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Internet Engineering Task Force                                   RMT WG
INTERNET-DRAFT                             Joerg Widmer/DoCoMo Euro-Labs
draft-ietf-rmt-bb-tfmcc-06.txt                          Mark Handley/UCL

                                                            3 March 2006
                                                 Expires: September 2006


           TCP-Friendly Multicast Congestion Control (TFMCC):
                         Protocol Specification



Status of this Document

By submitting this Internet-Draft, each author represents that any
applicable patent or other IPR claims of which he or she is aware have
been or will be disclosed, and any of which he or she becomes aware will
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This document is a product of the IETF RMT WG.  Comments should be
addressed to the authors, or the WG's mailing list at rmt@lbl.gov.


                                Abstract


     This document specifies TCP-Friendly Multicast Congestion
     Control (TFMCC).  TFMCC is a congestion control mechanism for
     multicast transmissions in a best-effort Internet environment.



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     It is a single-rate congestion control scheme, where the
     sending rate is adapted to the receiver experiencing the worst
     network conditions.  TFMCC is reasonably fair when competing
     for bandwidth with TCP flows and has a relatively low
     variation of throughput over time, making it 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. . . . . . . . . . . . . . . . . . . . . .   4
      1.1. Terminology. . . . . . . . . . . . . . . . . . . . .   4
      1.2. Related Documents. . . . . . . . . . . . . . . . . .   5
      1.3. Environmental Requirements and Considerations. . . .   5
     2. Protocol Overview . . . . . . . . . . . . . . . . . . .   6
      2.1. TCP Throughput Equation. . . . . . . . . . . . . . .   7
      2.2. Packet Contents. . . . . . . . . . . . . . . . . . .   8
       2.2.1. Sender Packets. . . . . . . . . . . . . . . . . .   8
       2.2.2. Feedback Packets. . . . . . . . . . . . . . . . .  10
     3. Data Sender Protocol. . . . . . . . . . . . . . . . . .  10
      3.1. Sender Initialization. . . . . . . . . . . . . . . .  11
      3.2. Determining the Maximum RTT. . . . . . . . . . . . .  11
      3.3. Adjusting the Sending Rate . . . . . . . . . . . . .  12
      3.4. Controlling Receiver Feedback. . . . . . . . . . . .  13
      3.5. Assisting Receiver-Side RTT Measurements . . . . . .  14
      3.6. Slowstart. . . . . . . . . . . . . . . . . . . . . .  15
      3.7. Scheduling of Packet Transmissions . . . . . . . . .  16
     4. Data Receiver Protocol. . . . . . . . . . . . . . . . .  16
      4.1. Receiver Initialization. . . . . . . . . . . . . . .  17
      4.2. Receiver Leave . . . . . . . . . . . . . . . . . . .  17
      4.3. Measurement of the Network Conditions. . . . . . . .  17
       4.3.1. Updating the Loss Event Rate. . . . . . . . . . .  17
       4.3.2. Basic Round-Trip Time Measurement . . . . . . . .  17
       4.3.3. One-Way Delay Adjustments . . . . . . . . . . . .  18
       4.3.4. Receive Rate Measurements . . . . . . . . . . . .  19
      4.4. Setting the Desired Rate . . . . . . . . . . . . . .  19
      4.5. Feedback and Feedback Suppression. . . . . . . . . .  20
     5. Calculation of the Loss Event Rate. . . . . . . . . . .  21
      5.1. Detection of Lost or Marked Packets. . . . . . . . .  22
      5.2. Translation from Loss History to Loss Events . . . .  22
      5.3. Inter-Loss Event Interval. . . . . . . . . . . . . .  23
      5.4. Average Loss Interval. . . . . . . . . . . . . . . .  24
      5.5. History Discounting. . . . . . . . . . . . . . . . .  25
      5.6. Initializing the Loss History after the First
      Loss Event. . . . . . . . . . . . . . . . . . . . . . . .  27
     6. Security Considerations . . . . . . . . . . . . . . . .  28
     7. IANA Considerations . . . . . . . . . . . . . . . . . .  29
     8. Authors' Addresses. . . . . . . . . . . . . . . . . . .  29
     9. Acknowledgments . . . . . . . . . . . . . . . . . . . .  29
     10. References . . . . . . . . . . . . . . . . . . . . . .  30
     11. Copyright Notice . . . . . . . . . . . . . . . . . . .  31







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1.  Introduction

This document specifies TCP-Friendly Multicast Congestion Control
(TFMCC) [10]. TFMCC is a source-based, single-rate congestion control
scheme that builds upon the unicast TCP-Friendly Rate Control mechanism
(TFRC) [2]. TFMCC is stable and responsive under a wide range of network
conditions and scales to receiver sets on the order of several thousand
receivers.  To support scalability, as much congestion control
functionality as possible is located at the receivers.  Each receiver
continuously determines a desired receive rate that is TCP-friendly for
the path from the sender to this receiver.  Selected receivers then
report the rate to the sender in feedback packets.

