Internet Engineering Task Force                              Sally Floyd
INTERNET-DRAFT                                                      ICIR
draft-ietf-dccp-tfrc-voip-02.txt                            Eddie Kohler
Expires: January 2006                                               UCLA
                                                            19 July 2005


              TCP Friendly Rate Control (TFRC) for Voice:
                              VoIP Variant



Status of this Memo

    This document is an Internet-Draft and is subject to all provisions
    of section 3 of RFC 3667.  By submitting this Internet-Draft, each
    author represents that any applicable patent or other IPR claims of
    which he or she is aware have been or will be disclosed, and any of
    which he or she becomes aware will be disclosed, in accordance with
    Section 6 of BCP 79.

    Internet-Drafts are working documents of the Internet Engineering
    Task Force (IETF), its areas, and its working groups.  Note that
    other groups may also distribute working documents as Internet-
    Drafts.

    Internet-Drafts are draft documents valid for a maximum of six
    months and may be updated, replaced, or obsoleted by other documents
    at any time.  It is inappropriate to use Internet-Drafts as
    reference material or to cite them other than as "work in progress."

    The list of current Internet-Drafts can be accessed at
    http://www.ietf.org/ietf/1id-abstracts.txt.

    The list of Internet-Draft Shadow Directories can be accessed at
    http://www.ietf.org/shadow.html.

    This Internet-Draft will expire on January 2006.

Copyright Notice

    Copyright (C) The Internet Society (2005). All Rights Reserved.






Floyd/Kohler                                                    [Page 1]


INTERNET-DRAFT            Expires: January 2006                July 2005


Abstract

    TCP-Friendly Rate Control (TFRC) is a congestion control mechanism
    for unicast flows operating in a best-effort Internet environment
    [RFC 3448]. This document proposes a VoIP variant to TFRC.  TFRC was
    intended for applications that use a fixed packet size, and was
    designed to be reasonably fair when competing for bandwidth with TCP
    connections using the same packet size.  The VoIP variant of TFRC is
    designed for applications that send small packets, where the design
    goal is to achieve the same bandwidth in bps as a TCP flow using
    1500-byte data packets.  This variant is referred to in RFC 3448 as
    TFRC-PS, for applications that might vary their packet size in
    response to congestion.  The VoIP variant of TFRC enforces a Min
    Interval of 10 ms between data packets, to prevent a single flow
    from sending small packets arbitrarily frequently.

    Flows using the VoIP variant of TFRC compete reasonably fairly with
    large-packet TCP and TFRC flows in environments where large-packet
    flows and small-packet flows experience similar packet drop rates.
    However, in environments where small-packet flows experience lower
    packet drop rates than large-packet flows (e.g., with DropTail
    queues in units of bytes), the current VoIP variant of TFRC can
    receive considerably more than its share of the bandwidth.  (We note
    however that in all scenarios the VoIP variant of TFRC is better, in
    terms of congestion in the network, than the same application in the
    absence of congestion control).

























Floyd/Kohler                                                    [Page 2]


INTERNET-DRAFT            Expires: January 2006                July 2005


    TO BE DELETED BY THE RFC EDITOR UPON PUBLICATION:
     Changes from draft-ietf-dccp-tfrc-voip-01.txt
      * Added modified algorithm for calculating the loss event rate,
          for short intervals with multiple packet drops.
      * Moved Faster Restart to a separate document.
      * Added simulations with a configured byte drop rate.
      * Added many more simulations, including DropTail with a queue
        in bytes.
      * Added a discussion of unfairness for DropTail with a queue
        in bytes.
     Changes from draft-ietf-dccp-tfrc-voip-00.txt
      * Added more simulations.
      * Added a Related Work section.






































Floyd/Kohler                                                    [Page 3]


INTERNET-DRAFT            Expires: January 2006                July 2005


                             Table of Contents

    1. Conventions . . . . . . . . . . . . . . . . . . . . . . . . .   5
    2. VoIP Variant Introduction . . . . . . . . . . . . . . . . . .   5
    3. VoIP Variant Congestion Control . . . . . . . . . . . . . . .   7
    4. VoIP Variant Discussion . . . . . . . . . . . . . . . . . . .   8
       4.1. The TCP Throughput Equation. . . . . . . . . . . . . . .   8
       4.2. Accounting for Header Size . . . . . . . . . . . . . . .   9
       4.3. The VoIP Min Interval. . . . . . . . . . . . . . . . . .   9
       4.4. Counting Packet Losses . . . . . . . . . . . . . . . . .  11
    5. A Comparison with RFC 3714. . . . . . . . . . . . . . . . . .  11
    6. The VoIP Variant with Applications that Modify the
    Packet Size. . . . . . . . . . . . . . . . . . . . . . . . . . .  11
    7. Simulation Results. . . . . . . . . . . . . . . . . . . . . .  12
       7.1. Simulations with Configured Drop Rates . . . . . . . . .  12
       7.2. Packet Dropping Behavior at Routers with Drop-
       Tail Queues . . . . . . . . . . . . . . . . . . . . . . . . .  17
       7.3. Packet Dropping Behavior at Routers with AQM . . . . . .  19
    8. Discussion. . . . . . . . . . . . . . . . . . . . . . . . . .  22
    9. Implementation Issues . . . . . . . . . . . . . . . . . . . .  23
    10. Security Considerations. . . . . . . . . . . . . . . . . . .  23
    11. IANA Considerations. . . . . . . . . . . . . . . . . . . . .  23
    12. Thanks . . . . . . . . . . . . . . . . . . . . . . . . . . .  23
    Normative References . . . . . . . . . . . . . . . . . . . . . .  23
    Informative References . . . . . . . . . . . . . . . . . . . . .  23
    Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . .  24
    13. Related Work on VoIP Variants of TFRC. . . . . . . . . . . .  25
    14. Simulation Results with RED Queue Management . . . . . . . .  26
    15. A Discussion of Packet Size and Packet Dropping. . . . . . .  27
    Full Copyright Statement . . . . . . . . . . . . . . . . . . . .  28
    Intellectual Property. . . . . . . . . . . . . . . . . . . . . .  28




















Floyd/Kohler                                                    [Page 4]


INTERNET-DRAFT            Expires: January 2006                July 2005


1.  Conventions

    The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
    "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
    document are to be interpreted as described in [RFC 2119].

2.  VoIP Variant Introduction

    This document specifies a VoIP variant for TCP-friendly rate control
    (TFRC) [RFC 3448].  TFRC was designed to be reasonably fair when
    competing for bandwidth with TCP flows, but to avoid the abrupt
    changes in the sending rate characteristic of TCP's congestion
    control mechanisms.  TFRC is intended for applications such as
    streaming media applications where a relatively smooth sending rate
    is of importance.

    The VoIP variant is intended for flows that need to send frequent
    small packets (limited by a minimum interval between packets of 10
    ms).  Conventional TFRC measures loss rates by estimating the loss
    event ratio as described in [RFC 3448], and uses this loss event
    rate to determine the sending rate in packets per round-trip time.
    This has consequences for the rate a TFRC flow can achieve when
    sharing a bottleneck with large-packet TCP flows.  In particular, a
    low-bandwidth, small-packet TFRC flow sharing a bottleneck with
    high-bandwidth, large-packet TCP flows may be forced to slow down,
    even though the application's nominal rate in bytes per second is
    less than the rate achieved by the TCP flows.  Intuitively, this
    would be "fair" only if the network limitation was in packets per
    second (such as a routing lookup), rather than bytes per second
    (such as link bandwidth).  Conventional wisdom is that many of the
    network limitations in today's Internet are in bytes per second,
    even though the network limitations of the future might move back
    towards limitations in packets per second.

