      4.2. Sender Initialization. . . . . . . . . . . . . . . .   9
      4.3. Sender behavior when a feedback packet is
      received. . . . . . . . . . . . . . . . . . . . . . . . .   9
      4.4. Expiration of nofeedback timer . . . . . . . . . . .  10
      4.5. Preventing Oscillations. . . . . . . . . . . . . . .  11
      4.6. Scheduling of Packet Transmissions . . . . . . . . .  12
     5. Calculation of the Loss Event Rate (p). . . . . . . . .  13
      5.1. Detection of Lost or Marked Packets. . . . . . . . .  13
      5.2. Translation from Loss History to Loss Events . . . .  13
      5.3. Inter-loss Event Interval. . . . . . . . . . . . . .  15
      5.4. Average Loss Interval. . . . . . . . . . . . . . . .  15
      5.5. History Discounting. . . . . . . . . . . . . . . . .  16
     6. Data Receiver Protocol. . . . . . . . . . . . . . . . .  18
      6.1. Receiver behavior when a data packet is
      received. . . . . . . . . . . . . . . . . . . . . . . . .  19
      6.2. Expiration of feedback timer . . . . . . . . . . . .  19
      6.3. Receiver initialization. . . . . . . . . . . . . . .  20
       6.3.1. Initializing the Loss History after the
       First Loss Event . . . . . . . . . . . . . . . . . . . .  20
     7. Security Considerations . . . . . . . . . . . . . . . .  20
     8. Authors' Addresses. . . . . . . . . . . . . . . . . . .  21
     9. Acknowledgments . . . . . . . . . . . . . . . . . . . .  21
     10. References . . . . . . . . . . . . . . . . . . . . . .  22

Handley/Padhye/Floyd/Widmer                                     [Page 3]

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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 a
Internet environment and competing with TCP traffic.  Instead of
specifying a complete protocol, this document simply specifies a
congestion control mechanism that could be used in a transport protocol
such as RTP [6], 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, reliability, or implementation-related issues.

TFRC is designed to be reasonably fair when competing for bandwidth with
TCP flows [2]. However it 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 applications that use a fixed packet size, and vary
their sending rate in packets per second in response to congestion.
Some audio applications require a fixed interval of time between packets
and vary their packet size instead of their packet rate in response to
congestion.  The congestion control mechanism in this document cannot be
used by those applications; variants of TFRC for applications that have
a fixed sending rate but vary their packet size in response to
congestion will be addressed in a separate document.

TFRC is 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.

Handley/Padhye/Floyd/Widmer                                     [Page 4]

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2.  Terminology

In this document, the key words "MUST", "MUST NOT", "REQUIRED", "SHALL",
"OPTIONAL" are to be interpreted as described in RFC 2119 and indicate
requirement levels for compliant TFRC implementations.

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 packet
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) [7].

Generally speaking, TFRC's congestion control mechanism works as

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

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

Handley/Padhye/Floyd/Widmer                                     [Page 5]

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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

The throughput equation we currently recommend for TFRC is a slightly
simplified version of the throughput equation for Reno TCP from [4].
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:

     X =  ----------------------------------------------------------
          R*sqrt(2*b*p/3) + (t_RTO * (3*sqrt(3*b*p/8) * p * (1+32*p^2)))


     X is the transmit rate in bytes/second.

     s is the packet 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

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 [5]. Another possibility would be to set t_RTO = max(4R,
one second), to match the recommended minimum of one second on the RTO

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 =

Handley/Padhye/Floyd/Widmer                                     [Page 6]

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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

The parameters s (packet 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.

3.2.1.  Data Packets

Each data packet sent by the data sender contains the following

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

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.

o    The sender's current estimate of the round trip time. The estimate
     reported in packet i is denoted by R_i.

o    The sender's current transmit rate. The estimate reported in packet
     i is denoted by X_i.

Handley/Padhye/Floyd/Widmer                                     [Page 7]

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3.2.2.  Feedback Packets

Each feedback packet sent by the data receiver contains the following

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.

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 two 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 packet 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 Packet Size

The parameter s (packet size) is normally known to an application.  This
may not be so in two cases:

Handley/Padhye/Floyd/Widmer                                     [Page 8]

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o    The packet size naturally varies depending on the data.  In this
     case, although the packet size varies, that variation is not
     coupled to the transmit rate.  It should normally be safe to use an
     estimate of the mean packet size for s.

o    The application needs to change the packet size rather than the
     number of packets 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.

The second class of applications are discussed separately in a separate
document.  For the remainder of this section we assume the sender can
estimate the packet size, and that congestion control is performed by
adjusting the number of packets sent per second.

4.2.  Sender Initialization

To initialize the sender, the value of X is set to 1 packet/second and
the nofeedback timer is set to expire after 2 seconds. The initial
values for R (RTT) and t_RTO are undefined until they are set as
described above. The intial value of tld, for the Time Last Doubled
during slow-start, is set to -1.

4.3.  Sender behavior when a feedback packet is received

The sender knows its current 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;
              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.

Handley/Padhye/Floyd/Widmer                                     [Page 9]

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3)   Update the timeout interval:

          t_RTO = 4*R.

4)   Update the sending rate as follows:

               If (p > 0)
                   Calculate X_calc using the TCP throughput equation.
                   X = max(min(X_calc, 2*X_recv), s/64);
                   If (t_now - tld >= RTT)
                       X = max(min(2*X, 2*X_recv), s/RTT);
                       tld = t_now;

          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/RTT term gives a minimum
          sending rate during slow-start of one packet per RTT.  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

1)   Cut the sending rate in half.  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 messages
     resume reporting no losses.

