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 INTERNET-DRAFT             DCCP User Guide                 April 2005
    DCCP User Guide
    Internet Draft                                             T. Phelan
    Document: draft-ietf-dccp-user-guide-03.txt           Sonus Networks
    Expires: October 2005                                     April 2005
           Datagram Congestion Control Protocol (DCCP) User Guide
 Status of this Memo
    By submitting this Internet-Draft, I certify that any applicable
    patent or other IPR claims of which I am aware have been disclosed,
    or will be disclosed, and any of which I become aware will be
    disclosed, in accordance with RFC 3668.
    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-
    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
    The list of Internet-Draft Shadow Directories can be accessed at
    This document is an informative reference discussing strategies for
    using DCCP as the transport protocol for various applications.  The
    focus is on how applications can make use of the capabilities, and
    deal with the idiosyncrasies, of DCCP.  Of particular interest is how
    UDP applications, which have traditionally ignored congestion control
    issues, can adapt to a congestion controlled transport.
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 Table of Contents
    1. Introduction...................................................3
       1.1 Candidate Applications.....................................3
       1.2 Which CCID?................................................4
       1.3 Document Organization......................................5
    2. Streaming Media Applications...................................5
       2.1 Types of Media Applications................................6
       2.2 Stream Switching...........................................6
       2.3 Media Buffers..............................................7
       2.4 CCID Choice................................................7
       2.5 Variable Rate Media Streams................................7
       2.6 TFRC Basics................................................8
       2.7 First Attempt -- One-way Pre-recorded Media................9
          2.7.1 Strategy 1............................................9
          2.7.2 Issues With Strategy 1...............................11
       2.8 Second Try -- One-way Live Media..........................12
          2.8.1 Strategy 2...........................................12
          2.8.2 Issues with Strategy 2...............................13
       2.9 One More Time -- Two-way Interactive Media................14
          2.9.1 Strategy 3...........................................14
          2.9.2 Issues with Strategy 3...............................14
          2.9.3 A Thought Experiment.................................15
          2.9.4 Fairness.............................................17
    3. Interactive Games.............................................18
       3.1 Rate Limiting.............................................19
          3.1.1 Idleness Problems....................................19
       3.2 Partial Reliability.......................................20
       3.3 Sequence Numbers..........................................21
    4. Miscellaneous Capabilities....................................21
       4.1 Path MTU Discovery........................................21
       4.2 Mobility and Multihoming..................................22
       4.3 Partial Checksums and Payload Checksums...................22
       4.4 ECN Support...............................................23
       4.5 Slow Receiver Option......................................23
       4.6 Detecting Lost Application Data...........................23
    5. Security Considerations.......................................24
    6. IANA Considerations...........................................24
    7. Thanks........................................................24
    8. Informative References........................................25
    9. Author's Address..............................................26
    10. Full Copyright Statement.....................................26
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 1. Introduction
    The Datagram Congestion Control Protocol (DCCP), as defined in
    [DCCP], is a transport protocol that implements a congestion-
    controlled unreliable service.  Currently, there are two congestion
    control algorithms for DCCP, referred to by Congestion Control
    Identifiers (CCIDs).  CCID2 is a TCP-like additive increase
    multiplicative decrease (AIMD) algorithm [CCID2].  CCID3 is an
    implementation of TCP-Friendly Rate Control (TFRC) [RFC 3448],
    [CCID3].  The congestion control algorithm in effect for each
    direction of data transfer is chosen at connection setup time.
    Many applications that currently use UDP [RFC 768] are candidates for
    DCCP.  The main difference applications will see between UDP and DCCP
    is congestion control.  Because it is complicated to get right, many
    UDP applications ignore or greatly simplify congestion control
    issues, even though this can lead to application and network
    misbehavior.  Because of this, some UDP applications employ
    strategies that are unsuitable for a congestion-controlled transport.
    Adapting these applications to DCCP will likely require some new
    modes of thought.
    This document explores issues to consider and strategies to employ
    when adapting or creating applications to use DCCP.  The approach
    here is one of successive refinement.  Strategies are described and
    their strengths and weaknesses are explored.  New strategies are then
    presented that improve on the previous ones and the process iterates.
    The intent is to illuminate the issues, rather than to jump to
    solutions, in order to provide guidance to application designers.
    The reader is assumed to be familiar with the mechanisms of DCCP and
    the CCIDs.
 1.1 Candidate Applications
    Many applications that currently use UDP, and some that use TCP, are
    candidates for DCCP.  Basically, applications with some of the
    following characteristics should consider DCCP:
     o  There are flows of packets from one end system to another that
        are larger than a few handfuls of packets.
     o  Lost packets should be ignored or replaced with updated data --
        timeliness is preferred over reliability.
     o  There is a preference for immediate delivery of packets as they
        arrive over strict in-order delivery by waiting for out-of-order
        packets to arrive.
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     o  There is a preference for immediate transmission of small chunks
        of data.
     o  The large transmission rate variations that are typical of TCP-
        like congestion control are problematic.
    Major examples of applications that fit these characteristics are
    streaming media, including Internet telephony, and multiplayer on-
    line games.  Without DCCP, these applications either use UDP and
    mostly ignore or greatly simplify congestion control issues, or use
    TCP and live with the timeliness and rate variation problems.
    Another major use of UDP involves one-shot request-response cycles,
    with packet flows limited to at most a few packets in each direction
    (for example, DNS, SNMP, MGCP).  These applications are unlikely to
    benefit from DCCP.
 1.2 Which CCID?
    CCID2, TCP-like congestion control, uses a packet-oriented
    modification of TCP's SACK-based Additive-Increase-Multiplicative-
    Decrease (AIMD) congestion control [RFC 3517].  CCID2 uses a
    congestion window (the maximum number of packets in flight) to limit
    the transmitter.  The congestion window is increased by one for each
    acknowledged packet, or for each window of acknowledged packets,
    depending on the phase of operation.  If any packet is dropped (or
    ECN-marked [ECN]; for simplicity in the rest of the document assume
    that "dropped" equals "dropped or ECN-marked"), the congestion window
    is halved.  This produces a characteristic saw-tooth wave variation
    in throughput, where the throughput increases linearly up to the
    network capacity and then drops abruptly (roughly shown in Figure 1).
                  |      /|    /|    /|    /|    /
                  |     / |   / |   / |   / |   /
        Throughput|    /  |  /  |  /  |  /  |  /
                  |   /   | /   | /   | /   | /
                  |  /    |/    |/    |/    |/
    Figure 1: Characteristic throughput for TCP-like congestion control.
