Internet Engineering Task Force
INTERNET-DRAFT                                               Sally Floyd
draft-floyd-dcp-problem-01.txt                              Mark Handley
                                                            Eddie Kohler
                                                          26 August 2002
                                                  Expires: February 2003

                       Problem Statement for DCCP

Status of this Document

    This document is an Internet-Draft and is in full conformance with
    all provisions of Section 10 of [RFC 2026].  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

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     This document gives the problem statement underlying the
     development of an unreliable transport protocol incorporating
     end-to-end congestion control.  This is also the problem
     statement underlying the development of DCCP, the Datagram
     Congestion Control Protocol. DCCP implements a congestion-
     controlled, unreliable flow of datagrams suitable for use by
     applications such as streaming media or on-line games.

Floyd/Handley/Kohler                                            [Page 1]

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                           Table of Contents

     1. Introduction. . . . . . . . . . . . . . . . . . . . . .   3
     2. Problem Space . . . . . . . . . . . . . . . . . . . . .   4
      2.1. Congestion Control for Unreliable Transfer . . . . .   4
      2.2. Overhead . . . . . . . . . . . . . . . . . . . . . .   6
      2.3. Firewall Traversal . . . . . . . . . . . . . . . . .   7
      2.4. Parameter Negotiation. . . . . . . . . . . . . . . .   7
     3. Solution Space for Congestion Control of Unreli-
     able Flows . . . . . . . . . . . . . . . . . . . . . . . .   7
      3.1. Providing Congestion Control Above UDP . . . . . . .   8
       3.1.1. The Burden on the Application Designer. . . . . .   8
       3.1.2. Difficulties with ECN . . . . . . . . . . . . . .   9
       3.1.3. The Evasion of Congestion Control . . . . . . . .  10
      3.2. Providing Congestion Control Below UDP . . . . . . .  10
       3.2.1. Case 1: Congestion Feedback at the Applica-
       tion . . . . . . . . . . . . . . . . . . . . . . . . . .  11
       3.2.2. Case 2: Congestion Feedback at a Layer
       below UDP. . . . . . . . . . . . . . . . . . . . . . . .  11
      3.3. Providing Congestion Control at the Transport
      Layer . . . . . . . . . . . . . . . . . . . . . . . . . .  11
       3.3.1. Modifying TCP?. . . . . . . . . . . . . . . . . .  12
       3.3.2. Unreliable Variants of SCTP?. . . . . . . . . . .  12
       3.3.3. Modifying RTP?. . . . . . . . . . . . . . . . . .  13
       3.3.4. Designing a New Transport Protocol. . . . . . . .  14
     4. Selling Congestion Control to Reluctant Applica-
     tions. . . . . . . . . . . . . . . . . . . . . . . . . . .  15
     5. Additional Design Considerations. . . . . . . . . . . .  15
     6. Summary of Recommendations. . . . . . . . . . . . . . .  16
     7. References. . . . . . . . . . . . . . . . . . . . . . .  17
     8. Authors' Addresses. . . . . . . . . . . . . . . . . . .  18

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1.  Introduction

    Historically, the great majority of Internet unicast traffic has
    used congestion-controlled TCP, with UDP making up most of the
    remainder. UDP has mainly been used for short, request-response
    transfers, like DNS and SNMP, that wish to avoid TCP's three-way
    handshake, retransmission, and/or stateful connections. UDP also
    avoids TCP's built-in end-to-end congestion control, and UDP
    applications tended not to implement their own congestion control.
    However, since UDP traffic volume was small relative to congestion-
    controlled TCP flows, the network didn't collapse.

    Recent years have seen the growth of applications that use UDP in a
    different way. These applications, including RealAudio, Internet
    telephony, and on-line games such as Half Life, Quake, and
    Starcraft, share a preference for timeliness over reliability. TCP
    can introduce arbitrary delay because of its reliability and in-
    order delivery requirements; thus, the applications use UDP instead.
    This growth of long-lived non-congestion-controlled traffic,
    relative to congestion-controlled traffic, poses a real threat to
    the overall health of the Internet.

