| Rfc | 4654 |
|---|
| Title | TCP-Friendly Multicast Congestion Control (TFMCC): Protocol
Specification |
|---|
| Author | J. Widmer, M. Handley |
|---|
| Date | August 2006 |
|---|
| Format: | TXT,
HTML |
|---|
| Status: | EXPERIMENTAL |
|---|
|
Network Working Group J. Widmer
Request for Comments: 4654 DoCoMo Euro-Labs
Category: Experimental M. Handley
UCL
August 2006
TCP-Friendly Multicast Congestion Control (TFMCC):
Protocol Specification
Status of This Memo
This memo defines an Experimental Protocol for the Internet
community. It does not specify an Internet standard of any kind.
Discussion and suggestions for improvement are requested.
Distribution of this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2006).
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. 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 where a relatively smooth sending rate is
of importance, such as streaming media.
Table of Contents
1. Introduction ....................................................3
1.1. Related Documents ..........................................4
1.2. Environmental Requirements and Considerations ..............4
2. Protocol Overview ...............................................5
2.1. TCP Throughput Equation ....................................6
2.2. Packet Contents ............................................7
2.2.1. Sender Packets ......................................8
2.2.2. Feedback Packets ....................................9
3. Data Sender Protocol ...........................................10
3.1. Sender Initialization .....................................10
3.2. Determining the Maximum RTT ...............................10
3.3. Adjusting the Sending Rate ................................11
3.4. Controlling Receiver Feedback .............................12
3.5. Assisting Receiver-Side RTT Measurements ..................14
3.6. Slowstart .................................................15
3.7. Scheduling of Packet Transmissions ........................15
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 .............................22
5.1. Detection of Lost or Marked Packets .......................22
5.2. Translation from Loss History to Loss Events ..............23
5.3. Inter-Loss Event Interval .................................24
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. Acknowledgments ................................................29
8. References .....................................................29
8.1. Normative References ......................................29
8.2. Informative References ....................................29
1. Introduction
This document specifies TCP-Friendly Multicast Congestion Control
(TFMCC) [3]. TFMCC is a source-based, single-rate congestion control
scheme that builds upon the unicast TCP-Friendly Rate Control
mechanism (TFRC) [4]. 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 [1]. 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 [11], 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 where a relatively smooth sending rate is
of importance, such as streaming media. 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 [10]
may be more suitable.
This memo contains part of the definitions necessary to fully specify
a Reliable Multicast Transport protocol in accordance with RFC 2357.
As per RFC 2357, the use of any reliable multicast protocol in the
Internet requires an adequate congestion control scheme. This
document specifies an experimental congestion control scheme. While
waiting for initial deployment and experience to show this scheme to
be effective and scalable, the IETF publishes this scheme in the
"Experimental" category.
It is the intent of the Reliable Multicast Transport (RMT) Working
Group to re-submit the specification as an IETF Proposed Standard as
soon as the scheme is deemed adequate.
1.1. Related Documents
As described in RFC 3048 [1], TFMCC is a building block that is
intended to be used, in conjunction with other building blocks, to
help specify a protocol instantiation. It follows the general
guidelines provided in RFC 3269 [2]. In particular, TFMCC is a
suitable congestion control building block for NACK-Oriented Reliable
Multicast (NORM) [5].
1.2. 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 on 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 [6] and the Source-Specific Multicast (SSM) model as
defined in [7]. TFMCC works with both multicast models, but in a
slightly different way. When ASM is used, 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 that allow bi-
directional communication, 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, whereas TCP 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 applications (e.g., those using audio) 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) [9]. 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. 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 to 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 [8]:
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 the future, different TCP equations may be substituted for this
equation. The requirement is that the throughput equation be a
reasonable approximation of the sending rate of TCP for conformant
TCP congestion control.
The 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.
Sender and receivers of a specific TFMCC instance need to agree on a
common encoding for the header fields.