    With CCID3, TCP-Friendly Rate Control (TFRC), the immediate response
    to packet drops is less dramatic.  To compensate for this CCID3 is
    less aggressive in probing for new capacity after a loss.  The
    characteristic throughput graph for a CCID3 connection looks like a
    flattened sine wave (extremely roughly shown in Figure 2).
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                  |    --        --        --
                  |   /  \      /  \      /  \
        Throughput|  /    \    /    \    /    \
                  | /      \  /      \  /      \
                  |-        --        --        -
    Figure 2: Characteristic throughput for TFRC congestion control.
    The CCID that is appropriate for a given application depends on the
    tradeoff between the application's sensitivity to abrupt rate
    changes, and its need to rapidly consume available capacity.
    Applications that simply want to transfer as much data in the
    shortest time possible probably should use CCID2.  Applications that
    have some natural limits on transmission rates are probably better
    served by CCID3.  Applications that perform some progressive display
    of incoming data, and want to avoid abrupt variations in the update
    rate, would prefer CCID3.
 1.3 Document Organization
    Sections 2 and 3 explore the specific issues involved in using DCCP
    for streaming media and interactive game applications, respectively.
    Many of the issues discussed in these sections are also applicable to
    other applications.  Section 4 discusses capabilities of DCCP not
    mentioned in the application-specific sections that can be used to
    offload features that are normally built at the application layer for
    UDP-based applications.  Section 5 discusses the security-related
    features of DCCP.
 2. Streaming Media Applications
    The canonical streaming media application emits fixed-sized (often
    small) RTP/UDP packets at a regular interval [RFC 3550].  It relies
    on the network to deliver the packets to the receiver in roughly the
    same regular interval.  Often, the transmitter operates in a fire-
    and-forget mode, receiving no indications of packet delivery.  This
    often still holds true even if RTCP [RFC 3550] is used to get
    receiver information; it's rare that the RTCP reports trigger changes
    in the transmitted stream.
    The IAB has expressed concerns over the stability of the Internet if
    these applications become too popular with regard to TCP-based
    applications [IABCONG].  They suggest that media applications should
    monitor their packet loss rate, and abort if they exceed certain
    thresholds.  Unfortunately, up until this threshold is reached, the
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    network, the media applications and the other applications are
    experiencing considerable duress.
    DCCP offers an opportunity for media applications to satisfy the IAB
    concerns in a way that is better for both the network and the
    applications themselves.  However, this does require some rather
    significant shifts from current practices.  These shifts are explored
    in this section.
 2.1 Types of Media Applications
    While all streaming media applications have some characteristics in
    common (e.g. data must arrive at the receiver at some minimum rate
    for reasonable operation), other characteristics (e.g. tolerance of
    delay) vary considerably from application to application.  For the
    purposes of this document, it's useful to divide streaming media
    applications into three subtypes:
     o  One-way pre-recorded media
     o  One-way live media
     o  Two-way interactive media
    The relevant difference, as far as this discussion goes, between
    recorded and live media is that recorded media can be transmitted as
    fast as the network allows (assuming adequate buffering at the
    receiver) -- it could be viewed as a special file transfer operation.
    Live media can't be transmitted faster than the rate that it's
    The difference between one-way and two-way media is the sensitivity
    to delay.  For one-way applications, delays from transmit at the
    sender to playout at the receiver of several or even tens of seconds
    are acceptable.  For two-way applications delays from transmit to
    playout of as little as 150 to 200 ms are often problematic [XTIME].
 2.2 Stream Switching
    The discussion here assumes that media transmitters are able to
    provide their data in a number of encodings with various bit rate
    requirements, as described in [SWITCH], and are able to dynamically
    change between these encodings with low overhead.  It also assumes
    that switching back and forth between coding rates does not cause
    excessive user annoyance.
    Given the current state of codec art, these are big assumptions.  The
    algorithms and results described here, however, hold even if the
    media sources can only supply media at one rate.  Obviously the
    statements about switching encoding rates don't apply, and an
    application with only one encoding rate behaves as if it is
    simultaneously at its minimum and maximum rate.
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    For convenience in the discussion below, assume that all media
    streams have two encodings, a high bit rate and a low bit rate,
    unless otherwise indicated.
 2.3 Media Buffers
    Many of the strategies below make use of the concept of a media
    buffer.  A media buffer is a first-in-first-out queue of media data.
    The buffer is filled by some source of data and drained by some sink.
    It provides rate and jitter compensation between the source and the
    Media buffer contents are measured in seconds of media play time, not
    bytes or packets.  Media buffers are completely application-level
    constructs and are separate from transport-layer transmit and receive
 2.4 CCID Choice
    For two-way media applications, CCID3 (TFRC) is by far the most
    appropriate congestion control algorithm.  Since CCID2 halves the
    transmit rate when packets are lost, the media encoding steps would
    need to be at least a factor of two apart.  If the encoding steps are
    less than a factor of two, the application will need to additive
    increase up after a packet loss; two-way media applications will
    rarely be able to afford the delays necessary.  With the smooth
    variations in CCID3 the application has much more freedom to choose
    encoding rate steps.
    One-way applications could possibly use CCID2, since they can
    tolerate more delay during the rate shifts.  However, one-way
    applications often have self-imposed limits on maximum transmission
    rates that mean they will be unable to reap the higher throughput
    benefits of CCID2.
 2.5 Variable Rate Media Streams
    The canonical media codec encodes its media as a constant rate bit
    stream.  As the technology has progressed from its time-division
    multiplexing roots, this constant rate stream has become not so
    constant.  Voice codecs often employ silence suppression, where the
    stream (in at least one direction) goes totally idle for sometimes
    several seconds while one side listens to what the other side has to
    say.  When the one side wants to start talking again, the codec
    resumes sending immediately at its "constant" rate.
    Video codecs similarly employ what could be called "stillness"
    suppression.  Often these codecs effectively transmit the changes
    from one video frame to another.  When there is little change from
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    frame to frame (as when the background is constant and a talking head
    is just moving its lips) the amount of information to send is small.
    When there is a major motion, or change of scene, much more
    information must be sent.  For some codecs, the variation from the
    minimum rate to the maximum rate can be a factor of ten [MPEG4].
    Unlike voice codecs, though, video codecs typically never go
    completely idle.