    Applications could implement their own congestion control mechanisms
    on a case-by-case basis, with encouragement from the IETF. Some
    already do this. However, experience shows that congestion control
    is difficult to get right, and many application writers would like
    to avoid reinventing this particular wheel. We believe that a new
    protocol is needed, one that combines unreliable datagram delivery
    with built-in congestion control. This protocol would act as an
    enabling technology: existing and new applications could easily use
    it to transfer timely data without destabilizing the Internet.

    This document provides a problem statement for such a protocol. We
    list the properties the protocol should have, then explain why those
    properties are necessary. We also describe why a new protocol is the
    best solution for the more general problem of bringing congestion
    control to unreliable flows of unicast datagrams.

    This problem statement began as a formalization of the goals of
    DCCP, a proposed protocol already described in several Internet-
    Drafts. We intended DCCP to satisfy the problem statement laid out
    here. However, we believe the problem should be solved whether or
    not the chosen solution is DCCP, and the DCCP drafts should be
    judged based on this document or its successors.  Nevertheless, for
    convenience, we write "DCCP" to mean "any protocol that satisfies
    this problem statement".

Floyd/Handley/Kohler                                Section 1.  [Page 3]

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2.  Problem Space

    We perceive a number of problems related to the use of unreliable
    data flows in the Internet.  The major issues are:

    o The potential for non-congestion-controlled datagram flows to
      cause congestion collapse of the network.

    o The difficulty of correctly implementing effective congestion
      control mechanisms for unreliable datagram flows.

    o The lack of a standard solution for performing reliable congestion
      feedback for unreliable data flows.

    o The need for a standard solution for negotiating ECN usage for
      unreliable flows.

    o The need for applications to have a choice of TCP-friendly
      congestion control mechanisms.

    While undoubtedly some application writers would prefer not to
    perform congestion control at all for unreliable flows, we expect
    that most responsible application writers would use congestion
    control if it were available in a standard, easy to use form.

    More minor problems are:

    o The difficulty of deploying applications using UDP-based flows in
      the presence of firewalls.

    o The desire to have a single way to negotiate congestion control
      parameters for unreliable flows, independently of the signalling
      protocol used to set up the flow.

    o The desire for low per-packet byte overhead.

    The subsections below discuss these problems of providing congestion
    control, traversing firewalls, and negotiating parameters in more
    detail.  A separate subsection also discusses the problem of
    minimizing the overhead of packet headers.

2.1.  Congestion Control for Unreliable Transfer

    This project aims to bring easy-to-use congestion control mechanisms
    to applications that generate long-lived, large flows of unreliable
    datagrams, such as RealAudio, Internet telephony, and multiplayer
    games. Our motivation is to avoid congestion collapse. (Request-

Floyd/Handley/Kohler                              Section 2.1.  [Page 4]

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    response applications, such as DNS, SNMP, and SIP, are not our
    concern; their short flows don't cause congestion in practice, and
    any congestion control mechanism would take effect between flows,
    not within a single end-to-end transfer of information.) However,
    before designing a congestion control mechanism for these
    applications, we must understand why they use unreliable datagrams
    in the first place, lest we destroy the very properties they

    There are several reasons why protocols currently use UDP instead of
    TCP, amongst them:

    o Startup Delay: they wish to avoid the delay of a three-way

    o Statelessness: they wish to avoid holding connection state, and
      the potential state-holding attacks that come with this.

    o Trading of Reliability against Timing: the data being sent is
      timely in the sense that if it is not delivered by some deadline
      (typically a small number of RTTs) then the data will not be
      useful at the receiver.

    Of these, our target applications -- media transfer, games, and the
    like -- mostly care about having control over the tradeoff between
    timing and reliability. Such applications use UDP because when they
    send a datagram, they wish to send the most appropriate data in that
    datagram. If the datagram is lost, they may or may not resend the
    same data, depending on whether the data will still be useful at the
    receiver. Data may no longer be useful for many reasons:

    o In a telephony or streaming video session, data in a packet
      comprises a timeslice of a continuous stream.  Once a timeslice
      has been played out, the next timeslice is required immediately.
      If the data comprising that timeslice arrives at some later time,
      then it is no longer useful.  Such applications can cope with
      masking the effects of missing packets to some extent, so when the
      sender transmits its next packet, it is important for it to only
      send data that has a good chance of arriving in time for its

    o In a interactive game or VR session, position information is
      transient. If a datagram containing position information is lost,
      resending the old position does not usually make sense -- rather,
      every position information datagram should contain the latest
      position information.