TFMCC is a building block as defined in RFC 3048.  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 [7], in
an application incorporating end-to-end congestion control at the
application level. This document does not discuss packet formats,
reliability, or implementation-related issues.

TFMCC is designed to be reasonably fair when competing for bandwidth
with TCP flows.  A multicast flow is ``reasonably fair'' if its sending
rate is generally within a factor of two of the sending rate of a TCP
flow from the sender to the slowest receiver of the multicast group
under the same network conditions.

In general, TFMCC has a low variation of throughput, which makes it
suitable for applications such as streaming media where a relatively
smooth sending rate is of importance.  The penalty of having smooth
throughput while competing fairly for bandwidth is a reduced
responsiveness to changes in available bandwidth.  Thus TFMCC should be
used when the application has a requirement for smooth throughput, in
particular, avoiding halving of the sending rate in response to a single
packet drop.  For applications that simply need to multicast as much
data as possible in as short a time as possible, PGMCC [6] may be more
suitable.


1.1.  Terminology

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







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1.2.  Related Documents

As described in RFC 3048, TFMCC is a building block that is intended to
be used, in conjunction with other building blocks, to help specify a
protocol instantiation.  In particular, TFMCC is a suitable congestion
control building block for NACK-Oriented Reliable Multicast (NORM) [1].


1.3.  Environmental Requirements and Considerations

TFMCC is intended to be a congestion control scheme that can be used in
a complete protocol instantiation that delivers objects and streams
(both reliable content delivery and streaming of multimedia
information).

TFMCC is most applicable for sessions that deliver a substantial amount
of data, i.e., in length from hundreds of kilobytes to many gigabytes,
and whose duration is in the order of tens of seconds or more.

TFMCC is intended for multicast delivery.  There are currently two
models of multicast delivery, the Any-Source Multicast (ASM) model as
defined in RFC 1112 and the Source-Specific Multicast (SSM) model as
defined in [3]. TFMCC works with both multicast models, but in a
slightly different way.  When using ASM, feedback from the receivers is
multicast to the sender as well as to all other receivers.  Feedback can
be either multicast on the same group address used for sending data or
on a separate multicast feedback group address.  For SSM, the receivers
must unicast the feedback directly to the sender.  Hence, feedback from
a receiver will not be received by other receivers.

TFMCC inherently works with all types of networks, including LANs, WANs,
Intranets, the Internet, asymmetric networks, wireless networks, and
satellite networks.  However, in some network environments varying the
sending rate to the receivers may not be advantageous (e.g., for a
satellite or wireless network, there may be no mechanism for receivers
to effectively reduce their reception rate since there may be a fixed
transmission rate allocated to the session).

The difference in responsiveness of TFMCC and TCP may result in
significant throughput differences in case of a very low bitrate.  TFMCC
requires an estimate of the loss event rate to calculate a fair sending
rate.  This estimate may be inaccurate in case TFMCC receives only very
few packets per RTT.  TFMCC should not be used together with TCP if the
capacity of the bottleneck link is less than 30KBit/s (e.g., a very slow
modem connection).  TFMCC may also achieve a rate that is very different
from the average TCP rate in case buffer space at the bottleneck is
severely underprovisioned. In particular, TFMCC is less susceptible to
small buffer sizes since TFMCC spaces out packets in time, while TCP



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sends them back-to-back. Thus TCP is much more likely to see a packet
loss if buffer space is scarce.

TFMCC 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.


2.  Protocol Overview

TFMCC extends the basic mechanisms of TFRC into the multicast domain.
In order to compete fairly with TCP, TFMCC receivers individually
measure the prevalent network conditions and calculate a rate that is
TCP-friendly on the path from the sender to themselves.  The rate is
determined using an equation for TCP throughput, which roughly describes
TCP's sending rate as a function of the loss event rate, round-trip time
(RTT), and packet size.  We define a loss event as one or more lost or
marked packets from the packets received during one RTT, where a marked
packet refers to a congestion indication from Explicit Congestion
Notification (ECN) [5]. The sending rate of the multicast transmission
is adapted to the receiver experiencing the worst network conditions.

Basically, TFMCC's congestion control mechanism works as follows:

o Each receiver measures the loss event rate and its RTT to the sender.

o Each receiver then uses this information, together with an equation
  for TCP throughput, to derive a TCP-friendly sending rate.

o Through a distributed feedback suppression mechanism, only a subset of
  the receivers are allowed to give feedback to prevent a feedback
  implosion at the sender.  The feedback mechanism ensures that
  receivers reporting a low desired transmission rate have a high
  probability of sending feedback.

o Receivers whose feedback is not suppressed report the calculated
  transmission rate back to the sender in so-called receiver reports.
  The receiver reports serve two purposes: they inform the sender about
  the appropriate transmit rate, and they allow the receivers to measure
  their RTT.

o The sender selects the receiver that reports the lowest rate as
  current limiting receiver (CLR).  Whenever feedback with an even lower
  rate reaches the sender, the corresponding receiver becomes CLR and
  the sending rate is reduced to match that receiver's calculated rate.