    These abrupt jumps in transmission rate are problematic for any
    congestion control algorithm.  A basic tenet of all existing
    algorithms assumes that increases in transmission rate must be
    gradual and smooth to avoid damaging other connections in the
    CCID3 uses a maximum rate of two packets per RTT after an idle
    period.  This rate might support immediate restart of voice data
    after a silence period, at least when the RTT is in the suitable
    range for two-way media.  More problematic are the factor of ten
    variations in video codecs.  In some circumstances, CCID3 (and CCID2)
    allows an application to double its transmit rate over one RTT
    (assuming no recent packet loss events), but an immediate ten times
    increase is not possible.
 2.6 TFRC Basics
    The job of mapping media applications onto the packet formats and
    connection handshake mechanisms of DCCP proper is straightforward,
    and won't be dealt with here.  The problem for this section is how
    media stream applications can make use of and adapt to the
    idiosyncrasies of TCP-Friendly Rate Control (TFRC), as implemented in
    Data streams controlled by TFRC must vary their transmission rates in
    ways that, at first blush, seem at odds with common media stream
    transmission practices.  Some particular considerations are:
     o  Slow Start -- A connection starts out with a transmission rate of
        up to four packets per round trip time (RTT).  After the first
        RTT, the rate is doubled each RTT until a packet is lost.  At
        this point the transmission rate is halved and we enter the
        additive increase phase of operation.  It's likely that in many
        situations the initial transmit rate is slower than the lowest
        bit rate encoding of the media.  This will require the
        application to deal with a ramp up period.
     o  Capacity Probing and Lost Packets -- If the application transmits
        for some time at the maximum rate that TFRC will allow without
        packet loss, TFRC will continuously raise the allowed rate until
        a packet is lost.  This means that, in many circumstances, if an
        application wants to transmit at the maximum possible rate,
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        packet loss will not be an exceptional event, but will happen
        routinely in the course of probing for more capacity.
     o  Idleness Penalty -- TFRC follows a "use it or lose it" policy.
        If the transmitter goes idle for a few RTTs, as it would if, for
        instance, silence suppression were being used, the transmit rate
        returns to two packets per RTT, and then doubles every RTT until
        the previous rate is achieved.  This can make restarting after a
        silence suppression interval problematic.
     o  Contentment Penalty -- TFRC likes to satisfy greed.  If you are
        transmitting at the maximum allowed rate, TFRC will try to raise
        that rate.  However, if your application is transmitting below
        the maximum allowed rate, the maximum allowed rate will not be
        increased higher than twice the current transmit rate, no matter
        how long it has been since the last increase.  This can create
        problems when attempting to shift to a higher rate encoding, or
        with video codecs that vary the transmission rate with the amount
        of movement in the image.
     o  Packet Rate, not Bit Rate -- TFRC controls the rate that packets
        may enter the network, not bytes.  To respond to a lowered
        transmit rate you must reduce the packet transmission rate.
        Making the packets smaller while still keeping the same packet
        rate will not be effective.
     o  Smooth Variance of Transmit Rate -- The strength and purpose of
        TFRC (over TCP-Like Congestion Control, CCID2) is that it
        smoothly decreases the transmission rate in response to recent
        packet loss events, and smoothly increases the rate in the
        absence of loss events.  This smoothness is somewhat at odds with
        most media stream encodings, where the transition from one rate
        to another is often a step function.
 2.7 First Attempt -- One-way Pre-recorded Media
    The first strategy is suitable for use with pre-recorded media, and
    takes advantage of the fact that the data for pre-recorded media can
    be transferred to the receiver as fast as the network will allow it,
    assuming that the receiver has sufficient buffer space.
 2.7.1 Strategy 1
    Assume a recorded program resides on a media server, and the server
    and its clients are capable of stream switching between two encoding
    rates, as described in section 2.2.
    The client (receiver) implements a media buffer as a playout buffer.
    This buffer is potentially big enough to hold the entire recording.
    The playout buffer has three thresholds: a low threshold, a playback
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    start threshold, and a high threshold, in order of increasing size.
    These values will typically be in the several to tens of seconds
    range.  The buffer is filled by data arriving from the network and
    drained at the decoding rate necessary to display the data to the
    user.  Figure 3 shows this schematically.
                              high threshold
                                  |  playback start threshold
                                  |    |  low threshold
    +-------+                     |    |    |
    | Media |  transmit at    +---v----v----v--+
    | File  |---------------->| Playout buffer |-------> display
    |       |  TFRC max rate  +----------------+ drain at
    +-------+                 fill at network    decode rate
                              arrival rate
    Figure 3: One-way pre-recorded media.
    During the connection the server needs to be able to determine the
    depth of data in the playout buffer.  This could be provided by
    direct feedback from the client to the server, or the server could
    estimate its depth (e.g. the server knows how much data has been
    sent, and how much time has passed).
    To start the connection, the server begins transmitting data in the
    high bit rate encoding as fast as TFRC allows.  Since TFRC is in slow
    start, this is probably too slow initially, but eventually the rate
    should increase to fast enough and more.  As the client receives data
    from the network it adds it to the playout buffer.  Once the buffer
    depth reaches the playback start threshold, the receiver begins
    draining the buffer and playing the contents to the user.
    If the network has sufficient capacity, TFRC will eventually raise
    the transmit rate to greater than necessary to keep up with the
    decoding rate, the playout buffer will back up as necessary, and the
    entire program will eventually be transferred.
    If the TFRC transmit rate never gets fast enough, or a loss event
    makes TFRC drop the rate, the receiver will drain the playout buffer
    faster than it is filled.  If the playout buffer drops below the low
    threshold the server switches to the low bit rate encoding.  Assuming
    that the network has a bit more capacity than the low bit rate
    requires, the playout buffer will begin filling again.
    When the buffer crosses the high threshold the server switches back
    to the high encoding rate.  Assuming that the network still doesn't
    have enough capacity for the high bit rate, the playout buffer will
    start draining again.  When it reaches the low threshold the server
    switches again to the low bit rate encoding.  The server will
    oscillate back and forth like this until the connection is concluded.
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    If the network has insufficient capacity to support the low bit rate
    encoding, the playout buffer will eventually drain completely, and
    playback will need to be paused until the buffer refills to some
    level (presumably the playback start level).
    Note that, in this scheme, the server doesn't need to explicitly know
    the rate that TFRC has determined; it simply always sends as fast as
    TFRC allows (perhaps alternately reading a chunk of data from disk
    and then blocking on the socket write call until it's transmitted).
    TFRC shapes the stream to the network's requirements, and the playout
    buffer feedback allows the server to shape the stream to the
    application's requirements.