    The VoIP variant of TFRC described here will better support
    applications that do not want their sending rates in bytes per
    second to suffer from their use of small packets.  This variant is
    restricted to applications that send packets no more than once every
    10 ms (the Min Interval).  Given this restriction, the VoIP variant
    effectively calculates the TFRC fair rate as if the bottleneck
    restriction was in bytes per second.  Applications using the VoIP
    variant of TFRC could have a fixed packet size, or could vary their
    packet size in response to congestion.

    The VoIP variant of TFRC is motivated in part by the approach in RFC
    3714, which argues that it is acceptable for VoIP flows to assume
    that the network limitation is in bytes per second (Bps) rather in
    packets per second (pps), and to have the same sending rate in bytes



Floyd/Kohler                                        Section 2.  [Page 5]


INTERNET-DRAFT            Expires: January 2006                July 2005


    per second as a TCP flow with 1500-byte packets and the same packet
    drop rate.  RFC 3714 states the following:

        "While the ideal would be to have a transport protocol that is
        able to detect whether the bottleneck links along the path are
        limited in Bps or in pps, and to respond appropriately when the
        limitation is in pps, such an ideal is hard to achieve. We would
        not want to delay the deployment of congestion control for
        telephony traffic until such an ideal could be accomplished.  In
        addition, we note that the current TCP congestion control
        mechanisms are themselves not very effective in an environment
        where there is a limitation along the reverse path in pps.
        While the TCP mechanisms do provide an incentive to use large
        data packets, TCP does not include any effective congestion
        control mechanisms for the stream of small acknowledgement
        packets on the reverse path.  Given the arguments above, it
        seems acceptable to us to assume a network limitation in Bps
        rather than in pps in considering the minimum sending rate of
        telephony traffic."

    Translating the discussion in [RFC 3714] to the congestion control
    mechanisms of TFRC, it seems acceptable to standardize a variant of
    TFRC that allows VoIP flows sending small packets to achieve a rough
    fairness with TCP flows in terms of the sending rate in Bps, rather
    than in terms of the sending rate in pps.  This is accomplished by a
    small modification to TFRC, as described below.

    However, because the bottlenecks in the network in fact can include
    limitations in pps as well as in Bps, we pay special attention to
    the potential dangers of encouraging a large deployment of best-
    effort traffic in the Internet consisting entirely of small packets.
    This is discussed in more detail in Section 4.3. In addition, as
    again discussed in Section 4.3, the VoIP variant of TFRC includes
    the limitation of the Min Interval between packets of 10 ms.

    The VoIP variant of TFRC essentially assumes that the small-packet
    VoIP TFRC flow receives roughly the same packet drop rate as a
    large-packet TFRC or TCP flow.  As we show, this assumption is not
    correct for all of the simulations in this document.

    The VoIP variant of TFRC requires a modification in TFRC's
    calculation of the loss event rate, because a VoIP TFRC connection
    can send many small packets when a standard TFRC or TCP connection
    would send a single large packet.  It is not possible for a standard
    TFRC or TCP connection to repeatedly send multiple packets per
    round-trip time in the face of a high packet drop rate.  As a
    result, TCP and standard TFRC only respond to a single loss event
    per round-trip time, and are still able to detect and respond to



Floyd/Kohler                                        Section 2.  [Page 6]


INTERNET-DRAFT            Expires: January 2006                July 2005


    increasingly heavy packet loss rates.  However, in a highly-
    congested environment, when a TCP connection might be sending, on
    average, one large packet per round-trip time, a corresponding VoIP
    TFRC connection might be sending many small packets per round-trip
    time.  As a result, in order to maintain fairness with TCP, and to
    be able to detect changes in the degree of congestion, VoIP TFRC
    needs to be sensitive to the actual packet drop rate during periods
    of high congestion.  This is discussed in more detail in the section
    below.

3.  VoIP Variant Congestion Control

    TFRC uses the TCP throughput equation given in Section 3.1 of [RFC
    3448], which gives the allowed sending rate X in bytes per second as
    a function of the loss event rate, packet size, and round-trip time.
    [RFC 3448] specifies that the packet size s used in the throughput
    equation should be the packet size used by the application, or the
    estimated mean packet size if there are variations in the packet
    size depending on the data.  This gives rough fairness with TCP
    flows using the same packet size.

    The VoIP variant changes this behavior in the following ways.

    o  The nominal packet size: The nominal packet size s is set to
       1460 bytes.  Following [RFC 3714], this provides a goal of
       fairness, in terms of the sending rate in bytes per second, with
       a TCP flow with 1460 bytes of application data per packet but
       with the same packet drop rate.

    o  Taking packet headers into account: The allowed transmit rate X
       in bytes per second is reduced by a factor that accounts for
       packet header size.  This gives the application some incentive,
       beyond the Min Interval, not to use unnecessarily small packets.
       In particular, we introduce a new parameter H, which represents
       the expected size in bytes of network and transport headers to be
       used on the TFRC connection's packets.  This is used to reduce
       the allowed transmit rate X as follows:

       X <- X * s_true / (s_true + H),

       where s_true is the true average data packet size for the
       connection in bytes, excluding the transport and network headers.

       The H parameter is set to the constant 40 bytes.  Thus, if the
       VoIP TFRC application used 40-byte data segments, the allowed
       transmit rate X would be halved to account for the fact that half
       of the sending rate would be used by the headers.  Section 4.2
       justifies this definition.  However, a connection using the VoIP



Floyd/Kohler                                        Section 3.  [Page 7]


INTERNET-DRAFT            Expires: January 2006                July 2005


       variant MAY instead use a more precise estimate of H, based on
       the actual network and transport headers to be used on the
       connection's packets.  For example, a DCCP connection [DCCP] over
       IPv4, where data packets use the DCCP-Data packet type, and there
       are no IP or DCCP options, could set H to 20 + 12 = 32 bytes.

    o  Measuring the loss event rate in times of high loss: During short
       loss intervals (those at most two round-trip times), the loss
       rate is computed by counting the actual number of packets lost or
       marked, not by counting at most one loss event per loss interval.
       Without this change, VoIP TFRC could send multiple packets per
       round-trip time even in the face of heavy congestion, for a
       steady-state behavior of multiple packets dropped each round-trip
       time.

       In standard TFRC, the TFRC receiver estimates the loss event rate
       by calculating the average loss interval in packets, and
       inverting to get the loss event rate.  Thus, for a short loss
       interval with N packets and K losses, standard TFRC calculates
       the size of that loss interval as N packets, contributing to a
       loss event rate of 1/N.  However, for VoIP TFRC, for small loss
       intervals of at most two round-trip times, if the loss interval
       consists of N packets including K losses, the size of the loss
       interval is calculated as N/K, contributing to a loss event rate
       of K/N instead of 1/N.

    o  A minimum interval between packets: The VoIP variant of TFRC
       enforces a Min Interval between packets of 10 ms.  A flow that
       wishes its transport protocol to exceed this Min Interval MUST
       use the conventional TFRC equations, rather than the VoIP
       variant.  The motivation for this is discussed below.