2.2.1. Sender Packets
Each packet sent by the data sender contains the following
information:
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 tr_r' = tr_r of that
receiver's last report, which allows the receiver to measure its
RTT. If there is a delay ts_d between receiving the report from
receiver r and sending the data packet, then tr_r' = tr_r + ts_d is
included in the packet instead. 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 congestion control messages
are used instead of piggybacked ones, 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 that 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.
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 tr_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 ts_i' of the timestamp of the last data packet received.
If the last packet received at the receiver has sequence number i,
then ts_i' = ts_i is included in the feedback. If there is a delay
tr_d between receiving that last data packet and sending feedback,
then ts_i' = ts_i + tr_d is included in the feedback instead. 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.
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 = ts_now - ts_i'
where ts_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 ts_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 decrease below the maximum time interval between
consecutive data packets:
R_max = max(R_max, 8s/X + ts_gran)
where ts_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:
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 reduce the sending rate
further. 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.
An application using TFMCC may have a minimum sending rate
requirement, where the application becomes unusable if the sending
rate continuously falls below this minimum rate. The application
should exclude receivers that report such a low rate from the
multicast group. The specific mechanism to do this is application
dependent and beyond the scope of this document.
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 selected as CLR, but that 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 [13]). 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 the RTT. The same applies if
separate congestion control messages allow echoing 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., 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 an 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.
Slowstart may also be used after the sender has been idle for some
time, to quickly reach the previous sending rate. When the sender
stops sending data packets, it records the current sending rate X' =
X. Every 10 RTTs, the allowed sending rate will be halved due to
lack of receiver feedback, as specified in Section 3.3. This halving
may take place multiple times. When the sender resumes, it may
perform a slowstart from the current allowed rate up to the recorded
rate X'. Slowstart ends after the first packet loss by any of the
receivers or as soon as X' is reached.
To this end, receivers have to clear the have_loss flag after 10 RTTs
without data packets as specified in Section 4.3.1. The have_loss
flag is only used during slowstart. Therefore, clearing the flag has
no effect if no packets arrived due to network partitioning or packet
loss.
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, ts_ipi, as follows:
ts_ipi = 8s/X
When a sender first starts sending at time t_0, it calculates ts_ipi
and calculates a nominal send time, t_1 = t_0 + ts_ipi, for packet 1.
When the application becomes idle, it checks the current time,
ts_now, and then requests re-scheduling after (ts_ipi - (ts_now -
t_0)) seconds. When the application is re-scheduled, it checks the
current time, ts_now, again. If (ts_now > t_1 - delta) then packet 1
is sent (see below for delta).
Now, a new ts_ipi may be calculated and used to calculate a nominal
send time, t_2, for packet 2: t_2 = t_1 + ts_ipi. The process then
repeats with each successive packet's send time being calculated from
the nominal send time of the previous packet. Note that the actual
send time ts_i, and not the nominal send time, is included as
timestamp in the packet header.
In some cases, when the nominal send time, t_i, of the next packet is
calculated, it may already be the case that ts_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 ts_gran seconds, then delta would typically be set to:
delta = min(ts_ipi/2, ts_gran/2)
ts_gran is 10 milliseconds on many Unix systems. If ts_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.
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.
Also, the flags receiver_leave, have_RTT, and have_loss are cleared.
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.
When a loss event is detected, the flag have_loss is set. In case no
data packets are received for 10 consecutive RTTs, the flag is
cleared to allow the sender to slowstart. It is set again when new
data packets arrive and a loss event is detected.
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 = tr_now - tr_r'
where tr_now is the time the data packet arrives at the receiver
and tr_r' is the receiver report timestamp echoed in the data
packet. If the actual RTT is smaller than the resolution of the
timestamps and tr_now equals tr_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 RTT measurements may be used instead of
receiver-based ones. 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 an RTT measurement is performed, the receiver also determines
the one-way delay D_r from itself to the sender:
D_r = tr_r' - ts_i
where ts_i and tr_r' are the timestamp and receiver report timestamp
echo contained in the data packet. With each new data packet j, the
receiver can now calculate an updated RTT estimate as:
R' = max(D_r + tr_now - ts_j, 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 the interim one-way delay
adjustments into account).