 2.7.2 Issues With Strategy 1
    The advantage of this strategy is that it provides insurance against
    an unpredictable future.  Since there's no guarantee that a currently
    supported transmit rate will continue to be supported, the strategy
    takes what the network is willing to give when it's willing to give
    it.  The data is transferred from the server to the client perhaps
    faster than is strictly necessary, but once it's there no network
    problems (or new sources of traffic) can affect the display.
    Silence suppression can be used with this strategy, since the
    transmitter doesn't actually go idle during the silence û- it just
    gets further ahead.
    One obvious disadvantage, if the client is a "thin" device, is the
    large buffer at the client.  A subtler disadvantage involves the way
    TFRC probes the network to determine its capacity.  Basically, TFRC
    does not have an a priori idea of what the network capacity is; it
    simply gradually increases the transmit rate until packets are lost,
    then backs down.  After a period of time with no losses, the rate is
    gradually increased again until more packets are lost.  Over the long
    term, the transmit rate will oscillate up and down, with packet loss
    events occurring at the rate peaks.
    This means that packet loss will likely be routine with this
    strategy.  For any given transfer, the number of lost packets is
    likely to be small, but non-zero.  Whether this causes noticeable
    quality problems depends on the characteristics of the particular
    codec in use.
    If the DCCP connection uses the Ack Vector option for the DCCP-Ack
    packets in the return direction, DCCP will be able to tell which
    packets are lost.  An API could inform the application of lost
    packets, and they could be retransmitted at the application layer.
    See section 3.2 for more on this.
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 2.8 Second Try -- One-way Live Media
    With one-way live media you can only transmit the data as fast as
    it's created, but end-to-end delays of several or tens of seconds are
    usually acceptable.
 2.8.1 Strategy 2
    Assume that we have a playout media buffer at the receiver and a
    transmit media buffer at the sender.  The transmit buffer is filled
    at the encoding rate and drained at the TFRC transmit rate.  The
    playout buffer is filled at the network arrival rate and drained at
    the decoding rate.  The playout buffer has a playback start threshold
    and the transmit buffer has a switch encoding threshold and a discard
    data threshold.  These thresholds are on the order of several to tens
    of seconds.  Switch encoding is less than discard data, which is less
    than playback start.  Figure 4 shows this schematically.
                    discard data
                      |  switch encoding
                      |   |                 playback start
                      |   |                   |
    media   +---------v---v---+          +----v-----------+
    ------->| Transmit buffer |--------->| Playout buffer |---> display
    source  +-----------------+ transmit +----------------+
            fill at             at TFRC rate          drain at
            encode rate                               decode rate
    Figure 4: One-way live media.
    At the start of the connection, the sender places data into the
    transmit buffer at the high encoding rate.  The buffer is drained at
    the TFRC transmit rate, which at this point is in slow-start and is
    probably slower than the encoding rate.  This will cause a backup in
    the transmit buffer.  Eventually TFRC will slow-start to a rate
    slightly above the rate necessary to sustain the encoding rate
    (assuming the network has sufficient capacity).  When this happens
    the transmit buffer will drain and we'll reach a steady state
    condition where the transmit buffer is normally empty and we're
    transmitting at a rate that is probably below the maximum allowed by
    Meanwhile at the receiver, the playout buffer is filling, and when it
    reaches the playback start threshold playback will start.  After TFRC
    slow-start is complete and the transmit buffer is drained, this
    buffer will reach a steady state where packets are arriving from the
    network at the encoding rate (ignoring jitter) and being drained at
    the (equal) decoding rate.  The depth of the buffer will be the
    playback start threshold plus the maximum depth of the transmit
    buffer during slow start.
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    Now assume that network congestion (packet losses) forces TFRC to
    drop its rate to below that needed by the high encoding rate.  The
    transmit buffer will begin to fill and the playout buffer will begin
    to drain.  When the transmit buffer reaches the switch encoding
    threshold, the sender switches to the low encoding rate, and converts
    all of the data in the transmit buffer to low rate encoding.
    Assuming that the network can support the new, lower, rate (and a
    little more) the transmit buffer will begin to drain and the playout
    buffer will begin to fill.  Eventually the transmit buffer will empty
    and the playout buffer will be back to its steady state level.
    At this point (or perhaps after a slight delay) the sender can switch
    back to the higher rate encoding.  If the new rate can't be sustained
    the transmit buffer will fill again, and the playout buffer will
    drain.  When the transmit buffer reaches the switch encoding
    threshold the sender goes back to the lower encoding rate.  This
    oscillation continues until the stream ends or the network is able to
    support the high encoding rate for the long term.
    If the network can't support the low encoding rate, the transmit
    buffer will continue to fill (and the playout buffer will continue to
    drain).  When the transmit buffer reaches the discard data threshold,
    the sender must discard data from the transmit buffer for every data
    added.  Preferably, the discard should happen from the head of the
    transmit buffer, as these are the stalest data, but the application
    could make other choices (e.g. discard the earliest silence in the
    buffer).  This discard behavior continues until the transmit buffer
    falls below the switch encoding threshold.  If the playout buffer
    ever drains completely, the receiver should fill the output with
    suitable material (e.g. silence or stillness).
    Note that this strategy is also suitable for one-way pre-recorded
    media, as long as the transmit buffer is only filled at the encoding
    rate, not at the disk read rate.
 2.8.2 Issues with Strategy 2
    Silence suppression can be a problem with strategy 2.  If the
    encoding rate is low enough -- if it's in the range used by most
    telephony applications -- the ramp up to the required rate can be
    short or nonexistent, and silence suppression can be used.  If the
    encoding rate is in the range of high-quality music or video, then
    silence or stillness suppression could cause problems, although the
    playout buffer might provide sufficient elasticity to overcome the
    rate ramping issues.
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 2.9 One More Time -- Two-way Interactive Media
    Two-way interactive media is characterized by its low tolerance for
    end-to-end delay, usually requiring less than 200 ms., including
    jitter buffering at the receiver.  Rate adapting buffers will insert
    too much delay and the slow start period is likely to be noticeable,
    so another strategy is needed.
 2.9.1 Strategy 3
    To start, the calling party sends an INVITE (loosely using SIP [RFC
    3261] terminology) indicating the IP address and DCCP port to use for
    media at its end.  Without informing the called user, the called
    system responds to the INVITE by connecting to the calling party
    media port.  Both end systems then begin exchanging test data, at the
    (slowly increasing) rate allowed by TFRC.  The purpose of this test
    data is to see what rate the connection can be ramped up to.  If a
    minimum acceptable rate cannot be achieved within some time period,
    the call is cleared (conceptually, the calling party hears "fast
    busy" and the called user is never informed of the incoming call).