4.  VoIP Variant Discussion

4.1.  The TCP Throughput Equation

    The VoIP variant of TFRC uses the TCP throughput equation given in
    [RFC 3448].  As shown in Table 1 of [RFC 3714], for high packet drop
    rates, this throughput equation gives rough fairness with most
    aggressive possible current TCP: a SACK TCP flow using timestamps
    and ECN.  Because it is not recommended for routers to use ECN-
    marking in highly-congested environments (e.g., with packet drop
    rates greater than 10%), we note that it would be useful to have a
    throughput equation with a somewhat more moderate sending rate for
    packet drop rates of 40% and above.






Floyd/Kohler                                      Section 4.1.  [Page 8]


INTERNET-DRAFT            Expires: January 2006                July 2005


4.2.  Accounting for Header Size

    [RFC 3714] makes the optimistic assumption that the limitation of
    the network is in bandwidth in bytes per second (Bps), and not in
    CPU cycles or in packets per second (pps).  However, some attention
    must be paid to the load in pps as well as to the load in Bps.  Even
    aside from the Min Interval, the VoIP variant of TFRC gives the
    application some incentive to use fewer but larger packets, when
    larger packets would suffice, by including the bandwidth used by the
    packet header in the allowed sending rate.

    As an example, a sender using 120-byte packets needs a TCP-friendly
    rate of 128 Kbps to send 96 Kbps of application data.  This is
    because the TCP-friendly rate is reduced by a factor of
    s_true/(s_true + H) = 120/160, to account for the effect of packet
    headers.  If the sender suddenly switched to 40-byte data segments,
    the allowed sending rate would reduce to 64 Kbps of application
    data; and one-byte data segments would reduce the allowed sending
    rate to 3.12 Kbps of application data.  (In fact, the Min Interval
    would prevent senders from achieving these rates, since applications
    using the VoIP variant cannot send more than 100 packets per
    second.)

    The VoIP variant assumes 40 bytes for the header size, although the
    header could be larger (due to IP options, IPv6, IP tunnels, and the
    like) or smaller (due to header compression) on the wire, because
    using the exact header size in bytes would have little additional
    benefit.  The VoIP variant's use of an assumed 40-byte header is
    sufficient to get a rough estimate of the throughput, and to give
    the application some incentive not to use unnecessarily-many small
    packets.  Because we are only aiming at rough fairness, and at a
    rough incentive for applications, the use of a 40-byte header in the
    calculations of the header bandwidth seems sufficient.

4.3.  The VoIP Min Interval

    The header size calculation provides an incentive for applications
    to use fewer, but larger, packets.  Another incentive is that when
    the path limitation is in pps, the application using more small
    packets is likely to cause higher packet drop rates, and to have to
    reduce its sending rate accordingly.  That is, if the congestion is
    in terms of pps, then the flow sending more pps will increase the
    packet drop rate, and have to adjust its sending rate accordingly.
    However, the increased congestion caused by the use of small packets
    in an environment limited by pps is experienced not only by the flow
    using the small packets, but by all of the competing traffic on that
    congested link.  These incentives are therefore insufficient to
    provide sufficient protection for pps network limitations.



Floyd/Kohler                                      Section 4.3.  [Page 9]


INTERNET-DRAFT            Expires: January 2006                July 2005


    The VoIP variant for TFRC, then, includes a Min Interval of 10 ms.
    This provides additional restrictions on the use of unnecessarily
    many small packets.

    One justification for the Min Interval is the practical one that the
    applications that currently want to send small packets are the VoIP
    applications that send at most one packet every 10 ms, so this
    restriction does not affect current traffic.  A second justification
    is that there is no pressing need for best-effort traffic in the
    current Internet to send small packets more frequently than once
    every 10 ms (rather than taking the 10 ms delay at the sender, and
    merging the two small packets into one larger one).  This 10 ms
    delay for merging small packets is likely to be dominated by the
    network propagation, transmission, and queueing delays of best-
    effort traffic in the current Internet.  As a result, our judgement
    would be that the benefit to the user of having less than 10 ms
    between packets is outweighed by the benefit to the network of
    avoiding unnecessarily many small packets.

    The Min Interval causes the VoIP variant of TFRC not to support
    applications sending small packets very frequently.  Consider a TFRC
    flow with a fixed packet size of 100 bytes, but with a variable
    sending rate and a fairly uncongested path.  When this flow was
    sending at most 100 pps, it would be able to use the VoIP variant of
    TFRC.  If the flow wished to increase its sending rate to more than
    100 pps, but to keep the same packet size, it would no longer be
    able to achieve this with the VoIP variant to TFRC, and would have
    to switch to the default TFRC, receiving a dramatic, discontinuous
    decrease in its allowed sending rate.  This seems not only
    acceptable, but desirable for the global Internet.

    What is to prevent flows from opening multiple connections, each
    with a 10 ms Min Interval, and thereby getting around the limitation
    of the Min Interval?  Obviously, there is nothing to prevent flows
    from doing this, just as there is currently nothing to prevent flows
    from using UDP, or from opening multiple parallel TCP connections,
    or from using their own congestion control mechanism.  Of course,
    implementations or middleboxes are also free to limit the number of
    parallel TFRC connections opened to the same destination in times of
    congestion, if that seems desirable.  And flows that open multiple
    parallel connections are subject to the inconveniences of reordering
    and the like.  But even without a mechanism to prevent flows from
    subverting the Min Interval by opening multiple parallel
    connections, it seems useful to include the Min Interval in the VoIP
    variant of TFRC.






Floyd/Kohler                                     Section 4.3.  [Page 10]


INTERNET-DRAFT            Expires: January 2006                July 2005


4.4.  Counting Packet Losses

    It is not possible for a TCP connection to persistently send
    multiple packets per round-trip time in the face of high congestion,
    with a steady-state with multiple packets dropped per round-trip
    time.  For TCP, when one or more packets are dropped each round-trip
    time, the sending rate is quickly dropped to less than one packet
    per round-trip time.  In addition, for TCP with Tahoe, NewReno, or
    SACK congestion control mechanisms, the response to congestion is
    largely independent of the number of packets dropped per round-trip
    time.

    As a result, standard TFRC can best achieve fairness with TCP, even
    in highly congested environments, by calculating the loss event rate
    rather than the packet drop rate, where a loss event is one or more
    packets dropped or marked from a window of data.

    However, with the VoIP variant of TFRC, it is no longer possible to
    achieve fairness with TCP or with standard TFRC by counting only the
    loss event rate [WBL04].  Instead of sending one large packet per
    round-trip time, the VoIP variant of TFRC could be sending N small
    packets (where N small packets equal one large 1500-byte packet).
    The loss measurement used with the VoIP variant of TFRC has to be
    able to detect a connection that is sending multiple packets per
    round-trip time in the face of multiple packet losses or marks per
    round-trip time.

    In the VoIP variant of TFRC, the loss event rate is calculated by
    counting at most one loss event in loss intervals longer than two
    round-trip times, and by counting each packet lost or marked in
    shorter loss intervals.  In particular, for a short loss interval
    with N packets, including K lost or marked packets, the loss
    interval length is calculated as N/K, instead as N.  The average
    loss interval I_mean is still averaged over the most recent eight
    loss intervals, as specified in Section 5.4 of RFC 3448.  Thus, if
    eight successive loss intervals are short loss intervals with N
    packets and K losses, the loss event rate is calculated as K/N,
    rather than as 1/N.

5.  A Comparison with RFC 3714

6.  The VoIP Variant with Applications that Modify the Packet Size

    To be done.







