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RFC 4654

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

Pages: 32

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Network Working Group                                          J. Widmer
Request for Comments: 4654                              DoCoMo Euro-Labs
Category: Experimental                                        M. Handley
                                                             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).


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.
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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
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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.
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   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).
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   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

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.
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   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)))
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      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

   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.
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   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.
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   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.
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   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
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   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

   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.
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   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

   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.
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   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
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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.
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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:
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      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.
Top   ToC   RFC4654 - Page 17

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).
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   2.  The smoothed RTT estimate R is updated:

       If no feedback has been received before
           R = R_sample

           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).
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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 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.
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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).
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   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

      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

   More details on the characteristics of the feedback suppression
   mechanism can be found in [13] and [3].
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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.
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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.
Top   ToC   RFC4654 - Page 24
   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.
Top   ToC   RFC4654 - Page 25
   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.
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   As described in Section 5.4, the average loss interval is calculated
   using the n previous loss intervals I_1, ..., I_n, and the interval
   I_0 that represents the number of packets received since the last
   loss event.  The computation of the average loss interval using the
   discount factors is a simple modification of the procedure in Section
   5.4, as 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;
Top   ToC   RFC4654 - Page 27
   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

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).
Top   ToC   RFC4654 - Page 28
      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,
Top   ToC   RFC4654 - Page 29
   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.
Top   ToC   RFC4654 - Page 30
   [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.
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Authors' Addresses

Joerg Widmer DoCoMo Euro-Labs Landsberger Str. 312, Munich, Germany EMail: Mark Handley UCL (University College London) Gower Street, London WC1E 6BT, UK EMail:
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