Floyd/Handley/Kohler                              Section 2.1.  [Page 5]

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    In a congestion-controlled flow, the allowed packet sending rate
    depends on measured network congestion. Thus, applications must give
    up some control to the congestion control mechanism, which
    determines precisely when packets can be sent. However, applications
    could still decide, at transmission time, which information to put
    in a packet. TCP doesn't allow control over this; these applications
    demand it.

    Often, these applications (especially games and telephony
    applications) work on very short playout timescales.  Whilst they
    are usually able to adjust their transmission rate based on
    congestion feedback, they do have constraints on how this adaptation
    can be performed so that it has minimal impact on the quality of the
    session. Thus, they tend to need some control over the short-term
    dynamics of the congestion control algorithm, whilst being fair to
    other traffic on medium timescales. This control includes, but is
    not limited to, some influence on which congestion control algorithm
    should be used -- for example, TFRC rather than strict TCP-like
    congestion control.

2.2.  Overhead

    The applications we are concerned with often send compressed data,
    or send frequent small packets. For example, when internet telephony
    or streaming media are used over low-bandwidth modem links, highly
    compressing the payload data is essential. For internet telephony
    applications and for games, the requirement is for low delay, and
    hence small packets are sent frequently.

    For example, a telephony application sending a 20Kb/s packet stream
    wanting moderately low delay may send a packet every 20ms, and so
    each packet would be only 50 bytes.  Of this, 20 bytes is taken up
    by the IP header, leaving only 30 bytes for the transport header and
    payload.  Of course this is a relatively extreme example, but it
    serves to illustrate the degree to which some of these applications
    care that the transport protocol is low overhead.

    In some cases the correct solution would be to use link-based packet
    header compression to compress the packet headers, although we
    cannot guarantee the availability of such compression schemes on any
    particular link.

    The delay of data until after the completion of a handshake also
    represents potentially unnecessary overhead. A new protocol might
    therefore allow senders to include some data on their initial

Floyd/Handley/Kohler                              Section 2.2.  [Page 6]

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2.3.  Firewall Traversal

    Applications requiring a flow of unreliable datagrams currently tend
    to use signalling protocols such as RTSP, SIP and H.323 in
    conjunction with UDP for the data flow.  The initial setup request
    uses a signalling protocol to locate the correct remote end-system
    for the data flow, sometimes being redirected or relayed to other
    machines, before the data flow is established.

    As UDP flows contain no explicit setup and teardown, it is hard for
    firewalls to handle them correctly.  Typically the firewall needs to
    parse RTSP, SIP and H.323 to obtain the information necessary to
    open a hole in the firewall.  Alternatively, for bi-directional
    flows, the firewall can open a bi-directional hole if it receives a
    UDP packet from inside the firewall, but in this case the firewall
    can't easily know when to close the hole again.

    While we do not consider these to be major problems, they are
    nonetheless issues that application designers face.  Currently
    streaming media players such as RealPlayer attempt UDP first, and
    then switch to TCP if UDP is not successful.  Streaming media over
    TCP is undesirable, and can result in the receiver needing to
    temporarily halt playout while it "rebuffers" data.  Telephony
    applications don't even have this option.

2.4.  Parameter Negotiation

    Different applications have different requirements of congestion
    control, which may map into different congestion feedback.  Examples
    are the ECN capability, and desired congestion control dynamics,
    which in turn determines the choice of congestion control algorithm
    and the form of feedback information required.  Such parameters need
    to be reliably negotiated before congestion control can function

    While this negotiation could be performed using signalling protocols
    such as SIP, RTSP and H.323, it would be desirable to have a single
    standard way of negotiating these transport parameters.  This is of
    particular importance with ECN, where sending ECN-marked packets to
    a non-ECN-capable receiver can cause significant congestion problems
    to other flows.  We discuss the ECN issue in more detail below.