    Note that once the rate has ramped up sufficiently for the highest
    rate codec there's no need to go further.
    If an acceptable rate can be achieved (in both directions), the
    called user is informed of the incoming call.  The test data is
    continued during this period.  Once the called user accepts the call,
    the test data is replaced by real data at the same rate.
    If congestion is encountered during the call, TFRC will reduce its
    allowed sending rate.  When that rate falls below the codec currently
    in use, the sender switches to a lower rate codec, but should pad its
    transmission out to the allowed TFRC rate.  If the TFRC rate
    continues to fall past the lowest rate codec, the sender must discard
    packets to conform to that rate.
    If the network capacity is sufficient to support one of the lower
    rate codecs, eventually the congestion will clear and TFRC will
    slowly increase the allowed transmit rate.  The application should
    increase its transmission padding to keep up with the increasing TFRC
    rate.  The application switches back to the higher rate codec when
    the TFRC rate reaches a sufficient value.
    Note that the receiver would normally implement a short playout
    buffer (with playback start on the order of 100 ms) to smooth out
    jitter in the packet arrival gaps.
 2.9.2 Issues with Strategy 3
    An obvious issue with strategy 3 is the post-dial call connection
    delay imposed by the slow-start ramp up.  This is perhaps less of an
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    issue for two-way video applications, where post-dial delays of
    several seconds are accepted practice.  For telephony applications,
    however, post-dial delays significantly greater than a second are a
    problem, given that users have been conditioned to that behavior by
    the public telephone network.  On the other hand, the four packets
    per RTT initial transmit rate is likely to be sufficient for many
    telephony applications, and the ramp up will be very quick.
    Strategy 3 might not support silence suppression well.  During the
    silence period, TFRC will lower the transmit rate to two packets per
    RTT.  After the silence the application will need to ramp up to the
    necessary data sending rate.
    However, there are many telephony codecs and network situations where
    two packets per RTT is a sufficient data rate.  An application that
    knows it's in this situation could conceivably use silence
    suppression, knowing that there's no ramp up needed when it returns
    to transmission.
    The next section explores some more subtle issues.
 2.9.3 A Thought Experiment
    In [IABCONG], the authors describe a VoIP demonstration given at the
    IEPREP working group meeting at the 2002 Atlanta IETF.  A VoIP call
    was made from a nearby hotel room to a system in Nairobi, Kenya.  The
    data traveled over a wide variety of interfaces, the slowest of which
    was a 128 kbps link between an ISP in Kenya and several of its
    subscribers.  The media data was contained in typical RTP/UDP
    framing, and, as is the usual current practice, the transmitter
    transmitted at a constant rate with no feedback for or adjustment to
    loss events (although forward error correction was used).  The focus
    of [IABCONG] was on the fairness of this behavior with regard to TCP
    applications sharing that Kenyan link.
    Let's imagine this situation if we replace the RTP/UDP media
    application with an RTP/DCCP application using strategy 3.  Imagine
    the media application has two encoding rates that it can switch
    between at little cost, a high-rate at 32 kbps and a low-rate at 18
    kbps (these are bits-on-the-wire per second).  Furthermore, at the
    low rate, the receiver can withstand packet loss down to 14 kbps
    before the output is unintelligible.  These numbers are chosen more
    for computational convenience than to represent real codecs, but they
    should be conceptually representative.
    Let's also imagine that there is a TCP-based application, say large
    file transfer, whose connection lifetime is of the same order of
    magnitude as a voice call.
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    Now imagine that one media connection is using the Kenyan link.  This
    connection will slow-start up to 32 kbps and then self-limit at that
    rate.  The TFRC maximum rate will continue to increase up to 64 kbps,
    and then hold because of the lack of demand from the application.
    Now add one TCP connection to the Kenyan link.  Ideally, the media
    connection would receive 32 kbps and the TCP application would get
    the remaining 96 kbps.
    The situation is not quite that simple, though.  A significant
    difference between the two applications is their degree of
    contentment or greediness.  If the media application can achieve 32
    kbps throughput, it's satisfied, and won't push for more.  The TCP
    application, on the hand, is never satisfied with its current
    throughput and will always push for more.
    Periodically, TCP will probe for more throughput, causing congestion
    on our link, and eventually lost packets.  If some of the media
    packets are lost, DCCP (through TFRC) will back off its transmit
    rate.  Since that rate is twice what the application actually needs,
    and TFRC makes gradual rate changes, it will be a while before the
    rate is reduced to below 32 kbps.  It's likely that the TCP
    connection will experience packet loss before that, though, and halve
    its rate, relieving the congestion.
    The media connection is therefore somewhat resilient to the TCP
    probing.  In the steady state, the media connection will transmit at
    a constant 32 kbps (with occasional lost packets), while the TCP
    connection will vary between 48 and 96 kbps.
    Now let's consider what happens when a second media application
    connection is added to the existing situation.  During the new
    connection's test data phase, it will encounter congestion very
    quickly, but, after a period of time, it should muscle its way in and
    the three connections will coexist, with the media applications
    receiving 32 kpbs apiece and the TCP connection getting between 32
    and 64 kbps.  We'll assume that this jostling period is within the
    bounds of the acceptable post-dial delay, and the new connection is
    When we add one more media connection we'll end up with approximately
    32 kbps per media connection and between 16 and 32 kbps for the TCP
    connection.  With the three TFRC and one TCP connection all jostling
    with each other, some of the media streams will momentarily drop
    below 32 kbps and need to switch to the low encoding rate.  During
    that time it's possible for the TCP application to get more than 32
    kbps, but eventually things will even out again.
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    Adding a fourth media connection will leave approximately 25 kbps per
    connection, forcing the media connections to all permanently switch
    to the low encoding rate (with nominally 7 kbps of padding).
    By the time we have six media connections (plus the one TCP
    connection) we have reduced the per-connection bandwidth share to
    just over 18 kbps.  At this point some media connections are
    discarding packets as the connections jostle for bandwidth and some
    TFRC rates drop below 18 kbps.