Floyd/Kohler                                       Section 6.  [Page 11]


INTERNET-DRAFT            Expires: January 2006                July 2005


7.  Simulation Results

7.1.  Simulations with Configured Drop Rates

    In this section we describe simulation results from simulations
    comparing the throughput of standard (SACK) TCP flows, TCP flows
    with timestamps and ECN, VoIP TFRC flows, and standard TFRC (Stnd
    TFRC) flows.  In these simulations we configure the router to
    randomly drop or mark packets at a specified rate, independently of
    the packet size.  For each specified packet drop rate, we give a
    flow's average sending rate in Kbps over the second half of a
    100-second simulation, averaged over ten flows.

      Packet      TCP        ECN TCP     TFRC
      DropRate    SendRate   SendRate    SendRate
      --------    --------   --------    --------
         0.001    2020.85    1904.61     982.09
         0.005     811.10     792.11     878.08
         0.01      515.45     533.19     598.90
         0.02      362.93     382.67     431.41
         0.04      250.06     252.64     284.82
         0.05      204.48     218.16     268.51
         0.066     176.40     178.16     211.05
         0.1       143.30     148.41     146.03
         0.2        78.65      93.23*     55.14
         0.3        26.26      59.65*     32.87
         0.4         9.87      47.79*     25.45
         0.5         3.53      32.01*     18.52

    Table 1:
    "Total Sending Rates (Kbps) for Configured Packet Drop Rates.
    The TFRC flow uses 1460-byte data packets, with a maximum
    data sending rate of 1000 Kbps."

    * Note: These ECN scenarios are not realistic, as routers are
    not likely to mark packets when packet drop/mark rates are 20%
    or higher.


    Table 1 shows the sending rate for a TCP and a standard TFRC flow
    for a range of configured packet drop rates, when both flows have
    1460-byte data packets, in order to illustrate the relative fairness
    of TCP and TFRC when both flows use the same packet size.  For
    example, a packet drop rate of 0.1 means that 10% of the TCP and
    TFRC packets are dropped.  There is good relative fairness until the
    packet drop percentages reach 40 and 50%, when the TFRC flow
    receives three to five times more bandwidth than the standard TCP
    flow.  We note that an ECN TCP flow would receive a higher



Floyd/Kohler                                     Section 7.1.  [Page 12]


INTERNET-DRAFT            Expires: January 2006                July 2005


    throughput than the TFRC flow, but this would not be relevant in
    practice, as routers are advised to drop rather than mark packets
    during high levels of congestion.


      Packet      TCP        ECN TCP      VoIP TFRC  Stnd TFRC
      DropRate    SendRate   SendRate     SendRate   SendRate
      --------    --------   --------     --------   --------
         0.001    1787.54    1993.03      17.71      17.69
         0.005     785.11     823.75      18.11      17.69
         0.01      533.38     529.01      17.69      17.80
         0.02      317.16     399.62      17.69      13.41
         0.04      245.42     260.57      17.69       8.84
         0.05      216.38     223.75      17.69       7.63
         0.066     174.07     184.37      17.69       6.46
         0.1       142.75     138.36      17.69       4.29
         0.2        58.61      91.54*     17.80       1.94
         0.3        21.62      63.96*     10.26       1.00
         0.4        10.51      41.74*      4.78       0.77
         0.5         1.92      19.03*      2.41       0.56

    Table 2:
    "Total Sending Rates (Kbps) for Configured Packet Drop Rates.
    The TFRC flows use 14-byte data packets, with a maximum
    data sending rate of 5.6 Kbps."

    * Note: These ECN scenarios are not realistic, as routers are
    not likely to mark packets when packet drop/mark rates are 20%
    or higher.


    Table 2 shows the results of simulations where each VoIP TFRC flow
    has a maximum data sending rate of 5.6 Kbps, with 14-byte data
    packets and a 32-byte packet header for DCCP and IP.  Each TCP flow
    has a receive window of 100 packets and a data packet size of 1460
    bytes, with a 40-byte packet header for TCP and IP.  The TCP flow
    uses Sack TCP with Limited Transmit, but without timestamps or ECN.
    Each flow has a round-trip time of 240 ms.

    The TFRC sending rate in Table 2 is the sending rate for the 14-byte
    data packet with the 32-byte packet header.  Thus, only 30% of the
    TFRC sending rate is for data, and with a packet drop rate of p,
    only a fraction 1-p of that data makes it to the receiver.  Thus,
    the TFRC data receive rate can be considerably less than the TFRC
    sending rate in the table.  Because TCP uses large packets, 97% of
    the TCP sending rate is for data, and the same fraction 1-p of that
    data makes it to the receiver.




Floyd/Kohler                                     Section 7.1.  [Page 13]


INTERNET-DRAFT            Expires: January 2006                July 2005


    Table 2 shows that for the 5.6 Kbps data stream with TFRC, Standard
    TFRC (Stnd TFRC) gives a very poor sending rate in bps, relative to
    the sending rate for the large-packet TCP connection.  In contrast,
    the sending rate for the VoIP TFRC flow is relatively close to the
    desired goal of fairness in bps with TCP.

    Table 2 shows that with VoIP TFRC, the 5.6 Kbps data stream doesn't
    reduce its sending rate until packet drop rates greater than 20%, as
    desired.  With packet drop rates of 30-40%, the sending rate for the
    VoIP TFRC flow is somewhat less than that of the average large-
    packet TCP flow, while for packet drop rates of 50% the sending rate
    for the VoIP TFRC flow is somewhat greater than that of the average
    large-packet TCP flow.

    We note that the high sending rate for ECN TCP in environments with
    high marking rates is largely irrelevant, as routers are advised to
    drop rather than mark ECN-capable packets in times of high
    congestion.  The sending rate of a TCP connection using timestamps
    is similar to the sending rate shown for a standard TCP connection
    without timestamps.

      Byte        TCP        ECN TCP      VoIP TFRC  Stnd TFRC
      DropRate    SendRate   SendRate     SendRate   SendRate
      --------    --------   --------     --------   --------
         0.00001   423.02     406.44      17.69      17.69
         0.0001    117.41     114.34      17.69      17.69
         0.001       0.41       3.38*     17.69       8.37
         0.005       0.26       0.26*     18.39       1.91
         0.01        0.31       0.26*      7.07       0.84
         0.02        0.29       0.26*      1.61       0.43
         0.04        0.12       0.26*      0.17       0.12
         0.05        0.15       0.26*      0.08       0.06

    Table 3:
    "Total Sending Rates (Kbps) for Configured Byte Drop Rates.
    The TFRC flows use 14-byte data packets, with a maximum
    data sending rate of 5.6 Kbps."

    * Note: These ECN scenarios are not realistic, as routers are
    not likely to mark packets when packet drop/mark rates are 20%
    or higher.










Floyd/Kohler                                     Section 7.1.  [Page 14]


INTERNET-DRAFT            Expires: January 2006                July 2005


      Byte        TCP Pkt    TFRC Pkt   TCP/TFRC
      DropRate    DropRate   DropRate   Pkt Drop Ratio
      --------    --------   --------   --------------
       0.00001      0.015     0.0006     26.59
       0.0001       0.13      0.0056     24.94
       0.001        0.77      0.054      14.26
       0.005        0.99      0.24        4.08
       0.01         1.0       0.43        2.32
       0.05         1.0       0.94        1.05

    Table 4:
    "Converting Byte Drop Rates to Packet Drop Rates,
    for 1500-byte TCP packets and 56-byte TFRC packets."