3.  Solution Space for Congestion Control of Unreliable Flows

    We thus want to provide congestion control for unreliable flows,
    providing both ECN and the choice of different forms of congestion
    control, and providing moderate overhead in terms of packet size,
    state, and CPU processing.  There are a number of options for

Floyd/Handley/Kohler                                Section 3.  [Page 7]

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    providing end-to-end congestion control for the unicast traffic that
    currently uses UDP, in terms of the layer that provides the
    congestion control mechanism:

    o Congestion control above UDP.

    o Congestion control below UDP.

    o Congestion control at the transport layer as an alternative to

    We explore these alternatives in the sections below.  The concerns
    from the discussions below have convinced us that the best way to
    provide congestion control for unreliable flows is to provide
    congestion control at the transport layer, as an alternative to the
    use of UDP and TCP.

3.1.  Providing Congestion Control Above UDP

    One possibility would be to provide congestion control at the
    application layer, or at some other layer above UDP.  Implementing
    congestion control at a layer above UDP would allow the congestion
    control mechanism to be closely integrated with the application

3.1.1.  The Burden on the Application Designer

    A key disadvantage of providing congestion control above UDP is that
    it places an unnecessary burden on the application-level designer,
    who might be just as happy to use the congestion control provided by
    a lower layer.  If the application can rely on a lower layer that
    gives a choice between TCP-like or TFRC-like congestion control, and
    that offers ECN, then this might be highly satisfactory to many
    application designers.

    The long history of debugging TCP implementations [RFC 2525] [TBIT]
    makes the difficulties in implementing end-to-end congestion control
    abundantly clear.  It is clearly more robust for congestion control
    to be provided for the application by a lower layer.  In rare cases
    there might be compelling reasons for the congestion control
    mechanism to be implemented in the application itself, but we do not
    expect this to be the general case.  For example, applications that
    use RTP over UDP might be just as happy if RTP itself implemented
    end-to-end congestion control.  (See Section 3.3.3 for more
    discussion of RTP.)

    In addition to congestion control issues, we also note the problems
    with firewall traversal and parameter negotiation discussed in

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    sections 2.3 and 2.4. Implementing on top of UDP requires that the
    application designer also address these issues.

3.1.2.  Difficulties with ECN

    A second problem of providing congestion control above UDP is that
    it would require either giving up the use of ECN, or giving the
    application direct control over setting and reading the ECN field in
    the IP header.  Giving up the use of ECN would be problematic, since
    ECN can be particularly useful for unreliable flows, where a dropped
    packet will not be retransmitted by the data sender.

    With the development of the ECN nonce, ECN can also be useful even
    in the absence of ECN support from the network.  The data sender can
    use the ECN nonce, along with feedback from the data receiver, to
    verify that the data receiver is correctly reporting all lost
    packets.  This use of ECN can be particularly useful for an
    application using unreliable delivery, where the receiver might
    otherwise have little incentive to report lost packets.

    In order to allow the use of ECN by a layer above UDP, the UDP
    socket would have to allow the application to control the ECN field
    in the IP header.  In particular, the UDP socket would have to allow
    the application to specify whether or not the ECN-Capable Transport
    (ECT) codepoints should be set in the ECN field of the IP header.

    The ECT codepoint can only safely be set in the packet header of a
    UDP packet if the following is guaranteed:

    o If the Congestion Experienced (CE) codepoint is set by a router,
      the receiving UDP will pass that CE status to the receiving
      application at the data receiver, and:

    o Upon receiving a packet that had the CE codepoint set, the
      receiving application will take the appropriate congestion control
      action, such as informing the data sender.

    However, the UDP implementation at the data sender has no way of
    knowing if the UDP implementation at the data receiver has been
    upgraded to pass a CE status up to the receiving application, let
    alone whether or not the application will use the conformant end-to-
    end congestion control that goes along with use of ECN.

    In the absence of the widespread deployment of mechanisms in routers
    to detect flows that are not using conformant congestion control,
    allowing applications arbitrary control of the ECT codepoints for
    UDP packets would seem like an unnecessary opportunity for
    applications to use ECN while evading the use of end-to-end

Floyd/Handley/Kohler                            Section 3.1.2.  [Page 9]

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    congestion control.  Thus, there is an inherent "chicken-and-egg"
    problem of whether first to deploy policing mechanisms in routers,
    or first to enable the use of ECN by UDP flows.  Without the
    policing mechanisms in routers, we would not advise adding ECN-
    capability to UDP sockets at this time.