    If we try to introduce another media stream connection, reducing the
    per-connection share further to 16 kbps, the new connection won't be
    able to achieve a sufficient rate during the test period, and the
    connection will be blocked.  After a moment of packet discard in the
    existing connections (during the probe period), things will return
    back to the 18 kbps per-connection state.  We won't be able to add a
    new media connection until one of the existing connections
    But nothing prevents new TCP connections from being added.  By the
    time we have three more TCP connections (for a total of six media
    connections and four TCP connections) per-connection share has
    reduced to just under 13 kbps, and the media applications are
    unusable.  The TCP applications, although slow, are likely still
    useable, and will continue a graceful decline as more TCP connections
    are added.
 2.9.4 Fairness
    The model used above for the interactions of several TCP and TFRC
    streams -- roughly equal sharing of the available capacity -- is of
    course a highly simplified version of the real world interactions.  A
    more detailed discussion is presented in [EQCC], however, it seems
    that the model used here is adequate for the purpose.
    The behavior described above seems to be eminently fair to TCP
    applications -- a TCP connection gets the same (or more) bandwidth
    over a congested link that it would get if there were only other TCP
    The behavior also seems fair to the network.  It avoids persistent
    packet loss in the network, as occurs in the behavior model in
    [IABCONG], by discarding media data at the transmitter.
    Just how fair this is to media applications is debatable, but it
    seems better than the method of feeding packets into the network
    regardless of the congestion situation, but terminate if not enough
    packets are delivered, as described in [IABCONG].  A media
    application can choose to not start a connection, if at the moment
    there are insufficient network resources.  A media connection that
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    encounters major congestion after starting up can choose to wait out
    the congestion, rather than terminate, since the excess packets are
    discarded before entering the network.  The application can perhaps
    improve quality in a congested situation by discarding packets
    intelligently, rather than allowing the network to discard randomly.
    What the likelihood is of a user hanging on through this situation
    depends on the length and severity of the incident.
 3. Interactive Games
    Another application area that could benefit from the use of DCCP is
    real-time interactive, multi-player, on-line games.  These games
    usually consist of a set of clients (players) that display the game
    environment to and take commands from a user, and a server computer
    that maintains the entire game state.  Massively Multi-Player (MMP)
    games can use a grid of server computers connected in a peer-to-peer
    fashion to distribute the game state computation and increase the
    number of simultaneous players into the hundreds of thousands or even
    millions [MMPGRID].
    Communications between the various components (client-to-server,
    server-to-client and server-to-server) usually take two forms.
    Transient data would like to be delivered as soon as possible, but if
    lost will be replaced by later data.  For example, a player moving
    through the game space will send a continuous stream of "move-to"
    messages.  If one is lost it isn't worth retransmitting -- the next
    "move-to" message will give better information.  On the other hand,
    actions that represent permanent changes in the game state must be
    reliably delivered (e.g., "you're-dead" messages).
    It's important for the receiver to able to determine the sequence of
    the received data, although data received out of order should be
    delivered immediately, without waiting for the missing data.  If two
    "move-to" messages are reordered it's important to know which one was
    sent first for the object to end up in the right location, but if the
    two moves are delivered out of order the late one can be discarded.
    Messages are often given an application-layer header, which usually
    includes sequence numbers and reliability requirements.  Each message
    is usually encapsulated in a single UDP packet and transient
    information is transmitted in fire-and-forget mode.  Persistent data
    requires an acknowledgement from the receiver, and a retransmit timer
    at the sender.
    The devices would like to transmit their state changes as frequently
    as possible, as this leads to smoother rendering of the changes at
    the user display.  However, a device can often generate data faster
    than the network or receiver can handle it, so some rate limits must
    be applied.  Many applications handle this by simply setting a peak
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    data rate limit, rather than dynamically responding to network
    If DCCP were used instead of UDP, a multi-player game application
    could offload much of work required for the functions of rate
    limiting, partial reliability and sequencing.
 3.1 Rate Limiting
    The main purpose of DCCP is to provide congestion control for
    unreliable streams -- to not only limit the transmit rate of an
    application in times of congestion, but to also provide the
    application with the most throughput it's entitled to.  Congestion
    control is difficult to get right, and many application-level
    implementations greatly simplify things, often leading to various
    inefficiencies.  As mentioned above, many applications simply set a
    peak data rate limit.  This makes extra capacity unavailable, and
    causes unnecessary congestion in the network when sufficient capacity
    isn't available.
    By using DCCP, a multi-player game application could completely
    offload congestion control considerations, and benefit from a much
    more complete implementation than would normally be provided at the
    application layer, including support for ECN and the ability to use
    all of the available bandwidth.  The application's flow control
    algorithm could be to simply write a message whenever the DCCP socket
    is ready.  DCCP would manage probing the network for available
    capacity and backing off in the presence of network congestion or
    server load (via the slow receiver indications).
    The CCID to use would depend on tradeoffs between smoothness and
    throughput.  CCID3 would provide less dramatic changes in the rate of
    state transmissions (and receptions), perhaps eliminating abrupt
    changes in the display update rate.  CCID2 would more rapidly consume
    available capacity, perhaps leading to noticeable changes in the
    update rate, but more updates at the peak.
 3.1.1 Idleness Problems
    Often the amount of data that needs to be transmitted depends on the
    amount of user activity at a given point.  When the user is idle,
    little needs to be sent.  When the user is active, a great deal can
    be sent.  Often the transition between these two states is immediate.
    Even though it has long been known to be harmful for the Internet
    [CONGAVOID], one of the simplifications often made by applications
    implementing congestion control is to ignore the issues of fast
    startup or restart after idle.  Often these applications simply start
    transmitting immediately at a high rate, without ramping up.
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    The DCCP congestion control algorithms enforce slow-start and
    restart.  For applications used to ignoring these issues, however,
    this could lead to unexpected behavior.  In particular, the
    application could appear less responsive to quick shifts in the
    activity rate.
 3.2 Partial Reliability
    An application that implements reliability on top of UDP for some (or
    all) of its packets must implement application level acknowledgements
    and retransmit timers.  The most efficient implementations of
    retransmission timers take the current RTT into account.  This is
    often difficult to do at the application layer, so RTT is often
    ignored in favor of a pre-configured value likely to be "large
    enough".  This can lead to long delays in the face of lost packets.
    Although DCCP doesn't implement retransmissions, the nature of
    congestion control forces it to implement most of what is required
    for reliable delivery.  An application using DCCP could simplify (and
    improve) its implementation of reliability by offloading some of the
    functions to DCCP.
    By its nature, congestion control must know if packets are lost.  For
    CCID2, which requires the use of the Ack Vector option, it even knows
    specifically which packets are lost.  If a DCCP implementation were
    to make this information available to an application, it could ease
    the burden of implementing reliability.