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 an 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 not to 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. In that case,
twice the rate at which data is received is included in the receiver
reports as X_r to allow the sender to slowstart as described in
Section 3.6. This is also done when the sender resumes sending data
packets after the have_loss flag was cleared due to the sender being
idle.
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 is received that has a lower
suppression rate than the receiver's calculated rate and a higher or
equal maximum RTT than the receiver's RTT. Likewise, a data packet
indicating 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 is received from another non-CLR receiver reporting a
lower rate.
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(tr_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 [13] and [3].
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.
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
it 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 need 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 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 be either 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,
the sequence numbers must be modified to take this into account
before the calculation is performed. 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 if 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).
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:
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 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.
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 respond 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 it 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).
8s
X_recv = sqrt(3/2) * -----------------
R * sqrt(1/l_0)
X_recv * R
==> l_0 = (----------------)^2
sqrt(3/2) * 8s
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 too high a 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.
However, the precise mechanism to achieve this will 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 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 the 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 that 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 [12]. 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. Acknowledgments
We would like to acknowledge feedback and discussions on equation-
based congestion control with a wide range of people, including
members of the Reliable Multicast Research Group, the Reliable
Multicast Transport Working Group, and the End-to-End Research Group.
We would particularly like to thank Brian Adamson, Mark Pullen, Fei
Zhao, and Magnus Westerlund for feedback on earlier versions of this
document.
8. References
8.1. Normative References
[1] Whetten, B., Vicisano, L., Kermode, R., Handley, M., Floyd, S.,
and M. Luby, "Reliable Multicast Transport Building Blocks for
One-to-Many Bulk-Data Transfer", RFC 3048, January 2001.
[2] Kermode, R. and L. Vicisano, "Author Guidelines for Reliable
Multicast Transport (RMT) Building Blocks and Protocol
Instantiation documents", RFC 3269, April 2002.
8.2. Informative References
[3] J. Widmer and M. Handley, "Extending Equation-Based Congestion
Control to Multicast Applications", Proc ACM Sigcomm 2001, San
Diego, August 2001.
[4] S. Floyd, M. Handley, J. Padhye, and J. Widmer, "Equation-Based
Congestion Control for Unicast Applications", Proc ACM SIGCOMM
2000, Stockholm, August 2000.
[5] Adamson, B., Bormann, C., Handley, M., and J. Macker,
"Negative-Acknowledgment (NACK)-Oriented Reliable Multicast
(NORM) Building Blocks", RFC 3941, November 2004.
[6] Deering, S., "Host extensions for IP multicasting", STD 5, RFC
1112, August 1989.
[7] H. W. Holbrook, "A Channel Model for Multicast," Ph.D.
Dissertation, Stanford University, Department of Computer
Science, Stanford, California, August 2001.
[8] J. Padhye, V. Firoiu, D. Towsley, and J. Kurose, "Modeling TCP
Throughput: A Simple Model and its Empirical Validation", Proc
ACM SIGCOMM 1998.
[9] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition of
Explicit Congestion Notification (ECN) to IP", RFC 3168,
September 2001.
[10] L. Rizzo, "pgmcc: a TCP-friendly single-rate multicast
congestion control scheme", Proc ACM Sigcomm 2000, Stockholm,
August 2000.
[11] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications", STD 64,
RFC 3550, July 2003.
[12] Spring, N., Wetherall, D., and D. Ely, "Robust Explicit
Congestion Notification (ECN) Signaling with Nonces", RFC 3540,
June 2003.
[13] J. Widmer and T. Fuhrmann, "Extremum Feedback for Very Large
Multicast Groups", Proc NGC 2001, London, November 2001.
Authors' Addresses
Joerg Widmer
DoCoMo Euro-Labs
Landsberger Str. 312, Munich, Germany
EMail: widmer@acm.org
Mark Handley
UCL (University College London)
Gower Street, London WC1E 6BT, UK
EMail: m.handley@cs.ucl.ac.uk
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Acknowledgement
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