3.1.3.  The Evasion of Congestion Control

    A third problem of providing congestion control above UDP is that
    relying on congestion control at the application level makes it
    somewhat easier for some users to evade end-to-end congestion
    control.  We do not claim that a transport protocol such as DCCP
    would always be implemented in the kernel, and do not attempt to
    evaluate the relative difficulty of modifying code inside the kernel
    vs. outside the kernel in any case.  However, we believe that
    putting the congestion control at the transport level rather than at
    the application level makes it just slightly less likely that users
    will go to the trouble of modifying the code in order to avoid using
    end-to-end congestion control.

3.2.  Providing Congestion Control Below UDP

    Instead of providing congestion control above UDP, a second
    possibility would be to provide congestion control for unreliable
    applications at a layer below UDP, and the applications would use
    UDP as their transport protocol.  Given that UDP does not itself
    provide sequence numbers or congestion feedback, there are two
    possible forms for this congestion feedback:

    o (1) Feedback at the application: The application above UDP could
      provide sequence numbers and feedback to the sender, which would
      then communicate loss information to the congestion control
      mechanism.  This is the approach currently standardized by the
      Congestion Manager [RFC 3124].

    o (2) Feedback at the layer below UDP: The application could use
      UDP, and a protocol could be implemented using a shim header
      between IP and UDP to provide sequence number information for data
      packets and return feedback to the data sender.  The original
      proposal for the Congestion Manager [Bala99] suggested providing
      this layer for applications that did not have their own feedback
      about dropped packets.

    We discuss these two cases separately below.

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3.2.1.  Case 1: Congestion Feedback at the Application

    In this case, the application provides sequence numbers and
    congestion feedback above UDP, but communicates that feedback to a
    congestion manager below UDP, which regulates when packets can be
    sent.  This approach suffers from most of the problems described in
    section 3.1, namely forcing the application designer to reinvent the
    wheel each time for packet formats and parameter negotiation, and
    problems with ECN usage, firewalls and evasion.

    It would avoid the application writer needing to implement the
    control part of the congestion control mechanism, but it is unclear
    how easily multiple congestion control algorithms (such as receiver-
    based TFRC) can be supported, given that the form of congestion
    feedback usually needs to be closely coupled to the congestion
    control algorithm being used.

3.2.2.  Case 2: Congestion Feedback at a Layer below UDP

    Providing feedback at a layer below UDP would require an additional
    packet header below UDP to carry sequence numbers in addition to the
    eight-byte header for UDP itself.  Unless this header were an IP
    option (which is likely to cause problems for many IPv4 routers)
    then its presence would need to be indicated using a different IP
    protocol value from UDP.  Thus, the packets would no longer look
    like UDP on the wire.

    To use this mechanism most effectively, the semantics of the UDP
    socket API (Application Programming Interface) would also need
    changing, both to support a late decision on what to send, and to
    provide access to the sequence numbers to avoid the application
    needing to duplicate them for its own purposes.  Thus, the socket
    API would no longer look like UDP in the end hosts.  This would
    effectively be a new transport protocol.

    Given these complications, it seems cleaner to actually design a new
    transport protocol, which also allows us to address the issues of
    firewall traversal, flow setup, and parameter negotiation.  We note
    that any new transport protocol could also use a Congestion Manager
    approach to share congestion state between flows using the same
    congestion control algorithm, if this were deemed to be desirable.

3.3.  Providing Congestion Control at the Transport Layer

    The concerns from the discussions above have convinced us that the
    best way to provide congestion control to applications that
    currently use UDP is to provide congestion control at the transport
    layer, in a transport protocol used as an alternative to UDP.  One

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    advantage of providing end-to-end congestion control in an
    unreliable transport protocol is that it could be used easily by a
    wide range of the applications that currently use UDP, with minimal
    changes to the application itself.  The application itself would not
    have to provide the congestion control mechanism, or even the
    feedback from the data receiver to the data sender about lost or
    marked packets.

    The question then arises of whether to adapt TCP for use by
    unreliable applications, to use an unreliable variant of SCTP or a
    version of RTP with built-in congestion control, or to design a new
    transport protocol.