    Say, for example, that a DCCP allowed the application to request
    delivery indication for certain packets.  DCCP would inform the
    application when either the packet was acknowledged, or DCCP had
    decided that the packet was dropped.  The application would need to
    save the packet for retransmission, but it wouldn't need to maintain
    a retransmission timer or implement application-layer
    Because DCCP maintains connection state, such as current RTT, it can
    often make this decision more efficiently than the application.  By
    tying the retransmission timeout to the current RTT, lost packets can
    be retransmitted sooner, with less chance of unneeded
    retransmissions.  In addition, application layer acknowledgements
    would often be redundant with DCCP acknowledgements.
    When CCID3 is used, the application could request the use of Ack
    Vector options and receive the same service.  On the other hand, the
    normal use in CCID3 of the Loss Event Rate option might provide
    sufficient information.  In this case the reliable packets would be
    considered delivered if there were no losses in the interval that
    contained the packet, and not delivered if there were losses.  This
    might lead to some unnecessary retransmissions, but the
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    acknowledgement savings (Loss Event Rate options are smaller and
    simpler to deal with than Ack Vector options) might make up for it.
 3.3 Sequence Numbers
    Game applications using UDP normally include sequence numbers in
    their application headers to allow the receiver to detect packet
    reordering.  These application layer sequence numbers would often be
    redundant with the DCCP sequence numbers.  If a DCCP application made
    the DCCP sequence numbers available to receiving applications, the
    application could determine the transmission order.  Note that, since
    DCCP sequence numbers increase for Ack packets as well as Data
    packets, the application wouldn't be able to infer missing packets
    from holes in the delivered sequence.
 4. Miscellaneous Capabilities
    This section covers DCCP capabilities not covered earlier that might
    be unfamiliar to a developer accustomed to UDP communications.
 4.1 Path MTU Discovery
    DCCP mandates the use of Path Maximum Transfer Unit (PMTU) discovery.
    In general, an application will not be allowed to transmit a packet
    larger than the currently known PMTU (or the maximum packet size
    allowed by the CCID in effect).  Applications can be allowed to
    override this restriction (at least for PMTU) and send packets larger
    than the PMTU (but not larger than the CCID maximum).  These large
    packets will of course be fragmented as they travel through the
    network.  However, since most applications would like to avoid
    fragmenting packets, the default DCCP behavior fits nicely.
    UDP applications that wish to avoid packet fragmentation usually
    limit the size of their packets to 576 bytes, even though most
    connection paths can support much larger sizes.  With PMTU discovery
    in DCCP, larger packet sizes can be used safely.
    PMTU discovery in DCCP is based on the use of ICMP "packet too big"
    messages, as defined in [RFC 1911].  The PMTU is initially set to the
    MTU of the first-hop interface.  Transmitted packets have the "don't
    fragment" bit set in the IP header.  If an interface with a lower MTU
    is encountered, the router sends an ICMP Destination Unreachable
    message, with cause set to "fragmentation needed and DF set", to the
    originator (colloquially referred to as a "packet too big" message).
    When the originating DCCP receives the packet too big message, it
    adjusts the PMTU according to [RFC 1911].
    Of course the packet that caused the packet too big message will be
    dropped, and since DCCP provides an unreliable service, it won't be
    retransmitted.  There are also several other known issues with [RFC
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    1911]-style PMTU discovery (e.g., some firewalls block incoming ICMP
    messages).  Applications that would like to use the biggest packets
    possible might want to consider an application-level supplement to
    DCCP's PMTU discovery (e.g, send an initial packet stream in various
    sizes and choose packet size based on the largest packet acknowledged
    by the receiver).
    DCCP implementations should consider providing this procedure for
    applications, based on sending gratuitous DCCP-Sync packets padded
    out to various sizes with Padding Options.  The receiving DCCP will
    acknowledge the DCCP-Sync packets that got through (with normal-sized
    DCCP-Sync packets of its own), and the PMTU can be set to the largest
    acknowledged packet.
 4.2 Mobility and Multihoming
    DCCP supports the ability to change the location of a connection
    endpoint (IP address and port) during the connection.  This
    capability provides simple support for mobile hosts or high-
    availability applications.  The use of mobility must be negotiated at
    connection setup, but the set of possible new addresses is not
    constrained at that time.
    When an endpoint would like to move, it sends a DCCP-Move packet.
    The source address in the IP header is the new address, and the
    source port in the DCCP header is the new port.  The packet also
    includes values for Mobility ID and Identification Option that were
    negotiated at connection setup.  If the other end accepts the change
    it sends a DCCP-Ack to the new address/port; otherwise it sends a
    DCCP-Ack to the old address/port.  The DCCP-Move packet also may
    include user data, but usually data transfer is interrupted until the
    move is complete.  An endpoint may move any number of times during a
    connection.  However, after each move, a new Mobility ID must be
 4.3 Partial Checksums and Payload Checksums
    Applications that would rather receive corrupted data than have it
    discarded by the network can use the DCCP partial checksum
    capability.  This capability allows the sender to specify how much of
    the user data should be covered by the DCCP header checksum.  Options
    include all of the user data, none, or up to 14 32-words (this final
    option is considered experimental).  DCCP implementations should
    allow senders to specify the checksum coverage per packet and allow
    receivers to specify the minimum checksum coverage they'd like to
    receive.  Defaults for both of these values should be to cover all
    Of course this is really only useful if there are link layers that
    detect that a DCCP packet is being sent and provide strong error
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    checking only for the portion of the packet covered by the DCCP
    checksum.  Only time will tell if any link layers will implement
    Applications that would like greater protection on their data than
    the Internet checksum provides can use the Payload Checksum option.
    This option contains a CRC-32 checksum of the payload data only.
 4.4 ECN Support
    DCCP supports the use of Explicit Congestion Notification [ECN].  ECN
    allows routers to mark packets that have experienced congestion,
    rather than drop them.  This provides DCCP with the capability to
    adjust to network resources before packets are lost.  This capability
    is usually impossible to implement at the application layer, due to
    kernel restraints often imposed on setting and receiving ECN-marks in
 4.5 Slow Receiver Option
    The Slow Receiver Option is sent by a DCCP when the receiving
    application indicates that it is unable to keep up with the incoming
    data rate.  A DCCP that receives a Slow Receiver Option is required
    to not increase its sending rate for one RTT, and should indicate to
    its application that the option has been received.