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  The sending rate increases when the CLR reports a calculated rate
  higher than the current sending rate.

The dynamics of TFMCC are sensitive to how the measurements are
performed and applied and what feedback suppression mechanism is chosen.
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 TFMCC.


2.1.  TCP Throughput Equation

Any realistic equation giving TCP throughput as a function of loss event
rate and RTT should be suitable for use in TFMCC.  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 TFMCC is a slightly
simplified version of the throughput equation for Reno TCP from [4]:



                                    8 s
     X =  ---------------------------------------------------------   (1)
           R * (sqrt(2*p/3) + (12*sqrt(3*p/8) * p * (1+32*p^2)))



where

     X is the transmit rate in bits/second.

     s is the packet size in bytes.

     R is the round-trip time in seconds.

     p is the loss event rate, between 0.0 and 1.0, of the number of
     loss events as a fraction of the number of packets transmitted.

In future, different TCP equations may be substituted for this equation.
The requirement is that the throughput equation be a reasonable



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approximation of the sending rate of TCP for conformant TCP congestion
control.

The parameters s (packet size), p (loss event rate) and R (RTT) need to
be measured or calculated by a TFMCC implementation.  The measurement of
R is specified in Section 4.3.2, and the measurement of p is specified
in Section 5. The parameter s (packet size) is normally known to an
application.  This may not be so in two cases:

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 different way of measuring parameters.

Currently, TFMCC cannot be used for the second class of applications.


2.2.  Packet Contents

Before specifying the sender and receiver functionality, we describe the
congestion control information contained in packets sent by the sender
and feedback packets from the receivers.  Information from the sender
can either be sent in separate congestion control messages or
piggybacked onto data packets.  If separate congestion control messages
are sent at time intervals larger than the time interval between data
packets (e.g., once per feedback round), it is necessary to be able to
include timestamp information destined for more than one receiver to
allow a sufficient number of receivers to measure their RTT.

As TFMCC will be used along with a transport protocol, we do not specify
packet formats, since these depend on the details of the transport
protocol used.  The recommended representation of the header fields is
given below.  Alternatively, if the computational overhead of a floating
point representation is prohibitive, fixed point arithmetic can be used
at the expense of larger packet headers.


2.2.1.  Sender Packets

Each packet sent by the data sender contains the following information:





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o A sequence number i.  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.  In
  most cases the sequence number will be supplied by the transport
  protocol used along with TFMCC.

o A suppression rate X_supp in bits/s.  Only receivers with a calculated
  rate lower than the suppression rate are eligible to give feedback,
  unless their RTT is higher than the maximum RTT described below in
  which case they are also eligible to give feedback.  The suppression
  rate should be represented as a 12 bit floating point value with 5
  bits for the unsigned exponent and 7 bits for the unsigned mantissa
  (to represent rates from 100 bit/s to 400 Gbit/s with an error of less
  than 1%).

o A timestamp ts_i indicating when the packet is sent.  The resolution
  of the timestamp should typically be milliseconds and the timestamp
  should be an unsigned integer value no less than 16 bits wide.

o A receiver ID r and a copy of the timestamp ts_r of that receiver's
  last report, which allows the receiver to measure its RTT.  The
  receiver ID is described in the next section.  The resolution of the
  timestamp echo should be milliseconds and the timestamp should be an
  unsigned integer value no less than 16 bits wide.  If separate instead
  of piggybacked congestion control messages are used, the packet needs
  to contain a list of receiver IDs with corresponding timestamps to
  allow a sufficient number of receivers to simultaneously measure their
  RTT.  For the default values used for the feedback process this
  corresponds to a list size on the order of 10 to 20 entries.

o A flag is_CLR indicating whether the receiver with ID r is the CLR.

o A feedback round counter fb_nr.  This counter is incremented by the
  sender at the beginning of a new feedback round to notify the
  receivers than all feedback for older rounds should be suppressed.
  The feedback round counter should be at least 4 bits wide.

o A maximum RTT value R_max, representing the maximum of the RTTs of all
  receivers.  The RTT should be measured in milliseconds.  An 8 bit
  floating point value with 4 bits for the unsigned exponent and 4 bits
  for the unsigned mantissa (to represent RTTs from 1 millisecond to 64
  seconds with an error of ca. 6%) should be used for the
  representation.







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2.2.2.  Feedback Packets

Each feedback packet sent by a data receiver contains the following
information:

o A unique receiver ID r.  In most cases the receiver ID will be
  supplied by the transport protocol, but it may simply be the IP
  address of the receiver.

o A flag have_RTT indicating whether the receiver has made at least one
  RTT measurement since it joined the session.

o A flag have_loss indicating whether the receiver experienced at least
  one loss event since it joined the session.

o A flag receiver_leave indicating that the receiver will leave the
  session (and should therefore not be CLR).

o A timestamp ts_r indicating when the feedback packet is sent.  The
  representation of the timestamp should be the same as that of the
  timestamp echo in the data packets.

o An echo of the timestamp of the last data packet received.  If the
  last packet received at the receiver has sequence number i, then ts_i
  is included in the feedback.  If there is a delay t_d between
  receiving that last data packet and sending feedback, then ts_i' =
  ts_i + t_d is included in the feedback instead of ts_i. The
  representation of the timestamp echo should be the same as that of the
  timestamp in the data packets.

o A feedback round echo fb_nr, reflecting the highest feedback round
  counter value received so far.  The representation of the feedback
  round echo should be the same as the one used for the feedback round
  counter in the data packets.

o The desired sending rate X_r.  This is the rate calculated by the
  receiver to be TCP-friendly on the path from the sender to this
  receiver.  The representation of the desired sending rate should be
  the same as that of the suppression rate in the data packets.