    In contrast, Table 3 shows the TCP and TFRC send rates for various
    byte drop rates.  For these simulations, for each *byte*, the packet
    containing that byte is dropped with probability p.  Table 4
    converts the byte drop rate p to packet drop rates for the TCP and
    TFRC packets, where the packet drop rate for an N-byte packet is
    1-(1-p)^N.  Thus, a byte drop rate of 0.001, with each byte being
    dropped with probability 0.001, converts to a packet drop rate of
    0.77, or 77%, for the 1500-byte TCP packets, and a packet drop rate
    of 0.054, or 5.4%, for the 56-byte TFRC packets.

    The last column of Table 3 shows the ratio between the TCP packet
    drop rate and the TFRC packet drop rate.  For low byte drop rates,
    this ratio is close to 26.8, the ratio between the TCP and TFRC
    packet sizes.  For high byte drop rates, where even most small TFRC
    packets are likely to be dropped, this drop ratio approaches 1.

    As Table 3 shows. with moderate byte drop rates (between 0.1% and
    4%), even the standard TFRC flow with 14-byte data packets has
    higher throughput than the large-packet TCP flow.  In these regimes,
    the packet drop rate for TCP is greater than 50%, and the TCP
    sending rate no longer varies as 1/sqrt(p), as it does with smaller
    packet drop rates.  With these equal byte drop rates for the TCP and
    TFRC flows, the VoIP variant of TFRC makes the unfairness in favor
    of TFRC even worse, with the VoIP TFRC flow receiving significantly
    higher throughput than the TCP flow for byte drop rates from 0.1% to
    2%.










Floyd/Kohler                                     Section 7.1.  [Page 15]


INTERNET-DRAFT            Expires: January 2006                July 2005


      Packet      TCP        ECN TCP     VoIP TFRC  Stnd TFRC
      DropRate    SendRate   SendRate    SendRate   SendRate
      --------    --------   --------    --------   --------
         0.001    1908.98    1869.24     183.45     178.35
         0.005     854.69     835.10     185.06     138.06
         0.01      564.10     531.10     185.33      92.43
         0.02      365.38     369.10     185.57      62.18
         0.04      220.80     257.81     185.14      45.43
         0.05      208.97     219.41     180.08      39.44
         0.066     179.04     184.68     168.51      31.16
         0.1       141.67     143.88     127.33      21.96
         0.2        62.66      91.87*     54.66       9.40
         0.3        16.63      65.52*     24.50       4.73
         0.4         6.62      42.00*     13.47       3.35
         0.5         1.01      21.34*     10.51       2.92

    Table 5:
    "Total Sending Rates (in Kbps) for Configured Packet Drop Rates.
    The TFRC flows use 200-byte data packets, with a maximum
    data sending rate of 160 Kbps."

    * Note: These ECN scenarios are not realistic, as routers are
    not likely to mark packets when packet drop/mark rates are 20%
    or higher.


    Using configured packet drop rates, Table 5 compares the average
    per-flow sending rates when the TFRC flow has a maximum data sending
    rate of 160 Kbps, with the application generating 200-byte data
    packets at 100 packets per second.  As expected with equal packet
    drop rates, the performance of Standard TFRC is quite poor, while
    the performance of VoIP TFRC is essentially as desired for packet
    drop rates up to 30%.  Again as expected, with packet drop rates of
    40-50% the VoIP TFRC sending rate is somewhat higher than desired.

    In general, Tables 2 and 5 show acceptable performance for VoIP TFRC
    in environments with stable packet drop rates, where the decision to
    drop a packet is independent of the packet size.  However, in
    realistic environments, the packet size might affect the likelihood
    that a packet is dropped.  For example, with heavy congestion and a
    Drop Tail queue with a fixed number of bytes rather than a fixed
    number of packet-sized buffers, small packets might be more likely
    than large packets to find room at the end of an almost-full queue.
    As a further complication, in a scenario with Active Queue
    Management, the AQM mechanism could either be in packet mode,
    dropping each packet with equal probability, or in byte mode,
    dropping each byte with equal probability.  Sections 7.2 and 7.3
    show simulations with packets dropped at Drop Tail or AQM queues,



Floyd/Kohler                                     Section 7.1.  [Page 16]


INTERNET-DRAFT            Expires: January 2006                July 2005


    rather that from a probabilistic process.

    VoIP mode for TFRC has been added to the NS simulator, and is
    illustrated in the validation test "./test-all-friendly" in the
    directory tcl/tests.  The simulation scripts for the simulations in
    this document will be made available in
    "http://www.icir.org/tfrc/voipsims.html".

7.2.  Packet Dropping Behavior at Routers with DropTail Queues

    One of the problems with comparing the throughput of two flows using
    different packet sizes is that the packet size itself can influence
    the packet drop rate [V00, WBL04].

    The default TFRC, without the VoIP variant, was designed for rough
    fairness with TCP, for TFRC and TCP flows with the same packet size
    and experiencing the same packet drop rate.  When the issue of
    fairness between flows with different packets sizes is addressed, it
    matters whether the packet drop rates experienced by the flows is
    related to the packet size.  That is, are small VoIP packets just as
    likely to be dropped as large TCP packets, or are the smaller
    packets less likely to be dropped [WBL04]? And what is the
    relationship between the packet-dropping behavior of the path, and
    the loss event measurements of TFRC?

          Web        TCP       TCP       VoIP_TFRC VoIP_TFRC
        Sessions   DropRate  SendRate    DropRate  SendRate
        --------   --------  --------    --------  --------
            10       0.04     316.18       0.05     183.05
            25       0.07     227.47       0.07     181.23
            50       0.08     181.10       0.08     178.32
           100       0.14      85.97       0.12     151.42
           200       0.17      61.20       0.14      73.88
           400       0.20      27.79       0.18      36.81
           800       0.29       3.50       0.27      16.33
          1600       0.37       0.63       0.33       6.29

    Table 6:
    "Simulation Results with Drop-Tail Queues in Packets."


    Table 6 shows the results of the second half of 100-second
    simulations, with five TCP connections and five VoIP TFRC
    connections competing with web traffic in a topology with a 3 Mbps
    shared link.  The VoIP TFRC application generates 200-byte data
    packets every 10 ms, for a maximum data rate of 160 Kbps.  The five
    TCP connections have roundtrip times from 40 to 240 ms, and the five
    TFRC connections have the same set of round-trip times.  The SACK



Floyd/Kohler                                     Section 7.2.  [Page 17]


INTERNET-DRAFT            Expires: January 2006                July 2005


    TCP connections in these simulations use the default parameters in
    the NS simulator, with Limited Transmit, and a minimum RTO of 200
    ms.  Adding timestamps to the TCP connection didn't change the
    results appreciably.  The simulations include reverse-path traffic,
    to add some small control packets to the forward path, and some
    queueing delay to the reverse path.  The number of web sessions is
    varied to create different levels of congestion.  The DropTail queue
    is in units of packets, which each packet in the queue requires a
    single buffer, regardless of the packet size.

    Table 6 shows the average TCP and TFRC sending rates, each averaged
    over the five flows.  As expected, the VoIP TCP flows see similar
    packet drop rates as the TCP flows, though the VoIP TFRC flows
    receives higher throughput than the TCP flows with packet drop rates
    of 25% or higher.

          Web        TCP       TCP       VoIP_TFRC VoIP_TFRC
        Sessions   DropRate  SendRate    DropRate  SendRate
        --------   --------  --------    --------  --------
            10       0.06     239.81       0.00     185.19
            25       0.09     189.02       0.01     184.95
            50       0.14      99.46       0.01     185.07
           100       0.20      16.42       0.02     183.77
           200       0.26       4.46       0.03     181.98
           400       0.29       4.61       0.05     151.88
           800       0.49       1.01       0.08     113.10
          1600       0.65       0.67       0.12      65.17

    Table 7:
    "Simulation Results with Drop-Tail Queues in Bytes."