    As we argue below, the desire for minimal overhead results in the
    design decision to use a transport protocol containing only the
    minimal necessary functionality, and to leave other functionality
    such as reliability, semi-reliability, or Forward Error Correction
    (FEC) to be layered on top.

3.3.1.  Modifying TCP?

    One alternative to be considered would be to create an unreliable
    variant of TCP, with the reliability layered on top for applications
    desiring reliable delivery.  However, our requirement is not simply
    for TCP minus the in-order reliable delivery, but also for the
    application to be able to choose congestion control algorithms.
    TCP's feedback mechanism works well for TCP-like congestion control,
    but is inappropriate (or at the very least, inefficient) for TFRC.
    In addition, TCP sequence numbers are in bytes, which is not optimal
    for an unreliable datagram protocol.  Finally, there is the issue of
    whether a modified TCP would require a new IP protocol number as
    well; a significantly modified TCP using the same IP protocol number
    could have unwanted interactions with some of the middleboxes
    already deployed in the network.

    In conclusion, it seems best simply to leave TCP as it is, and to
    create a new congestion control protocol for unreliable transfer.

3.3.2.  Unreliable Variants of SCTP?

    SCTP was in part designed to accommodate multiple streams within a
    single end-to-end connection, modifying TCP's semantics of reliable,
    in-order delivery to allow out-of-order delivery.  However, explicit
    support for multiple streams over a single flow at the transport
    layer is not necessary for an unreliable transport protocol such as
    DCCP, which of necessity allows out-of-order delivery.  Because an
    unreliable transport does not need streams support, applications
    should not have to pay the penalties in terms of increased header

Floyd/Handley/Kohler                           Section 3.3.2.  [Page 12]

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    size that accompany the use of streams in SCTP.

    The basic underlying structure of the SCTP packet, into a common
    SCTP header followed by chunks for data, SACK information, and so
    on, is motivated by SCTP's goal of accommodating multiple streams,
    However, this use of chunks comes at the cost of an increased header
    size for packets, as each chunk must be aligned on 32-bit
    boundaries, and therefore requires a fixed-size 4-byte chunk header.
    For example, for a connection using ECN, SCTP includes separate
    control chunks for the Explicit Congestion Notification Echo and
    Congestion Window Reduced functions, with the ECNE and CWR chunks
    each requiring 8 bytes.  As another example, the common header
    includes a 4-byte verification tag to validate the sender.  The
    issue of a minimal header size is just one of the ways that the
    underlying framework of SCTP does not fit the needs of an unreliable
    transport as outlined in this document.

    As a second concern, SCTP as currently specified uses TCP-like
    congestion control, and does not provide support for alternative
    congestion control algorithms such as TFRC that would be more
    attractive to some of the applications currently using UDP flows.
    Thus, the current version of SCTP would not meet the requirements
    for a choice between forms of end-to-end congestion control.

    One could suggest adding support for alternative congestion control
    mechanisms as an option to SCTP, and adding a fully-unreliable
    variant that does not include the mechanisms for multiple streams.
    However, that path leads one to a choice between a ``kitchen sink''
    protocol that tries to include options to accommodate all
    applications (i.e., the modified version of SCTP), and a clean and
    minimal protocol that provides only end-to-end congestion control
    and any other mechanisms that cannot be provided in a higher layer.
    We would argue against the ``kitchen sink'' approach in this case.

    Applications that desire limited retransmission with multi-stream
    support, or that desire multi-homing support, might be good
    candidates for a semi-reliable version of SCTP such as U-SCTP.
    However, we believe that variants of SCTP are not suitable for
    fully-unreliable applications, or for the bulk of applications that
    today use UDP.  We would argue that instead, an additional transport
    protocol is needed for unreliable transfer.

3.3.3.  Modifying RTP?

    Several of our target applications currently use RTP layered above
    UDP to transfer their data. Why not modify RTP to provide end-to-end
    congestion control?

Floyd/Handley/Kohler                           Section 3.3.3.  [Page 13]

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    As RTP lives above UDP, modifying it to support congestion control
    might encounter some of the problems described in Section 3.1. In
    particular, user-level RTP implementations would want access to ECN
    bits in UDP datagrams. It might be difficult or undesirable to allow
    that access for RTP, but not for other user-level programs.