3.  Data Sender Protocol

The data sender multicasts a stream of data packets to the data
receivers at a controlled rate.  Whenever feedback is received, the
sender checks if it is necessary to switch CLRs and to readjust the
sending rate.




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The main tasks that have to be provided by a TFMCC sender are:

o adjusting the sending rate,

o controlling receiver feedback, and

o assisting receiver-side RTT measurements.


3.1.  Sender Initialization

At initialization of the sender, the maximum RTT is set to a value that
should be larger than the highest RTT to any of the receivers.  It
should not be smaller than 500 milliseconds for operation in the public
Internet.  The sending rate X is initialized to 1 packet per maximum
RTT.


3.2.  Determining the Maximum RTT

For each feedback packet that arrives at the sender, the sender computes
the instantaneous RTT to the receiver as

     R_r = t_now - ts_i

where t_now is the time the feedback packet arrived.  Receivers will
have adjusted ts_i for the time interval between receiving the last data
packet and sending the corresponding report so that this interval will
not be included in R_r.  If the actual RTT is smaller than the
resolution of the timestamps and t_now equals ts_i, then R_r is set to
the smallest positive RTT value larger than 0 (i.e., 1 millisecond in
our case).  If the instantaneous RTT is larger than the current maximum
RTT, the maximum RTT is increased to that value

     R_max = R_r

Otherwise, if no feedback with a higher instantaneous RTT than the
maximum RTT is received during a feedback round (see Section 3.4), the
maximum RTT is reduced to

     R_max = MAX(R_max * 0.9, R_peak)

where R_peak is the peak receiver RTT measured during the feedback
round.

The maximum RTT is mainly used for feedback suppression among receivers
with heterogeneous RTTs. Feedback suppression is closely coupled to the
sending of data packets and for this reason, the maximum RTT must not



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decrease below the maximum time interval between consecutive data
packets:

     R_max = max(R_max, s/X + t_gran)

where t_gran is the granularity of the sender's system clock (see
Section 3.7).


3.3.  Adjusting the Sending Rate

When a feedback packet from receiver r arrives at the sender, the sender
has to check whether it is necessary to adjust the transmission rate and
to switch to a new CLR.

How the rate is adjusted depends on the desired rate X_r of the receiver
report.  We distinguish four cases:

1    If no CLR is present, receiver r becomes the current limiting
     receiver.  The sending rate X is directly set to X_r, so long as
     this would result in a rate increase of less than 8s/R_max bits/s
     (i.e., 1 packet per R_max).  Otherwise X is gradually increased to
     X_r at an increase rate of no more than 8s/R_max bits/s every R_max
     seconds.

2    If receiver r is not the CLR but a CLR is present, then receiver r
     becomes the current limiting receiver if X_r is less than the
     current sending rate X and the receiver_leave flag of that
     receiver's report is not set.  Furthermore, the sending rate is
     reduced to X_r.

3    If receiver r is not the CLR but a CLR is present and the
     receiver_leave flag of the CLR's last report was set, then receiver
     r becomes the current limiting receiver.  However, if X_r > X, the
     sending rate is not increased to X_r for the duration of a feedback
     round to allow other (lower rate) receivers to give feedback and be
     selected as CLR.

4    If receiver r is the CLR, the sending rate is set to the minimum of
     X_r and X + 8s/R_max bits/s.

If the receiver has not yet measured its RTT but already experienced
packet loss (indicated by the corresponding flags in the receiver
report), the receiver report will include a desired rate that is based
on the maximum RTT rather than the actual RTT to that receiver.  In this
case, the sender adjusts the desired rate using its measurement of the
instantaneous RTT R_r to that receiver:




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     X_r' = X_r * R_max / R_r

X_r' is then used instead of X_r to detect whether to switch to a new
CLR.

If the TFMCC sender receives no reports from the CLR for 4 RTTs, the
sending rate is cut in half unless the CLR was selected less than 10
RTTs ago.  In addition, if the sender receives no reports from the CLR
for at least 10 RTTs, it assumes that the CLR crashed or left the group.
A new CLR is selected from the feedback that subsequently arrives at the
sender, and we increase as in case 3 above.

If no new CLR can be selected (i.e., in the absence of any feedback from
any of the receivers) it is necessary to further reduce the sending
rate.  For every 10 consecutive RTTs without feedback, the sending rate
is cut in half.  The rate is at most reduced to one packet every 8
seconds.