    However, the fairness results can change significantly if the
    DropTail queue at the congested output link is in units of bytes
    rather than packets.  For a queue in packets, the queue has a fixed
    number of buffers, and each buffer can hold exactly one packet,
    regardless of its size in bytes.  For a queue in bytes, the queue
    has a fixed number of *bytes*, and an almost-full queue might have
    room for a small packet but not for a large one.  This, for a
    simulation with a Drop-Tail queue in bytes, large packets are more
    likely to be dropped than are small ones.  The NS simulator doesn't
    yet have a more realistic intermediate model, where the queue has a
    fixed number of buffers, each buffer has a fixed number of bytes,
    and each packet would require one or more free buffers.  In this
    model, a small packet would use one buffer, while a larger packet
    would require several buffers.





Floyd/Kohler                                     Section 7.2.  [Page 18]


INTERNET-DRAFT            Expires: January 2006                July 2005


    As Table 7 shows, with a DropTail queue in bytes, the VoIP TFRC flow
    sees a much smaller drop rate than the TCP flow, and as a
    consequence receives a much larger sending rate.  For example, when
    the five TCP flows and five VoIP TFRC flows share the link with 800
    web sessions, the five TCP flows see an average drop rate of 49% in
    the second half of the simulation, while the five VoIP TFRC flows
    receive an average drop rate of 8%, and as a consequence receive
    more than 100 times the throughput of the TCP flows.  This raises
    serious questions about making the assumption that flows with small
    packets see the same packet drop rate as flows with larger packets.
    Further work will have to include an investigation into the range of
    realistic Internet scenarios, in terms of whether large packets are
    considerably more likely to be dropped than are small ones.

7.3.  Packet Dropping Behavior at Routers with AQM

    As expected, the packet dropping behavior also can be varied by
    varying the Active Queue Management (AQM) mechanism in the router.
    When the routers use RED (Random Early Detection), there are several
    parameters than can affect the packet dropping or marking behavior
    as a function of the packet size.

    First, as with DropTail, the RED queue can be either in units of
    packets or of bytes.  This can affect the packet dropping behavior
    when the RED mechanism is unable to control the average queue size,
    and the queue overflows.

    Second, and orthogonally, RED can be either in packet mode or in
    byte mode.  In packet mode, each *packet* has the same probability
    of being dropped by RED, while in byte mode, each *byte* has the
    same probability of being dropped.  In packet mode, large-packet and
    small-packet flows receive roughly the same packet drop rate, while
    in byte mode, with small to moderate levels of congestion, large-
    packet and small-packet flows with the same throughput in bps
    receive roughly the same *number* of packet drops.  The simulations
    reported in the appendix show that for RED in packet mode, the
    packet drop rates for the VoIP TFRC flows are similar to those for
    the TCP flows, with a resulting acceptable throughput for the VoIP
    TFRC flows.   This is true with the queue in packets or in bytes,
    and with or without Adaptive RED (discussed below).  As we show
    below, this fairness between TCP and VoIP TFRC flows does not hold
    for RED in byte mode.

    The third RED parameter that affects the packet dropping or marking
    behavior as a function of packet size is whether the RED mechanism
    is using Standard RED or Adaptive RED, where the dropping function
    is varied as the packet drop rate changes.  The use of Adaptive RED
    allows the RED mechanism to function more effectively in the



Floyd/Kohler                                     Section 7.3.  [Page 19]


INTERNET-DRAFT            Expires: January 2006                July 2005


    presence of high packet drop rates (e.g., greater than 10%).
    Without Adaptive RED, there is a fixed dropping threshold, and all
    arriving packets are dropped when the dropping or marking rate
    exceeds this threshold.  In contrast, with Adaptive RED, the
    dropping function is adapted to accommodate these high-drop-rate
    regimes.  One consequence is that when byte mode is combined with
    Adaptive RED, the byte mode extends even to high-drop-rate regimes.
    When byte mode is used with standard RED, however, the byte mode is
    no longer in use when the drop rate exceeds the fixed dropping
    threshold (set by default to 10% in the NS simulator).

    In the simulations in this section, we explore the VoIP TFRC
    behavior over some of this range of scenarios.  In this simulations,
    as in Section 7.2 above, the application for the VoIP TFRC flow uses
    200-byte data packets, generating 100 packets per second.


          Web        TCP       TCP       VoIP_TFRC VoIP_TFRC
        Sessions   DropRate  SendRate    DropRate  SendRate
        --------   --------  --------    --------  --------
            10       0.05     305.76       0.04     182.82
            25       0.06     224.16       0.06     175.91
            50       0.09     159.12       0.08     152.51
           100       0.13      90.77       0.11     106.13
           200       0.14      48.53       0.14      70.25
           400       0.20      22.08       0.19      41.50
           800       0.27       3.55       0.25      17.50
          1600       0.42       1.87       0.34       8.81

    Table 8:
    "Simulation Results with RED Queues, Packet Mode.
    RED in packet mode, queue in packets, standard RED."


    For the simulations in Table 8, with a congested router with a RED
    queue in packet mode, the results are similar to those in Table 6
    above.  The VoIP TCP flow receives similar packet drop rates as the
    TCP flow, though it receives higher throughput in the more congested
    environments.  The simulations are similar with the queue in bytes,
    and with or without Adaptive RED.  These results seems generally
    acceptable.










Floyd/Kohler                                     Section 7.3.  [Page 20]


INTERNET-DRAFT            Expires: January 2006                July 2005


          Web        TCP       TCP       VoIP_TFRC VoIP_TFRC
        Sessions   DropRate  SendRate    DropRate  SendRate
        --------   --------  --------    --------  --------
            10       0.04     286.90       0.01     184.92
            25       0.07     192.67       0.02     184.06
            50       0.11      92.98       0.04     181.69
           100       0.17      57.02       0.07     154.88
           200       0.17       9.55       0.11     120.83
           400       0.29       7.68       0.16      77.38
           800       0.36       1.97       0.23      23.79
          1600       0.51       0.87       0.30      11.00

    Table 9:
    "Simulation Results with RED Queues, Byte Mode.
    RED in byte mode, queue in bytes, standard RED."


    Table 9 shows that with a standard RED queue in byte mode, there is
    a somewhat greater different between the packet drop rates between
    the TCP and VoIP TFRC flows, particularly for lower packet drop
    rates.  For the simulation in Table 9, the packet drop rates for the
    TCP flows can range from 1 1/2 to four times greater than the packet
    drop rates for the VoIP TFRC flows.  However, because the VoIP TFRC
    flow has an upper bound on the sending rate, its sending rate is not
    affected in the lower packet-drop-rate regimes; its sending rate is
    only affected in the regimes with packet drop rates of 10% or more.
    While the sending rate for VoIP TFRC in the scenarios in Table 9
    with higher packet drop rates are greater than desired, the results
    are significantly better than that of VoIP flows with no congestion
    control at all.