    Kernel implementations of RTP would not suffer from this problem. In
    the end, the argument against modifying RTP is the same as that
    against modifying SCTP: Some applications, such as the export of
    flow information from routers, need congestion control but don't
    need much of RTP's functionality. From these applications' point of
    view, RTP would induce unnecessary overhead. Again, we would argue
    for a clean and minimal protocol focused on end-to-end congestion

    RTP would commonly be used as a layer above any new transport
    protocol, however. The design of that new transport protocol should
    take this into account, either by avoiding undue duplication of
    information available in the RTP header, or by suggesting
    modifications to RTP.

3.3.4.  Designing a New Transport Protocol

    In the first half of this document we have argued for providing
    congestion control at the transport layer as an alternative to UDP,
    instead of relying on congestion control supplied only above or
    below UDP.  In this section, we have examined the possibilities of
    modifying SCTP, modifying TCP, and designing a new transport
    protocol.  In large part because of the requirement for unreliable
    transport, and for accommodating TFRC and well as TCP-like
    congestion control, we have concluded that modifications of SCTP or
    TCP are not the best answer, and that a new transport protocol is
    needed.  Thus, we have argued for the need for a new transport
    protocol that offers unreliable delivery; accommodates TFRC as well
    as TCP-like congestion control; accommodates the use of ECN; and
    requires minimal overhead in packet size and in the state and CPU
    processing required at the data receiver.

    This is the line of reasoning that has lead us to the development of
    DCCP.  Both the problem statement presented in this document, and
    the detailed design decisions made in the design of DCCP, have been
    brought to the IETF for further feedback.  This document addresses
    only the problem statement; the design decisions resulting from this
    problem statement will be addressed in a later document.

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4.  Selling Congestion Control to Reluctant Applications

    The goal of this work is to provide general congestion control
    mechanisms that will actually be used by many of the applications
    that currently use UDP.  This may include applications that are
    perfectly happy without end-to-end congestion control.  Several of
    our design requirements follow from a desire to design and deploy a
    congestion-controlled protocol that is actually attractive to these
    "reluctant" applications.  These include the use of Explicit
    Congestion Notification (ECN) and the ECN Nonce, which both provide
    positive benefit to the application itself; the choice between
    different forms of congestion control; and moderate overhead in the
    size of the packet header.

    There will always be a few flows that are resistant to the use of
    end-to-end congestion control, preferring an environment where end-
    to-end congestion control is used by everyone else, but not by
    themselves.  There has been substantial agreement [RFC 2309] [FF99]
    that in order to maintain the continued use of end-to-end congestion
    control, router mechanisms are needed to detect and penalize
    uncontrolled high-bandwidth flows in times of high congestion; these
    router mechanisms are colloquially known as `penalty boxes'.
    However, before undertaking a concerted effort towards the
    deployment of penalty boxes in the Internet, it seems reasonable,
    and more effective, to first make a concerted effort to make end-to-
    end congestion control easily available to applications currently
    using UDP.

5.  Additional Design Considerations

    This section mentions some additional design considerations that
    should be considered in designing a new transport protocol.

    o Mobility: Mechanisms for multi-homing and mobility are one area of
      additional functionality that cannot necessarily be layered
      cleanly and effectively on top of a transport protocol.  Thus, one
      outstanding design decision with any new transport protocol
      concerns whether to incorporate mechanisms for multi-homing and
      mobility into the protocol itself.

    o Defense against DoS attacks and spoofing: A reliable handshake for
      connection setup and teardown offers protection against DoS and
      spoofing attacks.  Mechanisms allowing a server to avoid holding
      any state for unacknowledged connection attempts or already-
      finished connections offers additional protection against DoS
      attacks.  Thus, in designed a new transport protocol, even one
      designed to provide minimal functionality, the requirements for
      providing defense against DoS attacks and spoofing need to be

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    o Interoperation with RTP: As Section 3.3.3 describes, attention
      should be paid to the changes that would be required by RTP in
      order to use any new protocol.

    o API: Some functionality required by the protocol, or useful for
      applications using the protocol, may require the definition of new
      API mechanisms.  Examples include allowing applications to decide
      what information to put in a packet  at transmission timer, and
      providing applications with some information about packet sequence

    o Consider general experiences with unicast transport: A
      Requirements for Unicast Transport/Sessions (ruts) BOF was held at
      the IETF meeting in December, 1998, with the goal of understanding
      the requirements of applications whose needs were not met by TCP
      [RUTS]. Not all of those unmet needs are relevant to or
      appropriate for a unicast, congestion-controlled, unreliable flow
      of datagrams designed for long-lived transfers. Some are, however,
      and any new protocol should address those needs, and other
      requirements derived from the community's experience. We believe
      that the DCCP problem statement addresses the relevant
      requirements brought up at the ruts BOF.