Note that when receivers stop receiving data packets, they will stop
sending feedback.  This eventually causes the sending rate to be reduced
in the case of network failure.  If the network subsequently recovers, a
linear increase to the calculated rate of the CLR will occur at 8s/R_max
bits/s every R_max.


3.4.  Controlling Receiver Feedback

The receivers allowed to send a receiver report are determined in so-
called feedback rounds.  Feedback rounds have a duration T of six times
the maximum RTT.  In case the multicast model is ASM, i.e., receiver
feedback is multicast to the whole group, the duration of a feedback
round may be reduced to four times the maximum RTT.

Only receivers wishing to report a rate that is lower than the
suppression rate X_supp, or those with a higher RTT than R_max may send
feedback.  At the beginning of each feedback round, X_supp is set to the
highest possible value that can be represented.  When feedback arrives
at the sender over the course of a feedback round, X_supp is decreased
such that more and more feedback is suppressed towards the end of the
round.  How receiver feedback is spread out over the feedback round is
discussed in Section 4.5.

Whenever non-CLR feedback for the current round arrives at the sender,
X_supp is reduced to

     X_supp = (1-g) * X_r

if X_supp > X_r.  Feedback that causes the corresponding receiver to be



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selected as CLR, but was from a non-CLR receiver at the time of sending
also contributes to the feedback suppression.  Note that X_r must not be
adjusted by the sender to reflect the receiver's real RTT in case X_r
was calculated using the maximum RTT, as is done for setting the sending
rate (Section 3.3), otherwise a feedback implosion is possible.  The
parameter g determines to what extent higher rate feedback can suppress
lower rate feedback.  This mechanism guarantees, that the lowest
calculated rate reported lies within a factor of g of the actual lowest
calculated rate of the receiver set (see [9]). A value of g of 0.1 is
recommended.

To allow receivers to suppress their feedback, the sender's suppression
rate needs to be updated whenever feedback is received.  This
suppression rate has to be communicated to the receivers in a timely
manner, either by including it in the data packet header or, if separate
congestion control messages are used, by sending a message with the
suppression rate whenever the rate changes significantly (i.e., when it
is reduced to less than (1-g) times the previously advertised
suppression rate).

After a time span of T, the feedback round ends if non-CLR feedback was
received during that time.  Otherwise, the feedback round ends as soon
as the first non-CLR feedback message arrives at the sender but at most
after 2T.  The feedback round counter is incremented by one and the
suppression rate X_supp is reset to the highest representable value.
The feedback round counter restarts with round 0 after a wrap-around.


3.5.  Assisting Receiver-Side RTT Measurements

Receivers measure their RTT by sending a timestamp with a receiver
report, which is echoed by the sender.  If congestion control
information is piggybacked onto data packets, usually only one receiver
ID and timestamp can be included.  If multiple feedback messages from
different receivers arrive at the sender during the time interval
between two data packets, the sender has to decide which receiver to
allow to measure RTT.  The same applies if separate congestion control
messages allow to echo multiple receiver timestamps simultaneously but
the number of receivers that gave feedback since the last congestion
control message exceeds the list size.

The sender's timestamp echoes are prioritized in the following order:

1.   a new CLR (after a change of CLR's) or a CLR without any previous
     RTT measurements

2.   receivers without any previous RTT measurements in the order of the
     feedback round echo of the corresponding receiver report (i.e.,



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     older feedback first)

3.   non-CLR receivers with previous RTT measurements, again in
     ascending order of the feedback round echo of the report

4.   the CLR

Ties are broken in favor of the receiver with the lowest reported rate.

It is necessary to account for the time that elapses between receiving a
report and sending the next data packet.  This time needs to be deducted
from the RTT and thus has to be added to the receiver's timestamp value.

Whenever no feedback packets arrive in the interval between two data
packets, the CLR's last timestamp, adjusted by the appropriate offset,
is echoed.  When the number of packets per RTT is so low that all
packets carry a non-CLR receiver's timestamp, the CLR's timestamp and ID
are included in a data packet at least once per feedback round.


3.6.  Slowstart

TFMCC uses a slowstart mechanism to quickly approach its fair bandwidth
share at the start of a session.  During slowstart, the sending rate
increases exponentially.  The rate increase is limited to the minimum of
the rates included in the receiver reports and receivers report twice
the rate at which they currently receive data.  As in normal congestion
control mode, the receiver with the smallest reported rate becomes CLR.
Since a receiver can never receive data at a rate higher than its link
bandwidth, this effectively limits the overshoot to twice this
bandwidth.  In case the resulting increase over R_max is less than
8s/R_max bits/s, the sender may choose to increase the rate by up to
8s/R_max bits/s every R_max.  The current sending rate is gradually
adjusted to the target rate reported in the receiver reports over the
course of a RTT.  Slowstart is terminated as soon as any one of the
receivers experiences its first packet loss.  Since that receiver's
calculated rate will be lower than the current sending rate, the
receiver will be selected as CLR.