          Web        TCP       TCP       VoIP_TFRC VoIP_TFRC
        Sessions   DropRate  SendRate    DropRate  SendRate
        --------   --------  --------    --------  --------
            10       0.04     297.74       0.02     185.06
            25       0.07     209.42       0.03     184.06
            50       0.10      85.34       0.04     182.30
           100       0.16      28.18       0.05     181.17
           200       0.19       4.18       0.06     177.70
           400       0.31       1.87       0.08     154.40
           800       0.29       0.77       0.06     170.01
          1600       0.59       0.48       0.02     173.91

    Table 10:
    "Simulation Results with RED Queues, Byte Mode.
    RED in byte mode, queue in bytes, Adaptive RED."




Floyd/Kohler                                     Section 7.3.  [Page 21]


INTERNET-DRAFT            Expires: January 2006                July 2005


    In contrast, for the simulations in Table 10, the congested router
    uses an Adaptive RED queue in byte mode.   For this router, the
    output queue is in units of bytes rather than of packets, each
    *byte* is dropped with the same probability, and because of the use
    of Adaptive RED, this byte-dropping mode extends even for the high-
    packet-drop-rate regime.

    As Table 10 shows, for a scenario with Adaptive RED in byte mode,
    the packet drop rate for the VoIP TFRC traffic is *much* lower than
    that for the TCP traffic, and as a consequence, the sending rate for
    the VoIP TFRC traffic in a highly congested environment is *much*
    higher than that of the TCP traffic.  In fact, in these scenarios
    the TFRC VoIP congestion control mechanisms are largely ineffective
    for the VoIP traffic.

    We note that the unfairness in these simulations, in favor of VoIP
    TFRC, is even greater than the unfairness shown in Table 7 for a
    DropTail queue in bytes.  At the same time, it is not known if there
    is any deployment in the Internet of any routers with Adaptive RED
    in byte mode, or of any AQM mechanisms with similar behavior;  we
    don't even know the extent of the deployment of standard RED, or or
    any of the proposed AQM mechanisms.


8.  Discussion

    The goal of the VoIP variant of TFRC has been for the TCP flows and
    the VoIP TFRC flows to have rough fairness in the sending rate in
    bps, in a scenario where each packet receives roughly the same
    probability of being dropped.  In a scenario where large packets are
    more likely to be dropped than small packets, or where flows with a
    bursty sending rate are more likely to have packets dropped than are
    flows with a smooth sending rate, flows using the VoIP variant of
    TFRC could receive more bandwidth than competing TCP flows.

    The VoIP variant of TFRC limits the sending rate in packet per
    second.  The simulations by Tom Phelan explore how a limitation in
    sending rate complicates the issue of exploring the fairness between
    TCP and VoIP TFRC flows.

    For applications with a maximum sending rate of 96 Kbps or less,
    VoIP TFRC only restricts the sending rate when the packet drop rate
    is fairly high.  In this regime, the performance of TFRC is very
    much determined by the accuracy of the TCP response function in
    representing the actual sending rate of a TCP connection.  In this
    regime of high packet drop rates, the performance of the TCP
    connection is very much affected by the TCP algorithm (e.g., SACK or
    not), by the minimum RTO, by the use or not of Limited Transmit, by



Floyd/Kohler                                       Section 8.  [Page 22]


INTERNET-DRAFT            Expires: January 2006                July 2005


    the use of timestamps and/or of ECN, and the like.  It is good to
    insure that simulations or experiments exploring fairness include
    the exploration of fairness with the most aggressive TCP mechanisms
    conformance with the current standards.  Our simulations use SACK
    TCP with Limited Transmit and with a minimum RTO of 200 ms.  Adding
    the use of timestamps has not made a big difference.  We haven't
    used ECN, because our judgement is that the use of ECN is not
    advisable in high-packet-dropping regimes.

9.  Implementation Issues

    TBA

10.  Security Considerations

    TBA

11.  IANA Considerations

    There are no IANA considerations in this document.

12.  Thanks

    We thank Tom Phelan for discussions of the VoIP variant of TFRC and
    for his paper exploring the fairness between TCP and VoIP TFRC
    flows.  We thank Joerg Widmer for feedback on earlier versions of
    this draft.  We also thank the DCCP Working Group for feedback and
    discussions.

Normative References

    [RFC 2119] S. Bradner. Key Words For Use in RFCs to Indicate
        Requirement Levels. RFC 2119.

    [RFC 2434] T. Narten and H. Alvestrand.  Guidelines for Writing an
        IANA Considerations Section in RFCs.  RFC 2434.

    [RFC 2581] M. Allman, V. Paxson, and W. Stevens.  TCP Congestion
        Control.  RFC 2581.

    [RFC 3448] M. Handley, S. Floyd, J. Padhye, and J. Widmer, TCP
        Friendly Rate Control (TFRC): Protocol Specification, RFC 3448,
        Proposed Standard, January 2003.

Informative References

    [CCID 3 PROFILE] S. Floyd, E. Kohler, and J. Padhye.  Profile for
        DCCP Congestion Control ID 3: TFRC Congestion Control.  draft-



Floyd/Kohler                                                   [Page 23]


INTERNET-DRAFT            Expires: January 2006                July 2005


        ietf-dccp-ccid3-06.txt, work in progress, October 2004.

    [DCCP] E. Kohler, M. Handley, and S. Floyd.  Datagram Congestion
        Control Protocol, draft-ietf-dccp-spec-08.txt, work in progress,
        October 2004.

    [JFAS05] A. Jain, S. Floyd, M. Allman, and P. Sarolahti.  Quick-
        Start for TCP and IP.  Internet-draft draft-amit-quick-
        start-04.txt, work in progress, February 2004.

    [MAF04] A. Medina, M. Allman, and A. Floyd, Measuring the Evolution
        of Transport Protocols in the Internet, May 2004, URL
        "http://www.icir.org/tbit/".

    [P04] T. Phelan, TFRC with Self-Limiting Sources, October 2004.  URL
        "http://www.phelan-4.com/dccp/".

    [RFC 2861] M. Handley, J. Padhye, and S. Floyd.  TCP Congestion
        Window Validation.  RFC 2861, June 2000.

    [RFC 3714] S. Floyd and J. Kempf, Editors.  IAB Concerns Regarding
        Congestion Control for Voice Traffic in the Internet.  RFC 3714.

    [V00] P. Vasallo.  Variable Packet Size Equation-Based Congestion
        Control.  ICSI Technical Report TR-00-008, April 2000.  URL
        "http://www.icsi.berkeley.edu/techreports/2000.abstracts/
        tr-00-008.html".


    [WBL04] J. Widmer, C. Boutremans, and Jean-Yves Le Boudec.
        Congestion Control for Flows with Variable Packet Size, ACM CCR,
        34(2), 2004.

Authors' Addresses

    Sally Floyd <floyd@icir.org>
    ICSI Center for Internet Research
    1947 Center Street, Suite 600
    Berkeley, CA 94704
    USA

    Eddie Kohler <kohler@cs.ucla.edu>
    4531C Boelter Hall
    UCLA Computer Science Department
    Los Angeles, CA 90095
    USA





Floyd/Kohler                                                   [Page 24]


INTERNET-DRAFT            Expires: January 2006                July 2005


13.  Related Work on VoIP Variants of TFRC

    Other proposals for variants of TFRC for applications with variable
    packet sizes include [WBL04] and [V00]. [V00] proposed that TFRC
    should be modified so that flows are not penalized by sending
    smaller packets.  In particular, [V00] proposes using the path MTU
    in the TCP-friendly equation, instead of the actual packet size used
    by TFRC, and counting the packet drop rate by using an estimation
    algorithm that aggregates both packet drops and received packets
    into MTU-sized units.