               If (X_calc > 2*X_recv)
                   X_recv = max(X_recv/2, s/128);
                   X_recv = X_calc/4;

     The s/128 term 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.

Handley/Padhye/Floyd/Widmer                                    [Page 10]

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     If the nofeedback timer expires when the sender does not yet have
     an RTT sample, and has not yet received any feedback from the
     sender, then step (1) can be skipped, and the sending rate cut in
     half directly:

            X = max(X/2, s/64)

3)   Restart the nofeedback timer to expire after max(4*R, 2*s/X)

Note that when the sender stops sending, the receiver will stop sending
feedback.  This will cause the nofeedback timer to start to expire and
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_calc.

4.5.  Preventing Oscillations

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);
         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 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

Handley/Padhye/Floyd/Widmer                                    [Page 11]

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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.

4.6.  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 course-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;

When a sender first starts sending at time t_0, it calculates t_ipi, and
calculates a nominal send time t_1 = t_0 + t_ipi for packet 1.  When the
application becomes idle, it checks the current time, t_now, and then
requests re-scheduling after (t_ipi - (t_now - t_0)) seconds.  When the
application is re-scheduled, it checks the current time, t_now, again.
If (t_now > t_1 - delta) then packet 1 is sent.

Now a new t_ipi may be calculated, and used to calculate a nominal send
time t_2 for packet 2: t2 = t_1 + t_ipi.  The process then repeats, with
each successive packet's send time being calculated from the nominal
send time of the previous packet.

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 - 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.

The parameter delta is to allow a degree of flexibility in the send time
of a packet.  If the operating system has a scheduling timer granularity
of t_gran seconds, then delta would typically be set to:

     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.

Handley/Padhye/Floyd/Widmer                                    [Page 12]

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5.  Calculation of the Loss Event Rate (p)

Obtaining a 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.

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-

The loss of a packet is detected by the arrival of at least three
packets with a higher sequence number than the lost packet.  The
requirement for three 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 3 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 three 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 where those packets are part of the same loss event.  This

Handley/Padhye/Floyd/Widmer                                    [Page 13]

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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 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:

     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.

     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

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

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

Handley/Padhye/Floyd/Widmer                                    [Page 14]

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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 packet
sequence number S_B, then the number of packets in loss interval A is
given by (S_B - S_A).

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;
        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 TRFC'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

Handley/Padhye/Floyd/Widmer                                    [Page 15]

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Thus if the most recent loss intervals are I_0 to I_n, with I_0 being
the interval since the most recent loss event, then we calculate the
average loss interval I_mean as:

     I_tot0 = 0;
     I_tot1 = 0;
     W_tot = 0;
     for (i = 0 to n-1) {
       I_tot0 = I_tot0 + (I_i * w_i);
       W_tot = W_tot + w_i;
     for (i = 1 to n) {
       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, 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 [3] and [5], 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 = 1 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 received since 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

     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_(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
received since 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;

A nonzero value for THRESHOLD ensures that older loss intervals from an

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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

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.  However
there is little gain from sending a large number of feedback messages
per RTT.

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6.1.  Receiver behavior when a data packet is received

When a data packet is received, the receiver performs the following

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.
If data packets have been received since the pervious feedback was sent,
the receiver performs the following steps:

1)   Calculate the average loss event rate using the algorithm described

2)   Calculate the measured receive rate, X_recv, based on the packets
     received within the previous R_m seconds.

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.

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.

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6.3.  Receiver initialization

The receiver is initialized by the first 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 = X_send, where X_send is the sending rate the sender
     reports in the first data packet.

o    Prepare and send a feedback packet.

o    Set the feedback timer to expire after R_i seconds.

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 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 sending rate when the loss occurred.  TFRC approximates
this target rate by X_recv, the receive rate over the most recent round-
trip time.  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_recv, 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_recv, given the
current packet size s and round-trip time R.  The first loss interval is
then set to 1/p.  (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.)

7.  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

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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

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
[WES01].   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.

8.  Authors' Addresses

     Mark Handley, Jitendra Padhye, Sally Floyd
     1947 Center St, Suite 600
     Berkeley, CA 94708,,

     Joerg Widmer
     Lehrstuhl Praktische Informatik IV
     Universitat Mannheim
     L 15, 16 - Room 415
     D-68131 Mannheim

9.  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

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Working Group, and the End-to-End Research Group.   We would like to
thank Eduardo Urzaiz, Vladica Stanisic, and Shushan Wen for feedback on
earlier versions of this document, and to thank Mark Allman for his
extensive feedback from using the draft to produce a working

10.  References

[1] Balakrishnan, H., Rahul, H., and Seshan, S., "An Integrated
     Congestion Management Architecture for Internet Hosts," Proc. ACM
     SIGCOMM, Cambridge, MA, September 1999.

[2] S. Floyd, M. Handley, J. Padhye, and J. Widmer, "Equation-Based
     Congestion Control for Unicast Applications", August 2000, Proc
     SIGCOMM 2000.

[3] 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.

[4] 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.

[5] Widmer, J., "Equation-Based Congestion Control", Diploma Thesis,
     University of Mannheim, February 2000.  URL

[6] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: A
     Transport Protocol for Real-Time Applications", RFC 1889, January

[7] K. Ramakrishnan and S. Floyd, "A Proposal to add Explicit Congestion
     Notification (ECN) to IP", RFC 2481, January 1999.

[8] V. Paxson and M. Allman, "Computing TCP's Retransmission Timer", RFC
     2988, November 2000.

[9] Wetherall, D., Ely, D., and Spring, N., "Robust ECN Signaling with
     Nonces", draft-ietf-tsvwg-tcp-nonce-00.txt, internet draft, work in
     progress, January 2001.  Citation for informational purposes only.

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