During slowstart, the upper bound on the rate increase of 8s/R_max
bits/s every RTT does not apply.  Only after the TFMCC sender receives
the first report with the have_loss flag set is the rate increase
limited in this way.








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3.7.  Scheduling of Packet Transmissions

As TFMCC 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

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 (see below for delta).

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 TFMCC 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 10 milliseconds on many Unix systems.  If t_gran is not known,
a value of 10 milliseconds can be safely assumed.


4.  Data Receiver Protocol

Receivers measure the current network conditions, namely RTT and loss
event rate, and use this information to calculate a rate that is fair to
competing traffic.  The rate is then communicated to the sender in
receiver reports.  Due to the potentially large number of receivers, it
is undesirable that all receivers send reports, especially not at the
same time.



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In the description of the receiver functionality, we will first address
how the receivers measure the network parameters and then discuss the
feedback process.


4.1.  Receiver Initialization

The receiver is initialized when it receives the first data packet.  The
RTT is set to the maximum RTT value contained in the data packet.  This
initial value is used as the receiver's RTT until the first real RTT
measurement is made.  The loss event rate is initialized to 0.


4.2.  Receiver Leave

A receiver that sends feedback but wishes to leave the TFMCC session
within the next feedback round may indicate the pending leave by setting
the receiver_leave flag in its report.  If the leaving receiver is the
CLR, the receiver_leave flag should be set for all the reports within
the feedback round before the leave takes effect.


4.3.  Measurement of the Network Conditions

Receivers have to update their estimate of the network parameters with
each new data packet they receive.


4.3.1.  Updating the Loss Event Rate

When a data packet is received, the receiver adds the packet to the
packet history.  It then recalculates the new value of the loss event
rate p.  The loss event rate measurement mechanism is described
separately in Section 5.


4.3.2.  Basic Round-Trip Time Measurement

When a receiver gets a data packet that carries the receiver's own ID in
the r field, the receiver updates its RTT estimate.

1.   The current RTT is calculated as:

          R_sample = t_now - ts_r

     where t_now is the time the data packet arrives at the receiver and
     ts_r is the receiver report timestamp echoed in the data packet. If
     the actual RTT is smaller than the resolution of the timestamps and



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     t_now equals ts_r, then R_sample is set to the smallest positive
     RTT value larger than 0 (i.e., 1 millisecond in our case).

2.   The smoothed RTT estimate R is updated:

          If no feedback has been received before
              R = R_sample
          Else
              R = q*R + (1-q)*R_sample

     A filter parameter q of 0.5 is recommended for non-CLR receivers.
     The CLR performs RTT measurements much more frequently and hence
     should use a higher filter value.  We recommend using q=0.9.  Note
     that TFMCC is not sensitive to the precise value for the filter
     constant.

Optionally, sender-based instead of receiver-based RTT measurements may
be used.  The sender already determines the RTT to a receiver from the
receiver's echo of the sender's own timestamp for the calculation of the
maximum RTT.  For sender-based RTT measurements, this RTT measurement
needs to be communicated to the receiver.  Instead of including an echo
of the receiver's timestamp, the sender includes the receiver's RTT in
the next data packet, using the prioritization rules described in
Section 3.5.

To simplify sender operation, smoothing of RTT samples as described
above should still be done at the receiver.


4.3.3.  One-Way Delay Adjustments

When a RTT measurement is performed, the receiver also determines the
one-way delay D_r from itself to the sender:

     D_r = ts_r - ts_i

where ts_i and ts_r are the timestamp and timestamp echo contained in
the data packet.  With each new data packet i', the receiver can now
calculate an updated RTT estimate as:

     R' = max(D_r + t_now - ts_i', 1 millisecond)

In between RTT measurements, the updated R' is used instead of the
smoothed RTT R. Like the RTT samples, R' must be strictly positive. When
a new measurement is made, all interim one-way delay measurements are
discarded (i.e., the smoothed RTT is updated according to Section 4.3.2
without taking one-way delay adjustments into account).




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For the one-way delay measurements, the clocks of sender and receivers
need not be synchronized.  Clock skew will cancel itself out when both
one-way measurements are added to form a RTT estimate, as long as clock
drift between real RTT measurements is negligible.

The same one-way delay adjustments should be applied to the RTT supplied
by the sender when using sender-based RTT measurements.


4.3.4.  Receive Rate Measurements

When a receiver has not experienced any loss events, it cannot calculate
a TCP-friendly rate to include in the receiver reports.  Instead, the
receiver measures the current receive rate and sets the desired rate X_r
to twice the receive rate.

The receive rate in bits/s is measured as the number of bits received
over the last k RTTs, taking into account the IP and transport packet
headers, but excluding the link-layer packet headers. A value for k
between 2 and 4 is recommended.


4.4.  Setting the Desired Rate

When a receiver measures a non-zero loss event rate, it calculates the
desired rate using Equation (1).  In case no RTT measurement is
available yet, the maximum RTT is used instead of the receiver's RTT.
The desired rate X_r is updated whenever the loss event rate or the RTT
changes.