    This perhaps can be used by applications as a lightweight
    application-level flow control.  If application data needs to be
    paced outside of congestion control, an overflow at the receiver can
    be communicated back with the Slow Receiver Option, instead of using
    an entire application-level message.  Upon receiving a Slow Receiver
    Option, the transmitting application can make whatever adjustment is
    necessary.  The application will of course be constrained by the
    requirement on DCCP to not allow the transmit rate to increase for
    one RTT.
 4.6 Detecting Lost Application Data
    In general, DCCP knows when packets are lost, but since data and
    acknowledgment packets use the same sequence number space, it can't
    tell whether or not application data has been lost.  Applications
    that want to detect data loss at the receiver could implement their
    own application-layer sequence numbers for this purpose, but the DCCP
    NDP Count Feature and Option could allow the application to push this
    to DCCP.
    When the NDP (stands for Non-Data Packets) Count Feature is in use,
    DCCP sends an NDP Count Option on every packet whose immediate
    predecessor was a non-data packet (mostly DCCP-Acks).  With the NDP
    Count Options, the receiving DCCP can determine whether or not a hole
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    in the received sequence numbers represents lost data.  The receiving
    DCCP could then inform the application that data was lost.
    Note that another use for application-layer sequence numbers is to
    reorder packets received out of transmission order.  However, as
    mentioned in section 3.3, the DCCP sequence numbers are adequate for
 5. Security Considerations
    DCCP includes several non-cryptographic security features designed to
    limit vulnerability to some common Denial of Service (DoS) and other
    attacks.  In UDP-based applications these capabilities would need to
    be implemented at the application layer, but are often ignored.
    To insert packets into a DCCP connection, an attacker must guess
    proper sequence numbers.  With randomly chosen initial sequence
    numbers, an attacker must snoop the connection to have any reasonable
    chance of success.  Other mechanisms (such as ignoring invalid DCCP-
    Move packets) prevent leakage of information to attackers.
    To hijack a connection (with DCCP-Move packets), an attacker must
    know the Mobility ID and Identification Option in use.  Again,
    without snooping the connection, there is little chance of guessing
    these accurately.
    The Init Cookie Option allows a server to delay holding state for a
    connection until the client has proved its aliveness.  Basically, the
    server responds to a DCCP-Request packet with a DCCP-Response packet
    that contains an Init Cookie Option.  This Init Cookie Option wraps
    up all information necessary for the connection to proceed in an
    encrypted and authenticated package.  After sending the DCCP-
    Response, the server needn't remember that a connection handshake is
    in progress.  The client responds to the DCCP-Response with a DCCP-
    Ack that includes the Init Cookie.  The server can then instantiate
    the necessary connection state.
 6. IANA Considerations
    There are no IANA actions required for this document.
 7. Thanks
    Thanks to Damon Lanphear for the original, API-centric, version of
    the user guide, the AVT working group, especially Philippe Gentric
    and Brian Rosen, for comments on the streaming-media-centric earlier
    version of this, Bart Whitebrook and Vladimir Moltchanov for
    information on game programming, and Jukka Manner for comments.
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 8. Informative References
    [DCCP]      E. Kohler, M. Handley, S. Floyd, J. Padhye, Datagram
                Congestion Control Protocol (DCCP), February 2004, draft-
                ietf-dccp-spec-06.txt, work in progress.
    [CCID2]     S. Floyd, E. Kohler, Profile for DCCP Congestion Control
                2: TCP-Like Congestion Control, February 2004, draft-
                ietf-dccp-ccid2-05.txt, work in progress.
    [CCID3]     S. Floyd, E. Kohler, J. Padhye, Profile for DCCP
                Congestion Control 3: TFRC Congestion Control, February
                2004, draft-ietf-dccp-ccid3-04.txt, work in progress.
    [RFC 3448]  M. Handley, S. Floyd, J. Padhye, J. Widmer, TCP Friendly
                Rate Control (TFRC): Protocol Specification, RFC 3448.
    [RFC 768]   J. Postel, User Datagram Protocol, August 1980, RFC 768.
    [IABCONG]   S. Floyd, J, Kempf, IAB Concerns Regarding Congestion for
                Voice Traffic in the Internet, October 2003, draft-iab-
                congestion-01.txt, work in progress.
    [SWITCH]    P. Gentric, RTSP Stream Switching, January 2004, draft-
                gentric-mmusic-stream-switching-01.txt, work in progress.
    [RFC 3261]  J. Rosenberg, et al, SIP: Session Initiation Protocol,
                June 2002, RFC 3261
    [EQCC]      S. Floyd, M. Handley, J. Padhye, J. Widmer, Equation-
                Based Congestion Control For Unicast Applications: the
                Extended Version, March 2000, International Computer
                Science Institute, http://www.icir.org/tfrc/tcp-
    [MMPGRID]   Butterfly.net: Powering Next-Generation Gaming with
                Computing On-Demand,
    [CONGAVOID] V. Jacobson, M. Karels, Congestion Avoidance and Control,
                November 1988, ftp://ftp.ee.lbl.gov/papers/congavoid.ps.Z
    [RFC 1911]  J. Mogul, S. Deering, Path MTU Discovery, February 1996,
                RFC 1911
    [RFC 3168]  K. Ramakrishnan, S. Floyd, D. Black, The addition of
                Explicit Congestion Notification (ECN) to IP, September
                2001, RFC 3168
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 INTERNET-DRAFT             DCCP User Guide                 April 2005
    [RFC 3517]  E. Blanton, M. Allman, K. Fall, L. Wang, A Conservative
                Selective Acknowledgment (SACK)-based Loss Recovery
                Algorithm for TCP, April 2003, RFC 3517
    [RFC 3550]  H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson,
                RTP: A Transport Protocol for Real-Time Applications,
                July 2003, RFC 3550
    [XTIME]     ITU-T: Series G: Transmission Systems and Media, Digital
                Systems and Networks, Recommendation G.114, One-way
                Transmission Time, May 2000
    [ECN]       K. Ramakrishnan, S. Floyd, D. Black, The Addition of
                Explicit Congestion Notification (ECN) to IP, September
                2001, RFC 3168
    [MPEG4]     ISO/IEC International Standard 14496 (MPEG-4),
                Information technology - Coding of audio-visual objects,
                January 2000
 9. Author's Address
    Tom Phelan
    Sonus Networks
    5 Carlisle Rd.
    Westford, MA USA 01886
    Phone: 978-681-8456
    Email: tphelan@sonusnet.com
 10. Full Copyright Statement
    Copyright ¨ The Internet Society (2004).  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
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