6.  Summary of Recommendations

    Our problem statement has discussed the need for implementing
    congestion control for unreliable flows.  Additional problems
    concern the need for low overhead, the problems of firewall
    traversal, and the need for reliable parameter negotiation.  Our
    consideration of the problem statement has resulted in the following
    general recommendations:

    o A unicast transport protocol for unreliable datagrams should be
      developed, including mandatory, built-in congestion control.

    o The protocol must provide a set of congestion control mechanisms
      from which the application may choose. These mechanisms should
      include, at minimum, TCP-like congestion control and a more
      slowly-responding congestion control such as TFRC.

    o Important features of the connection, such as the congestion
      control mechanism in use, should be reliably negotiated by both

    o Support for ECN should be included.

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    o The overhead must be low, in terms of both packet size and
      protocol complexity.

    o Some DoS protection for servers must be included. In particular,
      servers can make themselves invulnerable to spoofed connection
      attacks ("SYN floods").

    o Connection setup and teardown must use explicit handshakes,
      facilitating transmission through stateful firewalls.

    If there is a consensus about the need for a new unicast transport
    protocol for unreliable datagrams, then the next step can be the
    consideration of more detailed architectural issues.

7.  References

    [Bala99] H. Balakrishnan, H. Rahul, and S. Seshan, An Integrated
        Congestion Management Architecture for Internet Hosts, SIGCOMM,
        Sept. 1999.

    [MEASWEB] Ramon Caceres and Sally Floyd, Measurement Studies of End-
        to-End Congestion Control in the Internet, Web Page, 2001.

    [FF99] S. Floyd and K. Fall, Promoting the Use of End-to-End
        Congestion Control in the Internet, IEEE/ACM Transactions on
        Networking, August 1999.

    [MC01] S. McCreary and K.C. Claffy, Trends in Wide Area IP Traffic
        Patterns: A View from Ames Internet Exchange, ITC Specialist
        Seminar, 2001.  URL

    [RFC 2026] S. Bradner. The Internet Standards Process -- Revision 3.
        RFC 2026.

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

    [RFC 1191] J. C. Mogul, S. E. Deering. Path MTU Discovery. RFC 1191.

    [RFC 2309] B. Braden et al., Recommendations on Queue Management and
        Congestion Avoidance in the Internet.  RFC 2309, April 1998.

    [RFC 2525] V. Paxson et al., Known TCP Implementation Problems, RFC
        2525, March 1999.

    [RFC 2914] S. Floyd. Congestion Control Principles.  RFC 2914, Sept.

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    [RFC 2960] R. Stewart, Q. Xie, K. Morneault, C. Sharp, H.
        Schwarzbauer, T. Taylor, I.  Rytina, M. Kalla, L. Zhang, V.
        Paxson. Stream Control Transmission Protocol. RFC 2960.

    [RFC 3124] H. Balakrishnan, S. Seshan. The Congestion Manager. RFC

    [RUTS] Requirements for Unicast Transport/Sessions (ruts) BOF, Dec.
        7, 1998.  URL "

    [TBIT] J. Padhye and S. Floyd, Identifying the TCP Behavior of Web
        Servers, SIGCOMM 2001.

    [WES01] David Wetherall, David Ely, Neil Spring. Robust ECN
        Signaling with Nonces.  draft-ietf-tsvwg-tcp-nonce-03.txt, work
        in progress, April 2002.

8.  Authors' Addresses

    Sally Floyd <>
    Mark Handley <>
    Eddie Kohler <>

    ICSI Center for Internet Research (ICIR),
    International Computer Science Institute,
    1947 Center Street, Suite 600
    Berkeley, CA 94704.

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