A receiver may decide to not report desired rates that are below 1
packet per 8 seconds, since a sender is very slow to recover from such
low sending rates. In this case, the receiver reports a desired rate of
1 packet per 8 seconds. However, it must leave the multicast group if
for more than 120 seconds the calculated rate falls below the reported
rate and the current sending rate is higher than the receiver's
calculated rate.

As mentioned above, calculation of the desired rate is not possible
before the receiver experiences the first loss event and in that case
twice the rate at which data is received is included in the receiver
reports as X_r.  This mechanism allows the sender to slowstart as
described in Section 3.6.








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4.5.  Feedback and Feedback Suppression

Let fb_nr be the highest feedback round counter value received by a
receiver.  When a new data packet arrives with a higher feedback round
counter than fb_nr, a new feedback round begins and fb_nr is updated.
Outstanding feedback for the old round is canceled.  In case a feedback
number with a value that is more than half the feedback number space
lower than fb_nr is received, the receiver assumes that the feedback
round counter wrapped and also cancels the feedback timer and updates
fb_nr.

The CLR sends its feedback independently from all the other receivers
once per RTT. Its feedback does not suppress other feedback and cannot
be suppressed by other receiver's feedback.

Non-CLR receivers set a feedback timer at the beginning of a feedback
round.  Using an exponentially weighted random timer mechanism, the
feedback timer is set to expire after

     t = max(T * (1 + log(x)/log(N)), 0)

where

     x is a random variable uniformly distributed in (0,1],

     T is the duration of a feedback round (i.e., 6 * R_max),

     N is an estimated upper bound on the number of receivers.

N is a constant specific to the TFMCC protocol.  Since TFMCC scales to
up to thousands of receivers, setting N to 10,000 for all receivers (and
limiting the TFMCC session to at most 10,000 receivers) is recommended.

A feedback packet is sent when the feedback timer expires, unless the
timer is canceled beforehand.  When the multicast model is ASM, feedback
is multicast to the whole group, otherwise the feedback is unicast to
the sender.  The feedback packet includes the calculated rate valid at
the time the feedback packet is sent (not the rate at the point of time
when the feedback timer is set).  The copy of the timestamp ts_i of the
last data packet received, which is included in the feedback packet,
needs to be adjusted by the time interval between receiving the data
packet and sending the report to allow the sender to correctly infer the
instantaneous RTT (i.e., that time interval has to be added to the
timestamp value).

The timer is canceled if a data packet with a lower suppression rate
than the receiver's calculated rate and a higher or equal maximum RTT
than the receiver's RTT is received.  Likewise, a data packet indicating



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the beginning of a new feedback round cancels all feedback for older
rounds.  In case of ASM, the timer is also canceled if a feedback packet
from another non-CLR receiver reporting a lower rate is received.

The feedback suppression process is complicated by the fact that the
calculated rates of the receivers will change during a feedback round.
If the calculated rates decrease rapidly for all receivers, feedback
suppression can no longer prevent a feedback implosion since earlier
feedback will always report a higher rate than current feedback. To make
the feedback suppression mechanism robust in the face of changing rates,
it is necessary to introduce X_fbr, the calculated rate of a receiver at
the beginning of a feedback round. A receiver needs to suppress its
feedback not only when the suppression rate is less than the receiver's
current calculated rate but also in the case that the suppression rate
falls below X_fbr.

When the maximum RTT changes significantly during one feedback round, it
is necessary to reschedule the feedback timer in proportion to the
change.

     t = t * R_max / R_max'

where R_max is the new maximum RTT and R_max' is the previous maximum
RTT.  The same considerations hold, when the last data packets were
received more than a time interval of R_max ago.  In this case, it is
necessary to add the difference of the inter-packet gap and the maximum
RTT to the feedback time to prevent a feedback implosion (e.g., in case
the sender crashed).

     t = t + max(t_now - tr_i - R_max, 0)

where tr_i is the time when the last data packet arrived at the
receiver.

More details on the characteristics of the feedback suppression
mechanism can be found in [9] and [10].


5.  Calculation of the Loss Event Rate

Obtaining an accurate and stable measurement of the loss event rate is
of primary importance for TFMCC.  Loss rate measurement is performed at
the receiver, based on the detection of lost or marked packets from the
sequence numbers of arriving packets.







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5.1.  Detection of Lost or Marked Packets

TFMCC 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 receivers each maintain 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 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 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 TFMCC more robust in the presence of reordering.  In contrast to
TCP, if a packet arrives late (after 3 subsequent packets arrived) at a
receiver, the late packet can fill the hole in the reception record, and
the receiver can recalculate the loss event rate.  Future versions of
TFMCC 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

TFMCC requires that the loss event rate be robust to several consecutive
packets lost 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 receivers needs to
map the packet loss history into a loss event record, where a loss event
is one or more packets lost in a RTT.

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



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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
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 the 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 of 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).




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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 TFMCC's speed in responding to changes in the
level of congestion.  As currently specified, TFMCC 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 TFMCC'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.