    [WBL04] also argues that adapting TFRC for variable packet sizes by
    just using the packet size of a reference TCP flow in the TFRC
    equation would not suffice in the high-packet-loss regime; such a
    modified TFRC would have a strong bias in favor of smaller packets,
    because multiple lost packets in a single round-trip time would be
    aggregated into a single packet loss.  [WBL04] proposes modifying
    the loss measurement process to account for the bias in favor of
    smaller packets.

    The VoIP Variant proposed in our document differs from [WBL04] in
    restricting its attention to flows that send at most 100 packets per
    second.  The VoIP Variant proposed in our document also differs from
    the straw proposal discussed in [WBL04] in that the allowed
    bandwidth includes the bandwidth used by packet headers.

    [WBL04] discusses four methods for modifying the loss measurement
    process, "unbiasing, "virtual packets", "random sampling", and "Loss
    Insensitive Period (LIP) scaling".  [WBL04] finds only the second
    and third methods sufficiently robust when the network drops packets
    independently of packet size.  They find only the second method
    sufficiently robust when the network is more likely to drop large
    packets than small packets.  Our method for calculating the loss
    event rate is somewhat similar to the random sampling method
    proposed in [WBL04], except that randomization is not used; instead
    of randomization, the exact packet loss rate is computed for short
    loss intervals, and the standard loss event rate calculation is used
    for longer loss intervals.  [WBL04] includes simulations with a
    Bernoulli loss model, a Bernoulli loss model with a drop rate
    varying over time, and a Gilbert loss model, as well as more
    realistic simulations with a range of TCP and TFRC flows.

    [WBL04] produces both a byte-mode and a packet-mode variant of the
    TFRC transport protocol, for connections over paths with per-byte
    and per-packet dropping respectively.  We would argue that in the
    absence of transport-level mechanisms for determining whether the
    packet dropping in the network is per-packet, per-byte, or somewhere
    in between, a single TFRC implementation is needed, independently of



Floyd/Kohler                                      Section 13.  [Page 25]


INTERNET-DRAFT            Expires: January 2006                July 2005


    the packet-dropping behaviors of the routers along the path.  It
    would of course be preferable to have a mechanism that gives roughly
    acceptable behavior, to the connection and to the network as a
    whole, on paths with both per-byte and per-packet dropping (and on
    paths with multiple congested routers, some with per-byte dropping
    mechanisms, some with per-packet dropping mechanisms, and some with
    dropping mechanisms that lie somewhere between per-byte and per-
    packet).  A first step will be to investigate the range of behaviors
    actually present in today's networks.


14.  Simulation Results with RED Queue Management


          Web        TCP       TCP       VoIP_TFRC VoIP_TFRC
        Sessions   DropRate  SendRate    DropRate  SendRate
        --------   --------  --------    --------  --------
            10       0.04     299.95       0.04     184.15
            25       0.06     234.00       0.06     176.43
            50       0.08     172.42       0.08     147.82
           100       0.11     110.83       0.12      89.78
           200       0.13      41.90       0.15      62.37
           400       0.23      11.38       0.20      25.05
           800       0.27       2.88       0.29      12.26
          1600       0.34       0.62       0.35       6.78

    Table 11:
    "Simulation Results with RED Queues, Packet Mode.
    RED in packet mode, queue in packets, Adaptive RED."






















Floyd/Kohler                                      Section 14.  [Page 26]


INTERNET-DRAFT            Expires: January 2006                July 2005


          Web        TCP       TCP       VoIP_TFRC VoIP_TFRC
        Sessions   DropRate  SendRate    DropRate  SendRate
        --------   --------  --------    --------  --------
            10       0.04     319.87       0.03     183.32
            25       0.05     246.38       0.05     175.27
            50       0.08     148.18       0.07     152.02
           100       0.12      96.38       0.11      92.87
           200       0.18      43.30       0.17      55.87
           400       0.24      12.58       0.21      20.25
           800       0.30       4.22       0.27      20.08
          1600       0.35       1.34       0.35       7.21

    Table 12:
    "Simulation Results with RED Queues, Packet Mode.
    RED in packet mode, queue in bytes, Adaptive RED."


          Web        TCP       TCP       VoIP_TFRC VoIP_TFRC
        Sessions   DropRate  SendRate    DropRate  SendRate
        --------   --------  --------    --------  --------
            10       0.04     326.74       0.03     184.79
            25       0.05     260.50       0.04     180.28
            50       0.08     173.62       0.07     154.04
           100       0.11      81.94       0.10     103.87
           200       0.16      48.10       0.14      60.50
           400       0.25       8.64       0.19      32.55
           800       0.35       7.68       0.26      14.62
          1600       0.38       0.77       0.36       7.13

    Table 13:
    "Simulation Results with RED Queues, Packet Mode.
    RED in packet mode, queue in bytes, standard RED."



15.  A Discussion of Packet Size and Packet Dropping

    This section gives a general summary of the relative advantages of
    packet-dropping behavior independent of packet size, versus packet
    dropping behavior where large packets are more likely to be dropped
    than small ones.










Floyd/Kohler                                      Section 15.  [Page 27]


INTERNET-DRAFT            Expires: January 2006                July 2005


    Advantages of Packet Dropping Independent of Packet Size:
    ---------------------------------------------------------
    1.  Adds another incentive for end nodes to use large packets.
    2.  Matches an environment with a limitation in pps rather than
        bps.
    ---------------------------------------------------------

    Advantages of Packet Dropping as a Function of Packet Size:
    ---------------------------------------------------------
    1.  Small control packets are less likely to be dropped than are
        large data packets, improving TCP performance.
    2.  Matches an environment with a limitation in bps rather than
        pps.
    3.  Reduces the penalty of TCP and other transport protocols
        against flows with small packets (where the allowed sending
        rate is roughly a linear function of packet size).
    4.  A queue limited in bytes is better than a queue limited in
        packets for matching the worst-case queue size to the
        worst-case queueing delay in seconds.
    ---------------------------------------------------------



Full Copyright Statement

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

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

Intellectual Property

    The IETF takes no position regarding the validity or scope of any
    Intellectual Property Rights or other rights that might be claimed
    to pertain to the implementation or use of the technology described
    in this document or the extent to which any license under such
    rights might or might not be available; nor does it represent that
    it has made any independent effort to identify any such rights.
    Information on the procedures with respect to rights in RFC
    documents can be found in BCP 78 and BCP 79.




Floyd/Kohler                                                   [Page 28]


INTERNET-DRAFT            Expires: January 2006                July 2005


    Copies of IPR disclosures made to the IETF Secretariat and any
    assurances of licenses to be made available, or the result of an
    attempt made to obtain a general license or permission for the use
    of such proprietary rights by implementers or users of this
    specification can be obtained from the IETF on-line IPR repository
    at http://www.ietf.org/ipr.

    The IETF invites any interested party to bring to its attention any
    copyrights, patents or patent applications, or other proprietary
    rights that may cover technology that may be required to implement
    this standard.  Please address the information to the IETF at ietf-
    ipr@ietf.org.







































Floyd/Kohler                                                   [Page 29]

Internet-Drafts are not archival documents, and copies of Internet-Drafts
that have been deleted from the directory are not available.  The
Secretariat does not have any information regarding the future plans of
the author(s) or working group, if applicable, with respect to this
deleted Internet-Draft.  For more information, or to request a copy of
the document, please contact the author(s) directly.

Draft Author(s):

 S. Floyd <floyd@icir.org>
 E. Kohler <kohler@aciri.org>