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:















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     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 4/(3*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 that allows
the TFMCC receivers 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, 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.

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



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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
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, a 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
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.



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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 TFMCC to response somewhat more quickly to the
sudden absence of congestion, as represented by a long current loss
interval.


5.6.  Initializing the Loss History after the First Loss Event

The number of packets received before the first loss event usually does
not reflect the current loss event rate.  When the first loss event
occurs, a TFMCC receiver assumes that the correct data rate is the rate
at which data was received during the last RTT when the loss occurred.
Instead of initializing the first loss interval to the number of packets
sent until the first loss event, the TFMCC receiver calculates the loss
interval that would be required to produce the receive rate X_recv, and
uses this synthetic loss interval l_0 to seed the loss history
mechanism.

The initial loss interval is calculated by inverting a simplified
version of the TCP Equation (1).












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                                  s
     X_recv = sqrt(3/2) * -----------------
                           R * sqrt(1/l_0)

                  X_recv * R
     ==> l_0 = (---------------)^2
                 sqrt(3/2) * s



The resulting initial loss interval is too small at higher loss rates
compared to using the more accurate Equation (1), which leads to a more
conservative initial loss event rate.

If a receiver still uses the initial RTT R_max instead of its real RTT,
the initial loss interval is too large in case the initial RTT is higher
than the actual RTT.  As a consequence, the receiver will calculate a
too high desired rate when the first RTT measurement R is made and the
initial loss interval is still in the loss history.  The receiver has to
adjust l_0 as follows:


     l_0 = l_0 * (R/R_max)^2



No action needs to be taken when the first RTT measurement is made after
the initial loss interval left the loss history.


6.  Security Considerations

TFMCC 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 TFMCC should take care to ensure that
feedback is only accepted from valid receivers of the data.  The precise
mechanism to achieve this will however depend on the transport protocol
itself.

Congestion control mechanisms may potentially be manipulated by a greedy
receiver that wishes to receive more than its fair share of network
bandwidth.  However, in TFMCC a receiver can only influence the sending



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rate if it is the CLR and thus has the lowest calculated rate of all
receivers.  If the calculated rate is then manipulated such that it
exceeds the calculated rate of the second to lowest receiver, it will
cease to be CLR.  A greedy receiver can only significantly increase then
transmission rate if it is the only participant in the session.  If such
scenarios are of concern, 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.

It is possible that a receiver sends feedback claiming it has a very low
calculated rate.  This will reduce the rate of the multicast session and
might render it useless but obviously cannot hurt the network itself.

We expect that protocols incorporating ECN with TFMCC will also want to
incorporate feedback from the receiver to the sender using the ECN nonce
[8]. 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.


7.  IANA Considerations

There are no IANA actions required for this document.


8.  Authors' Addresses

     Joerg Widmer
     DoCoMo Euro-Labs
     Landsberger Str. 312, Munich, Germany
     widmer@acm.org


     Mark Handley
     UCL (University College London)
     Gower Street, London WC1E 6BT, UK
     m.handley@cs.ucl.ac.uk



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 particularly
like to thank Brian Adamson, Mark Pullen, and Fei Zhao for feedback on
earlier versions of this document.


10.  References


[1] B. Adamson, C. Bormann, M. Handley, and J. Macker, "NACK-Oriented
     Reliable Multicast (NORM) Protocol Building Blocks", Internet Draft
     draft-ietf-rmt-norm-bb-09.txt, July 2004, work in progress.
     Citation for informational purposes only.

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

[3] H. W. Holbrook, "A Channel Model for Multicast," Ph.D.
     Dissertation, Stanford University, Department of Computer Science,
     Stanford, California, August 2001.

[4] J. Padhye, V. Firoiu, D. Towsley, and J. Kurose, "Modeling TCP
     Throughput: A Simple Model and its Empirical Validation", Proc ACM
     SIGCOMM 1998.

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

[6] L. Rizzo, "pgmcc: a TCP-friendly single-rate multicast congestion
     control scheme", Proc ACM SIGCOMM 2000, Stockholm, August 2000

[7] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: A
     Transport Protocol for Real-Time Applications", RFC 1889, January
     1996.

[8] N. Spring, D. Wetherall, and D. Ely, "Robust Explicit Congestion
     Notification (ECN) Signaling with Nonces", RFC 3540, June 2003.

[9] J. Widmer and T. Fuhrmann, "Extremum Feedback for Very Large
     Multicast Groups", Proc NGC 2001, London, November 2001.

[10] J. Widmer and M. Handley, "Extending Equation-Based Congestion
     Control to Multicast Applications", Proc ACM SIGCOMM 2001, San
     Diego, August 2001







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11.  Copyright Notice

Copyright (C) The Internet Society (2005).  This document is subject to
the rights, licenses and restrictions contained in BCP 78, and except as
set forth therein, the authors retain all their rights.

This document and the information contained herein are provided on an
"AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS OR
IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE INTERNET
ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED,
INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE
INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.






































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